From dmitry.bely at gmail.com  Wed Dec  1 00:59:29 2010
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Wed, 1 Dec 2010 11:59:29 +0300
Subject: [Freeswitch-users] FsGui Windows build
In-Reply-To: <1291133351932-5788649.post@n2.nabble.com>
References: <AANLkTin4hT0yUkr2gPd3bhq+NWTGmbhk3229-kFTDMOs@mail.gmail.com>
	<1291133351932-5788649.post@n2.nabble.com>
Message-ID: <AANLkTi=-qLk4iy-J5JkPCqzYG1FKq_0OAjjwzOmXVBPQ@mail.gmail.com>

On Tue, Nov 30, 2010 at 7:09 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
>
> Look at the current build of FSComm for windows for an example. It currently
> builds fine in Git Head with VS2008

Well, finally I have it compiling and working (don't see how FSComm
could help). I'm no expert in qmake so I just fixed qmake-generated
makefiles manually to make Microsoft's nmake happy. Now CPU 100% load
problem is over.

- Dmitry Bely


From abubacker at bksystems.co.in  Wed Dec  1 02:23:15 2010
From: abubacker at bksystems.co.in (abubacker)
Date: Wed, 01 Dec 2010 15:53:15 +0530
Subject: [Freeswitch-users] mod_fifo or mod_callcenter
Message-ID: <4CF62213.2000707@bksys.co.in>

Dear all,
      I want to use mod_fifo or mod_callcenter to perform queuing 
operation , but I dont want this application to
connect the agent and the customer I have the external script to do that 
but it should give me the dial string of the
agent where the customer is likely to  connect with.

Is this possible or can we do this using some work around ?
Thanks in advance !

-- 
Best regards,
N.Abubacker ,
Associate system engineer ,
bk systems pvt ltd ,
Ph : 9144-43902701

Disclaimer: http://www.bksystems.co.in/email-policy



From kornev at ural.ru  Wed Dec  1 02:47:26 2010
From: kornev at ural.ru (korn)
Date: Wed, 1 Dec 2010 15:47:26 +0500
Subject: [Freeswitch-users] Cant compile last git on FreeBSD
In-Reply-To: <AANLkTikKL9f_O=Np0D2iw1weDsT-FDduBoMBXOd+Aq0M@mail.gmail.com>
References: <201011231704.01945.kornev@ural.ru>
	<AANLkTikKL9f_O=Np0D2iw1weDsT-FDduBoMBXOd+Aq0M@mail.gmail.com>
Message-ID: <201012011547.26772.kornev@ural.ru>

Thanks Michael, I have made fresh checkout at 29.11, and there is no changes

> Is this a fresh checkout or no? If not, bootstrap.sh and configure again
> and then make all.
> -MC
>
> On Tue, Nov 23, 2010 at 4:04 AM, korn <kornev at ural.ru> wrote:
> > Hello, I,m trying to compile last git on FreeBSD 7.1 and receive this
> > error in
> > mod_spandsp:
> >
> > making all mod_spandsp
> > Creating mod_spandsp_la-mod_spandsp_fax.lo
> > Compiling mod_spandsp_fax.c ...
> > cc1: warnings being treated as errors
> > mod_spandsp_fax.c: In function 'configure_t38':
> > mod_spandsp_fax.c:721: warning: implicit declaration of function
> > 't38_set_fastest_image_data_rate'
> > mod_spandsp_fax.c: In function 't38_gateway_on_consume_media':
> > mod_spandsp_fax.c:1637: warning: implicit declaration of function
> > 't38_gateway_rx_fillin'
> > gmake[4]: *** [mod_spandsp_la-mod_spandsp_fax.lo] Error 1
> > gmake[3]: *** [mod_spandsp-all] Error 1
> > gmake[2]: *** [all-recursive] Error 1
> >
> >
> > configure and gmake in ./libs/spandsp are successful
> >
> > can anybody tell me, why I'm getting error
> >
> > WBR
> > Evgeny
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org



From a.afzali2003 at gmail.com  Wed Dec  1 02:50:32 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Wed, 1 Dec 2010 14:20:32 +0330
Subject: [Freeswitch-users] mod_fifo or mod_callcenter
In-Reply-To: <4CF62213.2000707@bksys.co.in>
References: <4CF62213.2000707@bksys.co.in>
Message-ID: <AANLkTikfc6RPG+BanLUkCr_46pc2hWvuJsxR13R166_L@mail.gmail.com>

Abubacker,

I think you can use loopback endpoint in your dial-string to go through an
extension. Look at the "Another example of On-hook Agent Login/Logout with
loopback members"
in wiki.

-- afshin

On Wed, Dec 1, 2010 at 1:53 PM, abubacker <abubacker at bksystems.co.in> wrote:

> Dear all,
>      I want to use mod_fifo or mod_callcenter to perform queuing
> operation , but I dont want this application to
> connect the agent and the customer I have the external script to do that
> but it should give me the dial string of the
> agent where the customer is likely to  connect with.
>
> Is this possible or can we do this using some work around ?
> Thanks in advance !
>
> --
> Best regards,
> N.Abubacker ,
> Associate system engineer ,
> bk systems pvt ltd ,
> Ph : 9144-43902701
>
> Disclaimer: http://www.bksystems.co.in/email-policy
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From erik.dekkers at wvds.nl  Wed Dec  1 02:58:08 2010
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Wed, 1 Dec 2010 11:58:08 +0100
Subject: [Freeswitch-users] Cant compile last git on FreeBSD
In-Reply-To: <201012011547.26772.kornev@ural.ru>
References: <201011231704.01945.kornev@ural.ru>
	<AANLkTikKL9f_O=Np0D2iw1weDsT-FDduBoMBXOd+Aq0M@mail.gmail.com>
	<201012011547.26772.kornev@ural.ru>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE0772CBE909@FS1.wvds.local>

Already tried http://wiki.freeswitch.org/wiki/Installation_Guide#FreeBSD with first installing libtiff?

I've compiled yesterday on Freebsd 8.1 with no problems.

-----Oorspronkelijk bericht-----
Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens korn
Verzonden: woensdag 1 december 2010 11:47
Aan: FreeSWITCH Users Help
Onderwerp: Re: [Freeswitch-users] Cant compile last git on FreeBSD

Thanks Michael, I have made fresh checkout at 29.11, and there is no changes

> Is this a fresh checkout or no? If not, bootstrap.sh and configure 
> again and then make all.
> -MC
>
> On Tue, Nov 23, 2010 at 4:04 AM, korn <kornev at ural.ru> wrote:
> > Hello, I,m trying to compile last git on FreeBSD 7.1 and receive 
> > this error in
> > mod_spandsp:
> >
> > making all mod_spandsp
> > Creating mod_spandsp_la-mod_spandsp_fax.lo Compiling 
> > mod_spandsp_fax.c ...
> > cc1: warnings being treated as errors
> > mod_spandsp_fax.c: In function 'configure_t38':
> > mod_spandsp_fax.c:721: warning: implicit declaration of function 
> > 't38_set_fastest_image_data_rate'
> > mod_spandsp_fax.c: In function 't38_gateway_on_consume_media':
> > mod_spandsp_fax.c:1637: warning: implicit declaration of function 
> > 't38_gateway_rx_fillin'
> > gmake[4]: *** [mod_spandsp_la-mod_spandsp_fax.lo] Error 1
> > gmake[3]: *** [mod_spandsp-all] Error 1
> > gmake[2]: *** [all-recursive] Error 1
> >
> >
> > configure and gmake in ./libs/spandsp are successful
> >
> > can anybody tell me, why I'm getting error
> >
> > WBR
> > Evgeny
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u
> > sers
> > http://www.freeswitch.org


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


From kornev at ural.ru  Wed Dec  1 03:31:45 2010
From: kornev at ural.ru (korn)
Date: Wed, 1 Dec 2010 16:31:45 +0500
Subject: [Freeswitch-users] Cant compile last git on FreeBSD
In-Reply-To: <CDF5E00590C9BA4AB96BE7CDD28C86AE0772CBE909@FS1.wvds.local>
References: <201011231704.01945.kornev@ural.ru>
	<201012011547.26772.kornev@ural.ru>
	<CDF5E00590C9BA4AB96BE7CDD28C86AE0772CBE909@FS1.wvds.local>
Message-ID: <201012011631.46248.kornev@ural.ru>


server# ll /usr/local/lib/ |grep tiff
-rw-r--r--  1 root  wheel   405936 Jun 29  2009 libtiff.a
-rwxr-xr-x  1 root  wheel      837 Jun 29  2009 libtiff.la
lrwxr-xr-x  1 root  wheel       12 Jun 29  2009 libtiff.so -> libtiff.so.4
-rwxr-xr-x  1 root  wheel   372497 Jun 29  2009 libtiff.so.4
-rw-r--r--  1 root  wheel     6384 Jun 29  2009 libtiffxx.a
-rwxr-xr-x  1 root  wheel      873 Jun 29  2009 libtiffxx.la
lrwxr-xr-x  1 root  wheel       14 Jun 29  2009 libtiffxx.so -> libtiffxx.so.4
-rwxr-xr-x  1 root  wheel    11331 Jun 29  2009 libtiffxx.so.4
server#
server# ll /usr/local/include/ | grep tiff
-r--r--r--  1 root  wheel  33725 Jun 29  2009 tiff.h
-r--r--r--  1 root  wheel   2867 Jun 29  2009 tiffconf.h
-r--r--r--  1 root  wheel  19711 Jun 29  2009 tiffio.h
-r--r--r--  1 root  wheel   1610 Jun 29  2009 tiffio.hxx
-r--r--r--  1 root  wheel    410 Jun 29  2009 tiffvers.h

and configure and gmake in ./libs/spandsp are successful

> Already tried http://wiki.freeswitch.org/wiki/Installation_Guide#FreeBSD
> with first installing libtiff?
>
> I've compiled yesterday on Freebsd 8.1 with no problems.
>
> -----Oorspronkelijk bericht-----
> Van: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens korn
> Verzonden: woensdag 1 december 2010 11:47
> Aan: FreeSWITCH Users Help
> Onderwerp: Re: [Freeswitch-users] Cant compile last git on FreeBSD
>
> Thanks Michael, I have made fresh checkout at 29.11, and there is no
> changes
>
> > Is this a fresh checkout or no? If not, bootstrap.sh and configure
> > again and then make all.
> > -MC
> >
> > On Tue, Nov 23, 2010 at 4:04 AM, korn <kornev at ural.ru> wrote:
> > > Hello, I,m trying to compile last git on FreeBSD 7.1 and receive
> > > this error in
> > > mod_spandsp:
> > >
> > > making all mod_spandsp
> > > Creating mod_spandsp_la-mod_spandsp_fax.lo Compiling
> > > mod_spandsp_fax.c ...
> > > cc1: warnings being treated as errors
> > > mod_spandsp_fax.c: In function 'configure_t38':
> > > mod_spandsp_fax.c:721: warning: implicit declaration of function
> > > 't38_set_fastest_image_data_rate'
> > > mod_spandsp_fax.c: In function 't38_gateway_on_consume_media':
> > > mod_spandsp_fax.c:1637: warning: implicit declaration of function
> > > 't38_gateway_rx_fillin'
> > > gmake[4]: *** [mod_spandsp_la-mod_spandsp_fax.lo] Error 1
> > > gmake[3]: *** [mod_spandsp-all] Error 1
> > > gmake[2]: *** [all-recursive] Error 1
> > >
> > >
> > > configure and gmake in ./libs/spandsp are successful
> > >
> > > can anybody tell me, why I'm getting error
> > >
> > > WBR
> > > Evgeny
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u
> > > sers
> > > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From dujinfang at gmail.com  Wed Dec  1 05:51:52 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 1 Dec 2010 21:51:52 +0800
Subject: [Freeswitch-users] dead call records
Message-ID: <AANLkTimhxMGBzZkcpO1GzvPDYUE6empmE1NxAXrjm4d5@mail.gmail.com>

Hi,

I found a wired problem where left dead call records.

originate {origination_uuid=1,hangup_after_bridge=false}user/1000 &park
originate {origination_uuid=2,hangup_after_bridge=false}user/1001 &park
uuid_bridge 1 2
uuid_transfer 1 -both park inline
hupall

show channels # show nothing
show calls # the record still there, why the uuid is not 1 and 2 but a
long uuid str?

but there's no problem if not use origination_uuid.

Also, there are no problems if no uuid_transfer.

originate {origination_uuid=1,hangup_after_bridge=false}user/1000 &park
originate {origination_uuid=2,hangup_after_bridge=false}user/1001 &park
uuid_bridge 1 2
hupall


I'm on last git, anyone can help take a look?

http://pastebin.freeswitch.org/14681

Thanks.

-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From babak.freeswitch at gmail.com  Wed Dec  1 06:53:27 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Wed, 1 Dec 2010 18:23:27 +0330
Subject: [Freeswitch-users] building mod_managed fails
In-Reply-To: <AANLkTi=xLHN7SAL9t0A+aiDtGTwYoCDwqNFjMapSGQpu@mail.gmail.com>
References: <AANLkTi=96X=WyXFZ1ZTU8AE3fKiZUm6Q=tr-ds4kOCWa@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703681DD6C9@mse17be1.mse17.exchange.ms>
	<AANLkTi=xLHN7SAL9t0A+aiDtGTwYoCDwqNFjMapSGQpu@mail.gmail.com>
Message-ID: <AANLkTimWZYJJEyp9Ayu-Ou3asR0kqbZ2EcjVq8mLAk_c@mail.gmail.com>

Hi Micheal
I've attached errors from building with mono 2.6.7
thanx
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From renjian at gmail.com  Wed Dec  1 07:03:13 2010
From: renjian at gmail.com (Jian Ren)
Date: Wed, 1 Dec 2010 10:03:13 -0500
Subject: [Freeswitch-users] : Skypopen error
In-Reply-To: <AANLkTimgOsmXR6jC8pOhB7URwdn19H91ZtkRC9sfBasU@mail.gmail.com>
References: <AANLkTi=W65whZv0f7p6GxfjJWmsG9=OXSs-Tt3jmG1tX@mail.gmail.com>
	<AANLkTi=2f9B2upJx8kg64K4NbRGaiw9fOcULET=4THoM@mail.gmail.com>
	<AANLkTinBeEcX0-6gffYsWaJLNU8j8R+t31hZwMbzX7R+@mail.gmail.com>
	<AANLkTinYqCUDiK=hLKPUJ9Q04VUkGgyHE_BKujasd4Cg@mail.gmail.com>
	<AANLkTiks99vbGmG72fheTRb2QdCkQqQQ01gFZRsjxT2h@mail.gmail.com>
	<AANLkTimcYMYuV1pxo60JhQkfK0+P94sf=b_Er8FJrwP_@mail.gmail.com>
	<AANLkTimcCSMfyaajth4qQBiyDGDfhVpPCJyeVt9-VxdF@mail.gmail.com>
	<AANLkTimgOsmXR6jC8pOhB7URwdn19H91ZtkRC9sfBasU@mail.gmail.com>
Message-ID: <AANLkTinks_EtXZ-y7a=Xux80XrJey1dAoOr8hnJT64aS@mail.gmail.com>

Hi Giovanni,
Could you explain more to a bizarre person like me :-)
What's DTMF "D"? Sent from one of my SIP client? Or Someone sent to my skype
through chat? If it's abnormal, I think I should fix the root cause.
Thank you very much for looking at the issue, with 2M logs :-).
Jian

On Tue, Nov 30, 2010 at 7:15 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> Jian,
>
> you're very very bizarre :)
>
> Who is that is sending to you DTMF "D"????
>
> First time I see this.
>
> Normally only 0-9 * and # are used.
>
> Good to know, no need to do the other debug log.
>
> Skype client does not accept DTMF "A-D"
>
> I'll fix that in the code, no problem.
>
> Just for my curiosity: how is happening someone is sending you the "D"
> DTMF?
>
> -giovanni
>
> On Wed, Dec 1, 2010 at 12:54 AM, Jian Ren <renjian at gmail.com> wrote:
> > Hi Giovanni,
> > I got it happened with loglevel set to 9 on CentOS and created an issue
> as
> > below, switched to Ubuntu, will add the Ubuntu log once it happened.
> Sorry
> > that I can only do it one by one.
> > http://jira.freeswitch.org/browse/FS-2891
> > Regards!
> > Jian
> >
> > On Fri, Nov 19, 2010 at 2:15 PM, Giovanni Maruzzelli <
> gmaruzz at celliax.org>
> > wrote:
> >>
> >> Jian,
> >> I repeat:
> >> post a full debug session.
> >> Do like this:
> >> from the console
> >> "fsctl loglevel 7"
> >> then use the system, and when you encounter the problem post on
> >> pastebin.freeswitch.org a relevant part of the log you find in
> >> /usr/local/freeswitch/log/freeswitch.log
> >>
> >> Not only the line with the problem, at least 300 line before
> >>
> >> -giovanni
> >>
> >> On Fri, Nov 19, 2010 at 7:38 PM, Jian Ren <renjian at gmail.com> wrote:
> >> > You are right, I know nothing about this system. The reason I post
> here
> >> > is
> >> > to expect there are smart people, or at least people not as stupid as
> me
> >> > to
> >> > help. As I said, so far I haven't figured out how to reproduce this
> >> > consistently, which would be the first required step to report this as
> >> > an
> >> > issue, according to your reply. I thought the error number or line
> >> > number
> >> > could remind people something  but they didn't seem to, which is fine.
> >> > It's a free product and has already been working at certain level,
> thank
> >> > you
> >> > for creating this.
> >> > Jian
> >> >
> >> >
> >> > On Fri, Nov 19, 2010 at 8:19 AM, Giovanni Maruzzelli
> >> > <gmaruzz at celliax.org>
> >> > wrote:
> >> >>
> >> >> On Fri, Nov 19, 2010 at 2:12 PM, Jian Ren <renjian at gmail.com> wrote:
> >> >> > I still don't know how to reproduce. Since there are error number,
> >> >> > and
> >> >> > even
> >> >> > the line number of the source code, I am wondering the author or
> >> >> > whoever
> >> >> > knows this part well could help.
> >> >> > It's the same environment you helped me on, Ubuntu 8.04 server
> 64bit,
> >> >> > skype
> >> >> > 2.0.0.72, snd-dummy built according to the wiki. 4 skype instances
> >> >> > with
> >> >> > same
> >> >> > user account launched. The only extra mod I added is dingaling to
> >> >> > connect my
> >> >> > GV account.
> >> >>
> >> >> Jian,
> >> >>
> >> >> I am the author.
> >> >>
> >> >> And I am a little annoyed by your unwillingness to cooperate, but lot
> >> >> of will to whine.
> >> >>
> >> >> Also, you don't understand how things works, but you are sure the
> >> >> people is asking for more info, and that tells you how to provide
> that
> >> >> info, are somewhat stupid, heh?
> >> >>
> >> >>
> >> >> -giovanni
> >> >>
> >> >>
> >> >>
> >> >> > Thanks!
> >> >> > Jian
> >> >> >
> >> >> > On Fri, Nov 19, 2010 at 2:10 AM, Giovanni Maruzzelli
> >> >> > <gmaruzz at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >> Jian
> >> >> >> you have an ah, so cryptic way to communicate.
> >> >> >> How can your question be answered?
> >> >> >> There are instructions in the skypopen wiki page on how to report
> a
> >> >> >> problem.
> >> >> >> Please follow those instructions and give us all the infos we need
> >> >> >> for
> >> >> >> answering you.
> >> >> >> Particularly, how to replicate the problem, a procedure to obtain
> >> >> >> the
> >> >> >> same problem/issue, andd what are you doing when the problem shows
> >> >> >> up.
> >> >> >> Also, version of skype client, operating system, machine,
> freeswitch
> >> >> >> version, etc etc etc etc
> >> >> >> -giovanni
> >> >> >>
> >> >> >> On 11/19/10, Jian Ren <renjian at gmail.com> wrote:
> >> >> >> > Could anyone tell me what's this error about? It happened to me
> >> >> >> > quite
> >> >> >> > often.
> >> >> >> >
> >> >> >> > 2010-11-18 20:45:52.690158 [ERR] skypopen_protocol.c:259
>  [|]
> >> >> >> > [ERRORA
> >> >> >> >     259  ][interface00    ][UP,INPROGRS] Skype got ERROR:
> |||ERROR
> >> >> >> > 21
> >> >> >> > Unknown/disallowed call prop|||
> >> >> >> > 2010-11-18 20:45:52.690158 [ERR] skypopen_protocol.c:261
>  [|]
> >> >> >> > [ERRORA
> >> >> >> >     261  ][interface00    ][UP,FNSHED] skype_call now is DOWN
> >> >> >> >
> >> >> >> > Thanks!
> >> >> >> > Jian
> >> >> >> >
> >> >> >>
> >> >> >> --
> >> >> >> Sent from my mobile device
> >> >> >>
> >> >> >> Sincerely,
> >> >> >>
> >> >> >> Giovanni Maruzzelli
> >> >> >> Cell : +39-347-2665618
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >>
> >> >>
> >> >> --
> >> >> Sincerely,
> >> >>
> >> >> Giovanni Maruzzelli
> >> >> Cell : +39-347-2665618
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch-list at puzzled.xs4all.nl  Wed Dec  1 07:41:51 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Wed, 01 Dec 2010 16:41:51 +0100
Subject: [Freeswitch-users] building mod_managed fails
In-Reply-To: <AANLkTimWZYJJEyp9Ayu-Ou3asR0kqbZ2EcjVq8mLAk_c@mail.gmail.com>
References: <AANLkTi=96X=WyXFZ1ZTU8AE3fKiZUm6Q=tr-ds4kOCWa@mail.gmail.com>	<6E8D2069C08AA84A83D336E996AE4C6703681DD6C9@mse17be1.mse17.exchange.ms>	<AANLkTi=xLHN7SAL9t0A+aiDtGTwYoCDwqNFjMapSGQpu@mail.gmail.com>
	<AANLkTimWZYJJEyp9Ayu-Ou3asR0kqbZ2EcjVq8mLAk_c@mail.gmail.com>
Message-ID: <4CF66CBF.9080308@puzzled.xs4all.nl>

On 12/01/2010 03:53 PM, babak yakhchali wrote:
> Hi Micheal
> I've attached errors from building with mono 2.6.7

Did you actually read the error or did you just attach it?
Your answer is on line 1 and 2 of the error file. Clearly you are 
missing some header files.

Patrick


From freeswitch-list at puzzled.xs4all.nl  Wed Dec  1 07:45:52 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Wed, 01 Dec 2010 16:45:52 +0100
Subject: [Freeswitch-users] : Skypopen error
In-Reply-To: <AANLkTinks_EtXZ-y7a=Xux80XrJey1dAoOr8hnJT64aS@mail.gmail.com>
References: <AANLkTi=W65whZv0f7p6GxfjJWmsG9=OXSs-Tt3jmG1tX@mail.gmail.com>	<AANLkTi=2f9B2upJx8kg64K4NbRGaiw9fOcULET=4THoM@mail.gmail.com>	<AANLkTinBeEcX0-6gffYsWaJLNU8j8R+t31hZwMbzX7R+@mail.gmail.com>	<AANLkTinYqCUDiK=hLKPUJ9Q04VUkGgyHE_BKujasd4Cg@mail.gmail.com>	<AANLkTiks99vbGmG72fheTRb2QdCkQqQQ01gFZRsjxT2h@mail.gmail.com>	<AANLkTimcYMYuV1pxo60JhQkfK0+P94sf=b_Er8FJrwP_@mail.gmail.com>	<AANLkTimcCSMfyaajth4qQBiyDGDfhVpPCJyeVt9-VxdF@mail.gmail.com>	<AANLkTimgOsmXR6jC8pOhB7URwdn19H91ZtkRC9sfBasU@mail.gmail.com>
	<AANLkTinks_EtXZ-y7a=Xux80XrJey1dAoOr8hnJT64aS@mail.gmail.com>
Message-ID: <4CF66DB0.9090807@puzzled.xs4all.nl>

On 12/01/2010 04:03 PM, Jian Ren wrote:
> Hi Giovanni,
> Could you explain more to a bizarre person like me :-)
> What's DTMF "D"? Sent from one of my SIP client? Or Someone sent to my
> skype through chat? If it's abnormal, I think I should fix the root cause.
> Thank you very much for looking at the issue, with 2M logs :-).

DTMF A-D seems to be used only on military phones:
http://www.telecomdictionary.com/telecom_dictionary_DTMF_definition.html

Regards,
Patrick


From renjian at gmail.com  Wed Dec  1 08:33:15 2010
From: renjian at gmail.com (Jian Ren)
Date: Wed, 1 Dec 2010 11:33:15 -0500
Subject: [Freeswitch-users] : Skypopen error
In-Reply-To: <4CF66DB0.9090807@puzzled.xs4all.nl>
References: <AANLkTi=W65whZv0f7p6GxfjJWmsG9=OXSs-Tt3jmG1tX@mail.gmail.com>
	<AANLkTi=2f9B2upJx8kg64K4NbRGaiw9fOcULET=4THoM@mail.gmail.com>
	<AANLkTinBeEcX0-6gffYsWaJLNU8j8R+t31hZwMbzX7R+@mail.gmail.com>
	<AANLkTinYqCUDiK=hLKPUJ9Q04VUkGgyHE_BKujasd4Cg@mail.gmail.com>
	<AANLkTiks99vbGmG72fheTRb2QdCkQqQQ01gFZRsjxT2h@mail.gmail.com>
	<AANLkTimcYMYuV1pxo60JhQkfK0+P94sf=b_Er8FJrwP_@mail.gmail.com>
	<AANLkTimcCSMfyaajth4qQBiyDGDfhVpPCJyeVt9-VxdF@mail.gmail.com>
	<AANLkTimgOsmXR6jC8pOhB7URwdn19H91ZtkRC9sfBasU@mail.gmail.com>
	<AANLkTinks_EtXZ-y7a=Xux80XrJey1dAoOr8hnJT64aS@mail.gmail.com>
	<4CF66DB0.9090807@puzzled.xs4all.nl>
Message-ID: <AANLkTimqo4ZeodHr8WVOz28Uq9MyOuE1a11y33+diE2N@mail.gmail.com>

Interesting. I don't have these keys on my phone. Here is the dialplan
string I am using in the ATA(SPA1001):
(<:1>[2-9]xx[2-9]xxxxxx|011xx.|1[2-9]xx[2-9]xxxxxx|1xxx|00xx.)
Shouldn't it only take numbers?
Could this be caused by any dialplan XML files in freeswitch?
Thanks!
Jian

On Wed, Dec 1, 2010 at 10:45 AM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 12/01/2010 04:03 PM, Jian Ren wrote:
> > Hi Giovanni,
> > Could you explain more to a bizarre person like me :-)
> > What's DTMF "D"? Sent from one of my SIP client? Or Someone sent to my
> > skype through chat? If it's abnormal, I think I should fix the root
> cause.
> > Thank you very much for looking at the issue, with 2M logs :-).
>
> DTMF A-D seems to be used only on military phones:
> http://www.telecomdictionary.com/telecom_dictionary_DTMF_definition.html
>
> Regards,
> Patrick
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From renjian at gmail.com  Wed Dec  1 08:35:32 2010
From: renjian at gmail.com (Jian Ren)
Date: Wed, 1 Dec 2010 11:35:32 -0500
Subject: [Freeswitch-users] : Skypopen error
In-Reply-To: <AANLkTimqo4ZeodHr8WVOz28Uq9MyOuE1a11y33+diE2N@mail.gmail.com>
References: <AANLkTi=W65whZv0f7p6GxfjJWmsG9=OXSs-Tt3jmG1tX@mail.gmail.com>
	<AANLkTi=2f9B2upJx8kg64K4NbRGaiw9fOcULET=4THoM@mail.gmail.com>
	<AANLkTinBeEcX0-6gffYsWaJLNU8j8R+t31hZwMbzX7R+@mail.gmail.com>
	<AANLkTinYqCUDiK=hLKPUJ9Q04VUkGgyHE_BKujasd4Cg@mail.gmail.com>
	<AANLkTiks99vbGmG72fheTRb2QdCkQqQQ01gFZRsjxT2h@mail.gmail.com>
	<AANLkTimcYMYuV1pxo60JhQkfK0+P94sf=b_Er8FJrwP_@mail.gmail.com>
	<AANLkTimcCSMfyaajth4qQBiyDGDfhVpPCJyeVt9-VxdF@mail.gmail.com>
	<AANLkTimgOsmXR6jC8pOhB7URwdn19H91ZtkRC9sfBasU@mail.gmail.com>
	<AANLkTinks_EtXZ-y7a=Xux80XrJey1dAoOr8hnJT64aS@mail.gmail.com>
	<4CF66DB0.9090807@puzzled.xs4all.nl>
	<AANLkTimqo4ZeodHr8WVOz28Uq9MyOuE1a11y33+diE2N@mail.gmail.com>
Message-ID: <AANLkTinp7W6d3GOh2C=rA748ceQmni+tFA3QNkSu5qj=@mail.gmail.com>

Also this happened during a call, which got hanged up because of the error.
So the phone sends out A-D during a call?
Jian
On Wed, Dec 1, 2010 at 11:33 AM, Jian Ren <renjian at gmail.com> wrote:

> Interesting. I don't have these keys on my phone. Here is the dialplan
> string I am using in the ATA(SPA1001):
> (<:1>[2-9]xx[2-9]xxxxxx|011xx.|1[2-9]xx[2-9]xxxxxx|1xxx|00xx.)
> Shouldn't it only take numbers?
> Could this be caused by any dialplan XML files in freeswitch?
> Thanks!
> Jian
>
>
> On Wed, Dec 1, 2010 at 10:45 AM, Patrick Lists <
> freeswitch-list at puzzled.xs4all.nl> wrote:
>
>> On 12/01/2010 04:03 PM, Jian Ren wrote:
>> > Hi Giovanni,
>> > Could you explain more to a bizarre person like me :-)
>> > What's DTMF "D"? Sent from one of my SIP client? Or Someone sent to my
>> > skype through chat? If it's abnormal, I think I should fix the root
>> cause.
>> > Thank you very much for looking at the issue, with 2M logs :-).
>>
>> DTMF A-D seems to be used only on military phones:
>> http://www.telecomdictionary.com/telecom_dictionary_DTMF_definition.html
>>
>> Regards,
>> Patrick
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From anthony.minessale at gmail.com  Wed Dec  1 08:39:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 1 Dec 2010 10:39:57 -0600
Subject: [Freeswitch-users] dead call records
In-Reply-To: <AANLkTimhxMGBzZkcpO1GzvPDYUE6empmE1NxAXrjm4d5@mail.gmail.com>
References: <AANLkTimhxMGBzZkcpO1GzvPDYUE6empmE1NxAXrjm4d5@mail.gmail.com>
Message-ID: <AANLkTi=4u-wH4+9CiUdvM50dC0JKx7+7kSdA7ZO=mVHT@mail.gmail.com>

commit ca28a80658d50a9fab14f3022c40536b385b6b9b
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Wed Dec 1 10:31:28 2010 -0600

    update caller_profile to have correct uuid when using custom uuid
from originate string


On Wed, Dec 1, 2010 at 7:51 AM, Seven Du <dujinfang at gmail.com> wrote:
> Hi,
>
> I found a wired problem where left dead call records.
>
> originate {origination_uuid=1,hangup_after_bridge=false}user/1000 &park
> originate {origination_uuid=2,hangup_after_bridge=false}user/1001 &park
> uuid_bridge 1 2
> uuid_transfer 1 -both park inline
> hupall
>
> show channels # show nothing
> show calls # the record still there, why the uuid is not 1 and 2 but a
> long uuid str?
>
> but there's no problem if not use origination_uuid.
>
> Also, there are no problems if no uuid_transfer.
>
> originate {origination_uuid=1,hangup_after_bridge=false}user/1000 &park
> originate {origination_uuid=2,hangup_after_bridge=false}user/1001 &park
> uuid_bridge 1 2
> hupall
>
>
> I'm on last git, anyone can help take a look?
>
> http://pastebin.freeswitch.org/14681
>
> Thanks.
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:? http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From gmaruzz at gmail.com  Wed Dec  1 08:56:30 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 1 Dec 2010 17:56:30 +0100
Subject: [Freeswitch-users] : Skypopen error
In-Reply-To: <AANLkTinp7W6d3GOh2C=rA748ceQmni+tFA3QNkSu5qj=@mail.gmail.com>
References: <AANLkTi=W65whZv0f7p6GxfjJWmsG9=OXSs-Tt3jmG1tX@mail.gmail.com>
	<AANLkTi=2f9B2upJx8kg64K4NbRGaiw9fOcULET=4THoM@mail.gmail.com>
	<AANLkTinBeEcX0-6gffYsWaJLNU8j8R+t31hZwMbzX7R+@mail.gmail.com>
	<AANLkTinYqCUDiK=hLKPUJ9Q04VUkGgyHE_BKujasd4Cg@mail.gmail.com>
	<AANLkTiks99vbGmG72fheTRb2QdCkQqQQ01gFZRsjxT2h@mail.gmail.com>
	<AANLkTimcYMYuV1pxo60JhQkfK0+P94sf=b_Er8FJrwP_@mail.gmail.com>
	<AANLkTimcCSMfyaajth4qQBiyDGDfhVpPCJyeVt9-VxdF@mail.gmail.com>
	<AANLkTimgOsmXR6jC8pOhB7URwdn19H91ZtkRC9sfBasU@mail.gmail.com>
	<AANLkTinks_EtXZ-y7a=Xux80XrJey1dAoOr8hnJT64aS@mail.gmail.com>
	<4CF66DB0.9090807@puzzled.xs4all.nl>
	<AANLkTimqo4ZeodHr8WVOz28Uq9MyOuE1a11y33+diE2N@mail.gmail.com>
	<AANLkTinp7W6d3GOh2C=rA748ceQmni+tFA3QNkSu5qj=@mail.gmail.com>
Message-ID: <AANLkTi=aOgFgELyKVHBAWpLq33a29AepH7Ur+v4YCw1r@mail.gmail.com>

Hi Jian,

I hope in English "bizarre" does not sound bad, in Italian would be
like "original and out of standard in a funny way" :).

>From the log you attached to the Jira, the incoming SIP calls that are
then bridged to skypopen are sending both the "A" dtmf and the "D"
dtmf (can't remember if any other).

You can peruse the log looking for "error 21", and you'll  see is
anytime that they sent to you (via SIP) one of the A-B-C-D dtmfs that
mod_skypopen duly passes to skype.

Problem is: the Skype client does not accept or relay dtmf "A-D", and
spit out an error.

Out of curiosity you may want to check why your customers are using
dtrmf A-D, but is not an absolut need.

Anyway, I'll fix this in mod_skypopen code asap, so that if another
channel (SIP in your case) try to send A-D to skype, that dtmf will be
ignored and a warning line will be emitted to console and to logfile.
And no more errors or aborted calls.

-giovanni



On Wed, Dec 1, 2010 at 5:35 PM, Jian Ren <renjian at gmail.com> wrote:
> Also this happened during a call, which got hanged up because of the error.
> So the phone sends out A-D during a call?
> Jian
> On Wed, Dec 1, 2010 at 11:33 AM, Jian Ren <renjian at gmail.com> wrote:
>>
>> Interesting. I don't have these keys on my phone. Here is the dialplan
>> string I am using in the ATA(SPA1001):
>> (<:1>[2-9]xx[2-9]xxxxxx|011xx.|1[2-9]xx[2-9]xxxxxx|1xxx|00xx.)
>> Shouldn't it only take numbers?
>> Could this be caused by any dialplan XML files in freeswitch?
>> Thanks!
>> Jian
>>
>> On Wed, Dec 1, 2010 at 10:45 AM, Patrick Lists
>> <freeswitch-list at puzzled.xs4all.nl> wrote:
>>>
>>> On 12/01/2010 04:03 PM, Jian Ren wrote:
>>> > Hi Giovanni,
>>> > Could you explain more to a bizarre person like me :-)
>>> > What's DTMF "D"? Sent from one of my SIP client? Or Someone sent to my
>>> > skype through chat? If it's abnormal, I think I should fix the root
>>> > cause.
>>> > Thank you very much for looking at the issue, with 2M logs :-).
>>>
>>> DTMF A-D seems to be used only on military phones:
>>> http://www.telecomdictionary.com/telecom_dictionary_DTMF_definition.html
>>>
>>> Regards,
>>> Patrick
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From msc at freeswitch.org  Wed Dec  1 09:13:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Dec 2010 09:13:26 -0800
Subject: [Freeswitch-users] uuid_bridge on bridged channels
In-Reply-To: <AANLkTi=EKFWvU3JGJrch4=zHdZdVO7gcTwTPRSN=3RJv@mail.gmail.com>
References: <AANLkTi=EKFWvU3JGJrch4=zHdZdVO7gcTwTPRSN=3RJv@mail.gmail.com>
Message-ID: <AANLkTimh0h5EHhS1jizxHBYAf+CZ0n3Qqzc=0h6vuwyf@mail.gmail.com>

You can call uuid_bridge from the dialplan if you really wanted to. Just use
the "set" app with data="${uuid_bridge foo bar}"

-MC

On Tue, Nov 30, 2010 at 7:32 AM, Seven Du <dujinfang at gmail.com> wrote:

> Hi,
>
> Is it make sence to brige a bridged channel to another channel?
>
> I'm on FreeSWITCH Version 1.0.head (git-28f5d28 2010-11-09 14-11-02 -0500)
>
> And trying this:
>
> originate
> {origination_uuid=1,hangup_after_bridge=false,exec_after_bridge_app=park}user/1000
> &park
> originate {origination_uuid=2,exec_after_bridge_app=park}user/1001 &park
> originate
> {origination_uuid=3,exec_after_bridge_app=park}portaudio/auto_answer
> &park
>
> and
>
> uuid_bridge 1 2   # Then it bridged, and then I did
>
> uuid_bridge 1 3   # 1 bridged to 3 and 2 parked, then
>
> uuid_bridge 2 3   # no bridges all channels parked
>
> Then I tried a few uuid_bridge between two of 1,2,3  until all
> channels hangup with destination out of order
>
> http://pastebin.freeswitch.org/14661
>
>
> I was trying to move calls between live channels, should I unbridge
> all channels completely and bridge later? If so, is it possible to use
> the bridge app to bridge to a existing UUID instead of a dial string?
> (makes sense, since then we can use transfer bridge:other_uuid inline
> instead of uuid_bridge
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From renjian at gmail.com  Wed Dec  1 09:18:06 2010
From: renjian at gmail.com (Jian Ren)
Date: Wed, 1 Dec 2010 12:18:06 -0500
Subject: [Freeswitch-users] : Skypopen error
In-Reply-To: <AANLkTi=aOgFgELyKVHBAWpLq33a29AepH7Ur+v4YCw1r@mail.gmail.com>
References: <AANLkTi=W65whZv0f7p6GxfjJWmsG9=OXSs-Tt3jmG1tX@mail.gmail.com>
	<AANLkTi=2f9B2upJx8kg64K4NbRGaiw9fOcULET=4THoM@mail.gmail.com>
	<AANLkTinBeEcX0-6gffYsWaJLNU8j8R+t31hZwMbzX7R+@mail.gmail.com>
	<AANLkTinYqCUDiK=hLKPUJ9Q04VUkGgyHE_BKujasd4Cg@mail.gmail.com>
	<AANLkTiks99vbGmG72fheTRb2QdCkQqQQ01gFZRsjxT2h@mail.gmail.com>
	<AANLkTimcYMYuV1pxo60JhQkfK0+P94sf=b_Er8FJrwP_@mail.gmail.com>
	<AANLkTimcCSMfyaajth4qQBiyDGDfhVpPCJyeVt9-VxdF@mail.gmail.com>
	<AANLkTimgOsmXR6jC8pOhB7URwdn19H91ZtkRC9sfBasU@mail.gmail.com>
	<AANLkTinks_EtXZ-y7a=Xux80XrJey1dAoOr8hnJT64aS@mail.gmail.com>
	<4CF66DB0.9090807@puzzled.xs4all.nl>
	<AANLkTimqo4ZeodHr8WVOz28Uq9MyOuE1a11y33+diE2N@mail.gmail.com>
	<AANLkTinp7W6d3GOh2C=rA748ceQmni+tFA3QNkSu5qj=@mail.gmail.com>
	<AANLkTi=aOgFgELyKVHBAWpLq33a29AepH7Ur+v4YCw1r@mail.gmail.com>
Message-ID: <AANLkTiks+WPpo6P5qvaeqmEydNQX3RbTBNECkbcsx612@mail.gmail.com>

Hmm, so far the only "customer" is my wife, who keeps complaining to me
everyday. It explains why never happened to me. I will ask her tonight how
she sent A or D while calling. Or maybe it's the problem of the phone. She
is using a dual mode phone. Before, it's connected to PC with USB for skype
calls on Windows. Now I stopped running any version of skype on windows and
the USB was disconnected so she is using the phone as a normal one.
Thanks!
Jian

On Wed, Dec 1, 2010 at 11:56 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> Hi Jian,
>
> I hope in English "bizarre" does not sound bad, in Italian would be
> like "original and out of standard in a funny way" :).
>
> >From the log you attached to the Jira, the incoming SIP calls that are
> then bridged to skypopen are sending both the "A" dtmf and the "D"
> dtmf (can't remember if any other).
>
> You can peruse the log looking for "error 21", and you'll  see is
> anytime that they sent to you (via SIP) one of the A-B-C-D dtmfs that
> mod_skypopen duly passes to skype.
>
> Problem is: the Skype client does not accept or relay dtmf "A-D", and
> spit out an error.
>
> Out of curiosity you may want to check why your customers are using
> dtrmf A-D, but is not an absolut need.
>
> Anyway, I'll fix this in mod_skypopen code asap, so that if another
> channel (SIP in your case) try to send A-D to skype, that dtmf will be
> ignored and a warning line will be emitted to console and to logfile.
> And no more errors or aborted calls.
>
> -giovanni
>
>
>
> On Wed, Dec 1, 2010 at 5:35 PM, Jian Ren <renjian at gmail.com> wrote:
> > Also this happened during a call, which got hanged up because of the
> error.
> > So the phone sends out A-D during a call?
> > Jian
> > On Wed, Dec 1, 2010 at 11:33 AM, Jian Ren <renjian at gmail.com> wrote:
> >>
> >> Interesting. I don't have these keys on my phone. Here is the dialplan
> >> string I am using in the ATA(SPA1001):
> >> (<:1>[2-9]xx[2-9]xxxxxx|011xx.|1[2-9]xx[2-9]xxxxxx|1xxx|00xx.)
> >> Shouldn't it only take numbers?
> >> Could this be caused by any dialplan XML files in freeswitch?
> >> Thanks!
> >> Jian
> >>
> >> On Wed, Dec 1, 2010 at 10:45 AM, Patrick Lists
> >> <freeswitch-list at puzzled.xs4all.nl> wrote:
> >>>
> >>> On 12/01/2010 04:03 PM, Jian Ren wrote:
> >>> > Hi Giovanni,
> >>> > Could you explain more to a bizarre person like me :-)
> >>> > What's DTMF "D"? Sent from one of my SIP client? Or Someone sent to
> my
> >>> > skype through chat? If it's abnormal, I think I should fix the root
> >>> > cause.
> >>> > Thank you very much for looking at the issue, with 2M logs :-).
> >>>
> >>> DTMF A-D seems to be used only on military phones:
> >>>
> http://www.telecomdictionary.com/telecom_dictionary_DTMF_definition.html
> >>>
> >>> Regards,
> >>> Patrick
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Dec  1 09:20:13 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Dec 2010 09:20:13 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <AANLkTikMG3OcHanKz_nTLU=G4fp=zpJ6ajwnG31+bJ-N@mail.gmail.com>

Hello all,

Today's agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_01

We'll be doing another community documentation session but first we'll have
Chad Phillips, aka hunmonk, talking to us about his Jester Mail project.
Talk to you in a bit.

-MC
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From msc at freeswitch.org  Wed Dec  1 09:22:12 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Dec 2010 09:22:12 -0800
Subject: [Freeswitch-users] Record File
In-Reply-To: <699521.99191.qm@web55705.mail.re3.yahoo.com>
References: <699521.99191.qm@web55705.mail.re3.yahoo.com>
Message-ID: <AANLkTinM3xFNVYgY8dwubhOkyu70HeBk=p-UH3vd1+7C@mail.gmail.com>

How about uuid_record?
-MC

On Mon, Nov 15, 2010 at 7:38 AM, Will Smith <willbelair at yahoo.com> wrote:

> hi,
> I am trying to record a call after bridging the 2 legs. 2 parties can
> talk/listen to each other. There is an audio file created, but just silence,
> and always stopped at 44bytes. I just wonder if there is any other setup
> that I missed.
> I used 2 ways to record the call, and they both gave the same result:
> 1-   session.execute("record_session", "foo.wav");
>
> 2- rtn = session.recordFile("foo.wav", true, "", 500,500,3);
>
> Thank you for your help.
>
> Will
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From adminjew at gmail.com  Wed Dec  1 10:09:24 2010
From: adminjew at gmail.com (Yitzchok)
Date: Wed, 1 Dec 2010 13:09:24 -0500
Subject: [Freeswitch-users] building mod_managed fails
In-Reply-To: <4CF66CBF.9080308@puzzled.xs4all.nl>
References: <AANLkTi=96X=WyXFZ1ZTU8AE3fKiZUm6Q=tr-ds4kOCWa@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703681DD6C9@mse17be1.mse17.exchange.ms>
	<AANLkTi=xLHN7SAL9t0A+aiDtGTwYoCDwqNFjMapSGQpu@mail.gmail.com>
	<AANLkTimWZYJJEyp9Ayu-Ou3asR0kqbZ2EcjVq8mLAk_c@mail.gmail.com>
	<4CF66CBF.9080308@puzzled.xs4all.nl>
Message-ID: <AANLkTinrNQXY=3VosgdNFPs5398OO_fCD3BtrQ+4FP1J@mail.gmail.com>

Are you sure you have Mono 2.6.7 and not Mono 2.8?



Yitzchok


On Wed, Dec 1, 2010 at 10:41 AM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 12/01/2010 03:53 PM, babak yakhchali wrote:
> > Hi Micheal
> > I've attached errors from building with mono 2.6.7
>
> Did you actually read the error or did you just attach it?
> Your answer is on line 1 and 2 of the error file. Clearly you are
> missing some header files.
>
> Patrick
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From marcdecorny at gmail.com  Wed Dec  1 04:17:50 2010
From: marcdecorny at gmail.com (Marc de Corny)
Date: Wed, 1 Dec 2010 12:17:50 +0000
Subject: [Freeswitch-users] Simple LUA not playing message issue
Message-ID: <AANLkTi=gCYtmBo_gPbBkX+BG05-61RPJmjz+SN8D240y@mail.gmail.com>

Hi all,

I have run into an issue on something so basic that I must be as simple as
enabling a feature somewhere.

I have been trying to get lua to play a message from a WAV file. I have
tried session:execute("playback", main_msg) and
session:streamFile(ivr_invalid_msg) but neither of them play any music to
the caller. I tried both to answer and preAnswer the call first but it made
no difference. However if I put the same file into the XML dialplan and play
it with the commands below I hear the music fine.
<action application="set" data="playback_terminators=#"/>
<action application="playback" data="${401_sound_file_path}"/>

The issue only seems to be from lua when playing any type of wav file and
those files are definitelly there as can be read by the XML

The error message is below for the execute(playback) command, but nothing
can be seen for the
2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application
playback Requires media! pre_answering channel
sofia/external/2031701665 at 194.0.147.16:5060 EXECUTE
sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswitch/sounds/svc_sound_files/default_autoattendant.wav)

But there is no mention of the streamFile command. I have had similar issue
with the PlayAndGetDigits command.
Is there something that I need to enable in lua so that is can playback
messages to the caller.

Many thanks to anyone who can help.
Marc


below is the XML dialplan and lua script as well as the log at the very end.


*XML DIALPLAN:*
<include>
<extension name="IVR_FROM_MYSQL">
<condition field="caller_id_number" expression="Anonymous" break="never">
<action application="set" data="effective_caller_id_number=0000000000"/>
</condition>
<condition field="destination_number" expression="^(404)">
<action inline="true" application="lua" data="ivr_mysql.lua
${destination_number:3}"/>
<action application="set" data="effective_caller_id_name=${404_tag}"/>
<action application="bridge" data="${404_dial}"/>
</condition>
</extension>
</include>

*The LUA script ivr_mysql.lua is callsed and this is it.
-- IVR : PLAY IVR WAV FILES
*-- Global Variables:
local dialstr_prefix = "sofia/gateway/CS2k/"
local dialstr_main = ""
local breakoutcode = "184"
local sound_file_folder = "/usr/local/freeswitch/sounds/svc_sound_files/"
local ddi = argv[1]
-- answer the call
session:preAnswer();
freeswitch.consoleLog("info", "All Answered\n");

ivr_invalid_msg = sound_file_folder .. "invalid_msg.wav"
main_msg = sound_file_folder .. "default_autoattendant.wav"
-- Play with Execute
session:execute("playback", main_msg)
-- Play with StreamFile
session:streamFile(ivr_invalid_msg);
dialstr_main  = dialstr_main .. dialstr_prefix .. breakoutcode ..
"02031701665"
session:setVariable("404_dial",dialstr_main)
session:setVariable("404_tag","IVR")


*RELEVANT LOGS :*
Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Regex (PASS)
[IVR_FROM_MYS                          QL] destination_number(4042031956241)
=~ /^(404)/ break=on-false
Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
lua(ivr_mysql.lua $                          {destination_number:3}) INLINE
EXECUTE sofia/external/2031701665 at 194.0.147.16:5060 lua(ivr_mysql.lua
2031956241                          )
2010-12-01 11:56:01.525426 [INFO] switch_cpp.cpp:584 Sending early media
2010-12-01 11:56:01.525426 [DEBUG] sofia_glue.c:2972 AUDIO RTP
[sofia/external/2                          031701665 at 194.0.147.16:5060]
10.5.2.105 port 29900 -> 194.0.147.164 port 50202 c
odec: 8 ms: 20
2010-12-01 11:56:01.525426 [DEBUG] switch_rtp.c:1418 Starting timer [soft]
160 b                          ytes per 20ms
2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3190 Set 2833 dtmf send
payload                           to 101
2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3195 Set 2833 dtmf receive
paylo                          ad to 101
2010-12-01 11:56:01.532280 [DEBUG] mod_sofia.c:2172 Ring SDP:
v=0
o=FreeSWITCH 1291174661 1291174662 IN IP4 10.5.2.105
s=FreeSWITCH
c=IN IP4 10.5.2.105
t=0 0
m=audio 29900 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
2010-12-01 11:56:01.532280 [NOTICE] mod_sofia.c:2175 Pre-Answer
sofia/external/2                          031701665 at 194.0.147.16:5060!
2010-12-01 11:56:01.532280 [DEBUG] switch_channel.c:2544
(sofia/external/2031701                          665 at 194.0.147.16:5060)
Callstate Change RINGING -> EARLY
2010-12-01 11:56:01.534727 [DEBUG] sofia.c:4576 Channel
sofia/external/203170166
5 at 194.0.147.16:5060skipping state [early][183]
2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:676 Send signal
sofia/e                          xternal/2031701665 at 194.0.147.16:5060[BREAK]
2010-12-01 11:56:01.534727 [INFO] switch_cpp.cpp:1181 All Answered
2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application
playba                          ck Requires media! pre_answering channel
sofia/external/2031701665 at 194.0.147.16:                          5060
EXECUTE sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswit
ch/sounds/svc_sound_files/default_autoattendant.wav)
2010-12-01 11:56:01.537644 [DEBUG] switch_cpp.cpp:972
sofia/external/2031701665@
194.0.147.16:5060destroy/unlink session from object
Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
set(effective_calle                          r_id_name=${404_tag})
Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
bridge(${404_dial})
2010-12-01 11:56:01.537644 [DEBUG] switch_core_state_machine.c:119
(sofia/extern                          al/2031701665 at 194.0.147.16:5060)
State Change CS_ROUTING -> CS_EXECUTE
2010-12-01 11:56:01.537644 [DEBUG] switch_core_session.c:1057 Send signal
sofia/                          external/2031701665 at 194.0.147.16:5060[BREAK]
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From raimund.sacherer at logitravel.com  Wed Dec  1 07:44:03 2010
From: raimund.sacherer at logitravel.com (Raimund Sacherer)
Date: Wed, 1 Dec 2010 16:44:03 +0100 (CET)
Subject: [Freeswitch-users] mod_callcenter and peridic announcements
Message-ID: <2601137.175251291218243111.JavaMail.root@pina>

Hello Freeswitch List.

We are evaluating switching our asterisk based call center (around 150 agents) over to freeswitch. 

Among other things I noticed that the mod_callcenter module does not have the possibility to use periodic announcements. Could these be implemented somehow ... like, with ESL or something else?


I want to try out the Spice Technologies (or is it now OpenACD) callcenter solution, but this will be a further project as we first want to switch to freeswitch as soon as possible as we have instability problems which we want to get rid off.

Thank you for your support,

Ray


-- 
Raimund Sacherer 
Dpto. de Sistemas 
Agencia de Viajes Online 


From infos at madovsky.org  Wed Dec  1 08:53:09 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 1 Dec 2010 11:53:09 -0500
Subject: [Freeswitch-users] can't get channel variables after hangup
Message-ID: <AE2E563CA6B0424FA874C7E68CC77C2A@e1705>

I set this in my dialplan :

   <action application="set" data="session_in_hangup_hook=true"/>
   <action application="set" data="api_hangup_hook=system /usr/local/bin/php /usr/local/freeswitch/scripts/php/test.php ${uuid}"/>

  and this in test.php:

  exec("/usr/local/freeswitch/bin/fs_cli -x \"uuid_getvar ".$argv[1]."  test_var\"", $exTabRet, $req);


 it results :

-ERR No Such Channel!


I tried to replace ${uuid} by ${call_uuid} without success.

Thanks

Franck
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From davidjbrazier at gmail.com  Wed Dec  1 09:16:46 2010
From: davidjbrazier at gmail.com (David Brazier)
Date: Wed, 1 Dec 2010 17:16:46 +0000
Subject: [Freeswitch-users] building mod_managed fails
In-Reply-To: <4CF66CBF.9080308@puzzled.xs4all.nl>
References: <AANLkTi=96X=WyXFZ1ZTU8AE3fKiZUm6Q=tr-ds4kOCWa@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703681DD6C9@mse17be1.mse17.exchange.ms>
	<AANLkTi=xLHN7SAL9t0A+aiDtGTwYoCDwqNFjMapSGQpu@mail.gmail.com>
	<AANLkTimWZYJJEyp9Ayu-Ou3asR0kqbZ2EcjVq8mLAk_c@mail.gmail.com>
	<4CF66CBF.9080308@puzzled.xs4all.nl>
Message-ID: <AANLkTim9EFf4HxHt=hh+d2YzP-e+mBzoCodfGqMdQCDg@mail.gmail.com>

I had a similar problem - try using mono 2.8.1 and the patch mentioned
in http://jira.freeswitch.org/browse/FS-2774.  This tells you to
re-swig (make reswig) & after that I found that there were compile
errors in the swig output - double declarations etc - easy enough to
remove.  I'll try to investigate why the swig output wasn't right.
This was on Linux (CentOS 5.5) - you may find that on Windows it just
works.

David

On Wed, Dec 1, 2010 at 3:41 PM, Patrick Lists
<freeswitch-list at puzzled.xs4all.nl> wrote:
>
> On 12/01/2010 03:53 PM, babak yakhchali wrote:
> > Hi Micheal
> > I've attached errors from building with mono 2.6.7
>
> Did you actually read the error or did you just attach it?
> Your answer is on line 1 and 2 of the error file. Clearly you are
> missing some header files.
>
> Patrick
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Wed Dec  1 11:08:21 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Dec 2010 11:08:21 -0800
Subject: [Freeswitch-users] how to use mod_curl in Lua on window
	platform ???
In-Reply-To: <AANLkTincmVBfsT3bHi1=SGDo1kGyUKdvEGPkoi86HhxK@mail.gmail.com>
References: <AANLkTincmVBfsT3bHi1=SGDo1kGyUKdvEGPkoi86HhxK@mail.gmail.com>
Message-ID: <AANLkTikcJyS4--jD3tsL99NDaTH1CXkFu2yM9DZBmGgw@mail.gmail.com>

Is it not working for you? You can use the curl API and call it from Lua.
See chapter 7 of the FreeSWITCH book for explicit details.

-MC

On Thu, Nov 25, 2010 at 1:58 AM, dhiraj bharti <dhiraj.bharti at gmail.com>wrote:

> Hi,
> Anybody have any idea how to user cURL in Lua on window platform ?
>
> Thanks in Advance
> --
> Dhiraj bharti
>
> DBS Technologies Pvt. Ltd
> 9990448967
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From willbelair at yahoo.com  Wed Dec  1 11:14:35 2010
From: willbelair at yahoo.com (Will Smith)
Date: Wed, 1 Dec 2010 11:14:35 -0800 (PST)
Subject: [Freeswitch-users] session_record
Message-ID: <37890.5730.qm@web55706.mail.re3.yahoo.com>

hi ,I have a problem to record files.here is my javascript:---------------session = new Session('sofia/gateway/mygateway/14164516999');session.waitForAnswer(10000);
if (session.ready()) {?? ?session.execute("record_session", "/usr/local/freeswitch/recordings/" + session.uuid + ".wav"); ? ??}---------------after press answer key on my keypad, the call disconnected. What I am missing here?
Thank you for your help.
Will


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From anthony.minessale at gmail.com  Wed Dec  1 11:18:58 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 1 Dec 2010 13:18:58 -0600
Subject: [Freeswitch-users] mod_fifo vs mod_callcenter
In-Reply-To: <AANLkTimftpTpLLYXED0Fk1=NwVowzxv=8T0ufQBkrAZ8@mail.gmail.com>
References: <AANLkTimftpTpLLYXED0Fk1=NwVowzxv=8T0ufQBkrAZ8@mail.gmail.com>
Message-ID: <AANLkTinC85pzthxCVcjopW9q=cjBkkjS0jXGBhKBjyHV@mail.gmail.com>

no there are no plans.
The calls are distributed to the agents based on the number of missed
calls they earn and the number of calls they have made.
Call strategies are mythical.


2010/11/29 Eugene Prokopiev <enp at altlinux.org>:
> Hi,
>
> No mod_fifo nor mod_?all?enter not fit me completely. There are no
> strategy option in mod_fifo, so incoming calls are distributed to
> agents unfairly. There are no ?something like "fifo in" and "fifo out
> wait" in mod_?all?enter. It it can be implemented with
> "callcenter_config agent add" and "callcenter_config tier add" but it
> is impossible to execute "callcenter_config agent del" and
> "callcenter_config tier del" automatically after agent hungup.
>
> Whether there are plans to add something like strategy option to
> mod_fifo or simplify mod_?all?enter usage?
>
> --
> Thanks,
> Eugene Prokopiev
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From willbelair at yahoo.com  Wed Dec  1 11:19:50 2010
From: willbelair at yahoo.com (Will Smith)
Date: Wed, 1 Dec 2010 11:19:50 -0800 (PST)
Subject: [Freeswitch-users] Record File
In-Reply-To: <AANLkTinM3xFNVYgY8dwubhOkyu70HeBk=p-UH3vd1+7C@mail.gmail.com>
Message-ID: <154087.81865.qm@web55701.mail.re3.yahoo.com>

Thanks for answering.I did not see your email before posting another thread.I tried to run :apiExecute("uuid_record ......", "");?but it does not work.Also, I try this, and it stops right after I answer the call:------session = new Session('sofia/gateway/mygateway/19054516999');session.waitForAnswer(10000);
if (session.ready()) {?? ?session.execute("record_session", "/usr/local/freeswitch/recordings/" + session.uuid + ".wav"); ? ??}--------what I am missing. I spent 2 weeks looking for a way to record file, but still going nowhere.
Thank you

--- On Wed, 12/1/10, Michael Collins <msc at freeswitch.org> wrote:

From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Record File
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Received: Wednesday, December 1, 2010, 9:22 AM

How about uuid_record?-MC

On Mon, Nov 15, 2010 at 7:38 AM, Will Smith <willbelair at yahoo.com> wrote:

hi,I am trying to record a call after bridging the 2 legs. 2 parties can talk/listen to each other. There is an audio file created, but just silence, and always stopped at 44bytes. I just wonder if there is any other setup that I missed.
I used 2 ways to record the call, and they both gave the same result:1- ? session.execute("record_session", "foo.wav");
2- rtn = session.recordFile("foo.wav", true, "", 500,500,3);

Thank you for your help.
Will

_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From gmaruzz at gmail.com  Wed Dec  1 11:19:48 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 1 Dec 2010 20:19:48 +0100
Subject: [Freeswitch-users] /usr/local/freeswitch/mod/mod_gsmopen.so:
 cannot open shared object file: No such file or directory
In-Reply-To: <AANLkTikejPqZ8gL0qZDLb+Bz2yBPCCpSD=Df=mbt95D+@mail.gmail.com>
References: <AANLkTikejPqZ8gL0qZDLb+Bz2yBPCCpSD=Df=mbt95D+@mail.gmail.com>
Message-ID: <AANLkTinb7E4uR_Mc6A1xR7hRxFj3pd8Q1X8--jTrqeGT@mail.gmail.com>

Hi Anshu,

have you compiled mod_gsmopen? Do you have installed the needed
dependencies (eg: libgssme-dev and gsmutils)?

I modified yesterday the wiki page to be more update, maybe you want
to take a look at it again.

-giovanni

On Mon, Nov 29, 2010 at 7:14 AM, Anshu Sah <sah.anshu at ymail.com> wrote:
> Hi,
>
> I was following steps of http://wiki.freeswitch.org/wiki/GSMopen to Install
> GSMOpen,
>
> OS : UBUNTU Server v10
> FreeSWITCH Version 1.0.6
> gcc (Ubuntu 4.4.3-4ubuntu5) 4.4.3
>
> After Completing Last step
>
> -----------------------------------------------------------------------
> cd /usr/src/freeswitch.trunk/src/mod/endpoints/mod_gsmopen/configs/
>
> cp gsmopen.conf.xml /usr/local/freeswitch/conf/autoload_configs/
> vi /usr/local/freeswitch/conf/autoload_configs/gsmopen.conf.xml
>
> -----------------------------------------------------------------------
>
> When I run
>
> -----------------------------------------------------------------------
> /usr/local/freeswitch/bin/freeswitch
> load mod_gsmopen
>
> -----------------------------------------------------------------------
>
> ERROR
>
> -----------------------------------------------------------------------
>
> 2010-11-29 10:32:43.064571 [CRIT] switch_loadable_module.c:882 Error Loading
> module /usr/local/freeswitch/mod/mod_gsmopen.so
> **/usr/local/freeswitch/mod/mod_gsmopen.so: cannot open shared object file:
> No such file or directory**
>
> -----------------------------------------------------------------------
>
> I tried to search mod_gsmopen.so, But didn't found anywhere :(
> Please guide.
>
> Regards
> Anshu Sah
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From msc at freeswitch.org  Wed Dec  1 11:32:08 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Dec 2010 11:32:08 -0800
Subject: [Freeswitch-users] SNMP - FS_cli -x syntax
In-Reply-To: <AANLkTikG7qtAxt=MR1UjOO-wrYY0ouYA75_zHGZ-sCJx@mail.gmail.com>
References: <AANLkTikG7qtAxt=MR1UjOO-wrYY0ouYA75_zHGZ-sCJx@mail.gmail.com>
Message-ID: <AANLkTimhsucLjZbB1k+QFOYoswSrnJN-79wKHjLXcEt2@mail.gmail.com>

At approx line 3578 in mod_commands.c you have this:

sprintf(sql,
"select * from channels where hostname='%s' and uuid like '%s' or name like
'%s' or cid_name like '%s' or cid_num like '%s' order by created_epoch",
hostname, argv[2], argv[2], argv[2], argv[2]);

So the five fields that the 'like' param includes are:
hostname
uuid
name (that is, channel name)
cid_name
cid_num

So in theory you can do something like this:
show channels like sofia
or
show channels like freetdm
or
show channels like loopback

It's my understanding that relying on the channel name in this manner has
some drawbacks, as the name of the channel can change. Therefore it may be
better for you to dump the whole list of channels and perform your parsing
offline with a perl/python/whatever script.

What is the ultimate output format that you are wanting?
-MC

On Thu, Nov 25, 2010 at 10:17 PM, Francis Trevor <fiberhaus at gmail.com>wrote:

> I have created the activechannels script as recommended in another post to
> export active calls so that they may be displayed in a NMS. This is
> basically a "fs_cli -x distinct_channels"  However, is it possible to show
> channels for a particular tech prefix? Or rather put, what is the syntax for
> the fs_cli -x show channels like <999>. Whereas 999 would be the tech
> prefix. Am I on the right track for building the logic to show active calls
> per customer?
>
> TGF
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Dec  1 11:33:14 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Dec 2010 11:33:14 -0800
Subject: [Freeswitch-users] freeswitch transfer call problem
In-Reply-To: <E1PM1zr-0007Wu-00.melkybes-mail-ru@f200.mail.ru>
References: <E1PM1zr-0007Wu-00.melkybes-mail-ru@f200.mail.ru>
Message-ID: <AANLkTi=szKk=QNVSghbxYzyPJUeR2ZqYeVJfNhhVjjkf@mail.gmail.com>

Pastebin the debug output so we can see what is happening.
-MC

2010/11/26 ?????? ???????? <melkybes at mail.ru>

>
>
> HelloI have some problem with freeswitch call transfer by python  script.
>
> -----------------------------------------------
> import os
> from freeswitch import *
> import psycopg2
> import socket
>
> def handler(session, args):
>       print 'ready to transfer'
>       number_to_transfer = session.getVariable('number_')
>       print number_to_transfer
>       session.setAutoHangup('false')
>       session.execute('set', 'hangup_after_bridge=false')
>       session.execute('set', 'continue_on_fail=true')
>       session.transfer( number_to_transfer, "XML", "default")
>
> Description of my problem
>
> Phone # 1  call to Phone #2
> Phone #2   input some digits for transfer incoming call to phone #3  [phone
> #1-->phone#3]
> Phone #1 hangup, Phone 2 connect to phone # 3
>
> where the my mistake?
>
> When I  make transfer by xml extensions all works fine
>
>
> ? ?????????
> ?????? ????????
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Dec  1 12:16:51 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Dec 2010 12:16:51 -0800
Subject: [Freeswitch-users] FreeSWITCH "Staff" Conference Call on Thursday
	Dec 2
Message-ID: <AANLkTinmn4WkQ7AcgCQfXHiY_u0z_r2FXkBURhmr8vDJ@mail.gmail.com>

Hello all,

We are planning a 30 minute staff conf call for tomorrow, December 2 at 1PM
EST / 10AM PST. If you are willing and able to assist with the FreeSWITCH
infrastructure maintenance and are ready to roll up your sleeves and do some
work then please join us tomorrow. We will use the FreeSWITCH public
conference:

SIP: 888 at conference.freeswitch.org
PSTN: 1-919-386-9900
Skype: skypiax5

Some of the things we will be discussing:
Who can do what? (i.e. what are you good at and what would you like to learn
how to do)
FS infrastructure overview: what we have, what we are planning to do, etc.
and stuff that needs management:
* git
* JIRA
* Fisheye
* Drupal sites (freeswitch.org, cluecon.com, ostag.org)
* Mailing lists (-users, -dev, etc.)
* other servers (db, conf box, etc.)

We have a lot to go over, but initially we just need to get everything out
on the table. If you want to help please email me off list and provide me
your contact information and what your skills are. We are filling out this
list:
http://wiki.freeswitch.org/wiki/Community_Staff

Thank you all for supporting FreeSWITCH!

-Michael
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From msc at freeswitch.org  Wed Dec  1 12:48:25 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Dec 2010 12:48:25 -0800
Subject: [Freeswitch-users] can't get channel variables after hangup
In-Reply-To: <AE2E563CA6B0424FA874C7E68CC77C2A@e1705>
References: <AE2E563CA6B0424FA874C7E68CC77C2A@e1705>
Message-ID: <AANLkTim3Y+FYRqmy+wyyZKoeOv5keAGoNEe6fPKUTtcK@mail.gmail.com>

This won't work. The "session_in_hangup_hook" literally means that the
$session object is available in the hangup hook script. You need to use Lua,
Perl, or Javascript to access the $session object. What you are doing is
actually trying to access a uuid that does not exist. Once the call is over
it's gone. The "session" object will be available only in the script that is
called in the hangup hook.

Here's a simple script you can tinker with:
-- hook-test.lua


dat = env:serialize()


--freeswitch.consoleLog("INFO","Here you go:\n" .. dat .. "\nn")





uuid = env:getHeader("uuid")


freeswitch.consoleLog("INFO","Inside hangup hook, uuid is: " .. dat .. "\n")





my_var = env:getHeader("my_custom_var")


freeswitch.consoleLog("INFO","my_custom_var is '" .. dat .. "'\n")





api = freeswitch.API()


res = api:execute("uuid_dump",uuid)


freeswitch.consoleLog("INFO","result of 'uuid_dump " .. uuid .. "' is:\n" ..
res .. "\n\n")


Hopefully that will illustrate the issue...
-MC

On Wed, Dec 1, 2010 at 8:53 AM, Madovsky <infos at madovsky.org> wrote:

>  I set this in my dialplan :
>
>    <action application="set" data="session_in_hangup_hook=true"/>
>    <action application="set" data="api_hangup_hook=system
> /usr/local/bin/php /usr/local/freeswitch/scripts/php/test.php ${uuid}"/>
>   and this in test.php:
>
>   exec("/usr/local/freeswitch/bin/fs_cli -x \"uuid_getvar ".$argv[1]."
> test_var\"", $exTabRet, $req);
>
>
>  it results :
>
> -ERR No Such Channel!
>
>
> I tried to replace ${uuid} by ${call_uuid} without success.
>
> Thanks
>
> Franck
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Wed Dec  1 12:59:22 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 1 Dec 2010 15:59:22 -0500
Subject: [Freeswitch-users] can't get channel variables after hangup
References: <AE2E563CA6B0424FA874C7E68CC77C2A@e1705>
	<AANLkTim3Y+FYRqmy+wyyZKoeOv5keAGoNEe6fPKUTtcK@mail.gmail.com>
Message-ID: <96A290BECF924A0FA371E425E0C052DA@e1705>

Thanks Mike,

and with PHP is $session available ?
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, December 01, 2010 3:48 PM
  Subject: Re: [Freeswitch-users] can't get channel variables after hangup


  This won't work. The "session_in_hangup_hook" literally means that the $session object is available in the hangup hook script. You need to use Lua, Perl, or Javascript to access the $session object. What you are doing is actually trying to access a uuid that does not exist. Once the call is over it's gone. The "session" object will be available only in the script that is called in the hangup hook. 


  Here's a simple script you can tinker with:
  -- hook-test.lua                                                                                                                                                
  dat = env:serialize()                                                                                                                                           
  --freeswitch.consoleLog("INFO","Here you go:\n" .. dat .. "\nn")                                                                                                
                                                                                                                                                                  
  uuid = env:getHeader("uuid")                                                                                                                                    
  freeswitch.consoleLog("INFO","Inside hangup hook, uuid is: " .. dat .. "\n")                                                                                    
                                                                                                                                                                  
  my_var = env:getHeader("my_custom_var")                                                                                                                         
  freeswitch.consoleLog("INFO","my_custom_var is '" .. dat .. "'\n")                                                                                              
                                                                                                                                                                  
  api = freeswitch.API()                                                                                                                                          
  res = api:execute("uuid_dump",uuid)                                                                                                                             
  freeswitch.consoleLog("INFO","result of 'uuid_dump " .. uuid .. "' is:\n" .. res .. "\n\n")                                                                     


  Hopefully that will illustrate the issue...
  -MC


  On Wed, Dec 1, 2010 at 8:53 AM, Madovsky <infos at madovsky.org> wrote:

    I set this in my dialplan :

       <action application="set" data="session_in_hangup_hook=true"/>
       <action application="set" data="api_hangup_hook=system /usr/local/bin/php /usr/local/freeswitch/scripts/php/test.php ${uuid}"/>

      and this in test.php:

      exec("/usr/local/freeswitch/bin/fs_cli -x \"uuid_getvar ".$argv[1]."  test_var\"", $exTabRet, $req);


     it results :

    -ERR No Such Channel!


    I tried to replace ${uuid} by ${call_uuid} without success.

    Thanks

    Franck

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






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


  _______________________________________________
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From msc at freeswitch.org  Wed Dec  1 13:11:41 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Dec 2010 13:11:41 -0800
Subject: [Freeswitch-users] can't get channel variables after hangup
In-Reply-To: <96A290BECF924A0FA371E425E0C052DA@e1705>
References: <AE2E563CA6B0424FA874C7E68CC77C2A@e1705>
	<AANLkTim3Y+FYRqmy+wyyZKoeOv5keAGoNEe6fPKUTtcK@mail.gmail.com>
	<96A290BECF924A0FA371E425E0C052DA@e1705>
Message-ID: <AANLkTi=jdYWWbG2i1UDMiQgHXFdYLgPnN6OLOriEgznO@mail.gmail.com>

On Wed, Dec 1, 2010 at 12:59 PM, Madovsky <infos at madovsky.org> wrote:

>  Thanks Mike,
>
> and with PHP is $session available ?
>

No, it's only available in the "built in" languages: Perl, Lua, and
Javascript.
-MC
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From xyangni at gmail.com  Wed Dec  1 13:23:57 2010
From: xyangni at gmail.com (xuyan yang)
Date: Wed, 1 Dec 2010 21:23:57 +0000
Subject: [Freeswitch-users] How to filter Skypopen chat message in ESL mode?
Message-ID: <AANLkTi=T8G8LjKytzpSTw+bjGUZS7_3TAJDySxKevC0+@mail.gmail.com>

hi,

I am writing ESL program on FreeSWITCH Version 1.0.head (git-8825b6e
2010-11-28 17-15-39 -0500)
I need to handle skype chat message with a inbound ESL connection. But there
are 1 or 2 esl events received randomly for each message.
the first short one is alway generated, but the second one is random. So
when trying to filter according to UUID, nothing is caught in many cases.
If I make filter based on skype ID, duplicated messages are often heard.
I do not why the behavior of the second event is random.
How should I setup the filter to get 1 and only 1 event for each chat
message? Thanks.


the first is a short one with the following header:

 [Event-Name] = [MESSAGE]
 [Event-Calling-Function] = [incoming_chatmessage]
 [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
 [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
 [Content-Length] = [1]
 [subject] = [SIMPLE MESSAGE]
 [FreeSWITCH-IPv4] = [192.168.0.3]
 [hint] = [niqizhi]
 [from] = [niqizhi]
 [Event-Date-Local] = [2010-12-01 21:02:37]
 [proto] = [skype]
 [FreeSWITCH-IPv6] = [::1]
 [id] = [5334]
 [Event-Calling-File] = [mod_skypopen.c]
 [Event-Date-Timestamp] = [1291237357051788]
 [FreeSWITCH-Hostname] = [EYSRV]
 [login] = [interface1]
 [during-call] = [true]
 [Event-Calling-Line-Number] = [2915]
 [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]

the second is much longer:

 [Caller-Source] = [mod_skypopen]
 [Event-Calling-Function] = [incoming_chatmessage]
 [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
 [Answer-State] = [answered]
 [FreeSWITCH-IPv4] = [192.168.0.3]
 [Channel-State] = [CS_EXECUTE]
 [Channel-Read-Codec-Bit-Rate] = [256000]
 [FreeSWITCH-IPv6] = [::1]
 [Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
 [Channel-Read-Codec-Rate] = [16000]
 [Caller-Destination-Number] = [5655]
 [Caller-Channel-Transfer-Time] = [0]
 [Channel-Call-State] = [ACTIVE]
 [Caller-Channel-Progress-Media-Time] = [0]
 [FreeSWITCH-Hostname] = [EYSRV]
 [Caller-Channel-Answered-Time] = [1291237326697085]
 [login] = [interface1]
 [during-call] = [true]
 [Channel-Name] = [skypopen/interface1]
 [Caller-Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
 [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
 [Channel-Read-Codec-Name] = [L16]
 [Caller-Channel-Name] = [skypopen/interface1]
 [Caller-Caller-ID-Number] = [niqizhi]
 [Event-Date-Timestamp] = [1291237357051788]
 [Channel-State-Number] = [4]
 [Event-Calling-Line-Number] = [2888]
 [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
 [Event-Name] = [MESSAGE]
 [Content-Length] = [1]
 [subject] = [SIMPLE MESSAGE]
 [Caller-Caller-ID-Name] = [niqizhi]
 [from] = [niqizhi]
 [Caller-Dialplan] = [XML]
 [Caller-Channel-Hangup-Time] = [0]
 [id] = [5334]
 [Caller-Profile-Index] = [1]
 [Caller-Direction] = [inbound]
 [Caller-Username] = [skypopen]
 [Channel-Write-Codec-Name] = [L16]
 [Call-Direction] = [inbound]
 [Caller-Screen-Bit] = [true]
 [hint] = [niqizhi]
 [Caller-Privacy-Hide-Number] = [false]
 [Event-Date-Local] = [2010-12-01 21:02:37]
 [proto] = [skype]
 [Caller-Channel-Created-Time] = [1291237326468855]
 [Event-Calling-File] = [mod_skypopen.c]
 [Caller-Channel-Progress-Time] = [0]
 [Caller-Privacy-Hide-Name] = [false]
 [Channel-Write-Codec-Rate] = [16000]
 [Caller-Context] = [default]
 [Channel-Write-Codec-Bit-Rate] = [256000]
 [Presence-Call-Direction] = [inbound]
 [Caller-Profile-Created-Time] = [1291237326468855]
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From bcxml at hotmail.com  Wed Dec  1 13:25:10 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Wed, 1 Dec 2010 16:25:10 -0500
Subject: [Freeswitch-users] can't get channel variables after hangup
In-Reply-To: <AANLkTi=jdYWWbG2i1UDMiQgHXFdYLgPnN6OLOriEgznO@mail.gmail.com>
References: <AE2E563CA6B0424FA874C7E68CC77C2A@e1705>,
	<AANLkTim3Y+FYRqmy+wyyZKoeOv5keAGoNEe6fPKUTtcK@mail.gmail.com>,
	<96A290BECF924A0FA371E425E0C052DA@e1705>,
	<AANLkTi=jdYWWbG2i1UDMiQgHXFdYLgPnN6OLOriEgznO@mail.gmail.com>
Message-ID: <SNT135-w6572837B46FADE180AF45EB7260@phx.gbl>



 


Date: Wed, 1 Dec 2010 13:11:41 -0800
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] can't get channel variables after hangup




On Wed, Dec 1, 2010 at 12:59 PM, Madovsky <infos at madovsky.org> wrote:



Thanks Mike,
 
and with PHP is $session available ?


No, it's only available in the "built in" languages: Perl, Lua, and Javascript.
-MC


_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 		 	   		  
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From gmaruzz at gmail.com  Wed Dec  1 13:35:18 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 1 Dec 2010 22:35:18 +0100
Subject: [Freeswitch-users] How to filter Skypopen chat message in ESL
	mode?
In-Reply-To: <AANLkTi=T8G8LjKytzpSTw+bjGUZS7_3TAJDySxKevC0+@mail.gmail.com>
References: <AANLkTi=T8G8LjKytzpSTw+bjGUZS7_3TAJDySxKevC0+@mail.gmail.com>
Message-ID: <AANLkTi=HstdSAsVqe836AXpYwvTg8r3_UdovPCbrLUaO@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#MESSAGE_.28Chat.29

On Wed, Dec 1, 2010 at 10:23 PM, xuyan yang <xyangni at gmail.com> wrote:
> hi,
> I am writing ESL program on?FreeSWITCH Version 1.0.head (git-8825b6e
> 2010-11-28 17-15-39 -0500)
> I need to handle skype chat message with a inbound ESL connection. But there
> are 1 or 2 esl events received randomly for each message.
> the first short one is alway generated, but the second one is random. So
> when trying to filter according to UUID, nothing is caught in many cases.
> If I make filter based on skype ID, duplicated messages are often heard.
> I do not why the?behavior?of the second event is random.
> How should I setup the filter to get 1 and only 1 event for each chat
> message? Thanks.
>
> the first is a short one with the following header:
> ?[Event-Name] = [MESSAGE]
> ?[Event-Calling-Function] = [incoming_chatmessage]
> ?[Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
> ?[Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
> ?[Content-Length] = [1]
> ?[subject] = [SIMPLE MESSAGE]
> ?[FreeSWITCH-IPv4] = [192.168.0.3]
> ?[hint] = [niqizhi]
> ?[from] = [niqizhi]
> ?[Event-Date-Local] = [2010-12-01 21:02:37]
> ?[proto] = [skype]
> ?[FreeSWITCH-IPv6] = [::1]
> ?[id] = [5334]
> ?[Event-Calling-File] = [mod_skypopen.c]
> ?[Event-Date-Timestamp] = [1291237357051788]
> ?[FreeSWITCH-Hostname] = [EYSRV]
> ?[login] = [interface1]
> ?[during-call] = [true]
> ?[Event-Calling-Line-Number] = [2915]
> ?[chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
> the second is much longer:
> ?[Caller-Source] = [mod_skypopen]
> ?[Event-Calling-Function] = [incoming_chatmessage]
> ?[Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
> ?[Answer-State] = [answered]
> ?[FreeSWITCH-IPv4] = [192.168.0.3]
> ?[Channel-State] = [CS_EXECUTE]
> ?[Channel-Read-Codec-Bit-Rate] = [256000]
> ?[FreeSWITCH-IPv6] = [::1]
> ?[Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
> ?[Channel-Read-Codec-Rate] = [16000]
> ?[Caller-Destination-Number] = [5655]
> ?[Caller-Channel-Transfer-Time] = [0]
> ?[Channel-Call-State] = [ACTIVE]
> ?[Caller-Channel-Progress-Media-Time] = [0]
> ?[FreeSWITCH-Hostname] = [EYSRV]
> ?[Caller-Channel-Answered-Time] = [1291237326697085]
> ?[login] = [interface1]
> ?[during-call] = [true]
> ?[Channel-Name] = [skypopen/interface1]
> ?[Caller-Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
> ?[Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
> ?[Channel-Read-Codec-Name] = [L16]
> ?[Caller-Channel-Name] = [skypopen/interface1]
> ?[Caller-Caller-ID-Number] = [niqizhi]
> ?[Event-Date-Timestamp] = [1291237357051788]
> ?[Channel-State-Number] = [4]
> ?[Event-Calling-Line-Number] = [2888]
> ?[chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
> ?[Event-Name] = [MESSAGE]
> ?[Content-Length] = [1]
> ?[subject] = [SIMPLE MESSAGE]
> ?[Caller-Caller-ID-Name] = [niqizhi]
> ?[from] = [niqizhi]
> ?[Caller-Dialplan] = [XML]
> ?[Caller-Channel-Hangup-Time] = [0]
> ?[id] = [5334]
> ?[Caller-Profile-Index] = [1]
> ?[Caller-Direction] = [inbound]
> ?[Caller-Username] = [skypopen]
> ?[Channel-Write-Codec-Name] = [L16]
> ?[Call-Direction] = [inbound]
> ?[Caller-Screen-Bit] = [true]
> ?[hint] = [niqizhi]
> ?[Caller-Privacy-Hide-Number] = [false]
> ?[Event-Date-Local] = [2010-12-01 21:02:37]
> ?[proto] = [skype]
> ?[Caller-Channel-Created-Time] = [1291237326468855]
> ?[Event-Calling-File] = [mod_skypopen.c]
> ?[Caller-Channel-Progress-Time] = [0]
> ?[Caller-Privacy-Hide-Name] = [false]
> ?[Channel-Write-Codec-Rate] = [16000]
> ?[Caller-Context] = [default]
> ?[Channel-Write-Codec-Bit-Rate] = [256000]
> ?[Presence-Call-Direction] = [inbound]
> ?[Caller-Profile-Created-Time] = [1291237326468855]
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From bcxml at hotmail.com  Wed Dec  1 13:46:51 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Wed, 1 Dec 2010 16:46:51 -0500
Subject: [Freeswitch-users] Custom Channel Variables via C#
Message-ID: <SNT135-w336CAB94DFC78D328AE6E0B7260@phx.gbl>


 
I can set a custom channel variable in my incomming dial plan like this...
 
<action application='set' data='MyChannelVariable=12345' />
 
And it shows up fine in the CDR
 
I am now attempting to use mod_event_socket to set a custom channel variable and have it appear in the CDR
 
Here is the C# code so far, the code is part of a Microsoft Speech Server application that is answering the call
 
private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
{
  try
  {
    TcpClient newClient = new TcpClient();
 
    newClient.Connect("127.0.0.1", 8021);
 
    NetworkStream tcpStream = newClient.GetStream();
 
    byte[] sendBytes = Encoding.ASCII.GetBytes("<action application='set' data='MyChannelVariable=12345' />");
 
    tcpStream.Write(sendBytes, 0, sendBytes.Length);
 
    tcpStream.Close();
 
    newClient.Close();
  }
  catch(Exception ex)
  {
    throw ex;
  }
}
 
After the call is answered, the C# code seems to run fine, but I dont see the custom channel variable in the resulting CDR
 
I figure I am not setting it correctly
 
Can anyone advise on what I am doing wrong ?
 
Thanks
 
 
Brian
 
  		 	   		  
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From curriegrad2004 at gmail.com  Wed Dec  1 13:50:58 2010
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 1 Dec 2010 13:50:58 -0800
Subject: [Freeswitch-users] mod_spandsp fax question
In-Reply-To: <A781D8F4E37D4073B447724BF5D5C4BC@e1705>
References: <A781D8F4E37D4073B447724BF5D5C4BC@e1705>
Message-ID: <AANLkTikhu=+w-qfAzY76_ESePjMFd2L3TS-i2cBr0585@mail.gmail.com>

Yes, send them in a tiff file with multiple pages and mod_spandsp
should do the trick nicely for you.

On Sun, Nov 28, 2010 at 10:00 PM, Madovsky <infos at madovsky.org> wrote:
> Hi,
>
> is there any way to send multiple fax pages with one call with originate and
> &txfax() ?
>
> Thanks
>
> Franck
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mrene_lists at avgs.ca  Wed Dec  1 13:53:53 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 1 Dec 2010 16:53:53 -0500
Subject: [Freeswitch-users] Custom Channel Variables via C#
In-Reply-To: <SNT135-w336CAB94DFC78D328AE6E0B7260@phx.gbl>
References: <SNT135-w336CAB94DFC78D328AE6E0B7260@phx.gbl>
Message-ID: <0E462B4C-C436-462E-9DA7-9EC1241FB380@avgs.ca>

Hi,

You can't send XML dialplan actions on the socket and expect FreeSWITCH to understand it, you must follow the event socket protocol.

See http://wiki.freeswitch.org/wiki/Event_socket for more information.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-12-01, at 4:46 PM, Brian Campbell wrote:

>  
> I can set a custom channel variable in my incomming dial plan like this...
>  
> <action application='set' data='MyChannelVariable=12345' />
>  
> And it shows up fine in the CDR
>  
> I am now attempting to use mod_event_socket to set a custom channel variable and have it appear in the CDR
>  
> Here is the C# code so far, the code is part of a Microsoft Speech Server application that is answering the call
>  
> private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
> {
>   try
>   {
>     TcpClient newClient = new TcpClient();
>  
>     newClient.Connect("127.0.0.1", 8021);
>  
>     NetworkStream tcpStream = newClient.GetStream();
>  
>     byte[] sendBytes = Encoding.ASCII.GetBytes("<action application='set' data='MyChannelVariable=12345' />");
>  
>     tcpStream.Write(sendBytes, 0, sendBytes.Length);
>  
>     tcpStream.Close();
>  
>     newClient.Close();
>   }
>   catch(Exception ex)
>   {
>     throw ex;
>   }
> }
>  
> After the call is answered, the C# code seems to run fine, but I dont see the custom channel variable in the resulting CDR
>  
> I figure I am not setting it correctly
>  
> Can anyone advise on what I am doing wrong ?
>  
> Thanks
>  
>  
> Brian
>  
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From peder at networkoblivion.com  Wed Dec  1 13:57:32 2010
From: peder at networkoblivion.com (Peder)
Date: Wed, 1 Dec 2010 15:57:32 -0600
Subject: [Freeswitch-users] Fax Issue
Message-ID: <058101cb91a2$c182dee0$44889ca0$@com>

Anybody know how to debug a failed fax issue?

LOG:
http://pastebin.freeswitch.org/14687

VERSION:
FreeSWITCH Version 1.0.head (git-13e8de4 2010-12-01 13-07-47 -0600)

FAX.CONF.XML:
<configuration name="fax.conf" description="FAX application configuration">
    <settings>
        <param name="use-ecm"           value="true"/>
        <param name="verbose"           value="true"/>
        <param name="disable-v17"       value="false"/>
        <param name="ident"             value="SpanDSP Fax Ident"/>
        <param name="header"            value="SpanDSP Fax Header"/>
        <param name="spool-dir"         value="/tmp"/>
        <param name="file-prefix"       value="faxrx"/>
    </settings>
</configuration>

DIALPLAN:
<include>
<extension name="fax_receive">
     <condition field="destination_number" expression="^(6083)$">
        <action application="set" data="api_hangup_hook=system
${base_dir}/scrip
ts/emailfax.php $1 '${caller_id_name}-${caller_id_number}' test at test.com /tm
p/${uuid}.rxfax.tiff"/>
        <action application="answer"/>
        <action application="playback" data="silence_stream://2000"/>
        <action application="rxfax" data="/tmp/${uuid}.rxfax.tiff"/>
        <action application="log" data="CRIT Success - [${fax_success}]"/>
        <action application="log" data="CRIT Rate -
[${fax_transfer_rate}]"/>
        <action application="hangup"/>
     </condition>
   </extension>
</include>


I've got 3-4 different people trying to fax to me and every one fails with a
similar log to what I have above.  I had it working at one point in time,
but I get about 2 faxes a month, so I have no clue when it broke or what
broke it.  The log just seems to show that negotiation failed, but I can't
tell why.

Thanks.

Peder





From slim at thegreek.com  Wed Dec  1 14:12:42 2010
From: slim at thegreek.com (Jeroen C. van Gelderen)
Date: Wed, 1 Dec 2010 17:12:42 -0500
Subject: [Freeswitch-users] "No Digits to send" on purpose?
In-Reply-To: <AANLkTinE1YAOq_k7PG6fTEvpioO3zo+=mMscXcGdSsQp@mail.gmail.com>
References: <6EFE58A23D6B4E65A16462DCDDE5AD1E@mbnet.local>
	<AANLkTinE1YAOq_k7PG6fTEvpioO3zo+=mMscXcGdSsQp@mail.gmail.com>
Message-ID: <782B83D12FAE4495A35C0DEED2EB6A84@mbnet.local>

Hi Daniel,

 

Indeed. I have configured the FXS port in "hotline" mode.

 

As soon as the FXS port goes off-hook it drops into extension 1000. That
extension bridges to the FXO port so that the FXS ports gets the dialtone
from the CO:

 

  <context name="csd-outbound">

    <extension name="csd-outbound">

      <condition field="destination_number" expression="^1000$">

        <action application="export" data="freetdm_disable_dtmf=yes"/>

        <action application="disable_dtmf"/>

        <action application="answer"/>

        <action application="bridge"
data="freetdm/span_fxo_csd/${freetdm_chan_number}/w"/>

        <action application="hangup"/>

      </condition>

    </extension>

  </context>

 

The problem is that Freeswitch in its current incarnation insists that I
send at least one DTMF digit out on the FXO port. I work around this issue
by sending a 'w' "digit" as follows:

 

  <action application="bridge" data="freetdm/span_fxo_csd/1/w"/>

 

This does not actually cause any DTMF tones to be sent out on the FXO port
(so my dialtone is not disturbed) but it does cause a 5-second wait before
the dialtone is heard on the FXS port.

 

What I really need is this:

 

  <action application="bridge" data="freetdm/span_fxo_csd/1/"/>

 

Using this however results in the "No Digits to send" error.

 

Is there a specific reason why FS insists on always sending at least one
digit? What would break if we allow zero-digit DNIS in the bridge call?

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Daniel
Pizarro
Sent: Tuesday, November 30, 2010 10:04
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] "No Digits to send" on purpose?

 

FXS analog devices need a DTMF string to communicate with the PBX and the
PBX could process the call (bridge) ...at least you need a "batman phone"
with auto-dialing (inmmediate). 



2010/11/29 Jeroen C. van Gelderen <slim at thegreek.com>

Hi Guys,

I need to bridge an FXS port (when taken off-hook) to an FXO port without
dialing any digits like so:

<action application="bridge" data="freetdm/span_fxo_csd/1"/>

This results in the following error:

ftmod/ftmod_analog/ftmod_analog.c:

if (ftdm_strlen_zero(ftdmchan->caller_data.dnis.digits)) {
 ftdm_log_chan_msg(ftdmchan, FTDM_LOG_ERROR, "No Digits to send!\n");

For now I have worked around this by using

<action application="bridge" data="freetdm/span_fxo_csd/1/w"/>
or
<action application="bridge" data="freetdm/span_fxo_csd/1/F"/>

I would like to avoid the unnecessary delay this incurs.

Any suggestions? Under what circumstances is it bad to send zero digits? Can
I simply take out the check for zero-length dnis.digits?

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen



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

 

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From djbinter at gmail.com  Wed Dec  1 14:12:55 2010
From: djbinter at gmail.com (DJB International)
Date: Wed, 1 Dec 2010 14:12:55 -0800
Subject: [Freeswitch-users] Fax Issue
In-Reply-To: <058101cb91a2$c182dee0$44889ca0$@com>
References: <058101cb91a2$c182dee0$44889ca0$@com>
Message-ID: <AANLkTinJhs-9Q6wyTHqLLbfQda2n0bm5XwVpuG50nqoZ@mail.gmail.com>

You probably might need to set fax_enable_t38_request=true in your dialplan.

-djbinter


On Wed, Dec 1, 2010 at 1:57 PM, Peder <peder at networkoblivion.com> wrote:

> Anybody know how to debug a failed fax issue?
>
> LOG:
> http://pastebin.freeswitch.org/14687
>
> VERSION:
> FreeSWITCH Version 1.0.head (git-13e8de4 2010-12-01 13-07-47 -0600)
>
> FAX.CONF.XML:
> <configuration name="fax.conf" description="FAX application configuration">
>    <settings>
>        <param name="use-ecm"           value="true"/>
>        <param name="verbose"           value="true"/>
>        <param name="disable-v17"       value="false"/>
>        <param name="ident"             value="SpanDSP Fax Ident"/>
>        <param name="header"            value="SpanDSP Fax Header"/>
>        <param name="spool-dir"         value="/tmp"/>
>        <param name="file-prefix"       value="faxrx"/>
>    </settings>
> </configuration>
>
> DIALPLAN:
> <include>
> <extension name="fax_receive">
>     <condition field="destination_number" expression="^(6083)$">
>        <action application="set" data="api_hangup_hook=system
> ${base_dir}/scrip
> ts/emailfax.php $1 '${caller_id_name}-${caller_id_number}' test at test.com/tm
> p/${uuid}.rxfax.tiff"/>
>        <action application="answer"/>
>        <action application="playback" data="silence_stream://2000"/>
>        <action application="rxfax" data="/tmp/${uuid}.rxfax.tiff"/>
>        <action application="log" data="CRIT Success - [${fax_success}]"/>
>        <action application="log" data="CRIT Rate -
> [${fax_transfer_rate}]"/>
>        <action application="hangup"/>
>     </condition>
>   </extension>
> </include>
>
>
> I've got 3-4 different people trying to fax to me and every one fails with
> a
> similar log to what I have above.  I had it working at one point in time,
> but I get about 2 faxes a month, so I have no clue when it broke or what
> broke it.  The log just seems to show that negotiation failed, but I can't
> tell why.
>
> Thanks.
>
> Peder
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Dec  1 14:22:35 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 1 Dec 2010 16:22:35 -0600
Subject: [Freeswitch-users] "No Digits to send" on purpose?
In-Reply-To: <782B83D12FAE4495A35C0DEED2EB6A84@mbnet.local>
References: <6EFE58A23D6B4E65A16462DCDDE5AD1E@mbnet.local>
	<AANLkTinE1YAOq_k7PG6fTEvpioO3zo+=mMscXcGdSsQp@mail.gmail.com>
	<782B83D12FAE4495A35C0DEED2EB6A84@mbnet.local>
Message-ID: <FB8961D5-AA00-4C75-87E0-7049C2F332A8@freeswitch.org>

This is NOT specifically a FreeSWITCH issue its mod_freetdm that has the parsing setup to require digits... small bug on jira would get this resolved.  http://jira.freeswitch.org

/b

On Dec 1, 2010, at 4:12 PM, Jeroen C. van Gelderen wrote:

> The problem is that Freeswitch in its current incarnation insists that I send at least one DTMF digit out on the FXO port. I work around this issue by sending a ?w? ?digit? as follows:
>  
>   <action application="bridge" data="freetdm/span_fxo_csd/1/w"/>
>  
> This does not actually cause any DTMF tones to be sent out on the FXO port (so my dialtone is not disturbed) but it does cause a 5-second wait before the dialtone is heard on the FXS port.
>  
> What I really need is this:
>  
>   <action application="bridge" data="freetdm/span_fxo_csd/1/"/>
>  
> Using this however results in the "No Digits to send" error.

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From xyangni at gmail.com  Wed Dec  1 14:35:18 2010
From: xyangni at gmail.com (xuyan yang)
Date: Wed, 1 Dec 2010 22:35:18 +0000
Subject: [Freeswitch-users] How to filter Skypopen chat message in ESL
	mode?
In-Reply-To: <AANLkTi=HstdSAsVqe836AXpYwvTg8r3_UdovPCbrLUaO@mail.gmail.com>
References: <AANLkTi=T8G8LjKytzpSTw+bjGUZS7_3TAJDySxKevC0+@mail.gmail.com>
	<AANLkTi=HstdSAsVqe836AXpYwvTg8r3_UdovPCbrLUaO@mail.gmail.com>
Message-ID: <AANLkTimW=fueqE-iMhPQ03+UEO5dj1-Mghdj7b-ZoM1-@mail.gmail.com>

Thanks for your reply. I have read this page before. but the random emerging
of the second verbose esl events is causing trouble. And I guess it may be a
bug, or it should be predictable.
Before reporting to jira, I just want to check whether I have made any
mistake.


On Wed, Dec 1, 2010 at 9:35 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

>
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#MESSAGE_.28Chat.29
>
> On Wed, Dec 1, 2010 at 10:23 PM, xuyan yang <xyangni at gmail.com> wrote:
> > hi,
> > I am writing ESL program on FreeSWITCH Version 1.0.head (git-8825b6e
> > 2010-11-28 17-15-39 -0500)
> > I need to handle skype chat message with a inbound ESL connection. But
> there
> > are 1 or 2 esl events received randomly for each message.
> > the first short one is alway generated, but the second one is random. So
> > when trying to filter according to UUID, nothing is caught in many cases.
> > If I make filter based on skype ID, duplicated messages are often heard.
> > I do not why the behavior of the second event is random.
> > How should I setup the filter to get 1 and only 1 event for each chat
> > message? Thanks.
> >
> > the first is a short one with the following header:
> >  [Event-Name] = [MESSAGE]
> >  [Event-Calling-Function] = [incoming_chatmessage]
> >  [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
> >  [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
> >  [Content-Length] = [1]
> >  [subject] = [SIMPLE MESSAGE]
> >  [FreeSWITCH-IPv4] = [192.168.0.3]
> >  [hint] = [niqizhi]
> >  [from] = [niqizhi]
> >  [Event-Date-Local] = [2010-12-01 21:02:37]
> >  [proto] = [skype]
> >  [FreeSWITCH-IPv6] = [::1]
> >  [id] = [5334]
> >  [Event-Calling-File] = [mod_skypopen.c]
> >  [Event-Date-Timestamp] = [1291237357051788]
> >  [FreeSWITCH-Hostname] = [EYSRV]
> >  [login] = [interface1]
> >  [during-call] = [true]
> >  [Event-Calling-Line-Number] = [2915]
> >  [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
> > the second is much longer:
> >  [Caller-Source] = [mod_skypopen]
> >  [Event-Calling-Function] = [incoming_chatmessage]
> >  [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
> >  [Answer-State] = [answered]
> >  [FreeSWITCH-IPv4] = [192.168.0.3]
> >  [Channel-State] = [CS_EXECUTE]
> >  [Channel-Read-Codec-Bit-Rate] = [256000]
> >  [FreeSWITCH-IPv6] = [::1]
> >  [Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
> >  [Channel-Read-Codec-Rate] = [16000]
> >  [Caller-Destination-Number] = [5655]
> >  [Caller-Channel-Transfer-Time] = [0]
> >  [Channel-Call-State] = [ACTIVE]
> >  [Caller-Channel-Progress-Media-Time] = [0]
> >  [FreeSWITCH-Hostname] = [EYSRV]
> >  [Caller-Channel-Answered-Time] = [1291237326697085]
> >  [login] = [interface1]
> >  [during-call] = [true]
> >  [Channel-Name] = [skypopen/interface1]
> >  [Caller-Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
> >  [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
> >  [Channel-Read-Codec-Name] = [L16]
> >  [Caller-Channel-Name] = [skypopen/interface1]
> >  [Caller-Caller-ID-Number] = [niqizhi]
> >  [Event-Date-Timestamp] = [1291237357051788]
> >  [Channel-State-Number] = [4]
> >  [Event-Calling-Line-Number] = [2888]
> >  [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
> >  [Event-Name] = [MESSAGE]
> >  [Content-Length] = [1]
> >  [subject] = [SIMPLE MESSAGE]
> >  [Caller-Caller-ID-Name] = [niqizhi]
> >  [from] = [niqizhi]
> >  [Caller-Dialplan] = [XML]
> >  [Caller-Channel-Hangup-Time] = [0]
> >  [id] = [5334]
> >  [Caller-Profile-Index] = [1]
> >  [Caller-Direction] = [inbound]
> >  [Caller-Username] = [skypopen]
> >  [Channel-Write-Codec-Name] = [L16]
> >  [Call-Direction] = [inbound]
> >  [Caller-Screen-Bit] = [true]
> >  [hint] = [niqizhi]
> >  [Caller-Privacy-Hide-Number] = [false]
> >  [Event-Date-Local] = [2010-12-01 21:02:37]
> >  [proto] = [skype]
> >  [Caller-Channel-Created-Time] = [1291237326468855]
> >  [Event-Calling-File] = [mod_skypopen.c]
> >  [Caller-Channel-Progress-Time] = [0]
> >  [Caller-Privacy-Hide-Name] = [false]
> >  [Channel-Write-Codec-Rate] = [16000]
> >  [Caller-Context] = [default]
> >  [Channel-Write-Codec-Bit-Rate] = [256000]
> >  [Presence-Call-Direction] = [inbound]
> >  [Caller-Profile-Created-Time] = [1291237326468855]
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peder at networkoblivion.com  Wed Dec  1 14:40:00 2010
From: peder at networkoblivion.com (Peder)
Date: Wed, 1 Dec 2010 16:40:00 -0600
Subject: [Freeswitch-users] Fax Issue
In-Reply-To: <AANLkTinJhs-9Q6wyTHqLLbfQda2n0bm5XwVpuG50nqoZ@mail.gmail.com>
References: <058101cb91a2$c182dee0$44889ca0$@com>
	<AANLkTinJhs-9Q6wyTHqLLbfQda2n0bm5XwVpuG50nqoZ@mail.gmail.com>
Message-ID: <062001cb91a8$b0549ea0$10fddbe0$@com>

I don't recall ever having that in there before, but I must have and deleted
it somehow.  FYI, I added two lines to the dialplan and it works now:

 

<action application="set" data="fax_enable_t38_request=true"/>

<action application="set" data="fax_enable_t38=true"/>

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of DJB
International
Sent: Wednesday, December 01, 2010 4:13 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Fax Issue

 

You probably might need to set fax_enable_t38_request=true in your dialplan.

-djbinter



On Wed, Dec 1, 2010 at 1:57 PM, Peder <peder at networkoblivion.com> wrote:

Anybody know how to debug a failed fax issue?

LOG:
http://pastebin.freeswitch.org/14687

VERSION:
FreeSWITCH Version 1.0.head (git-13e8de4 2010-12-01 13-07-47 -0600)

FAX.CONF.XML:
<configuration name="fax.conf" description="FAX application configuration">
   <settings>
       <param name="use-ecm"           value="true"/>
       <param name="verbose"           value="true"/>
       <param name="disable-v17"       value="false"/>
       <param name="ident"             value="SpanDSP Fax Ident"/>
       <param name="header"            value="SpanDSP Fax Header"/>
       <param name="spool-dir"         value="/tmp"/>
       <param name="file-prefix"       value="faxrx"/>
   </settings>
</configuration>

DIALPLAN:
<include>
<extension name="fax_receive">
    <condition field="destination_number" expression="^(6083)$">
       <action application="set" data="api_hangup_hook=system
${base_dir}/scrip
ts/emailfax.php $1 '${caller_id_name}-${caller_id_number}' test at test.com /tm
p/${uuid}.rxfax.tiff"/>
       <action application="answer"/>
       <action application="playback" data="silence_stream://2000"/>
       <action application="rxfax" data="/tmp/${uuid}.rxfax.tiff"/>
       <action application="log" data="CRIT Success - [${fax_success}]"/>
       <action application="log" data="CRIT Rate -
[${fax_transfer_rate}]"/>
       <action application="hangup"/>
    </condition>
  </extension>
</include>


I've got 3-4 different people trying to fax to me and every one fails with a
similar log to what I have above.  I had it working at one point in time,
but I get about 2 faxes a month, so I have no clue when it broke or what
broke it.  The log just seems to show that negotiation failed, but I can't
tell why.

Thanks.

Peder




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

 

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From gmaruzz at celliax.org  Wed Dec  1 15:15:43 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 2 Dec 2010 00:15:43 +0100
Subject: [Freeswitch-users] How to filter Skypopen chat message in ESL
	mode?
In-Reply-To: <AANLkTimW=fueqE-iMhPQ03+UEO5dj1-Mghdj7b-ZoM1-@mail.gmail.com>
References: <AANLkTi=T8G8LjKytzpSTw+bjGUZS7_3TAJDySxKevC0+@mail.gmail.com>
	<AANLkTi=HstdSAsVqe836AXpYwvTg8r3_UdovPCbrLUaO@mail.gmail.com>
	<AANLkTimW=fueqE-iMhPQ03+UEO5dj1-Mghdj7b-ZoM1-@mail.gmail.com>
Message-ID: <AANLkTik5fMn4Yi_bU_KFKyufdJ1b13AOBD2-H47UHNWQ@mail.gmail.com>

If you subscribe only to "MESSAGE" kind of events, you'll get only those.

The other events are "raw" events, that other users have requested for
other purposes.

-giovanni

On Wed, Dec 1, 2010 at 11:35 PM, xuyan yang <xyangni at gmail.com> wrote:
> Thanks for your reply. I have read this page before. but the random emerging
> of the second verbose esl events is causing trouble. And I guess it may be a
> bug, or it should be predictable.
> Before reporting to jira, I just want to check whether I have made any
> mistake.
>
>
> On Wed, Dec 1, 2010 at 9:35 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>>
>>
>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#MESSAGE_.28Chat.29
>>
>> On Wed, Dec 1, 2010 at 10:23 PM, xuyan yang <xyangni at gmail.com> wrote:
>> > hi,
>> > I am writing ESL program on?FreeSWITCH Version 1.0.head (git-8825b6e
>> > 2010-11-28 17-15-39 -0500)
>> > I need to handle skype chat message with a inbound ESL connection. But
>> > there
>> > are 1 or 2 esl events received randomly for each message.
>> > the first short one is alway generated, but the second one is random. So
>> > when trying to filter according to UUID, nothing is caught in many
>> > cases.
>> > If I make filter based on skype ID, duplicated messages are often heard.
>> > I do not why the?behavior?of the second event is random.
>> > How should I setup the filter to get 1 and only 1 event for each chat
>> > message? Thanks.
>> >
>> > the first is a short one with the following header:
>> > ?[Event-Name] = [MESSAGE]
>> > ?[Event-Calling-Function] = [incoming_chatmessage]
>> > ?[Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
>> > ?[Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
>> > ?[Content-Length] = [1]
>> > ?[subject] = [SIMPLE MESSAGE]
>> > ?[FreeSWITCH-IPv4] = [192.168.0.3]
>> > ?[hint] = [niqizhi]
>> > ?[from] = [niqizhi]
>> > ?[Event-Date-Local] = [2010-12-01 21:02:37]
>> > ?[proto] = [skype]
>> > ?[FreeSWITCH-IPv6] = [::1]
>> > ?[id] = [5334]
>> > ?[Event-Calling-File] = [mod_skypopen.c]
>> > ?[Event-Date-Timestamp] = [1291237357051788]
>> > ?[FreeSWITCH-Hostname] = [EYSRV]
>> > ?[login] = [interface1]
>> > ?[during-call] = [true]
>> > ?[Event-Calling-Line-Number] = [2915]
>> > ?[chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
>> > the second is much longer:
>> > ?[Caller-Source] = [mod_skypopen]
>> > ?[Event-Calling-Function] = [incoming_chatmessage]
>> > ?[Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
>> > ?[Answer-State] = [answered]
>> > ?[FreeSWITCH-IPv4] = [192.168.0.3]
>> > ?[Channel-State] = [CS_EXECUTE]
>> > ?[Channel-Read-Codec-Bit-Rate] = [256000]
>> > ?[FreeSWITCH-IPv6] = [::1]
>> > ?[Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
>> > ?[Channel-Read-Codec-Rate] = [16000]
>> > ?[Caller-Destination-Number] = [5655]
>> > ?[Caller-Channel-Transfer-Time] = [0]
>> > ?[Channel-Call-State] = [ACTIVE]
>> > ?[Caller-Channel-Progress-Media-Time] = [0]
>> > ?[FreeSWITCH-Hostname] = [EYSRV]
>> > ?[Caller-Channel-Answered-Time] = [1291237326697085]
>> > ?[login] = [interface1]
>> > ?[during-call] = [true]
>> > ?[Channel-Name] = [skypopen/interface1]
>> > ?[Caller-Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
>> > ?[Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
>> > ?[Channel-Read-Codec-Name] = [L16]
>> > ?[Caller-Channel-Name] = [skypopen/interface1]
>> > ?[Caller-Caller-ID-Number] = [niqizhi]
>> > ?[Event-Date-Timestamp] = [1291237357051788]
>> > ?[Channel-State-Number] = [4]
>> > ?[Event-Calling-Line-Number] = [2888]
>> > ?[chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
>> > ?[Event-Name] = [MESSAGE]
>> > ?[Content-Length] = [1]
>> > ?[subject] = [SIMPLE MESSAGE]
>> > ?[Caller-Caller-ID-Name] = [niqizhi]
>> > ?[from] = [niqizhi]
>> > ?[Caller-Dialplan] = [XML]
>> > ?[Caller-Channel-Hangup-Time] = [0]
>> > ?[id] = [5334]
>> > ?[Caller-Profile-Index] = [1]
>> > ?[Caller-Direction] = [inbound]
>> > ?[Caller-Username] = [skypopen]
>> > ?[Channel-Write-Codec-Name] = [L16]
>> > ?[Call-Direction] = [inbound]
>> > ?[Caller-Screen-Bit] = [true]
>> > ?[hint] = [niqizhi]
>> > ?[Caller-Privacy-Hide-Number] = [false]
>> > ?[Event-Date-Local] = [2010-12-01 21:02:37]
>> > ?[proto] = [skype]
>> > ?[Caller-Channel-Created-Time] = [1291237326468855]
>> > ?[Event-Calling-File] = [mod_skypopen.c]
>> > ?[Caller-Channel-Progress-Time] = [0]
>> > ?[Caller-Privacy-Hide-Name] = [false]
>> > ?[Channel-Write-Codec-Rate] = [16000]
>> > ?[Caller-Context] = [default]
>> > ?[Channel-Write-Codec-Bit-Rate] = [256000]
>> > ?[Presence-Call-Direction] = [inbound]
>> > ?[Caller-Profile-Created-Time] = [1291237326468855]
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Wed Dec  1 15:18:40 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 2 Dec 2010 00:18:40 +0100
Subject: [Freeswitch-users] How to filter Skypopen chat message in ESL
	mode?
In-Reply-To: <AANLkTik5fMn4Yi_bU_KFKyufdJ1b13AOBD2-H47UHNWQ@mail.gmail.com>
References: <AANLkTi=T8G8LjKytzpSTw+bjGUZS7_3TAJDySxKevC0+@mail.gmail.com>
	<AANLkTi=HstdSAsVqe836AXpYwvTg8r3_UdovPCbrLUaO@mail.gmail.com>
	<AANLkTimW=fueqE-iMhPQ03+UEO5dj1-Mghdj7b-ZoM1-@mail.gmail.com>
	<AANLkTik5fMn4Yi_bU_KFKyufdJ1b13AOBD2-H47UHNWQ@mail.gmail.com>
Message-ID: <AANLkTi=eE+AQ-F-xndR-6zBajomHY-a5A_8D1ULrZwM_@mail.gmail.com>

On Thu, Dec 2, 2010 at 12:15 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
> If you subscribe only to "MESSAGE" kind of events, you'll get only those.
>
> The other events are "raw" events, that other users have requested for
> other purposes.

or at  least that is the expected behavior, please let me know if I
introduced some regression in integrating that "raw event" thingy.

-giovanni

>
> -giovanni
>
> On Wed, Dec 1, 2010 at 11:35 PM, xuyan yang <xyangni at gmail.com> wrote:
>> Thanks for your reply. I have read this page before. but the random emerging
>> of the second verbose esl events is causing trouble. And I guess it may be a
>> bug, or it should be predictable.
>> Before reporting to jira, I just want to check whether I have made any
>> mistake.
>>
>>
>> On Wed, Dec 1, 2010 at 9:35 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>>
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#MESSAGE_.28Chat.29
>>>
>>> On Wed, Dec 1, 2010 at 10:23 PM, xuyan yang <xyangni at gmail.com> wrote:
>>> > hi,
>>> > I am writing ESL program on?FreeSWITCH Version 1.0.head (git-8825b6e
>>> > 2010-11-28 17-15-39 -0500)
>>> > I need to handle skype chat message with a inbound ESL connection. But
>>> > there
>>> > are 1 or 2 esl events received randomly for each message.
>>> > the first short one is alway generated, but the second one is random. So
>>> > when trying to filter according to UUID, nothing is caught in many
>>> > cases.
>>> > If I make filter based on skype ID, duplicated messages are often heard.
>>> > I do not why the?behavior?of the second event is random.
>>> > How should I setup the filter to get 1 and only 1 event for each chat
>>> > message? Thanks.
>>> >
>>> > the first is a short one with the following header:
>>> > ?[Event-Name] = [MESSAGE]
>>> > ?[Event-Calling-Function] = [incoming_chatmessage]
>>> > ?[Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
>>> > ?[Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
>>> > ?[Content-Length] = [1]
>>> > ?[subject] = [SIMPLE MESSAGE]
>>> > ?[FreeSWITCH-IPv4] = [192.168.0.3]
>>> > ?[hint] = [niqizhi]
>>> > ?[from] = [niqizhi]
>>> > ?[Event-Date-Local] = [2010-12-01 21:02:37]
>>> > ?[proto] = [skype]
>>> > ?[FreeSWITCH-IPv6] = [::1]
>>> > ?[id] = [5334]
>>> > ?[Event-Calling-File] = [mod_skypopen.c]
>>> > ?[Event-Date-Timestamp] = [1291237357051788]
>>> > ?[FreeSWITCH-Hostname] = [EYSRV]
>>> > ?[login] = [interface1]
>>> > ?[during-call] = [true]
>>> > ?[Event-Calling-Line-Number] = [2915]
>>> > ?[chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
>>> > the second is much longer:
>>> > ?[Caller-Source] = [mod_skypopen]
>>> > ?[Event-Calling-Function] = [incoming_chatmessage]
>>> > ?[Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
>>> > ?[Answer-State] = [answered]
>>> > ?[FreeSWITCH-IPv4] = [192.168.0.3]
>>> > ?[Channel-State] = [CS_EXECUTE]
>>> > ?[Channel-Read-Codec-Bit-Rate] = [256000]
>>> > ?[FreeSWITCH-IPv6] = [::1]
>>> > ?[Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
>>> > ?[Channel-Read-Codec-Rate] = [16000]
>>> > ?[Caller-Destination-Number] = [5655]
>>> > ?[Caller-Channel-Transfer-Time] = [0]
>>> > ?[Channel-Call-State] = [ACTIVE]
>>> > ?[Caller-Channel-Progress-Media-Time] = [0]
>>> > ?[FreeSWITCH-Hostname] = [EYSRV]
>>> > ?[Caller-Channel-Answered-Time] = [1291237326697085]
>>> > ?[login] = [interface1]
>>> > ?[during-call] = [true]
>>> > ?[Channel-Name] = [skypopen/interface1]
>>> > ?[Caller-Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
>>> > ?[Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
>>> > ?[Channel-Read-Codec-Name] = [L16]
>>> > ?[Caller-Channel-Name] = [skypopen/interface1]
>>> > ?[Caller-Caller-ID-Number] = [niqizhi]
>>> > ?[Event-Date-Timestamp] = [1291237357051788]
>>> > ?[Channel-State-Number] = [4]
>>> > ?[Event-Calling-Line-Number] = [2888]
>>> > ?[chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
>>> > ?[Event-Name] = [MESSAGE]
>>> > ?[Content-Length] = [1]
>>> > ?[subject] = [SIMPLE MESSAGE]
>>> > ?[Caller-Caller-ID-Name] = [niqizhi]
>>> > ?[from] = [niqizhi]
>>> > ?[Caller-Dialplan] = [XML]
>>> > ?[Caller-Channel-Hangup-Time] = [0]
>>> > ?[id] = [5334]
>>> > ?[Caller-Profile-Index] = [1]
>>> > ?[Caller-Direction] = [inbound]
>>> > ?[Caller-Username] = [skypopen]
>>> > ?[Channel-Write-Codec-Name] = [L16]
>>> > ?[Call-Direction] = [inbound]
>>> > ?[Caller-Screen-Bit] = [true]
>>> > ?[hint] = [niqizhi]
>>> > ?[Caller-Privacy-Hide-Number] = [false]
>>> > ?[Event-Date-Local] = [2010-12-01 21:02:37]
>>> > ?[proto] = [skype]
>>> > ?[Caller-Channel-Created-Time] = [1291237326468855]
>>> > ?[Event-Calling-File] = [mod_skypopen.c]
>>> > ?[Caller-Channel-Progress-Time] = [0]
>>> > ?[Caller-Privacy-Hide-Name] = [false]
>>> > ?[Channel-Write-Codec-Rate] = [16000]
>>> > ?[Caller-Context] = [default]
>>> > ?[Channel-Write-Codec-Bit-Rate] = [256000]
>>> > ?[Presence-Call-Direction] = [inbound]
>>> > ?[Caller-Profile-Created-Time] = [1291237326468855]
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From steveayre at gmail.com  Wed Dec  1 15:27:40 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 1 Dec 2010 23:27:40 +0000
Subject: [Freeswitch-users] session_record
In-Reply-To: <37890.5730.qm@web55706.mail.re3.yahoo.com>
References: <37890.5730.qm@web55706.mail.re3.yahoo.com>
Message-ID: <B932268F-1DD2-498C-94BB-166C786B6E74@gmail.com>

Is that your entire script?

Once the record_session executes the script will continue, reach the end. The call will then hangup.

Look at calling session.sleep(ms) within a loop while session.ready() is true.

Steve on iPhone

On 1 Dec 2010, at 19:14, Will Smith <willbelair at yahoo.com> wrote:

> hi ,
> I have a problem to record files.
> here is my javascript:
> ---------------
> session = new Session('sofia/gateway/mygateway/14164516999');
> session.waitForAnswer(10000);
> 
> if (session.ready()) {
>     session.execute("record_session", "/usr/local/freeswitch/recordings/" + session.uuid + ".wav");     
> }
> ---------------
> after press answer key on my keypad, the call disconnected. What I am missing here?
> 
> Thank you for your help.
> 
> Will
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From steveayre at gmail.com  Wed Dec  1 15:31:58 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 1 Dec 2010 23:31:58 +0000
Subject: [Freeswitch-users] manage FS's users by an Java application
In-Reply-To: <AANLkTimmHFdQAa_Cm6sXT0E++XJL2-ek_cFL2p-8JjfN@mail.gmail.com>
References: <AANLkTimmHFdQAa_Cm6sXT0E++XJL2-ek_cFL2p-8JjfN@mail.gmail.com>
Message-ID: <1D0CF5F4-1851-4188-A9DB-183D28A795FD@gmail.com>

Either edit the files from Java and call reloadxml via ESL, or serve the user directory from the Java application using mod_xml_curl (directory lookups are done via a XML request/response over HTTP).

Steve on iPhone

On 27 Nov 2010, at 13:56, hoailt <hoai0906 at gmail.com> wrote:

> Hi, 
> 
> Normally, we create user by modify "/usr/local/freeswitch/conf/directory/default/*.xml" file, and call "reloadxml" command from FS console. 
> 
> But now, I want to write a Java application to manage FS users (create, delete, modify). So, could you suggest me what module/api/command I should use ?
> 
> -- 
> Luong Thanh Hoai
> +84983454810
> Senior Software Engineer
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Wed Dec  1 17:00:30 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Dec 2010 17:00:30 -0800
Subject: [Freeswitch-users] session_record
In-Reply-To: <B932268F-1DD2-498C-94BB-166C786B6E74@gmail.com>
References: <37890.5730.qm@web55706.mail.re3.yahoo.com>
	<B932268F-1DD2-498C-94BB-166C786B6E74@gmail.com>
Message-ID: <AANLkTimkx47OifsEJK4ZsW9h5pn85PwU+w3BE=chmg1D@mail.gmail.com>

FYI,

Will and I figured it out. It was much more involved. Long story short we
did a combo of originate on the cmd line and having a long
destination_number that gets parsed up into several values (target
extension, password, etc.). It seems to be working. Once it's tested we'll
throw the example up on the wiki

-MC

On Wed, Dec 1, 2010 at 3:27 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Is that your entire script?
>
> Once the record_session executes the script will continue, reach the end.
> The call will then hangup.
>
> Look at calling session.sleep(ms) within a loop while session.ready() is
> true.
>
> Steve on iPhone
>
> On 1 Dec 2010, at 19:14, Will Smith <willbelair at yahoo.com> wrote:
>
> hi ,
> I have a problem to record files.
> here is my javascript:
> ---------------
> session = new Session('sofia/gateway/mygateway/14164516999');
> session.waitForAnswer(10000);
>
> if (session.ready()) {
>     session.execute("record_session", "/usr/local/freeswitch/recordings/" +
> session.uuid + ".wav");
> }
> ---------------
> after press answer key on my keypad, the call disconnected. What I am
> missing here?
>
> Thank you for your help.
>
> Will
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Dec  1 17:03:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Dec 2010 17:03:00 -0800
Subject: [Freeswitch-users] loopback question
In-Reply-To: <918F31117A894F2A917947AF7940507A@e1705>
References: <918F31117A894F2A917947AF7940507A@e1705>
Message-ID: <AANLkTin4Me4zC1x=g2d3hQ3Bg3W+EU+5vuaA_aiEKt0J@mail.gmail.com>

I'm guessing that "sip_to_uri" may not be matching. Pastebin the debug log
of the call failing and we'll take a look.

-MC

On Tue, Nov 30, 2010 at 3:01 PM, Madovsky <infos at madovsky.org> wrote:

>
> Hi,
>
> what is the logical to send a fax outside from gateway with originate that
> sends the call to the default dialplan ?
>
> I tried ths :
>
> originalte {filetofax=myfax.tiff}loopback/0000numberToSendFax/default
> 99999999
>
> and put this in default dialplan
>
> <extension name="send_fax">
>   <condition field="${sip_to_uri}"
> expression="^\+([2-9]\d{10,15})@$${domain}$">
>    <action application="set" data="hangup_after_bridge=false"/>
>    <action application="set" data="continue_on_fail=false"/>
>    <action application="ring_ready"/>
>    <action application="set" data="absolute_codec_string=PCMU"/>
>    <action inline="true" application="set" data="fax_file=${fax_file}"/>
>    <action application="transfer" data="sendfax XML features"/>
>    </condition>
> </extension>
>
>
> but doesn't work.
>
> Any ?
>
> Thanks
>
> Franck
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Dec  1 17:04:31 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Dec 2010 17:04:31 -0800
Subject: [Freeswitch-users] hangup_after_hook after txfax
In-Reply-To: <926F75A9D9D345F790A101E798C2795A@e1705>
References: <926F75A9D9D345F790A101E798C2795A@e1705>
Message-ID: <AANLkTiknk3PwpQOEhcrBRf1BqRGEdtYiOe=02TAhKuiB@mail.gmail.com>

The answer to this question is the same as the answer to the other one: need
to use the session_in_hangup_hook and process it in Lua, Perl, or js.
-MC

On Tue, Nov 30, 2010 at 8:27 PM, Madovsky <infos at madovsky.org> wrote:

>  I'm trying to get the mod_span vaialbes after fax sent
> fax_success and fax_result_text with following in dialplan
>
> <action application="set" data="api_hangup_hook=system /usr/local/bin/php
> /usr/local/freeswitch/scripts/phptest.php ${dialed_extension} ${fax_file}
> ${fax_success} ${fax_result_text}"/>
> but these vars are empty.
>
> is there any example to get these vars after hangup ?
>
> Thanks
>
> Franck
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Dec  1 17:08:54 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Dec 2010 17:08:54 -0800
Subject: [Freeswitch-users] mod_callcenter and peridic announcements
In-Reply-To: <2601137.175251291218243111.JavaMail.root@pina>
References: <2601137.175251291218243111.JavaMail.root@pina>
Message-ID: <AANLkTik6C5UK8G=29dnFD5fj_FJESM=VcAxac8bjYAF=@mail.gmail.com>

mod_fifo supports periodic announcements. mod_fifo and mod_callcenter are
two different ways to accomplish the same basic task, namely to queue up
incoming calls and pass them to agents as efficiently as possible.

-MC

On Wed, Dec 1, 2010 at 7:44 AM, Raimund Sacherer <
raimund.sacherer at logitravel.com> wrote:

> Hello Freeswitch List.
>
> We are evaluating switching our asterisk based call center (around 150
> agents) over to freeswitch.
>
> Among other things I noticed that the mod_callcenter module does not have
> the possibility to use periodic announcements. Could these be implemented
> somehow ... like, with ESL or something else?
>
>
> I want to try out the Spice Technologies (or is it now OpenACD) callcenter
> solution, but this will be a further project as we first want to switch to
> freeswitch as soon as possible as we have instability problems which we want
> to get rid off.
>
> Thank you for your support,
>
> Ray
>
>
> --
> Raimund Sacherer
> Dpto. de Sistemas
> Agencia de Viajes Online
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Wed Dec  1 17:18:01 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 1 Dec 2010 19:18:01 -0600
Subject: [Freeswitch-users] ringback on aleg during bleg_app
In-Reply-To: <AANLkTikibrHRVRnUObROismBMBOs3gK42cSgVdLxCBbq@mail.gmail.com>
References: <AANLkTikibrHRVRnUObROismBMBOs3gK42cSgVdLxCBbq@mail.gmail.com>
Message-ID: <AANLkTind--0nS1gC3n86U54BNi7YXhJUeE=pvk08NdyP@mail.gmail.com>

if you are using bridge_pre_exec you already answered it, so of course
it's not going to ring anymore.

you had the right idea with group_confirm, just because you are not
trying to confirm doesn't mean you can't use it for this:

group_confirm_key=exec
group_confirm_file=playback foo.wav

as soon as the file plays the call will bridge.


On Sun, Nov 28, 2010 at 10:22 AM, Alexandre Fiori <fiorix at gmail.com> wrote:
>
> Hi guys,
> When?bridge_pre_execute_bleg_app?is executing, aleg remain in silence.
> More specifically, let's say aleg is listening the ringback, bleg answer the
> call and start executing the bleg_app. After the call is answer by bleg,
> aleg immediately stops ringing and remain in silence while bleg_app is
> executing.
> How can I make aleg keep listening the ringback until bleg_app is finished
> and media (audio) is bridged?
> That is similar to the group_confirm_key=exec behavior, but I have to
> use?bridge_pre_execute_bleg_app because it's not about confirming the call,
> it's about playing some audio files before media is bridged. The problem is,
> aleg should keep the ringback before bleg_app is finished.
> Thanks!
> --
> Ship ahoy! Hast seen the While Whale?
> ? - Melville's Captain Ahab
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From steveu at coppice.org  Wed Dec  1 17:18:33 2010
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 02 Dec 2010 09:18:33 +0800
Subject: [Freeswitch-users] mod_spandsp fax question
In-Reply-To: <A781D8F4E37D4073B447724BF5D5C4BC@e1705>
References: <A781D8F4E37D4073B447724BF5D5C4BC@e1705>
Message-ID: <4CF6F3E9.4000806@coppice.org>

On 11/29/2010 02:00 PM, Madovsky wrote:
> Hi,
> is there any way to send multiple fax pages with one call with 
> originate and &txfax() ?
>
That is the default behaviour, to send all the pages in a TIFF file. 
There are variables to constrain the transmission to a limited range of 
page numbers, if you want to. This is useful if you want to restart a 
long transmission from the point at which it failed.

Spandsp has the ability to send multiple TIFF files within a single 
call, or to turn around and receive within a call. It can even handled 
mixed sizes and resolutions within a file, or between files. However, I 
don't think we have exposed that functionality in mod_spandsp so far.

Steve



From slim at thegreek.com  Wed Dec  1 17:22:45 2010
From: slim at thegreek.com (Jeroen C. van Gelderen)
Date: Wed, 1 Dec 2010 20:22:45 -0500
Subject: [Freeswitch-users] "No Digits to send" on purpose?
In-Reply-To: <FB8961D5-AA00-4C75-87E0-7049C2F332A8@freeswitch.org>
References: <6EFE58A23D6B4E65A16462DCDDE5AD1E@mbnet.local><AANLkTinE1YAOq_k7PG6fTEvpioO3zo+=mMscXcGdSsQp@mail.gmail.com><782B83D12FAE4495A35C0DEED2EB6A84@mbnet.local>
	<FB8961D5-AA00-4C75-87E0-7049C2F332A8@freeswitch.org>
Message-ID: <D5D7DBD69EB94C94B423774D936A8B15@mbnet.local>

Hi Brian,

 

Thanks for your feedback.

 

I've attempted to create a decent bug:

 http://jira.freeswitch.org/browse/OPENZAP-120

 

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen



  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Wednesday, December 01, 2010 17:23
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] "No Digits to send" on purpose?

 

This is NOT specifically a FreeSWITCH issue its mod_freetdm that has the
parsing setup to require digits... small bug on jira would get this
resolved.  http://jira.freeswitch.org

 

/b

 

On Dec 1, 2010, at 4:12 PM, Jeroen C. van Gelderen wrote:





The problem is that Freeswitch in its current incarnation insists that I
send at least one DTMF digit out on the FXO port. I work around this issue
by sending a 'w' "digit" as follows:

 

  <action application="bridge" data="freetdm/span_fxo_csd/1/w"/>

 

This does not actually cause any DTMF tones to be sent out on the FXO port
(so my dialtone is not disturbed) but it does cause a 5-second wait before
the dialtone is heard on the FXS port.

 

What I really need is this:

 

  <action application="bridge" data="freetdm/span_fxo_csd/1/"/>

 

Using this however results in the "No Digits to send" error.

 

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From xyangni at gmail.com  Wed Dec  1 17:39:19 2010
From: xyangni at gmail.com (xuyan yang)
Date: Thu, 2 Dec 2010 01:39:19 +0000
Subject: [Freeswitch-users] How to filter Skypopen chat message in ESL
	mode?
In-Reply-To: <AANLkTi=eE+AQ-F-xndR-6zBajomHY-a5A_8D1ULrZwM_@mail.gmail.com>
References: <AANLkTi=T8G8LjKytzpSTw+bjGUZS7_3TAJDySxKevC0+@mail.gmail.com>
	<AANLkTi=HstdSAsVqe836AXpYwvTg8r3_UdovPCbrLUaO@mail.gmail.com>
	<AANLkTimW=fueqE-iMhPQ03+UEO5dj1-Mghdj7b-ZoM1-@mail.gmail.com>
	<AANLkTik5fMn4Yi_bU_KFKyufdJ1b13AOBD2-H47UHNWQ@mail.gmail.com>
	<AANLkTi=eE+AQ-F-xndR-6zBajomHY-a5A_8D1ULrZwM_@mail.gmail.com>
Message-ID: <AANLkTinjo9k-UgOm-kF4dkYVn7XbXZz4MzS=g_RrmZmS@mail.gmail.com>

I got it. Then the problem should be the unstable behavior of raw events
which has about 25% chances of being missed.

Fortunately, I have found a way to avoid this issue. Ignore all message
events which contains Unique-ID field.

On Wed, Dec 1, 2010 at 11:18 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> On Thu, Dec 2, 2010 at 12:15 AM, Giovanni Maruzzelli
> <gmaruzz at celliax.org> wrote:
> > If you subscribe only to "MESSAGE" kind of events, you'll get only those.
> >
> > The other events are "raw" events, that other users have requested for
> > other purposes.
>
> or at  least that is the expected behavior, please let me know if I
> introduced some regression in integrating that "raw event" thingy.
>
> -giovanni
>
> >
> > -giovanni
> >
> > On Wed, Dec 1, 2010 at 11:35 PM, xuyan yang <xyangni at gmail.com> wrote:
> >> Thanks for your reply. I have read this page before. but the random
> emerging
> >> of the second verbose esl events is causing trouble. And I guess it may
> be a
> >> bug, or it should be predictable.
> >> Before reporting to jira, I just want to check whether I have made any
> >> mistake.
> >>
> >>
> >> On Wed, Dec 1, 2010 at 9:35 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> >> wrote:
> >>>
> >>>
> >>>
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#MESSAGE_.28Chat.29
> >>>
> >>> On Wed, Dec 1, 2010 at 10:23 PM, xuyan yang <xyangni at gmail.com> wrote:
> >>> > hi,
> >>> > I am writing ESL program on FreeSWITCH Version 1.0.head (git-8825b6e
> >>> > 2010-11-28 17-15-39 -0500)
> >>> > I need to handle skype chat message with a inbound ESL connection.
> But
> >>> > there
> >>> > are 1 or 2 esl events received randomly for each message.
> >>> > the first short one is alway generated, but the second one is random.
> So
> >>> > when trying to filter according to UUID, nothing is caught in many
> >>> > cases.
> >>> > If I make filter based on skype ID, duplicated messages are often
> heard.
> >>> > I do not why the behavior of the second event is random.
> >>> > How should I setup the filter to get 1 and only 1 event for each chat
> >>> > message? Thanks.
> >>> >
> >>> > the first is a short one with the following header:
> >>> >  [Event-Name] = [MESSAGE]
> >>> >  [Event-Calling-Function] = [incoming_chatmessage]
> >>> >  [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
> >>> >  [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
> >>> >  [Content-Length] = [1]
> >>> >  [subject] = [SIMPLE MESSAGE]
> >>> >  [FreeSWITCH-IPv4] = [192.168.0.3]
> >>> >  [hint] = [niqizhi]
> >>> >  [from] = [niqizhi]
> >>> >  [Event-Date-Local] = [2010-12-01 21:02:37]
> >>> >  [proto] = [skype]
> >>> >  [FreeSWITCH-IPv6] = [::1]
> >>> >  [id] = [5334]
> >>> >  [Event-Calling-File] = [mod_skypopen.c]
> >>> >  [Event-Date-Timestamp] = [1291237357051788]
> >>> >  [FreeSWITCH-Hostname] = [EYSRV]
> >>> >  [login] = [interface1]
> >>> >  [during-call] = [true]
> >>> >  [Event-Calling-Line-Number] = [2915]
> >>> >  [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
> >>> > the second is much longer:
> >>> >  [Caller-Source] = [mod_skypopen]
> >>> >  [Event-Calling-Function] = [incoming_chatmessage]
> >>> >  [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
> >>> >  [Answer-State] = [answered]
> >>> >  [FreeSWITCH-IPv4] = [192.168.0.3]
> >>> >  [Channel-State] = [CS_EXECUTE]
> >>> >  [Channel-Read-Codec-Bit-Rate] = [256000]
> >>> >  [FreeSWITCH-IPv6] = [::1]
> >>> >  [Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
> >>> >  [Channel-Read-Codec-Rate] = [16000]
> >>> >  [Caller-Destination-Number] = [5655]
> >>> >  [Caller-Channel-Transfer-Time] = [0]
> >>> >  [Channel-Call-State] = [ACTIVE]
> >>> >  [Caller-Channel-Progress-Media-Time] = [0]
> >>> >  [FreeSWITCH-Hostname] = [EYSRV]
> >>> >  [Caller-Channel-Answered-Time] = [1291237326697085]
> >>> >  [login] = [interface1]
> >>> >  [during-call] = [true]
> >>> >  [Channel-Name] = [skypopen/interface1]
> >>> >  [Caller-Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
> >>> >  [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
> >>> >  [Channel-Read-Codec-Name] = [L16]
> >>> >  [Caller-Channel-Name] = [skypopen/interface1]
> >>> >  [Caller-Caller-ID-Number] = [niqizhi]
> >>> >  [Event-Date-Timestamp] = [1291237357051788]
> >>> >  [Channel-State-Number] = [4]
> >>> >  [Event-Calling-Line-Number] = [2888]
> >>> >  [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
> >>> >  [Event-Name] = [MESSAGE]
> >>> >  [Content-Length] = [1]
> >>> >  [subject] = [SIMPLE MESSAGE]
> >>> >  [Caller-Caller-ID-Name] = [niqizhi]
> >>> >  [from] = [niqizhi]
> >>> >  [Caller-Dialplan] = [XML]
> >>> >  [Caller-Channel-Hangup-Time] = [0]
> >>> >  [id] = [5334]
> >>> >  [Caller-Profile-Index] = [1]
> >>> >  [Caller-Direction] = [inbound]
> >>> >  [Caller-Username] = [skypopen]
> >>> >  [Channel-Write-Codec-Name] = [L16]
> >>> >  [Call-Direction] = [inbound]
> >>> >  [Caller-Screen-Bit] = [true]
> >>> >  [hint] = [niqizhi]
> >>> >  [Caller-Privacy-Hide-Number] = [false]
> >>> >  [Event-Date-Local] = [2010-12-01 21:02:37]
> >>> >  [proto] = [skype]
> >>> >  [Caller-Channel-Created-Time] = [1291237326468855]
> >>> >  [Event-Calling-File] = [mod_skypopen.c]
> >>> >  [Caller-Channel-Progress-Time] = [0]
> >>> >  [Caller-Privacy-Hide-Name] = [false]
> >>> >  [Channel-Write-Codec-Rate] = [16000]
> >>> >  [Caller-Context] = [default]
> >>> >  [Channel-Write-Codec-Bit-Rate] = [256000]
> >>> >  [Presence-Call-Direction] = [inbound]
> >>> >  [Caller-Profile-Created-Time] = [1291237326468855]
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>>
> >>>
> >>> --
> >>> Sincerely,
> >>>
> >>> Giovanni Maruzzelli
> >>> Cell : +39-347-2665618
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Dec  1 17:40:02 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Dec 2010 17:40:02 -0800
Subject: [Freeswitch-users] send fax
In-Reply-To: <4FA114B0D2634B158C9260B97E53075D@e1705>
References: <4FA114B0D2634B158C9260B97E53075D@e1705>
Message-ID: <AANLkTik-RUkkGu9UC=Mc+esONs8ELc9oEHtkA3Y0AUHL@mail.gmail.com>

This was a limit in fs_cli, which could only handle 256 chars in the -x
argument. Per Tony I bumped it up to 1024 chars in the latest git. Also, I
just tested libs/esl/perl/single_command.pl and it can have really long
arguments, so if you cannot update to latest git right this minute then you
can always send your command using single_command.pl.

-MC

On Sun, Nov 28, 2010 at 9:17 AM, Madovsky <infos at madovsky.org> wrote:

>  Hi All,
>
> I'm currently playing with send a fax like this :
>
> /usr/local/freeswitch/bin/fs_cli -x "originate
> {originate_caller_id_number=9999999999999,fax_enable_t38=true,fax_enable_t38_request=true,absolute_codec_string=PCMU,nibble_account=9999999999999,nibble_rate=(lcr(1111111111111)*2.3)}sofia/gateway/trunk/1111111111111
> &txfax(/usr/local/freeswitch/storage/FAX/test/test.tiff)"
>
> but it seems that the originate command line has chars length limit, so the
> command line is trunkated and give this result :
> 2010-11-28 12:15:24.929729 [ERR] mod_spandsp_fax.c:1019 Cannot send
> non-existant fax file [/usr/local/fre]
> also do I need to set tot true T38 the 2 vars if my trunkn is T38
> compatible ? I noticed that send a fax has not a high percentage of success.
>
> Any help would be appreciated
>
> Franck
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Wed Dec  1 16:03:41 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 1 Dec 2010 19:03:41 -0500
Subject: [Freeswitch-users] can't get channel variables after hangup
References: <AE2E563CA6B0424FA874C7E68CC77C2A@e1705><AANLkTim3Y+FYRqmy+wyyZKoeOv5keAGoNEe6fPKUTtcK@mail.gmail.com><96A290BECF924A0FA371E425E0C052DA@e1705>
	<AANLkTi=jdYWWbG2i1UDMiQgHXFdYLgPnN6OLOriEgznO@mail.gmail.com>
Message-ID: <8A3974D87D234C4084D673EF42913AD5@e1705>

Ok understood.
but maybe I took the wrong way to catch
the mod_spandsp variables fax_success and fax_result_text.
I would like to add some examples on wiki soon if anyone
knows how to catch them in a dialplan.

Thanks

Franck
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, December 01, 2010 4:11 PM
  Subject: Re: [Freeswitch-users] can't get channel variables after hangup





  On Wed, Dec 1, 2010 at 12:59 PM, Madovsky <infos at madovsky.org> wrote:

    Thanks Mike,

    and with PHP is $session available ?


  No, it's only available in the "built in" languages: Perl, Lua, and Javascript.
  -MC




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


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Wed Dec  1 17:22:19 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 1 Dec 2010 20:22:19 -0500
Subject: [Freeswitch-users] loopback question
References: <918F31117A894F2A917947AF7940507A@e1705>
	<AANLkTin4Me4zC1x=g2d3hQ3Bg3W+EU+5vuaA_aiEKt0J@mail.gmail.com>
Message-ID: <A10F394DF3E34E50A1CC124D249DAF4F@e1705>

I solved it by adding (\+)
<condition field="${sip_to_uri}" expression="^(\+)([2-9]\d{10,15})@$${domain}$">

thanks

  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, December 01, 2010 8:03 PM
  Subject: Re: [Freeswitch-users] loopback question


  I'm guessing that "sip_to_uri" may not be matching. Pastebin the debug log of the call failing and we'll take a look.


  -MC


  On Tue, Nov 30, 2010 at 3:01 PM, Madovsky <infos at madovsky.org> wrote:


    Hi,

    what is the logical to send a fax outside from gateway with originate that sends the call to the default dialplan ?

    I tried ths :

    originalte {filetofax=myfax.tiff}loopback/0000numberToSendFax/default 99999999

    and put this in default dialplan

    <extension name="send_fax">
      <condition field="${sip_to_uri}" expression="^\+([2-9]\d{10,15})@$${domain}$">
       <action application="set" data="hangup_after_bridge=false"/>
       <action application="set" data="continue_on_fail=false"/>
       <action application="ring_ready"/>
       <action application="set" data="absolute_codec_string=PCMU"/>
       <action inline="true" application="set" data="fax_file=${fax_file}"/>
       <action application="transfer" data="sendfax XML features"/>
       </condition>
    </extension>


    but doesn't work.

    Any ?

    Thanks

    Franck



    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Wed Dec  1 17:28:11 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 1 Dec 2010 20:28:11 -0500
Subject: [Freeswitch-users] mod_spandsp fax question
References: <A781D8F4E37D4073B447724BF5D5C4BC@e1705>
	<4CF6F3E9.4000806@coppice.org>
Message-ID: <C3FC8114772245E3B7DEAB2D90519C9B@e1705>


----- Original Message ----- 
From: "Steve Underwood" <steveu at coppice.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, December 01, 2010 8:18 PM
Subject: Re: [Freeswitch-users] mod_spandsp fax question


> On 11/29/2010 02:00 PM, Madovsky wrote:
>> Hi,
>> is there any way to send multiple fax pages with one call with 
>> originate and &txfax() ?
>>
> That is the default behaviour, to send all the pages in a TIFF file. 
> There are variables to constrain the transmission to a limited range of 
> page numbers, if you want to. This is useful if you want to restart a 
> long transmission from the point at which it failed.
> 
> Spandsp has the ability to send multiple TIFF files within a single 
> call, or to turn around and receive within a call. It can even handled 
> mixed sizes and resolutions within a file, or between files. However, I 
> don't think we have exposed that functionality in mod_spandsp so far.
> 
> Steve
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

ok got it, for those who wants to mix several tiff pages to one :
cd /path/to/tiffolder
   tiffcp fax_????.tiff One.tiff

the ???? will search automatically numbers

hope this helps



From infos at madovsky.org  Wed Dec  1 17:50:29 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 1 Dec 2010 20:50:29 -0500
Subject: [Freeswitch-users] send fax
References: <4FA114B0D2634B158C9260B97E53075D@e1705>
	<AANLkTik-RUkkGu9UC=Mc+esONs8ELc9oEHtkA3Y0AUHL@mail.gmail.com>
Message-ID: <CC8F87353BF24D00B7983A548D369779@e1705>

Ok thanks, I will update from git

  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, December 01, 2010 8:40 PM
  Subject: Re: [Freeswitch-users] send fax


  This was a limit in fs_cli, which could only handle 256 chars in the -x argument. Per Tony I bumped it up to 1024 chars in the latest git. Also, I just tested libs/esl/perl/single_command.pl and it can have really long arguments, so if you cannot update to latest git right this minute then you can always send your command using single_command.pl.


  -MC


  On Sun, Nov 28, 2010 at 9:17 AM, Madovsky <infos at madovsky.org> wrote:

    Hi All,

    I'm currently playing with send a fax like this :

    /usr/local/freeswitch/bin/fs_cli -x "originate {originate_caller_id_number=9999999999999,fax_enable_t38=true,fax_enable_t38_request=true,absolute_codec_string=PCMU,nibble_account=9999999999999,nibble_rate=(lcr(1111111111111)*2.3)}sofia/gateway/trunk/1111111111111 &txfax(/usr/local/freeswitch/storage/FAX/test/test.tiff)"

    but it seems that the originate command line has chars length limit, so the command line is trunkated and give this result :
    2010-11-28 12:15:24.929729 [ERR] mod_spandsp_fax.c:1019 Cannot send non-existant fax file [/usr/local/fre]

    also do I need to set tot true T38 the 2 vars if my trunkn is T38 compatible ? I noticed that send a fax has not a high percentage of success.

    Any help would be appreciated

    Franck




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






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From gmaruzz at celliax.org  Wed Dec  1 18:01:17 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 2 Dec 2010 03:01:17 +0100
Subject: [Freeswitch-users] How to filter Skypopen chat message in ESL
	mode?
In-Reply-To: <AANLkTinjo9k-UgOm-kF4dkYVn7XbXZz4MzS=g_RrmZmS@mail.gmail.com>
References: <AANLkTi=T8G8LjKytzpSTw+bjGUZS7_3TAJDySxKevC0+@mail.gmail.com>
	<AANLkTi=HstdSAsVqe836AXpYwvTg8r3_UdovPCbrLUaO@mail.gmail.com>
	<AANLkTimW=fueqE-iMhPQ03+UEO5dj1-Mghdj7b-ZoM1-@mail.gmail.com>
	<AANLkTik5fMn4Yi_bU_KFKyufdJ1b13AOBD2-H47UHNWQ@mail.gmail.com>
	<AANLkTi=eE+AQ-F-xndR-6zBajomHY-a5A_8D1ULrZwM_@mail.gmail.com>
	<AANLkTinjo9k-UgOm-kF4dkYVn7XbXZz4MzS=g_RrmZmS@mail.gmail.com>
Message-ID: <AANLkTik-NdpdJRYxBWZ1KYOOk8xxS59KWiJJmMeFERiP@mail.gmail.com>

I repeat, if you subscribe to "message" events you get only those.
Also, how I can replicate your problem?
I've tested events and I had no problem at all with spurious or
unreliable events in mod-skypopen.
Please, can you indicate a detailed way to reproduce your problem?
-giovanni

On 12/2/10, xuyan yang <xyangni at gmail.com> wrote:
> I got it. Then the problem should be the unstable behavior of raw events
> which has about 25% chances of being missed.
>
> Fortunately, I have found a way to avoid this issue. Ignore all message
> events which contains Unique-ID field.
>
> On Wed, Dec 1, 2010 at 11:18 PM, Giovanni Maruzzelli
> <gmaruzz at celliax.org>wrote:
>
>> On Thu, Dec 2, 2010 at 12:15 AM, Giovanni Maruzzelli
>> <gmaruzz at celliax.org> wrote:
>> > If you subscribe only to "MESSAGE" kind of events, you'll get only
>> > those.
>> >
>> > The other events are "raw" events, that other users have requested for
>> > other purposes.
>>
>> or at  least that is the expected behavior, please let me know if I
>> introduced some regression in integrating that "raw event" thingy.
>>
>> -giovanni
>>
>> >
>> > -giovanni
>> >
>> > On Wed, Dec 1, 2010 at 11:35 PM, xuyan yang <xyangni at gmail.com> wrote:
>> >> Thanks for your reply. I have read this page before. but the random
>> emerging
>> >> of the second verbose esl events is causing trouble. And I guess it may
>> be a
>> >> bug, or it should be predictable.
>> >> Before reporting to jira, I just want to check whether I have made any
>> >> mistake.
>> >>
>> >>
>> >> On Wed, Dec 1, 2010 at 9:35 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> >> wrote:
>> >>>
>> >>>
>> >>>
>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#MESSAGE_.28Chat.29
>> >>>
>> >>> On Wed, Dec 1, 2010 at 10:23 PM, xuyan yang <xyangni at gmail.com> wrote:
>> >>> > hi,
>> >>> > I am writing ESL program on FreeSWITCH Version 1.0.head (git-8825b6e
>> >>> > 2010-11-28 17-15-39 -0500)
>> >>> > I need to handle skype chat message with a inbound ESL connection.
>> But
>> >>> > there
>> >>> > are 1 or 2 esl events received randomly for each message.
>> >>> > the first short one is alway generated, but the second one is
>> >>> > random.
>> So
>> >>> > when trying to filter according to UUID, nothing is caught in many
>> >>> > cases.
>> >>> > If I make filter based on skype ID, duplicated messages are often
>> heard.
>> >>> > I do not why the behavior of the second event is random.
>> >>> > How should I setup the filter to get 1 and only 1 event for each
>> >>> > chat
>> >>> > message? Thanks.
>> >>> >
>> >>> > the first is a short one with the following header:
>> >>> >  [Event-Name] = [MESSAGE]
>> >>> >  [Event-Calling-Function] = [incoming_chatmessage]
>> >>> >  [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
>> >>> >  [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
>> >>> >  [Content-Length] = [1]
>> >>> >  [subject] = [SIMPLE MESSAGE]
>> >>> >  [FreeSWITCH-IPv4] = [192.168.0.3]
>> >>> >  [hint] = [niqizhi]
>> >>> >  [from] = [niqizhi]
>> >>> >  [Event-Date-Local] = [2010-12-01 21:02:37]
>> >>> >  [proto] = [skype]
>> >>> >  [FreeSWITCH-IPv6] = [::1]
>> >>> >  [id] = [5334]
>> >>> >  [Event-Calling-File] = [mod_skypopen.c]
>> >>> >  [Event-Date-Timestamp] = [1291237357051788]
>> >>> >  [FreeSWITCH-Hostname] = [EYSRV]
>> >>> >  [login] = [interface1]
>> >>> >  [during-call] = [true]
>> >>> >  [Event-Calling-Line-Number] = [2915]
>> >>> >  [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
>> >>> > the second is much longer:
>> >>> >  [Caller-Source] = [mod_skypopen]
>> >>> >  [Event-Calling-Function] = [incoming_chatmessage]
>> >>> >  [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
>> >>> >  [Answer-State] = [answered]
>> >>> >  [FreeSWITCH-IPv4] = [192.168.0.3]
>> >>> >  [Channel-State] = [CS_EXECUTE]
>> >>> >  [Channel-Read-Codec-Bit-Rate] = [256000]
>> >>> >  [FreeSWITCH-IPv6] = [::1]
>> >>> >  [Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
>> >>> >  [Channel-Read-Codec-Rate] = [16000]
>> >>> >  [Caller-Destination-Number] = [5655]
>> >>> >  [Caller-Channel-Transfer-Time] = [0]
>> >>> >  [Channel-Call-State] = [ACTIVE]
>> >>> >  [Caller-Channel-Progress-Media-Time] = [0]
>> >>> >  [FreeSWITCH-Hostname] = [EYSRV]
>> >>> >  [Caller-Channel-Answered-Time] = [1291237326697085]
>> >>> >  [login] = [interface1]
>> >>> >  [during-call] = [true]
>> >>> >  [Channel-Name] = [skypopen/interface1]
>> >>> >  [Caller-Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
>> >>> >  [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
>> >>> >  [Channel-Read-Codec-Name] = [L16]
>> >>> >  [Caller-Channel-Name] = [skypopen/interface1]
>> >>> >  [Caller-Caller-ID-Number] = [niqizhi]
>> >>> >  [Event-Date-Timestamp] = [1291237357051788]
>> >>> >  [Channel-State-Number] = [4]
>> >>> >  [Event-Calling-Line-Number] = [2888]
>> >>> >  [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
>> >>> >  [Event-Name] = [MESSAGE]
>> >>> >  [Content-Length] = [1]
>> >>> >  [subject] = [SIMPLE MESSAGE]
>> >>> >  [Caller-Caller-ID-Name] = [niqizhi]
>> >>> >  [from] = [niqizhi]
>> >>> >  [Caller-Dialplan] = [XML]
>> >>> >  [Caller-Channel-Hangup-Time] = [0]
>> >>> >  [id] = [5334]
>> >>> >  [Caller-Profile-Index] = [1]
>> >>> >  [Caller-Direction] = [inbound]
>> >>> >  [Caller-Username] = [skypopen]
>> >>> >  [Channel-Write-Codec-Name] = [L16]
>> >>> >  [Call-Direction] = [inbound]
>> >>> >  [Caller-Screen-Bit] = [true]
>> >>> >  [hint] = [niqizhi]
>> >>> >  [Caller-Privacy-Hide-Number] = [false]
>> >>> >  [Event-Date-Local] = [2010-12-01 21:02:37]
>> >>> >  [proto] = [skype]
>> >>> >  [Caller-Channel-Created-Time] = [1291237326468855]
>> >>> >  [Event-Calling-File] = [mod_skypopen.c]
>> >>> >  [Caller-Channel-Progress-Time] = [0]
>> >>> >  [Caller-Privacy-Hide-Name] = [false]
>> >>> >  [Channel-Write-Codec-Rate] = [16000]
>> >>> >  [Caller-Context] = [default]
>> >>> >  [Channel-Write-Codec-Bit-Rate] = [256000]
>> >>> >  [Presence-Call-Direction] = [inbound]
>> >>> >  [Caller-Profile-Created-Time] = [1291237326468855]
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> Sincerely,
>> >>>
>> >>> Giovanni Maruzzelli
>> >>> Cell : +39-347-2665618
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> >
>> >
>> > --
>> > Sincerely,
>> >
>> > Giovanni Maruzzelli
>> > Cell : +39-347-2665618
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From dujinfang at gmail.com  Wed Dec  1 18:19:24 2010
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 2 Dec 2010 10:19:24 +0800
Subject: [Freeswitch-users] dead call records
In-Reply-To: <AANLkTi=4u-wH4+9CiUdvM50dC0JKx7+7kSdA7ZO=mVHT@mail.gmail.com>
References: <AANLkTimhxMGBzZkcpO1GzvPDYUE6empmE1NxAXrjm4d5@mail.gmail.com>
	<AANLkTi=4u-wH4+9CiUdvM50dC0JKx7+7kSdA7ZO=mVHT@mail.gmail.com>
Message-ID: <AANLkTin6nUj2L-M1PPHip9Y+JXk8VU-4a5ai=0Mwnr_v@mail.gmail.com>

great, confirmed it works, thanks.

On Thu, Dec 2, 2010 at 12:39 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> commit ca28a80658d50a9fab14f3022c40536b385b6b9b
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date: ? Wed Dec 1 10:31:28 2010 -0600
>
> ? ?update caller_profile to have correct uuid when using custom uuid
> from originate string
>
>
> On Wed, Dec 1, 2010 at 7:51 AM, Seven Du <dujinfang at gmail.com> wrote:
>> Hi,
>>
>> I found a wired problem where left dead call records.
>>
>> originate {origination_uuid=1,hangup_after_bridge=false}user/1000 &park
>> originate {origination_uuid=2,hangup_after_bridge=false}user/1001 &park
>> uuid_bridge 1 2
>> uuid_transfer 1 -both park inline
>> hupall
>>
>> show channels # show nothing
>> show calls # the record still there, why the uuid is not 1 and 2 but a
>> long uuid str?
>>
>> but there's no problem if not use origination_uuid.
>>
>> Also, there are no problems if no uuid_transfer.
>>
>> originate {origination_uuid=1,hangup_after_bridge=false}user/1000 &park
>> originate {origination_uuid=2,hangup_after_bridge=false}user/1001 &park
>> uuid_bridge 1 2
>> hupall
>>
>>
>> I'm on last git, anyone can help take a look?
>>
>> http://pastebin.freeswitch.org/14681
>>
>> Thanks.
>>
>> --
>> About: http://about.me/dujinfang
>> Blog: http://www.dujinfang.com
>> Proj:? http://www.freeswitch.org.cn
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From xyangni at gmail.com  Wed Dec  1 19:45:37 2010
From: xyangni at gmail.com (xuyan yang)
Date: Thu, 2 Dec 2010 03:45:37 +0000
Subject: [Freeswitch-users] How to filter Skypopen chat message in ESL
	mode?
In-Reply-To: <AANLkTik-NdpdJRYxBWZ1KYOOk8xxS59KWiJJmMeFERiP@mail.gmail.com>
References: <AANLkTi=T8G8LjKytzpSTw+bjGUZS7_3TAJDySxKevC0+@mail.gmail.com>
	<AANLkTi=HstdSAsVqe836AXpYwvTg8r3_UdovPCbrLUaO@mail.gmail.com>
	<AANLkTimW=fueqE-iMhPQ03+UEO5dj1-Mghdj7b-ZoM1-@mail.gmail.com>
	<AANLkTik5fMn4Yi_bU_KFKyufdJ1b13AOBD2-H47UHNWQ@mail.gmail.com>
	<AANLkTi=eE+AQ-F-xndR-6zBajomHY-a5A_8D1ULrZwM_@mail.gmail.com>
	<AANLkTinjo9k-UgOm-kF4dkYVn7XbXZz4MzS=g_RrmZmS@mail.gmail.com>
	<AANLkTik-NdpdJRYxBWZ1KYOOk8xxS59KWiJJmMeFERiP@mail.gmail.com>
Message-ID: <AANLkTi=n4nbyjKcxGR4pM74XWLzC5VqR2nSJtuiN0WHa@mail.gmail.com>

I tried it again. When the line is parked and idle. the raw events can
always be generated.
but if the line is kept busy such as the following case this events is trend
to have problem

<extension name="skype-in"  break="on-false">
<condition field="destination_number" expression="^5655$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
<!--<action application="park"/> -->
<action application="playback" data="test.wav"/>
<action application="playback" data="test.wav"/>
<action application="playback" data="test.wav"/>
                                <action application="playback"
data="test.wav"/>
<action application="playback" data="test.wav"/>
<action application="playback" data="test.wav"/>
                                <action application="playback"
data="test.wav"/>
<action application="playback" data="test.wav"/>
<action application="playback" data="test.wav"/>
   </condition>
</extension>



On Thu, Dec 2, 2010 at 2:01 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> I repeat, if you subscribe to "message" events you get only those.
> Also, how I can replicate your problem?
> I've tested events and I had no problem at all with spurious or
> unreliable events in mod-skypopen.
> Please, can you indicate a detailed way to reproduce your problem?
> -giovanni
>
> On 12/2/10, xuyan yang <xyangni at gmail.com> wrote:
> > I got it. Then the problem should be the unstable behavior of raw events
> > which has about 25% chances of being missed.
> >
> > Fortunately, I have found a way to avoid this issue. Ignore all message
> > events which contains Unique-ID field.
> >
> > On Wed, Dec 1, 2010 at 11:18 PM, Giovanni Maruzzelli
> > <gmaruzz at celliax.org>wrote:
> >
> >> On Thu, Dec 2, 2010 at 12:15 AM, Giovanni Maruzzelli
> >> <gmaruzz at celliax.org> wrote:
> >> > If you subscribe only to "MESSAGE" kind of events, you'll get only
> >> > those.
> >> >
> >> > The other events are "raw" events, that other users have requested for
> >> > other purposes.
> >>
> >> or at  least that is the expected behavior, please let me know if I
> >> introduced some regression in integrating that "raw event" thingy.
> >>
> >> -giovanni
> >>
> >> >
> >> > -giovanni
> >> >
> >> > On Wed, Dec 1, 2010 at 11:35 PM, xuyan yang <xyangni at gmail.com>
> wrote:
> >> >> Thanks for your reply. I have read this page before. but the random
> >> emerging
> >> >> of the second verbose esl events is causing trouble. And I guess it
> may
> >> be a
> >> >> bug, or it should be predictable.
> >> >> Before reporting to jira, I just want to check whether I have made
> any
> >> >> mistake.
> >> >>
> >> >>
> >> >> On Wed, Dec 1, 2010 at 9:35 PM, Giovanni Maruzzelli <
> gmaruzz at gmail.com>
> >> >> wrote:
> >> >>>
> >> >>>
> >> >>>
> >>
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#MESSAGE_.28Chat.29
> >> >>>
> >> >>> On Wed, Dec 1, 2010 at 10:23 PM, xuyan yang <xyangni at gmail.com>
> wrote:
> >> >>> > hi,
> >> >>> > I am writing ESL program on FreeSWITCH Version 1.0.head
> (git-8825b6e
> >> >>> > 2010-11-28 17-15-39 -0500)
> >> >>> > I need to handle skype chat message with a inbound ESL connection.
> >> But
> >> >>> > there
> >> >>> > are 1 or 2 esl events received randomly for each message.
> >> >>> > the first short one is alway generated, but the second one is
> >> >>> > random.
> >> So
> >> >>> > when trying to filter according to UUID, nothing is caught in many
> >> >>> > cases.
> >> >>> > If I make filter based on skype ID, duplicated messages are often
> >> heard.
> >> >>> > I do not why the behavior of the second event is random.
> >> >>> > How should I setup the filter to get 1 and only 1 event for each
> >> >>> > chat
> >> >>> > message? Thanks.
> >> >>> >
> >> >>> > the first is a short one with the following header:
> >> >>> >  [Event-Name] = [MESSAGE]
> >> >>> >  [Event-Calling-Function] = [incoming_chatmessage]
> >> >>> >  [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
> >> >>> >  [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
> >> >>> >  [Content-Length] = [1]
> >> >>> >  [subject] = [SIMPLE MESSAGE]
> >> >>> >  [FreeSWITCH-IPv4] = [192.168.0.3]
> >> >>> >  [hint] = [niqizhi]
> >> >>> >  [from] = [niqizhi]
> >> >>> >  [Event-Date-Local] = [2010-12-01 21:02:37]
> >> >>> >  [proto] = [skype]
> >> >>> >  [FreeSWITCH-IPv6] = [::1]
> >> >>> >  [id] = [5334]
> >> >>> >  [Event-Calling-File] = [mod_skypopen.c]
> >> >>> >  [Event-Date-Timestamp] = [1291237357051788]
> >> >>> >  [FreeSWITCH-Hostname] = [EYSRV]
> >> >>> >  [login] = [interface1]
> >> >>> >  [during-call] = [true]
> >> >>> >  [Event-Calling-Line-Number] = [2915]
> >> >>> >  [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
> >> >>> > the second is much longer:
> >> >>> >  [Caller-Source] = [mod_skypopen]
> >> >>> >  [Event-Calling-Function] = [incoming_chatmessage]
> >> >>> >  [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
> >> >>> >  [Answer-State] = [answered]
> >> >>> >  [FreeSWITCH-IPv4] = [192.168.0.3]
> >> >>> >  [Channel-State] = [CS_EXECUTE]
> >> >>> >  [Channel-Read-Codec-Bit-Rate] = [256000]
> >> >>> >  [FreeSWITCH-IPv6] = [::1]
> >> >>> >  [Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
> >> >>> >  [Channel-Read-Codec-Rate] = [16000]
> >> >>> >  [Caller-Destination-Number] = [5655]
> >> >>> >  [Caller-Channel-Transfer-Time] = [0]
> >> >>> >  [Channel-Call-State] = [ACTIVE]
> >> >>> >  [Caller-Channel-Progress-Media-Time] = [0]
> >> >>> >  [FreeSWITCH-Hostname] = [EYSRV]
> >> >>> >  [Caller-Channel-Answered-Time] = [1291237326697085]
> >> >>> >  [login] = [interface1]
> >> >>> >  [during-call] = [true]
> >> >>> >  [Channel-Name] = [skypopen/interface1]
> >> >>> >  [Caller-Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
> >> >>> >  [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
> >> >>> >  [Channel-Read-Codec-Name] = [L16]
> >> >>> >  [Caller-Channel-Name] = [skypopen/interface1]
> >> >>> >  [Caller-Caller-ID-Number] = [niqizhi]
> >> >>> >  [Event-Date-Timestamp] = [1291237357051788]
> >> >>> >  [Channel-State-Number] = [4]
> >> >>> >  [Event-Calling-Line-Number] = [2888]
> >> >>> >  [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
> >> >>> >  [Event-Name] = [MESSAGE]
> >> >>> >  [Content-Length] = [1]
> >> >>> >  [subject] = [SIMPLE MESSAGE]
> >> >>> >  [Caller-Caller-ID-Name] = [niqizhi]
> >> >>> >  [from] = [niqizhi]
> >> >>> >  [Caller-Dialplan] = [XML]
> >> >>> >  [Caller-Channel-Hangup-Time] = [0]
> >> >>> >  [id] = [5334]
> >> >>> >  [Caller-Profile-Index] = [1]
> >> >>> >  [Caller-Direction] = [inbound]
> >> >>> >  [Caller-Username] = [skypopen]
> >> >>> >  [Channel-Write-Codec-Name] = [L16]
> >> >>> >  [Call-Direction] = [inbound]
> >> >>> >  [Caller-Screen-Bit] = [true]
> >> >>> >  [hint] = [niqizhi]
> >> >>> >  [Caller-Privacy-Hide-Number] = [false]
> >> >>> >  [Event-Date-Local] = [2010-12-01 21:02:37]
> >> >>> >  [proto] = [skype]
> >> >>> >  [Caller-Channel-Created-Time] = [1291237326468855]
> >> >>> >  [Event-Calling-File] = [mod_skypopen.c]
> >> >>> >  [Caller-Channel-Progress-Time] = [0]
> >> >>> >  [Caller-Privacy-Hide-Name] = [false]
> >> >>> >  [Channel-Write-Codec-Rate] = [16000]
> >> >>> >  [Caller-Context] = [default]
> >> >>> >  [Channel-Write-Codec-Bit-Rate] = [256000]
> >> >>> >  [Presence-Call-Direction] = [inbound]
> >> >>> >  [Caller-Profile-Created-Time] = [1291237326468855]
> >> >>> > _______________________________________________
> >> >>> > FreeSWITCH-users mailing list
> >> >>> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> > UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> > http://www.freeswitch.org
> >> >>> >
> >> >>> >
> >> >>>
> >> >>>
> >> >>>
> >> >>> --
> >> >>> Sincerely,
> >> >>>
> >> >>> Giovanni Maruzzelli
> >> >>> Cell : +39-347-2665618
> >> >>>
> >> >>> _______________________________________________
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >>
> >> >
> >> >
> >> >
> >> > --
> >> > Sincerely,
> >> >
> >> > Giovanni Maruzzelli
> >> > Cell : +39-347-2665618
> >> >
> >>
> >>
> >>
> >> --
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
>
> --
> Sent from my mobile device
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From babak.freeswitch at gmail.com  Wed Dec  1 22:09:30 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Thu, 2 Dec 2010 09:39:30 +0330
Subject: [Freeswitch-users] building mod_managed fails
In-Reply-To: <AANLkTim9EFf4HxHt=hh+d2YzP-e+mBzoCodfGqMdQCDg@mail.gmail.com>
References: <AANLkTi=96X=WyXFZ1ZTU8AE3fKiZUm6Q=tr-ds4kOCWa@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703681DD6C9@mse17be1.mse17.exchange.ms>
	<AANLkTi=xLHN7SAL9t0A+aiDtGTwYoCDwqNFjMapSGQpu@mail.gmail.com>
	<AANLkTimWZYJJEyp9Ayu-Ou3asR0kqbZ2EcjVq8mLAk_c@mail.gmail.com>
	<4CF66CBF.9080308@puzzled.xs4all.nl>
	<AANLkTim9EFf4HxHt=hh+d2YzP-e+mBzoCodfGqMdQCDg@mail.gmail.com>
Message-ID: <AANLkTin-r-sxmaeuD677-__DkFJo5TiXN+ZaX6NcLDxB@mail.gmail.com>

thank u all for ur answers
Yitzchok yes I'm sure I'm using mono 2.6.7. I started from mono 2.8 and
after it failed I used mono 2.6.7 (on a clean centos 5.5 not the one I
installed mono 2.8) and I got the same results. I tried 2.6 and it worked
and David I'm on centos 5.5 too.
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From raimund.sacherer at logitravel.com  Thu Dec  2 00:06:26 2010
From: raimund.sacherer at logitravel.com (Raimund Sacherer)
Date: Thu, 2 Dec 2010 09:06:26 +0100 (CET)
Subject: [Freeswitch-users] mod_callcenter and peridic announcements
In-Reply-To: <18962667.179141291277125404.JavaMail.root@pina>
Message-ID: <4528960.179161291277186310.JavaMail.root@pina>

Hello Michael, 

thank you for pointing this out, it seems mod_fifo looks exactly like what we would need, well, apart from the asterisk call-strategies, but I think we can live with that. 

Best, 
Ray 

----- Mensaje original ----- 
De: "Michael Collins" <msc at freeswitch.org> 
Para: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org> 
Enviados: Jueves, 2 de Diciembre 2010 2:08:54 GMT +01:00 Amsterdam / Berl?n / Berna / Roma / Estocolmo / Viena 
Asunto: Re: [Freeswitch-users] mod_callcenter and peridic announcements 

mod_fifo supports periodic announcements. mod_fifo and mod_callcenter are two different ways to accomplish the same basic task, namely to queue up incoming calls and pass them to agents as efficiently as possible. 


-MC 


On Wed, Dec 1, 2010 at 7:44 AM, Raimund Sacherer < raimund.sacherer at logitravel.com > wrote: 


Hello Freeswitch List. 

We are evaluating switching our asterisk based call center (around 150 agents) over to freeswitch. 

Among other things I noticed that the mod_callcenter module does not have the possibility to use periodic announcements. Could these be implemented somehow ... like, with ESL or something else? 


I want to try out the Spice Technologies (or is it now OpenACD) callcenter solution, but this will be a further project as we first want to switch to freeswitch as soon as possible as we have instability problems which we want to get rid off. 

Thank you for your support, 

Ray 


-- 
Raimund Sacherer 
Dpto. de Sistemas 
Agencia de Viajes Online 

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


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

-- 

Raimund Sacherer Espa?a 
Dpto. de Sistemas 
Agencia de Viajes Online 
www.logitravel.com 

Edificio Logitravel 
Parcela 3B, Parc Bit 
Ctra. Palma - Valldemossa km 7,4 
07121 Palma de Mallorca 
Tel 902 366 847 
Fax 971 213 495 


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From gmaruzz at celliax.org  Thu Dec  2 00:50:07 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 2 Dec 2010 09:50:07 +0100
Subject: [Freeswitch-users] How to filter Skypopen chat message in ESL
	mode?
In-Reply-To: <AANLkTi=n4nbyjKcxGR4pM74XWLzC5VqR2nSJtuiN0WHa@mail.gmail.com>
References: <AANLkTi=T8G8LjKytzpSTw+bjGUZS7_3TAJDySxKevC0+@mail.gmail.com>
	<AANLkTi=HstdSAsVqe836AXpYwvTg8r3_UdovPCbrLUaO@mail.gmail.com>
	<AANLkTimW=fueqE-iMhPQ03+UEO5dj1-Mghdj7b-ZoM1-@mail.gmail.com>
	<AANLkTik5fMn4Yi_bU_KFKyufdJ1b13AOBD2-H47UHNWQ@mail.gmail.com>
	<AANLkTi=eE+AQ-F-xndR-6zBajomHY-a5A_8D1ULrZwM_@mail.gmail.com>
	<AANLkTinjo9k-UgOm-kF4dkYVn7XbXZz4MzS=g_RrmZmS@mail.gmail.com>
	<AANLkTik-NdpdJRYxBWZ1KYOOk8xxS59KWiJJmMeFERiP@mail.gmail.com>
	<AANLkTi=n4nbyjKcxGR4pM74XWLzC5VqR2nSJtuiN0WHa@mail.gmail.com>
Message-ID: <AANLkTin8rCY8t6+buLn_N3RhBR768txZHPgrUOwCdacD@mail.gmail.com>

Xuyan,

please.

Can you give a clear defined procedure to reproduce the problem?

Please, take the time to write it, complete of all relevant info
(dialplan, script, etc etc) so I can reproduce it from a freshly
installed FreeSWITCH.

If you don't give this information I cannot fix the problem. And I
cannot neither read your mind, nor finding the time to try every
possible combination.

Please, post something someone can cut and paste and a clear step by
step procedure to replicate.

-giovanni

On Thu, Dec 2, 2010 at 4:45 AM, xuyan yang <xyangni at gmail.com> wrote:
> I tried it again. When the line is parked and idle. the raw events can
> always be generated.
> but if the line is kept busy such as the following case this events is trend
> to have problem
> <extension name="skype-in" ?break="on-false">
> <condition field="destination_number" expression="^5655$">
> ?? ? ? ?<action application="answer"/>
> ?? ? ? ?<action application="sleep" data="1000"/>
> <!--<action application="park"/> -->
> <action application="playback" data="test.wav"/>
> <action application="playback" data="test.wav"/>
> <action application="playback" data="test.wav"/>
> ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<action application="playback"
> data="test.wav"/>
> <action application="playback" data="test.wav"/>
> <action application="playback" data="test.wav"/>
> ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<action application="playback"
> data="test.wav"/>
> <action application="playback" data="test.wav"/>
> <action application="playback" data="test.wav"/>
> ?? </condition>
> </extension>
>
>
> On Thu, Dec 2, 2010 at 2:01 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> I repeat, if you subscribe to "message" events you get only those.
>> Also, how I can replicate your problem?
>> I've tested events and I had no problem at all with spurious or
>> unreliable events in mod-skypopen.
>> Please, can you indicate a detailed way to reproduce your problem?
>> -giovanni
>>
>> On 12/2/10, xuyan yang <xyangni at gmail.com> wrote:
>> > I got it. Then the problem should be the unstable behavior of raw events
>> > which has about 25% chances of being missed.
>> >
>> > Fortunately, I have found a way to avoid this issue. Ignore all message
>> > events which contains Unique-ID field.
>> >
>> > On Wed, Dec 1, 2010 at 11:18 PM, Giovanni Maruzzelli
>> > <gmaruzz at celliax.org>wrote:
>> >
>> >> On Thu, Dec 2, 2010 at 12:15 AM, Giovanni Maruzzelli
>> >> <gmaruzz at celliax.org> wrote:
>> >> > If you subscribe only to "MESSAGE" kind of events, you'll get only
>> >> > those.
>> >> >
>> >> > The other events are "raw" events, that other users have requested
>> >> > for
>> >> > other purposes.
>> >>
>> >> or at ?least that is the expected behavior, please let me know if I
>> >> introduced some regression in integrating that "raw event" thingy.
>> >>
>> >> -giovanni
>> >>
>> >> >
>> >> > -giovanni
>> >> >
>> >> > On Wed, Dec 1, 2010 at 11:35 PM, xuyan yang <xyangni at gmail.com>
>> >> > wrote:
>> >> >> Thanks for your reply. I have read this page before. but the random
>> >> emerging
>> >> >> of the second verbose esl events is causing trouble. And I guess it
>> >> >> may
>> >> be a
>> >> >> bug, or it should be predictable.
>> >> >> Before reporting to jira, I just want to check whether I have made
>> >> >> any
>> >> >> mistake.
>> >> >>
>> >> >>
>> >> >> On Wed, Dec 1, 2010 at 9:35 PM, Giovanni Maruzzelli
>> >> >> <gmaruzz at gmail.com>
>> >> >> wrote:
>> >> >>>
>> >> >>>
>> >> >>>
>> >>
>> >> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#MESSAGE_.28Chat.29
>> >> >>>
>> >> >>> On Wed, Dec 1, 2010 at 10:23 PM, xuyan yang <xyangni at gmail.com>
>> >> >>> wrote:
>> >> >>> > hi,
>> >> >>> > I am writing ESL program on FreeSWITCH Version 1.0.head
>> >> >>> > (git-8825b6e
>> >> >>> > 2010-11-28 17-15-39 -0500)
>> >> >>> > I need to handle skype chat message with a inbound ESL
>> >> >>> > connection.
>> >> But
>> >> >>> > there
>> >> >>> > are 1 or 2 esl events received randomly for each message.
>> >> >>> > the first short one is alway generated, but the second one is
>> >> >>> > random.
>> >> So
>> >> >>> > when trying to filter according to UUID, nothing is caught in
>> >> >>> > many
>> >> >>> > cases.
>> >> >>> > If I make filter based on skype ID, duplicated messages are often
>> >> heard.
>> >> >>> > I do not why the behavior of the second event is random.
>> >> >>> > How should I setup the filter to get 1 and only 1 event for each
>> >> >>> > chat
>> >> >>> > message? Thanks.
>> >> >>> >
>> >> >>> > the first is a short one with the following header:
>> >> >>> > ?[Event-Name] = [MESSAGE]
>> >> >>> > ?[Event-Calling-Function] = [incoming_chatmessage]
>> >> >>> > ?[Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
>> >> >>> > ?[Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
>> >> >>> > ?[Content-Length] = [1]
>> >> >>> > ?[subject] = [SIMPLE MESSAGE]
>> >> >>> > ?[FreeSWITCH-IPv4] = [192.168.0.3]
>> >> >>> > ?[hint] = [niqizhi]
>> >> >>> > ?[from] = [niqizhi]
>> >> >>> > ?[Event-Date-Local] = [2010-12-01 21:02:37]
>> >> >>> > ?[proto] = [skype]
>> >> >>> > ?[FreeSWITCH-IPv6] = [::1]
>> >> >>> > ?[id] = [5334]
>> >> >>> > ?[Event-Calling-File] = [mod_skypopen.c]
>> >> >>> > ?[Event-Date-Timestamp] = [1291237357051788]
>> >> >>> > ?[FreeSWITCH-Hostname] = [EYSRV]
>> >> >>> > ?[login] = [interface1]
>> >> >>> > ?[during-call] = [true]
>> >> >>> > ?[Event-Calling-Line-Number] = [2915]
>> >> >>> > ?[chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
>> >> >>> > the second is much longer:
>> >> >>> > ?[Caller-Source] = [mod_skypopen]
>> >> >>> > ?[Event-Calling-Function] = [incoming_chatmessage]
>> >> >>> > ?[Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
>> >> >>> > ?[Answer-State] = [answered]
>> >> >>> > ?[FreeSWITCH-IPv4] = [192.168.0.3]
>> >> >>> > ?[Channel-State] = [CS_EXECUTE]
>> >> >>> > ?[Channel-Read-Codec-Bit-Rate] = [256000]
>> >> >>> > ?[FreeSWITCH-IPv6] = [::1]
>> >> >>> > ?[Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
>> >> >>> > ?[Channel-Read-Codec-Rate] = [16000]
>> >> >>> > ?[Caller-Destination-Number] = [5655]
>> >> >>> > ?[Caller-Channel-Transfer-Time] = [0]
>> >> >>> > ?[Channel-Call-State] = [ACTIVE]
>> >> >>> > ?[Caller-Channel-Progress-Media-Time] = [0]
>> >> >>> > ?[FreeSWITCH-Hostname] = [EYSRV]
>> >> >>> > ?[Caller-Channel-Answered-Time] = [1291237326697085]
>> >> >>> > ?[login] = [interface1]
>> >> >>> > ?[during-call] = [true]
>> >> >>> > ?[Channel-Name] = [skypopen/interface1]
>> >> >>> > ?[Caller-Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
>> >> >>> > ?[Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
>> >> >>> > ?[Channel-Read-Codec-Name] = [L16]
>> >> >>> > ?[Caller-Channel-Name] = [skypopen/interface1]
>> >> >>> > ?[Caller-Caller-ID-Number] = [niqizhi]
>> >> >>> > ?[Event-Date-Timestamp] = [1291237357051788]
>> >> >>> > ?[Channel-State-Number] = [4]
>> >> >>> > ?[Event-Calling-Line-Number] = [2888]
>> >> >>> > ?[chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
>> >> >>> > ?[Event-Name] = [MESSAGE]
>> >> >>> > ?[Content-Length] = [1]
>> >> >>> > ?[subject] = [SIMPLE MESSAGE]
>> >> >>> > ?[Caller-Caller-ID-Name] = [niqizhi]
>> >> >>> > ?[from] = [niqizhi]
>> >> >>> > ?[Caller-Dialplan] = [XML]
>> >> >>> > ?[Caller-Channel-Hangup-Time] = [0]
>> >> >>> > ?[id] = [5334]
>> >> >>> > ?[Caller-Profile-Index] = [1]
>> >> >>> > ?[Caller-Direction] = [inbound]
>> >> >>> > ?[Caller-Username] = [skypopen]
>> >> >>> > ?[Channel-Write-Codec-Name] = [L16]
>> >> >>> > ?[Call-Direction] = [inbound]
>> >> >>> > ?[Caller-Screen-Bit] = [true]
>> >> >>> > ?[hint] = [niqizhi]
>> >> >>> > ?[Caller-Privacy-Hide-Number] = [false]
>> >> >>> > ?[Event-Date-Local] = [2010-12-01 21:02:37]
>> >> >>> > ?[proto] = [skype]
>> >> >>> > ?[Caller-Channel-Created-Time] = [1291237326468855]
>> >> >>> > ?[Event-Calling-File] = [mod_skypopen.c]
>> >> >>> > ?[Caller-Channel-Progress-Time] = [0]
>> >> >>> > ?[Caller-Privacy-Hide-Name] = [false]
>> >> >>> > ?[Channel-Write-Codec-Rate] = [16000]
>> >> >>> > ?[Caller-Context] = [default]
>> >> >>> > ?[Channel-Write-Codec-Bit-Rate] = [256000]
>> >> >>> > ?[Presence-Call-Direction] = [inbound]
>> >> >>> > ?[Caller-Profile-Created-Time] = [1291237326468855]
>> >> >>> > _______________________________________________
>> >> >>> > FreeSWITCH-users mailing list
>> >> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>> > UNSUBSCRIBE:
>> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> > http://www.freeswitch.org
>> >> >>> >
>> >> >>> >
>> >> >>>
>> >> >>>
>> >> >>>
>> >> >>> --
>> >> >>> Sincerely,
>> >> >>>
>> >> >>> Giovanni Maruzzelli
>> >> >>> Cell : +39-347-2665618
>> >> >>>
>> >> >>> _______________________________________________
>> >> >>> FreeSWITCH-users mailing list
>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>> UNSUBSCRIBE:
>> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> http://www.freeswitch.org
>> >> >>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> UNSUBSCRIBE:
>> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >>
>> >> >
>> >> >
>> >> >
>> >> > --
>> >> > Sincerely,
>> >> >
>> >> > Giovanni Maruzzelli
>> >> > Cell : +39-347-2665618
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Sincerely,
>> >>
>> >> Giovanni Maruzzelli
>> >> Cell : +39-347-2665618
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>>
>> --
>> Sent from my mobile device
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From xyangni at gmail.com  Thu Dec  2 03:09:03 2010
From: xyangni at gmail.com (xuyan yang)
Date: Thu, 2 Dec 2010 11:09:03 +0000
Subject: [Freeswitch-users] How to filter Skypopen chat message in ESL
	mode?
In-Reply-To: <AANLkTin8rCY8t6+buLn_N3RhBR768txZHPgrUOwCdacD@mail.gmail.com>
References: <AANLkTi=T8G8LjKytzpSTw+bjGUZS7_3TAJDySxKevC0+@mail.gmail.com>
	<AANLkTi=HstdSAsVqe836AXpYwvTg8r3_UdovPCbrLUaO@mail.gmail.com>
	<AANLkTimW=fueqE-iMhPQ03+UEO5dj1-Mghdj7b-ZoM1-@mail.gmail.com>
	<AANLkTik5fMn4Yi_bU_KFKyufdJ1b13AOBD2-H47UHNWQ@mail.gmail.com>
	<AANLkTi=eE+AQ-F-xndR-6zBajomHY-a5A_8D1ULrZwM_@mail.gmail.com>
	<AANLkTinjo9k-UgOm-kF4dkYVn7XbXZz4MzS=g_RrmZmS@mail.gmail.com>
	<AANLkTik-NdpdJRYxBWZ1KYOOk8xxS59KWiJJmMeFERiP@mail.gmail.com>
	<AANLkTi=n4nbyjKcxGR4pM74XWLzC5VqR2nSJtuiN0WHa@mail.gmail.com>
	<AANLkTin8rCY8t6+buLn_N3RhBR768txZHPgrUOwCdacD@mail.gmail.com>
Message-ID: <AANLkTin0EPm2LLUMCNHHD8oh+2XZk59PkO0Jxu85+Uki@mail.gmail.com>

OK. I will try to make it clear.
In general, my test case is to forward skype call to a extention such as
5655. And then in the dialplan for 5655, continuously play some sound.
During playing, you can send skype chat messages from the caller and check
what esl events has been generated from it. According to my test, the raw
type event is not generated for most of chat messages.

In details,

this is my skypeopen.conf.xml:
<configuration name="skypopen.conf" description="Skypopen Configuration">
  <global_settings>
    <param name="debug" value="8"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="default"/>
    <param name="destination" value="1000"/>
    <param name="skype_user" value="user1"/>
    <param name="report_incoming_chatmessages" value="true"/>
    <param name="silent_mode" value="false"/>
    <param name="write_silence_when_idle" value="true"/>
  </global_settings>
  <!-- one entry here per each skypopen interface -->
  <per_interface_settings>
    <interface id="1" name="interface1">
        <param name="X11-display" value=":0.0"/>
        <param name="skype_user" value="xxxxxxxxx"/>
        <param name="destination" value="5655"/>
    </interface>

  </per_interface_settings>
</configuration>



in dialplan for default context, add the following to describe extention
5655:

<extension name="skype"  break="on-false">
<condition field="destination_number" expression="^5655$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
<action application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
application="playback" data="ivr/ivr-sample_submenu.wav"/>
   </condition>
</extension>

ivr-sample_submenu.wav is a sound file in en/us/callie/ivr/8000, but it can
be replaced by any other wav files.

after setup, make a call to your skype user xxxxxxxxx.
in the mean time use fs_cli to connect FS and execute /event plain MESSAGE
to listen esl events.
sent any chat messages from caller, then you are very likely to find only
the short version events.



On Thu, Dec 2, 2010 at 8:50 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> Xuyan,
>
> please.
>
> Can you give a clear defined procedure to reproduce the problem?
>
> Please, take the time to write it, complete of all relevant info
> (dialplan, script, etc etc) so I can reproduce it from a freshly
> installed FreeSWITCH.
>
> If you don't give this information I cannot fix the problem. And I
> cannot neither read your mind, nor finding the time to try every
> possible combination.
>
> Please, post something someone can cut and paste and a clear step by
> step procedure to replicate.
>
> -giovanni
>
> On Thu, Dec 2, 2010 at 4:45 AM, xuyan yang <xyangni at gmail.com> wrote:
> > I tried it again. When the line is parked and idle. the raw events can
> > always be generated.
> > but if the line is kept busy such as the following case this events is
> trend
> > to have problem
> > <extension name="skype-in"  break="on-false">
> > <condition field="destination_number" expression="^5655$">
> >         <action application="answer"/>
> >         <action application="sleep" data="1000"/>
> > <!--<action application="park"/> -->
> > <action application="playback" data="test.wav"/>
> > <action application="playback" data="test.wav"/>
> > <action application="playback" data="test.wav"/>
> >                                 <action application="playback"
> > data="test.wav"/>
> > <action application="playback" data="test.wav"/>
> > <action application="playback" data="test.wav"/>
> >                                 <action application="playback"
> > data="test.wav"/>
> > <action application="playback" data="test.wav"/>
> > <action application="playback" data="test.wav"/>
> >    </condition>
> > </extension>
> >
> >
> > On Thu, Dec 2, 2010 at 2:01 AM, Giovanni Maruzzelli <gmaruzz at celliax.org
> >
> > wrote:
> >>
> >> I repeat, if you subscribe to "message" events you get only those.
> >> Also, how I can replicate your problem?
> >> I've tested events and I had no problem at all with spurious or
> >> unreliable events in mod-skypopen.
> >> Please, can you indicate a detailed way to reproduce your problem?
> >> -giovanni
> >>
> >> On 12/2/10, xuyan yang <xyangni at gmail.com> wrote:
> >> > I got it. Then the problem should be the unstable behavior of raw
> events
> >> > which has about 25% chances of being missed.
> >> >
> >> > Fortunately, I have found a way to avoid this issue. Ignore all
> message
> >> > events which contains Unique-ID field.
> >> >
> >> > On Wed, Dec 1, 2010 at 11:18 PM, Giovanni Maruzzelli
> >> > <gmaruzz at celliax.org>wrote:
> >> >
> >> >> On Thu, Dec 2, 2010 at 12:15 AM, Giovanni Maruzzelli
> >> >> <gmaruzz at celliax.org> wrote:
> >> >> > If you subscribe only to "MESSAGE" kind of events, you'll get only
> >> >> > those.
> >> >> >
> >> >> > The other events are "raw" events, that other users have requested
> >> >> > for
> >> >> > other purposes.
> >> >>
> >> >> or at  least that is the expected behavior, please let me know if I
> >> >> introduced some regression in integrating that "raw event" thingy.
> >> >>
> >> >> -giovanni
> >> >>
> >> >> >
> >> >> > -giovanni
> >> >> >
> >> >> > On Wed, Dec 1, 2010 at 11:35 PM, xuyan yang <xyangni at gmail.com>
> >> >> > wrote:
> >> >> >> Thanks for your reply. I have read this page before. but the
> random
> >> >> emerging
> >> >> >> of the second verbose esl events is causing trouble. And I guess
> it
> >> >> >> may
> >> >> be a
> >> >> >> bug, or it should be predictable.
> >> >> >> Before reporting to jira, I just want to check whether I have made
> >> >> >> any
> >> >> >> mistake.
> >> >> >>
> >> >> >>
> >> >> >> On Wed, Dec 1, 2010 at 9:35 PM, Giovanni Maruzzelli
> >> >> >> <gmaruzz at gmail.com>
> >> >> >> wrote:
> >> >> >>>
> >> >> >>>
> >> >> >>>
> >> >>
> >> >>
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#MESSAGE_.28Chat.29
> >> >> >>>
> >> >> >>> On Wed, Dec 1, 2010 at 10:23 PM, xuyan yang <xyangni at gmail.com>
> >> >> >>> wrote:
> >> >> >>> > hi,
> >> >> >>> > I am writing ESL program on FreeSWITCH Version 1.0.head
> >> >> >>> > (git-8825b6e
> >> >> >>> > 2010-11-28 17-15-39 -0500)
> >> >> >>> > I need to handle skype chat message with a inbound ESL
> >> >> >>> > connection.
> >> >> But
> >> >> >>> > there
> >> >> >>> > are 1 or 2 esl events received randomly for each message.
> >> >> >>> > the first short one is alway generated, but the second one is
> >> >> >>> > random.
> >> >> So
> >> >> >>> > when trying to filter according to UUID, nothing is caught in
> >> >> >>> > many
> >> >> >>> > cases.
> >> >> >>> > If I make filter based on skype ID, duplicated messages are
> often
> >> >> heard.
> >> >> >>> > I do not why the behavior of the second event is random.
> >> >> >>> > How should I setup the filter to get 1 and only 1 event for
> each
> >> >> >>> > chat
> >> >> >>> > message? Thanks.
> >> >> >>> >
> >> >> >>> > the first is a short one with the following header:
> >> >> >>> >  [Event-Name] = [MESSAGE]
> >> >> >>> >  [Event-Calling-Function] = [incoming_chatmessage]
> >> >> >>> >  [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
> >> >> >>> >  [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
> >> >> >>> >  [Content-Length] = [1]
> >> >> >>> >  [subject] = [SIMPLE MESSAGE]
> >> >> >>> >  [FreeSWITCH-IPv4] = [192.168.0.3]
> >> >> >>> >  [hint] = [niqizhi]
> >> >> >>> >  [from] = [niqizhi]
> >> >> >>> >  [Event-Date-Local] = [2010-12-01 21:02:37]
> >> >> >>> >  [proto] = [skype]
> >> >> >>> >  [FreeSWITCH-IPv6] = [::1]
> >> >> >>> >  [id] = [5334]
> >> >> >>> >  [Event-Calling-File] = [mod_skypopen.c]
> >> >> >>> >  [Event-Date-Timestamp] = [1291237357051788]
> >> >> >>> >  [FreeSWITCH-Hostname] = [EYSRV]
> >> >> >>> >  [login] = [interface1]
> >> >> >>> >  [during-call] = [true]
> >> >> >>> >  [Event-Calling-Line-Number] = [2915]
> >> >> >>> >  [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
> >> >> >>> > the second is much longer:
> >> >> >>> >  [Caller-Source] = [mod_skypopen]
> >> >> >>> >  [Event-Calling-Function] = [incoming_chatmessage]
> >> >> >>> >  [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
> >> >> >>> >  [Answer-State] = [answered]
> >> >> >>> >  [FreeSWITCH-IPv4] = [192.168.0.3]
> >> >> >>> >  [Channel-State] = [CS_EXECUTE]
> >> >> >>> >  [Channel-Read-Codec-Bit-Rate] = [256000]
> >> >> >>> >  [FreeSWITCH-IPv6] = [::1]
> >> >> >>> >  [Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
> >> >> >>> >  [Channel-Read-Codec-Rate] = [16000]
> >> >> >>> >  [Caller-Destination-Number] = [5655]
> >> >> >>> >  [Caller-Channel-Transfer-Time] = [0]
> >> >> >>> >  [Channel-Call-State] = [ACTIVE]
> >> >> >>> >  [Caller-Channel-Progress-Media-Time] = [0]
> >> >> >>> >  [FreeSWITCH-Hostname] = [EYSRV]
> >> >> >>> >  [Caller-Channel-Answered-Time] = [1291237326697085]
> >> >> >>> >  [login] = [interface1]
> >> >> >>> >  [during-call] = [true]
> >> >> >>> >  [Channel-Name] = [skypopen/interface1]
> >> >> >>> >  [Caller-Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
> >> >> >>> >  [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
> >> >> >>> >  [Channel-Read-Codec-Name] = [L16]
> >> >> >>> >  [Caller-Channel-Name] = [skypopen/interface1]
> >> >> >>> >  [Caller-Caller-ID-Number] = [niqizhi]
> >> >> >>> >  [Event-Date-Timestamp] = [1291237357051788]
> >> >> >>> >  [Channel-State-Number] = [4]
> >> >> >>> >  [Event-Calling-Line-Number] = [2888]
> >> >> >>> >  [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
> >> >> >>> >  [Event-Name] = [MESSAGE]
> >> >> >>> >  [Content-Length] = [1]
> >> >> >>> >  [subject] = [SIMPLE MESSAGE]
> >> >> >>> >  [Caller-Caller-ID-Name] = [niqizhi]
> >> >> >>> >  [from] = [niqizhi]
> >> >> >>> >  [Caller-Dialplan] = [XML]
> >> >> >>> >  [Caller-Channel-Hangup-Time] = [0]
> >> >> >>> >  [id] = [5334]
> >> >> >>> >  [Caller-Profile-Index] = [1]
> >> >> >>> >  [Caller-Direction] = [inbound]
> >> >> >>> >  [Caller-Username] = [skypopen]
> >> >> >>> >  [Channel-Write-Codec-Name] = [L16]
> >> >> >>> >  [Call-Direction] = [inbound]
> >> >> >>> >  [Caller-Screen-Bit] = [true]
> >> >> >>> >  [hint] = [niqizhi]
> >> >> >>> >  [Caller-Privacy-Hide-Number] = [false]
> >> >> >>> >  [Event-Date-Local] = [2010-12-01 21:02:37]
> >> >> >>> >  [proto] = [skype]
> >> >> >>> >  [Caller-Channel-Created-Time] = [1291237326468855]
> >> >> >>> >  [Event-Calling-File] = [mod_skypopen.c]
> >> >> >>> >  [Caller-Channel-Progress-Time] = [0]
> >> >> >>> >  [Caller-Privacy-Hide-Name] = [false]
> >> >> >>> >  [Channel-Write-Codec-Rate] = [16000]
> >> >> >>> >  [Caller-Context] = [default]
> >> >> >>> >  [Channel-Write-Codec-Bit-Rate] = [256000]
> >> >> >>> >  [Presence-Call-Direction] = [inbound]
> >> >> >>> >  [Caller-Profile-Created-Time] = [1291237326468855]
> >> >> >>> > _______________________________________________
> >> >> >>> > FreeSWITCH-users mailing list
> >> >> >>> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>> > UNSUBSCRIBE:
> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >>> > http://www.freeswitch.org
> >> >> >>> >
> >> >> >>> >
> >> >> >>>
> >> >> >>>
> >> >> >>>
> >> >> >>> --
> >> >> >>> Sincerely,
> >> >> >>>
> >> >> >>> Giovanni Maruzzelli
> >> >> >>> Cell : +39-347-2665618
> >> >> >>>
> >> >> >>> _______________________________________________
> >> >> >>> FreeSWITCH-users mailing list
> >> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>> UNSUBSCRIBE:
> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >>> http://www.freeswitch.org
> >> >> >>
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> UNSUBSCRIBE:
> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >>
> >> >> >>
> >> >> >
> >> >> >
> >> >> >
> >> >> > --
> >> >> > Sincerely,
> >> >> >
> >> >> > Giovanni Maruzzelli
> >> >> > Cell : +39-347-2665618
> >> >> >
> >> >>
> >> >>
> >> >>
> >> >> --
> >> >> Sincerely,
> >> >>
> >> >> Giovanni Maruzzelli
> >> >> Cell : +39-347-2665618
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >>
> >> --
> >> Sent from my mobile device
> >>
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jgallartm at gmail.com  Thu Dec  2 04:55:33 2010
From: jgallartm at gmail.com (Javier Gallart)
Date: Thu, 2 Dec 2010 13:55:33 +0100
Subject: [Freeswitch-users] Sangoma D100 usage
Message-ID: <AANLkTim=3aA4UqjTPATpxVs__0xti22ugXoxCST0Rg43@mail.gmail.com>

Hello

we got a Sangoma D100 transcoding card, and we like to get it working with
our freeswitch. Freeswitch and the Sangoma card are running on the same
server. The module is properly loaded:

 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1102 Reading sangoma
codec configuration
2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1126 Found Sangoma
RTP IP x.x.x.x
2010-12-02 07:46:33.393953 [NOTICE] mod_sangoma_codec.c:1193 Detected 1 and
activated 1 Sangoma codec vocallo modules
2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1207 Loading codecs,
register='all', noregister=''
2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1239 Registering
implementations for codec PCMU
...
2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering
implementations for codec PCMA
...
2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering
implementations for codec G729
...
2010-12-02 07:46:33.402964 [CONSOLE] switch_loadable_module.c:946
Successfully Loaded [mod_sangoma_codec]
2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding
Codec PCMU 0 Sangoma PCMU 8000hz 10ms 64000bps
...
2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding
Codec PCMA 8 Sangoma PCMA 8000hz 20ms 64000bps
...
2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding
Codec G729 18 Sangoma G729 8000hz 40ms 8000bps
...
2010-12-02 07:46:33.403967 [NOTICE] switch_loadable_module.c:274 Adding API
Function 'sangoma_codec'

The codecs look also good:
freeswitch at internal> show codecs
codec,Sangoma G723,mod_sangoma_codec
codec,Sangoma G729,mod_sangoma_codec
codec,Sangoma GSM,mod_sangoma_codec
codec,Sangoma PCMA,mod_sangoma_codec
codec,Sangoma PCMU,mod_sangoma_codec

What we don't know is how to actually use the card. We have forced a call
with g729 in the A-leg and g711 in the b-leg but freeswitch doesn't know how
to bridge them. I've tried unsuccessfully this config in the vars.xml:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=Sangoma G729,Sangoma
PCMU,Sangoma PCMA"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=Sangoma PCMU,Sangoma
PCMA"/>

Does anybody has any experience usgin this card?

Thanks

Javier
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From steveayre at gmail.com  Thu Dec  2 05:48:42 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 2 Dec 2010 13:48:42 +0000
Subject: [Freeswitch-users] Sangoma D100 usage
In-Reply-To: <AANLkTim=3aA4UqjTPATpxVs__0xti22ugXoxCST0Rg43@mail.gmail.com>
References: <AANLkTim=3aA4UqjTPATpxVs__0xti22ugXoxCST0Rg43@mail.gmail.com>
Message-ID: <AANLkTimVmiFnNv960R_Vw9eS4BxYuaMtDoihKXHcd3FP@mail.gmail.com>

Sangoma is not part of the codec IANA name, just part of the interface
name (description). Try it like this:

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA"/>

Don't have mod_g729 or mod_com_g729 loaded at the same time.

-Steve


On 2 December 2010 12:55, Javier Gallart <jgallartm at gmail.com> wrote:
> Hello
> we got a Sangoma D100 transcoding card, and we like to get it working with
> our freeswitch. Freeswitch and the Sangoma card are running on the same
> server. The module is properly loaded:
> ?2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1102 Reading sangoma
> codec configuration
> 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1126 Found Sangoma
> RTP IP x.x.x.x
> 2010-12-02 07:46:33.393953 [NOTICE] mod_sangoma_codec.c:1193 Detected 1 and
> activated 1 Sangoma codec vocallo modules
> 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1207 Loading codecs,
> register='all', noregister=''
> 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1239 Registering
> implementations for codec PCMU
> ...
> 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering
> implementations for codec PCMA
> ...
> 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering
> implementations for codec G729
> ...
> 2010-12-02 07:46:33.402964 [CONSOLE] switch_loadable_module.c:946
> Successfully Loaded [mod_sangoma_codec]
> 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding
> Codec PCMU 0 Sangoma PCMU 8000hz 10ms 64000bps
> ...
> 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding
> Codec PCMA 8 Sangoma PCMA 8000hz 20ms 64000bps
> ...
> 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding
> Codec G729 18 Sangoma G729 8000hz 40ms 8000bps
> ...
> 2010-12-02 07:46:33.403967 [NOTICE] switch_loadable_module.c:274 Adding API
> Function 'sangoma_codec'
> The codecs look also good:
> freeswitch at internal> show codecs
> codec,Sangoma G723,mod_sangoma_codec
> codec,Sangoma G729,mod_sangoma_codec
> codec,Sangoma GSM,mod_sangoma_codec
> codec,Sangoma PCMA,mod_sangoma_codec
> codec,Sangoma PCMU,mod_sangoma_codec
> What we don't know is how to actually use the card. We have forced a call
> with g729 in the A-leg and g711 in the b-leg but freeswitch doesn't know how
> to bridge them. I've tried unsuccessfully this config in the vars.xml:
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=Sangoma G729,Sangoma
> PCMU,Sangoma PCMA"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=Sangoma PCMU,Sangoma
> PCMA"/>
> Does anybody has any experience usgin this card?
> Thanks
> Javier
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From david.ponzone at ipeva.fr  Thu Dec  2 05:48:48 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 2 Dec 2010 14:48:48 +0100
Subject: [Freeswitch-users] Sangoma D100 usage
In-Reply-To: <AANLkTim=3aA4UqjTPATpxVs__0xti22ugXoxCST0Rg43@mail.gmail.com>
References: <AANLkTim=3aA4UqjTPATpxVs__0xti22ugXoxCST0Rg43@mail.gmail.com>
Message-ID: <0CADFCC9-3FDF-4C2C-A729-1E558FC429E0@ipeva.fr>

Javier,

as Sangoma is working quite closely with FreeSWITCH devs (some Sangoma people are actually on this ML), I am pretty sure you can ask Sangoma support to help you out.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 02/12/2010 ? 13:55, Javier Gallart a ?crit :

> Hello
> 
> we got a Sangoma D100 transcoding card, and we like to get it working with our freeswitch. Freeswitch and the Sangoma card are running on the same server. The module is properly loaded:
> 
>  2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1102 Reading sangoma codec configuration
> 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1126 Found Sangoma RTP IP x.x.x.x
> 2010-12-02 07:46:33.393953 [NOTICE] mod_sangoma_codec.c:1193 Detected 1 and activated 1 Sangoma codec vocallo modules
> 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1207 Loading codecs, register='all', noregister=''
> 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1239 Registering implementations for codec PCMU
> ...
> 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering implementations for codec PCMA
> ...
> 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering implementations for codec G729
> ...
> 2010-12-02 07:46:33.402964 [CONSOLE] switch_loadable_module.c:946 Successfully Loaded [mod_sangoma_codec]
> 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding Codec PCMU 0 Sangoma PCMU 8000hz 10ms 64000bps
> ...
> 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding Codec PCMA 8 Sangoma PCMA 8000hz 20ms 64000bps
> ...
> 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 Sangoma G729 8000hz 40ms 8000bps
> ...
> 2010-12-02 07:46:33.403967 [NOTICE] switch_loadable_module.c:274 Adding API Function 'sangoma_codec'
> 
> The codecs look also good:
> freeswitch at internal> show codecs
> codec,Sangoma G723,mod_sangoma_codec
> codec,Sangoma G729,mod_sangoma_codec
> codec,Sangoma GSM,mod_sangoma_codec
> codec,Sangoma PCMA,mod_sangoma_codec
> codec,Sangoma PCMU,mod_sangoma_codec
> 
> What we don't know is how to actually use the card. We have forced a call with g729 in the A-leg and g711 in the b-leg but freeswitch doesn't know how to bridge them. I've tried unsuccessfully this config in the vars.xml:
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=Sangoma G729,Sangoma PCMU,Sangoma PCMA"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=Sangoma PCMU,Sangoma PCMA"/>
> 
> Does anybody has any experience usgin this card?
> 
> Thanks
> 
> Javier
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From gmaruzz at celliax.org  Thu Dec  2 05:55:51 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 2 Dec 2010 14:55:51 +0100
Subject: [Freeswitch-users] How to filter Skypopen chat message in ESL
	mode?
In-Reply-To: <AANLkTin0EPm2LLUMCNHHD8oh+2XZk59PkO0Jxu85+Uki@mail.gmail.com>
References: <AANLkTi=T8G8LjKytzpSTw+bjGUZS7_3TAJDySxKevC0+@mail.gmail.com>
	<AANLkTi=HstdSAsVqe836AXpYwvTg8r3_UdovPCbrLUaO@mail.gmail.com>
	<AANLkTimW=fueqE-iMhPQ03+UEO5dj1-Mghdj7b-ZoM1-@mail.gmail.com>
	<AANLkTik5fMn4Yi_bU_KFKyufdJ1b13AOBD2-H47UHNWQ@mail.gmail.com>
	<AANLkTi=eE+AQ-F-xndR-6zBajomHY-a5A_8D1ULrZwM_@mail.gmail.com>
	<AANLkTinjo9k-UgOm-kF4dkYVn7XbXZz4MzS=g_RrmZmS@mail.gmail.com>
	<AANLkTik-NdpdJRYxBWZ1KYOOk8xxS59KWiJJmMeFERiP@mail.gmail.com>
	<AANLkTi=n4nbyjKcxGR4pM74XWLzC5VqR2nSJtuiN0WHa@mail.gmail.com>
	<AANLkTin8rCY8t6+buLn_N3RhBR768txZHPgrUOwCdacD@mail.gmail.com>
	<AANLkTin0EPm2LLUMCNHHD8oh+2XZk59PkO0Jxu85+Uki@mail.gmail.com>
Message-ID: <AANLkTimUWBDyvN7-NhFR1JWEc3pBnQtnnfcHC8UEjVXd@mail.gmail.com>

1) Is that the complete skypopen.conf.xml? I mean, do you use one only
interface?

2) An incoming skype call goes directly (because is in
skypopen.conf.xml) to 5655 and ear the continuous message.

3) While the incoming skype call is hearing the message, the same
incoming skypeclient sends chatmessages to the skypopen skypeclient

4) You connect via telnet to the ESL port, do the "events plain message"

5) You get sometimes those longer events intermixed with the regular
message events, but not everytime

Is this correct? If this is correct I will test asap and fix the
possible problem.

Btw, I tested yesterday with skypeclient not in a call and I got no
problems at all. I don't think to be in a call would make any
difference.

Please be certain you gave me all the info needed to exactly replicate
your problem.

-giovanni

On Thu, Dec 2, 2010 at 12:09 PM, xuyan yang <xyangni at gmail.com> wrote:
> OK. I will try to make it clear.
> In general, my test case is to forward skype call to a extention such as
> 5655. And then in the dialplan for 5655,?continuously?play some sound.
> During playing, you can send skype chat messages from the caller and check
> what esl events has been generated from it. According to my test, the raw
> type event is not generated for most of chat messages.
> In details,
> this is my skypeopen.conf.xml:
> <configuration name="skypopen.conf" description="Skypopen Configuration">
> ??<global_settings>
> ?? ?<param name="debug" value="8"/>
> ?? ?<param name="dialplan" value="XML"/>
> ?? ?<param name="context" value="default"/>
> ?? ?<param name="destination" value="1000"/>
> ?? ?<param name="skype_user" value="user1"/>
> ?? ?<param name="report_incoming_chatmessages" value="true"/>
> ?? ?<param name="silent_mode" value="false"/>
> ?? ?<param name="write_silence_when_idle" value="true"/>
> ??</global_settings>
> ??<!-- one entry here per each skypopen interface -->
> ??<per_interface_settings>
> ?? ?<interface id="1" name="interface1">
> ?? ? ? ?<param name="X11-display" value=":0.0"/>
> ?? ? ? ?<param name="skype_user" value="xxxxxxxxx"/>
> ?? ? ? ?<param name="destination" value="5655"/>
> ?? ?</interface>
> ??</per_interface_settings>
> </configuration>
>
>
> in dialplan for default context, add the following to describe extention
> 5655:
> <extension name="skype" ?break="on-false">
> <condition field="destination_number" expression="^5655$">
> ?? ? ? ?<action application="answer"/>
> ?? ? ? ?<action application="sleep" data="1000"/>
> <action application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> application="playback" data="ivr/ivr-sample_submenu.wav"/>
> ?? </condition>
> </extension>
> ivr-sample_submenu.wav is a sound file in en/us/callie/ivr/8000, but it can
> be replaced by any other wav files.
> after setup, make a call to your skype user xxxxxxxxx.
> in the mean time use fs_cli to connect FS and execute /event plain MESSAGE
> to listen esl events.
> sent any chat messages from caller, then you are very likely to find only
> the short version events.
>
>
> On Thu, Dec 2, 2010 at 8:50 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> Xuyan,
>>
>> please.
>>
>> Can you give a clear defined procedure to reproduce the problem?
>>
>> Please, take the time to write it, complete of all relevant info
>> (dialplan, script, etc etc) so I can reproduce it from a freshly
>> installed FreeSWITCH.
>>
>> If you don't give this information I cannot fix the problem. And I
>> cannot neither read your mind, nor finding the time to try every
>> possible combination.
>>
>> Please, post something someone can cut and paste and a clear step by
>> step procedure to replicate.
>>
>> -giovanni
>>
>> On Thu, Dec 2, 2010 at 4:45 AM, xuyan yang <xyangni at gmail.com> wrote:
>> > I tried it again. When the line is parked and idle. the raw events can
>> > always be generated.
>> > but if the line is kept busy such as the following case this events is
>> > trend
>> > to have problem
>> > <extension name="skype-in" ?break="on-false">
>> > <condition field="destination_number" expression="^5655$">
>> > ?? ? ? ?<action application="answer"/>
>> > ?? ? ? ?<action application="sleep" data="1000"/>
>> > <!--<action application="park"/> -->
>> > <action application="playback" data="test.wav"/>
>> > <action application="playback" data="test.wav"/>
>> > <action application="playback" data="test.wav"/>
>> > ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<action application="playback"
>> > data="test.wav"/>
>> > <action application="playback" data="test.wav"/>
>> > <action application="playback" data="test.wav"/>
>> > ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<action application="playback"
>> > data="test.wav"/>
>> > <action application="playback" data="test.wav"/>
>> > <action application="playback" data="test.wav"/>
>> > ?? </condition>
>> > </extension>
>> >
>> >
>> > On Thu, Dec 2, 2010 at 2:01 AM, Giovanni Maruzzelli
>> > <gmaruzz at celliax.org>
>> > wrote:
>> >>
>> >> I repeat, if you subscribe to "message" events you get only those.
>> >> Also, how I can replicate your problem?
>> >> I've tested events and I had no problem at all with spurious or
>> >> unreliable events in mod-skypopen.
>> >> Please, can you indicate a detailed way to reproduce your problem?
>> >> -giovanni
>> >>
>> >> On 12/2/10, xuyan yang <xyangni at gmail.com> wrote:
>> >> > I got it. Then the problem should be the unstable behavior of raw
>> >> > events
>> >> > which has about 25% chances of being missed.
>> >> >
>> >> > Fortunately, I have found a way to avoid this issue. Ignore all
>> >> > message
>> >> > events which contains Unique-ID field.
>> >> >
>> >> > On Wed, Dec 1, 2010 at 11:18 PM, Giovanni Maruzzelli
>> >> > <gmaruzz at celliax.org>wrote:
>> >> >
>> >> >> On Thu, Dec 2, 2010 at 12:15 AM, Giovanni Maruzzelli
>> >> >> <gmaruzz at celliax.org> wrote:
>> >> >> > If you subscribe only to "MESSAGE" kind of events, you'll get only
>> >> >> > those.
>> >> >> >
>> >> >> > The other events are "raw" events, that other users have requested
>> >> >> > for
>> >> >> > other purposes.
>> >> >>
>> >> >> or at ?least that is the expected behavior, please let me know if I
>> >> >> introduced some regression in integrating that "raw event" thingy.
>> >> >>
>> >> >> -giovanni
>> >> >>
>> >> >> >
>> >> >> > -giovanni
>> >> >> >
>> >> >> > On Wed, Dec 1, 2010 at 11:35 PM, xuyan yang <xyangni at gmail.com>
>> >> >> > wrote:
>> >> >> >> Thanks for your reply. I have read this page before. but the
>> >> >> >> random
>> >> >> emerging
>> >> >> >> of the second verbose esl events is causing trouble. And I guess
>> >> >> >> it
>> >> >> >> may
>> >> >> be a
>> >> >> >> bug, or it should be predictable.
>> >> >> >> Before reporting to jira, I just want to check whether I have
>> >> >> >> made
>> >> >> >> any
>> >> >> >> mistake.
>> >> >> >>
>> >> >> >>
>> >> >> >> On Wed, Dec 1, 2010 at 9:35 PM, Giovanni Maruzzelli
>> >> >> >> <gmaruzz at gmail.com>
>> >> >> >> wrote:
>> >> >> >>>
>> >> >> >>>
>> >> >> >>>
>> >> >>
>> >> >>
>> >> >> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#MESSAGE_.28Chat.29
>> >> >> >>>
>> >> >> >>> On Wed, Dec 1, 2010 at 10:23 PM, xuyan yang <xyangni at gmail.com>
>> >> >> >>> wrote:
>> >> >> >>> > hi,
>> >> >> >>> > I am writing ESL program on FreeSWITCH Version 1.0.head
>> >> >> >>> > (git-8825b6e
>> >> >> >>> > 2010-11-28 17-15-39 -0500)
>> >> >> >>> > I need to handle skype chat message with a inbound ESL
>> >> >> >>> > connection.
>> >> >> But
>> >> >> >>> > there
>> >> >> >>> > are 1 or 2 esl events received randomly for each message.
>> >> >> >>> > the first short one is alway generated, but the second one is
>> >> >> >>> > random.
>> >> >> So
>> >> >> >>> > when trying to filter according to UUID, nothing is caught in
>> >> >> >>> > many
>> >> >> >>> > cases.
>> >> >> >>> > If I make filter based on skype ID, duplicated messages are
>> >> >> >>> > often
>> >> >> heard.
>> >> >> >>> > I do not why the behavior of the second event is random.
>> >> >> >>> > How should I setup the filter to get 1 and only 1 event for
>> >> >> >>> > each
>> >> >> >>> > chat
>> >> >> >>> > message? Thanks.
>> >> >> >>> >
>> >> >> >>> > the first is a short one with the following header:
>> >> >> >>> > ?[Event-Name] = [MESSAGE]
>> >> >> >>> > ?[Event-Calling-Function] = [incoming_chatmessage]
>> >> >> >>> > ?[Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
>> >> >> >>> > ?[Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
>> >> >> >>> > ?[Content-Length] = [1]
>> >> >> >>> > ?[subject] = [SIMPLE MESSAGE]
>> >> >> >>> > ?[FreeSWITCH-IPv4] = [192.168.0.3]
>> >> >> >>> > ?[hint] = [niqizhi]
>> >> >> >>> > ?[from] = [niqizhi]
>> >> >> >>> > ?[Event-Date-Local] = [2010-12-01 21:02:37]
>> >> >> >>> > ?[proto] = [skype]
>> >> >> >>> > ?[FreeSWITCH-IPv6] = [::1]
>> >> >> >>> > ?[id] = [5334]
>> >> >> >>> > ?[Event-Calling-File] = [mod_skypopen.c]
>> >> >> >>> > ?[Event-Date-Timestamp] = [1291237357051788]
>> >> >> >>> > ?[FreeSWITCH-Hostname] = [EYSRV]
>> >> >> >>> > ?[login] = [interface1]
>> >> >> >>> > ?[during-call] = [true]
>> >> >> >>> > ?[Event-Calling-Line-Number] = [2915]
>> >> >> >>> > ?[chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
>> >> >> >>> > the second is much longer:
>> >> >> >>> > ?[Caller-Source] = [mod_skypopen]
>> >> >> >>> > ?[Event-Calling-Function] = [incoming_chatmessage]
>> >> >> >>> > ?[Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
>> >> >> >>> > ?[Answer-State] = [answered]
>> >> >> >>> > ?[FreeSWITCH-IPv4] = [192.168.0.3]
>> >> >> >>> > ?[Channel-State] = [CS_EXECUTE]
>> >> >> >>> > ?[Channel-Read-Codec-Bit-Rate] = [256000]
>> >> >> >>> > ?[FreeSWITCH-IPv6] = [::1]
>> >> >> >>> > ?[Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
>> >> >> >>> > ?[Channel-Read-Codec-Rate] = [16000]
>> >> >> >>> > ?[Caller-Destination-Number] = [5655]
>> >> >> >>> > ?[Caller-Channel-Transfer-Time] = [0]
>> >> >> >>> > ?[Channel-Call-State] = [ACTIVE]
>> >> >> >>> > ?[Caller-Channel-Progress-Media-Time] = [0]
>> >> >> >>> > ?[FreeSWITCH-Hostname] = [EYSRV]
>> >> >> >>> > ?[Caller-Channel-Answered-Time] = [1291237326697085]
>> >> >> >>> > ?[login] = [interface1]
>> >> >> >>> > ?[during-call] = [true]
>> >> >> >>> > ?[Channel-Name] = [skypopen/interface1]
>> >> >> >>> > ?[Caller-Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
>> >> >> >>> > ?[Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
>> >> >> >>> > ?[Channel-Read-Codec-Name] = [L16]
>> >> >> >>> > ?[Caller-Channel-Name] = [skypopen/interface1]
>> >> >> >>> > ?[Caller-Caller-ID-Number] = [niqizhi]
>> >> >> >>> > ?[Event-Date-Timestamp] = [1291237357051788]
>> >> >> >>> > ?[Channel-State-Number] = [4]
>> >> >> >>> > ?[Event-Calling-Line-Number] = [2888]
>> >> >> >>> > ?[chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
>> >> >> >>> > ?[Event-Name] = [MESSAGE]
>> >> >> >>> > ?[Content-Length] = [1]
>> >> >> >>> > ?[subject] = [SIMPLE MESSAGE]
>> >> >> >>> > ?[Caller-Caller-ID-Name] = [niqizhi]
>> >> >> >>> > ?[from] = [niqizhi]
>> >> >> >>> > ?[Caller-Dialplan] = [XML]
>> >> >> >>> > ?[Caller-Channel-Hangup-Time] = [0]
>> >> >> >>> > ?[id] = [5334]
>> >> >> >>> > ?[Caller-Profile-Index] = [1]
>> >> >> >>> > ?[Caller-Direction] = [inbound]
>> >> >> >>> > ?[Caller-Username] = [skypopen]
>> >> >> >>> > ?[Channel-Write-Codec-Name] = [L16]
>> >> >> >>> > ?[Call-Direction] = [inbound]
>> >> >> >>> > ?[Caller-Screen-Bit] = [true]
>> >> >> >>> > ?[hint] = [niqizhi]
>> >> >> >>> > ?[Caller-Privacy-Hide-Number] = [false]
>> >> >> >>> > ?[Event-Date-Local] = [2010-12-01 21:02:37]
>> >> >> >>> > ?[proto] = [skype]
>> >> >> >>> > ?[Caller-Channel-Created-Time] = [1291237326468855]
>> >> >> >>> > ?[Event-Calling-File] = [mod_skypopen.c]
>> >> >> >>> > ?[Caller-Channel-Progress-Time] = [0]
>> >> >> >>> > ?[Caller-Privacy-Hide-Name] = [false]
>> >> >> >>> > ?[Channel-Write-Codec-Rate] = [16000]
>> >> >> >>> > ?[Caller-Context] = [default]
>> >> >> >>> > ?[Channel-Write-Codec-Bit-Rate] = [256000]
>> >> >> >>> > ?[Presence-Call-Direction] = [inbound]
>> >> >> >>> > ?[Caller-Profile-Created-Time] = [1291237326468855]
>> >> >> >>> > _______________________________________________
>> >> >> >>> > FreeSWITCH-users mailing list
>> >> >> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>> > UNSUBSCRIBE:
>> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >>> > http://www.freeswitch.org
>> >> >> >>> >
>> >> >> >>> >
>> >> >> >>>
>> >> >> >>>
>> >> >> >>>
>> >> >> >>> --
>> >> >> >>> Sincerely,
>> >> >> >>>
>> >> >> >>> Giovanni Maruzzelli
>> >> >> >>> Cell : +39-347-2665618
>> >> >> >>>
>> >> >> >>> _______________________________________________
>> >> >> >>> FreeSWITCH-users mailing list
>> >> >> >>> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>> UNSUBSCRIBE:
>> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >>> http://www.freeswitch.org
>> >> >> >>
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> UNSUBSCRIBE:
>> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> http://www.freeswitch.org
>> >> >> >>
>> >> >> >>
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > --
>> >> >> > Sincerely,
>> >> >> >
>> >> >> > Giovanni Maruzzelli
>> >> >> > Cell : +39-347-2665618
>> >> >> >
>> >> >>
>> >> >>
>> >> >>
>> >> >> --
>> >> >> Sincerely,
>> >> >>
>> >> >> Giovanni Maruzzelli
>> >> >> Cell : +39-347-2665618
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >
>> >>
>> >> --
>> >> Sent from my mobile device
>> >>
>> >> Sincerely,
>> >>
>> >> Giovanni Maruzzelli
>> >> Cell : +39-347-2665618
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From rupa at rupa.com  Thu Dec  2 06:15:33 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 2 Dec 2010 08:15:33 -0600
Subject: [Freeswitch-users] : Skypopen error
In-Reply-To: <AANLkTiks+WPpo6P5qvaeqmEydNQX3RbTBNECkbcsx612@mail.gmail.com>
References: <AANLkTi=W65whZv0f7p6GxfjJWmsG9=OXSs-Tt3jmG1tX@mail.gmail.com>
	<AANLkTi=2f9B2upJx8kg64K4NbRGaiw9fOcULET=4THoM@mail.gmail.com>
	<AANLkTinBeEcX0-6gffYsWaJLNU8j8R+t31hZwMbzX7R+@mail.gmail.com>
	<AANLkTinYqCUDiK=hLKPUJ9Q04VUkGgyHE_BKujasd4Cg@mail.gmail.com>
	<AANLkTiks99vbGmG72fheTRb2QdCkQqQQ01gFZRsjxT2h@mail.gmail.com>
	<AANLkTimcYMYuV1pxo60JhQkfK0+P94sf=b_Er8FJrwP_@mail.gmail.com>
	<AANLkTimcCSMfyaajth4qQBiyDGDfhVpPCJyeVt9-VxdF@mail.gmail.com>
	<AANLkTimgOsmXR6jC8pOhB7URwdn19H91ZtkRC9sfBasU@mail.gmail.com>
	<AANLkTinks_EtXZ-y7a=Xux80XrJey1dAoOr8hnJT64aS@mail.gmail.com>
	<4CF66DB0.9090807@puzzled.xs4all.nl>
	<AANLkTimqo4ZeodHr8WVOz28Uq9MyOuE1a11y33+diE2N@mail.gmail.com>
	<AANLkTinp7W6d3GOh2C=rA748ceQmni+tFA3QNkSu5qj=@mail.gmail.com>
	<AANLkTi=aOgFgELyKVHBAWpLq33a29AepH7Ur+v4YCw1r@mail.gmail.com>
	<AANLkTiks+WPpo6P5qvaeqmEydNQX3RbTBNECkbcsx612@mail.gmail.com>
Message-ID: <AANLkTinruDtk6J_cJHWp_uguS=tBE6H98NEmi1r8MtFD@mail.gmail.com>

Her phone might have a *really* crappy DTMF detector.  They are more likely
to fail miserably with female voices.

On Wed, Dec 1, 2010 at 11:18 AM, Jian Ren <renjian at gmail.com> wrote:

> Hmm, so far the only "customer" is my wife, who keeps complaining to me
> everyday. It explains why never happened to me. I will ask her tonight how
> she sent A or D while calling. Or maybe it's the problem of the phone. She
> is using a dual mode phone. Before, it's connected to PC with USB for skype
> calls on Windows. Now I stopped running any version of skype on windows and
> the USB was disconnected so she is using the phone as a normal one.
> Thanks!
> Jian
>
>
> On Wed, Dec 1, 2010 at 11:56 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>
>> Hi Jian,
>>
>> I hope in English "bizarre" does not sound bad, in Italian would be
>> like "original and out of standard in a funny way" :).
>>
>> >From the log you attached to the Jira, the incoming SIP calls that are
>> then bridged to skypopen are sending both the "A" dtmf and the "D"
>> dtmf (can't remember if any other).
>>
>> You can peruse the log looking for "error 21", and you'll  see is
>> anytime that they sent to you (via SIP) one of the A-B-C-D dtmfs that
>> mod_skypopen duly passes to skype.
>>
>> Problem is: the Skype client does not accept or relay dtmf "A-D", and
>> spit out an error.
>>
>> Out of curiosity you may want to check why your customers are using
>> dtrmf A-D, but is not an absolut need.
>>
>> Anyway, I'll fix this in mod_skypopen code asap, so that if another
>> channel (SIP in your case) try to send A-D to skype, that dtmf will be
>> ignored and a warning line will be emitted to console and to logfile.
>> And no more errors or aborted calls.
>>
>> -giovanni
>>
>>
>>
>> On Wed, Dec 1, 2010 at 5:35 PM, Jian Ren <renjian at gmail.com> wrote:
>> > Also this happened during a call, which got hanged up because of the
>> error.
>> > So the phone sends out A-D during a call?
>> > Jian
>> > On Wed, Dec 1, 2010 at 11:33 AM, Jian Ren <renjian at gmail.com> wrote:
>> >>
>> >> Interesting. I don't have these keys on my phone. Here is the dialplan
>> >> string I am using in the ATA(SPA1001):
>> >> (<:1>[2-9]xx[2-9]xxxxxx|011xx.|1[2-9]xx[2-9]xxxxxx|1xxx|00xx.)
>> >> Shouldn't it only take numbers?
>> >> Could this be caused by any dialplan XML files in freeswitch?
>> >> Thanks!
>> >> Jian
>> >>
>> >> On Wed, Dec 1, 2010 at 10:45 AM, Patrick Lists
>> >> <freeswitch-list at puzzled.xs4all.nl> wrote:
>> >>>
>> >>> On 12/01/2010 04:03 PM, Jian Ren wrote:
>> >>> > Hi Giovanni,
>> >>> > Could you explain more to a bizarre person like me :-)
>> >>> > What's DTMF "D"? Sent from one of my SIP client? Or Someone sent to
>> my
>> >>> > skype through chat? If it's abnormal, I think I should fix the root
>> >>> > cause.
>> >>> > Thank you very much for looking at the issue, with 2M logs :-).
>> >>>
>> >>> DTMF A-D seems to be used only on military phones:
>> >>>
>> http://www.telecomdictionary.com/telecom_dictionary_DTMF_definition.html
>> >>>
>> >>> Regards,
>> >>> Patrick
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From gmaruzz at celliax.org  Thu Dec  2 06:22:18 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 2 Dec 2010 15:22:18 +0100
Subject: [Freeswitch-users] : Skypopen error
In-Reply-To: <AANLkTinruDtk6J_cJHWp_uguS=tBE6H98NEmi1r8MtFD@mail.gmail.com>
References: <AANLkTi=W65whZv0f7p6GxfjJWmsG9=OXSs-Tt3jmG1tX@mail.gmail.com>
	<AANLkTi=2f9B2upJx8kg64K4NbRGaiw9fOcULET=4THoM@mail.gmail.com>
	<AANLkTinBeEcX0-6gffYsWaJLNU8j8R+t31hZwMbzX7R+@mail.gmail.com>
	<AANLkTinYqCUDiK=hLKPUJ9Q04VUkGgyHE_BKujasd4Cg@mail.gmail.com>
	<AANLkTiks99vbGmG72fheTRb2QdCkQqQQ01gFZRsjxT2h@mail.gmail.com>
	<AANLkTimcYMYuV1pxo60JhQkfK0+P94sf=b_Er8FJrwP_@mail.gmail.com>
	<AANLkTimcCSMfyaajth4qQBiyDGDfhVpPCJyeVt9-VxdF@mail.gmail.com>
	<AANLkTimgOsmXR6jC8pOhB7URwdn19H91ZtkRC9sfBasU@mail.gmail.com>
	<AANLkTinks_EtXZ-y7a=Xux80XrJey1dAoOr8hnJT64aS@mail.gmail.com>
	<4CF66DB0.9090807@puzzled.xs4all.nl>
	<AANLkTimqo4ZeodHr8WVOz28Uq9MyOuE1a11y33+diE2N@mail.gmail.com>
	<AANLkTinp7W6d3GOh2C=rA748ceQmni+tFA3QNkSu5qj=@mail.gmail.com>
	<AANLkTi=aOgFgELyKVHBAWpLq33a29AepH7Ur+v4YCw1r@mail.gmail.com>
	<AANLkTiks+WPpo6P5qvaeqmEydNQX3RbTBNECkbcsx612@mail.gmail.com>
	<AANLkTinruDtk6J_cJHWp_uguS=tBE6H98NEmi1r8MtFD@mail.gmail.com>
Message-ID: <AANLkTimAtVns8TxXby+F_dWLGju_u2h+wBiV81gBUXLo@mail.gmail.com>

On Thu, Dec 2, 2010 at 3:15 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> Her phone might have a *really* crappy DTMF detector. ?They are more likely
> to fail miserably with female voices.

Thanks Rupa!

Btw, should be fixed in mod_skypopen code since yesterday. You'll now
get a warning, but no more an error, and the call should be
unaffected.

-giovanni

>
> On Wed, Dec 1, 2010 at 11:18 AM, Jian Ren <renjian at gmail.com> wrote:
>>
>> Hmm, so far the only "customer" is my wife, who keeps complaining to me
>> everyday. It explains why never happened to me. I will ask her tonight how
>> she sent A or D while calling. Or maybe it's the problem of the phone. She
>> is using a dual mode phone. Before, it's connected to PC with USB for skype
>> calls on Windows. Now I stopped running any version of skype on windows and
>> the USB was disconnected so she is using the phone as a normal one.
>> Thanks!
>> Jian
>>
>> On Wed, Dec 1, 2010 at 11:56 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>>
>>> Hi Jian,
>>>
>>> I hope in English "bizarre" does not sound bad, in Italian would be
>>> like "original and out of standard in a funny way" :).
>>>
>>> >From the log you attached to the Jira, the incoming SIP calls that are
>>> then bridged to skypopen are sending both the "A" dtmf and the "D"
>>> dtmf (can't remember if any other).
>>>
>>> You can peruse the log looking for "error 21", and you'll ?see is
>>> anytime that they sent to you (via SIP) one of the A-B-C-D dtmfs that
>>> mod_skypopen duly passes to skype.
>>>
>>> Problem is: the Skype client does not accept or relay dtmf "A-D", and
>>> spit out an error.
>>>
>>> Out of curiosity you may want to check why your customers are using
>>> dtrmf A-D, but is not an absolut need.
>>>
>>> Anyway, I'll fix this in mod_skypopen code asap, so that if another
>>> channel (SIP in your case) try to send A-D to skype, that dtmf will be
>>> ignored and a warning line will be emitted to console and to logfile.
>>> And no more errors or aborted calls.
>>>
>>> -giovanni
>>>
>>>
>>>
>>> On Wed, Dec 1, 2010 at 5:35 PM, Jian Ren <renjian at gmail.com> wrote:
>>> > Also this happened during a call, which got hanged up because of the
>>> > error.
>>> > So the phone sends out A-D during a call?
>>> > Jian
>>> > On Wed, Dec 1, 2010 at 11:33 AM, Jian Ren <renjian at gmail.com> wrote:
>>> >>
>>> >> Interesting. I don't have these keys on my phone. Here is the dialplan
>>> >> string I am using in the ATA(SPA1001):
>>> >> (<:1>[2-9]xx[2-9]xxxxxx|011xx.|1[2-9]xx[2-9]xxxxxx|1xxx|00xx.)
>>> >> Shouldn't it only take numbers?
>>> >> Could this be caused by any dialplan XML files in freeswitch?
>>> >> Thanks!
>>> >> Jian
>>> >>
>>> >> On Wed, Dec 1, 2010 at 10:45 AM, Patrick Lists
>>> >> <freeswitch-list at puzzled.xs4all.nl> wrote:
>>> >>>
>>> >>> On 12/01/2010 04:03 PM, Jian Ren wrote:
>>> >>> > Hi Giovanni,
>>> >>> > Could you explain more to a bizarre person like me :-)
>>> >>> > What's DTMF "D"? Sent from one of my SIP client? Or Someone sent to
>>> >>> > my
>>> >>> > skype through chat? If it's abnormal, I think I should fix the root
>>> >>> > cause.
>>> >>> > Thank you very much for looking at the issue, with 2M logs :-).
>>> >>>
>>> >>> DTMF A-D seems to be used only on military phones:
>>> >>>
>>> >>> http://www.telecomdictionary.com/telecom_dictionary_DTMF_definition.html
>>> >>>
>>> >>> Regards,
>>> >>> Patrick
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>
>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From renjian at gmail.com  Thu Dec  2 06:34:31 2010
From: renjian at gmail.com (Jian Ren)
Date: Thu, 2 Dec 2010 09:34:31 -0500
Subject: [Freeswitch-users] : Skypopen error
In-Reply-To: <AANLkTimAtVns8TxXby+F_dWLGju_u2h+wBiV81gBUXLo@mail.gmail.com>
References: <AANLkTi=W65whZv0f7p6GxfjJWmsG9=OXSs-Tt3jmG1tX@mail.gmail.com>
	<AANLkTi=2f9B2upJx8kg64K4NbRGaiw9fOcULET=4THoM@mail.gmail.com>
	<AANLkTinBeEcX0-6gffYsWaJLNU8j8R+t31hZwMbzX7R+@mail.gmail.com>
	<AANLkTinYqCUDiK=hLKPUJ9Q04VUkGgyHE_BKujasd4Cg@mail.gmail.com>
	<AANLkTiks99vbGmG72fheTRb2QdCkQqQQ01gFZRsjxT2h@mail.gmail.com>
	<AANLkTimcYMYuV1pxo60JhQkfK0+P94sf=b_Er8FJrwP_@mail.gmail.com>
	<AANLkTimcCSMfyaajth4qQBiyDGDfhVpPCJyeVt9-VxdF@mail.gmail.com>
	<AANLkTimgOsmXR6jC8pOhB7URwdn19H91ZtkRC9sfBasU@mail.gmail.com>
	<AANLkTinks_EtXZ-y7a=Xux80XrJey1dAoOr8hnJT64aS@mail.gmail.com>
	<4CF66DB0.9090807@puzzled.xs4all.nl>
	<AANLkTimqo4ZeodHr8WVOz28Uq9MyOuE1a11y33+diE2N@mail.gmail.com>
	<AANLkTinp7W6d3GOh2C=rA748ceQmni+tFA3QNkSu5qj=@mail.gmail.com>
	<AANLkTi=aOgFgELyKVHBAWpLq33a29AepH7Ur+v4YCw1r@mail.gmail.com>
	<AANLkTiks+WPpo6P5qvaeqmEydNQX3RbTBNECkbcsx612@mail.gmail.com>
	<AANLkTinruDtk6J_cJHWp_uguS=tBE6H98NEmi1r8MtFD@mail.gmail.com>
	<AANLkTimAtVns8TxXby+F_dWLGju_u2h+wBiV81gBUXLo@mail.gmail.com>
Message-ID: <AANLkTimmhePNeGQxo_EQ+DxjfVG4-uneRkE9_7eDuENE@mail.gmail.com>

Cool, I did a git clone last night after 7pm, which should contain the fix
already, right?
Maybe that's why I haven't seen the problem so far.
Besides, I disconnected her 2.4G cordless phone and replaced with a 900Hz
one, in case if it's caused by the Wifi interference.

Thanks!
Jian

On Thu, Dec 2, 2010 at 9:22 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> On Thu, Dec 2, 2010 at 3:15 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> > Her phone might have a *really* crappy DTMF detector.  They are more
> likely
> > to fail miserably with female voices.
>
> Thanks Rupa!
>
> Btw, should be fixed in mod_skypopen code since yesterday. You'll now
> get a warning, but no more an error, and the call should be
> unaffected.
>
> -giovanni
>
> >
> > On Wed, Dec 1, 2010 at 11:18 AM, Jian Ren <renjian at gmail.com> wrote:
> >>
> >> Hmm, so far the only "customer" is my wife, who keeps complaining to me
> >> everyday. It explains why never happened to me. I will ask her tonight
> how
> >> she sent A or D while calling. Or maybe it's the problem of the phone.
> She
> >> is using a dual mode phone. Before, it's connected to PC with USB for
> skype
> >> calls on Windows. Now I stopped running any version of skype on windows
> and
> >> the USB was disconnected so she is using the phone as a normal one.
> >> Thanks!
> >> Jian
> >>
> >> On Wed, Dec 1, 2010 at 11:56 AM, Giovanni Maruzzelli <gmaruzz at gmail.com
> >
> >> wrote:
> >>>
> >>> Hi Jian,
> >>>
> >>> I hope in English "bizarre" does not sound bad, in Italian would be
> >>> like "original and out of standard in a funny way" :).
> >>>
> >>> >From the log you attached to the Jira, the incoming SIP calls that are
> >>> then bridged to skypopen are sending both the "A" dtmf and the "D"
> >>> dtmf (can't remember if any other).
> >>>
> >>> You can peruse the log looking for "error 21", and you'll  see is
> >>> anytime that they sent to you (via SIP) one of the A-B-C-D dtmfs that
> >>> mod_skypopen duly passes to skype.
> >>>
> >>> Problem is: the Skype client does not accept or relay dtmf "A-D", and
> >>> spit out an error.
> >>>
> >>> Out of curiosity you may want to check why your customers are using
> >>> dtrmf A-D, but is not an absolut need.
> >>>
> >>> Anyway, I'll fix this in mod_skypopen code asap, so that if another
> >>> channel (SIP in your case) try to send A-D to skype, that dtmf will be
> >>> ignored and a warning line will be emitted to console and to logfile.
> >>> And no more errors or aborted calls.
> >>>
> >>> -giovanni
> >>>
> >>>
> >>>
> >>> On Wed, Dec 1, 2010 at 5:35 PM, Jian Ren <renjian at gmail.com> wrote:
> >>> > Also this happened during a call, which got hanged up because of the
> >>> > error.
> >>> > So the phone sends out A-D during a call?
> >>> > Jian
> >>> > On Wed, Dec 1, 2010 at 11:33 AM, Jian Ren <renjian at gmail.com> wrote:
> >>> >>
> >>> >> Interesting. I don't have these keys on my phone. Here is the
> dialplan
> >>> >> string I am using in the ATA(SPA1001):
> >>> >> (<:1>[2-9]xx[2-9]xxxxxx|011xx.|1[2-9]xx[2-9]xxxxxx|1xxx|00xx.)
> >>> >> Shouldn't it only take numbers?
> >>> >> Could this be caused by any dialplan XML files in freeswitch?
> >>> >> Thanks!
> >>> >> Jian
> >>> >>
> >>> >> On Wed, Dec 1, 2010 at 10:45 AM, Patrick Lists
> >>> >> <freeswitch-list at puzzled.xs4all.nl> wrote:
> >>> >>>
> >>> >>> On 12/01/2010 04:03 PM, Jian Ren wrote:
> >>> >>> > Hi Giovanni,
> >>> >>> > Could you explain more to a bizarre person like me :-)
> >>> >>> > What's DTMF "D"? Sent from one of my SIP client? Or Someone sent
> to
> >>> >>> > my
> >>> >>> > skype through chat? If it's abnormal, I think I should fix the
> root
> >>> >>> > cause.
> >>> >>> > Thank you very much for looking at the issue, with 2M logs :-).
> >>> >>>
> >>> >>> DTMF A-D seems to be used only on military phones:
> >>> >>>
> >>> >>>
> http://www.telecomdictionary.com/telecom_dictionary_DTMF_definition.html
> >>> >>>
> >>> >>> Regards,
> >>> >>> Patrick
> >>> >>>
> >>> >>> _______________________________________________
> >>> >>> FreeSWITCH-users mailing list
> >>> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>>
> >>> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >>> http://www.freeswitch.org
> >>> >>
> >>> >
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>>
> >>>
> >>> --
> >>> Sincerely,
> >>>
> >>> Giovanni Maruzzelli
> >>> Cell : +39-347-2665618
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > -Rupa
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From xyangni at gmail.com  Thu Dec  2 06:40:01 2010
From: xyangni at gmail.com (xuyan yang)
Date: Thu, 2 Dec 2010 14:40:01 +0000
Subject: [Freeswitch-users] How to filter Skypopen chat message in ESL
	mode?
In-Reply-To: <AANLkTimUWBDyvN7-NhFR1JWEc3pBnQtnnfcHC8UEjVXd@mail.gmail.com>
References: <AANLkTi=T8G8LjKytzpSTw+bjGUZS7_3TAJDySxKevC0+@mail.gmail.com>
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Message-ID: <AANLkTi=HGgC4BS6chKMWmU+dwnEFXLvgOq1yFB=pUC+q@mail.gmail.com>

Hi Giovanni,

Please find my reply below in blue. Thanks.

On Thu, Dec 2, 2010 at 1:55 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> 1) Is that the complete skypopen.conf.xml? I mean, do you use one only
> interface?
>
Yes, it is complete. During the test, I use only 1 client with latest
version running on ubuntu GUI. The older version can not be downloaded from
skype now.

>

2) An incoming skype call goes directly (because is in
> skypopen.conf.xml) to 5655 and ear the continuous message.
>
yes, go directly to 5655.


> 3) While the incoming skype call is hearing the message, the same
> incoming skypeclient sends chatmessages to the skypopen skypeclient
>
yes, you are right


>
> 4) You connect via telnet to the ESL port, do the "events plain message"
>
not exactly, but I think fs_cli should function the same as telnet.

>

5) You get sometimes those longer events intermixed with the regular
> message events, but not everytime
> yes, you are right.



> Is this correct? If this is correct I will test asap and fix the
> possible problem.
>
> Btw, I tested yesterday with skypeclient not in a call and I got no
> problems at all. I don't think to be in a call would make any
> difference.

 It is not the call, but the continuous sound playback in the call  which
cause the problem. If the call is parked idle, there is no problem.

>

Please be certain you gave me all the info needed to exactly replicate
> your problem.
>
> -giovanni
>
> On Thu, Dec 2, 2010 at 12:09 PM, xuyan yang <xyangni at gmail.com> wrote:
> > OK. I will try to make it clear.
> > In general, my test case is to forward skype call to a extention such as
> > 5655. And then in the dialplan for 5655, continuously play some sound.
> > During playing, you can send skype chat messages from the caller and
> check
> > what esl events has been generated from it. According to my test, the raw
> > type event is not generated for most of chat messages.
> > In details,
> > this is my skypeopen.conf.xml:
> > <configuration name="skypopen.conf" description="Skypopen Configuration">
> >   <global_settings>
> >     <param name="debug" value="8"/>
> >     <param name="dialplan" value="XML"/>
> >     <param name="context" value="default"/>
> >     <param name="destination" value="1000"/>
> >     <param name="skype_user" value="user1"/>
> >     <param name="report_incoming_chatmessages" value="true"/>
> >     <param name="silent_mode" value="false"/>
> >     <param name="write_silence_when_idle" value="true"/>
> >   </global_settings>
> >   <!-- one entry here per each skypopen interface -->
> >   <per_interface_settings>
> >     <interface id="1" name="interface1">
> >         <param name="X11-display" value=":0.0"/>
> >         <param name="skype_user" value="xxxxxxxxx"/>
> >         <param name="destination" value="5655"/>
> >     </interface>
> >   </per_interface_settings>
> > </configuration>
> >
> >
> > in dialplan for default context, add the following to describe extention
> > 5655:
> > <extension name="skype"  break="on-false">
> > <condition field="destination_number" expression="^5655$">
> >         <action application="answer"/>
> >         <action application="sleep" data="1000"/>
> > <action application="playback" data="ivr/ivr-sample_submenu.wav"/>
> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/>
> >    </condition>
> > </extension>
> > ivr-sample_submenu.wav is a sound file in en/us/callie/ivr/8000, but it
> can
> > be replaced by any other wav files.
> > after setup, make a call to your skype user xxxxxxxxx.
> > in the mean time use fs_cli to connect FS and execute /event plain
> MESSAGE
> > to listen esl events.
> > sent any chat messages from caller, then you are very likely to find only
> > the short version events.
> >
> >
> > On Thu, Dec 2, 2010 at 8:50 AM, Giovanni Maruzzelli <gmaruzz at celliax.org
> >
> > wrote:
> >>
> >> Xuyan,
> >>
> >> please.
> >>
> >> Can you give a clear defined procedure to reproduce the problem?
> >>
> >> Please, take the time to write it, complete of all relevant info
> >> (dialplan, script, etc etc) so I can reproduce it from a freshly
> >> installed FreeSWITCH.
> >>
> >> If you don't give this information I cannot fix the problem. And I
> >> cannot neither read your mind, nor finding the time to try every
> >> possible combination.
> >>
> >> Please, post something someone can cut and paste and a clear step by
> >> step procedure to replicate.
> >>
> >> -giovanni
> >>
> >> On Thu, Dec 2, 2010 at 4:45 AM, xuyan yang <xyangni at gmail.com> wrote:
> >> > I tried it again. When the line is parked and idle. the raw events can
> >> > always be generated.
> >> > but if the line is kept busy such as the following case this events is
> >> > trend
> >> > to have problem
> >> > <extension name="skype-in"  break="on-false">
> >> > <condition field="destination_number" expression="^5655$">
> >> >         <action application="answer"/>
> >> >         <action application="sleep" data="1000"/>
> >> > <!--<action application="park"/> -->
> >> > <action application="playback" data="test.wav"/>
> >> > <action application="playback" data="test.wav"/>
> >> > <action application="playback" data="test.wav"/>
> >> >                                 <action application="playback"
> >> > data="test.wav"/>
> >> > <action application="playback" data="test.wav"/>
> >> > <action application="playback" data="test.wav"/>
> >> >                                 <action application="playback"
> >> > data="test.wav"/>
> >> > <action application="playback" data="test.wav"/>
> >> > <action application="playback" data="test.wav"/>
> >> >    </condition>
> >> > </extension>
> >> >
> >> >
> >> > On Thu, Dec 2, 2010 at 2:01 AM, Giovanni Maruzzelli
> >> > <gmaruzz at celliax.org>
> >> > wrote:
> >> >>
> >> >> I repeat, if you subscribe to "message" events you get only those.
> >> >> Also, how I can replicate your problem?
> >> >> I've tested events and I had no problem at all with spurious or
> >> >> unreliable events in mod-skypopen.
> >> >> Please, can you indicate a detailed way to reproduce your problem?
> >> >> -giovanni
> >> >>
> >> >> On 12/2/10, xuyan yang <xyangni at gmail.com> wrote:
> >> >> > I got it. Then the problem should be the unstable behavior of raw
> >> >> > events
> >> >> > which has about 25% chances of being missed.
> >> >> >
> >> >> > Fortunately, I have found a way to avoid this issue. Ignore all
> >> >> > message
> >> >> > events which contains Unique-ID field.
> >> >> >
> >> >> > On Wed, Dec 1, 2010 at 11:18 PM, Giovanni Maruzzelli
> >> >> > <gmaruzz at celliax.org>wrote:
> >> >> >
> >> >> >> On Thu, Dec 2, 2010 at 12:15 AM, Giovanni Maruzzelli
> >> >> >> <gmaruzz at celliax.org> wrote:
> >> >> >> > If you subscribe only to "MESSAGE" kind of events, you'll get
> only
> >> >> >> > those.
> >> >> >> >
> >> >> >> > The other events are "raw" events, that other users have
> requested
> >> >> >> > for
> >> >> >> > other purposes.
> >> >> >>
> >> >> >> or at  least that is the expected behavior, please let me know if
> I
> >> >> >> introduced some regression in integrating that "raw event" thingy.
> >> >> >>
> >> >> >> -giovanni
> >> >> >>
> >> >> >> >
> >> >> >> > -giovanni
> >> >> >> >
> >> >> >> > On Wed, Dec 1, 2010 at 11:35 PM, xuyan yang <xyangni at gmail.com>
> >> >> >> > wrote:
> >> >> >> >> Thanks for your reply. I have read this page before. but the
> >> >> >> >> random
> >> >> >> emerging
> >> >> >> >> of the second verbose esl events is causing trouble. And I
> guess
> >> >> >> >> it
> >> >> >> >> may
> >> >> >> be a
> >> >> >> >> bug, or it should be predictable.
> >> >> >> >> Before reporting to jira, I just want to check whether I have
> >> >> >> >> made
> >> >> >> >> any
> >> >> >> >> mistake.
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> On Wed, Dec 1, 2010 at 9:35 PM, Giovanni Maruzzelli
> >> >> >> >> <gmaruzz at gmail.com>
> >> >> >> >> wrote:
> >> >> >> >>>
> >> >> >> >>>
> >> >> >> >>>
> >> >> >>
> >> >> >>
> >> >> >>
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#MESSAGE_.28Chat.29
> >> >> >> >>>
> >> >> >> >>> On Wed, Dec 1, 2010 at 10:23 PM, xuyan yang <
> xyangni at gmail.com>
> >> >> >> >>> wrote:
> >> >> >> >>> > hi,
> >> >> >> >>> > I am writing ESL program on FreeSWITCH Version 1.0.head
> >> >> >> >>> > (git-8825b6e
> >> >> >> >>> > 2010-11-28 17-15-39 -0500)
> >> >> >> >>> > I need to handle skype chat message with a inbound ESL
> >> >> >> >>> > connection.
> >> >> >> But
> >> >> >> >>> > there
> >> >> >> >>> > are 1 or 2 esl events received randomly for each message.
> >> >> >> >>> > the first short one is alway generated, but the second one
> is
> >> >> >> >>> > random.
> >> >> >> So
> >> >> >> >>> > when trying to filter according to UUID, nothing is caught
> in
> >> >> >> >>> > many
> >> >> >> >>> > cases.
> >> >> >> >>> > If I make filter based on skype ID, duplicated messages are
> >> >> >> >>> > often
> >> >> >> heard.
> >> >> >> >>> > I do not why the behavior of the second event is random.
> >> >> >> >>> > How should I setup the filter to get 1 and only 1 event for
> >> >> >> >>> > each
> >> >> >> >>> > chat
> >> >> >> >>> > message? Thanks.
> >> >> >> >>> >
> >> >> >> >>> > the first is a short one with the following header:
> >> >> >> >>> >  [Event-Name] = [MESSAGE]
> >> >> >> >>> >  [Event-Calling-Function] = [incoming_chatmessage]
> >> >> >> >>> >  [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
> >> >> >> >>> >  [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
> >> >> >> >>> >  [Content-Length] = [1]
> >> >> >> >>> >  [subject] = [SIMPLE MESSAGE]
> >> >> >> >>> >  [FreeSWITCH-IPv4] = [192.168.0.3]
> >> >> >> >>> >  [hint] = [niqizhi]
> >> >> >> >>> >  [from] = [niqizhi]
> >> >> >> >>> >  [Event-Date-Local] = [2010-12-01 21:02:37]
> >> >> >> >>> >  [proto] = [skype]
> >> >> >> >>> >  [FreeSWITCH-IPv6] = [::1]
> >> >> >> >>> >  [id] = [5334]
> >> >> >> >>> >  [Event-Calling-File] = [mod_skypopen.c]
> >> >> >> >>> >  [Event-Date-Timestamp] = [1291237357051788]
> >> >> >> >>> >  [FreeSWITCH-Hostname] = [EYSRV]
> >> >> >> >>> >  [login] = [interface1]
> >> >> >> >>> >  [during-call] = [true]
> >> >> >> >>> >  [Event-Calling-Line-Number] = [2915]
> >> >> >> >>> >  [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
> >> >> >> >>> > the second is much longer:
> >> >> >> >>> >  [Caller-Source] = [mod_skypopen]
> >> >> >> >>> >  [Event-Calling-Function] = [incoming_chatmessage]
> >> >> >> >>> >  [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
> >> >> >> >>> >  [Answer-State] = [answered]
> >> >> >> >>> >  [FreeSWITCH-IPv4] = [192.168.0.3]
> >> >> >> >>> >  [Channel-State] = [CS_EXECUTE]
> >> >> >> >>> >  [Channel-Read-Codec-Bit-Rate] = [256000]
> >> >> >> >>> >  [FreeSWITCH-IPv6] = [::1]
> >> >> >> >>> >  [Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
> >> >> >> >>> >  [Channel-Read-Codec-Rate] = [16000]
> >> >> >> >>> >  [Caller-Destination-Number] = [5655]
> >> >> >> >>> >  [Caller-Channel-Transfer-Time] = [0]
> >> >> >> >>> >  [Channel-Call-State] = [ACTIVE]
> >> >> >> >>> >  [Caller-Channel-Progress-Media-Time] = [0]
> >> >> >> >>> >  [FreeSWITCH-Hostname] = [EYSRV]
> >> >> >> >>> >  [Caller-Channel-Answered-Time] = [1291237326697085]
> >> >> >> >>> >  [login] = [interface1]
> >> >> >> >>> >  [during-call] = [true]
> >> >> >> >>> >  [Channel-Name] = [skypopen/interface1]
> >> >> >> >>> >  [Caller-Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
> >> >> >> >>> >  [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
> >> >> >> >>> >  [Channel-Read-Codec-Name] = [L16]
> >> >> >> >>> >  [Caller-Channel-Name] = [skypopen/interface1]
> >> >> >> >>> >  [Caller-Caller-ID-Number] = [niqizhi]
> >> >> >> >>> >  [Event-Date-Timestamp] = [1291237357051788]
> >> >> >> >>> >  [Channel-State-Number] = [4]
> >> >> >> >>> >  [Event-Calling-Line-Number] = [2888]
> >> >> >> >>> >  [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
> >> >> >> >>> >  [Event-Name] = [MESSAGE]
> >> >> >> >>> >  [Content-Length] = [1]
> >> >> >> >>> >  [subject] = [SIMPLE MESSAGE]
> >> >> >> >>> >  [Caller-Caller-ID-Name] = [niqizhi]
> >> >> >> >>> >  [from] = [niqizhi]
> >> >> >> >>> >  [Caller-Dialplan] = [XML]
> >> >> >> >>> >  [Caller-Channel-Hangup-Time] = [0]
> >> >> >> >>> >  [id] = [5334]
> >> >> >> >>> >  [Caller-Profile-Index] = [1]
> >> >> >> >>> >  [Caller-Direction] = [inbound]
> >> >> >> >>> >  [Caller-Username] = [skypopen]
> >> >> >> >>> >  [Channel-Write-Codec-Name] = [L16]
> >> >> >> >>> >  [Call-Direction] = [inbound]
> >> >> >> >>> >  [Caller-Screen-Bit] = [true]
> >> >> >> >>> >  [hint] = [niqizhi]
> >> >> >> >>> >  [Caller-Privacy-Hide-Number] = [false]
> >> >> >> >>> >  [Event-Date-Local] = [2010-12-01 21:02:37]
> >> >> >> >>> >  [proto] = [skype]
> >> >> >> >>> >  [Caller-Channel-Created-Time] = [1291237326468855]
> >> >> >> >>> >  [Event-Calling-File] = [mod_skypopen.c]
> >> >> >> >>> >  [Caller-Channel-Progress-Time] = [0]
> >> >> >> >>> >  [Caller-Privacy-Hide-Name] = [false]
> >> >> >> >>> >  [Channel-Write-Codec-Rate] = [16000]
> >> >> >> >>> >  [Caller-Context] = [default]
> >> >> >> >>> >  [Channel-Write-Codec-Bit-Rate] = [256000]
> >> >> >> >>> >  [Presence-Call-Direction] = [inbound]
> >> >> >> >>> >  [Caller-Profile-Created-Time] = [1291237326468855]
> >> >> >> >>> > _______________________________________________
> >> >> >> >>> > FreeSWITCH-users mailing list
> >> >> >> >>> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >>> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>> > UNSUBSCRIBE:
> >> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >>> > http://www.freeswitch.org
> >> >> >> >>> >
> >> >> >> >>> >
> >> >> >> >>>
> >> >> >> >>>
> >> >> >> >>>
> >> >> >> >>> --
> >> >> >> >>> Sincerely,
> >> >> >> >>>
> >> >> >> >>> Giovanni Maruzzelli
> >> >> >> >>> Cell : +39-347-2665618
> >> >> >> >>>
> >> >> >> >>> _______________________________________________
> >> >> >> >>> FreeSWITCH-users mailing list
> >> >> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>> UNSUBSCRIBE:
> >> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >>> http://www.freeswitch.org
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> _______________________________________________
> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >> UNSUBSCRIBE:
> >> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> http://www.freeswitch.org
> >> >> >> >>
> >> >> >> >>
> >> >> >> >
> >> >> >> >
> >> >> >> >
> >> >> >> > --
> >> >> >> > Sincerely,
> >> >> >> >
> >> >> >> > Giovanni Maruzzelli
> >> >> >> > Cell : +39-347-2665618
> >> >> >> >
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >> --
> >> >> >> Sincerely,
> >> >> >>
> >> >> >> Giovanni Maruzzelli
> >> >> >> Cell : +39-347-2665618
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >>
> >> >> >
> >> >>
> >> >> --
> >> >> Sent from my mobile device
> >> >>
> >> >> Sincerely,
> >> >>
> >> >> Giovanni Maruzzelli
> >> >> Cell : +39-347-2665618
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bernhard.suttner at winet.ch  Thu Dec  2 06:48:50 2010
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Thu, 2 Dec 2010 15:48:50 +0100
Subject: [Freeswitch-users] Export a channel variable within Dial String
Message-ID: <481d3184-6518-4957-acca-b1e73c7fec7e@winet.ch>

Hi,

I call multiple destination with bridge like that:

[var1=value]sofia/default/user1,[var1=another_value]loopback/user2

Is it somehow possible, that var1=another_value will be exported but var1=value not? What I want is, that I could specify a variable to be exported _within_ the dialstring?

Thanks for every hint.

Best regards,
Bernhard




From bcxml at hotmail.com  Thu Dec  2 07:03:13 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Thu, 2 Dec 2010 10:03:13 -0500
Subject: [Freeswitch-users] Custom Channel Variables via C#
In-Reply-To: <0E462B4C-C436-462E-9DA7-9EC1241FB380@avgs.ca>
References: <SNT135-w336CAB94DFC78D328AE6E0B7260@phx.gbl>,
	<0E462B4C-C436-462E-9DA7-9EC1241FB380@avgs.ca>
Message-ID: <SNT135-w4BF48F8FEB9054D804759B7270@phx.gbl>


Thanks, the link was very helpful
 
I have taken a look at the link that you provided
 
I assume now that I can use uuid_setvar to accomplish what I want
 
Assuming that the uuid is 53d37581-1f90-44bf-860a-addbc8430e3a, it seems that my code would need to change to
 
 
private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
{
  try
  {
    TcpClient newClient = new TcpClient();
 
    newClient.Connect("127.0.0.1", 8021);
 
    NetworkStream tcpStream = newClient.GetStream();
 
    byte[] sendBytes = Encoding.ASCII.GetBytes("api uuid_setvar 53d37581-1f90-44bf-860a-addbc8430e3a myChannelVariable 12345");
 
    tcpStream.Write(sendBytes, 0, sendBytes.Length);
 
    tcpStream.Close();
 
    newClient.Close();
  }
  catch(Exception ex)
  {
    throw ex;
  }
}

My question now becomes, how do I go about finding out the uuid for the session ? Is that obtainable via mod_event_socket or is there another way ?
 
Thanks
 
Brian 


From: mrene_lists at avgs.ca
Date: Wed, 1 Dec 2010 16:53:53 -0500
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Custom Channel Variables via C#



Hi,


You can't send XML dialplan actions on the socket and expect FreeSWITCH to understand it, you must follow the event socket protocol.


See http://wiki.freeswitch.org/wiki/Event_socket for more information.




Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca






On 2010-12-01, at 4:46 PM, Brian Campbell wrote:

 
I can set a custom channel variable in my incomming dial plan like this...
 
<action application='set' data='MyChannelVariable=12345' />
 
And it shows up fine in the CDR
 
I am now attempting to use mod_event_socket to set a custom channel variable and have it appear in the CDR
 
Here is the C# code so far, the code is part of a Microsoft Speech Server application that is answering the call
 
private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
{
  try
  {
    TcpClient newClient = new TcpClient();
 
    newClient.Connect("127.0.0.1", 8021);
 
    NetworkStream tcpStream = newClient.GetStream();
 
    byte[] sendBytes = Encoding.ASCII.GetBytes("<action application='set' data='MyChannelVariable=12345' />");
 
    tcpStream.Write(sendBytes, 0, sendBytes.Length);
 
    tcpStream.Close();
 
    newClient.Close();
  }
  catch(Exception ex)
  {
    throw ex;
  }
}
 
After the call is answered, the C# code seems to run fine, but I dont see the custom channel variable in the resulting CDR
 
I figure I am not setting it correctly
 
Can anyone advise on what I am doing wrong ?
 
Thanks
 
 
Brian
 
 
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From david.ponzone at ipeva.fr  Thu Dec  2 07:13:44 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 2 Dec 2010 16:13:44 +0100
Subject: [Freeswitch-users] Export a channel variable within Dial String
In-Reply-To: <481d3184-6518-4957-acca-b1e73c7fec7e@winet.ch>
References: <481d3184-6518-4957-acca-b1e73c7fec7e@winet.ch>
Message-ID: <B724748E-DF66-4CD4-B987-BED15F266C78@ipeva.fr>

Bernhard,

AFAIU, if you specify the variable in the dialstring, it's for leg B, so it's exported.
If you wan't it, don't set it :)

And for leg A, well, you only have one leg A, so you can't set it twice.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 02/12/2010 ? 15:48, Bernhard Suttner a ?crit :

> Hi,
> 
> I call multiple destination with bridge like that:
> 
> [var1=value]sofia/default/user1,[var1=another_value]loopback/user2
> 
> Is it somehow possible, that var1=another_value will be exported but var1=value not? What I want is, that I could specify a variable to be exported _within_ the dialstring?
> 
> Thanks for every hint.
> 
> Best regards,
> Bernhard
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From marcdecorny at gmail.com  Thu Dec  2 07:17:27 2010
From: marcdecorny at gmail.com (Marc de Corny)
Date: Thu, 2 Dec 2010 15:17:27 +0000
Subject: [Freeswitch-users] Lua not playing any WAV files
Message-ID: <AANLkTikaJSj0DXM_ZWX_jgYN4pP-F52s3NnWWxyKZuAk@mail.gmail.com>

Hi all,

I have run into an issue on something so basic that I must be as simple as
enabling a feature somewhere.

I have been trying to get lua to play a message from a WAV file. I have
tried session:execute("playback", main_msg) and
session:streamFile(ivr_invalid_msg) but neither of them play any music to
the caller. I tried both to answer and preAnswer the call first but it made
no difference. However if I put the same file into the XML dialplan and play
it with the commands below I hear the music fine.
<action application="set" data="playback_terminators=#"/>
<action application="playback" data="${401_sound_file_path}"/>

The issue only seems to be from lua when playing any type of wav file and
those files are definitelly there as can be read by the XML

The error message is below for the execute(playback) command, but nothing
can be seen for the
2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application
playback Requires media! pre_answering channel
sofia/external/2031701665 at 194.0.147.16:5060 EXECUTE
sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswitch/sounds/svc_sound_files/default_autoattendant.wav)

But there is no mention of the streamFile command. I have had similar issue
with the PlayAndGetDigits command.
Is there something that I need to enable in lua so that is can playback
messages to the caller.

Many thanks to anyone who can help.
Marc


below is the XML dialplan and lua script as well as the log at the very end.


XML DIALPLAN:
<include>
<extension name="IVR_FROM_MYSQL">
<condition field="caller_id_number" expression="Anonymous" break="never">
<action application="set" data="effective_caller_id_number=0000000000"/>
</condition>
<condition field="destination_number" expression="^(404)">
<action inline="true" application="lua" data="ivr_mysql.lua
${destination_number:3}"/>
<action application="set" data="effective_caller_id_name=${404_tag}"/>
<action application="bridge" data="${404_dial}"/>
</condition>
</extension>
</include>

The LUA script ivr_mysql.lua is callsed and this is it.
-- IVR : PLAY IVR WAV FILES
-- Global Variables:
local dialstr_prefix = "sofia/gateway/CS2k/"
local dialstr_main = ""
local breakoutcode = "184"
local sound_file_folder = "/usr/local/freeswitch/sounds/svc_sound_files/"
local ddi = argv[1]
-- answer the call
session:preAnswer();
freeswitch.consoleLog("info", "All Answered\n");

ivr_invalid_msg = sound_file_folder .. "invalid_msg.wav"
main_msg = sound_file_folder .. "default_autoattendant.wav"
-- Play with Execute
session:execute("playback", main_msg)
-- Play with StreamFile
session:streamFile(ivr_invalid_msg);
dialstr_main  = dialstr_main .. dialstr_prefix .. breakoutcode ..
"02031701665"
session:setVariable("404_dial",dialstr_main)
session:setVariable("404_tag","IVR")


RELEVANT LOGS :
Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Regex (PASS)
[IVR_FROM_MYS                          QL] destination_number(4042031956241)
=~ /^(404)/ break=on-false
Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
lua(ivr_mysql.lua $                          {destination_number:3}) INLINE
EXECUTE sofia/external/2031701665 at 194.0.147.16:5060 lua(ivr_mysql.lua
2031956241                          )
2010-12-01 11:56:01.525426 [INFO] switch_cpp.cpp:584 Sending early media
2010-12-01 11:56:01.525426 [DEBUG] sofia_glue.c:2972 AUDIO RTP
[sofia/external/2                          031701665 at 194.0.147.16:5060]
10.5.2.105 port 29900 -> 194.0.147.164 port 50202 c
odec: 8 ms: 20
2010-12-01 11:56:01.525426 [DEBUG] switch_rtp.c:1418 Starting timer [soft]
160 b                          ytes per 20ms
2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3190 Set 2833 dtmf send
payload                           to 101
2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3195 Set 2833 dtmf receive
paylo                          ad to 101
2010-12-01 11:56:01.532280 [DEBUG] mod_sofia.c:2172 Ring SDP:
v=0
o=FreeSWITCH 1291174661 1291174662 IN IP4 10.5.2.105
s=FreeSWITCH
c=IN IP4 10.5.2.105
t=0 0
m=audio 29900 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
2010-12-01 11:56:01.532280 [NOTICE] mod_sofia.c:2175 Pre-Answer
sofia/external/2                          031701665 at 194.0.147.16:5060!
2010-12-01 11:56:01.532280 [DEBUG] switch_channel.c:2544
(sofia/external/2031701                          665 at 194.0.147.16:5060)
Callstate Change RINGING -> EARLY
2010-12-01 11:56:01.534727 [DEBUG] sofia.c:4576 Channel
sofia/external/203170166
5 at 194.0.147.16:5060skipping state [early][183]
2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:676 Send signal
sofia/e                          xternal/2031701665 at 194.0.147.16:5060[BREAK]
2010-12-01 11:56:01.534727 [INFO] switch_cpp.cpp:1181 All Answered
2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application
playba                          ck Requires media! pre_answering channel
sofia/external/2031701665 at 194.0.147.16:                          5060
EXECUTE sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswit
ch/sounds/svc_sound_files/default_autoattendant.wav)
2010-12-01 11:56:01.537644 [DEBUG] switch_cpp.cpp:972
sofia/external/2031701665@
194.0.147.16:5060destroy/unlink session from object
Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
set(effective_calle                          r_id_name=${404_tag})
Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
bridge(${404_dial})
2010-12-01 11:56:01.537644 [DEBUG] switch_core_state_machine.c:119
(sofia/extern                          al/2031701665 at 194.0.147.16:5060)
State Change CS_ROUTING -> CS_EXECUTE
2010-12-01 11:56:01.537644 [DEBUG] switch_core_session.c:1057 Send signal
sofia/                          external/2031701665 at 194.0.147.16:5060[BREAK]
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From adminjew at gmail.com  Thu Dec  2 07:23:25 2010
From: adminjew at gmail.com (Yitzchok)
Date: Thu, 2 Dec 2010 10:23:25 -0500
Subject: [Freeswitch-users] building mod_managed fails
In-Reply-To: <AANLkTin-r-sxmaeuD677-__DkFJo5TiXN+ZaX6NcLDxB@mail.gmail.com>
References: <AANLkTi=96X=WyXFZ1ZTU8AE3fKiZUm6Q=tr-ds4kOCWa@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703681DD6C9@mse17be1.mse17.exchange.ms>
	<AANLkTi=xLHN7SAL9t0A+aiDtGTwYoCDwqNFjMapSGQpu@mail.gmail.com>
	<AANLkTimWZYJJEyp9Ayu-Ou3asR0kqbZ2EcjVq8mLAk_c@mail.gmail.com>
	<4CF66CBF.9080308@puzzled.xs4all.nl>
	<AANLkTim9EFf4HxHt=hh+d2YzP-e+mBzoCodfGqMdQCDg@mail.gmail.com>
	<AANLkTin-r-sxmaeuD677-__DkFJo5TiXN+ZaX6NcLDxB@mail.gmail.com>
Message-ID: <AANLkTikP+FNuKxHZTP3k7PHBSM1_OzizeRTdKrwNWWX_@mail.gmail.com>

Is it possible that some of the changes that are in 2.8 are already in
2.6.7?

Try removing the reference to glib (in freeswitch.i, freeswitch_managed.h)
and replacing g_free in freeswitch_managed.h to mono_free then run reswig



Yitzchok


On Thu, Dec 2, 2010 at 1:09 AM, babak yakhchali
<babak.freeswitch at gmail.com>wrote:

> thank u all for ur answers
> Yitzchok yes I'm sure I'm using mono 2.6.7. I started from mono 2.8 and
> after it failed I used mono 2.6.7 (on a clean centos 5.5 not the one I
> installed mono 2.8) and I got the same results. I tried 2.6 and it worked
> and David I'm on centos 5.5 too.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at celliax.org  Thu Dec  2 07:31:15 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 2 Dec 2010 16:31:15 +0100
Subject: [Freeswitch-users] : Skypopen error
In-Reply-To: <AANLkTimmhePNeGQxo_EQ+DxjfVG4-uneRkE9_7eDuENE@mail.gmail.com>
References: <AANLkTi=W65whZv0f7p6GxfjJWmsG9=OXSs-Tt3jmG1tX@mail.gmail.com>
	<AANLkTi=2f9B2upJx8kg64K4NbRGaiw9fOcULET=4THoM@mail.gmail.com>
	<AANLkTinBeEcX0-6gffYsWaJLNU8j8R+t31hZwMbzX7R+@mail.gmail.com>
	<AANLkTinYqCUDiK=hLKPUJ9Q04VUkGgyHE_BKujasd4Cg@mail.gmail.com>
	<AANLkTiks99vbGmG72fheTRb2QdCkQqQQ01gFZRsjxT2h@mail.gmail.com>
	<AANLkTimcYMYuV1pxo60JhQkfK0+P94sf=b_Er8FJrwP_@mail.gmail.com>
	<AANLkTimcCSMfyaajth4qQBiyDGDfhVpPCJyeVt9-VxdF@mail.gmail.com>
	<AANLkTimgOsmXR6jC8pOhB7URwdn19H91ZtkRC9sfBasU@mail.gmail.com>
	<AANLkTinks_EtXZ-y7a=Xux80XrJey1dAoOr8hnJT64aS@mail.gmail.com>
	<4CF66DB0.9090807@puzzled.xs4all.nl>
	<AANLkTimqo4ZeodHr8WVOz28Uq9MyOuE1a11y33+diE2N@mail.gmail.com>
	<AANLkTinp7W6d3GOh2C=rA748ceQmni+tFA3QNkSu5qj=@mail.gmail.com>
	<AANLkTi=aOgFgELyKVHBAWpLq33a29AepH7Ur+v4YCw1r@mail.gmail.com>
	<AANLkTiks+WPpo6P5qvaeqmEydNQX3RbTBNECkbcsx612@mail.gmail.com>
	<AANLkTinruDtk6J_cJHWp_uguS=tBE6H98NEmi1r8MtFD@mail.gmail.com>
	<AANLkTimAtVns8TxXby+F_dWLGju_u2h+wBiV81gBUXLo@mail.gmail.com>
	<AANLkTimmhePNeGQxo_EQ+DxjfVG4-uneRkE9_7eDuENE@mail.gmail.com>
Message-ID: <AANLkTinab5bCsTW3S=qWMBPF=KrKntF4_3v6vVFOFHvO@mail.gmail.com>

On Thu, Dec 2, 2010 at 3:34 PM, Jian Ren <renjian at gmail.com> wrote:
> Cool, I did a git clone last night after 7pm, which should contain the fix
> already, right?

Don't know, you can check like this:

# cd /usr/src/freeswitch/src/mod/endpoint/mod_skypopen
# git log

And you'll see the last commits that you got.

-giovanni

> Maybe that's why I haven't seen the problem so far.
> Besides, I disconnected her 2.4G cordless phone and replaced with a 900Hz
> one, in case if it's caused by the Wifi interference.
>
> Thanks!
> Jian
>
> On Thu, Dec 2, 2010 at 9:22 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> On Thu, Dec 2, 2010 at 3:15 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>> > Her phone might have a *really* crappy DTMF detector. ?They are more
>> > likely
>> > to fail miserably with female voices.
>>
>> Thanks Rupa!
>>
>> Btw, should be fixed in mod_skypopen code since yesterday. You'll now
>> get a warning, but no more an error, and the call should be
>> unaffected.
>>
>> -giovanni
>>
>> >
>> > On Wed, Dec 1, 2010 at 11:18 AM, Jian Ren <renjian at gmail.com> wrote:
>> >>
>> >> Hmm, so far the only "customer" is my wife, who keeps complaining to me
>> >> everyday. It explains why never happened to me. I will ask her tonight
>> >> how
>> >> she sent A or D while calling. Or maybe it's the problem of the phone.
>> >> She
>> >> is using a dual mode phone. Before, it's connected to PC with USB for
>> >> skype
>> >> calls on Windows. Now I stopped running any version of skype on windows
>> >> and
>> >> the USB was disconnected so she is using the phone as a normal one.
>> >> Thanks!
>> >> Jian
>> >>
>> >> On Wed, Dec 1, 2010 at 11:56 AM, Giovanni Maruzzelli
>> >> <gmaruzz at gmail.com>
>> >> wrote:
>> >>>
>> >>> Hi Jian,
>> >>>
>> >>> I hope in English "bizarre" does not sound bad, in Italian would be
>> >>> like "original and out of standard in a funny way" :).
>> >>>
>> >>> >From the log you attached to the Jira, the incoming SIP calls that
>> >>> are
>> >>> then bridged to skypopen are sending both the "A" dtmf and the "D"
>> >>> dtmf (can't remember if any other).
>> >>>
>> >>> You can peruse the log looking for "error 21", and you'll ?see is
>> >>> anytime that they sent to you (via SIP) one of the A-B-C-D dtmfs that
>> >>> mod_skypopen duly passes to skype.
>> >>>
>> >>> Problem is: the Skype client does not accept or relay dtmf "A-D", and
>> >>> spit out an error.
>> >>>
>> >>> Out of curiosity you may want to check why your customers are using
>> >>> dtrmf A-D, but is not an absolut need.
>> >>>
>> >>> Anyway, I'll fix this in mod_skypopen code asap, so that if another
>> >>> channel (SIP in your case) try to send A-D to skype, that dtmf will be
>> >>> ignored and a warning line will be emitted to console and to logfile.
>> >>> And no more errors or aborted calls.
>> >>>
>> >>> -giovanni
>> >>>
>> >>>
>> >>>
>> >>> On Wed, Dec 1, 2010 at 5:35 PM, Jian Ren <renjian at gmail.com> wrote:
>> >>> > Also this happened during a call, which got hanged up because of the
>> >>> > error.
>> >>> > So the phone sends out A-D during a call?
>> >>> > Jian
>> >>> > On Wed, Dec 1, 2010 at 11:33 AM, Jian Ren <renjian at gmail.com> wrote:
>> >>> >>
>> >>> >> Interesting. I don't have these keys on my phone. Here is the
>> >>> >> dialplan
>> >>> >> string I am using in the ATA(SPA1001):
>> >>> >> (<:1>[2-9]xx[2-9]xxxxxx|011xx.|1[2-9]xx[2-9]xxxxxx|1xxx|00xx.)
>> >>> >> Shouldn't it only take numbers?
>> >>> >> Could this be caused by any dialplan XML files in freeswitch?
>> >>> >> Thanks!
>> >>> >> Jian
>> >>> >>
>> >>> >> On Wed, Dec 1, 2010 at 10:45 AM, Patrick Lists
>> >>> >> <freeswitch-list at puzzled.xs4all.nl> wrote:
>> >>> >>>
>> >>> >>> On 12/01/2010 04:03 PM, Jian Ren wrote:
>> >>> >>> > Hi Giovanni,
>> >>> >>> > Could you explain more to a bizarre person like me :-)
>> >>> >>> > What's DTMF "D"? Sent from one of my SIP client? Or Someone sent
>> >>> >>> > to
>> >>> >>> > my
>> >>> >>> > skype through chat? If it's abnormal, I think I should fix the
>> >>> >>> > root
>> >>> >>> > cause.
>> >>> >>> > Thank you very much for looking at the issue, with 2M logs :-).
>> >>> >>>
>> >>> >>> DTMF A-D seems to be used only on military phones:
>> >>> >>>
>> >>> >>>
>> >>> >>> http://www.telecomdictionary.com/telecom_dictionary_DTMF_definition.html
>> >>> >>>
>> >>> >>> Regards,
>> >>> >>> Patrick
>> >>> >>>
>> >>> >>> _______________________________________________
>> >>> >>> FreeSWITCH-users mailing list
>> >>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>>
>> >>> >>>
>> >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >>> http://www.freeswitch.org
>> >>> >>
>> >>> >
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> >
>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> Sincerely,
>> >>>
>> >>> Giovanni Maruzzelli
>> >>> Cell : +39-347-2665618
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > -Rupa
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From riedinger at sns.eu  Thu Dec  2 07:51:26 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Thu, 02 Dec 2010 16:51:26 +0100
Subject: [Freeswitch-users] How to force the usage of g729 instead of g729b
Message-ID: <4CF7C07E.9080707@sns.eu>

By setting global_codec_prefs and inbound_codec_prefs it's possible to 
specify, which codecs are supported for inbound calls. How can I specify 
that the switch shall support g729, but not g729b?

The incoming INVITE message contains a SDP with the media attribute 
annexb=yes. How has Freeswitch to answer to indicate that g729 is 
supported but g729b is not supported?

Thank you in advance
         Jan

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973



From gmaruzz at celliax.org  Thu Dec  2 08:29:31 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 2 Dec 2010 17:29:31 +0100
Subject: [Freeswitch-users] How to filter Skypopen chat message in ESL
	mode?
In-Reply-To: <AANLkTi=HGgC4BS6chKMWmU+dwnEFXLvgOq1yFB=pUC+q@mail.gmail.com>
References: <AANLkTi=T8G8LjKytzpSTw+bjGUZS7_3TAJDySxKevC0+@mail.gmail.com>
	<AANLkTi=HstdSAsVqe836AXpYwvTg8r3_UdovPCbrLUaO@mail.gmail.com>
	<AANLkTimW=fueqE-iMhPQ03+UEO5dj1-Mghdj7b-ZoM1-@mail.gmail.com>
	<AANLkTik5fMn4Yi_bU_KFKyufdJ1b13AOBD2-H47UHNWQ@mail.gmail.com>
	<AANLkTi=eE+AQ-F-xndR-6zBajomHY-a5A_8D1ULrZwM_@mail.gmail.com>
	<AANLkTinjo9k-UgOm-kF4dkYVn7XbXZz4MzS=g_RrmZmS@mail.gmail.com>
	<AANLkTik-NdpdJRYxBWZ1KYOOk8xxS59KWiJJmMeFERiP@mail.gmail.com>
	<AANLkTi=n4nbyjKcxGR4pM74XWLzC5VqR2nSJtuiN0WHa@mail.gmail.com>
	<AANLkTin8rCY8t6+buLn_N3RhBR768txZHPgrUOwCdacD@mail.gmail.com>
	<AANLkTin0EPm2LLUMCNHHD8oh+2XZk59PkO0Jxu85+Uki@mail.gmail.com>
	<AANLkTimUWBDyvN7-NhFR1JWEc3pBnQtnnfcHC8UEjVXd@mail.gmail.com>
	<AANLkTi=HGgC4BS6chKMWmU+dwnEFXLvgOq1yFB=pUC+q@mail.gmail.com>
Message-ID: <AANLkTimBd0msAdo0uGP7RfSek3MWd7g3vkQg_BPar5mL@mail.gmail.com>

On Thu, Dec 2, 2010 at 3:40 PM, xuyan yang <xyangni at gmail.com> wrote:
>>
>> 1) Is that the complete skypopen.conf.xml? I mean, do you use one only
>> interface?
>
> Yes, it is complete. During the test, I use only 1 client with latest
> version running on ubuntu GUI. The older version can not be downloaded from
> skype now.
>>

OK Xuyan, thanks for the infos.

I'll test it asap (some hours).

Maybe the problem comes from using the beta (2.1.81) version of the
Skype client.

The only Skype client version that is supported (because is the only
one that works correctly) is the 2.0.72. I know that version is not on
the Skype website (and I cannot give it to you, because Skype policies
on distribution are probably not allowing me to distribute it), but
maybe you can find someone that gives it to you.

2.1.81 is known to give a lot of problems of various nature (this one
is very small), and will never be supported by mod_skypopen.

Anyway, I'll test your problem asap with 2.0.72.

-giovanni

>>
>>
>> 2) An incoming skype call goes directly (because is in
>> skypopen.conf.xml) to 5655 and ear the continuous message.
>
> yes, go directly to 5655.
>
>>
>> 3) While the incoming skype call is hearing the message, the same
>> incoming skypeclient sends chatmessages to the skypopen skypeclient
>
> yes, you are right
>
>>
>> 4) You connect via telnet to the ESL port, do the "events plain message"
>
> not exactly, but I think fs_cli should function the same as telnet.
>>
>>
>>
>> 5) You get sometimes those longer events intermixed with the regular
>> message events, but not everytime
>> yes, you are right.
>
>
>>
>> Is this correct? If this is correct I will test asap and fix the
>> possible problem.
>>
>> Btw, I tested yesterday with skypeclient not in a call and I got no
>> problems at all. I don't think to be in a call would make any
>> difference.
>
> ?It is not the call, but the?continuous sound playback in the call??which
> cause the problem. If the call is parked idle, there is no problem.
>>
>>
>>
>> Please be certain you gave me all the info needed to exactly replicate
>> your problem.
>>
>> -giovanni
>>
>> On Thu, Dec 2, 2010 at 12:09 PM, xuyan yang <xyangni at gmail.com> wrote:
>> > OK. I will try to make it clear.
>> > In general, my test case is to forward skype call to a extention such as
>> > 5655. And then in the dialplan for 5655,?continuously?play some sound.
>> > During playing, you can send skype chat messages from the caller and
>> > check
>> > what esl events has been generated from it. According to my test, the
>> > raw
>> > type event is not generated for most of chat messages.
>> > In details,
>> > this is my skypeopen.conf.xml:
>> > <configuration name="skypopen.conf" description="Skypopen
>> > Configuration">
>> > ??<global_settings>
>> > ?? ?<param name="debug" value="8"/>
>> > ?? ?<param name="dialplan" value="XML"/>
>> > ?? ?<param name="context" value="default"/>
>> > ?? ?<param name="destination" value="1000"/>
>> > ?? ?<param name="skype_user" value="user1"/>
>> > ?? ?<param name="report_incoming_chatmessages" value="true"/>
>> > ?? ?<param name="silent_mode" value="false"/>
>> > ?? ?<param name="write_silence_when_idle" value="true"/>
>> > ??</global_settings>
>> > ??<!-- one entry here per each skypopen interface -->
>> > ??<per_interface_settings>
>> > ?? ?<interface id="1" name="interface1">
>> > ?? ? ? ?<param name="X11-display" value=":0.0"/>
>> > ?? ? ? ?<param name="skype_user" value="xxxxxxxxx"/>
>> > ?? ? ? ?<param name="destination" value="5655"/>
>> > ?? ?</interface>
>> > ??</per_interface_settings>
>> > </configuration>
>> >
>> >
>> > in dialplan for default context, add the following to describe extention
>> > 5655:
>> > <extension name="skype" ?break="on-false">
>> > <condition field="destination_number" expression="^5655$">
>> > ?? ? ? ?<action application="answer"/>
>> > ?? ? ? ?<action application="sleep" data="1000"/>
>> > <action application="playback" data="ivr/ivr-sample_submenu.wav"/>
>> > <action
>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>> > application="playback" data="ivr/ivr-sample_submenu.wav"/>
>> > ?? </condition>
>> > </extension>
>> > ivr-sample_submenu.wav is a sound file in en/us/callie/ivr/8000, but it
>> > can
>> > be replaced by any other wav files.
>> > after setup, make a call to your skype user xxxxxxxxx.
>> > in the mean time use fs_cli to connect FS and execute /event plain
>> > MESSAGE
>> > to listen esl events.
>> > sent any chat messages from caller, then you are very likely to find
>> > only
>> > the short version events.
>> >
>> >
>> > On Thu, Dec 2, 2010 at 8:50 AM, Giovanni Maruzzelli
>> > <gmaruzz at celliax.org>
>> > wrote:
>> >>
>> >> Xuyan,
>> >>
>> >> please.
>> >>
>> >> Can you give a clear defined procedure to reproduce the problem?
>> >>
>> >> Please, take the time to write it, complete of all relevant info
>> >> (dialplan, script, etc etc) so I can reproduce it from a freshly
>> >> installed FreeSWITCH.
>> >>
>> >> If you don't give this information I cannot fix the problem. And I
>> >> cannot neither read your mind, nor finding the time to try every
>> >> possible combination.
>> >>
>> >> Please, post something someone can cut and paste and a clear step by
>> >> step procedure to replicate.
>> >>
>> >> -giovanni
>> >>
>> >> On Thu, Dec 2, 2010 at 4:45 AM, xuyan yang <xyangni at gmail.com> wrote:
>> >> > I tried it again. When the line is parked and idle. the raw events
>> >> > can
>> >> > always be generated.
>> >> > but if the line is kept busy such as the following case this events
>> >> > is
>> >> > trend
>> >> > to have problem
>> >> > <extension name="skype-in" ?break="on-false">
>> >> > <condition field="destination_number" expression="^5655$">
>> >> > ?? ? ? ?<action application="answer"/>
>> >> > ?? ? ? ?<action application="sleep" data="1000"/>
>> >> > <!--<action application="park"/> -->
>> >> > <action application="playback" data="test.wav"/>
>> >> > <action application="playback" data="test.wav"/>
>> >> > <action application="playback" data="test.wav"/>
>> >> > ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<action application="playback"
>> >> > data="test.wav"/>
>> >> > <action application="playback" data="test.wav"/>
>> >> > <action application="playback" data="test.wav"/>
>> >> > ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<action application="playback"
>> >> > data="test.wav"/>
>> >> > <action application="playback" data="test.wav"/>
>> >> > <action application="playback" data="test.wav"/>
>> >> > ?? </condition>
>> >> > </extension>
>> >> >
>> >> >
>> >> > On Thu, Dec 2, 2010 at 2:01 AM, Giovanni Maruzzelli
>> >> > <gmaruzz at celliax.org>
>> >> > wrote:
>> >> >>
>> >> >> I repeat, if you subscribe to "message" events you get only those.
>> >> >> Also, how I can replicate your problem?
>> >> >> I've tested events and I had no problem at all with spurious or
>> >> >> unreliable events in mod-skypopen.
>> >> >> Please, can you indicate a detailed way to reproduce your problem?
>> >> >> -giovanni
>> >> >>
>> >> >> On 12/2/10, xuyan yang <xyangni at gmail.com> wrote:
>> >> >> > I got it. Then the problem should be the unstable behavior of raw
>> >> >> > events
>> >> >> > which has about 25% chances of being missed.
>> >> >> >
>> >> >> > Fortunately, I have found a way to avoid this issue. Ignore all
>> >> >> > message
>> >> >> > events which contains Unique-ID field.
>> >> >> >
>> >> >> > On Wed, Dec 1, 2010 at 11:18 PM, Giovanni Maruzzelli
>> >> >> > <gmaruzz at celliax.org>wrote:
>> >> >> >
>> >> >> >> On Thu, Dec 2, 2010 at 12:15 AM, Giovanni Maruzzelli
>> >> >> >> <gmaruzz at celliax.org> wrote:
>> >> >> >> > If you subscribe only to "MESSAGE" kind of events, you'll get
>> >> >> >> > only
>> >> >> >> > those.
>> >> >> >> >
>> >> >> >> > The other events are "raw" events, that other users have
>> >> >> >> > requested
>> >> >> >> > for
>> >> >> >> > other purposes.
>> >> >> >>
>> >> >> >> or at ?least that is the expected behavior, please let me know if
>> >> >> >> I
>> >> >> >> introduced some regression in integrating that "raw event"
>> >> >> >> thingy.
>> >> >> >>
>> >> >> >> -giovanni
>> >> >> >>
>> >> >> >> >
>> >> >> >> > -giovanni
>> >> >> >> >
>> >> >> >> > On Wed, Dec 1, 2010 at 11:35 PM, xuyan yang <xyangni at gmail.com>
>> >> >> >> > wrote:
>> >> >> >> >> Thanks for your reply. I have read this page before. but the
>> >> >> >> >> random
>> >> >> >> emerging
>> >> >> >> >> of the second verbose esl events is causing trouble. And I
>> >> >> >> >> guess
>> >> >> >> >> it
>> >> >> >> >> may
>> >> >> >> be a
>> >> >> >> >> bug, or it should be predictable.
>> >> >> >> >> Before reporting to jira, I just want to check whether I have
>> >> >> >> >> made
>> >> >> >> >> any
>> >> >> >> >> mistake.
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> On Wed, Dec 1, 2010 at 9:35 PM, Giovanni Maruzzelli
>> >> >> >> >> <gmaruzz at gmail.com>
>> >> >> >> >> wrote:
>> >> >> >> >>>
>> >> >> >> >>>
>> >> >> >> >>>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#MESSAGE_.28Chat.29
>> >> >> >> >>>
>> >> >> >> >>> On Wed, Dec 1, 2010 at 10:23 PM, xuyan yang
>> >> >> >> >>> <xyangni at gmail.com>
>> >> >> >> >>> wrote:
>> >> >> >> >>> > hi,
>> >> >> >> >>> > I am writing ESL program on FreeSWITCH Version 1.0.head
>> >> >> >> >>> > (git-8825b6e
>> >> >> >> >>> > 2010-11-28 17-15-39 -0500)
>> >> >> >> >>> > I need to handle skype chat message with a inbound ESL
>> >> >> >> >>> > connection.
>> >> >> >> But
>> >> >> >> >>> > there
>> >> >> >> >>> > are 1 or 2 esl events received randomly for each message.
>> >> >> >> >>> > the first short one is alway generated, but the second one
>> >> >> >> >>> > is
>> >> >> >> >>> > random.
>> >> >> >> So
>> >> >> >> >>> > when trying to filter according to UUID, nothing is caught
>> >> >> >> >>> > in
>> >> >> >> >>> > many
>> >> >> >> >>> > cases.
>> >> >> >> >>> > If I make filter based on skype ID, duplicated messages are
>> >> >> >> >>> > often
>> >> >> >> heard.
>> >> >> >> >>> > I do not why the behavior of the second event is random.
>> >> >> >> >>> > How should I setup the filter to get 1 and only 1 event for
>> >> >> >> >>> > each
>> >> >> >> >>> > chat
>> >> >> >> >>> > message? Thanks.
>> >> >> >> >>> >
>> >> >> >> >>> > the first is a short one with the following header:
>> >> >> >> >>> > ?[Event-Name] = [MESSAGE]
>> >> >> >> >>> > ?[Event-Calling-Function] = [incoming_chatmessage]
>> >> >> >> >>> > ?[Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
>> >> >> >> >>> > ?[Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
>> >> >> >> >>> > ?[Content-Length] = [1]
>> >> >> >> >>> > ?[subject] = [SIMPLE MESSAGE]
>> >> >> >> >>> > ?[FreeSWITCH-IPv4] = [192.168.0.3]
>> >> >> >> >>> > ?[hint] = [niqizhi]
>> >> >> >> >>> > ?[from] = [niqizhi]
>> >> >> >> >>> > ?[Event-Date-Local] = [2010-12-01 21:02:37]
>> >> >> >> >>> > ?[proto] = [skype]
>> >> >> >> >>> > ?[FreeSWITCH-IPv6] = [::1]
>> >> >> >> >>> > ?[id] = [5334]
>> >> >> >> >>> > ?[Event-Calling-File] = [mod_skypopen.c]
>> >> >> >> >>> > ?[Event-Date-Timestamp] = [1291237357051788]
>> >> >> >> >>> > ?[FreeSWITCH-Hostname] = [EYSRV]
>> >> >> >> >>> > ?[login] = [interface1]
>> >> >> >> >>> > ?[during-call] = [true]
>> >> >> >> >>> > ?[Event-Calling-Line-Number] = [2915]
>> >> >> >> >>> > ?[chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
>> >> >> >> >>> > the second is much longer:
>> >> >> >> >>> > ?[Caller-Source] = [mod_skypopen]
>> >> >> >> >>> > ?[Event-Calling-Function] = [incoming_chatmessage]
>> >> >> >> >>> > ?[Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
>> >> >> >> >>> > ?[Answer-State] = [answered]
>> >> >> >> >>> > ?[FreeSWITCH-IPv4] = [192.168.0.3]
>> >> >> >> >>> > ?[Channel-State] = [CS_EXECUTE]
>> >> >> >> >>> > ?[Channel-Read-Codec-Bit-Rate] = [256000]
>> >> >> >> >>> > ?[FreeSWITCH-IPv6] = [::1]
>> >> >> >> >>> > ?[Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
>> >> >> >> >>> > ?[Channel-Read-Codec-Rate] = [16000]
>> >> >> >> >>> > ?[Caller-Destination-Number] = [5655]
>> >> >> >> >>> > ?[Caller-Channel-Transfer-Time] = [0]
>> >> >> >> >>> > ?[Channel-Call-State] = [ACTIVE]
>> >> >> >> >>> > ?[Caller-Channel-Progress-Media-Time] = [0]
>> >> >> >> >>> > ?[FreeSWITCH-Hostname] = [EYSRV]
>> >> >> >> >>> > ?[Caller-Channel-Answered-Time] = [1291237326697085]
>> >> >> >> >>> > ?[login] = [interface1]
>> >> >> >> >>> > ?[during-call] = [true]
>> >> >> >> >>> > ?[Channel-Name] = [skypopen/interface1]
>> >> >> >> >>> > ?[Caller-Unique-ID] =
>> >> >> >> >>> > [412764c0-fd8e-11df-9019-835ae03a7500]
>> >> >> >> >>> > ?[Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
>> >> >> >> >>> > ?[Channel-Read-Codec-Name] = [L16]
>> >> >> >> >>> > ?[Caller-Channel-Name] = [skypopen/interface1]
>> >> >> >> >>> > ?[Caller-Caller-ID-Number] = [niqizhi]
>> >> >> >> >>> > ?[Event-Date-Timestamp] = [1291237357051788]
>> >> >> >> >>> > ?[Channel-State-Number] = [4]
>> >> >> >> >>> > ?[Event-Calling-Line-Number] = [2888]
>> >> >> >> >>> > ?[chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
>> >> >> >> >>> > ?[Event-Name] = [MESSAGE]
>> >> >> >> >>> > ?[Content-Length] = [1]
>> >> >> >> >>> > ?[subject] = [SIMPLE MESSAGE]
>> >> >> >> >>> > ?[Caller-Caller-ID-Name] = [niqizhi]
>> >> >> >> >>> > ?[from] = [niqizhi]
>> >> >> >> >>> > ?[Caller-Dialplan] = [XML]
>> >> >> >> >>> > ?[Caller-Channel-Hangup-Time] = [0]
>> >> >> >> >>> > ?[id] = [5334]
>> >> >> >> >>> > ?[Caller-Profile-Index] = [1]
>> >> >> >> >>> > ?[Caller-Direction] = [inbound]
>> >> >> >> >>> > ?[Caller-Username] = [skypopen]
>> >> >> >> >>> > ?[Channel-Write-Codec-Name] = [L16]
>> >> >> >> >>> > ?[Call-Direction] = [inbound]
>> >> >> >> >>> > ?[Caller-Screen-Bit] = [true]
>> >> >> >> >>> > ?[hint] = [niqizhi]
>> >> >> >> >>> > ?[Caller-Privacy-Hide-Number] = [false]
>> >> >> >> >>> > ?[Event-Date-Local] = [2010-12-01 21:02:37]
>> >> >> >> >>> > ?[proto] = [skype]
>> >> >> >> >>> > ?[Caller-Channel-Created-Time] = [1291237326468855]
>> >> >> >> >>> > ?[Event-Calling-File] = [mod_skypopen.c]
>> >> >> >> >>> > ?[Caller-Channel-Progress-Time] = [0]
>> >> >> >> >>> > ?[Caller-Privacy-Hide-Name] = [false]
>> >> >> >> >>> > ?[Channel-Write-Codec-Rate] = [16000]
>> >> >> >> >>> > ?[Caller-Context] = [default]
>> >> >> >> >>> > ?[Channel-Write-Codec-Bit-Rate] = [256000]
>> >> >> >> >>> > ?[Presence-Call-Direction] = [inbound]
>> >> >> >> >>> > ?[Caller-Profile-Created-Time] = [1291237326468855]
>> >> >> >> >>> > _______________________________________________
>> >> >> >> >>> > FreeSWITCH-users mailing list
>> >> >> >> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> >>> >
>> >> >> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >>> > UNSUBSCRIBE:
>> >> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> >>> > http://www.freeswitch.org
>> >> >> >> >>> >
>> >> >> >> >>> >
>> >> >> >> >>>
>> >> >> >> >>>
>> >> >> >> >>>
>> >> >> >> >>> --
>> >> >> >> >>> Sincerely,
>> >> >> >> >>>
>> >> >> >> >>> Giovanni Maruzzelli
>> >> >> >> >>> Cell : +39-347-2665618
>> >> >> >> >>>
>> >> >> >> >>> _______________________________________________
>> >> >> >> >>> FreeSWITCH-users mailing list
>> >> >> >> >>> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >>> UNSUBSCRIBE:
>> >> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> >>> http://www.freeswitch.org
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >> _______________________________________________
>> >> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >> UNSUBSCRIBE:
>> >> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> >> http://www.freeswitch.org
>> >> >> >> >>
>> >> >> >> >>
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > --
>> >> >> >> > Sincerely,
>> >> >> >> >
>> >> >> >> > Giovanni Maruzzelli
>> >> >> >> > Cell : +39-347-2665618
>> >> >> >> >
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> --
>> >> >> >> Sincerely,
>> >> >> >>
>> >> >> >> Giovanni Maruzzelli
>> >> >> >> Cell : +39-347-2665618
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> http://www.freeswitch.org
>> >> >> >>
>> >> >> >
>> >> >>
>> >> >> --
>> >> >> Sent from my mobile device
>> >> >>
>> >> >> Sincerely,
>> >> >>
>> >> >> Giovanni Maruzzelli
>> >> >> Cell : +39-347-2665618
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Sincerely,
>> >>
>> >> Giovanni Maruzzelli
>> >> Cell : +39-347-2665618
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From bernhard.suttner at winet.ch  Thu Dec  2 08:31:17 2010
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Thu, 2 Dec 2010 17:31:17 +0100
Subject: [Freeswitch-users] Export a channel variable within Dial String
In-Reply-To: <B724748E-DF66-4CD4-B987-BED15F266C78@ipeva.fr>
References: <481d3184-6518-4957-acca-b1e73c7fec7e@winet.ch>
	<B724748E-DF66-4CD4-B987-BED15F266C78@ipeva.fr>
Message-ID: <1ff0ebed-7410-4bdb-ab4c-619cadd564ed@winet.ch>

The channel variable will be set, but not exported like the ?export? dialplan function does export the variable to the new channel.
 
 
 
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von David Ponzone
Gesendet: Donnerstag, 2. Dezember 2010 16:14
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Export a channel variable within Dial String
 
Bernhard,
 
AFAIU, if you specify the variable in the dialstring, it's for leg B, so it's exported.
If you wan't it, don't set it :)
 
And for leg A, well, you only have one leg A, so you can't set it twice.
 
David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34
 
Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com
 
Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
 



 
Le 02/12/2010 ? 15:48, Bernhard Suttner a ?crit :



Hi,

I call multiple destination with bridge like that:

[var1=value]sofia/default/user1,[var1=another_value]loopback/user2

Is it somehow possible, that var1=another_value will be exported but var1=value not? What I want is, that I could specify a variable to be exported _within_ the dialstring?

Thanks for every hint.

Best regards,
Bernhard



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 
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From jgallartm at gmail.com  Thu Dec  2 08:31:03 2010
From: jgallartm at gmail.com (Javier Gallart)
Date: Thu, 2 Dec 2010 17:31:03 +0100
Subject: [Freeswitch-users] Sangoma D100 usage
Message-ID: <AANLkTim5J6nkczEvnDzi5Di+YanvWN+YpRva9W-ifY=8@mail.gmail.com>

On Thu, Dec 2, 2010 at 2:56 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:

> Send FreeSWITCH-users mailing list submissions to
>        freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
>        freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
>        freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
>   1. Sangoma D100 usage (Javier Gallart)
>   2. Re: Sangoma D100 usage (Steven Ayre)
>   3. Re: Sangoma D100 usage (David Ponzone)
>   4. Re: How to filter Skypopen chat message in ESL    mode?
>      (Giovanni Maruzzelli)
>
>
> ---------- Forwarded message ----------
> From: Javier Gallart <jgallartm at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 2 Dec 2010 13:55:33 +0100
> Subject: [Freeswitch-users] Sangoma D100 usage
> Hello
>
> we got a Sangoma D100 transcoding card, and we like to get it working with
> our freeswitch. Freeswitch and the Sangoma card are running on the same
> server. The module is properly loaded:
>
>  2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1102 Reading
> sangoma codec configuration
> 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1126 Found Sangoma
> RTP IP x.x.x.x
> 2010-12-02 07:46:33.393953 [NOTICE] mod_sangoma_codec.c:1193 Detected 1 and
> activated 1 Sangoma codec vocallo modules
> 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1207 Loading codecs,
> register='all', noregister=''
> 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1239 Registering
> implementations for codec PCMU
> ...
> 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering
> implementations for codec PCMA
> ...
> 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering
> implementations for codec G729
> ...
> 2010-12-02 07:46:33.402964 [CONSOLE] switch_loadable_module.c:946
> Successfully Loaded [mod_sangoma_codec]
> 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding
> Codec PCMU 0 Sangoma PCMU 8000hz 10ms 64000bps
> ...
> 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding
> Codec PCMA 8 Sangoma PCMA 8000hz 20ms 64000bps
> ...
> 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding
> Codec G729 18 Sangoma G729 8000hz 40ms 8000bps
> ...
> 2010-12-02 07:46:33.403967 [NOTICE] switch_loadable_module.c:274 Adding API
> Function 'sangoma_codec'
>
> The codecs look also good:
> freeswitch at internal> show codecs
> codec,Sangoma G723,mod_sangoma_codec
> codec,Sangoma G729,mod_sangoma_codec
> codec,Sangoma GSM,mod_sangoma_codec
> codec,Sangoma PCMA,mod_sangoma_codec
> codec,Sangoma PCMU,mod_sangoma_codec
>
> What we don't know is how to actually use the card. We have forced a call
> with g729 in the A-leg and g711 in the b-leg but freeswitch doesn't know how
> to bridge them. I've tried unsuccessfully this config in the vars.xml:
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=Sangoma G729,Sangoma
> PCMU,Sangoma PCMA"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=Sangoma PCMU,Sangoma
> PCMA"/>
>
> Does anybody has any experience usgin this card?
>
> Thanks
>
> Javier
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Steven Ayre <steveayre at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Thu, 2 Dec 2010 13:48:42 +0000
> Subject: Re: [Freeswitch-users] Sangoma D100 usage
> Sangoma is not part of the codec IANA name, just part of the interface
> name (description). Try it like this:
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA"/>
>
> Don't have mod_g729 or mod_com_g729 loaded at the same time.
>
> -Steve
>

Hi Steve

removing the mod_g729 module did the trick.

Thanks!

Javi

>
> On 2 December 2010 12:55, Javier Gallart <jgallartm at gmail.com> wrote:
> > Hello
> > we got a Sangoma D100 transcoding card, and we like to get it working
> with
> > our freeswitch. Freeswitch and the Sangoma card are running on the same
> > server. The module is properly loaded:
> >  2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1102 Reading
> sangoma
> > codec configuration
> > 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1126 Found Sangoma
> > RTP IP x.x.x.x
> > 2010-12-02 07:46:33.393953 [NOTICE] mod_sangoma_codec.c:1193 Detected 1
> and
> > activated 1 Sangoma codec vocallo modules
> > 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1207 Loading
> codecs,
> > register='all', noregister=''
> > 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1239 Registering
> > implementations for codec PCMU
> > ...
> > 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering
> > implementations for codec PCMA
> > ...
> > 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering
> > implementations for codec G729
> > ...
> > 2010-12-02 07:46:33.402964 [CONSOLE] switch_loadable_module.c:946
> > Successfully Loaded [mod_sangoma_codec]
> > 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding
> > Codec PCMU 0 Sangoma PCMU 8000hz 10ms 64000bps
> > ...
> > 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding
> > Codec PCMA 8 Sangoma PCMA 8000hz 20ms 64000bps
> > ...
> > 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding
> > Codec G729 18 Sangoma G729 8000hz 40ms 8000bps
> > ...
> > 2010-12-02 07:46:33.403967 [NOTICE] switch_loadable_module.c:274 Adding
> API
> > Function 'sangoma_codec'
> > The codecs look also good:
> > freeswitch at internal> show codecs
> > codec,Sangoma G723,mod_sangoma_codec
> > codec,Sangoma G729,mod_sangoma_codec
> > codec,Sangoma GSM,mod_sangoma_codec
> > codec,Sangoma PCMA,mod_sangoma_codec
> > codec,Sangoma PCMU,mod_sangoma_codec
> > What we don't know is how to actually use the card. We have forced a call
> > with g729 in the A-leg and g711 in the b-leg but freeswitch doesn't know
> how
> > to bridge them. I've tried unsuccessfully this config in the vars.xml:
> > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=Sangoma G729,Sangoma
> > PCMU,Sangoma PCMA"/>
> > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=Sangoma PCMU,Sangoma
> > PCMA"/>
> > Does anybody has any experience usgin this card?
> > Thanks
> > Javier
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
>
> ---------- Forwarded message ----------
> From: David Ponzone <david.ponzone at ipeva.fr>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Thu, 2 Dec 2010 14:48:48 +0100
> Subject: Re: [Freeswitch-users] Sangoma D100 usage
> Javier,
>
> as Sangoma is working quite closely with FreeSWITCH devs (some Sangoma
> people are actually on this ML), I am pretty sure you can ask Sangoma
> support to help you out.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 02/12/2010 ? 13:55, Javier Gallart a ?crit :
>
> Hello
>
> we got a Sangoma D100 transcoding card, and we like to get it working with
> our freeswitch. Freeswitch and the Sangoma card are running on the same
> server. The module is properly loaded:
>
>  2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1102 Reading
> sangoma codec configuration
> 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1126 Found Sangoma
> RTP IP x.x.x.x
> 2010-12-02 07:46:33.393953 [NOTICE] mod_sangoma_codec.c:1193 Detected 1 and
> activated 1 Sangoma codec vocallo modules
> 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1207 Loading codecs,
> register='all', noregister=''
> 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1239 Registering
> implementations for codec PCMU
> ...
> 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering
> implementations for codec PCMA
> ...
> 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering
> implementations for codec G729
> ...
> 2010-12-02 07:46:33.402964 [CONSOLE] switch_loadable_module.c:946
> Successfully Loaded [mod_sangoma_codec]
> 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding
> Codec PCMU 0 Sangoma PCMU 8000hz 10ms 64000bps
> ...
> 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding
> Codec PCMA 8 Sangoma PCMA 8000hz 20ms 64000bps
> ...
> 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding
> Codec G729 18 Sangoma G729 8000hz 40ms 8000bps
> ...
> 2010-12-02 07:46:33.403967 [NOTICE] switch_loadable_module.c:274 Adding API
> Function 'sangoma_codec'
>
> The codecs look also good:
> freeswitch at internal> show codecs
> codec,Sangoma G723,mod_sangoma_codec
> codec,Sangoma G729,mod_sangoma_codec
> codec,Sangoma GSM,mod_sangoma_codec
> codec,Sangoma PCMA,mod_sangoma_codec
> codec,Sangoma PCMU,mod_sangoma_codec
>
> What we don't know is how to actually use the card. We have forced a call
> with g729 in the A-leg and g711 in the b-leg but freeswitch doesn't know how
> to bridge them. I've tried unsuccessfully this config in the vars.xml:
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=Sangoma G729,Sangoma
> PCMU,Sangoma PCMA"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=Sangoma PCMU,Sangoma
> PCMA"/>
>
> Does anybody has any experience usgin this card?
>
> Thanks
>
> Javier
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> ---------- Forwarded message ----------
> From: Giovanni Maruzzelli <gmaruzz at celliax.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Thu, 2 Dec 2010 14:55:51 +0100
> Subject: Re: [Freeswitch-users] How to filter Skypopen chat message in ESL
> mode?
> 1) Is that the complete skypopen.conf.xml? I mean, do you use one only
> interface?
>
> 2) An incoming skype call goes directly (because is in
> skypopen.conf.xml) to 5655 and ear the continuous message.
>
> 3) While the incoming skype call is hearing the message, the same
> incoming skypeclient sends chatmessages to the skypopen skypeclient
>
> 4) You connect via telnet to the ESL port, do the "events plain message"
>
> 5) You get sometimes those longer events intermixed with the regular
> message events, but not everytime
>
> Is this correct? If this is correct I will test asap and fix the
> possible problem.
>
> Btw, I tested yesterday with skypeclient not in a call and I got no
> problems at all. I don't think to be in a call would make any
> difference.
>
> Please be certain you gave me all the info needed to exactly replicate
> your problem.
>
> -giovanni
>
> On Thu, Dec 2, 2010 at 12:09 PM, xuyan yang <xyangni at gmail.com> wrote:
> > OK. I will try to make it clear.
> > In general, my test case is to forward skype call to a extention such as
> > 5655. And then in the dialplan for 5655, continuously play some sound.
> > During playing, you can send skype chat messages from the caller and
> check
> > what esl events has been generated from it. According to my test, the raw
> > type event is not generated for most of chat messages.
> > In details,
> > this is my skypeopen.conf.xml:
> > <configuration name="skypopen.conf" description="Skypopen Configuration">
> >   <global_settings>
> >     <param name="debug" value="8"/>
> >     <param name="dialplan" value="XML"/>
> >     <param name="context" value="default"/>
> >     <param name="destination" value="1000"/>
> >     <param name="skype_user" value="user1"/>
> >     <param name="report_incoming_chatmessages" value="true"/>
> >     <param name="silent_mode" value="false"/>
> >     <param name="write_silence_when_idle" value="true"/>
> >   </global_settings>
> >   <!-- one entry here per each skypopen interface -->
> >   <per_interface_settings>
> >     <interface id="1" name="interface1">
> >         <param name="X11-display" value=":0.0"/>
> >         <param name="skype_user" value="xxxxxxxxx"/>
> >         <param name="destination" value="5655"/>
> >     </interface>
> >   </per_interface_settings>
> > </configuration>
> >
> >
> > in dialplan for default context, add the following to describe extention
> > 5655:
> > <extension name="skype"  break="on-false">
> > <condition field="destination_number" expression="^5655$">
> >         <action application="answer"/>
> >         <action application="sleep" data="1000"/>
> > <action application="playback" data="ivr/ivr-sample_submenu.wav"/>
> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
> > application="playback" data="ivr/ivr-sample_submenu.wav"/>
> >    </condition>
> > </extension>
> > ivr-sample_submenu.wav is a sound file in en/us/callie/ivr/8000, but it
> can
> > be replaced by any other wav files.
> > after setup, make a call to your skype user xxxxxxxxx.
> > in the mean time use fs_cli to connect FS and execute /event plain
> MESSAGE
> > to listen esl events.
> > sent any chat messages from caller, then you are very likely to find only
> > the short version events.
> >
> >
> > On Thu, Dec 2, 2010 at 8:50 AM, Giovanni Maruzzelli <gmaruzz at celliax.org
> >
> > wrote:
> >>
> >> Xuyan,
> >>
> >> please.
> >>
> >> Can you give a clear defined procedure to reproduce the problem?
> >>
> >> Please, take the time to write it, complete of all relevant info
> >> (dialplan, script, etc etc) so I can reproduce it from a freshly
> >> installed FreeSWITCH.
> >>
> >> If you don't give this information I cannot fix the problem. And I
> >> cannot neither read your mind, nor finding the time to try every
> >> possible combination.
> >>
> >> Please, post something someone can cut and paste and a clear step by
> >> step procedure to replicate.
> >>
> >> -giovanni
> >>
> >> On Thu, Dec 2, 2010 at 4:45 AM, xuyan yang <xyangni at gmail.com> wrote:
> >> > I tried it again. When the line is parked and idle. the raw events can
> >> > always be generated.
> >> > but if the line is kept busy such as the following case this events is
> >> > trend
> >> > to have problem
> >> > <extension name="skype-in"  break="on-false">
> >> > <condition field="destination_number" expression="^5655$">
> >> >         <action application="answer"/>
> >> >         <action application="sleep" data="1000"/>
> >> > <!--<action application="park"/> -->
> >> > <action application="playback" data="test.wav"/>
> >> > <action application="playback" data="test.wav"/>
> >> > <action application="playback" data="test.wav"/>
> >> >                                 <action application="playback"
> >> > data="test.wav"/>
> >> > <action application="playback" data="test.wav"/>
> >> > <action application="playback" data="test.wav"/>
> >> >                                 <action application="playback"
> >> > data="test.wav"/>
> >> > <action application="playback" data="test.wav"/>
> >> > <action application="playback" data="test.wav"/>
> >> >    </condition>
> >> > </extension>
> >> >
> >> >
> >> > On Thu, Dec 2, 2010 at 2:01 AM, Giovanni Maruzzelli
> >> > <gmaruzz at celliax.org>
> >> > wrote:
> >> >>
> >> >> I repeat, if you subscribe to "message" events you get only those.
> >> >> Also, how I can replicate your problem?
> >> >> I've tested events and I had no problem at all with spurious or
> >> >> unreliable events in mod-skypopen.
> >> >> Please, can you indicate a detailed way to reproduce your problem?
> >> >> -giovanni
> >> >>
> >> >> On 12/2/10, xuyan yang <xyangni at gmail.com> wrote:
> >> >> > I got it. Then the problem should be the unstable behavior of raw
> >> >> > events
> >> >> > which has about 25% chances of being missed.
> >> >> >
> >> >> > Fortunately, I have found a way to avoid this issue. Ignore all
> >> >> > message
> >> >> > events which contains Unique-ID field.
> >> >> >
> >> >> > On Wed, Dec 1, 2010 at 11:18 PM, Giovanni Maruzzelli
> >> >> > <gmaruzz at celliax.org>wrote:
> >> >> >
> >> >> >> On Thu, Dec 2, 2010 at 12:15 AM, Giovanni Maruzzelli
> >> >> >> <gmaruzz at celliax.org> wrote:
> >> >> >> > If you subscribe only to "MESSAGE" kind of events, you'll get
> only
> >> >> >> > those.
> >> >> >> >
> >> >> >> > The other events are "raw" events, that other users have
> requested
> >> >> >> > for
> >> >> >> > other purposes.
> >> >> >>
> >> >> >> or at  least that is the expected behavior, please let me know if
> I
> >> >> >> introduced some regression in integrating that "raw event" thingy.
> >> >> >>
> >> >> >> -giovanni
> >> >> >>
> >> >> >> >
> >> >> >> > -giovanni
> >> >> >> >
> >> >> >> > On Wed, Dec 1, 2010 at 11:35 PM, xuyan yang <xyangni at gmail.com>
> >> >> >> > wrote:
> >> >> >> >> Thanks for your reply. I have read this page before. but the
> >> >> >> >> random
> >> >> >> emerging
> >> >> >> >> of the second verbose esl events is causing trouble. And I
> guess
> >> >> >> >> it
> >> >> >> >> may
> >> >> >> be a
> >> >> >> >> bug, or it should be predictable.
> >> >> >> >> Before reporting to jira, I just want to check whether I have
> >> >> >> >> made
> >> >> >> >> any
> >> >> >> >> mistake.
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> On Wed, Dec 1, 2010 at 9:35 PM, Giovanni Maruzzelli
> >> >> >> >> <gmaruzz at gmail.com>
> >> >> >> >> wrote:
> >> >> >> >>>
> >> >> >> >>>
> >> >> >> >>>
> >> >> >>
> >> >> >>
> >> >> >>
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#MESSAGE_.28Chat.29
> >> >> >> >>>
> >> >> >> >>> On Wed, Dec 1, 2010 at 10:23 PM, xuyan yang <
> xyangni at gmail.com>
> >> >> >> >>> wrote:
> >> >> >> >>> > hi,
> >> >> >> >>> > I am writing ESL program on FreeSWITCH Version 1.0.head
> >> >> >> >>> > (git-8825b6e
> >> >> >> >>> > 2010-11-28 17-15-39 -0500)
> >> >> >> >>> > I need to handle skype chat message with a inbound ESL
> >> >> >> >>> > connection.
> >> >> >> But
> >> >> >> >>> > there
> >> >> >> >>> > are 1 or 2 esl events received randomly for each message.
> >> >> >> >>> > the first short one is alway generated, but the second one
> is
> >> >> >> >>> > random.
> >> >> >> So
> >> >> >> >>> > when trying to filter according to UUID, nothing is caught
> in
> >> >> >> >>> > many
> >> >> >> >>> > cases.
> >> >> >> >>> > If I make filter based on skype ID, duplicated messages are
> >> >> >> >>> > often
> >> >> >> heard.
> >> >> >> >>> > I do not why the behavior of the second event is random.
> >> >> >> >>> > How should I setup the filter to get 1 and only 1 event for
> >> >> >> >>> > each
> >> >> >> >>> > chat
> >> >> >> >>> > message? Thanks.
> >> >> >> >>> >
> >> >> >> >>> > the first is a short one with the following header:
> >> >> >> >>> >  [Event-Name] = [MESSAGE]
> >> >> >> >>> >  [Event-Calling-Function] = [incoming_chatmessage]
> >> >> >> >>> >  [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
> >> >> >> >>> >  [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
> >> >> >> >>> >  [Content-Length] = [1]
> >> >> >> >>> >  [subject] = [SIMPLE MESSAGE]
> >> >> >> >>> >  [FreeSWITCH-IPv4] = [192.168.0.3]
> >> >> >> >>> >  [hint] = [niqizhi]
> >> >> >> >>> >  [from] = [niqizhi]
> >> >> >> >>> >  [Event-Date-Local] = [2010-12-01 21:02:37]
> >> >> >> >>> >  [proto] = [skype]
> >> >> >> >>> >  [FreeSWITCH-IPv6] = [::1]
> >> >> >> >>> >  [id] = [5334]
> >> >> >> >>> >  [Event-Calling-File] = [mod_skypopen.c]
> >> >> >> >>> >  [Event-Date-Timestamp] = [1291237357051788]
> >> >> >> >>> >  [FreeSWITCH-Hostname] = [EYSRV]
> >> >> >> >>> >  [login] = [interface1]
> >> >> >> >>> >  [during-call] = [true]
> >> >> >> >>> >  [Event-Calling-Line-Number] = [2915]
> >> >> >> >>> >  [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
> >> >> >> >>> > the second is much longer:
> >> >> >> >>> >  [Caller-Source] = [mod_skypopen]
> >> >> >> >>> >  [Event-Calling-Function] = [incoming_chatmessage]
> >> >> >> >>> >  [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
> >> >> >> >>> >  [Answer-State] = [answered]
> >> >> >> >>> >  [FreeSWITCH-IPv4] = [192.168.0.3]
> >> >> >> >>> >  [Channel-State] = [CS_EXECUTE]
> >> >> >> >>> >  [Channel-Read-Codec-Bit-Rate] = [256000]
> >> >> >> >>> >  [FreeSWITCH-IPv6] = [::1]
> >> >> >> >>> >  [Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
> >> >> >> >>> >  [Channel-Read-Codec-Rate] = [16000]
> >> >> >> >>> >  [Caller-Destination-Number] = [5655]
> >> >> >> >>> >  [Caller-Channel-Transfer-Time] = [0]
> >> >> >> >>> >  [Channel-Call-State] = [ACTIVE]
> >> >> >> >>> >  [Caller-Channel-Progress-Media-Time] = [0]
> >> >> >> >>> >  [FreeSWITCH-Hostname] = [EYSRV]
> >> >> >> >>> >  [Caller-Channel-Answered-Time] = [1291237326697085]
> >> >> >> >>> >  [login] = [interface1]
> >> >> >> >>> >  [during-call] = [true]
> >> >> >> >>> >  [Channel-Name] = [skypopen/interface1]
> >> >> >> >>> >  [Caller-Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
> >> >> >> >>> >  [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
> >> >> >> >>> >  [Channel-Read-Codec-Name] = [L16]
> >> >> >> >>> >  [Caller-Channel-Name] = [skypopen/interface1]
> >> >> >> >>> >  [Caller-Caller-ID-Number] = [niqizhi]
> >> >> >> >>> >  [Event-Date-Timestamp] = [1291237357051788]
> >> >> >> >>> >  [Channel-State-Number] = [4]
> >> >> >> >>> >  [Event-Calling-Line-Number] = [2888]
> >> >> >> >>> >  [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
> >> >> >> >>> >  [Event-Name] = [MESSAGE]
> >> >> >> >>> >  [Content-Length] = [1]
> >> >> >> >>> >  [subject] = [SIMPLE MESSAGE]
> >> >> >> >>> >  [Caller-Caller-ID-Name] = [niqizhi]
> >> >> >> >>> >  [from] = [niqizhi]
> >> >> >> >>> >  [Caller-Dialplan] = [XML]
> >> >> >> >>> >  [Caller-Channel-Hangup-Time] = [0]
> >> >> >> >>> >  [id] = [5334]
> >> >> >> >>> >  [Caller-Profile-Index] = [1]
> >> >> >> >>> >  [Caller-Direction] = [inbound]
> >> >> >> >>> >  [Caller-Username] = [skypopen]
> >> >> >> >>> >  [Channel-Write-Codec-Name] = [L16]
> >> >> >> >>> >  [Call-Direction] = [inbound]
> >> >> >> >>> >  [Caller-Screen-Bit] = [true]
> >> >> >> >>> >  [hint] = [niqizhi]
> >> >> >> >>> >  [Caller-Privacy-Hide-Number] = [false]
> >> >> >> >>> >  [Event-Date-Local] = [2010-12-01 21:02:37]
> >> >> >> >>> >  [proto] = [skype]
> >> >> >> >>> >  [Caller-Channel-Created-Time] = [1291237326468855]
> >> >> >> >>> >  [Event-Calling-File] = [mod_skypopen.c]
> >> >> >> >>> >  [Caller-Channel-Progress-Time] = [0]
> >> >> >> >>> >  [Caller-Privacy-Hide-Name] = [false]
> >> >> >> >>> >  [Channel-Write-Codec-Rate] = [16000]
> >> >> >> >>> >  [Caller-Context] = [default]
> >> >> >> >>> >  [Channel-Write-Codec-Bit-Rate] = [256000]
> >> >> >> >>> >  [Presence-Call-Direction] = [inbound]
> >> >> >> >>> >  [Caller-Profile-Created-Time] = [1291237326468855]
> >> >> >> >>> > _______________________________________________
> >> >> >> >>> > FreeSWITCH-users mailing list
> >> >> >> >>> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >>> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>> > UNSUBSCRIBE:
> >> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >>> > http://www.freeswitch.org
> >> >> >> >>> >
> >> >> >> >>> >
> >> >> >> >>>
> >> >> >> >>>
> >> >> >> >>>
> >> >> >> >>> --
> >> >> >> >>> Sincerely,
> >> >> >> >>>
> >> >> >> >>> Giovanni Maruzzelli
> >> >> >> >>> Cell : +39-347-2665618
> >> >> >> >>>
> >> >> >> >>> _______________________________________________
> >> >> >> >>> FreeSWITCH-users mailing list
> >> >> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>> UNSUBSCRIBE:
> >> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >>> http://www.freeswitch.org
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> _______________________________________________
> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >> UNSUBSCRIBE:
> >> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> http://www.freeswitch.org
> >> >> >> >>
> >> >> >> >>
> >> >> >> >
> >> >> >> >
> >> >> >> >
> >> >> >> > --
> >> >> >> > Sincerely,
> >> >> >> >
> >> >> >> > Giovanni Maruzzelli
> >> >> >> > Cell : +39-347-2665618
> >> >> >> >
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >> --
> >> >> >> Sincerely,
> >> >> >>
> >> >> >> Giovanni Maruzzelli
> >> >> >> Cell : +39-347-2665618
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >>
> >> >> >
> >> >>
> >> >> --
> >> >> Sent from my mobile device
> >> >>
> >> >> Sincerely,
> >> >>
> >> >> Giovanni Maruzzelli
> >> >> Cell : +39-347-2665618
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Thu Dec  2 09:05:42 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 2 Dec 2010 12:05:42 -0500
Subject: [Freeswitch-users] Custom Channel Variables via C#
In-Reply-To: <SNT135-w4BF48F8FEB9054D804759B7270@phx.gbl>
References: <SNT135-w336CAB94DFC78D328AE6E0B7260@phx.gbl>,
	<0E462B4C-C436-462E-9DA7-9EC1241FB380@avgs.ca>
	<SNT135-w4BF48F8FEB9054D804759B7270@phx.gbl>
Message-ID: <9C89CC76-01FC-4F8D-B00B-89EE3A74EB05@avgs.ca>

You need to authenticate first.

Trying 127.0.0.1...
Connected to localhost
Escape character is '^]'.
Content-Type: auth/request

auth ClueCon

Content-Type: command/reply
Reply-Text: +OK accepted

api show channels

Content-Type: api/response
Content-Length: 748

uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid
aa6e6bee-1c3b-4493-b824-765c5a15f02c,outbound,2010-12-02 12:03:58,1291309438,loopback/park-a,CS_CONSUME_MEDIA,,0000000000,,park,,,inline,inline,L16,8000,128000,L16,8000,128000,,mrene.local,,,RINGING,,,,c9781e01-667e-446b-93e8-a081f0287d15
78ff4bb6-573e-4a60-a920-bfdb67f3ed6b,inbound,2010-12-02 12:03:58,1291309438,loopback/park-b,CS_EXECUTE,,0000000000,,park,park,,inline,inline,L16,8000,128000,L16,8000,128000,,mrene.local,,,RINGING,,,,

2 total.

Then you can use the uuid_* commands.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-12-02, at 10:03 AM, Brian Campbell wrote:

> Thanks, the link was very helpful
>  
> I have taken a look at the link that you provided
>  
> I assume now that I can use uuid_setvar to accomplish what I want
>  
> Assuming that the uuid is 53d37581-1f90-44bf-860a-addbc8430e3a, it seems that my code would need to change to
>  
>  
> private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
> {
>   try
>   {
>     TcpClient newClient = new TcpClient();
>  
>     newClient.Connect("127.0.0.1", 8021);
>  
>     NetworkStream tcpStream = newClient.GetStream();
>  
>     byte[] sendBytes = Encoding.ASCII.GetBytes("api uuid_setvar 53d37581-1f90-44bf-860a-addbc8430e3a myChannelVariable 12345");
>  
>     tcpStream.Write(sendBytes, 0, sendBytes.Length);
>  
>     tcpStream.Close();
>  
>     newClient.Close();
>   }
>   catch(Exception ex)
>   {
>     throw ex;
>   }
> }
> 
> My question now becomes, how do I go about finding out the uuid for the session ? Is that obtainable via mod_event_socket or is there another way ?
>  
> Thanks
>  
> Brian 
> From: mrene_lists at avgs.ca
> Date: Wed, 1 Dec 2010 16:53:53 -0500
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Custom Channel Variables via C#
> 
> Hi,
> 
> You can't send XML dialplan actions on the socket and expect FreeSWITCH to understand it, you must follow the event socket protocol.
> 
> See http://wiki.freeswitch.org/wiki/Event_socket for more information.
> 
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
> 
> 
> 
> 
> On 2010-12-01, at 4:46 PM, Brian Campbell wrote:
> 
>  
> I can set a custom channel variable in my incomming dial plan like this...
>  
> <action application='set' data='MyChannelVariable=12345' />
>  
> And it shows up fine in the CDR
>  
> I am now attempting to use mod_event_socket to set a custom channel variable and have it appear in the CDR
>  
> Here is the C# code so far, the code is part of a Microsoft Speech Server application that is answering the call
>  
> private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
> {
>   try
>   {
>     TcpClient newClient = new TcpClient();
>  
>     newClient.Connect("127.0.0.1", 8021);
>  
>     NetworkStream tcpStream = newClient.GetStream();
>  
>     byte[] sendBytes = Encoding.ASCII.GetBytes("<action application='set' data='MyChannelVariable=12345' />");
>  
>     tcpStream.Write(sendBytes, 0, sendBytes.Length);
>  
>     tcpStream.Close();
>  
>     newClient.Close();
>   }
>   catch(Exception ex)
>   {
>     throw ex;
>   }
> }
>  
> After the call is answered, the C# code seems to run fine, but I dont see the custom channel variable in the resulting CDR
>  
> I figure I am not setting it correctly
>  
> Can anyone advise on what I am doing wrong ?
>  
> Thanks
>  
>  
> Brian
>  
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Thu Dec  2 09:12:35 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 2 Dec 2010 12:12:35 -0500
Subject: [Freeswitch-users] How to force the usage of g729 instead of
	g729b
References: <4CF7C07E.9080707@sns.eu>
Message-ID: <47E99D5C67F84F49BAE67CCDAD2D2230@e1705>

Maybe
absolute_codec_string will do the trick

----- Original Message ----- 
From: "Jan Riedinger" <riedinger at sns.eu>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, December 02, 2010 10:51 AM
Subject: [Freeswitch-users] How to force the usage of g729 instead of g729b


By setting global_codec_prefs and inbound_codec_prefs it's possible to
specify, which codecs are supported for inbound calls. How can I specify
that the switch shall support g729, but not g729b?

The incoming INVITE message contains a SDP with the media attribute
annexb=yes. How has Freeswitch to answer to indicate that g729 is
supported but g729b is not supported?

Thank you in advance
         Jan

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973


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



From bcxml at hotmail.com  Thu Dec  2 09:44:09 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Thu, 2 Dec 2010 12:44:09 -0500
Subject: [Freeswitch-users] Custom Channel Variables via C#
In-Reply-To: <9C89CC76-01FC-4F8D-B00B-89EE3A74EB05@avgs.ca>
References: <SNT135-w336CAB94DFC78D328AE6E0B7260@phx.gbl>, ,
	<0E462B4C-C436-462E-9DA7-9EC1241FB380@avgs.ca>,
	<SNT135-w4BF48F8FEB9054D804759B7270@phx.gbl>,
	<9C89CC76-01FC-4F8D-B00B-89EE3A74EB05@avgs.ca>
Message-ID: <SNT135-w401BFA14D3758BCF6AADF1B7270@phx.gbl>


Thanks Mathieu
 
Ill give it a try
 
 
Brian
 


From: mrene_lists at avgs.ca
Date: Thu, 2 Dec 2010 12:05:42 -0500
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Custom Channel Variables via C#


You need to authenticate first.



Trying 127.0.0.1...
Connected to localhost
Escape character is '^]'.
Content-Type: auth/request


auth ClueCon


Content-Type: command/reply
Reply-Text: +OK accepted


api show channels


Content-Type: api/response
Content-Length: 748


uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid
aa6e6bee-1c3b-4493-b824-765c5a15f02c,outbound,2010-12-02 12:03:58,1291309438,loopback/park-a,CS_CONSUME_MEDIA,,0000000000,,park,,,inline,inline,L16,8000,128000,L16,8000,128000,,mrene.local,,,RINGING,,,,c9781e01-667e-446b-93e8-a081f0287d15
78ff4bb6-573e-4a60-a920-bfdb67f3ed6b,inbound,2010-12-02 12:03:58,1291309438,loopback/park-b,CS_EXECUTE,,0000000000,,park,park,,inline,inline,L16,8000,128000,L16,8000,128000,,mrene.local,,,RINGING,,,,


2 total.


Then you can use the uuid_* commands.





Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca






On 2010-12-02, at 10:03 AM, Brian Campbell wrote:

Thanks, the link was very helpful
 
I have taken a look at the link that you provided
 
I assume now that I can use uuid_setvar to accomplish what I want
 
Assuming that the uuid is 53d37581-1f90-44bf-860a-addbc8430e3a, it seems that my code would need to change to
 
 
private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
{
  try
  {
    TcpClient newClient = new TcpClient();
 
    newClient.Connect("127.0.0.1", 8021);
 
    NetworkStream tcpStream = newClient.GetStream();
 
    byte[] sendBytes = Encoding.ASCII.GetBytes("api uuid_setvar 53d37581-1f90-44bf-860a-addbc8430e3a myChannelVariable 12345");
 
    tcpStream.Write(sendBytes, 0, sendBytes.Length);
 
    tcpStream.Close();
 
    newClient.Close();
  }
  catch(Exception ex)
  {
    throw ex;
  }
}

My question now becomes, how do I go about finding out the uuid for the session ? Is that obtainable via mod_event_socket or is there another way ?
 
Thanks
 
Brian 


From: mrene_lists at avgs.ca
Date: Wed, 1 Dec 2010 16:53:53 -0500
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Custom Channel Variables via C#


Hi,


You can't send XML dialplan actions on the socket and expect FreeSWITCH to understand it, you must follow the event socket protocol.


See http://wiki.freeswitch.org/wiki/Event_socket for more information.




Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca






On 2010-12-01, at 4:46 PM, Brian Campbell wrote:

 
I can set a custom channel variable in my incomming dial plan like this...
 
<action application='set' data='MyChannelVariable=12345' />
 
And it shows up fine in the CDR
 
I am now attempting to use mod_event_socket to set a custom channel variable and have it appear in the CDR
 
Here is the C# code so far, the code is part of a Microsoft Speech Server application that is answering the call
 
private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
{
  try
  {
    TcpClient newClient = new TcpClient();
 
    newClient.Connect("127.0.0.1", 8021);
 
    NetworkStream tcpStream = newClient.GetStream();
 
    byte[] sendBytes = Encoding.ASCII.GetBytes("<action application='set' data='MyChannelVariable=12345' />");
 
    tcpStream.Write(sendBytes, 0, sendBytes.Length);
 
    tcpStream.Close();
 
    newClient.Close();
  }
  catch(Exception ex)
  {
    throw ex;
  }
}
 
After the call is answered, the C# code seems to run fine, but I dont see the custom channel variable in the resulting CDR
 
I figure I am not setting it correctly
 
Can anyone advise on what I am doing wrong ?
 
Thanks
 
 
Brian
 
 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


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


_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 		 	   		  
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From riedinger at sns.eu  Thu Dec  2 09:48:02 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Thu, 02 Dec 2010 18:48:02 +0100
Subject: [Freeswitch-users] How to force the usage of g729 instead of
 g729b
In-Reply-To: <47E99D5C67F84F49BAE67CCDAD2D2230@e1705>
References: <4CF7C07E.9080707@sns.eu> <47E99D5C67F84F49BAE67CCDAD2D2230@e1705>
Message-ID: <4CF7DBD2.6030404@sns.eu>

As far as I know the codec strings for g729 and g729b are the same, 
despite these codecs are  not compatible according RFC 3555 (at least 
Cisco is reading the RFC in that way, 
https://supportforums.cisco.com/docs/DOC-3188).


Am 02.12.2010 18:12, schrieb Madovsky:
> Maybe
> absolute_codec_string will do the trick
>
> ----- Original Message -----
> From: "Jan Riedinger"<riedinger at sns.eu>
> To: "FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, December 02, 2010 10:51 AM
> Subject: [Freeswitch-users] How to force the usage of g729 instead of g729b
>
>
> By setting global_codec_prefs and inbound_codec_prefs it's possible to
> specify, which codecs are supported for inbound calls. How can I specify
> that the switch shall support g729, but not g729b?
>
> The incoming INVITE message contains a SDP with the media attribute
> annexb=yes. How has Freeswitch to answer to indicate that g729 is
> supported but g729b is not supported?
>
> Thank you in advance
>           Jan
>

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973



From david.ponzone at ipeva.fr  Thu Dec  2 09:56:44 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 2 Dec 2010 18:56:44 +0100
Subject: [Freeswitch-users] How to force the usage of g729 instead of
	g729b
In-Reply-To: <4CF7DBD2.6030404@sns.eu>
References: <4CF7C07E.9080707@sns.eu> <47E99D5C67F84F49BAE67CCDAD2D2230@e1705>
	<4CF7DBD2.6030404@sns.eu>
Message-ID: <EBB773FA-E006-4015-9CC1-9C9BCA85FC3B@ipeva.fr>

Jan,

For outbound, I do it this way:

    <extension name="disable-annexB" continue="true">
      <condition field="${switch_r_sdp}" expression="/(.*)(m=audio \d+ RTP\/AVP)(.*)( 18 )(.*)/s">
        <action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/>
      </condition>
    </extension>

For inbound, it's going to be tricky, as I am not sure if there is a way to patch the SDP that FS will send back...

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 02/12/2010 ? 18:48, Jan Riedinger a ?crit :

> As far as I know the codec strings for g729 and g729b are the same, 
> despite these codecs are  not compatible according RFC 3555 (at least 
> Cisco is reading the RFC in that way, 
> https://supportforums.cisco.com/docs/DOC-3188).
> 
> 
> Am 02.12.2010 18:12, schrieb Madovsky:
>> Maybe
>> absolute_codec_string will do the trick
>> 
>> ----- Original Message -----
>> From: "Jan Riedinger"<riedinger at sns.eu>
>> To: "FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>
>> Sent: Thursday, December 02, 2010 10:51 AM
>> Subject: [Freeswitch-users] How to force the usage of g729 instead of g729b
>> 
>> 
>> By setting global_codec_prefs and inbound_codec_prefs it's possible to
>> specify, which codecs are supported for inbound calls. How can I specify
>> that the switch shall support g729, but not g729b?
>> 
>> The incoming INVITE message contains a SDP with the media attribute
>> annexb=yes. How has Freeswitch to answer to indicate that g729 is
>> supported but g729b is not supported?
>> 
>> Thank you in advance
>>          Jan
>> 
> 
> -- 
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
> 
> AG Charlottenburg - HRB 71973
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From chad at apartmentlines.com  Thu Dec  2 11:23:42 2010
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Thu, 2 Dec 2010 11:23:42 -0800
Subject: [Freeswitch-users] can't get channel variables after hangup
In-Reply-To: <AANLkTim3Y+FYRqmy+wyyZKoeOv5keAGoNEe6fPKUTtcK@mail.gmail.com>
References: <AE2E563CA6B0424FA874C7E68CC77C2A@e1705>
	<AANLkTim3Y+FYRqmy+wyyZKoeOv5keAGoNEe6fPKUTtcK@mail.gmail.com>
Message-ID: <75D2E81B-DCB0-49EA-90E0-21AAB9FF632C@apartmentlines.com>

on a related note, i wanted to point out that if you're already inside a running script and need to perform actions when the call is hung up, you can do it right inside the script, no need to set a hangup hook.  in this case, when the call hangs up, session:ready() (in Lua's case) returns false, the entire session object is still available (including channel variables), and you can perform hangup tasks and end the script when you're finished.

the fact that FreeSWITCH allows your script to continue running after hangup is quite handy, but it also means that you should be thorough in your use of session:ready() to ensure the call is still up when you're taking actions that depend on it.  this is especially important when using loops in your script -- very unpleasant things can happen if you don't include session:ready() in your loop conditional!

hope this further clarification helps.

chad

On Dec 1, 2010, at 12:48 PM, Michael Collins wrote:

> This won't work. The "session_in_hangup_hook" literally means that the $session object is available in the hangup hook script. You need to use Lua, Perl, or Javascript to access the $session object. What you are doing is actually trying to access a uuid that does not exist. Once the call is over it's gone. The "session" object will be available only in the script that is called in the hangup hook. 



From infos at madovsky.org  Thu Dec  2 11:37:33 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 2 Dec 2010 14:37:33 -0500
Subject: [Freeswitch-users] mod_perl comple problem
Message-ID: <8AD291679292407D9D9B3BF8ED591D8B@e1705>


have this problem at complie :

/usr/bin/ld: cannot find -ldb

fedora10 64bits.

thanks

Franck
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From dyatsin at sangoma.com  Thu Dec  2 11:46:12 2010
From: dyatsin at sangoma.com (David Yat Sin)
Date: Thu, 02 Dec 2010 14:46:12 -0500
Subject: [Freeswitch-users] Problem with SIP 180 + FreeTDM
In-Reply-To: <AANLkTi=24KMXRz1FMi7nLTPkjFj2nzqFycD8ZPZ2b+g9@mail.gmail.com>
References: <AANLkTi=3RuKF9YzGmvz=Xcga4zdH6_eOod7H0XMd=Kyn@mail.gmail.com>	<4CE5A68B.2060507@sangoma.com>
	<AANLkTi=24KMXRz1FMi7nLTPkjFj2nzqFycD8ZPZ2b+g9@mail.gmail.com>
Message-ID: <4CF7F784.7090201@sangoma.com>

Hi Juan Antonio,

Can you pull the latest Freeswitch git. This should be fixed now.

David

*David Yat Sin, BEng*
Software Engineer
Sangoma Technologies
100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada 	

t.   +1 800 388 2475 x119
t.   +1 905 474 1990 x119
f.   +1 905 474 9223 	SANGOMA 
<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 

Products 
<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Solutions 
<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Events 
<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Contact 
<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Wiki 
<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Facebook 
<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Twitter 
<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 



On 11/24/2010 1:38 PM, Juan Antonio Iba?ez Santorum wrote:
> Any news?
>
> 2010/11/18 David Yat Sin <dyatsin at sangoma.com 
> <mailto:dyatsin at sangoma.com>>
>
>     Hi Juan Antonio,
>     We are looking into this and will get back to you in a day or two.
>
>     David
>
>     *David Yat Sin, BEng*
>     Software Engineer
>     Sangoma Technologies
>     100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada 	
>
>     t.   +1 800 388 2475 x119
>     t.   +1 905 474 1990 x119
>     f.   +1 905 474 9223 	SANGOMA
>     <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>
>     Products
>     <http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>     | Solutions
>     <http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>     | Events
>     <http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>     | Contact
>     <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>     | Wiki
>     <http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>     | Facebook
>     <http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>     | Twitter
>     <http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>
>
>
>     On 11/18/2010 3:50 PM, Juan Antonio Iba?ez Santorum wrote:
>>     Hello boys!
>>
>>        I have one problem after updating from openzap to freetdm.
>>     Now, whe FS receives 'Proceeding' from PRI sends a SIP 180 which
>>     makes the phone to play a fake ring tone. After some seconds,
>>     when FS receives 'Alerting' from PRI, FS sends a SIP 183 with the
>>     real ring tone.
>>
>>     Is there any way to avoid first SIP 180? If the calle is busy, I
>>     can hear a ringing tone between SIP 180 and SIP 183 and a busy
>>     tone after SIP 183. Same if calle is a mobile phone and it is out
>>     of signal.
>>
>>     Regards
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From kilburna at gmail.com  Thu Dec  2 12:59:21 2010
From: kilburna at gmail.com (Kilburn Abrahams)
Date: Fri, 03 Dec 2010 07:59:21 +1100
Subject: [Freeswitch-users] No CDR
Message-ID: <4CF808A9.90302@gmail.com>

Hi List

Asterisk has useful feature called NoCDR. Prefer not to record CDR's for
internal dialing. I am using mod_xml_cdr. Is there an equivalent feature
or a work around to achieve this

Thanks
Kilburn
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From david.ponzone at ipeva.fr  Thu Dec  2 13:17:01 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 2 Dec 2010 22:17:01 +0100
Subject: [Freeswitch-users] Export a channel variable within Dial String
In-Reply-To: <1ff0ebed-7410-4bdb-ab4c-619cadd564ed@winet.ch>
References: <481d3184-6518-4957-acca-b1e73c7fec7e@winet.ch>
	<B724748E-DF66-4CD4-B987-BED15F266C78@ipeva.fr>
	<1ff0ebed-7410-4bdb-ab4c-619cadd564ed@winet.ch>
Message-ID: <C640E45F-505B-4700-9074-EDDBED4CA50B@ipeva.fr>

Bernhard,

I am not sure I follow you.

AFAIK, exporting and setting with {} in the dialstring is basically the same.
There is a difference, but it's a very minor technical one.
Also, if you export, the variable is set on both legs, while {} will only set it on leg B.

So you say you dont want to export on a specific leg B, but in your dialstring, you dont export, you just set a variable on the leg B.
Nothing will ever be exported from leg B.
Variables can be exported from leg A to leg B.

I could be wrong, but that has been my understanding for the last 12 months.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 02/12/2010 ? 17:31, Bernhard Suttner a ?crit :

> The channel variable will be set, but not exported like the ?export? dialplan function does export the variable to the new channel.
>  
>  
>  
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von David Ponzone
> Gesendet: Donnerstag, 2. Dezember 2010 16:14
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] Export a channel variable within Dial String
>  
> Bernhard,
>  
> AFAIU, if you specify the variable in the dialstring, it's for leg B, so it's exported.
> If you wan't it, don't set it :)
>  
> And for leg A, well, you only have one leg A, so you can't set it twice.
>  
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>  
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>  
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>  
> 
> 
>  
> Le 02/12/2010 ? 15:48, Bernhard Suttner a ?crit :
> 
> 
> Hi,
> 
> I call multiple destination with bridge like that:
> 
> [var1=value]sofia/default/user1,[var1=another_value]loopback/user2
> 
> Is it somehow possible, that var1=another_value will be exported but var1=value not? What I want is, that I could specify a variable to be exported _within_ the dialstring?
> 
> Thanks for every hint.
> 
> Best regards,
> Bernhard
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tayeb.meftah at gmail.com  Thu Dec  2 13:15:36 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Thu, 02 Dec 2010 22:15:36 +0100
Subject: [Freeswitch-users] No CDR
In-Reply-To: <4CF808A9.90302@gmail.com>
References: <4CF808A9.90302@gmail.com>
Message-ID: <4CF80C78.6000907@gmail.com>

<action application="set" data="process_cdr=false"/>

Le 02/12/2010 21:59, Kilburn Abrahams a ?crit :
> Hi List
>
> Asterisk has useful feature called NoCDR. Prefer not to record CDR's 
> for internal dialing. I am using mod_xml_cdr. Is there an equivalent 
> feature or a work around to achieve this
>
> Thanks
> Kilburn
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313

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From infos at madovsky.org  Thu Dec  2 13:47:48 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 2 Dec 2010 16:47:48 -0500
Subject: [Freeswitch-users] mod_perl comple problem
Message-ID: <A931A81C848A459199F12E3EA988FE63@e1705>

I solved it by installing 

yum install db4-devel

thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, December 02, 2010 2:37 PM
  Subject: mod_perl comple problem



  have this problem at complie :

  /usr/bin/ld: cannot find -ldb

  fedora10 64bits.

  thanks

  Franck
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From moises.silva at gmail.com  Thu Dec  2 13:48:54 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 2 Dec 2010 16:48:54 -0500
Subject: [Freeswitch-users] "No Digits to send" on purpose?
In-Reply-To: <D5D7DBD69EB94C94B423774D936A8B15@mbnet.local>
References: <6EFE58A23D6B4E65A16462DCDDE5AD1E@mbnet.local>
	<AANLkTinE1YAOq_k7PG6fTEvpioO3zo+=mMscXcGdSsQp@mail.gmail.com>
	<782B83D12FAE4495A35C0DEED2EB6A84@mbnet.local>
	<FB8961D5-AA00-4C75-87E0-7049C2F332A8@freeswitch.org>
	<D5D7DBD69EB94C94B423774D936A8B15@mbnet.local>
Message-ID: <AANLkTinb5OgVXoYC7+CcvbTCwvofK1dNPddpjatUT5VR@mail.gmail.com>

On Wed, Dec 1, 2010 at 8:22 PM, Jeroen C. van Gelderen <slim at thegreek.com>wrote:

>  Hi Brian,
>
>
>
> Thanks for your feedback.
>
>
>
> I?ve attempted to create a decent bug:
>
>  http://jira.freeswitch.org/browse/OPENZAP-120
>
>
>

Thanks for the report.

This commit should fix it, although I did not test. Please re-open if the
issue persists.

commit e0048ed24e7eee9cbfd07b7a9afc1d97941cd270
Author: Moises Silva <moy at sangoma.com>
Date:   Thu Dec 2 16:43:27 2010 -0500

    freetdm: OPENZAP-120 - Allow FXO to bridge calls without digits

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com
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From moises.silva at gmail.com  Thu Dec  2 13:53:11 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 2 Dec 2010 16:53:11 -0500
Subject: [Freeswitch-users] FreeSWITCH, FreeTDM, Windows
In-Reply-To: <AANLkTi=CtRJVa2Ff+HzbB-Xe+qannZhxdRFjuaRrh4dB@mail.gmail.com>
References: <AANLkTik-LE4UhqnJ2p3OxCJ5kxa+UToXpcWgGZ9qjRYH@mail.gmail.com>
	<AANLkTi=CtRJVa2Ff+HzbB-Xe+qannZhxdRFjuaRrh4dB@mail.gmail.com>
Message-ID: <AANLkTinpNnxFYTKPn_m8-0_ez7_4UFc=OYA=xrUVs1bd@mail.gmail.com>

On Mon, Nov 29, 2010 at 10:49 AM, Mikhail Zhirnov <michaeal at gmail.com>wrote:

> Hello!
>
> Could you please help me to solve error with FreeSWITCH, FreeTDM on Windows
> XP SP3  machine.
>
> I used wiki http://wiki.sangoma.com/wanpipe-api-freetdm . I installed
> Wanpipe and then compiled FreeSWITCH, FreeTDM by Visual Studio 2010.
> FreeSWITCH is fine but it couldn't load FreeTDM module with error:
>
> *freeswitch at prodan-a> reload freetdm*
> *2010-11-29 17:18:33.437500 [INFO] mod_enum.c:808 ENUM Reloaded*
> *
> *
> *-ERR unloading module [No such module!]*
> *+OK Reloading XML*
> *-ERR loading module [module load file routine returned an error]*
> *
> *
> *2010-11-29 17:18:33.437500 [CRIT] switch_loadable_module.c:928 Error
> Loading module D:\sangoma\FS_GIT\FS_GIT\Win32\Debug\mod\freetdm.dll*
> ***dll sym error [127l]*
> ****
> *2010-11-29 17:18:33.437500 [INFO] switch_time.c:950 Timezone reloaded 530
> definitions*
>
>
Did you solve this? Enable debugging messages and pastebin the output if you
did not solve it already.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com
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From juanito1982 at gmail.com  Thu Dec  2 13:55:36 2010
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Thu, 2 Dec 2010 22:55:36 +0100
Subject: [Freeswitch-users] Problem with SIP 180 + FreeTDM
In-Reply-To: <4CF7F784.7090201@sangoma.com>
References: <AANLkTi=3RuKF9YzGmvz=Xcga4zdH6_eOod7H0XMd=Kyn@mail.gmail.com>
	<4CE5A68B.2060507@sangoma.com>
	<AANLkTi=24KMXRz1FMi7nLTPkjFj2nzqFycD8ZPZ2b+g9@mail.gmail.com>
	<4CF7F784.7090201@sangoma.com>
Message-ID: <AANLkTikBL=rxXgkq68SHOzpwfWd3CNO-if4Q7GpGMuwD@mail.gmail.com>

I will try it. I'll tell you the result.

Regards

2010/12/2 David Yat Sin <dyatsin at sangoma.com>

>  Hi Juan Antonio,
>
> Can you pull the latest Freeswitch git. This should be fixed now.
>
>
> David
>
>   *David Yat Sin, BEng*
> Software Engineer
> Sangoma Technologies
> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
>
> t.   +1 800 388 2475 x119
> t.   +1 905 474 1990 x119
> f.   +1 905 474 9223 [image: SANGOMA]<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
> Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
> Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
> Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
> Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
> Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
> Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>
> On 11/24/2010 1:38 PM, Juan Antonio Iba?ez Santorum wrote:
>
> Any news?
>
> 2010/11/18 David Yat Sin <dyatsin at sangoma.com>
>
>>  Hi Juan Antonio,
>> We are looking into this and will get back to you in a day or two.
>>
>> David
>>
>>   *David Yat Sin, BEng*
>> Software Engineer
>> Sangoma Technologies
>> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
>>
>> t.   +1 800 388 2475 x119
>> t.   +1 905 474 1990 x119
>> f.   +1 905 474 9223 [image: SANGOMA]<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>
>> On 11/18/2010 3:50 PM, Juan Antonio Iba?ez Santorum wrote:
>>
>>  Hello boys!
>>
>>     I have one problem after updating from openzap to freetdm. Now, whe
>> FS receives 'Proceeding' from PRI sends a SIP 180 which makes the phone to
>> play a fake ring tone. After some seconds, when FS receives 'Alerting' from
>> PRI, FS sends a SIP 183 with the real ring tone.
>>
>>  Is there any way to avoid first SIP 180? If the calle is busy, I can
>> hear a ringing tone between SIP 180 and SIP 183 and a busy tone after SIP
>> 183. Same if calle is a mobile phone and it is out of signal.
>>
>>  Regards
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jeff at jefflenk.com  Thu Dec  2 14:24:43 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 2 Dec 2010 14:24:43 -0800 (PST)
Subject: [Freeswitch-users] is mod_python supported on windows?
In-Reply-To: <AANLkTin-MjRKsAGMs_Lk3TvB8gfmNnseE334n9wh4SGi@mail.gmail.com>
References: <AANLkTin-MjRKsAGMs_Lk3TvB8gfmNnseE334n9wh4SGi@mail.gmail.com>
Message-ID: <1291328683976-5797895.post@n2.nabble.com>


At this time this module does not have VS projects for building it.  If you
are so inclined to create the VS projects and wiki documentation you may add
a patch to Jira http://jira.freeswitch.org.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/is-mod-python-supported-on-windows-tp5793156p5797895.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Fri Dec  3 02:52:09 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 2 Dec 2010 17:52:09 -0600
Subject: [Freeswitch-users] TEST
Message-ID: <215DCB83-4BF0-4027-A857-AB13CA714931@freeswitch.org>

Ignore please.

/b



From bernhard.suttner at winet.ch  Fri Dec  3 12:28:19 2010
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Fri, 3 Dec 2010 10:28:19 +0100
Subject: [Freeswitch-users] Export a channel variable within Dial String
In-Reply-To: <C640E45F-505B-4700-9074-EDDBED4CA50B@ipeva.fr>
References: <481d3184-6518-4957-acca-b1e73c7fec7e@winet.ch>
	<B724748E-DF66-4CD4-B987-BED15F266C78@ipeva.fr>
	<1ff0ebed-7410-4bdb-ab4c-619cadd564ed@winet.ch>
	<C640E45F-505B-4700-9074-EDDBED4CA50B@ipeva.fr>
Message-ID: <80107a0f-79c8-4922-ae58-830100142291@winet.ch>

How embrassing?. You are right, of course. Thanks J
 
 
 
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von David Ponzone
Gesendet: Donnerstag, 2. Dezember 2010 22:17
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Export a channel variable within Dial String
 
Bernhard,
 
I am not sure I follow you.
 
AFAIK, exporting and setting with {} in the dialstring is basically the same.
There is a difference, but it's a very minor technical one.
Also, if you export, the variable is set on both legs, while {} will only set it on leg B.
 
So you say you dont want to export on a specific leg B, but in your dialstring, you dont export, you just set a variable on the leg B.
Nothing will ever be exported from leg B.
Variables can be exported from leg A to leg B.
 
I could be wrong, but that has been my understanding for the last 12 months.
 
David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34
 
Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com
 
Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
 



 
Le 02/12/2010 ? 17:31, Bernhard Suttner a ?crit :



The channel variable will be set, but not exported like the ?export? dialplan function does export the variable to the new channel.
 
 
 
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von David Ponzone
Gesendet: Donnerstag, 2. Dezember 2010 16:14
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Export a channel variable within Dial String
 
Bernhard,
 
AFAIU, if you specify the variable in the dialstring, it's for leg B, so it's exported.
If you wan't it, don't set it :)
 
And for leg A, well, you only have one leg A, so you can't set it twice.
 
David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34
 
Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com
 
Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
 




 
Le 02/12/2010 ? 15:48, Bernhard Suttner a ?crit :




Hi,

I call multiple destination with bridge like that:

[var1=value]sofia/default/user1,[var1=another_value]loopback/user2

Is it somehow possible, that var1=another_value will be exported but var1=value not? What I want is, that I could specify a variable to be exported _within_ the dialstring?

Thanks for every hint.

Best regards,
Bernhard



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 
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From david.ponzone at ipeva.fr  Fri Dec  3 12:38:04 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 3 Dec 2010 10:38:04 +0100
Subject: [Freeswitch-users] Export a channel variable within Dial String
In-Reply-To: <80107a0f-79c8-4922-ae58-830100142291@winet.ch>
References: <481d3184-6518-4957-acca-b1e73c7fec7e@winet.ch>
	<B724748E-DF66-4CD4-B987-BED15F266C78@ipeva.fr>
	<1ff0ebed-7410-4bdb-ab4c-619cadd564ed@winet.ch>
	<C640E45F-505B-4700-9074-EDDBED4CA50B@ipeva.fr>
	<80107a0f-79c8-4922-ae58-830100142291@winet.ch>
Message-ID: <5E8E20FC-9E35-43C7-BFD2-70837CEA7F94@ipeva.fr>

Don't be!
The one who should be embarassed is the one who never asks questions and never makes mistakes :)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 03/12/2010 ? 10:28, Bernhard Suttner a ?crit :

> How embrassing?. You are right, of course. Thanks J
>  
>  
>  
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von David Ponzone
> Gesendet: Donnerstag, 2. Dezember 2010 22:17
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] Export a channel variable within Dial String
>  
> Bernhard,
>  
> I am not sure I follow you.
>  
> AFAIK, exporting and setting with {} in the dialstring is basically the same.
> There is a difference, but it's a very minor technical one.
> Also, if you export, the variable is set on both legs, while {} will only set it on leg B.
>  
> So you say you dont want to export on a specific leg B, but in your dialstring, you dont export, you just set a variable on the leg B.
> Nothing will ever be exported from leg B.
> Variables can be exported from leg A to leg B.
>  
> I could be wrong, but that has been my understanding for the last 12 months.
>  
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>  
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>  
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>  
> 
> 
>  
> Le 02/12/2010 ? 17:31, Bernhard Suttner a ?crit :
> 
> 
> The channel variable will be set, but not exported like the ?export? dialplan function does export the variable to the new channel.
>  
>  
>  
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von David Ponzone
> Gesendet: Donnerstag, 2. Dezember 2010 16:14
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] Export a channel variable within Dial String
>  
> Bernhard,
>  
> AFAIU, if you specify the variable in the dialstring, it's for leg B, so it's exported.
> If you wan't it, don't set it :)
>  
> And for leg A, well, you only have one leg A, so you can't set it twice.
>  
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>  
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>  
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>  
> 
> 
> 
>  
> Le 02/12/2010 ? 15:48, Bernhard Suttner a ?crit :
> 
> 
> 
> Hi,
> 
> I call multiple destination with bridge like that:
> 
> [var1=value]sofia/default/user1,[var1=another_value]loopback/user2
> 
> Is it somehow possible, that var1=another_value will be exported but var1=value not? What I want is, that I could specify a variable to be exported _within_ the dialstring?
> 
> Thanks for every hint.
> 
> Best regards,
> Bernhard
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From vkozak at abisoft.spb.ru  Fri Dec  3 12:58:10 2010
From: vkozak at abisoft.spb.ru (Kozak Vladimir)
Date: Fri, 3 Dec 2010 12:58:10 +0300
Subject: [Freeswitch-users] how set caler info in deflect command
Message-ID: <A8C859F869544985BEEB3C9E9F314D43@abisoft.biz>

Hi all.
Operation deflect have next structure "uuid_deflect <uuid> <sip URL>"  or "<action application="deflect" data="sip:someone at somewhere.com" />"

How I can set specified caller info in this operation? It need set caller name and caller phone.

I try set sip_invite_from_params, sip_callee_id_name, sip_callee_id_number, sip_invite_req_uri, origination_caller_id_name, origination_caller_id_number variables before use deflect command. It dosn't work.

What I need to do?
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From steveayre at gmail.com  Fri Dec  3 15:07:04 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 3 Dec 2010 12:07:04 +0000
Subject: [Freeswitch-users] how set caler info in deflect command
In-Reply-To: <A8C859F869544985BEEB3C9E9F314D43@abisoft.biz>
References: <A8C859F869544985BEEB3C9E9F314D43@abisoft.biz>
Message-ID: <AANLkTine_p4Nba6OpecA+rXg92BgYqk_Wd4AYpCY+6wV@mail.gmail.com>

Try:
<action application="set" data="effective_caller_id_name=Bob Smith"/>
<action application="set" data="effective_caller_id_number=0123456789"/>
<action application="deflect" data="sip:someone at somewhere.com" />

Or, from cli/esl:
uuid_setvar <uuid> effective_caller_id_name Bob Smith
uuid_setvar <uuid> effective_caller_id_number 0123456789
uuid_deflect <uuid> sip:someone at somewhere.com

-Steve


2010/12/3 Kozak Vladimir <vkozak at abisoft.spb.ru>:
> Hi all.
> Operation deflect have next structure "uuid_deflect <uuid> <sip URL>"? or
> "<action application="deflect" data="sip:someone at somewhere.com" />"
> How I can set specified caller info in this operation? It need set caller
> name and caller phone.
>
> I try?set sip_invite_from_params, sip_callee_id_name, sip_callee_id_number,
> sip_invite_req_uri, origination_caller_id_name, origination_caller_id_number
> variables before use deflect command. It dosn't work.
>
> What I need to do?
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From enp at altlinux.org  Fri Dec  3 15:57:32 2010
From: enp at altlinux.org (Eugene Prokopiev)
Date: Fri, 3 Dec 2010 15:57:32 +0300
Subject: [Freeswitch-users] mod_fifo vs mod_callcenter
In-Reply-To: <AANLkTinC85pzthxCVcjopW9q=cjBkkjS0jXGBhKBjyHV@mail.gmail.com>
References: <AANLkTimftpTpLLYXED0Fk1=NwVowzxv=8T0ufQBkrAZ8@mail.gmail.com>
	<AANLkTinC85pzthxCVcjopW9q=cjBkkjS0jXGBhKBjyHV@mail.gmail.com>
Message-ID: <AANLkTikWttM3het5NTO7YJSu=0xT84=O9-4rpvgzNhdC@mail.gmail.com>

I see round robin call distribution for on-hook members -
http://jira.freeswitch.org/browse/FS-1674

Is it possible to implement the same logic for off-hook members? Can
you point me the line/function in mod_fifo.c in which off-hook member
is selecting for call?

--
Thanks,
Eugene Prokopiev


From rodrigoferreiralang at gmail.com  Fri Dec  3 15:51:47 2010
From: rodrigoferreiralang at gmail.com (Rodrigo Lang)
Date: Fri, 3 Dec 2010 10:51:47 -0200
Subject: [Freeswitch-users] Enquiry - Comparing Asterisk and FreeSwitch
Message-ID: <AANLkTik4yDxaz298bnutrPLAZ+RiRRpTxxM-7COGjoY3@mail.gmail.com>

Good day list,

My questions may seem a bit silly, but I always worked with Asterisk and I'm
seeing the potential of FreeSwitch. Two questions that I have are the
following:

1 - FreeSwitch has an interface similar to the Asterisk AMI? Where an
external software can connect and monitor everything that happens in real
time and have some control over the FS?

2 - Can you put your settings in FS in database like the Asterisk Realtime?


Thanks in advance,

-- 
Rodrigo Lang
Opening your mind - Just another Open Source
site<http://openingyourmind.wordpress.com/>
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From peder at networkoblivion.com  Fri Dec  3 16:32:37 2010
From: peder at networkoblivion.com (Peder)
Date: Fri, 3 Dec 2010 07:32:37 -0600
Subject: [Freeswitch-users] Enquiry - Comparing Asterisk and FreeSwitch
In-Reply-To: <AANLkTik4yDxaz298bnutrPLAZ+RiRRpTxxM-7COGjoY3@mail.gmail.com>
References: <AANLkTik4yDxaz298bnutrPLAZ+RiRRpTxxM-7COGjoY3@mail.gmail.com>
Message-ID: <01d201cb92ee$8d319620$a794c260$@com>

1.        EVENT SOCKET:  http://wiki.freeswitch.org/wiki/Event_socket

2.        XML -CURL:  http://wiki.freeswitch.org/wiki/Xml_curl

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rodrigo
Lang
Sent: Friday, December 03, 2010 6:52 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Enquiry - Comparing Asterisk and FreeSwitch

 

Good day list,

My questions may seem a bit silly, but I always worked with Asterisk and I'm
seeing the potential of FreeSwitch. Two questions that I have are the
following:

1 - FreeSwitch has an interface similar to the Asterisk AMI? Where an
external software can connect and monitor everything that happens in real
time and have some control over the FS?

2 - Can you put your settings in FS in database like the Asterisk Realtime?


Thanks in advance,

-- 
Rodrigo Lang
Opening your mind - Just another Open Source site
<http://openingyourmind.wordpress.com/> 

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From juanito1982 at gmail.com  Fri Dec  3 16:40:18 2010
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Fri, 3 Dec 2010 14:40:18 +0100
Subject: [Freeswitch-users] Enquiry - Comparing Asterisk and FreeSwitch
In-Reply-To: <AANLkTik4yDxaz298bnutrPLAZ+RiRRpTxxM-7COGjoY3@mail.gmail.com>
References: <AANLkTik4yDxaz298bnutrPLAZ+RiRRpTxxM-7COGjoY3@mail.gmail.com>
Message-ID: <AANLkTiki=szPWWz614UgzSywvAufcwq-uyoc2_E16A8R@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Rosetta_stone

2010/12/3 Rodrigo Lang <rodrigoferreiralang at gmail.com>

> Good day list,
>
> My questions may seem a bit silly, but I always worked with Asterisk and
> I'm seeing the potential of FreeSwitch. Two questions that I have are the
> following:
>
> 1 - FreeSwitch has an interface similar to the Asterisk AMI? Where an
> external software can connect and monitor everything that happens in real
> time and have some control over the FS?
>
> 2 - Can you put your settings in FS in database like the Asterisk Realtime?
>
>
> Thanks in advance,
>
> --
> Rodrigo Lang
> Opening your mind - Just another Open Source site<http://openingyourmind.wordpress.com/>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dome at tel.co.th  Fri Dec  3 18:14:22 2010
From: dome at tel.co.th (dome at tel.co.th)
Date: Fri, 3 Dec 2010 22:14:22 +0700
Subject: [Freeswitch-users] Need help about lua
Message-ID: <AANLkTintWwkL+7bYd0dpykKGhVNNDH=A+Bffgr8DsQeQ@mail.gmail.com>

Dear All,
   After upgrade from 1.0.6 to git current version. i got problem
about lua and i report to jira
(http://jira.freeswitch.org/browse/FS-2893)
i remove all lua devel package and recompile by make current (I'm sure
mod_lua use FS liblua.a) but i still got problem

i realy need to use luamemcahe , luasocket in my lua script

Someone help me please

BG

Dome C.


From srinivas.ksvreddy at gmail.com  Fri Dec  3 18:53:08 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Fri, 3 Dec 2010 21:23:08 +0530
Subject: [Freeswitch-users] Error loading mod_managed in CentOS
Message-ID: <AANLkTik6TREYzozsKWPWavHHytCERO89QST3_12JY227@mail.gmail.com>

All,
We are presently working on Windows Based FreeSwitch server but doing
piloting on using CentOS platform for better performance. We are presently
using modified Managed module (mod_managed.dll) using C# and .NET framework
3.5. We want to use the same dll in linux too. We found the possibility
using Mono 2.8 but when we place the mono built dll in the /mod folder, we
are getting error while loading saying that mod_managed.so is not found.

i understand that we are trying to load dll. could someone please help us on
how to link the dll so that FreeSwitch will load the module without errors?

We also found that mod_mono_managed is also available. I built the
mod_mono_managed.dll in mono windows and tried to load the module which also
failed.

Any help provided is highly appreciated.

Thanks
Srinivas
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From jeff at jefflenk.com  Fri Dec  3 19:06:24 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 3 Dec 2010 08:06:24 -0800 (PST)
Subject: [Freeswitch-users] Error loading mod_managed in CentOS
In-Reply-To: <AANLkTik6TREYzozsKWPWavHHytCERO89QST3_12JY227@mail.gmail.com>
References: <AANLkTik6TREYzozsKWPWavHHytCERO89QST3_12JY227@mail.gmail.com>
Message-ID: <1291392384758-5800351.post@n2.nabble.com>


Have you followed the instructions here?
http://wiki.freeswitch.org/wiki/Mod_managed

mod_managed.so has to be built on your target linux platform. Only the
managed dlls(that you author) are compatible accross platforms.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-loading-mod-managed-in-CentOS-tp5800322p5800351.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From haloha201 at yahoo.com  Fri Dec  3 19:01:09 2010
From: haloha201 at yahoo.com (ha do)
Date: Fri, 3 Dec 2010 08:01:09 -0800 (PST)
Subject: [Freeswitch-users] confusing on skypopen module
Message-ID: <740634.4551.qm@web32401.mail.mud.yahoo.com>

hi all

i try to use the skypopen and i follow on wiki page
there are some places that are very hard to understand


1/

let's edit the startskype script 
remember to add the removing of all the installed snd-* modules 
cp freeswitch/src/mod/endpoints/mod_skypopen/configs/startskype.sh ./
vi startskype.sh

start the X servers and the Skype clients 
sh ./startskype.sh


2/

on "How to prepare the configuration directory of Skype clients on Linux", see 
http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#How_to_prepare_the_configuration_directory_of_Skype_clients_on_Linux_using_ssh_-X_and_xauth
 
cp /mnt/root/configskypenew.tgz ./
tar xzf configskypenew.tgz
chown root.root .Skype

configskypenew.tgz is not a published file - you must generate it  yourself for 
your own configuration using the instructions at the above  link. This is 
because it contains your authentication details. 


3/

Copy and install the Skype clients configuration directory you previously 
prepared (see 
http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#How_to_prepare_the_configuration_directory_of_Skype_clients_on_Linux_using_ssh_-X_and_xauth
 for how to prepare it): 

cd /root
mount /dev/hda5 /mnt
cp /mnt/root/skypeconfig2.tgz ./
tar xzf skypeconfig2.tgz
chown -R root.root .Skype


4/

Install the skypopen configuration files, edit and execute the script that 
starts the Skype client instances: 

cd /usr/src
cd freeswitch/src/mod/endpoints/mod_skypopen/
cd configs/
cp skypopen.conf.xml /usr/local/freeswitch/conf/autoload_configs/ ==> copy the 
config to autoload

vi /usr/local/freeswitch/conf/autoload_configs/skypopen.conf.xml ==> what parts 
should i edit in the skypopen.conf.xml

cp startskype.sh 2startskype.sh => where do i put the 2startskype.sh
vi 2startskype.sh => what information should i input
sh ./2startskype.sh



when i start the sh ./startskype.sh i got the error
skype at skype-desktop:/$ sh ./startskype.sh
ERROR: Module snd_* does not exist in /proc/modules
ERROR: Module snd_hda_intel is in use
ERROR: Module snd_dummy is in use

Fatal server error:
Server is already active for display 101
        If this server is no longer running, remove /tmp/.X101-lock
        and start again.

Password:


what is the password i should input here


where i put the real password of skype account and the real account ID of skype 
account


please help
Thank you
Ha



      


From ross at ossiantelecom.co.uk  Fri Dec  3 19:38:46 2010
From: ross at ossiantelecom.co.uk (Ross McKillop)
Date: Fri, 3 Dec 2010 16:38:46 +0000
Subject: [Freeswitch-users] Nibblebill
In-Reply-To: <AANLkTimBsw13-SKboLUp_gNH6MqojCU4y0QD6eKNN5cO@mail.gmail.com>
References: <AANLkTimBsw13-SKboLUp_gNH6MqojCU4y0QD6eKNN5cO@mail.gmail.com>
Message-ID: <CAA3A1B5-9C49-4C3D-A61A-36C9E433D95E@ossiantelecom.co.uk>


On 26 Nov 2010, at 06:06, Francis Trevor wrote:

> Is it possible to bill two separate nibblebill accounts from the same dialplan entry? For instance, can I bill a customer and a vendor account I have in Nibblebill during the same call with different rates? This would be useful in a wholesale scenario. Further, if that isnt possible, can a call be posted to two separate account codes, so as to create an entry in two separate CDR files. Ultimately what I am wanting to accomplish is to make accounting entries in a customer and a vendor account for the same call. I can rate them later, but it would be useful to not have to query the master CDR file for TP's. Am I going about this in a rather complicated way and/or is my logic not lucid enough?

I do something similar (record a call accounting against two accounts) by using the xml_cdr facility to call a CGI script which updates both databases... not as seamless as what you want to do I'm sure but it might be an option?

Regards,
Ross

From renjian at gmail.com  Fri Dec  3 19:39:07 2010
From: renjian at gmail.com (Jian Ren)
Date: Fri, 3 Dec 2010 11:39:07 -0500
Subject: [Freeswitch-users] Fwd:  : Skypopen error
In-Reply-To: <AANLkTi=-Z_o0qZGWG-B3CMTF-jOfmo9npffWXLsNnTzJ@mail.gmail.com>
References: <AANLkTi=W65whZv0f7p6GxfjJWmsG9=OXSs-Tt3jmG1tX@mail.gmail.com>
	<AANLkTi=2f9B2upJx8kg64K4NbRGaiw9fOcULET=4THoM@mail.gmail.com>
	<AANLkTinBeEcX0-6gffYsWaJLNU8j8R+t31hZwMbzX7R+@mail.gmail.com>
	<AANLkTinYqCUDiK=hLKPUJ9Q04VUkGgyHE_BKujasd4Cg@mail.gmail.com>
	<AANLkTiks99vbGmG72fheTRb2QdCkQqQQ01gFZRsjxT2h@mail.gmail.com>
	<AANLkTimcYMYuV1pxo60JhQkfK0+P94sf=b_Er8FJrwP_@mail.gmail.com>
	<AANLkTimcCSMfyaajth4qQBiyDGDfhVpPCJyeVt9-VxdF@mail.gmail.com>
	<AANLkTimgOsmXR6jC8pOhB7URwdn19H91ZtkRC9sfBasU@mail.gmail.com>
	<AANLkTinks_EtXZ-y7a=Xux80XrJey1dAoOr8hnJT64aS@mail.gmail.com>
	<4CF66DB0.9090807@puzzled.xs4all.nl>
	<AANLkTimqo4ZeodHr8WVOz28Uq9MyOuE1a11y33+diE2N@mail.gmail.com>
	<AANLkTinp7W6d3GOh2C=rA748ceQmni+tFA3QNkSu5qj=@mail.gmail.com>
	<AANLkTi=aOgFgELyKVHBAWpLq33a29AepH7Ur+v4YCw1r@mail.gmail.com>
	<AANLkTiks+WPpo6P5qvaeqmEydNQX3RbTBNECkbcsx612@mail.gmail.com>
	<AANLkTinruDtk6J_cJHWp_uguS=tBE6H98NEmi1r8MtFD@mail.gmail.com>
	<AANLkTimAtVns8TxXby+F_dWLGju_u2h+wBiV81gBUXLo@mail.gmail.com>
	<AANLkTimmhePNeGQxo_EQ+DxjfVG4-uneRkE9_7eDuENE@mail.gmail.com>
	<AANLkTinab5bCsTW3S=qWMBPF=KrKntF4_3v6vVFOFHvO@mail.gmail.com>
	<AANLkTi=-Z_o0qZGWG-B3CMTF-jOfmo9npffWXLsNnTzJ@mail.gmail.com>
Message-ID: <AANLkTi=UuY+5-ud=6yDXw_SVze2jT+thtm3dSEs1LPxb@mail.gmail.com>

Try sending again.

---------- Forwarded message ----------
From: Jian Ren <renjian at gmail.com>
Date: Fri, Dec 3, 2010 at 11:13 AM
Subject: Re: [Freeswitch-users] : Skypopen error
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>


Hi Giovanni,
Unfortunately, my wife still got aborted calls with your fix, and
mine(another phone). I could see the warning line you added:
2010-12-02 21:12:45.470780 [DEBUG] switch_rtp.c:2925 RTP RECV DTMF C:400
2010-12-02 21:12:45.470780 [DEBUG] mod_skypopen.c:792          [|]
[DEBUG_SKYPE  792  ][interface00    ][UP,INPROGRS] interface00 CHANNEL
SEND_DTMF
2010-12-02 21:12:45.470780 [DEBUG] mod_skypopen.c:793          [|]
[DEBUG_SKYPE  793  ][interface00    ][UP,INPROGRS] DTMF: C
2010-12-02 21:12:45.470780 [DEBUG] skypopen_protocol.c:1122    [|]
[DEBUG_SKYPE  1122 ][interface00    ][UP,INPROGRS] DIGIT received: C
2010-12-02 21:12:45.470780 [WARNING] skypopen_protocol.c:1127  [|] [WARNINGA
    1127 ][interface00    ][UP,INPROGRS] Received DTMF DIGIT "C", but not
relayed to Skype client because Skype client accepts only 0-9*#
2010-12-02 21:12:45.687811 [DEBUG] skypopen_protocol.c:173     [|]
[DEBUG_SKYPE  173  ][interface00    ][UP,INPROGRS] READING: |||CALL 46932
DURATION 12|||

But the call was still ended within 1 min. Then there were other errors
like:
2010-12-02 21:14:08.708226 [DEBUG] skypopen_protocol.c:173     [|]
[DEBUG_SKYPE  173  ][interface00    ][DOWN,IDLE] READING: |||ERROR 559 CALL:
Action failed|||
2010-12-02 21:14:08.708226 [DEBUG] skypopen_protocol.c:228     [|]
[DEBUG_SKYPE  228  ][interface00    ][DOWN,IDLE] Skype got ERROR about a
failed action (probably TRYING to HANGUP A CALL), no problem: |||ERROR 559
CALL: Action failed|||
2010-12-02 21:14:08.708226 [DEBUG] switch_core_session.c:1993
sofia/internal/1000 at 192.168.1.104 skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)

Attached is the log file. I have turned on the debug level again to give
your more details.
The root cause might not be just the phone, it always happened after the
server has been running for a while (couple hours to 2 days), once server
rebooted, everything works fine.

Should I create another issue or open the original one?
Thanks!
Jian

On Thu, Dec 2, 2010 at 10:31 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> On Thu, Dec 2, 2010 at 3:34 PM, Jian Ren <renjian at gmail.com> wrote:
> > Cool, I did a git clone last night after 7pm, which should contain the
> fix
> > already, right?
>
> Don't know, you can check like this:
>
> # cd /usr/src/freeswitch/src/mod/endpoint/mod_skypopen
> # git log
>
> And you'll see the last commits that you got.
>
> -giovanni
>
> > Maybe that's why I haven't seen the problem so far.
> > Besides, I disconnected her 2.4G cordless phone and replaced with a 900Hz
> > one, in case if it's caused by the Wifi interference.
> >
> > Thanks!
> > Jian
> >
> > On Thu, Dec 2, 2010 at 9:22 AM, Giovanni Maruzzelli <gmaruzz at celliax.org
> >
> > wrote:
> >>
> >> On Thu, Dec 2, 2010 at 3:15 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> >> > Her phone might have a *really* crappy DTMF detector.  They are more
> >> > likely
> >> > to fail miserably with female voices.
> >>
> >> Thanks Rupa!
> >>
> >> Btw, should be fixed in mod_skypopen code since yesterday. You'll now
> >> get a warning, but no more an error, and the call should be
> >> unaffected.
> >>
> >> -giovanni
> >>
> >> >
> >> > On Wed, Dec 1, 2010 at 11:18 AM, Jian Ren <renjian at gmail.com> wrote:
> >> >>
> >> >> Hmm, so far the only "customer" is my wife, who keeps complaining to
> me
> >> >> everyday. It explains why never happened to me. I will ask her
> tonight
> >> >> how
> >> >> she sent A or D while calling. Or maybe it's the problem of the
> phone.
> >> >> She
> >> >> is using a dual mode phone. Before, it's connected to PC with USB for
> >> >> skype
> >> >> calls on Windows. Now I stopped running any version of skype on
> windows
> >> >> and
> >> >> the USB was disconnected so she is using the phone as a normal one.
> >> >> Thanks!
> >> >> Jian
> >> >>
> >> >> On Wed, Dec 1, 2010 at 11:56 AM, Giovanni Maruzzelli
> >> >> <gmaruzz at gmail.com>
> >> >> wrote:
> >> >>>
> >> >>> Hi Jian,
> >> >>>
> >> >>> I hope in English "bizarre" does not sound bad, in Italian would be
> >> >>> like "original and out of standard in a funny way" :).
> >> >>>
> >> >>> >From the log you attached to the Jira, the incoming SIP calls that
> >> >>> are
> >> >>> then bridged to skypopen are sending both the "A" dtmf and the "D"
> >> >>> dtmf (can't remember if any other).
> >> >>>
> >> >>> You can peruse the log looking for "error 21", and you'll  see is
> >> >>> anytime that they sent to you (via SIP) one of the A-B-C-D dtmfs
> that
> >> >>> mod_skypopen duly passes to skype.
> >> >>>
> >> >>> Problem is: the Skype client does not accept or relay dtmf "A-D",
> and
> >> >>> spit out an error.
> >> >>>
> >> >>> Out of curiosity you may want to check why your customers are using
> >> >>> dtrmf A-D, but is not an absolut need.
> >> >>>
> >> >>> Anyway, I'll fix this in mod_skypopen code asap, so that if another
> >> >>> channel (SIP in your case) try to send A-D to skype, that dtmf will
> be
> >> >>> ignored and a warning line will be emitted to console and to
> logfile.
> >> >>> And no more errors or aborted calls.
> >> >>>
> >> >>> -giovanni
> >> >>>
> >> >>>
> >> >>>
> >> >>> On Wed, Dec 1, 2010 at 5:35 PM, Jian Ren <renjian at gmail.com> wrote:
> >> >>> > Also this happened during a call, which got hanged up because of
> the
> >> >>> > error.
> >> >>> > So the phone sends out A-D during a call?
> >> >>> > Jian
> >> >>> > On Wed, Dec 1, 2010 at 11:33 AM, Jian Ren <renjian at gmail.com>
> wrote:
> >> >>> >>
> >> >>> >> Interesting. I don't have these keys on my phone. Here is the
> >> >>> >> dialplan
> >> >>> >> string I am using in the ATA(SPA1001):
> >> >>> >> (<:1>[2-9]xx[2-9]xxxxxx|011xx.|1[2-9]xx[2-9]xxxxxx|1xxx|00xx.)
> >> >>> >> Shouldn't it only take numbers?
> >> >>> >> Could this be caused by any dialplan XML files in freeswitch?
> >> >>> >> Thanks!
> >> >>> >> Jian
> >> >>> >>
> >> >>> >> On Wed, Dec 1, 2010 at 10:45 AM, Patrick Lists
> >> >>> >> <freeswitch-list at puzzled.xs4all.nl> wrote:
> >> >>> >>>
> >> >>> >>> On 12/01/2010 04:03 PM, Jian Ren wrote:
> >> >>> >>> > Hi Giovanni,
> >> >>> >>> > Could you explain more to a bizarre person like me :-)
> >> >>> >>> > What's DTMF "D"? Sent from one of my SIP client? Or Someone
> sent
> >> >>> >>> > to
> >> >>> >>> > my
> >> >>> >>> > skype through chat? If it's abnormal, I think I should fix the
> >> >>> >>> > root
> >> >>> >>> > cause.
> >> >>> >>> > Thank you very much for looking at the issue, with 2M logs
> :-).
> >> >>> >>>
> >> >>> >>> DTMF A-D seems to be used only on military phones:
> >> >>> >>>
> >> >>> >>>
> >> >>> >>>
> http://www.telecomdictionary.com/telecom_dictionary_DTMF_definition.html
> >> >>> >>>
> >> >>> >>> Regards,
> >> >>> >>> Patrick
> >> >>> >>>
> >> >>> >>> _______________________________________________
> >> >>> >>> FreeSWITCH-users mailing list
> >> >>> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>>
> >> >>> >>>
> >> >>> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> >>> http://www.freeswitch.org
> >> >>> >>
> >> >>> >
> >> >>> >
> >> >>> > _______________________________________________
> >> >>> > FreeSWITCH-users mailing list
> >> >>> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >
> >> >>> >
> >> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> > http://www.freeswitch.org
> >> >>> >
> >> >>> >
> >> >>>
> >> >>>
> >> >>>
> >> >>> --
> >> >>> Sincerely,
> >> >>>
> >> >>> Giovanni Maruzzelli
> >> >>> Cell : +39-347-2665618
> >> >>>
> >> >>> _______________________________________________
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>
> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> >
> >> > --
> >> > -Rupa
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From romerocarlos24 at gmail.com  Fri Dec  3 20:45:24 2010
From: romerocarlos24 at gmail.com (Carlos Romero)
Date: Fri, 3 Dec 2010 14:45:24 -0300
Subject: [Freeswitch-users] Asterisk SCF vs FreeSWITCH
Message-ID: <AANLkTin9ne0BXr1aRpEk6jNysF=_LGmMSaCc3incu2i0@mail.gmail.com>

hello

what is the difference between Asterisk SCF vs FreeSWITCH?

https://wiki.asterisk.org/wiki/display/TOP/Asterisk+SCF+Home


From dome at tel.co.th  Fri Dec  3 20:57:01 2010
From: dome at tel.co.th (dome at tel.co.th)
Date: Sat, 4 Dec 2010 00:57:01 +0700
Subject: [Freeswitch-users] Need help about lua
In-Reply-To: <AANLkTintWwkL+7bYd0dpykKGhVNNDH=A+Bffgr8DsQeQ@mail.gmail.com>
References: <AANLkTintWwkL+7bYd0dpykKGhVNNDH=A+Bffgr8DsQeQ@mail.gmail.com>
Message-ID: <AANLkTimCA51YPnkR0Wt8YxVHxEX25zdm9Rh5ibRL8WXx@mail.gmail.com>

liblua5.1-curl0 is problem
I'm testing with system lua lib (debian sid) and lualib from FS got
same problem
but it's workfine in 1.0.6
it's work fine when i remove require 'curl' from my lua script.

BG

Dome C.




2010/12/3 dome at tel.co.th <dome at tel.co.th>:
> Dear All,
> ? After upgrade from 1.0.6 to git current version. i got problem
> about lua and i report to jira
> (http://jira.freeswitch.org/browse/FS-2893)
> i remove all lua devel package and recompile by make current (I'm sure
> mod_lua use FS liblua.a) but i still got problem
>
> i realy need to use luamemcahe , luasocket in my lua script
>
> Someone help me please
>
> BG
>
> Dome C.
>


From david.ponzone at ipeva.fr  Fri Dec  3 21:09:00 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 3 Dec 2010 19:09:00 +0100
Subject: [Freeswitch-users] Asterisk SCF vs FreeSWITCH
In-Reply-To: <AANLkTin9ne0BXr1aRpEk6jNysF=_LGmMSaCc3incu2i0@mail.gmail.com>
References: <AANLkTin9ne0BXr1aRpEk6jNysF=_LGmMSaCc3incu2i0@mail.gmail.com>
Message-ID: <6D8FED3C-2F0D-4D67-A61C-FB312C34EDAF@ipeva.fr>

Hmm I would say the main one is that FreeSWICH is there now, and has been around for some years now.
Asterisk SCF is a project, not very advanced it seems.
If you can afford to wait 2 or 3 years...:)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 03/12/2010 ? 18:45, Carlos Romero a ?crit :

> hello
> 
> what is the difference between Asterisk SCF vs FreeSWITCH?
> 
> https://wiki.asterisk.org/wiki/display/TOP/Asterisk+SCF+Home
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From renjian at gmail.com  Fri Dec  3 22:43:14 2010
From: renjian at gmail.com (Jian Ren)
Date: Fri, 3 Dec 2010 14:43:14 -0500
Subject: [Freeswitch-users] confusing on skypopen module
In-Reply-To: <740634.4551.qm@web32401.mail.mud.yahoo.com>
References: <740634.4551.qm@web32401.mail.mud.yahoo.com>
Message-ID: <AANLkTi=tW_HgFigt_0tAfqsWOGas2n-QmdqA2=N13O5n@mail.gmail.com>

I am seeing where I was couple weeks ago :-).

Ha, all the steps below are to launch multiple skype instances(in different
X instances).

Step 2 is to prepare the base folder for skype. If you already followed the
step to launch Skype once from a remote client using XAuth. What you need to
do is just to copy the content under .Skype folder to as many places as the
number of skype instances you want to run. Forget about the step 2 and 3,
this part on wiki explains better:

mkdir -p /root/multi/01
mkdir -p /root/multi/02
cp -a /root/.Skype/* /root/multi/01/
cp -a /root/.Skype/* /root/multi/02/


Also I suggest you do everything with root login.

Those module not found errors are from the lines to remove all modules other
than snd-dummy(guess not required but strongly suggested). What I did is to
disable my onboard audio card in CMOS so when the server boots, no snd
modules loaded at all. You could comment out the remove module lines and
only leave the modprobe snd-dummy one there.

The skype launch script(no matter you use startskype.sh or multi.sh) just do
the below steps:

load snd-dummy
launch X number 1
launch skype 1 in X number 1
....
launch X number N
launch skype N in X number N

The parts you need to customize are the skype account and base folder(the
/root/multi/xx as above).

The skypopen.conf.xml file needs to be updated so it matches your skype
launch script(same skype account and same X number)

Good Luck!
Jian


On Fri, Dec 3, 2010 at 11:01 AM, ha do <haloha201 at yahoo.com> wrote:

> hi all
>
> i try to use the skypopen and i follow on wiki page
> there are some places that are very hard to understand
>
>
> 1/
>
> let's edit the startskype script
> remember to add the removing of all the installed snd-* modules
> cp freeswitch/src/mod/endpoints/mod_skypopen/configs/startskype.sh ./
> vi startskype.sh
>
> start the X servers and the Skype clients
> sh ./startskype.sh
>
>
> 2/
>
> on "How to prepare the configuration directory of Skype clients on Linux",
> see
>
> http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#How_to_prepare_the_configuration_directory_of_Skype_clients_on_Linux_using_ssh_-X_and_xauth
>
> cp /mnt/root/configskypenew.tgz ./
> tar xzf configskypenew.tgz
> chown root.root .Skype
>
> configskypenew.tgz is not a published file - you must generate it  yourself
> for
> your own configuration using the instructions at the above  link. This is
> because it contains your authentication details.
>
>
> 3/
>
> Copy and install the Skype clients configuration directory you previously
> prepared (see
>
> http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#How_to_prepare_the_configuration_directory_of_Skype_clients_on_Linux_using_ssh_-X_and_xauth
>  for how to prepare it):
>
> cd /root
> mount /dev/hda5 /mnt
> cp /mnt/root/skypeconfig2.tgz ./
> tar xzf skypeconfig2.tgz
> chown -R root.root .Skype
>
>
> 4/
>
> Install the skypopen configuration files, edit and execute the script that
> starts the Skype client instances:
>
> cd /usr/src
> cd freeswitch/src/mod/endpoints/mod_skypopen/
> cd configs/
> cp skypopen.conf.xml /usr/local/freeswitch/conf/autoload_configs/ ==> copy
> the
> config to autoload
>
> vi /usr/local/freeswitch/conf/autoload_configs/skypopen.conf.xml ==> what
> parts
> should i edit in the skypopen.conf.xml
>
> cp startskype.sh 2startskype.sh => where do i put the 2startskype.sh
> vi 2startskype.sh => what information should i input
> sh ./2startskype.sh
>
>
>
> when i start the sh ./startskype.sh i got the error
> skype at skype-desktop:/$ sh ./startskype.sh
> ERROR: Module snd_* does not exist in /proc/modules
> ERROR: Module snd_hda_intel is in use
> ERROR: Module snd_dummy is in use
>
> Fatal server error:
> Server is already active for display 101
>        If this server is no longer running, remove /tmp/.X101-lock
>        and start again.
>
> Password:
>
>
> what is the password i should input here
>
>
> where i put the real password of skype account and the real account ID of
> skype
> account
>
>
> please help
> Thank you
> Ha
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dftoro at yahoo.com  Sat Dec  4 03:00:08 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Fri, 3 Dec 2010 16:00:08 -0800 (PST)
Subject: [Freeswitch-users] Custom Channel Variables via C#
In-Reply-To: <0E462B4C-C436-462E-9DA7-9EC1241FB380@avgs.ca>
Message-ID: <165160.41076.qm@web33501.mail.mud.yahoo.com>

Hi, You can use ESL managed, is the easy way.

Diego 

http://voipensando.blogspot.com/

--- On Wed, 12/1/10, Mathieu Rene <mrene_lists at avgs.ca> wrote:

From: Mathieu Rene <mrene_lists at avgs.ca>
Subject: Re: [Freeswitch-users] Custom Channel Variables via C#
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Wednesday, December 1, 2010, 4:53 PM

Hi,
You can't send XML dialplan actions on the socket and expect FreeSWITCH to understand it, you must follow the event socket protocol.
See?http://wiki.freeswitch.org/wiki/Event_socket?for more information.

Mathieu ReneAvant-Garde Solutions IncOffice: + 1 (514) 664-1044 x100Cell: +1 (514) 664-1044 x200mrene at avgs.ca





On 2010-12-01, at 4:46 PM, Brian Campbell wrote:
?
I can set a custom channel variable in my incomming dial plan like this...
?
<action application='set' data='MyChannelVariable=12345' />
?
And it shows up fine in the CDR
?
I am now attempting to use mod_event_socket to set a custom channel variable and have it?appear in the CDR
?
Here is the C# code so far, the code is part of a Microsoft Speech Server application that is answering the call
?
private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
{
? try
? {
??? TcpClient newClient = new TcpClient();
?
??? newClient.Connect("127.0.0.1", 8021);
?
??? NetworkStream tcpStream = newClient.GetStream();
?
??? byte[] sendBytes = Encoding.ASCII.GetBytes("<action application='set' data='MyChannelVariable=12345' />");
?
??? tcpStream.Write(sendBytes, 0, sendBytes.Length);
?
??? tcpStream.Close();
?
??? newClient.Close();
? }
? catch(Exception ex)
? {
??? throw ex;
? }
}
?
After the call is answered, the C# code seems to run fine, but I dont see the custom channel variable in the resulting CDR
?
I figure I am not setting it correctly
?
Can anyone advise on what I am doing wrong ?
?
Thanks
?
?
Brian
?
?
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


-----Inline Attachment Follows-----

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From davidjbrazier at gmail.com  Sat Dec  4 03:39:25 2010
From: davidjbrazier at gmail.com (David Brazier)
Date: Sat, 4 Dec 2010 00:39:25 +0000
Subject: [Freeswitch-users] Error loading mod_managed in CentOS
In-Reply-To: <1291392384758-5800351.post@n2.nabble.com>
References: <AANLkTik6TREYzozsKWPWavHHytCERO89QST3_12JY227@mail.gmail.com>
	<1291392384758-5800351.post@n2.nabble.com>
Message-ID: <AANLkTi=fKFVHbAzNDk1RBfA2rGoJs0tijmE3qMGwu=Rr@mail.gmail.com>

Make sure you uncomment "languages/mod_managed" is uncommented in
modules.conf and follow the Mono 2.8 instructions on

http://wiki.freeswitch.org/wiki/Mod_managed

Don't try mod_mono & mod_mono_managed - that is old stuff.

David


From davidjbrazier at gmail.com  Sat Dec  4 03:43:43 2010
From: davidjbrazier at gmail.com (David Brazier)
Date: Sat, 4 Dec 2010 00:43:43 +0000
Subject: [Freeswitch-users] building mod_managed fails
In-Reply-To: <AANLkTim9EFf4HxHt=hh+d2YzP-e+mBzoCodfGqMdQCDg@mail.gmail.com>
References: <AANLkTi=96X=WyXFZ1ZTU8AE3fKiZUm6Q=tr-ds4kOCWa@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6703681DD6C9@mse17be1.mse17.exchange.ms>
	<AANLkTi=xLHN7SAL9t0A+aiDtGTwYoCDwqNFjMapSGQpu@mail.gmail.com>
	<AANLkTimWZYJJEyp9Ayu-Ou3asR0kqbZ2EcjVq8mLAk_c@mail.gmail.com>
	<4CF66CBF.9080308@puzzled.xs4all.nl>
	<AANLkTim9EFf4HxHt=hh+d2YzP-e+mBzoCodfGqMdQCDg@mail.gmail.com>
Message-ID: <AANLkTikMPEkXOtdGyZ=iNH_vkQeKeC8_d_=BavLKc+8s@mail.gmail.com>

On Wed, Dec 1, 2010 at 5:16 PM, David Brazier <davidjbrazier at gmail.com> wrote:
> I'll try to investigate why the swig output wasn't right.

I found out why - you need SWIG 2.0 - SWIG 1.3 that I had doesn't do it right.


From juanito1982 at gmail.com  Sat Dec  4 16:24:06 2010
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Sat, 4 Dec 2010 14:24:06 +0100
Subject: [Freeswitch-users] Problem with SIP 180 + FreeTDM
In-Reply-To: <4CF7F784.7090201@sangoma.com>
References: <AANLkTi=3RuKF9YzGmvz=Xcga4zdH6_eOod7H0XMd=Kyn@mail.gmail.com>
	<4CE5A68B.2060507@sangoma.com>
	<AANLkTi=24KMXRz1FMi7nLTPkjFj2nzqFycD8ZPZ2b+g9@mail.gmail.com>
	<4CF7F784.7090201@sangoma.com>
Message-ID: <AANLkTimsYTnJbUA=S-Mgb95=6BEWpNruUuLvJbe-2u3Q@mail.gmail.com>

You are true. Last git version works as expected.

Thank you very much

2010/12/2 David Yat Sin <dyatsin at sangoma.com>

>  Hi Juan Antonio,
>
> Can you pull the latest Freeswitch git. This should be fixed now.
>
>
> David
>
>   *David Yat Sin, BEng*
> Software Engineer
> Sangoma Technologies
> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
>
> t.   +1 800 388 2475 x119
> t.   +1 905 474 1990 x119
> f.   +1 905 474 9223 [image: SANGOMA]<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
> Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
> Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
> Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
> Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
> Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
> Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>
> On 11/24/2010 1:38 PM, Juan Antonio Iba?ez Santorum wrote:
>
> Any news?
>
> 2010/11/18 David Yat Sin <dyatsin at sangoma.com>
>
>>  Hi Juan Antonio,
>> We are looking into this and will get back to you in a day or two.
>>
>> David
>>
>>   *David Yat Sin, BEng*
>> Software Engineer
>> Sangoma Technologies
>> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
>>
>> t.   +1 800 388 2475 x119
>> t.   +1 905 474 1990 x119
>> f.   +1 905 474 9223 [image: SANGOMA]<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>
>> On 11/18/2010 3:50 PM, Juan Antonio Iba?ez Santorum wrote:
>>
>>  Hello boys!
>>
>>     I have one problem after updating from openzap to freetdm. Now, whe
>> FS receives 'Proceeding' from PRI sends a SIP 180 which makes the phone to
>> play a fake ring tone. After some seconds, when FS receives 'Alerting' from
>> PRI, FS sends a SIP 183 with the real ring tone.
>>
>>  Is there any way to avoid first SIP 180? If the calle is busy, I can
>> hear a ringing tone between SIP 180 and SIP 183 and a busy tone after SIP
>> 183. Same if calle is a mobile phone and it is out of signal.
>>
>>  Regards
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Sun Dec  5 02:35:02 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 4 Dec 2010 17:35:02 -0600
Subject: [Freeswitch-users] GIT/SVN Repo Maint.
Message-ID: <9C0A4345-C459-46B5-BB72-F300EBD9FB66@freeswitch.org>

I'm going to be doing some work on the repo to move it this weekend so please be patient if you can't access it while I'm moving it to Dallas Sunday December 5th 2010.

Thanks,
Brian



From slim at thegreek.com  Sun Dec  5 10:23:33 2010
From: slim at thegreek.com (Jeroen C. van Gelderen)
Date: Sun, 5 Dec 2010 02:23:33 -0500
Subject: [Freeswitch-users] Possible Loopback / Originate Race?
Message-ID: <AFAC63EEB7A54AE99FBD403D12FE2F87@mbnet.local>

Hi,

 

(Filed in JIRA FS-2899. Please let me know if posting this on the list as
well is frowned upon.)

 

I'm using switch_limit in a loopback extension with the following
problematic dialplan fragment (full dialplan context at end of message):

 

    <extension name="limit_exceeded">

      <condition field="destination_number" expression="^limit_exceeded$">

 

        <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>

 

      </condition>

    </extension>

 

This works fine in MOST cases with the expected result:

  switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 34
[NORMAL_CIRCUIT_CONGESTION]

 

Every once in a while though this very same extension will magically yield:

  switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 19
[NO_ANSWER]

 

Looking at the log snippets pasted below the correct result (34) appears
when this sequence of events occurs (first snippet):

 

  mod_loopback.c:425 Hangup loopback/18884447693-a [CS_CONSUME_MEDIA]
[NORMAL_CIRCUIT_CONGESTION]

  switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 34
[NORMAL_CIRCUIT_CONGESTION]

 

The incorrect answer (19) appears whenever this sequence shows in the logs:

 

  switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 19
[NO_ANSWER]

  mod_loopback.c:425 Hangup loopback/18884447701-a [CS_CONSUME_MEDIA]
[NORMAL_CIRCUIT_CONGESTION]

 

Pouring over my logs this ALWAYS is the case. Whenever the wrong error cause
appears the switch_ivr_originate call precedes the mod_loopback call.
Conversely whenever the correct error cause is given the call sequence is
reversed.

 

I can reproduce this at will by rapidly invoking the loopback extension in
the dialplan pasted at the bottom of this mail.

 

Any help is appreciated.

 

Cheers,

-Slim

 

----8<----8<----8<----8<----8<----8<----8<----

EXECUTE loopback/18884447693-b hangup(NORMAL_CIRCUIT_CONGESTION)

2010-12-05 01:27:13.318020 [DEBUG] switch_channel.c:2455
(loopback/18884447693-b) Callstate Change EARLY -> HANGUP

2010-12-05 01:27:13.318020 [NOTICE] mod_dptools.c:906 Hangup
loopback/18884447693-b [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION]

 

2010-12-05 01:27:13.319023 [DEBUG] switch_channel.c:2471 Send signal
loopback/18884447693-b [KILL]

2010-12-05 01:27:13.319023 [DEBUG] mod_loopback.c:468 loopback/18884447693-b
CHANNEL KILL

2010-12-05 01:27:13.319023 [DEBUG] switch_core_session.c:1083 Send signal
loopback/18884447693-b [BREAK]

2010-12-05 01:27:13.319023 [DEBUG] mod_loopback.c:468 loopback/18884447693-b
CHANNEL KILL

 

2010-12-05 01:27:13.320024 [DEBUG] switch_core_session.c:1993
loopback/18884447693-b skip receive message [APPLICATION_EXEC_COMPLETE]
(chann

el is hungup already)

 

2010-12-05 01:27:13.320024 [DEBUG] switch_core_state_machine.c:366
(loopback/18884447693-b) State EXECUTE going to sleep

2010-12-05 01:27:13.320024 [DEBUG] switch_core_state_machine.c:320
(loopback/18884447693-b) Running State Change CS_HANGUP

2010-12-05 01:27:13.321011 [DEBUG] switch_core_state_machine.c:557
(loopback/18884447693-b) State HANGUP

 

 

2010-12-05 01:27:13.321983 [DEBUG] mod_loopback.c:414 loopback/18884447693-b
CHANNEL HANGUP

2010-12-05 01:27:13.321983 [DEBUG] switch_channel.c:2455
(loopback/18884447693-a) Callstate Change EARLY -> HANGUP

 

2010-12-05 01:27:13.321983 [NOTICE] mod_loopback.c:425 Hangup
loopback/18884447693-a [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION]

 

2010-12-05 01:27:13.322995 [DEBUG] switch_ivr_originate.c:3448 Originate
Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]

 

----8<----8<----8<----8<----8<----8<----8<----

 

EXECUTE loopback/18884447701-b hangup(NORMAL_CIRCUIT_CONGESTION)

2010-12-05 01:27:14.364448 [DEBUG] switch_channel.c:2455
(loopback/18884447701-b) Callstate Change EARLY -> HANGUP

2010-12-05 01:27:14.365470 [NOTICE] mod_dptools.c:906 Hangup
loopback/18884447701-b [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION]

 

2010-12-05 01:27:14.365470 [DEBUG] switch_channel.c:2471 Send signal
loopback/18884447701-b [KILL]

2010-12-05 01:27:14.365470 [DEBUG] mod_loopback.c:468 loopback/18884447701-b
CHANNEL KILL

2010-12-05 01:27:14.365470 [DEBUG] switch_core_session.c:1083 Send signal
loopback/18884447701-b [BREAK]

2010-12-05 01:27:14.366485 [DEBUG] mod_loopback.c:468 loopback/18884447701-b
CHANNEL KILL

 

2010-12-05 01:27:14.366485 [DEBUG] switch_core_session.c:1993
loopback/18884447701-b skip receive message [APPLICATION_EXEC_COMPLETE]
(chann

el is hungup already)

 

2010-12-05 01:27:14.366485 [DEBUG] switch_core_state_machine.c:366
(loopback/18884447701-b) State EXECUTE going to sleep

2010-12-05 01:27:14.367508 [DEBUG] switch_core_state_machine.c:320
(loopback/18884447701-b) Running State Change CS_HANGUP

2010-12-05 01:27:14.369585 [DEBUG] switch_core_state_machine.c:557
(loopback/18884447701-b) State HANGUP

 

2010-12-05 01:27:14.369585 [DEBUG] mod_loopback.c:414 loopback/18884447701-b
CHANNEL HANGUP

2010-12-05 01:27:14.369585 [DEBUG] switch_channel.c:2455
(loopback/18884447701-a) Callstate Change EARLY -> HANGUP

 

2010-12-05 01:27:14.369585 [DEBUG] switch_ivr_originate.c:3448 Originate
Resulted in Error Cause: 19 [NO_ANSWER]

 

2010-12-05 01:27:14.369585 [NOTICE] mod_loopback.c:425 Hangup
loopback/18884447701-a [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION]

 

----8<----8<----8<----8<----8<----8<----8<----

 

 

  <context name="tech-outbound">

 

    <extension name="tech-outbound">

      <condition field="destination_number" expression="^1888(\d{7})$">

 

        <action application="info"/>

        <action application="set" data="loopback_bowout=false"/>

        <action application="limit" data="db outgoing goip 1" />

        <action application="bridge"
data="{ignore_early_media=true}sofia/internal/${destination_number}@192.168.
3.11:5060"/>

 

      </condition>

    </extension>

 

    <extension name="limit_exceeded">

      <condition field="destination_number" expression="^limit_exceeded$">

 

        <action application="<nothing> OR <answer> OR <pre_answer>"/> <!--
Tried all 3 variations, no difference -->

 

        <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>

 

      </condition>

    </extension>

 

  </context>

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From trob at freemail.hu  Sun Dec  5 20:11:28 2010
From: trob at freemail.hu (=?ISO-8859-2?Q?T=F3th_R=F3bert?=)
Date: Sun, 05 Dec 2010 18:11:28 +0100
Subject: [Freeswitch-users] event from LUA to phones
Message-ID: <4CFBC7C0.1050305@freemail.hu>

Hi

I would like send an event from LUA to a phone, with this header:
Event: x-gs-screen

(This message makes the Grandstream phone refresh its Idle-Screen)

I tried this:

             eout = freeswitch.Event("NOTIFY");
             eout.addHeader(eout, "profile", "mellek-belso");
             eout.addHeader(eout, "to-uri", "sip:50 at 192.168.1.10");
             eout.addHeader(eout, "from-uri", "sip:50 at 192.168.1.10");
             eout.addHeader(eout, "event-string", "x-gs-screen");
             eout.fire(eout);

and played with it a lot, e.g. add more headers, change the IPs, ...
but the SIP message does not arrive to the phone, altough there are not any error message.

How can i do this?

Thanks



From moises.silva at gmail.com  Mon Dec  6 01:08:29 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Sun, 5 Dec 2010 17:08:29 -0500
Subject: [Freeswitch-users] Asterisk SCF vs FreeSWITCH
In-Reply-To: <6D8FED3C-2F0D-4D67-A61C-FB312C34EDAF@ipeva.fr>
References: <AANLkTin9ne0BXr1aRpEk6jNysF=_LGmMSaCc3incu2i0@mail.gmail.com>
	<6D8FED3C-2F0D-4D67-A61C-FB312C34EDAF@ipeva.fr>
Message-ID: <AANLkTik2+_aCu3eVmcvK8EJOxXB-mCFUtP9XTOD19umq@mail.gmail.com>

2010/12/3 David Ponzone <david.ponzone at ipeva.fr>

> Hmm I would say the main one is that FreeSWICH is there now, and has been
> around for some years now.
> Asterisk SCF is a project, not very advanced it seems.
> If you can afford to wait 2 or 3 years...:)
>
>
Probably the most technically visibile difference (other than FreeSWITCH is
here now) is that Asterisk SCF aims to follow a distributed model
(components run in different processes or even different machines).

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com
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From moises.silva at gmail.com  Mon Dec  6 01:10:23 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Sun, 5 Dec 2010 17:10:23 -0500
Subject: [Freeswitch-users] Problem with SIP 180 + FreeTDM
In-Reply-To: <AANLkTimsYTnJbUA=S-Mgb95=6BEWpNruUuLvJbe-2u3Q@mail.gmail.com>
References: <AANLkTi=3RuKF9YzGmvz=Xcga4zdH6_eOod7H0XMd=Kyn@mail.gmail.com>
	<4CE5A68B.2060507@sangoma.com>
	<AANLkTi=24KMXRz1FMi7nLTPkjFj2nzqFycD8ZPZ2b+g9@mail.gmail.com>
	<4CF7F784.7090201@sangoma.com>
	<AANLkTimsYTnJbUA=S-Mgb95=6BEWpNruUuLvJbe-2u3Q@mail.gmail.com>
Message-ID: <AANLkTimAWtbv-TEwcV_c_zqxh6yH2MqyFOhZASm1OYTE@mail.gmail.com>

2010/12/4 Juan Antonio Iba?ez Santorum <juanito1982 at gmail.com>

> You are true. Last git version works as expected.
>
> Thank you very much
>
>
We're very interested in hearing about this kind of inter-networking issues
between SIP and the PSTN, if you find any other glitch, let us know.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com




> 2010/12/2 David Yat Sin <dyatsin at sangoma.com>
>
>>  Hi Juan Antonio,
>>
>> Can you pull the latest Freeswitch git. This should be fixed now.
>>
>>
>> David
>>
>>   *David Yat Sin, BEng*
>> Software Engineer
>> Sangoma Technologies
>> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
>>
>> t.   +1 800 388 2475 x119
>> t.   +1 905 474 1990 x119
>> f.   +1 905 474 9223 [image: SANGOMA]<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>
>> On 11/24/2010 1:38 PM, Juan Antonio Iba?ez Santorum wrote:
>>
>> Any news?
>>
>> 2010/11/18 David Yat Sin <dyatsin at sangoma.com>
>>
>>>  Hi Juan Antonio,
>>> We are looking into this and will get back to you in a day or two.
>>>
>>> David
>>>
>>>   *David Yat Sin, BEng*
>>> Software Engineer
>>> Sangoma Technologies
>>> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
>>>
>>> t.   +1 800 388 2475 x119
>>> t.   +1 905 474 1990 x119
>>> f.   +1 905 474 9223 [image: SANGOMA]<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>>
>>> On 11/18/2010 3:50 PM, Juan Antonio Iba?ez Santorum wrote:
>>>
>>>  Hello boys!
>>>
>>>     I have one problem after updating from openzap to freetdm. Now, whe
>>> FS receives 'Proceeding' from PRI sends a SIP 180 which makes the phone to
>>> play a fake ring tone. After some seconds, when FS receives 'Alerting' from
>>> PRI, FS sends a SIP 183 with the real ring tone.
>>>
>>>  Is there any way to avoid first SIP 180? If the calle is busy, I can
>>> hear a ringing tone between SIP 180 and SIP 183 and a busy tone after SIP
>>> 183. Same if calle is a mobile phone and it is out of signal.
>>>
>>>  Regards
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Mon Dec  6 01:13:57 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 5 Dec 2010 23:13:57 +0100
Subject: [Freeswitch-users] Asterisk SCF vs FreeSWITCH
In-Reply-To: <AANLkTik2+_aCu3eVmcvK8EJOxXB-mCFUtP9XTOD19umq@mail.gmail.com>
References: <AANLkTin9ne0BXr1aRpEk6jNysF=_LGmMSaCc3incu2i0@mail.gmail.com>
	<6D8FED3C-2F0D-4D67-A61C-FB312C34EDAF@ipeva.fr>
	<AANLkTik2+_aCu3eVmcvK8EJOxXB-mCFUtP9XTOD19umq@mail.gmail.com>
Message-ID: <FAA5C39B-45D1-488C-8661-8C12C4752107@ipeva.fr>

So SCF wants to be cloudy-cloudy ? :)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 05/12/2010 ? 23:08, Moises Silva a ?crit :

> 2010/12/3 David Ponzone <david.ponzone at ipeva.fr>
> Hmm I would say the main one is that FreeSWICH is there now, and has been around for some years now.
> Asterisk SCF is a project, not very advanced it seems.
> If you can afford to wait 2 or 3 years...:)
> 
> 
> Probably the most technically visibile difference (other than FreeSWITCH is here now) is that Asterisk SCF aims to follow a distributed model (components run in different processes or even different machines).
> 
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From william.suffill at gmail.com  Mon Dec  6 02:10:30 2010
From: william.suffill at gmail.com (William Suffill)
Date: Sun, 5 Dec 2010 18:10:30 -0500
Subject: [Freeswitch-users] Asterisk SCF vs FreeSWITCH
In-Reply-To: <FAA5C39B-45D1-488C-8661-8C12C4752107@ipeva.fr>
References: <AANLkTin9ne0BXr1aRpEk6jNysF=_LGmMSaCc3incu2i0@mail.gmail.com>
	<6D8FED3C-2F0D-4D67-A61C-FB312C34EDAF@ipeva.fr>
	<AANLkTik2+_aCu3eVmcvK8EJOxXB-mCFUtP9XTOD19umq@mail.gmail.com>
	<FAA5C39B-45D1-488C-8661-8C12C4752107@ipeva.fr>
Message-ID: <AANLkTikYrDa2Dnixcq9tDVp1UW-JWKAr2UYQcXZF234X@mail.gmail.com>

Also read up on the different software licenses as well since depending what
you have in mind that might come into play.

-- W
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From lakindia89 at gmail.com  Mon Dec  6 09:49:54 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 6 Dec 2010 12:19:54 +0530
Subject: [Freeswitch-users] how bind_meta_app works?
Message-ID: <AANLkTi=m0ww373iFn0nrZQ4hofyAp6duN=-Y4G-wyPgt@mail.gmail.com>

Hi all,
I was experimenting the bind_mea_app and I have a doubt.
There is a call from 1000 to FreeSwitch extension and it is connecting to
event outbound socket.
I've the following script in place.

#!/usr/bin/perl
use strict;
use warnings;
use Data::Dumper;
use lib '/usr/src/freeswitch/libs/esl/perl/';
use IO::Socket::INET;
use ESL;

my $SOCK = new IO::Socket::INET ( LocalHost => 'localhost',  LocalPort =>
'8447',  Proto => 'tcp',  Listen => 10,  Reuse => 1 );
unless ($SOCK)  {
                print("Could not create socket: $!");
                exit(2);
}

while (1)  {
                # Wait for any client through accept function in socket
module.
                my $new_sock = $SOCK->accept();
                next if (not defined($new_sock));
                # Get socket host
                my $host = $new_sock->sockhost();
                print("Got a client accepted from $host\n");
                my $pid = fork();
                if ($pid) {
                                close($new_sock);
                                next;
                }
                my $fd = fileno($new_sock);
                print("Newly forked child, pid: $$\n");
                my $EslCon = new ESL::ESLconnection($fd);
                print "Connection created successfully\n";
                my $info = $EslCon->getInfo();
                $EslCon->setEventLock("true");
                my $uuid = $info->getHeader("unique-id");
                print Dumper $info->serialize();
                $EslCon->execute("set","bind_meta_key=#");
                $EslCon->execute("bind_meta_app","1 b o
event::appli=testing");
                my $api =
$EslCon->execute("bridge","{ignore_early_media=true}freetdm/1/a/xxxxxxxxx");
}

The script called to the number that was dialed. In that number I pressed
#1, but the event was not sent,
and I assume that the bridge application is executing in the A leg ( is it
correct?? ) . So only after that bridge application gets completed the event
gets triggered out.

If I execute the application in the B leg, then it is working fine, since
there is no application that is running at that time.
Now I wanted to know if there is any way to run the application on A leg, I
need that application to be executed even the bridge is not completed.
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From abubacker at bksystems.co.in  Mon Dec  6 12:47:27 2010
From: abubacker at bksystems.co.in (abubacker)
Date: Mon, 06 Dec 2010 15:17:27 +0530
Subject: [Freeswitch-users] how bind_meta_app works?
In-Reply-To: <AANLkTi=m0ww373iFn0nrZQ4hofyAp6duN=-Y4G-wyPgt@mail.gmail.com>
References: <AANLkTi=m0ww373iFn0nrZQ4hofyAp6duN=-Y4G-wyPgt@mail.gmail.com>
Message-ID: <4CFCB12F.808@bksys.co.in>

On Monday 06 December 2010 12:19 PM, lakshmanan ganapathy wrote:
> Hi all,
> I was experimenting the bind_mea_app and I have a doubt.
> There is a call from 1000 to FreeSwitch extension and it is connecting 
> to event outbound socket.
> I've the following script in place.
>
> #!/usr/bin/perl
> use strict;
> use warnings;
> use Data::Dumper;
> use lib '/usr/src/freeswitch/libs/esl/perl/';
> use IO::Socket::INET;
> use ESL;
>
> my $SOCK = new IO::Socket::INET ( LocalHost => 'localhost',  LocalPort 
> => '8447',  Proto => 'tcp',  Listen => 10,  Reuse => 1 );
> unless ($SOCK)  {
>                 print("Could not create socket: $!");
>                 exit(2);
> }
>
> while (1)  {
>                 # Wait for any client through accept function in 
> socket module.
>                 my $new_sock = $SOCK->accept();
>                 next if (not defined($new_sock));
>                 # Get socket host
>                 my $host = $new_sock->sockhost();
>                 print("Got a client accepted from $host\n");
>                 my $pid = fork();
>                 if ($pid) {
>                                 close($new_sock);
>                                 next;
>                 }
>                 my $fd = fileno($new_sock);
>                 print("Newly forked child, pid: $$\n");
>                 my $EslCon = new ESL::ESLconnection($fd);
>                 print "Connection created successfully\n";
>                 my $info = $EslCon->getInfo();
>                 $EslCon->setEventLock("true");
>                 my $uuid = $info->getHeader("unique-id");
>                 print Dumper $info->serialize();
>                 $EslCon->execute("set","bind_meta_key=#");
>                 $EslCon->execute("bind_meta_app","1 b o 
> event::appli=testing");
>                 my $api = 
> $EslCon->execute("bridge","{ignore_early_media=true}freetdm/1/a/xxxxxxxxx");
> }
>
> The script called to the number that was dialed. In that number I 
> pressed #1, but the event was not sent,
> and I assume that the bridge application is executing in the A leg ( 
> is it correct?? ) . So only after that bridge application gets 
> completed the event gets triggered out.
>
> If I execute the application in the B leg, then it is working fine, 
> since there is no application that is running at that time.
> Now I wanted to know if there is any way to run the application on A 
> leg, I need that application to be executed even the bridge is not 
> completed.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
Try after running the answer application in leg A

-- 
Best regards,
N.Abubacker ,
Associate system engineer ,
bk systems pvt ltd ,
Ph : 9144-43902701

Disclaimer: http://www.bksystems.co.in/email-policy

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From lakindia89 at gmail.com  Mon Dec  6 13:10:30 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 6 Dec 2010 15:40:30 +0530
Subject: [Freeswitch-users] how bind_meta_app works?
In-Reply-To: <4CFCB12F.808@bksys.co.in>
References: <AANLkTi=m0ww373iFn0nrZQ4hofyAp6duN=-Y4G-wyPgt@mail.gmail.com>
	<4CFCB12F.808@bksys.co.in>
Message-ID: <AANLkTi=GsBcrYNJCTrHb8Daa+skPqSeK1A9SwL95Ba0_@mail.gmail.com>

The user can press #1 in any time of the call. Not necessarily after answer
and before bridge.


On Mon, Dec 6, 2010 at 3:17 PM, abubacker <abubacker at bksystems.co.in> wrote:

>  On Monday 06 December 2010 12:19 PM, lakshmanan ganapathy wrote:
>
> Hi all,
> I was experimenting the bind_mea_app and I have a doubt.
> There is a call from 1000 to FreeSwitch extension and it is connecting to
> event outbound socket.
> I've the following script in place.
>
> #!/usr/bin/perl
> use strict;
> use warnings;
> use Data::Dumper;
> use lib '/usr/src/freeswitch/libs/esl/perl/';
> use IO::Socket::INET;
> use ESL;
>
> my $SOCK = new IO::Socket::INET ( LocalHost => 'localhost',  LocalPort =>
> '8447',  Proto => 'tcp',  Listen => 10,  Reuse => 1 );
> unless ($SOCK)  {
>                 print("Could not create socket: $!");
>                 exit(2);
> }
>
> while (1)  {
>                 # Wait for any client through accept function in socket
> module.
>                 my $new_sock = $SOCK->accept();
>                 next if (not defined($new_sock));
>                 # Get socket host
>                 my $host = $new_sock->sockhost();
>                 print("Got a client accepted from $host\n");
>                 my $pid = fork();
>                 if ($pid) {
>                                 close($new_sock);
>                                 next;
>                 }
>                 my $fd = fileno($new_sock);
>                 print("Newly forked child, pid: $$\n");
>                 my $EslCon = new ESL::ESLconnection($fd);
>                 print "Connection created successfully\n";
>                 my $info = $EslCon->getInfo();
>                 $EslCon->setEventLock("true");
>                 my $uuid = $info->getHeader("unique-id");
>                 print Dumper $info->serialize();
>                 $EslCon->execute("set","bind_meta_key=#");
>                 $EslCon->execute("bind_meta_app","1 b o
> event::appli=testing");
>                 my $api =
> $EslCon->execute("bridge","{ignore_early_media=true}freetdm/1/a/xxxxxxxxx");
> }
>
> The script called to the number that was dialed. In that number I pressed
> #1, but the event was not sent,
> and I assume that the bridge application is executing in the A leg ( is it
> correct?? ) . So only after that bridge application gets completed the event
> gets triggered out.
>
> If I execute the application in the B leg, then it is working fine, since
> there is no application that is running at that time.
> Now I wanted to know if there is any way to run the application on A leg, I
> need that application to be executed even the bridge is not completed.
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>  Try after running the answer application in leg A
>
> --
> Best regards,
> N.Abubacker ,
> Associate system engineer ,
> bk systems pvt ltd ,
> Ph : 9144-43902701
>
> Disclaimer: http://www.bksystems.co.in/email-policy
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From srinivas.ksvreddy at gmail.com  Mon Dec  6 14:43:32 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 6 Dec 2010 17:13:32 +0530
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 54, Issue 34
In-Reply-To: <mailman.58.1291533815.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.58.1291533815.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <AANLkTim7LR9Pru3_doJ0yBsARkDW09d_S_HKwaps00Rv@mail.gmail.com>

HI David,

Thanks for the replay, when we trying to build mono_managed using linux in
CentOS We are getting the follwing error.

/home/hydadmin/freeSwitch/freeswitch-1.0.6/src/mod/languages/mod_managed/mod_managed.cpp:
In function 'switch_status_t loadRuntime()':
/home/hydadmin/freeSwitch/freeswitch-1.0.6/src/mod/languages/mod_managed/mod_managed.cpp:211:
error: aggregate 'MonoAssemblyName name' has incomplete type and cannot be
defined
make[1]: *** [mod_managed.lo] Error 1

I enable the module mod_managed in modules.conf.xml,  Can you please let me
know if you have any idea?


Thanks
Sriniavas


2010/12/5 <freeswitch-users-request at lists.freeswitch.org>

> Send FreeSWITCH-users mailing list submissions to
>        freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
>        freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
>        freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
>   1. Re: Error loading mod_managed in CentOS (David Brazier)
>   2. Re: building mod_managed fails (David Brazier)
>   3. Re: Problem with SIP 180 + FreeTDM (Juan Antonio Iba?ez Santorum)
>   4. GIT/SVN Repo Maint. (Brian West)
>   5. Possible Loopback / Originate Race? (Jeroen C. van Gelderen)
>
>
> ---------- Forwarded message ----------
> From: David Brazier <davidjbrazier at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Sat, 4 Dec 2010 00:39:25 +0000
> Subject: Re: [Freeswitch-users] Error loading mod_managed in CentOS
> Make sure you uncomment "languages/mod_managed" is uncommented in
> modules.conf and follow the Mono 2.8 instructions on
>
> http://wiki.freeswitch.org/wiki/Mod_managed
>
> Don't try mod_mono & mod_mono_managed - that is old stuff.
>
> David
>
>
>
>
> ---------- Forwarded message ----------
> From: David Brazier <davidjbrazier at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Sat, 4 Dec 2010 00:43:43 +0000
> Subject: Re: [Freeswitch-users] building mod_managed fails
> On Wed, Dec 1, 2010 at 5:16 PM, David Brazier <davidjbrazier at gmail.com>
> wrote:
> > I'll try to investigate why the swig output wasn't right.
>
> I found out why - you need SWIG 2.0 - SWIG 1.3 that I had doesn't do it
> right.
>
>
>
>
> ---------- Forwarded message ----------
> From: "Juan Antonio Iba?ez Santorum" <juanito1982 at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Sat, 4 Dec 2010 14:24:06 +0100
> Subject: Re: [Freeswitch-users] Problem with SIP 180 + FreeTDM
> You are true. Last git version works as expected.
>
> Thank you very much
>
> 2010/12/2 David Yat Sin <dyatsin at sangoma.com>
>
>>  Hi Juan Antonio,
>>
>> Can you pull the latest Freeswitch git. This should be fixed now.
>>
>>
>> David
>>
>>   *David Yat Sin, BEng*
>> Software Engineer
>> Sangoma Technologies
>> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
>>
>> t.   +1 800 388 2475 x119
>> t.   +1 905 474 1990 x119
>> f.   +1 905 474 9223 [image: SANGOMA]<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>> Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>
>> On 11/24/2010 1:38 PM, Juan Antonio Iba?ez Santorum wrote:
>>
>> Any news?
>>
>> 2010/11/18 David Yat Sin <dyatsin at sangoma.com>
>>
>>>  Hi Juan Antonio,
>>> We are looking into this and will get back to you in a day or two.
>>>
>>> David
>>>
>>>   *David Yat Sin, BEng*
>>> Software Engineer
>>> Sangoma Technologies
>>> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
>>>
>>> t.   +1 800 388 2475 x119
>>> t.   +1 905 474 1990 x119
>>> f.   +1 905 474 9223 [image: SANGOMA]<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>>
>>> On 11/18/2010 3:50 PM, Juan Antonio Iba?ez Santorum wrote:
>>>
>>>  Hello boys!
>>>
>>>     I have one problem after updating from openzap to freetdm. Now, whe
>>> FS receives 'Proceeding' from PRI sends a SIP 180 which makes the phone to
>>> play a fake ring tone. After some seconds, when FS receives 'Alerting' from
>>> PRI, FS sends a SIP 183 with the real ring tone.
>>>
>>>  Is there any way to avoid first SIP 180? If the calle is busy, I can
>>> hear a ringing tone between SIP 180 and SIP 183 and a busy tone after SIP
>>> 183. Same if calle is a mobile phone and it is out of signal.
>>>
>>>  Regards
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> ---------- Forwarded message ----------
> From: Brian West <brian at freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Sat, 4 Dec 2010 17:35:02 -0600
> Subject: [Freeswitch-users] GIT/SVN Repo Maint.
> I'm going to be doing some work on the repo to move it this weekend so
> please be patient if you can't access it while I'm moving it to Dallas
> Sunday December 5th 2010.
>
> Thanks,
> Brian
>
>
>
>
>
> ---------- Forwarded message ----------
> From: "Jeroen C. van Gelderen" <slim at thegreek.com>
> To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org>
> Date: Sun, 5 Dec 2010 02:23:33 -0500
> Subject: [Freeswitch-users] Possible Loopback / Originate Race?
>
> Hi,
>
>
>
> (Filed in JIRA FS-2899. Please let me know if posting this on the list as
> well is frowned upon.)
>
>
>
> I?m using switch_limit in a loopback extension with the following
> problematic dialplan fragment (full dialplan context at end of message):
>
>
>
>     <extension name="limit_exceeded">
>
>       <condition field="destination_number" expression="^limit_exceeded$">
>
>
>
>         <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
>
>
>
>       </condition>
>
>     </extension>
>
>
>
> This works fine in MOST cases with the expected result:
>
>   switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 34
> [NORMAL_CIRCUIT_CONGESTION]
>
>
>
> Every once in a while though this very same extension will magically yield:
>
>   switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 19
> [NO_ANSWER]
>
>
>
> Looking at the log snippets pasted below the correct result (34) appears
> when this sequence of events occurs (first snippet):
>
>
>
>   mod_loopback.c:425 Hangup loopback/18884447693-a [CS_CONSUME_MEDIA]
> [NORMAL_CIRCUIT_CONGESTION]
>
>   switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 34
> [NORMAL_CIRCUIT_CONGESTION]
>
>
>
> The incorrect answer (19) appears whenever this sequence shows in the logs:
>
>
>
>   switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 19
> [NO_ANSWER]
>
>   mod_loopback.c:425 Hangup loopback/18884447701-a [CS_CONSUME_MEDIA]
> [NORMAL_CIRCUIT_CONGESTION]
>
>
>
> Pouring over my logs this ALWAYS is the case. Whenever the wrong error
> cause appears the switch_ivr_originate call precedes the mod_loopback call.
> Conversely whenever the correct error cause is given the call sequence is
> reversed.
>
>
>
> I can reproduce this at will by rapidly invoking the loopback extension in
> the dialplan pasted at the bottom of this mail.
>
>
>
> Any help is appreciated.
>
>
>
> Cheers,
>
> -Slim
>
>
>
> ----8<----8<----8<----8<----8<----8<----8<----
>
> EXECUTE loopback/18884447693-b hangup(NORMAL_CIRCUIT_CONGESTION)
>
> 2010-12-05 01:27:13.318020 [DEBUG] switch_channel.c:2455
> (loopback/18884447693-b) Callstate Change EARLY -> HANGUP
>
> 2010-12-05 01:27:13.318020 [NOTICE] mod_dptools.c:906 Hangup
> loopback/18884447693-b [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION]
>
>
>
> 2010-12-05 01:27:13.319023 [DEBUG] switch_channel.c:2471 Send signal
> loopback/18884447693-b [KILL]
>
> 2010-12-05 01:27:13.319023 [DEBUG] mod_loopback.c:468
> loopback/18884447693-b CHANNEL KILL
>
> 2010-12-05 01:27:13.319023 [DEBUG] switch_core_session.c:1083 Send signal
> loopback/18884447693-b [BREAK]
>
> 2010-12-05 01:27:13.319023 [DEBUG] mod_loopback.c:468
> loopback/18884447693-b CHANNEL KILL
>
>
>
> 2010-12-05 01:27:13.320024 [DEBUG] switch_core_session.c:1993
> loopback/18884447693-b skip receive message [APPLICATION_EXEC_COMPLETE]
> (chann
>
> el is hungup already)
>
>
>
> 2010-12-05 01:27:13.320024 [DEBUG] switch_core_state_machine.c:366
> (loopback/18884447693-b) State EXECUTE going to sleep
>
> 2010-12-05 01:27:13.320024 [DEBUG] switch_core_state_machine.c:320
> (loopback/18884447693-b) Running State Change CS_HANGUP
>
> 2010-12-05 01:27:13.321011 [DEBUG] switch_core_state_machine.c:557
> (loopback/18884447693-b) State HANGUP
>
>
>
>
>
> 2010-12-05 01:27:13.321983 [DEBUG] mod_loopback.c:414
> loopback/18884447693-b CHANNEL HANGUP
>
> 2010-12-05 01:27:13.321983 [DEBUG] switch_channel.c:2455
> (loopback/18884447693-a) Callstate Change EARLY -> HANGUP
>
>
>
> 2010-12-05 01:27:13.321983 [NOTICE] mod_loopback.c:425 Hangup
> loopback/18884447693-a [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION]
>
>
>
> 2010-12-05 01:27:13.322995 [DEBUG] switch_ivr_originate.c:3448 Originate
> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>
>
>
> ----8<----8<----8<----8<----8<----8<----8<----
>
>
>
> EXECUTE loopback/18884447701-b hangup(NORMAL_CIRCUIT_CONGESTION)
>
> 2010-12-05 01:27:14.364448 [DEBUG] switch_channel.c:2455
> (loopback/18884447701-b) Callstate Change EARLY -> HANGUP
>
> 2010-12-05 01:27:14.365470 [NOTICE] mod_dptools.c:906 Hangup
> loopback/18884447701-b [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION]
>
>
>
> 2010-12-05 01:27:14.365470 [DEBUG] switch_channel.c:2471 Send signal
> loopback/18884447701-b [KILL]
>
> 2010-12-05 01:27:14.365470 [DEBUG] mod_loopback.c:468
> loopback/18884447701-b CHANNEL KILL
>
> 2010-12-05 01:27:14.365470 [DEBUG] switch_core_session.c:1083 Send signal
> loopback/18884447701-b [BREAK]
>
> 2010-12-05 01:27:14.366485 [DEBUG] mod_loopback.c:468
> loopback/18884447701-b CHANNEL KILL
>
>
>
> 2010-12-05 01:27:14.366485 [DEBUG] switch_core_session.c:1993
> loopback/18884447701-b skip receive message [APPLICATION_EXEC_COMPLETE]
> (chann
>
> el is hungup already)
>
>
>
> 2010-12-05 01:27:14.366485 [DEBUG] switch_core_state_machine.c:366
> (loopback/18884447701-b) State EXECUTE going to sleep
>
> 2010-12-05 01:27:14.367508 [DEBUG] switch_core_state_machine.c:320
> (loopback/18884447701-b) Running State Change CS_HANGUP
>
> 2010-12-05 01:27:14.369585 [DEBUG] switch_core_state_machine.c:557
> (loopback/18884447701-b) State HANGUP
>
>
>
> 2010-12-05 01:27:14.369585 [DEBUG] mod_loopback.c:414
> loopback/18884447701-b CHANNEL HANGUP
>
> 2010-12-05 01:27:14.369585 [DEBUG] switch_channel.c:2455
> (loopback/18884447701-a) Callstate Change EARLY -> HANGUP
>
>
>
> 2010-12-05 01:27:14.369585 [DEBUG] switch_ivr_originate.c:3448 Originate
> Resulted in Error Cause: 19 [NO_ANSWER]
>
>
>
> 2010-12-05 01:27:14.369585 [NOTICE] mod_loopback.c:425 Hangup
> loopback/18884447701-a [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION]
>
>
>
> ----8<----8<----8<----8<----8<----8<----8<----
>
>
>
>
>
>   <context name="tech-outbound">
>
>
>
>     <extension name="tech-outbound">
>
>       <condition field="destination_number" expression="^1888(\d{7})$">
>
>
>
>         <action application="info"/>
>
>         <action application="set" data="loopback_bowout=false"/>
>
>         <action application="limit" data="db outgoing goip 1" />
>
>         <action application="bridge"
> data="{ignore_early_media=true}sofia/internal/${
> destination_number}@192.168.3.11:5060<http://destination_number%7D%40192.168.3.11%3A5060>
> "/>
>
>
>
>       </condition>
>
>     </extension>
>
>
>
>     <extension name="limit_exceeded">
>
>       <condition field="destination_number" expression="^limit_exceeded$">
>
>
>
>         <action application="<nothing> OR <answer> OR <pre_answer>"/> <!--
> Tried all 3 variations, no difference -->
>
>
>
>         <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
>
>
>
>       </condition>
>
>     </extension>
>
>
>
>   </context>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Srinivasula Reddy K
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From enp at altlinux.org  Mon Dec  6 14:55:36 2010
From: enp at altlinux.org (Eugene Prokopiev)
Date: Mon, 6 Dec 2010 14:55:36 +0300
Subject: [Freeswitch-users] mod_fifo vs mod_callcenter
In-Reply-To: <AANLkTikWttM3het5NTO7YJSu=0xT84=O9-4rpvgzNhdC@mail.gmail.com>
References: <AANLkTimftpTpLLYXED0Fk1=NwVowzxv=8T0ufQBkrAZ8@mail.gmail.com>
	<AANLkTinC85pzthxCVcjopW9q=cjBkkjS0jXGBhKBjyHV@mail.gmail.com>
	<AANLkTikWttM3het5NTO7YJSu=0xT84=O9-4rpvgzNhdC@mail.gmail.com>
Message-ID: <AANLkTinc1RBRgYFn7K0HshRsRXXBf2fHA84QTMkxYyvN@mail.gmail.com>

> Is it possible to implement the same logic for off-hook members? Can
> you point me the line/function in mod_fifo.c in which off-hook member
> is selecting for call?

I tried to debug callflow with help of switch_log_printf. Two off-hook
agents executes:

<action application="fifo" data="$1 out wait"/>

and waits in:

mod_fifo.c:2588 - moh_status = switch_ivr_play_file(session, NULL, moh, &args);

After executing:

<action application="fifo" data="$1 in"/>

only one off-hook agent leaves switch_ivr_play_file and process
incoming call. Why can it leaves switch_ivr_play_file function?

--
Thanks,
Eugene Prokopiev


From srinivas.ksvreddy at gmail.com  Mon Dec  6 17:23:41 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 6 Dec 2010 19:53:41 +0530
Subject: [Freeswitch-users] managed code in CentOS
Message-ID: <AANLkTi=en-78jt7LT4vKOdFS-6W28cMqPBFW26sXhvdn@mail.gmail.com>

HI All,

We are trying to migrate from windows to linux, if we are trying to build
managed code in centos, we are facing few issues,

/home/hydadmin/freeSwitch/
freeswitch-1.0.6/src/mod/languages/mod_managed/mod_managed.cpp:211: error:
aggregate 'MonoAssemblyName name' has incomplete type and cannot be defined
make[1]: *** [mod_managed.lo] Error 1

we enabled the mod_managed module in modules_conf.xml, i have installed
mono2.8.1,

following instruction given by freeswitch, they are asking us to apply
mono28.patch,  we could not able to get that patch,  if mono28.patch is the
problem for the above error?

if that is the error where will we get that patch?

if any other reason please let us know.

THanks
Srinivas
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From brian at freeswitch.org  Mon Dec  6 17:57:08 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 6 Dec 2010 08:57:08 -0600
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 54, Issue 34
In-Reply-To: <AANLkTim7LR9Pru3_doJ0yBsARkDW09d_S_HKwaps00Rv@mail.gmail.com>
References: <mailman.58.1291533815.24555.freeswitch-users@lists.freeswitch.org>
	<AANLkTim7LR9Pru3_doJ0yBsARkDW09d_S_HKwaps00Rv@mail.gmail.com>
Message-ID: <94CE349F-2528-49F4-8CF9-10F0DCD457D8@freeswitch.org>

You might want to try git head.... and please subscribe non-digest if you plan on responding to emails please.

/b

On Dec 6, 2010, at 5:43 AM, srinivasula reddy wrote:

> HI David,
> 
> Thanks for the replay, when we trying to build mono_managed using linux in CentOS We are getting the follwing error.
> 
> /home/hydadmin/freeSwitch/freeswitch-1.0.6/src/mod/languages/mod_managed/mod_managed.cpp: In function 'switch_status_t loadRuntime()':
> /home/hydadmin/freeSwitch/freeswitch-1.0.6/src/mod/languages/mod_managed/mod_managed.cpp:211: error: aggregate 'MonoAssemblyName name' has incomplete type and cannot be defined
> make[1]: *** [mod_managed.lo] Error 1
> 
> I enable the module mod_managed in modules.conf.xml,  Can you please let me know if you have any idea?
> 
> 
> Thanks
> Sriniavas



From jmmbuthia at gmail.com  Mon Dec  6 17:49:33 2010
From: jmmbuthia at gmail.com (James Mbuthia)
Date: Mon, 6 Dec 2010 16:49:33 +0200
Subject: [Freeswitch-users] SIP Phone Integration
In-Reply-To: <AANLkTikQrkp43CetufJsBDO+70metjjh9in7ePOuQnU_@mail.gmail.com>
References: <AANLkTi=CJYQ2g6NO+YfnPz8wRkX79_xjBc3RBRfM+xEs@mail.gmail.com>
	<AANLkTikQrkp43CetufJsBDO+70metjjh9in7ePOuQnU_@mail.gmail.com>
Message-ID: <AANLkTim=GLEN9tvam461WB7A3KjW0qNm_9hXKOuC7Sq4@mail.gmail.com>

Hi guys,

Am new to Freeswitch and am looking for info that can help me develop a
web-based SIP Phone

 I am developing a web-based SIP Phone based on PHP/MySQL and OpenSIPS.
I have developed the component of the SIP phone responsible for the
3way handshake and SDP offer.

My challenge is to now integrate the application to a rtp stack which will
enable the app to pick up audio and transmit it over the internet.
Ultimately I want to connect the app to the PSTN using a media server such
as Freeswitch or Asterisk. I want to use the Speex codec. I wanted to know
whether Freeswitch has a rtp stack integrated with speex and whether there
any tutorials on how integration to a softphone can be done. Any pointers or
ideas from you would be very helpful and highly appreaciated.

regards,
James Mbuthia
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From brian at freeswitch.org  Mon Dec  6 18:06:00 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 6 Dec 2010 09:06:00 -0600
Subject: [Freeswitch-users] managed code in CentOS
In-Reply-To: <AANLkTi=en-78jt7LT4vKOdFS-6W28cMqPBFW26sXhvdn@mail.gmail.com>
References: <AANLkTi=en-78jt7LT4vKOdFS-6W28cMqPBFW26sXhvdn@mail.gmail.com>
Message-ID: <5C25E1D1-1E15-4506-9733-C7307BF41CD7@freeswitch.org>

You have already sent the message once to the list and I replied please be patient when posting to the list if you do not get immediate responses to your questions.

thanks,
/b

On Dec 6, 2010, at 8:23 AM, srinivasula reddy wrote:

> 
> HI All,
> 
> We are trying to migrate from windows to linux, if we are trying to build managed code in centos, we are facing few issues, 
> 
> /home/hydadmin/freeSwitch/
> freeswitch-1.0.6/src/mod/languages/mod_managed/mod_managed.cpp:211: error: aggregate 'MonoAssemblyName name' has incomplete type and cannot be defined
> make[1]: *** [mod_managed.lo] Error 1
> 
> we enabled the mod_managed module in modules_conf.xml, i have installed mono2.8.1,
> 
> following instruction given by freeswitch, they are asking us to apply mono28.patch,  we could not able to get that patch,  if mono28.patch is the problem for the above error?
> 
> if that is the error where will we get that patch?
> 
> if any other reason please let us know. 
> 
> THanks
> Srinivas
> 



From infos at madovsky.org  Mon Dec  6 18:17:44 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 6 Dec 2010 10:17:44 -0500
Subject: [Freeswitch-users] SIP Phone Integration
References: <AANLkTi=CJYQ2g6NO+YfnPz8wRkX79_xjBc3RBRfM+xEs@mail.gmail.com><AANLkTikQrkp43CetufJsBDO+70metjjh9in7ePOuQnU_@mail.gmail.com>
	<AANLkTim=GLEN9tvam461WB7A3KjW0qNm_9hXKOuC7Sq4@mail.gmail.com>
Message-ID: <3D0A80D2CE2244A7A28E94508390CEB7@e1705>

You can transcode with FS, 
I did already a web phone (boophone) with FS but ;keep in mind
that transcode=CPU
  ----- Original Message ----- 
  From: James Mbuthia 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, December 06, 2010 9:49 AM
  Subject: [Freeswitch-users] SIP Phone Integration


  Hi guys,

  Am new to Freeswitch and am looking for info that can help me develop a web-based SIP Phone

   I am developing a web-based SIP Phone based on PHP/MySQL and OpenSIPS. I have developed the component of the SIP phone responsible for the 3way handshake and SDP offer. 


  My challenge is to now integrate the application to a rtp stack which will enable the app to pick up audio and transmit it over the internet. Ultimately I want to connect the app to the PSTN using a media server such as Freeswitch or Asterisk. I want to use the Speex codec. I wanted to know whether Freeswitch has a rtp stack integrated with speex and whether there any tutorials on how integration to a softphone can be done. Any pointers or ideas from you would be very helpful and highly appreaciated.


  regards,
  James Mbuthia









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


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From juanito1982 at gmail.com  Mon Dec  6 18:24:18 2010
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Mon, 6 Dec 2010 16:24:18 +0100
Subject: [Freeswitch-users] Problem with SIP 180 + FreeTDM
In-Reply-To: <AANLkTimAWtbv-TEwcV_c_zqxh6yH2MqyFOhZASm1OYTE@mail.gmail.com>
References: <AANLkTi=3RuKF9YzGmvz=Xcga4zdH6_eOod7H0XMd=Kyn@mail.gmail.com>
	<4CE5A68B.2060507@sangoma.com>
	<AANLkTi=24KMXRz1FMi7nLTPkjFj2nzqFycD8ZPZ2b+g9@mail.gmail.com>
	<4CF7F784.7090201@sangoma.com>
	<AANLkTimsYTnJbUA=S-Mgb95=6BEWpNruUuLvJbe-2u3Q@mail.gmail.com>
	<AANLkTimAWtbv-TEwcV_c_zqxh6yH2MqyFOhZASm1OYTE@mail.gmail.com>
Message-ID: <AANLkTimiYyxseAYnVLD-E7Nb9zHCKAXYV4r0NSzg4Gpk@mail.gmail.com>

Don't worry about it Moises. I'll tell you all issues I could find...



2010/12/5 Moises Silva <moises.silva at gmail.com>

> 2010/12/4 Juan Antonio Iba?ez Santorum <juanito1982 at gmail.com>
>
> You are true. Last git version works as expected.
>>
>> Thank you very much
>>
>>
> We're very interested in hearing about this kind of inter-networking issues
> between SIP and the PSTN, if you find any other glitch, let us know.
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R
> 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
>
>
>
>> 2010/12/2 David Yat Sin <dyatsin at sangoma.com>
>>
>>  Hi Juan Antonio,
>>>
>>> Can you pull the latest Freeswitch git. This should be fixed now.
>>>
>>>
>>> David
>>>
>>>   *David Yat Sin, BEng*
>>> Software Engineer
>>> Sangoma Technologies
>>> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
>>>
>>> t.   +1 800 388 2475 x119
>>> t.   +1 905 474 1990 x119
>>> f.   +1 905 474 9223 [image: SANGOMA]<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>> Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>>
>>> On 11/24/2010 1:38 PM, Juan Antonio Iba?ez Santorum wrote:
>>>
>>> Any news?
>>>
>>> 2010/11/18 David Yat Sin <dyatsin at sangoma.com>
>>>
>>>>  Hi Juan Antonio,
>>>> We are looking into this and will get back to you in a day or two.
>>>>
>>>> David
>>>>
>>>>   *David Yat Sin, BEng*
>>>> Software Engineer
>>>> Sangoma Technologies
>>>> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
>>>>
>>>> t.   +1 800 388 2475 x119
>>>> t.   +1 905 474 1990 x119
>>>> f.   +1 905 474 9223 [image: SANGOMA]<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>>> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>>> Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>>> Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>>> Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>>> Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>>> Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>|
>>>> Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>>>
>>>> On 11/18/2010 3:50 PM, Juan Antonio Iba?ez Santorum wrote:
>>>>
>>>>  Hello boys!
>>>>
>>>>     I have one problem after updating from openzap to freetdm. Now, whe
>>>> FS receives 'Proceeding' from PRI sends a SIP 180 which makes the phone to
>>>> play a fake ring tone. After some seconds, when FS receives 'Alerting' from
>>>> PRI, FS sends a SIP 183 with the real ring tone.
>>>>
>>>>  Is there any way to avoid first SIP 180? If the calle is busy, I can
>>>> hear a ringing tone between SIP 180 and SIP 183 and a busy tone after SIP
>>>> 183. Same if calle is a mobile phone and it is out of signal.
>>>>
>>>>  Regards
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vipkilla at gmail.com  Mon Dec  6 18:47:54 2010
From: vipkilla at gmail.com (vip killa)
Date: Mon, 6 Dec 2010 10:47:54 -0500
Subject: [Freeswitch-users] SIP Phone Integration
In-Reply-To: <AANLkTim=GLEN9tvam461WB7A3KjW0qNm_9hXKOuC7Sq4@mail.gmail.com>
References: <AANLkTi=CJYQ2g6NO+YfnPz8wRkX79_xjBc3RBRfM+xEs@mail.gmail.com>
	<AANLkTikQrkp43CetufJsBDO+70metjjh9in7ePOuQnU_@mail.gmail.com>
	<AANLkTim=GLEN9tvam461WB7A3KjW0qNm_9hXKOuC7Sq4@mail.gmail.com>
Message-ID: <AANLkTim2TCSbmw3Vi94Vr-+pZ4a6xW10y8g+Bh1PJ=rO@mail.gmail.com>

already been done, check out red5phone. i have it working w/ FS and red5
server

On Mon, Dec 6, 2010 at 9:49 AM, James Mbuthia <jmmbuthia at gmail.com> wrote:

> Hi guys,
>
> Am new to Freeswitch and am looking for info that can help me develop a
> web-based SIP Phone
>
>  I am developing a web-based SIP Phone based on PHP/MySQL and OpenSIPS.
> I have developed the component of the SIP phone responsible for the
> 3way handshake and SDP offer.
>
> My challenge is to now integrate the application to a rtp stack which will
> enable the app to pick up audio and transmit it over the internet.
> Ultimately I want to connect the app to the PSTN using a media server such
> as Freeswitch or Asterisk. I want to use the Speex codec. I wanted to know
> whether Freeswitch has a rtp stack integrated with speex and whether there
> any tutorials on how integration to a softphone can be done. Any pointers or
> ideas from you would be very helpful and highly appreaciated.
>
> regards,
> James Mbuthia
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From adminjew at gmail.com  Mon Dec  6 19:58:35 2010
From: adminjew at gmail.com (Yitzchok)
Date: Mon, 6 Dec 2010 11:58:35 -0500
Subject: [Freeswitch-users] managed code in CentOS
In-Reply-To: <AANLkTi=en-78jt7LT4vKOdFS-6W28cMqPBFW26sXhvdn@mail.gmail.com>
References: <AANLkTi=en-78jt7LT4vKOdFS-6W28cMqPBFW26sXhvdn@mail.gmail.com>
Message-ID: <AANLkTik98HSrO3SV8L41tsVTiqrh129OLeh=8tatT6WU@mail.gmail.com>

If you get the latest files from git you should have the mono28.patch file
in freeswitch/src/mod/languages/mod_managed

Just run
cd src/mod/languages/mod_managed/
git apply mono28.patch
make reswig
make


Yitzchok


On Mon, Dec 6, 2010 at 9:23 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

>
> HI All,
>
> We are trying to migrate from windows to linux, if we are trying to build
> managed code in centos, we are facing few issues,
>
> /home/hydadmin/freeSwitch/
> freeswitch-1.0.6/src/mod/languages/mod_managed/mod_managed.cpp:211: error:
> aggregate 'MonoAssemblyName name' has incomplete type and cannot be defined
> make[1]: *** [mod_managed.lo] Error 1
>
> we enabled the mod_managed module in modules_conf.xml, i have installed
> mono2.8.1,
>
> following instruction given by freeswitch, they are asking us to apply
> mono28.patch,  we could not able to get that patch,  if mono28.patch is the
> problem for the above error?
>
> if that is the error where will we get that patch?
>
> if any other reason please let us know.
>
> THanks
> Srinivas
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From grw.freeswitch at gmail.com  Mon Dec  6 19:32:53 2010
From: grw.freeswitch at gmail.com (Geovani Ricardo Wiedenhoft)
Date: Mon, 6 Dec 2010 14:32:53 -0200
Subject: [Freeswitch-users] mod_khomp
Message-ID: <AANLkTi=zaMD1NsXJ_pEz8h=wCg0VT6bYcFWAz=_edfeb@mail.gmail.com>

The Endpoint and the documentation for the Khomp boards (mod_khomp) are
available on the official FreeSWTICH branch. Also, the documents are
available on the wiki.

http://wiki.freeswitch.org/wiki/Khomp

The version is compatible with all Khomp boards (SPX series):

- FXS
- FXO
- E1 - R2, R2 DTMF and OpenCAS in Hardware, ISDN (User, Network), OpenCCS,
LineSide, CAS_EL7, E1LC
- GSM with send/receive SMS (boards and usb devices)
- Passive record (R2, ISDN, OpenCAS, OpenCCS and FXO)
- kommuter

Visit our site to more information about Khomp products:

http://www.khomp.com.br



Thank you.

:)

Khomp development team
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From jmmbuthia at gmail.com  Mon Dec  6 18:23:53 2010
From: jmmbuthia at gmail.com (James Mbuthia)
Date: Mon, 6 Dec 2010 17:23:53 +0200
Subject: [Freeswitch-users] SIP Phone Integration
In-Reply-To: <3D0A80D2CE2244A7A28E94508390CEB7@e1705>
References: <AANLkTi=CJYQ2g6NO+YfnPz8wRkX79_xjBc3RBRfM+xEs@mail.gmail.com>
	<AANLkTikQrkp43CetufJsBDO+70metjjh9in7ePOuQnU_@mail.gmail.com>
	<AANLkTim=GLEN9tvam461WB7A3KjW0qNm_9hXKOuC7Sq4@mail.gmail.com>
	<3D0A80D2CE2244A7A28E94508390CEB7@e1705>
Message-ID: <AANLkTim_o3YdYob+3RsdEOSSpN4XZ5Dam1zv6mtVp_mG@mail.gmail.com>

Thanks,  about how many concurrent calls are you able to when transcoding
with FS?

On Mon, Dec 6, 2010 at 5:17 PM, Madovsky <infos at madovsky.org> wrote:

>  You can transcode with FS,
> I did already a web phone (boophone) with FS but ;keep in mind
> that transcode=CPU
>
> ----- Original Message -----
> *From:* James Mbuthia <jmmbuthia at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, December 06, 2010 9:49 AM
> *Subject:* [Freeswitch-users] SIP Phone Integration
>
>    Hi guys,
>
> Am new to Freeswitch and am looking for info that can help me develop a
> web-based SIP Phone
>
>  I am developing a web-based SIP Phone based on PHP/MySQL and OpenSIPS.
> I have developed the component of the SIP phone responsible for the
> 3way handshake and SDP offer.
>
> My challenge is to now integrate the application to a rtp stack which will
> enable the app to pick up audio and transmit it over the internet.
> Ultimately I want to connect the app to the PSTN using a media server such
> as Freeswitch or Asterisk. I want to use the Speex codec. I wanted to know
> whether Freeswitch has a rtp stack integrated with speex and whether there
> any tutorials on how integration to a softphone can be done. Any pointers or
> ideas from you would be very helpful and highly appreaciated.
>
> regards,
> James Mbuthia
>
>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Mon Dec  6 20:36:09 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 6 Dec 2010 12:36:09 -0500
Subject: [Freeswitch-users] SIP Phone Integration
References: <AANLkTi=CJYQ2g6NO+YfnPz8wRkX79_xjBc3RBRfM+xEs@mail.gmail.com><AANLkTikQrkp43CetufJsBDO+70metjjh9in7ePOuQnU_@mail.gmail.com><AANLkTim=GLEN9tvam461WB7A3KjW0qNm_9hXKOuC7Sq4@mail.gmail.com><3D0A80D2CE2244A7A28E94508390CEB7@e1705>
	<AANLkTim_o3YdYob+3RsdEOSSpN4XZ5Dam1zv6mtVp_mG@mail.gmail.com>
Message-ID: <AACF8843B59E4A2C951F071534D0CB25@e1705>

depend the transcoding, from 10 to 20


  ----- Original Message ----- 
  From: James Mbuthia 
  To: FreeSWITCH Users Help 
  Sent: Monday, December 06, 2010 10:23 AM
  Subject: Re: [Freeswitch-users] SIP Phone Integration


  Thanks,  about how many concurrent calls are you able to when transcoding with FS?


  On Mon, Dec 6, 2010 at 5:17 PM, Madovsky <infos at madovsky.org> wrote:

    You can transcode with FS, 
    I did already a web phone (boophone) with FS but ;keep in mind
    that transcode=CPU
      ----- Original Message ----- 
      From: James Mbuthia 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, December 06, 2010 9:49 AM
      Subject: [Freeswitch-users] SIP Phone Integration


      Hi guys,

      Am new to Freeswitch and am looking for info that can help me develop a web-based SIP Phone

       I am developing a web-based SIP Phone based on PHP/MySQL and OpenSIPS. I have developed the component of the SIP phone responsible for the 3way handshake and SDP offer. 


      My challenge is to now integrate the application to a rtp stack which will enable the app to pick up audio and transmit it over the internet. Ultimately I want to connect the app to the PSTN using a media server such as Freeswitch or Asterisk. I want to use the Speex codec. I wanted to know whether Freeswitch has a rtp stack integrated with speex and whether there any tutorials on how integration to a softphone can be done. Any pointers or ideas from you would be very helpful and highly appreaciated.


      regards,
      James Mbuthia









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


      _______________________________________________
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      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org



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






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


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From david.ponzone at ipeva.fr  Mon Dec  6 20:56:15 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 6 Dec 2010 18:56:15 +0100
Subject: [Freeswitch-users] SIP Phone Integration
In-Reply-To: <AANLkTim_o3YdYob+3RsdEOSSpN4XZ5Dam1zv6mtVp_mG@mail.gmail.com>
References: <AANLkTi=CJYQ2g6NO+YfnPz8wRkX79_xjBc3RBRfM+xEs@mail.gmail.com>
	<AANLkTikQrkp43CetufJsBDO+70metjjh9in7ePOuQnU_@mail.gmail.com>
	<AANLkTim=GLEN9tvam461WB7A3KjW0qNm_9hXKOuC7Sq4@mail.gmail.com>
	<3D0A80D2CE2244A7A28E94508390CEB7@e1705>
	<AANLkTim_o3YdYob+3RsdEOSSpN4XZ5Dam1zv6mtVp_mG@mail.gmail.com>
Message-ID: <4DAAA038-8EDC-4262-8B15-26AB77D8ACAA@ipeva.fr>

If you choose another codec, you could use a HW-based transcoding solution like the Sangoma D100/500, which goes from 30 to 2000 concurrent calls.
According their website, they don't support Speex.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 06/12/2010 ? 16:23, James Mbuthia a ?crit :

> Thanks,  about how many concurrent calls are you able to when transcoding with FS?
> 
> On Mon, Dec 6, 2010 at 5:17 PM, Madovsky <infos at madovsky.org> wrote:
> You can transcode with FS,
> I did already a web phone (boophone) with FS but ;keep in mind
> that transcode=CPU
> ----- Original Message -----
> From: James Mbuthia
> To: freeswitch-users at lists.freeswitch.org
> Sent: Monday, December 06, 2010 9:49 AM
> Subject: [Freeswitch-users] SIP Phone Integration
> 
> Hi guys,
> 
> Am new to Freeswitch and am looking for info that can help me develop a web-based SIP Phone
> 
>  I am developing a web-based SIP Phone based on PHP/MySQL and OpenSIPS. I have developed the component of the SIP phone responsible for the 3way handshake and SDP offer. 
> 
> My challenge is to now integrate the application to a rtp stack which will enable the app to pick up audio and transmit it over the internet. Ultimately I want to connect the app to the PSTN using a media server such as Freeswitch or Asterisk. I want to use the Speex codec. I wanted to know whether Freeswitch has a rtp stack integrated with speex and whether there any tutorials on how integration to a softphone can be done. Any pointers or ideas from you would be very helpful and highly appreaciated.
> 
> regards,
> James Mbuthia
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Mon Dec  6 21:01:22 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 6 Dec 2010 13:01:22 -0500
Subject: [Freeswitch-users] SIP Phone Integration
References: <AANLkTi=CJYQ2g6NO+YfnPz8wRkX79_xjBc3RBRfM+xEs@mail.gmail.com><AANLkTikQrkp43CetufJsBDO+70metjjh9in7ePOuQnU_@mail.gmail.com><AANLkTim=GLEN9tvam461WB7A3KjW0qNm_9hXKOuC7Sq4@mail.gmail.com><3D0A80D2CE2244A7A28E94508390CEB7@e1705><AANLkTim_o3YdYob+3RsdEOSSpN4XZ5Dam1zv6mtVp_mG@mail.gmail.com>
	<4DAAA038-8EDC-4262-8B15-26AB77D8ACAA@ipeva.fr>
Message-ID: <8987768B4C2D43C394490F4C51EE1DF7@e1705>

or use acme company,
but very expensive
  ----- Original Message ----- 
  From: David Ponzone 
  To: FreeSWITCH Users Help 
  Sent: Monday, December 06, 2010 12:56 PM
  Subject: Re: [Freeswitch-users] SIP Phone Integration


  If you choose another codec, you could use a HW-based transcoding solution like the Sangoma D100/500, which goes from 30 to 2000 concurrent calls.
  According their website, they don't support Speex.


  David Ponzone  Direction Technique
  email: david.ponzone at ipeva.fr
  tel:      01 74 03 18 97
  gsm:   06 66 98 76 34


  Service Client IPeva
  tel:      0811 46 26 26
  www.ipeva.fr  -   www.ipeva-studio.com


  Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.







  Le 06/12/2010 ? 16:23, James Mbuthia a ?crit :


    Thanks,  about how many concurrent calls are you able to when transcoding with FS?


    On Mon, Dec 6, 2010 at 5:17 PM, Madovsky <infos at madovsky.org> wrote:

      You can transcode with FS, 
      I did already a web phone (boophone) with FS but ;keep in mind
      that transcode=CPU
        ----- Original Message ----- 
        From: James Mbuthia 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Monday, December 06, 2010 9:49 AM
        Subject: [Freeswitch-users] SIP Phone Integration


        Hi guys,

        Am new to Freeswitch and am looking for info that can help me develop a web-based SIP Phone

         I am developing a web-based SIP Phone based on PHP/MySQL and OpenSIPS. I have developed the component of the SIP phone responsible for the 3way handshake and SDP offer. 


        My challenge is to now integrate the application to a rtp stack which will enable the app to pick up audio and transmit it over the internet. Ultimately I want to connect the app to the PSTN using a media server such as Freeswitch or Asterisk. I want to use the Speex codec. I wanted to know whether Freeswitch has a rtp stack integrated with speex and whether there any tutorials on how integration to a softphone can be done. Any pointers or ideas from you would be very helpful and highly appreaciated.


        regards,
        James Mbuthia










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



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




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



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





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From david.ponzone at ipeva.fr  Mon Dec  6 21:14:51 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 6 Dec 2010 19:14:51 +0100
Subject: [Freeswitch-users] SIP Phone Integration
In-Reply-To: <8987768B4C2D43C394490F4C51EE1DF7@e1705>
References: <AANLkTi=CJYQ2g6NO+YfnPz8wRkX79_xjBc3RBRfM+xEs@mail.gmail.com><AANLkTikQrkp43CetufJsBDO+70metjjh9in7ePOuQnU_@mail.gmail.com><AANLkTim=GLEN9tvam461WB7A3KjW0qNm_9hXKOuC7Sq4@mail.gmail.com><3D0A80D2CE2244A7A28E94508390CEB7@e1705><AANLkTim_o3YdYob+3RsdEOSSpN4XZ5Dam1zv6mtVp_mG@mail.gmail.com>
	<4DAAA038-8EDC-4262-8B15-26AB77D8ACAA@ipeva.fr>
	<8987768B4C2D43C394490F4C51EE1DF7@e1705>
Message-ID: <A16E064D-BEB1-4CD7-A534-F96F85CE1E82@ipeva.fr>

Acme company ?

You meant ACME Packet ?

A ACME SBC just for transcoding would be a little bit overkill, except if he really wants to be able to go up to 30,000 concurrent calls.
And I am pretty sure FreeSWITCH with 15 D500 cards maxed out could achieve the same result, for a fraction of the price.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 06/12/2010 ? 19:01, Madovsky a ?crit :

> or use acme company,
> but very expensive
> ----- Original Message -----
> From: David Ponzone
> To: FreeSWITCH Users Help
> Sent: Monday, December 06, 2010 12:56 PM
> Subject: Re: [Freeswitch-users] SIP Phone Integration
> 
> If you choose another codec, you could use a HW-based transcoding solution like the Sangoma D100/500, which goes from 30 to 2000 concurrent calls.
> According their website, they don't support Speex.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 06/12/2010 ? 16:23, James Mbuthia a ?crit :
> 
>> Thanks,  about how many concurrent calls are you able to when transcoding with FS?
>> 
>> On Mon, Dec 6, 2010 at 5:17 PM, Madovsky <infos at madovsky.org> wrote:
>> You can transcode with FS,
>> I did already a web phone (boophone) with FS but ;keep in mind
>> that transcode=CPU
>> ----- Original Message -----
>> From: James Mbuthia
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Monday, December 06, 2010 9:49 AM
>> Subject: [Freeswitch-users] SIP Phone Integration
>> 
>> Hi guys,
>> 
>> Am new to Freeswitch and am looking for info that can help me develop a web-based SIP Phone
>> 
>>  I am developing a web-based SIP Phone based on PHP/MySQL and OpenSIPS. I have developed the component of the SIP phone responsible for the 3way handshake and SDP offer. 
>> 
>> My challenge is to now integrate the application to a rtp stack which will enable the app to pick up audio and transmit it over the internet. Ultimately I want to connect the app to the PSTN using a media server such as Freeswitch or Asterisk. I want to use the Speex codec. I wanted to know whether Freeswitch has a rtp stack integrated with speex and whether there any tutorials on how integration to a softphone can be done. Any pointers or ideas from you would be very helpful and highly appreaciated.
>> 
>> regards,
>> James Mbuthia
>> 
>> 
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mail at jankubr.com  Mon Dec  6 21:16:53 2010
From: mail at jankubr.com (Jan Kubr)
Date: Mon, 6 Dec 2010 19:16:53 +0100
Subject: [Freeswitch-users] mod_pocketsphinx
In-Reply-To: <C4BCDAB3-5371-48D5-810F-1A8F299529F6@freeswitch.org>
References: <4C1997A0.1080909@gmail.com>
	<BA0A1EAE-474B-4FE1-8BB3-03BC4B1C3655@freeswitch.org>
	<4C1ACD2A.70906@gmail.com>
	<D7122611-F34A-40B4-B0BE-C95A3D551E09@freeswitch.org>
	<4C1AE301.30405@gmail.com> <4C3D48A5.6030508@todandlorna.com>
	<C4BCDAB3-5371-48D5-810F-1A8F299529F6@freeswitch.org>
Message-ID: <AANLkTi=aTwdANAkNR1V5HXnnrZD8FDeRVWKJktWYCGFa@mail.gmail.com>

I've run into the same issue and can't find any pocketsphinx related
packages on this machine.
Did you guys figure out what the problem was? If it was a system package,
what was its name?

Thanks!
Jan

On Wed, Jul 14, 2010 at 7:39 AM, Brian West <brian at freeswitch.org> wrote:

> Yes it works fine if you don't use Ubuntu's packages and wipe them off the
> system 100%.  We download and build a more bleeding edge version.
>
> /b
>
> On Jul 14, 2010, at 12:18 AM, Tod Hansmann wrote:
>
> > Did this ever get resolved?  I'm seeing Roy's exact problem (when
> > loading, pocketsphinx fails with:
> > **/usr/local/freeswitch/mod/mod_pocketsphinx.so: undefined symbol:
> > ngram_model_get_counts**).  I'm on Ubuntu 10.04 Server, 32 bit (It's a
> > Via C7, don't hate me).  I have verified I have no sphinx or
> > pocketsphinx packages installed, and it doesn't work with them installed
> > either.  (I tried installing them before coming to the mailing list).
> >
> > Thoughts?
> >
> > -Tod Hansmann
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Dec  6 22:19:24 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 6 Dec 2010 13:19:24 -0600
Subject: [Freeswitch-users] mod_pocketsphinx
In-Reply-To: <AANLkTi=aTwdANAkNR1V5HXnnrZD8FDeRVWKJktWYCGFa@mail.gmail.com>
References: <4C1997A0.1080909@gmail.com>
	<BA0A1EAE-474B-4FE1-8BB3-03BC4B1C3655@freeswitch.org>
	<4C1ACD2A.70906@gmail.com>
	<D7122611-F34A-40B4-B0BE-C95A3D551E09@freeswitch.org>
	<4C1AE301.30405@gmail.com> <4C3D48A5.6030508@todandlorna.com>
	<C4BCDAB3-5371-48D5-810F-1A8F299529F6@freeswitch.org>
	<AANLkTi=aTwdANAkNR1V5HXnnrZD8FDeRVWKJktWYCGFa@mail.gmail.com>
Message-ID: <AF932101-9B49-43F6-A488-8E1C5897E88F@freeswitch.org>

No clue I'll try to compile it again today.

/b

On Dec 6, 2010, at 12:16 PM, Jan Kubr wrote:

> I've run into the same issue and can't find any pocketsphinx related packages on this machine.
> Did you guys figure out what the problem was? If it was a system package, what was its name?
> 
> Thanks!
> Jan
> 
> On Wed, Jul 14, 2010 at 7:39 AM, Brian West <brian at freeswitch.org> wrote:
> Yes it works fine if you don't use Ubuntu's packages and wipe them off the system 100%.  We download and build a more bleeding edge version.
> 
> /b

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From mthakershi at gmail.com  Tue Dec  7 03:38:04 2010
From: mthakershi at gmail.com (Malay Thakershi)
Date: Mon, 6 Dec 2010 18:38:04 -0600
Subject: [Freeswitch-users] Cepstral + FS question
Message-ID: <AANLkTi=N4yG-z5FLUo_hn8YCqDZezSpTcmn74=Qv7c8u@mail.gmail.com>

Hello, it would be great help if someone who has used Cepstral from FS can
share their views.

1. I just have one Cepstral Allison voice license (1 port) on my FS server.
I use swift command to convert text files to WAV which are then played by
mod_managed in FS call process. First question is regarding limitations on
simultaneous conversions (TXT to WAV) using swift command. If I have
multiple threads doing this, will there be any degradation because of
Cepstral? I tried running two BAT files with 3 commands each. But running
them simultaneously or separate produced same outcome. Does anyone know when
Cepstral licensing kicks in and starts degrading quality ( or worse
inserting "not licensed") prompt?

2. When I call session speak from mod_managed (or stream file) after
selecting Allison / Cepstral as my voice, does Cepstral engine interfere
with quality of the playback? If yes, when will I see it and how can I
produce their effects?

3. What is the sensible number of ports (from Cepstral) I should be prepared
to buy if findings in the previous points imposes significant limitations?

Thank you for any help or guidance.

Malay
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From haloha201 at yahoo.com  Tue Dec  7 07:34:13 2010
From: haloha201 at yahoo.com (ha do)
Date: Mon, 6 Dec 2010 20:34:13 -0800 (PST)
Subject: [Freeswitch-users] get error on skypopen module
Message-ID: <42647.26863.qm@web32403.mail.mud.yahoo.com>

Hi list

i setup freeswitch and skypopen running fine

there are 2 skype clients run on freeswitch
freeswitch at internal> sk list
sk console is NOT yet assigned
F ID        Name        IB (F/T)    OB (F/T)    State   CallFlw         UUID
= ====    ========      =======     =======     ======  ============    ======
  1     [interface1]      0/1        3/7        IDLE    IDLE
  2     [interface2]      0/5        1/5        IDLE    IDLE

the skype clients are used the same username + password of skype account


the skypopen works fine but i get error below in the debug mode

2010-12-07 04:24:35.794611 [DEBUG] switch_core_state_machine.c:462 (skypopen/interface1) State DESTROY going to sleep
2010-12-07 04:24:35.795807 [DEBUG] skypopen_protocol.c:173     [|] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING: |||ERROR 559 CALL: Action failed|||
2010-12-07 04:24:35.795807 [DEBUG] skypopen_protocol.c:228     [|] [DEBUG_SKYPE  228  ][interface1     ][IDLE,IDLE] Skype got ERROR about a failed action (probably TRYING to HANGUP A CALL), no problem: |||ERROR 559 CALL: Action failed|||
2010-12-07 04:24:35.795807 [DEBUG] skypopen_protocol.c:173     [|] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING: |||ERROR 559 CALL: Action failed|||
2010-12-07 04:24:35.795807 [DEBUG] skypopen_protocol.c:228     [|] [DEBUG_SKYPE  228  ][interface1     ][IDLE,IDLE] Skype got ERROR about a failed action (probably TRYING to HANGUP A CALL), no problem: |||ERROR 559 CALL: Action failed|||

How to fix it

Thank you
Ha`


      


From gmaruzz at gmail.com  Tue Dec  7 10:56:07 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 7 Dec 2010 08:56:07 +0100
Subject: [Freeswitch-users] get error on skypopen module
In-Reply-To: <42647.26863.qm@web32403.mail.mud.yahoo.com>
References: <42647.26863.qm@web32403.mail.mud.yahoo.com>
Message-ID: <AANLkTi=cHoLFUDnw31d7LT2-hM2VimEdmv2kwg3bgdzM@mail.gmail.com>

On Tue, Dec 7, 2010 at 5:34 AM, ha do <haloha201 at yahoo.com> wrote:
> the skypopen works fine but i get error below in the debug mode

don't run in debug mode if you are not debugging :)

> [interface1 ? ? ][IDLE,IDLE] Skype got ERROR about a failed action (probably TRYING to HANGUP A CALL), no problem: |||ERROR 559 CALL: Action failed|||

as is clearly written, this is not a problem. Is a debug message that
explains we tried to hangup a skype call that the skype client closed
before. Is just to be double sure that we try to close it anyway. The
debug message is just an explanation.

>
> How to fix it

nothing to fix, no problems here.

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From ranjtech at gmail.com  Tue Dec  7 13:51:25 2010
From: ranjtech at gmail.com (RR)
Date: Tue, 7 Dec 2010 05:51:25 -0500
Subject: [Freeswitch-users] FreeSWITCH 64bit compilation error (Solaris
	10)
In-Reply-To: <2C944C29-80D7-4F4F-ABD4-EC5202FD08B7@jerris.com>
References: <4ab3d091.9453f10a.0891.2fa4@mx.google.com>
	<2C944C29-80D7-4F4F-ABD4-EC5202FD08B7@jerris.com>
Message-ID: <AANLkTi=gWy4R7DGfYrWkVRho44ZaA6V8hU6w5xOEOt5s@mail.gmail.com>

Hi Guys,

I have a really stupid question to ask. Having skimmed through the steps to
take to compile FreeSWITCH in a Solaris environment, everywhere it seems to
talk about the jds-cbe and sun studio etc...basically meaning that I have to
have the X11 framework/graphical environment/packages installed on the
machine? Is it possible to compile it without this using the standard,
./configure, make combo and the standard gcc etc?

Thanks
RR


On Wed, Sep 23, 2009 at 4:20 AM, Michael Jerris <mike at jerris.com> wrote:

> Try taking a list at the info here:
> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Solaris You need to be passing any necessary cflags in on configure
>
> Mike
>
>   On Sep 18, 2009, at 2:26 PM, email lists wrote:
>
>    Forwarding the issue below to see if anyone is familiar with this
> issue, and/or what our next steps should be.
>
> Thanks,
> Vladimir
>
>
>  Looks like a problem with a Makefile not honoring CFLAGS,etc.  Perhaps
> you can report this to the dev team.  Other components built fine but this
> damn spidermonkey is buggering.
>
> # file /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch*.o |
> head
> /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-g711.o: ELF
> 64-bit LSB relocatable AMD64 Version 1
> /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-getgateway.o:
> ELF 64-bit LSB relocatable AMD64 Version 1 [CMOV]
> /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-igd_desc_parse.o:
> ELF 64-bit LSB relocatable AMD64 Version 1 [CMOV]
> /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-libteletone_detect.o:
> ELF 64-bit LSB relocatable AMD64 Version 1 [SSE2 SSE AMD_3DNow CMOV FPU]
> /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-libteletone_generate.o:
> ELF 64-bit LSB relocatable AMD64 Version 1 [SSE2 SSE AMD_3DNow CMOV]
> /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-minisoap.o:
> ELF 64-bit LSB relocatable AMD64 Version 1
> /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-minissdpc.o:
> ELF 64-bit LSB relocatable AMD64 Version 1 [SSE]
> /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniupnpc.o:
> ELF 64-bit LSB relocatable AMD64 Version 1 [SSE AMD_3DNow]
> /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniwget.o:
> ELF 64-bit LSB relocatable AMD64 Version 1 [SSE]
> /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-minixml.o:
> ELF 64-bit LSB relocatable AMD64 Version 1 [SSE]
>
> # file /opt/freeradius-client/sbin/radacct
> /opt/freeradius-client/sbin/radacct:    ELF 64-bit LSB executable AMD64
> Version 1 [SSE2 SSE FXSR AMD_3DNow CMOV FPU], dynamically linked, not
> stripped
>
> *Build output:*
> Making all in nua
>          LTCOMPILE nua.lo
>          LTCOMPILE nua_common.lo
>          LTCOMPILE nua_stack.lo
>          LTCOMPILE nua_server.lo
>          LTCOMPILE nua_client.lo
>          LTCOMPILE nua_extension.lo
>          LTCOMPILE nua_dialog.lo
>          LTCOMPILE outbound.lo
>          LTCOMPILE nua_params.lo
>          LTCOMPILE nua_register.lo
>          LTCOMPILE nua_registrar.lo
>          LTCOMPILE nua_session.lo
>          LTCOMPILE nua_options.lo
>          LTCOMPILE nua_message.lo
>          LTCOMPILE nua_publish.lo
>          LTCOMPILE nua_subnotref.lo
>          LTCOMPILE nua_notifier.lo
>          LTCOMPILE nua_event_server.lo
>          LTCOMPILE nua_tag.lo
>          LTCOMPILE nua_tag_ref.lo
>          LINK libnua.la
>          LINK libsofia-sip-ua.la
> libtool: link: warning: `-version-info/-version-number' is ignored for
> convenience libraries
> Making all in packages
> Creating mod_sofia_la-mod_sofia.lo
> mkdir .libs
> Compiling mod_sofia.c ...
> "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofia-sip-ua/nua/nua_params.h",
> line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE)
> Creating mod_sofia_la-sofia.lo
> Compiling sofia.c ...
> "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofia-sip-ua/nua/nua_params.h",
> line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE)
> "sofia.c", line 3522: warning: enum type mismatch: arg #2
> (E_ENUM_TYPE_MISMATCH_ARG)
> Creating mod_sofia_la-sofia_glue.lo
> Compiling sofia_glue.c ...
> "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofia-sip-ua/nua/nua_params.h",
> line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE)
> Creating mod_sofia_la-sofia_presence.lo
> Compiling sofia_presence.c ...
> "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofia-sip-ua/nua/nua_params.h",
> line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE)
> Creating mod_sofia_la-sofia_reg.lo
> Compiling sofia_reg.c ...
> "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofia-sip-ua/nua/nua_params.h",
> line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE)
> Creating mod_sofia_la-sofia_sla.lo
> Compiling sofia_sla.c ...
> "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofia-sip-ua/nua/nua_params.h",
> line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE)
> Creating mod_sofia.la
>
> making all mod_speex
> Compiling mod_speex.c...
> mkdir .libs
> Compiling mod_speex.c ...
> Creating mod_speex.so...
>
> making all mod_spidermonkey
> cd config; /usr/sfw/bin/gmake -j1 export
> ld: fatal: file now.o: wrong ELF class: ELFCLASS64
> ld: fatal: File processing errors. No output written to now
> gmake[7]: *** [now] Error 1
> gmake[6]: *** [export] Error 2
> gmake[5]: *** [/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/js/
> libjs.la] Error 2
> gmake[4]: *** [all] Error 1
> gmake[3]: *** [mod_spidermonkey-all] Error 1
> gmake[2]: *** [all-recursive] Error 1
> Making all in build
>  +-------- FreeSWITCH Build Complete -----------+
>  + FreeSWITCH has been successfully built.      +
>  + Install by running:                          +
>  +                                              +
>  +               /usr/sfw/bin/gmake install                   +
>  +----------------------------------------------+
> gmake[1]: *** [all-recursive] Error 1
> gmake: *** [all] Error 2
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From enp at altlinux.org  Tue Dec  7 12:40:45 2010
From: enp at altlinux.org (Eugene Prokopiev)
Date: Tue, 7 Dec 2010 12:40:45 +0300
Subject: [Freeswitch-users] mod_fifo vs mod_callcenter
In-Reply-To: <AANLkTinc1RBRgYFn7K0HshRsRXXBf2fHA84QTMkxYyvN@mail.gmail.com>
References: <AANLkTimftpTpLLYXED0Fk1=NwVowzxv=8T0ufQBkrAZ8@mail.gmail.com>
	<AANLkTinC85pzthxCVcjopW9q=cjBkkjS0jXGBhKBjyHV@mail.gmail.com>
	<AANLkTikWttM3het5NTO7YJSu=0xT84=O9-4rpvgzNhdC@mail.gmail.com>
	<AANLkTinc1RBRgYFn7K0HshRsRXXBf2fHA84QTMkxYyvN@mail.gmail.com>
Message-ID: <AANLkTim-gELfGsYoeoGSfJXUsEZyb-ZJMsipSgu1HbpO@mail.gmail.com>

2010/12/6 Eugene Prokopiev <enp at altlinux.org>:
>> Is it possible to implement the same logic for off-hook members? Can
>> you point me the line/function in mod_fifo.c in which off-hook member
>> is selecting for call?
>
> I tried to debug callflow with help of switch_log_printf. Two off-hook
> agents executes:
>
> <action application="fifo" data="$1 out wait"/>
>
> and waits in:
>
> mod_fifo.c:2588 - moh_status = switch_ivr_play_file(session, NULL, moh, &args);
>
> After executing:
>
> <action application="fifo" data="$1 in"/>
>
> only one off-hook agent leaves switch_ivr_play_file and process
> incoming call. Why can it leaves switch_ivr_play_file function?

I tried to debug switch_ivr_play_file function in
switch_ivr_play_say.c with with help of switch_log_printf. I've found
cycle with for (;;) which is interrupted by this code:

if (args && (args->read_frame_callback)) {
    int ok = 1;
    switch_set_flag(fh, SWITCH_FILE_CALLBACK);
    if ((status = args->read_frame_callback(session, read_frame,
args->user_data)) != SWITCH_STATUS_SUCCESS) {
        ok = 0;
    }
    switch_clear_flag(fh, SWITCH_FILE_CALLBACK);
    if (!ok) {
        break;
    }
}

But I can't find the reason to interruption. I tried to find it in
mod_fifo.c:node_thread_run but it seems cycle in this function works
only for on-hook agents.

--
Thanks,
Eugene Prokopiev


From tayeb.meftah at gmail.com  Tue Dec  7 14:43:40 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 07 Dec 2010 12:43:40 +0100
Subject: [Freeswitch-users] get error on skypopen module
In-Reply-To: <42647.26863.qm@web32403.mail.mud.yahoo.com>
References: <42647.26863.qm@web32403.mail.mud.yahoo.com>
Message-ID: <4CFE1DEC.5040000@gmail.com>

did you autorised skypopen to access skype from the skype client?
and did you configured the skypopen interfaces to each one of the skype 
clients?
thanks
Le 07/12/2010 05:34, ha do a ?crit :
> Hi list
>
> i setup freeswitch and skypopen running fine
>
> there are 2 skype clients run on freeswitch
> freeswitch at internal>  sk list
> sk console is NOT yet assigned
> F ID        Name        IB (F/T)    OB (F/T)    State   CallFlw         UUID
> = ====    ========      =======     =======     ======  ============    ======
>    1     [interface1]      0/1        3/7        IDLE    IDLE
>    2     [interface2]      0/5        1/5        IDLE    IDLE
>
> the skype clients are used the same username + password of skype account
>
>
> the skypopen works fine but i get error below in the debug mode
>
> 2010-12-07 04:24:35.794611 [DEBUG] switch_core_state_machine.c:462 (skypopen/interface1) State DESTROY going to sleep
> 2010-12-07 04:24:35.795807 [DEBUG] skypopen_protocol.c:173     [|] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING: |||ERROR 559 CALL: Action failed|||
> 2010-12-07 04:24:35.795807 [DEBUG] skypopen_protocol.c:228     [|] [DEBUG_SKYPE  228  ][interface1     ][IDLE,IDLE] Skype got ERROR about a failed action (probably TRYING to HANGUP A CALL), no problem: |||ERROR 559 CALL: Action failed|||
> 2010-12-07 04:24:35.795807 [DEBUG] skypopen_protocol.c:173     [|] [DEBUG_SKYPE  173  ][interface1     ][IDLE,IDLE] READING: |||ERROR 559 CALL: Action failed|||
> 2010-12-07 04:24:35.795807 [DEBUG] skypopen_protocol.c:228     [|] [DEBUG_SKYPE  228  ][interface1     ][IDLE,IDLE] Skype got ERROR about a failed action (probably TRYING to HANGUP A CALL), no problem: |||ERROR 559 CALL: Action failed|||
>
> How to fix it
>
> Thank you
> Ha`
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313



From wstephen80 at gmail.com  Tue Dec  7 15:26:28 2010
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 7 Dec 2010 13:26:28 +0100
Subject: [Freeswitch-users] Scale UP Freeswitch
Message-ID: <AANLkTi=4n_wMSDE7KA9JTDhP+ToGggt-nJzf=EYwTd=6@mail.gmail.com>

Hi,
I have one server running Freeswitch with some ISDN connections (via
FreeTDM+Sangoma boards) and some SIP connections with service providers and
customer.
The usage of Freeswitch is as switching so it "bridge" each incoming call to
a new outgoing call.
SIP calls use G.729 and ISDN calls use ALaw for voice encoding.
Now the number of call is grow up and also the CPU load is a little high so
I have the necessity to scale UP my Freeswitch to handle more calls: what is
the best way to do that?
My first idea is to use a Sangoma D500 board to reduce the CPU load. Can be
this a solution?
There are different way to scale UP?
Thanks in advance,

Stephen
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From haloha201 at yahoo.com  Tue Dec  7 15:15:50 2010
From: haloha201 at yahoo.com (ha do)
Date: Tue, 7 Dec 2010 04:15:50 -0800 (PST)
Subject: [Freeswitch-users] get error on skypopen module
In-Reply-To: <4CFE1DEC.5040000@gmail.com>
Message-ID: <123359.43159.qm@web32401.mail.mud.yahoo.com>

Hi Meftab

> did you autorised skypopen to access
> skype from the skype client?
you mean the skypopen_auth ??? if so, the answer is yes

> did you configured the skypopen interfaces to each one
> of the skype
i dont understand, because i use multi skypy client(skype software) with 1 username and 1 password 
and use the sample config file in $source.../../configs/multiple-instances-same-skype-user/ 

2 skype clients run in difference folder
/home/cucku/multi/interfaces01
/home/cucku/multi/interfaces02

so i only need to config the skypopen.conf.xml look like below:
<configuration name="skypopen.conf" description="Skypopen Configuration">
  <global_settings>
    <param name="debug" value="8"/> 
    <param name="dialplan" value="XML"/>
    <param name="context" value="default"/>
    <param name="destination" value="5000"/>
    <param name="skype_user" value="do_nguyen_ha"/>
    <param name="report_incoming_chatmessages" value="false"/>
    <param name="silent_mode" value="false"/>
    <param name="write_silence_when_idle" value="true"/>
    <param name="setsockopt" value="false"/>
  </global_settings>
  <!-- one entry here per each skypopen interface -->
  <per_interface_settings>
    <interface id="1" name="interface1">
        <param name="X11-display" value=":101"/>
    </interface>
    <interface id="2" name="interface2">
        <param name="X11-display" value=":102"/>
    </interface>
  </per_interface_settings>
</configuration>

does the conifg look ok?? if not, please guide me to make it right

i have a question on event socket, what is the event name for skypopen to monitor the interface IDLE/not IDLE or ANSWER or IN PROGRESS...

which event plain should i take care of


Thank you
Ha`
--- On Tue, 12/7/10, Meftah Tayeb <tayeb.meftah at gmail.com> wrote:

> From: Meftah Tayeb <tayeb.meftah at gmail.com>
> Subject: Re: [Freeswitch-users] get error on skypopen module
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Cc: "ha do" <haloha201 at yahoo.com>
> Date: Tuesday, December 7, 2010, 4:43 AM
> did you autorised skypopen to access
> skype from the skype client?
> and did you configured the skypopen interfaces to each one
> of the skype 
> clients?
> thanks
> Le 07/12/2010 05:34, ha do a ?crit :
> > Hi list
> >
> > i setup freeswitch and skypopen running fine
> >
> > there are 2 skype clients run on freeswitch
> > freeswitch at internal>? sk list
> > sk console is NOT yet assigned
> > F ID? ? ? ? Name? ?
> ? ? IB (F/T)? ? OB (F/T)? ?
> State???CallFlw? ? ?
> ???UUID
> > = ====? ? ========? ? ?
> =======? ???=======?
> ???======? ============? ?
> ======
> >? ? 1?
> ???[interface1]? ? ? 0/1?
> ? ? ? 3/7? ? ? ?
> IDLE? ? IDLE
> >? ? 2?
> ???[interface2]? ? ? 0/5?
> ? ? ? 1/5? ? ? ?
> IDLE? ? IDLE
> >
> > the skype clients are used the same username +
> password of skype account
> >
> >
> > the skypopen works fine but i get error below in the
> debug mode
> >
> > 2010-12-07 04:24:35.794611 [DEBUG]
> switch_core_state_machine.c:462 (skypopen/interface1) State
> DESTROY going to sleep
> > 2010-12-07 04:24:35.795807 [DEBUG]
> skypopen_protocol.c:173? ???[|]
> [DEBUG_SKYPE? 173? ][interface1?
> ???][IDLE,IDLE] READING: |||ERROR 559 CALL:
> Action failed|||
> > 2010-12-07 04:24:35.795807 [DEBUG]
> skypopen_protocol.c:228? ???[|]
> [DEBUG_SKYPE? 228? ][interface1?
> ???][IDLE,IDLE] Skype got ERROR about a
> failed action (probably TRYING to HANGUP A CALL), no
> problem: |||ERROR 559 CALL: Action failed|||
> > 2010-12-07 04:24:35.795807 [DEBUG]
> skypopen_protocol.c:173? ???[|]
> [DEBUG_SKYPE? 173? ][interface1?
> ???][IDLE,IDLE] READING: |||ERROR 559 CALL:
> Action failed|||
> > 2010-12-07 04:24:35.795807 [DEBUG]
> skypopen_protocol.c:228? ???[|]
> [DEBUG_SKYPE? 228? ][interface1?
> ???][IDLE,IDLE] Skype got ERROR about a
> failed action (probably TRYING to HANGUP A CALL), no
> problem: |||ERROR 559 CALL: Action failed|||
> >
> > How to fix it
> >
> > Thank you
> > Ha`
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >? ? 
> 
> 
> -- 
> Meftah Tayeb
> inum: +883510001288000
> Phone: +13602276297
> Fax: +12538020313
> 
> 


      


From haloha201 at yahoo.com  Tue Dec  7 15:29:34 2010
From: haloha201 at yahoo.com (ha do)
Date: Tue, 7 Dec 2010 04:29:34 -0800 (PST)
Subject: [Freeswitch-users]  get error on skypopen module
Message-ID: <978672.50991.qm@web32401.mail.mud.yahoo.com>

Hi Giovanny

how to turn off the debug?? remove the below line in skypopen.xml.conf
    <param name="debug" value="8"/>

the error i get when i run the /usr/local/freeswitch/bin/fs_cli

i do check the CPU performance so 1 call from/to skype client it will cost 5% CPU of server
if 2 calls, CPU will be 10%

i dont know that is normal when freeswitch run with skypopen module


how do i change the 
Would you like to receive list mail batched in a daily digest? 
from yes to no  in http://lists.freeswitch.org/mailman/listinfo/freeswitch-users


Thank you
Ha


On Tue, Dec 7, 2010 at 5:34 AM, ha do <haloha201 at yahoo.com> wrote:
> the skypopen works fine but i get error below in the debug mode

don't run in debug mode if you are not debugging :)

> [interface1     ][IDLE,IDLE] Skype got ERROR about a failed action (probably TRYING to HANGUP A CALL), no problem: |||ERROR 559 CALL: Action failed|||

as is clearly written, this is not a problem. Is a debug message that
explains we tried to hangup a skype call that the skype client closed
before. Is just to be double sure that we try to close it anyway. The
debug message is just an explanation.

>
> How to fix it

nothing to fix, no problems here.

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


      


From juanito1982 at gmail.com  Tue Dec  7 15:42:58 2010
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Tue, 7 Dec 2010 13:42:58 +0100
Subject: [Freeswitch-users] Scale UP Freeswitch
In-Reply-To: <AANLkTi=4n_wMSDE7KA9JTDhP+ToGggt-nJzf=EYwTd=6@mail.gmail.com>
References: <AANLkTi=4n_wMSDE7KA9JTDhP+ToGggt-nJzf=EYwTd=6@mail.gmail.com>
Message-ID: <AANLkTi=sNVJ70h7MrH3t5ypVO2RrNTPswfWad4qcwhmV@mail.gmail.com>

It is a temp solution. If your system continues growing up, you will have
those problems again. Take a look at:

http://wiki.freeswitch.org/wiki/Enterprise_Deployment

You might find useful to install a second server.

<http://wiki.freeswitch.org/wiki/Enterprise_Deployment>Regards

2010/12/7 Stephen Wilde <wstephen80 at gmail.com>

> Hi,
> I have one server running Freeswitch with some ISDN connections (via
> FreeTDM+Sangoma boards) and some SIP connections with service providers and
> customer.
> The usage of Freeswitch is as switching so it "bridge" each incoming call
> to a new outgoing call.
> SIP calls use G.729 and ISDN calls use ALaw for voice encoding.
> Now the number of call is grow up and also the CPU load is a little high so
> I have the necessity to scale UP my Freeswitch to handle more calls: what is
> the best way to do that?
> My first idea is to use a Sangoma D500 board to reduce the CPU load. Can be
> this a solution?
> There are different way to scale UP?
> Thanks in advance,
>
> Stephen
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From w8hdkim at gmail.com  Tue Dec  7 15:51:56 2010
From: w8hdkim at gmail.com (Kim Culhan)
Date: Tue, 7 Dec 2010 07:51:56 -0500
Subject: [Freeswitch-users] FreeSWITCH 64bit compilation error (Solaris
	10)
Message-ID: <AANLkTikWGfii8zhPRtXLRWUf_0zkyB_jJCo1yT80Jn83@mail.gmail.com>

On Tue, December 7, 2010 5:51 am, RR wrote:
> Hi Guys,
>
> I have a really stupid question to ask. Having skimmed through the steps
to
> take to compile FreeSWITCH in a Solaris environment, everywhere it seems
to
> talk about the jds-cbe and sun studio etc...basically meaning that I have
to
> have the X11 framework/graphical environment/packages installed on the
> machine? Is it possible to compile it without this using the standard,
> ./configure, make combo and the standard gcc etc?
Installing the jds-cbe and sun studio does not involve installing 'the X11
framework' on the machine.

refer to::

http://wiki.freeswitch.org/wiki/Installation_Guide#OpenSolaris_os200906

A step in the above refers to: /opt/dtbld/bin/env.sh which is installed by
the jds-cbe

The script sets some variables in the environment which are read by
Freeswitch configure
as Mike mentioned:

> On Wed, Sep 23, 2009 at 4:20 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> Try taking a list at the info here:
>> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#SolarisYou need
>> to be passing any necessary cflags in on configure
Please follow the guide for compiling on Opensolaris and post if you run
into any problems.
**
Hope this helps.

-kim
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From steveayre at gmail.com  Tue Dec  7 16:31:31 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 7 Dec 2010 13:31:31 +0000
Subject: [Freeswitch-users] Scale UP Freeswitch
In-Reply-To: <AANLkTi=4n_wMSDE7KA9JTDhP+ToGggt-nJzf=EYwTd=6@mail.gmail.com>
References: <AANLkTi=4n_wMSDE7KA9JTDhP+ToGggt-nJzf=EYwTd=6@mail.gmail.com>
Message-ID: <AANLkTi=N+6=AcELMtbyje3O7W0k7qm3nz8z0CoO3LmLo@mail.gmail.com>

There are a few performance tweaking tips at
http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.

Yes a Sangoma card will reduce your CPU load since transcoding won't
be done on the CPU any longer, that will then mean there's more CPU
available so you'll be able to handle more calls.

However, if you're looking to increase your number of calls then you
probably want a cluster of servers as Juan pointed out.

It'll mean you can increase the capacity by adding extra servers, so
there'd no longer be a limit to the number of calls you could handle
(just add another server).

It'll also make maintenance easier, as you'll be able to pull a server
from service for updates etc while traffic continues to run on the
other servers. Maintenance won't mean a service outage.

If you're handling that many calls then additional servers would make
your service more reliable. If a server crashes you'll still have the
calls running on the other servers while you're fixing the problem so
you won't have a complete outage. If FS is behind a load balancer then
your customers might not even notice anything apart from a few dropped
calls.

There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will
attempt to continue calls if FS crashes and restarts, but I think
that's only for SIP-SIP not SIP-ISDN.

-Steve




On 7 December 2010 12:26, Stephen Wilde <wstephen80 at gmail.com> wrote:
> Hi,
> I have one server running Freeswitch with some ISDN connections (via
> FreeTDM+Sangoma boards) and some SIP connections with service providers and
> customer.
> The usage of Freeswitch is as switching so it "bridge" each incoming call to
> a new outgoing call.
> SIP calls use G.729 and ISDN calls use ALaw for voice encoding.
> Now the number of call is grow up and also the CPU load is a little high so
> I have the necessity to scale UP my Freeswitch to handle more calls: what is
> the best way to do that?
> My first idea is to use a Sangoma D500 board to reduce the CPU load. Can be
> this a solution?
> There are different way to scale UP?
> Thanks in advance,
> Stephen
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Tue Dec  7 16:40:15 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 7 Dec 2010 13:40:15 +0000
Subject: [Freeswitch-users] get error on skypopen module
In-Reply-To: <978672.50991.qm@web32401.mail.mud.yahoo.com>
References: <978672.50991.qm@web32401.mail.mud.yahoo.com>
Message-ID: <AANLkTi=Ujq7KCHWsD-PSUdLDkTs9GUxQak5Ln9Z6s4XT@mail.gmail.com>

> how to turn off the debug?? remove the below line in skypopen.xml.conf
>    <param name="debug" value="8"/>

Yes, remove that line, then do reloadxml and reload mod_skypopen to
pick up the new config. Or restart FS.

> i do check the CPU performance so 1 call from/to skype client it will cost 5% CPU of server
> if 2 calls, CPU will be 10%
>
> i dont know that is normal when freeswitch run with skypopen module

Are you using the dummy sound driver? That's meant to reduce the CPU
usage, but I don't know how much. % will also depend on what CPU
you're using.

> how do i change the
> Would you like to receive list mail batched in a daily digest?
> from yes to no  in http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

1) Visit that link
2) Enter your email and click 'Unsubscribe or edit options' near the bottom
3) Enter your password (there's a reminder button that'll email it to
you if you don't know it)
4) Scroll down to the 'Your FreeSWITCH-users Subscription Options' section
5) Flip the 'Set Digest Mode' setting to Off and click 'Submit my changes'

-Steve






On 7 December 2010 12:29, ha do <haloha201 at yahoo.com> wrote:
> Hi Giovanny
>
> how to turn off the debug?? remove the below line in skypopen.xml.conf
> ? ?<param name="debug" value="8"/>
>
> the error i get when i run the /usr/local/freeswitch/bin/fs_cli
>
> i do check the CPU performance so 1 call from/to skype client it will cost 5% CPU of server
> if 2 calls, CPU will be 10%
>
> i dont know that is normal when freeswitch run with skypopen module
>
>
> how do i change the
> Would you like to receive list mail batched in a daily digest?
> from yes to no ?in http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
> Thank you
> Ha
>
>
> On Tue, Dec 7, 2010 at 5:34 AM, ha do <haloha201 at yahoo.com> wrote:
>> the skypopen works fine but i get error below in the debug mode
>
> don't run in debug mode if you are not debugging :)
>
>> [interface1 ? ? ][IDLE,IDLE] Skype got ERROR about a failed action (probably TRYING to HANGUP A CALL), no problem: |||ERROR 559 CALL: Action failed|||
>
> as is clearly written, this is not a problem. Is a debug message that
> explains we tried to hangup a skype call that the skype client closed
> before. Is just to be double sure that we try to close it anyway. The
> debug message is just an explanation.
>
>>
>> How to fix it
>
> nothing to fix, no problems here.
>
> -giovanni
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From steveayre at gmail.com  Tue Dec  7 16:48:02 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 7 Dec 2010 13:48:02 +0000
Subject: [Freeswitch-users] get error on skypopen module
In-Reply-To: <123359.43159.qm@web32401.mail.mud.yahoo.com>
References: <4CFE1DEC.5040000@gmail.com>
	<123359.43159.qm@web32401.mail.mud.yahoo.com>
Message-ID: <AANLkTikvBq2__wxPK8PYfqMLXSuQv3OuKVRUOtpCx3eR@mail.gmail.com>

> i have a question on event socket, what is the event name for skypopen to monitor the interface IDLE/not IDLE or ANSWER or IN PROGRESS...

Full ESL event list: http://wiki.freeswitch.org/wiki/Event_list

Skyopen doesn't generate events for state changes itself.

Check for CHANNEL_ANSWER and CHANNEL_PROGRESS/CHANNEL_PROGRESS_MEDIA.

For spotting an idle skype interface, CHANNEL_CREATE and
CHANNEL_DESTROY might tell you that by telling you when the interface
is in use.

If you want to poll for the interface states you can use the sk list
api function via ESL, although there's unfortunately no 'as xml'
formatting at the moment so it'll be slightly trickier to parse.

-Steve

On 7 December 2010 12:15, ha do <haloha201 at yahoo.com> wrote:
> Hi Meftab
>
>> did you autorised skypopen to access
>> skype from the skype client?
> you mean the skypopen_auth ??? if so, the answer is yes
>
>> did you configured the skypopen interfaces to each one
>> of the skype
> i dont understand, because i use multi skypy client(skype software) with 1 username and 1 password
> and use the sample config file in $source.../../configs/multiple-instances-same-skype-user/
>
> 2 skype clients run in difference folder
> /home/cucku/multi/interfaces01
> /home/cucku/multi/interfaces02
>
> so i only need to config the skypopen.conf.xml look like below:
> <configuration name="skypopen.conf" description="Skypopen Configuration">
> ?<global_settings>
> ? ?<param name="debug" value="8"/>
> ? ?<param name="dialplan" value="XML"/>
> ? ?<param name="context" value="default"/>
> ? ?<param name="destination" value="5000"/>
> ? ?<param name="skype_user" value="do_nguyen_ha"/>
> ? ?<param name="report_incoming_chatmessages" value="false"/>
> ? ?<param name="silent_mode" value="false"/>
> ? ?<param name="write_silence_when_idle" value="true"/>
> ? ?<param name="setsockopt" value="false"/>
> ?</global_settings>
> ?<!-- one entry here per each skypopen interface -->
> ?<per_interface_settings>
> ? ?<interface id="1" name="interface1">
> ? ? ? ?<param name="X11-display" value=":101"/>
> ? ?</interface>
> ? ?<interface id="2" name="interface2">
> ? ? ? ?<param name="X11-display" value=":102"/>
> ? ?</interface>
> ?</per_interface_settings>
> </configuration>
>
> does the conifg look ok?? if not, please guide me to make it right
>
> i have a question on event socket, what is the event name for skypopen to monitor the interface IDLE/not IDLE or ANSWER or IN PROGRESS...
>
> which event plain should i take care of
>
>
> Thank you
> Ha`
> --- On Tue, 12/7/10, Meftah Tayeb <tayeb.meftah at gmail.com> wrote:
>
>> From: Meftah Tayeb <tayeb.meftah at gmail.com>
>> Subject: Re: [Freeswitch-users] get error on skypopen module
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Cc: "ha do" <haloha201 at yahoo.com>
>> Date: Tuesday, December 7, 2010, 4:43 AM
>> did you autorised skypopen to access
>> skype from the skype client?
>> and did you configured the skypopen interfaces to each one
>> of the skype
>> clients?
>> thanks
>> Le 07/12/2010 05:34, ha do a ?crit :
>> > Hi list
>> >
>> > i setup freeswitch and skypopen running fine
>> >
>> > there are 2 skype clients run on freeswitch
>> > freeswitch at internal>? sk list
>> > sk console is NOT yet assigned
>> > F ID? ? ? ? Name
>> ? ? IB (F/T)? ? OB (F/T)
>> State???CallFlw
>> ???UUID
>> > = ====? ? ========
>> =======? ???=======
>> ???======? ============
>> ======
>> >? ? 1
>> ???[interface1]? ? ? 0/1
>> ? ? ? 3/7
>> IDLE? ? IDLE
>> >? ? 2
>> ???[interface2]? ? ? 0/5
>> ? ? ? 1/5
>> IDLE? ? IDLE
>> >
>> > the skype clients are used the same username +
>> password of skype account
>> >
>> >
>> > the skypopen works fine but i get error below in the
>> debug mode
>> >
>> > 2010-12-07 04:24:35.794611 [DEBUG]
>> switch_core_state_machine.c:462 (skypopen/interface1) State
>> DESTROY going to sleep
>> > 2010-12-07 04:24:35.795807 [DEBUG]
>> skypopen_protocol.c:173? ???[|]
>> [DEBUG_SKYPE? 173? ][interface1
>> ???][IDLE,IDLE] READING: |||ERROR 559 CALL:
>> Action failed|||
>> > 2010-12-07 04:24:35.795807 [DEBUG]
>> skypopen_protocol.c:228? ???[|]
>> [DEBUG_SKYPE? 228? ][interface1
>> ???][IDLE,IDLE] Skype got ERROR about a
>> failed action (probably TRYING to HANGUP A CALL), no
>> problem: |||ERROR 559 CALL: Action failed|||
>> > 2010-12-07 04:24:35.795807 [DEBUG]
>> skypopen_protocol.c:173? ???[|]
>> [DEBUG_SKYPE? 173? ][interface1
>> ???][IDLE,IDLE] READING: |||ERROR 559 CALL:
>> Action failed|||
>> > 2010-12-07 04:24:35.795807 [DEBUG]
>> skypopen_protocol.c:228? ???[|]
>> [DEBUG_SKYPE? 228? ][interface1
>> ???][IDLE,IDLE] Skype got ERROR about a
>> failed action (probably TRYING to HANGUP A CALL), no
>> problem: |||ERROR 559 CALL: Action failed|||
>> >
>> > How to fix it
>> >
>> > Thank you
>> > Ha`
>> >
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>> --
>> Meftah Tayeb
>> inum: +883510001288000
>> Phone: +13602276297
>> Fax: +12538020313
>>
>>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From johns1433 at gmail.com  Tue Dec  7 16:26:35 2010
From: johns1433 at gmail.com (John Smith)
Date: Tue, 7 Dec 2010 14:26:35 +0100
Subject: [Freeswitch-users] creating conferences
In-Reply-To: <mailman.105.1291654867.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.105.1291654867.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <AANLkTikKoHnemGu9OhWaQ7X0EE1V5X3bEdjjf2cxF3Ds@mail.gmail.com>

Hi All,

I would like to develop a web application to create conferences and their
associated PIN.
As a first step, I'm only developing a simple command line application to
perform this task.

I would like to know if the way I plan to do it is correct or if there is an
even more simple way to do it.
I'm new to Freeswitch and I'm very impressed by FreeSwitch capacilities. I
suspect there is a simple way to do it but I couldn't find it.
Below is what I'm planning to do:

1. create a client-server application to generate xml files in
conf/dialplan/public (the participants calling the conference won't be known
by Freeswitch so I use the public directory instead of the default one).
   Each of these xml files will contain a conference. For example:
 <extension name="conf_100001">
    <condition field="destination_number" expression="100001">
       <action application="conference" data="conf_100001+123123" />
    </condition>
 </extension>

2. Once this new xml file has been created, use the api to call reload_xml

Thanks for your help

John
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From srinivas.ksvreddy at gmail.com  Tue Dec  7 17:52:49 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 7 Dec 2010 20:22:49 +0530
Subject: [Freeswitch-users] error in mod_managed.so
Message-ID: <AANLkTin641=qPhCtob727Hab_w+TLOOPz+qAMXa_a_QC@mail.gmail.com>

HI All,

We are planning to migrate windows to Centos, i am using the mono2.8.1 to
build managed code,

i am successfull in creating the mod_managed.so file, when running the
freeswitch i am getting the follwing error and its not loading the
mod_managed.so.

*undefind symbol: mono_class_from_name.*
**
any idea?

Thanks
Srinivasula Reddy K
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From infos at madovsky.org  Tue Dec  7 19:07:50 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 7 Dec 2010 11:07:50 -0500
Subject: [Freeswitch-users] tone_detect and transfer
Message-ID: <5322385EB6644F2AB42F9A44790D5298@e1705>

If I do this :

<action application="tone_detect" data="fax 1100 r +5000 transfer 'newext XML features' 1" />

are the channel variables transferred automatically or must I set every vars with inline="true" ?

Thanks

Franck

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From davidjbrazier at gmail.com  Tue Dec  7 19:17:57 2010
From: davidjbrazier at gmail.com (David Brazier)
Date: Tue, 7 Dec 2010 16:17:57 +0000
Subject: [Freeswitch-users] error in mod_managed.so
In-Reply-To: <AANLkTin641=qPhCtob727Hab_w+TLOOPz+qAMXa_a_QC@mail.gmail.com>
References: <AANLkTin641=qPhCtob727Hab_w+TLOOPz+qAMXa_a_QC@mail.gmail.com>
Message-ID: <AANLkTinNF5UN3cnzL06LbA5PELRWe8XD==OzA1PtAcSJ@mail.gmail.com>

Is your newly-built mod_managed.so installed in
/usr/local/freeswitch/mod (or wherever your FS module directory is)?


From Avi at aMarcus.com  Tue Dec  7 19:25:45 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Tue, 7 Dec 2010 18:25:45 +0200
Subject: [Freeswitch-users] creating conferences
In-Reply-To: <AANLkTikKoHnemGu9OhWaQ7X0EE1V5X3bEdjjf2cxF3Ds@mail.gmail.com>
References: <mailman.105.1291654867.24555.freeswitch-users@lists.freeswitch.org>
	<AANLkTikKoHnemGu9OhWaQ7X0EE1V5X3bEdjjf2cxF3Ds@mail.gmail.com>
Message-ID: <AANLkTikgn=LonPR8VC+Oo9EVp9=VRnbKjeuJH6sdgNQQ@mail.gmail.com>

Hi.
1) If you want to create XML, it may be easier to use xml_curl to serve up
the xml dynamically rather than doing a reload.

2) The web GUI FusionPBX has a developed conference creation, assignation,
and management interface. I hate seeing people rebuilding things from
scratch unnecessarily. See if it suits your need.

-Avi

On Tue, Dec 7, 2010 at 3:26 PM, John Smith <johns1433 at gmail.com> wrote:

>  Hi All,
>
> I would like to develop a web application to create conferences and their
> associated PIN.
> As a first step, I'm only developing a simple command line application to
> perform this task.
>
> I would like to know if the way I plan to do it is correct or if there is
> an even more simple way to do it.
> I'm new to Freeswitch and I'm very impressed by FreeSwitch capacilities. I
> suspect there is a simple way to do it but I couldn't find it.
> Below is what I'm planning to do:
>
> 1. create a client-server application to generate xml files in
> conf/dialplan/public (the participants calling the conference won't be known
> by Freeswitch so I use the public directory instead of the default one).
>    Each of these xml files will contain a conference. For example:
>  <extension name="conf_100001">
>     <condition field="destination_number" expression="100001">
>        <action application="conference" data="conf_100001+123123" />
>     </condition>
>  </extension>
>
> 2. Once this new xml file has been created, use the api to call reload_xml
>
> Thanks for your help
>
> John
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Dec  7 19:43:18 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 7 Dec 2010 10:43:18 -0600
Subject: [Freeswitch-users] tone_detect and transfer
In-Reply-To: <5322385EB6644F2AB42F9A44790D5298@e1705>
References: <5322385EB6644F2AB42F9A44790D5298@e1705>
Message-ID: <C30DE747-642E-4F49-B820-C44914026C5C@freeswitch.org>

If you are not doing conditions on variables you just set the line above then NO you don't need inline and yes the variables are on the session during all transfers.

/b

On Dec 7, 2010, at 10:07 AM, Madovsky wrote:

> If I do this :
>  
> <action application="tone_detect" data="fax 1100 r +5000 transfer 'newext XML features' 1" />
>  
> are the channel variables transferred automatically or must I set every vars with inline="true" ?
>  
> Thanks
> 
> Franck

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From haloha201 at yahoo.com  Tue Dec  7 17:52:07 2010
From: haloha201 at yahoo.com (ha do)
Date: Tue, 7 Dec 2010 06:52:07 -0800 (PST)
Subject: [Freeswitch-users] get error on skypopen module
In-Reply-To: <AANLkTi=Ujq7KCHWsD-PSUdLDkTs9GUxQak5Ln9Z6s4XT@mail.gmail.com>
Message-ID: <342779.59516.qm@web32405.mail.mud.yahoo.com>

Hi Steven

yes, i use the snd-dummy as guide in startskype.sh script


thank you for your guide, it works now :D
Ha`

--- On Tue, 12/7/10, Steven Ayre <steveayre at gmail.com> wrote:

> From: Steven Ayre <steveayre at gmail.com>
> Subject: Re: [Freeswitch-users] get error on skypopen module
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Tuesday, December 7, 2010, 6:40 AM
> > how to turn off the debug??
> remove the below line in skypopen.xml.conf
> >? ? <param name="debug" value="8"/>
> 
> Yes, remove that line, then do reloadxml and reload
> mod_skypopen to
> pick up the new config. Or restart FS.
> 
> > i do check the CPU performance so 1 call from/to skype
> client it will cost 5% CPU of server
> > if 2 calls, CPU will be 10%
> >
> > i dont know that is normal when freeswitch run with
> skypopen module
> 
> Are you using the dummy sound driver? That's meant to
> reduce the CPU
> usage, but I don't know how much. % will also depend on
> what CPU
> you're using.
> 
> > how do i change the
> > Would you like to receive list mail batched in a daily
> digest?
> > from yes to no? in http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> 1) Visit that link
> 2) Enter your email and click 'Unsubscribe or edit options'
> near the bottom
> 3) Enter your password (there's a reminder button that'll
> email it to
> you if you don't know it)
> 4) Scroll down to the 'Your FreeSWITCH-users Subscription
> Options' section
> 5) Flip the 'Set Digest Mode' setting to Off and click
> 'Submit my changes'
> 
> -Steve
> 
> 
> 
> 
> 
> 
> On 7 December 2010 12:29, ha do <haloha201 at yahoo.com>
> wrote:
> > Hi Giovanny
> >
> > how to turn off the debug?? remove the below line in
> skypopen.xml.conf
> > ? ?<param name="debug" value="8"/>
> >
> > the error i get when i run the
> /usr/local/freeswitch/bin/fs_cli
> >
> > i do check the CPU performance so 1 call from/to skype
> client it will cost 5% CPU of server
> > if 2 calls, CPU will be 10%
> >
> > i dont know that is normal when freeswitch run with
> skypopen module
> >
> >
> > how do i change the
> > Would you like to receive list mail batched in a daily
> digest?
> > from yes to no ?in http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> >
> > Thank you
> > Ha
> >
> >
> > On Tue, Dec 7, 2010 at 5:34 AM, ha do <haloha201 at yahoo.com>
> wrote:
> >> the skypopen works fine but i get error below in
> the debug mode
> >
> > don't run in debug mode if you are not debugging :)
> >
> >> [interface1 ? ? ][IDLE,IDLE] Skype got ERROR
> about a failed action (probably TRYING to HANGUP A CALL), no
> problem: |||ERROR 559 CALL: Action failed|||
> >
> > as is clearly written, this is not a problem. Is a
> debug message that
> > explains we tried to hangup a skype call that the
> skype client closed
> > before. Is just to be double sure that we try to close
> it anyway. The
> > debug message is just an explanation.
> >
> >>
> >> How to fix it
> >
> > nothing to fix, no problems here.
> >
> > -giovanni
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From haloha201 at yahoo.com  Tue Dec  7 18:00:27 2010
From: haloha201 at yahoo.com (ha do)
Date: Tue, 7 Dec 2010 07:00:27 -0800 (PST)
Subject: [Freeswitch-users] get error on skypopen module
In-Reply-To: <AANLkTikvBq2__wxPK8PYfqMLXSuQv3OuKVRUOtpCx3eR@mail.gmail.com>
Message-ID: <320728.95894.qm@web32403.mail.mud.yahoo.com>

Hi Steven

the skypopen module generate the events which are same as other modules
so i have to wait the 'sk list' command which maybe support to event socket in future :D


Thank you
Ha`
--- On Tue, 12/7/10, Steven Ayre <steveayre at gmail.com> wrote:

> From: Steven Ayre <steveayre at gmail.com>
> Subject: Re: [Freeswitch-users] get error on skypopen module
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Tuesday, December 7, 2010, 6:48 AM
> > i have a question on event
> socket, what is the event name for skypopen to monitor the
> interface IDLE/not IDLE or ANSWER or IN PROGRESS...
> 
> Full ESL event list: http://wiki.freeswitch.org/wiki/Event_list
> 
> Skyopen doesn't generate events for state changes itself.
> 
> Check for CHANNEL_ANSWER and
> CHANNEL_PROGRESS/CHANNEL_PROGRESS_MEDIA.
> 
> For spotting an idle skype interface, CHANNEL_CREATE and
> CHANNEL_DESTROY might tell you that by telling you when the
> interface
> is in use.
> 
> If you want to poll for the interface states you can use
> the sk list
> api function via ESL, although there's unfortunately no 'as
> xml'
> formatting at the moment so it'll be slightly trickier to
> parse.
> 
> -Steve
> 
> On 7 December 2010 12:15, ha do <haloha201 at yahoo.com>
> wrote:
> > Hi Meftab
> >
> >> did you autorised skypopen to access
> >> skype from the skype client?
> > you mean the skypopen_auth ??? if so, the answer is
> yes
> >
> >> did you configured the skypopen interfaces to each
> one
> >> of the skype
> > i dont understand, because i use multi skypy
> client(skype software) with 1 username and 1 password
> > and use the sample config file in
> $source.../../configs/multiple-instances-same-skype-user/
> >
> > 2 skype clients run in difference folder
> > /home/cucku/multi/interfaces01
> > /home/cucku/multi/interfaces02
> >
> > so i only need to config the skypopen.conf.xml look
> like below:
> > <configuration name="skypopen.conf"
> description="Skypopen Configuration">
> > ?<global_settings>
> > ? ?<param name="debug" value="8"/>
> > ? ?<param name="dialplan" value="XML"/>
> > ? ?<param name="context" value="default"/>
> > ? ?<param name="destination" value="5000"/>
> > ? ?<param name="skype_user"
> value="do_nguyen_ha"/>
> > ? ?<param name="report_incoming_chatmessages"
> value="false"/>
> > ? ?<param name="silent_mode" value="false"/>
> > ? ?<param name="write_silence_when_idle"
> value="true"/>
> > ? ?<param name="setsockopt" value="false"/>
> > ?</global_settings>
> > ?<!-- one entry here per each skypopen interface
> -->
> > ?<per_interface_settings>
> > ? ?<interface id="1" name="interface1">
> > ? ? ? ?<param name="X11-display"
> value=":101"/>
> > ? ?</interface>
> > ? ?<interface id="2" name="interface2">
> > ? ? ? ?<param name="X11-display"
> value=":102"/>
> > ? ?</interface>
> > ?</per_interface_settings>
> > </configuration>
> >
> > does the conifg look ok?? if not, please guide me to
> make it right
> >
> > i have a question on event socket, what is the event
> name for skypopen to monitor the interface IDLE/not IDLE or
> ANSWER or IN PROGRESS...
> >
> > which event plain should i take care of
> >
> >
> > Thank you
> > Ha`
> > --- On Tue, 12/7/10, Meftah Tayeb <tayeb.meftah at gmail.com>
> wrote:
> >
> >> From: Meftah Tayeb <tayeb.meftah at gmail.com>
> >> Subject: Re: [Freeswitch-users] get error on
> skypopen module
> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> >> Cc: "ha do" <haloha201 at yahoo.com>
> >> Date: Tuesday, December 7, 2010, 4:43 AM
> >> did you autorised skypopen to access
> >> skype from the skype client?
> >> and did you configured the skypopen interfaces to
> each one
> >> of the skype
> >> clients?
> >> thanks
> >> Le 07/12/2010 05:34, ha do a ?crit :
> >> > Hi list
> >> >
> >> > i setup freeswitch and skypopen running fine
> >> >
> >> > there are 2 skype clients run on freeswitch
> >> > freeswitch at internal>? sk list
> >> > sk console is NOT yet assigned
> >> > F ID? ? ? ? Name
> >> ? ? IB (F/T)? ? OB (F/T)
> >> State???CallFlw
> >> ???UUID
> >> > = ====? ? ========
> >> =======? ???=======
> >> ???======? ============
> >> ======
> >> >? ? 1
> >> ???[interface1]? ? ? 0/1
> >> ? ? ? 3/7
> >> IDLE? ? IDLE
> >> >? ? 2
> >> ???[interface2]? ? ? 0/5
> >> ? ? ? 1/5
> >> IDLE? ? IDLE
> >> >
> >> > the skype clients are used the same username
> +
> >> password of skype account
> >> >
> >> >
> >> > the skypopen works fine but i get error below
> in the
> >> debug mode
> >> >
> >> > 2010-12-07 04:24:35.794611 [DEBUG]
> >> switch_core_state_machine.c:462
> (skypopen/interface1) State
> >> DESTROY going to sleep
> >> > 2010-12-07 04:24:35.795807 [DEBUG]
> >> skypopen_protocol.c:173? ???[|]
> >> [DEBUG_SKYPE? 173? ][interface1
> >> ???][IDLE,IDLE] READING: |||ERROR 559 CALL:
> >> Action failed|||
> >> > 2010-12-07 04:24:35.795807 [DEBUG]
> >> skypopen_protocol.c:228? ???[|]
> >> [DEBUG_SKYPE? 228? ][interface1
> >> ???][IDLE,IDLE] Skype got ERROR about a
> >> failed action (probably TRYING to HANGUP A CALL),
> no
> >> problem: |||ERROR 559 CALL: Action failed|||
> >> > 2010-12-07 04:24:35.795807 [DEBUG]
> >> skypopen_protocol.c:173? ???[|]
> >> [DEBUG_SKYPE? 173? ][interface1
> >> ???][IDLE,IDLE] READING: |||ERROR 559 CALL:
> >> Action failed|||
> >> > 2010-12-07 04:24:35.795807 [DEBUG]
> >> skypopen_protocol.c:228? ???[|]
> >> [DEBUG_SKYPE? 228? ][interface1
> >> ???][IDLE,IDLE] Skype got ERROR about a
> >> failed action (probably TRYING to HANGUP A CALL),
> no
> >> problem: |||ERROR 559 CALL: Action failed|||
> >> >
> >> > How to fix it
> >> >
> >> > Thank you
> >> > Ha`
> >> >
> >> >
> >> >
> >> >
> >> >
> _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >>
> >> --
> >> Meftah Tayeb
> >> inum: +883510001288000
> >> Phone: +13602276297
> >> Fax: +12538020313
> >>
> >>
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From infos at madovsky.org  Tue Dec  7 21:27:28 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 7 Dec 2010 13:27:28 -0500
Subject: [Freeswitch-users] api_hangup_hook question
Message-ID: <4BEF6419594F4281874094369DC3289A@e1705>

I have this in dialplan/default

        <extension name="9999999999">
                <condition field="destination_number" expression="^(9999999999)$">
                        <action application="set" data="domain_name=$${domain}"/>
                        <action application="export" data="dialed_extension=$1"/>
                        <action application="set" data="hangup_after_bridge=true"/>
                        <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,CALL_REJECTED,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
                        <action application="ring_ready"/>
                        <action application="tone_detect" data="fax 1100 r +5000 transfer 'receivefax XML features' 1" />
                        <action application="export" data="originate_timeout=15"/>
                        <action application="bridge" data="{absolute_codec_string='speex at 16000k,G729,GSM,G726,PCMU,PCMA',sip_from_uri=${sip_from_user}@${sip_from_host},origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number}}user/${dialed_extension}"/>
                        <action application="sleep" data="3000"/>
                        <action application="answer"/>
                        <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
                </condition>
        </extension>


and this in dialplan/features


    <extension name="receivefax">
      <condition field="destination_number" expression="^receivefax$">
        <action application="set" data="session_in_hangup_hook=true"/>
        <action application="set" data="fax_file=/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
        <action application="set" data="fax_mode=recv"/>
        <action application="set" data="api_hangup_hook=perl /usr/local/freeswitch/scripts/perl/faxStatus.pl"/>
        <action application="answer" />
        <action application="playback" data="silence_stream://2000"/>
        <action application="set" data="fax_enable_t38_request=true"/>
        <action application="set" data="fax_enable_t38=true"/>
        <action application="rxfax" data="/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
        <action application="hangup"/>
      </condition>
    </extension>


the fax is received well, but the perl script is not executed.

Am I missing anything ?

Thanks

Franck

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From ranjtech at gmail.com  Tue Dec  7 21:53:55 2010
From: ranjtech at gmail.com (RR)
Date: Tue, 7 Dec 2010 13:53:55 -0500
Subject: [Freeswitch-users] FreeSWITCH 64bit compilation error (Solaris
	10)
In-Reply-To: <AANLkTikWGfii8zhPRtXLRWUf_0zkyB_jJCo1yT80Jn83@mail.gmail.com>
References: <AANLkTikWGfii8zhPRtXLRWUf_0zkyB_jJCo1yT80Jn83@mail.gmail.com>
Message-ID: <AANLkTikR3E3sVU7_Qw+Lm2QFOHnLag+3Y+5G6j+kKxiN@mail.gmail.com>

On Tue, Dec 7, 2010 at 7:51 AM, Kim Culhan <w8hdkim at gmail.com> wrote:

>  On Tue, December 7, 2010 5:51 am, RR wrote:
> > Hi Guys,
> >
> > I have a really stupid question to ask. Having skimmed through the steps
> to
> > take to compile FreeSWITCH in a Solaris environment, everywhere it seems
> to
> > talk about the jds-cbe and sun studio etc...basically meaning that I have
> to
> > have the X11 framework/graphical environment/packages installed on the
> > machine? Is it possible to compile it without this using the standard,
> > ./configure, make combo and the standard gcc etc?
> Installing the jds-cbe and sun studio does not involve installing 'the X11
> framework' on the machine.
>
> refer to::
>
> http://wiki.freeswitch.org/wiki/Installation_Guide#OpenSolaris_os200906
>
> A step in the above refers to: /opt/dtbld/bin/env.sh which is installed by
> the jds-cbe
>
> The script sets some variables in the environment which are read by
> Freeswitch configure
> as Mike mentioned:
>
> > On Wed, Sep 23, 2009 at 4:20 AM, Michael Jerris <mike at jerris.com> wrote:
> >
> >> Try taking a list at the info here:
> >> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#SolarisYou need
> >> to be passing any necessary cflags in on configure
> Please follow the guide for compiling on Opensolaris and post if you run
> into any problems.
> **
> Hope this helps.
>
> -kim
>

Thanks Kim,

I'm guessing that Sun Studio along with jds-cbe runs using the JRE/Java etc
which is why it doens't need the X11 framework? I am hoping that is the case
as I'm running Solaris inside VMs and having trouble with bringing up the
GUI desktop interface on OpenSolaris although it comes up withour a problem
on Solaris 10. I thought it was Solaris that had problems with drivers etc.
but it had no problems running OpenSolaris inside a VM on the same VM Host
fails to bring up the desktop.

Either way, it's awesome if I don't need to have the X11 framework in place
to compile FS in Solaris.
Thanks
\RR
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From steveayre at gmail.com  Tue Dec  7 22:42:42 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 7 Dec 2010 19:42:42 +0000
Subject: [Freeswitch-users] get error on skypopen module
In-Reply-To: <320728.95894.qm@web32403.mail.mud.yahoo.com>
References: <AANLkTikvBq2__wxPK8PYfqMLXSuQv3OuKVRUOtpCx3eR@mail.gmail.com>
	<320728.95894.qm@web32403.mail.mud.yahoo.com>
Message-ID: <AANLkTinRG+qV-O_NJ2NnAn2f7xKo-3dm1m+Z+EjKLJVV@mail.gmail.com>

> the skypopen module generate the events which are same as other modules

That's because they're channel events generated by the freeswitch
core. But there may be variables in them that you can use to identify
the Skypopen interface used.

> so i have to wait the 'sk list' command which maybe support to event socket in future :D

It's an API command so it's supported right now - just do "api sk
list". You just won't get the response as XML.

-Steve



On 7 December 2010 15:00, ha do <haloha201 at yahoo.com> wrote:
> Hi Steven
>
> the skypopen module generate the events which are same as other modules
> so i have to wait the 'sk list' command which maybe support to event socket in future :D
>
>
> Thank you
> Ha`
> --- On Tue, 12/7/10, Steven Ayre <steveayre at gmail.com> wrote:
>
>> From: Steven Ayre <steveayre at gmail.com>
>> Subject: Re: [Freeswitch-users] get error on skypopen module
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Date: Tuesday, December 7, 2010, 6:48 AM
>> > i have a question on event
>> socket, what is the event name for skypopen to monitor the
>> interface IDLE/not IDLE or ANSWER or IN PROGRESS...
>>
>> Full ESL event list: http://wiki.freeswitch.org/wiki/Event_list
>>
>> Skyopen doesn't generate events for state changes itself.
>>
>> Check for CHANNEL_ANSWER and
>> CHANNEL_PROGRESS/CHANNEL_PROGRESS_MEDIA.
>>
>> For spotting an idle skype interface, CHANNEL_CREATE and
>> CHANNEL_DESTROY might tell you that by telling you when the
>> interface
>> is in use.
>>
>> If you want to poll for the interface states you can use
>> the sk list
>> api function via ESL, although there's unfortunately no 'as
>> xml'
>> formatting at the moment so it'll be slightly trickier to
>> parse.
>>
>> -Steve
>>
>> On 7 December 2010 12:15, ha do <haloha201 at yahoo.com>
>> wrote:
>> > Hi Meftab
>> >
>> >> did you autorised skypopen to access
>> >> skype from the skype client?
>> > you mean the skypopen_auth ??? if so, the answer is
>> yes
>> >
>> >> did you configured the skypopen interfaces to each
>> one
>> >> of the skype
>> > i dont understand, because i use multi skypy
>> client(skype software) with 1 username and 1 password
>> > and use the sample config file in
>> $source.../../configs/multiple-instances-same-skype-user/
>> >
>> > 2 skype clients run in difference folder
>> > /home/cucku/multi/interfaces01
>> > /home/cucku/multi/interfaces02
>> >
>> > so i only need to config the skypopen.conf.xml look
>> like below:
>> > <configuration name="skypopen.conf"
>> description="Skypopen Configuration">
>> > ?<global_settings>
>> > ? ?<param name="debug" value="8"/>
>> > ? ?<param name="dialplan" value="XML"/>
>> > ? ?<param name="context" value="default"/>
>> > ? ?<param name="destination" value="5000"/>
>> > ? ?<param name="skype_user"
>> value="do_nguyen_ha"/>
>> > ? ?<param name="report_incoming_chatmessages"
>> value="false"/>
>> > ? ?<param name="silent_mode" value="false"/>
>> > ? ?<param name="write_silence_when_idle"
>> value="true"/>
>> > ? ?<param name="setsockopt" value="false"/>
>> > ?</global_settings>
>> > ?<!-- one entry here per each skypopen interface
>> -->
>> > ?<per_interface_settings>
>> > ? ?<interface id="1" name="interface1">
>> > ? ? ? ?<param name="X11-display"
>> value=":101"/>
>> > ? ?</interface>
>> > ? ?<interface id="2" name="interface2">
>> > ? ? ? ?<param name="X11-display"
>> value=":102"/>
>> > ? ?</interface>
>> > ?</per_interface_settings>
>> > </configuration>
>> >
>> > does the conifg look ok?? if not, please guide me to
>> make it right
>> >
>> > i have a question on event socket, what is the event
>> name for skypopen to monitor the interface IDLE/not IDLE or
>> ANSWER or IN PROGRESS...
>> >
>> > which event plain should i take care of
>> >
>> >
>> > Thank you
>> > Ha`
>> > --- On Tue, 12/7/10, Meftah Tayeb <tayeb.meftah at gmail.com>
>> wrote:
>> >
>> >> From: Meftah Tayeb <tayeb.meftah at gmail.com>
>> >> Subject: Re: [Freeswitch-users] get error on
>> skypopen module
>> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> >> Cc: "ha do" <haloha201 at yahoo.com>
>> >> Date: Tuesday, December 7, 2010, 4:43 AM
>> >> did you autorised skypopen to access
>> >> skype from the skype client?
>> >> and did you configured the skypopen interfaces to
>> each one
>> >> of the skype
>> >> clients?
>> >> thanks
>> >> Le 07/12/2010 05:34, ha do a ?crit :
>> >> > Hi list
>> >> >
>> >> > i setup freeswitch and skypopen running fine
>> >> >
>> >> > there are 2 skype clients run on freeswitch
>> >> > freeswitch at internal>? sk list
>> >> > sk console is NOT yet assigned
>> >> > F ID? ? ? ? Name
>> >> ? ? IB (F/T)? ? OB (F/T)
>> >> State???CallFlw
>> >> ???UUID
>> >> > = ====? ? ========
>> >> =======? ???=======
>> >> ???======? ============
>> >> ======
>> >> >? ? 1
>> >> ???[interface1]? ? ? 0/1
>> >> ? ? ? 3/7
>> >> IDLE? ? IDLE
>> >> >? ? 2
>> >> ???[interface2]? ? ? 0/5
>> >> ? ? ? 1/5
>> >> IDLE? ? IDLE
>> >> >
>> >> > the skype clients are used the same username
>> +
>> >> password of skype account
>> >> >
>> >> >
>> >> > the skypopen works fine but i get error below
>> in the
>> >> debug mode
>> >> >
>> >> > 2010-12-07 04:24:35.794611 [DEBUG]
>> >> switch_core_state_machine.c:462
>> (skypopen/interface1) State
>> >> DESTROY going to sleep
>> >> > 2010-12-07 04:24:35.795807 [DEBUG]
>> >> skypopen_protocol.c:173? ???[|]
>> >> [DEBUG_SKYPE? 173? ][interface1
>> >> ???][IDLE,IDLE] READING: |||ERROR 559 CALL:
>> >> Action failed|||
>> >> > 2010-12-07 04:24:35.795807 [DEBUG]
>> >> skypopen_protocol.c:228? ???[|]
>> >> [DEBUG_SKYPE? 228? ][interface1
>> >> ???][IDLE,IDLE] Skype got ERROR about a
>> >> failed action (probably TRYING to HANGUP A CALL),
>> no
>> >> problem: |||ERROR 559 CALL: Action failed|||
>> >> > 2010-12-07 04:24:35.795807 [DEBUG]
>> >> skypopen_protocol.c:173? ???[|]
>> >> [DEBUG_SKYPE? 173? ][interface1
>> >> ???][IDLE,IDLE] READING: |||ERROR 559 CALL:
>> >> Action failed|||
>> >> > 2010-12-07 04:24:35.795807 [DEBUG]
>> >> skypopen_protocol.c:228? ???[|]
>> >> [DEBUG_SKYPE? 228? ][interface1
>> >> ???][IDLE,IDLE] Skype got ERROR about a
>> >> failed action (probably TRYING to HANGUP A CALL),
>> no
>> >> problem: |||ERROR 559 CALL: Action failed|||
>> >> >
>> >> > How to fix it
>> >> >
>> >> > Thank you
>> >> > Ha`
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >>
>> >>
>> >> --
>> >> Meftah Tayeb
>> >> inum: +883510001288000
>> >> Phone: +13602276297
>> >> Fax: +12538020313
>> >>
>> >>
>> >
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From tayeb.meftah at gmail.com  Tue Dec  7 23:31:48 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 07 Dec 2010 21:31:48 +0100
Subject: [Freeswitch-users] api_hangup_hook question
In-Reply-To: <4BEF6419594F4281874094369DC3289A@e1705>
References: <4BEF6419594F4281874094369DC3289A@e1705>
Message-ID: <4CFE99B4.3090703@gmail.com>

please can you try this:
1. run the info app from inside your perl script
and see what's going on
otherwise:
<action application="set" 
data="api_hangup_hook=system(/usr/local/freeswitch/scripts/perl/faxStatus.pl)"/>
if the info app run so your perl script is likely called
le 07/12/2010 19:27, Madovsky a ?crit :
> I have this in dialplan/default
> <extension name="9999999999">
> <condition field="destination_number" expression="^(9999999999)$">
> <action application="set" data="domain_name=$${domain}"/>
> <action application="export" data="dialed_extension=$1"/>
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="set" 
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,CALL_REJECTED,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
> <action application="ring_ready"/>
> <action application="tone_detect" data="fax 1100 r +5000 transfer 
> 'receivefax XML features' 1" />
> <action application="export" data="originate_timeout=15"/>
> <action application="bridge" 
> data="{absolute_codec_string='speex at 16000k,G729,GSM,G726,PCMU,PCMA',sip_from_uri=${sip_from_user}@${sip_from_host},origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number}}user/${dialed_extension}"/ 
> <mailto:%7Babsolute_codec_string=%27speex at 16000k,G729,GSM,G726,PCMU,PCMA%27,sip_from_uri=$%7Bsip_from_user%7D@$%7Bsip_from_host%7D,origination_caller_id_name=$%7Bcaller_id_name%7D,origination_caller_id_number=$%7Bcaller_id_number%7D%7Duser/$%7Bdialed_extension%7D%22/>>
> <action application="sleep" data="3000"/>
> <action application="answer"/>
> <action application="voicemail" data="default ${domain_name} 
> ${dialed_extension}"/>
> </condition>
> </extension>
> and this in dialplan/features
> <extension name="receivefax">
> <condition field="destination_number" expression="^receivefax$">
> <action application="set" data="session_in_hangup_hook=true"/>
> <action application="set" 
> data="fax_file=/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
> <action application="set" data="fax_mode=recv"/>
> <action application="set" data="api_hangup_hook=perl 
> /usr/local/freeswitch/scripts/perl/faxStatus.pl"/>
> <action application="answer" />
> <action application="playback" data="silence_stream://2000"/>
> <action application="set" data="fax_enable_t38_request=true"/>
> <action application="set" data="fax_enable_t38=true"/>
> <action application="rxfax" 
> data="/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
> <action application="hangup"/>
> </condition>
> </extension>
> the fax is received well, but the perl script is not executed.
> Am I missing anything ?
> Thanks
> Franck
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    



-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313

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From msc at freeswitch.org  Tue Dec  7 23:51:27 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 7 Dec 2010 12:51:27 -0800
Subject: [Freeswitch-users] api_hangup_hook question
In-Reply-To: <4BEF6419594F4281874094369DC3289A@e1705>
References: <4BEF6419594F4281874094369DC3289A@e1705>
Message-ID: <AANLkTinNnpAyEAfsx56bciyjwUSfqAN=arJxbvhV-mAm@mail.gmail.com>

Can you pastebin your logs? There might be a clue as to what is happening.
-MC

On Tue, Dec 7, 2010 at 10:27 AM, Madovsky <infos at madovsky.org> wrote:

>  I have this in dialplan/default
>
>         <extension name="9999999999">
>                 <condition field="destination_number"
> expression="^(9999999999)$">
>                         <action application="set"
> data="domain_name=$${domain}"/>
>                         <action application="export"
> data="dialed_extension=$1"/>
>                         <action application="set"
> data="hangup_after_bridge=true"/>
>                         <action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,CALL_REJECTED,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
>                         <action application="ring_ready"/>
>                         <action application="tone_detect" data="fax 1100 r
> +5000 transfer 'receivefax XML features' 1" />
>                         <action application="export"
> data="originate_timeout=15"/>
>                         <action application="bridge" data="
> {absolute_codec_string='speex at 16000k,G729,GSM,G726,PCMU,PCMA',sip_from_uri=${sip_from_user}@${sip_from_host},origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number}}user/${dialed_extension}"/<%7Babsolute_codec_string='speex at 16000k,G729,GSM,G726,PCMU,PCMA',sip_from_uri=$%7Bsip_from_user%7D@$%7Bsip_from_host%7D,origination_caller_id_name=$%7Bcaller_id_name%7D,origination_caller_id_number=$%7Bcaller_id_number%7D%7Duser/$%7Bdialed_extension%7D%22/>
> >
>                         <action application="sleep" data="3000"/>
>                         <action application="answer"/>
>                         <action application="voicemail" data="default
> ${domain_name} ${dialed_extension}"/>
>                 </condition>
>         </extension>
>
> and this in dialplan/features
>
>
>     <extension name="receivefax">
>       <condition field="destination_number" expression="^receivefax$">
>         <action application="set" data="session_in_hangup_hook=true"/>
>         <action application="set"
> data="fax_file=/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
>         <action application="set" data="fax_mode=recv"/>
>         <action application="set" data="api_hangup_hook=perl
> /usr/local/freeswitch/scripts/perl/faxStatus.pl"/>
>         <action application="answer" />
>         <action application="playback" data="silence_stream://2000"/>
>         <action application="set" data="fax_enable_t38_request=true"/>
>         <action application="set" data="fax_enable_t38=true"/>
>         <action application="rxfax"
> data="/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
>         <action application="hangup"/>
>       </condition>
>     </extension>
>
>
> the fax is received well, but the perl script is not executed.
>
> Am I missing anything ?
>
> Thanks
>
> Franck
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tayeb.meftah at gmail.com  Tue Dec  7 23:49:14 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 07 Dec 2010 21:49:14 +0100
Subject: [Freeswitch-users] api_hangup_hook question
In-Reply-To: <4BEF6419594F4281874094369DC3289A@e1705>
References: <4BEF6419594F4281874094369DC3289A@e1705>
Message-ID: <4CFE9DCA.1050207@gmail.com>

can you join irc please?
#freeswitch in freenode.net
http://conference.freeswitch.org and enter yourname and join
waiting for you to help you there.
Le 07/12/2010 19:27, Madovsky a ?crit :
> I have this in dialplan/default
> <extension name="9999999999">
> <condition field="destination_number" expression="^(9999999999)$">
> <action application="set" data="domain_name=$${domain}"/>
> <action application="export" data="dialed_extension=$1"/>
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="set" 
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,CALL_REJECTED,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
> <action application="ring_ready"/>
> <action application="tone_detect" data="fax 1100 r +5000 transfer 
> 'receivefax XML features' 1" />
> <action application="export" data="originate_timeout=15"/>
> <action application="bridge" 
> data="{absolute_codec_string='speex at 16000k,G729,GSM,G726,PCMU,PCMA',sip_from_uri=${sip_from_user}@${sip_from_host},origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number}}user/${dialed_extension}"/ 
> <mailto:%7Babsolute_codec_string=%27speex at 16000k,G729,GSM,G726,PCMU,PCMA%27,sip_from_uri=$%7Bsip_from_user%7D@$%7Bsip_from_host%7D,origination_caller_id_name=$%7Bcaller_id_name%7D,origination_caller_id_number=$%7Bcaller_id_number%7D%7Duser/$%7Bdialed_extension%7D%22/>>
> <action application="sleep" data="3000"/>
> <action application="answer"/>
> <action application="voicemail" data="default ${domain_name} 
> ${dialed_extension}"/>
> </condition>
> </extension>
> and this in dialplan/features
> <extension name="receivefax">
> <condition field="destination_number" expression="^receivefax$">
> <action application="set" data="session_in_hangup_hook=true"/>
> <action application="set" 
> data="fax_file=/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
> <action application="set" data="fax_mode=recv"/>
> <action application="set" data="api_hangup_hook=perl 
> /usr/local/freeswitch/scripts/perl/faxStatus.pl"/>
> <action application="answer" />
> <action application="playback" data="silence_stream://2000"/>
> <action application="set" data="fax_enable_t38_request=true"/>
> <action application="set" data="fax_enable_t38=true"/>
> <action application="rxfax" 
> data="/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
> <action application="hangup"/>
> </condition>
> </extension>
> the fax is received well, but the perl script is not executed.
> Am I missing anything ?
> Thanks
> Franck
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313

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From phone.bytes at gmail.com  Wed Dec  8 00:12:19 2010
From: phone.bytes at gmail.com (Phone)
Date: Tue, 07 Dec 2010 14:12:19 -0700
Subject: [Freeswitch-users] cepstral problem
In-Reply-To: <AANLkTi=tp_JK2H4cHtK8129TnWyYhArxaS2R1jmyPMPr@mail.gmail.com>
References: <AANLkTi=tp_JK2H4cHtK8129TnWyYhArxaS2R1jmyPMPr@mail.gmail.com>
Message-ID: <4CFEA333.1090706@gmail.com>

Just wondering if you have had any luck in resolving this issue. 

We are running Cent 5.5 on FreeSWITCH Version 1.0.head (git-8825b6e 
2010-11-28 17-15-39 -0500) with Cepstral 5.1 with the Callie voice.

We have good audio for a short period of time, then suddenly the audio 
is gone.  Watching the logs, it show that it is trying to play the 
correct TTS, but it is silent.

We have tried to unload and reload mod_cepstral, but the only way we 
have been able to restore the audio is with a FreeSWITCH restart.

These voices sound good, but with these issues, it is really not useable.

Any Ideas?  Thanks



stephen at stephenjc wrote:
> I'm working with cepstral support and they seem to have no idea. Does
> any one know why cepstral might produce no audio on a cent os 5.5
> machine. Its works fine on another box of mine.
>
>
> Thanks,
> Stephen C
> -All of my email addresses go to the same place
> -Save Paper, think before you print
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From renjian at gmail.com  Wed Dec  8 00:14:59 2010
From: renjian at gmail.com (Jian Ren)
Date: Tue, 7 Dec 2010 16:14:59 -0500
Subject: [Freeswitch-users] get error on skypopen module
In-Reply-To: <342779.59516.qm@web32405.mail.mud.yahoo.com>
References: <AANLkTi=Ujq7KCHWsD-PSUdLDkTs9GUxQak5Ln9Z6s4XT@mail.gmail.com>
	<342779.59516.qm@web32405.mail.mud.yahoo.com>
Message-ID: <AANLkTimB+Nkc=S5gbOqrGFBnZo7rXKwVXs7uTc3FrteR@mail.gmail.com>

I am getting 5-6% for each skype call and 1% idle instance, on my Q6600.
It's normal, 20 concurrent calls would be way enough for most people. If you
want more, can scale to more servers.

Jian

On Tue, Dec 7, 2010 at 9:52 AM, ha do <haloha201 at yahoo.com> wrote:

> Hi Steven
>
> yes, i use the snd-dummy as guide in startskype.sh script
>
>
> thank you for your guide, it works now :D
> Ha`
>
> --- On Tue, 12/7/10, Steven Ayre <steveayre at gmail.com> wrote:
>
> > From: Steven Ayre <steveayre at gmail.com>
> > Subject: Re: [Freeswitch-users] get error on skypopen module
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Date: Tuesday, December 7, 2010, 6:40 AM
> > > how to turn off the debug??
> > remove the below line in skypopen.xml.conf
> > >    <param name="debug" value="8"/>
> >
> > Yes, remove that line, then do reloadxml and reload
> > mod_skypopen to
> > pick up the new config. Or restart FS.
> >
> > > i do check the CPU performance so 1 call from/to skype
> > client it will cost 5% CPU of server
> > > if 2 calls, CPU will be 10%
> > >
> > > i dont know that is normal when freeswitch run with
> > skypopen module
> >
> > Are you using the dummy sound driver? That's meant to
> > reduce the CPU
> > usage, but I don't know how much. % will also depend on
> > what CPU
> > you're using.
> >
> > > how do i change the
> > > Would you like to receive list mail batched in a daily
> > digest?
> > > from yes to no  in
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > 1) Visit that link
> > 2) Enter your email and click 'Unsubscribe or edit options'
> > near the bottom
> > 3) Enter your password (there's a reminder button that'll
> > email it to
> > you if you don't know it)
> > 4) Scroll down to the 'Your FreeSWITCH-users Subscription
> > Options' section
> > 5) Flip the 'Set Digest Mode' setting to Off and click
> > 'Submit my changes'
> >
> > -Steve
> >
> >
> >
> >
> >
> >
> > On 7 December 2010 12:29, ha do <haloha201 at yahoo.com>
> > wrote:
> > > Hi Giovanny
> > >
> > > how to turn off the debug?? remove the below line in
> > skypopen.xml.conf
> > >    <param name="debug" value="8"/>
> > >
> > > the error i get when i run the
> > /usr/local/freeswitch/bin/fs_cli
> > >
> > > i do check the CPU performance so 1 call from/to skype
> > client it will cost 5% CPU of server
> > > if 2 calls, CPU will be 10%
> > >
> > > i dont know that is normal when freeswitch run with
> > skypopen module
> > >
> > >
> > > how do i change the
> > > Would you like to receive list mail batched in a daily
> > digest?
> > > from yes to no  in
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > >
> > > Thank you
> > > Ha
> > >
> > >
> > > On Tue, Dec 7, 2010 at 5:34 AM, ha do <haloha201 at yahoo.com>
> > wrote:
> > >> the skypopen works fine but i get error below in
> > the debug mode
> > >
> > > don't run in debug mode if you are not debugging :)
> > >
> > >> [interface1     ][IDLE,IDLE] Skype got ERROR
> > about a failed action (probably TRYING to HANGUP A CALL), no
> > problem: |||ERROR 559 CALL: Action failed|||
> > >
> > > as is clearly written, this is not a problem. Is a
> > debug message that
> > > explains we tried to hangup a skype call that the
> > skype client closed
> > > before. Is just to be double sure that we try to close
> > it anyway. The
> > > debug message is just an explanation.
> > >
> > >>
> > >> How to fix it
> > >
> > > nothing to fix, no problems here.
> > >
> > > -giovanni
> > >
> > > --
> > > Sincerely,
> > >
> > > Giovanni Maruzzelli
> > > Cell : +39-347-2665618
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Dec  8 00:17:36 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 7 Dec 2010 13:17:36 -0800
Subject: [Freeswitch-users] cepstral problem
In-Reply-To: <4CFEA333.1090706@gmail.com>
References: <AANLkTi=tp_JK2H4cHtK8129TnWyYhArxaS2R1jmyPMPr@mail.gmail.com>
	<4CFEA333.1090706@gmail.com>
Message-ID: <AANLkTinwWwfSY2uPhs5c8SuyxnwOn9myWv2u2WiKqGj8@mail.gmail.com>

Can you try the mod_tts_commandline way of doing it? The wiki talks about
using Cepstral IIRC.
-MC

On Tue, Dec 7, 2010 at 1:12 PM, Phone <phone.bytes at gmail.com> wrote:

> Just wondering if you have had any luck in resolving this issue.
>
> We are running Cent 5.5 on FreeSWITCH Version 1.0.head (git-8825b6e
> 2010-11-28 17-15-39 -0500) with Cepstral 5.1 with the Callie voice.
>
> We have good audio for a short period of time, then suddenly the audio
> is gone.  Watching the logs, it show that it is trying to play the
> correct TTS, but it is silent.
>
> We have tried to unload and reload mod_cepstral, but the only way we
> have been able to restore the audio is with a FreeSWITCH restart.
>
> These voices sound good, but with these issues, it is really not useable.
>
> Any Ideas?  Thanks
>
>
>
> stephen at stephenjc wrote:
> > I'm working with cepstral support and they seem to have no idea. Does
> > any one know why cepstral might produce no audio on a cent os 5.5
> > machine. Its works fine on another box of mine.
> >
> >
> > Thanks,
> > Stephen C
> > -All of my email addresses go to the same place
> > -Save Paper, think before you print
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Dec  8 02:00:20 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 7 Dec 2010 17:00:20 -0600
Subject: [Freeswitch-users] sipgate.de
Message-ID: <104F5003-373A-465E-85E3-F6C154026B81@freeswitch.org>

Anyone have FS working with sipgate?

/b



From johnrose at comtex.net  Wed Dec  8 02:02:36 2010
From: johnrose at comtex.net (John Rose)
Date: Tue, 7 Dec 2010 16:02:36 -0700
Subject: [Freeswitch-users] SIP MESSAGE requests
Message-ID: <004201cb9662$d770dad0$86529070$@comtex.net>

Hello,

 

Does anyone know if it is possible to construct, originate and receive SIP
MESSAGE requests in FreeSwitch or libfreeswitch? This would be in a
sessionless dialog as of RFC 3428 Page mode. 

 

Thanks,

John

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From tayeb.meftah at gmail.com  Wed Dec  8 02:32:05 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Wed, 08 Dec 2010 00:32:05 +0100
Subject: [Freeswitch-users] sipgate.de
In-Reply-To: <104F5003-373A-465E-85E3-F6C154026B81@freeswitch.org>
References: <104F5003-373A-465E-85E3-F6C154026B81@freeswitch.org>
Message-ID: <4CFEC3F5.2030600@gmail.com>

i do, brian
anything needed?
thanks
Le 08/12/2010 00:00, Brian West a ?crit :
> Anyone have FS working with sipgate?
>
> /b
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    



-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313



From lloyd.aloysius at gmail.com  Wed Dec  8 02:49:00 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 7 Dec 2010 18:49:00 -0500
Subject: [Freeswitch-users] Video Call
Message-ID: <AANLkTikrzHRkoapcGYrRPbfztop7YCqXsnKZgaAzb_5D@mail.gmail.com>

Hi All,

Does FreeSwitch support video call? Is there any special configurations for
video calls?

I am planning to test with Yealink IP Video Phone.

http://www.yealink.com/en/view.asp?ClassLayer=120

Any help appreciated.

Thanks
Lloyd
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From w8hdkim at gmail.com  Wed Dec  8 03:02:43 2010
From: w8hdkim at gmail.com (Kim Culhan)
Date: Tue, 7 Dec 2010 19:02:43 -0500
Subject: [Freeswitch-users] FreeSWITCH 64bit compilation error (Solaris
	10)
Message-ID: <AANLkTik3FLnHdGpY=7z=K=poqydVac6HBToq8FzfwHZ0@mail.gmail.com>

On Tue, December 7, 2010 1:53 pm, RR wrote:
> On Tue, Dec 7, 2010 at 7:51 AM, Kim Culhan <w8hdkim at gmail.com> wrote:
>
>>  On Tue, December 7, 2010 5:51 am, RR wrote:

> I'm guessing that Sun Studio along with jds-cbe runs using the JRE/Java
etc
> which is why it doens't need the X11 framework?

Actually Sun Studio doesn't use JRE-Java nor does the jds-cbe.

Its really straightforward and doesn't utilize an integrated graphic
environment
the way some other developement products do with 'Studio' in the name :)

> I am hoping that is the case
> as I'm running Solaris inside VMs and having trouble with bringing up the
> GUI desktop interface on OpenSolaris although it comes up withour a
problem
> on Solaris 10. I thought it was Solaris that had problems with drivers
etc.
> but it had no problems running OpenSolaris inside a VM on the same VM Host
> fails to bring up the desktop.
>
> Either way, it's awesome if I don't need to have the X11 framework in
place
> to compile FS in Solaris.
I'm sure you'll have very little problem with this, just install sun studio
on your system
with the instructions for your flavor of [Open]Solaris along with the
jds-cbe and your good to go.

-kim
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From dujinfang at gmail.com  Wed Dec  8 03:05:15 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 8 Dec 2010 08:05:15 +0800
Subject: [Freeswitch-users] Video Call
In-Reply-To: <AANLkTikrzHRkoapcGYrRPbfztop7YCqXsnKZgaAzb_5D@mail.gmail.com>
References: <AANLkTikrzHRkoapcGYrRPbfztop7YCqXsnKZgaAzb_5D@mail.gmail.com>
Message-ID: <AANLkTi==8eaKHDmfDFeBXX1GLh88QQn6ePVbSDMNCM6t@mail.gmail.com>

enable video codecs in vars.xml

On Wed, Dec 8, 2010 at 7:49 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com> wrote:
> Hi All,
> Does FreeSwitch support video call? Is there any special configurations for
> video calls?
> I am planning to test with Yealink?IP Video Phone.
> http://www.yealink.com/en/view.asp?ClassLayer=120
> Any help appreciated.
> Thanks
> Lloyd
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From ranjtech at gmail.com  Wed Dec  8 03:10:32 2010
From: ranjtech at gmail.com (RR)
Date: Tue, 7 Dec 2010 19:10:32 -0500
Subject: [Freeswitch-users] FreeSWITCH 64bit compilation error (Solaris
	10)
In-Reply-To: <AANLkTik3FLnHdGpY=7z=K=poqydVac6HBToq8FzfwHZ0@mail.gmail.com>
References: <AANLkTik3FLnHdGpY=7z=K=poqydVac6HBToq8FzfwHZ0@mail.gmail.com>
Message-ID: <AANLkTimWgkRs-mp8r3Ndjx4Ff90aTb5Bu6rwwmRuLJXz@mail.gmail.com>

Ok Cool thanks.

BTW, here's another dumb question. Have you or anyone else reading this
thread have had better success with Installing, compiling FS on Solaris 10
as compared to OpenSolaris 0906 etc OR the other way around? I am currently
experimenting with trying to compile Asterisk in Soalris for the last 2-3
weeks and it's not going well at all. I have created and destroyed over 4-5
VMs with different packages, libraries, paths, makefile mods and I don't
know what not, but still have not been able to compile it successfully. If
FS is more straightforward to install, then even though my initial platform
architecture was built around Asterisk as the media server, I will change
that to FS and figure out / re-write the code that was written for the
FastAGI for Asterisk.

Any comments from anyone about this will be greatly appreciated.

Thanks in advance,
RR


On Tue, Dec 7, 2010 at 7:02 PM, Kim Culhan <w8hdkim at gmail.com> wrote:

>  On Tue, December 7, 2010 1:53 pm, RR wrote:
> > On Tue, Dec 7, 2010 at 7:51 AM, Kim Culhan <w8hdkim at gmail.com> wrote:
> >
> >>  On Tue, December 7, 2010 5:51 am, RR wrote:
>
> > I'm guessing that Sun Studio along with jds-cbe runs using the JRE/Java
> etc
> > which is why it doens't need the X11 framework?
>
> Actually Sun Studio doesn't use JRE-Java nor does the jds-cbe.
>
> Its really straightforward and doesn't utilize an integrated graphic
> environment
> the way some other developement products do with 'Studio' in the name :)
>
> > I am hoping that is the case
> > as I'm running Solaris inside VMs and having trouble with bringing up the
> > GUI desktop interface on OpenSolaris although it comes up withour a
> problem
> > on Solaris 10. I thought it was Solaris that had problems with drivers
> etc.
> > but it had no problems running OpenSolaris inside a VM on the same VM
> Host
> > fails to bring up the desktop.
> >
> > Either way, it's awesome if I don't need to have the X11 framework in
> place
> > to compile FS in Solaris.
> I'm sure you'll have very little problem with this, just install sun studio
> on your system
> with the instructions for your flavor of [Open]Solaris along with the
> jds-cbe and your good to go.
>
> -kim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris.chen2004 at gmail.com  Wed Dec  8 03:25:22 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Tue, 7 Dec 2010 19:25:22 -0500
Subject: [Freeswitch-users] Video Call
In-Reply-To: <AANLkTikrzHRkoapcGYrRPbfztop7YCqXsnKZgaAzb_5D@mail.gmail.com>
References: <AANLkTikrzHRkoapcGYrRPbfztop7YCqXsnKZgaAzb_5D@mail.gmail.com>
Message-ID: <AANLkTinNAiB-+Q-vbybAtrBeZXXhpgHpvJ4xBWAOP1eA@mail.gmail.com>

Hi Lloyd, I have my Nortel IP1535 hooked up to my FreeSWITCH and another
Nortel IP1535 hooked up to Asterisk, they can do video calls between them
pretty well. Please keep in mind though video calls are pass-through under
both FS and Asterisk, the end points must support the same video codecs.

Thanks,
Chris



On Tue, Dec 7, 2010 at 6:49 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> Does FreeSwitch support video call? Is there any special configurations for
> video calls?
>
> I am planning to test with Yealink IP Video Phone.
>
> http://www.yealink.com/en/view.asp?ClassLayer=120
>
> Any help appreciated.
>
> Thanks
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at sunteltech.ca  Wed Dec  8 03:36:05 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Tue, 7 Dec 2010 19:36:05 -0500
Subject: [Freeswitch-users] Video Call
In-Reply-To: <AANLkTinNAiB-+Q-vbybAtrBeZXXhpgHpvJ4xBWAOP1eA@mail.gmail.com>
References: <AANLkTikrzHRkoapcGYrRPbfztop7YCqXsnKZgaAzb_5D@mail.gmail.com>
	<AANLkTinNAiB-+Q-vbybAtrBeZXXhpgHpvJ4xBWAOP1eA@mail.gmail.com>
Message-ID: <AANLkTi=FN2RYvPcFGYa2fjzxdgdkUFLiv3+KFJfajmPP@mail.gmail.com>

Hi Chris,

Thank you for the information. I know how video works in Asterisk. Just
enable the videosupport=yes and enable the codec.

But could not find any resources how to enable the video call in Free
Switch. I will enable the codec in vars.xml and make some video calls.

Thanks
Lloyd

On Tue, Dec 7, 2010 at 7:25 PM, Chris Chen <chris.chen2004 at gmail.com> wrote:

> Hi Lloyd, I have my Nortel IP1535 hooked up to my FreeSWITCH and another
> Nortel IP1535 hooked up to Asterisk, they can do video calls between them
> pretty well. Please keep in mind though video calls are pass-through under
> both FS and Asterisk, the end points must support the same video codecs.
>
> Thanks,
> Chris
>
>
>
> On Tue, Dec 7, 2010 at 6:49 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Hi All,
>>
>> Does FreeSwitch support video call? Is there any special configurations
>> for video calls?
>>
>> I am planning to test with Yealink IP Video Phone.
>>
>> http://www.yealink.com/en/view.asp?ClassLayer=120
>>
>> Any help appreciated.
>>
>> Thanks
>> Lloyd
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From michael at ostag.org  Wed Dec  8 04:11:22 2010
From: michael at ostag.org (Michael Collins)
Date: Tue, 7 Dec 2010 17:11:22 -0800
Subject: [Freeswitch-users] OSTAG - How You Can Help
Message-ID: <AANLkTim+17x7vApY9Ag6-hsD6gPS1CoyWWGSr8f13zDr@mail.gmail.com>

Greetings!

The OSTAG team would like to let everyone know that we are ready for
non-profit work. While monetary donations are always welcomed they certainly
are not the only way that you can help. We have added a new section to the
ostag.org Web site on how to donate <http://www.ostag.org/node/7>. Please
review it and think about how you can donate time, money, or resources to
the OSTAG project. All donations are tax-deductible.

Thank you for continuing to support Open Source software!

-The OSTAG Team
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From haloha201 at yahoo.com  Wed Dec  8 05:44:49 2010
From: haloha201 at yahoo.com (ha do)
Date: Tue, 7 Dec 2010 18:44:49 -0800 (PST)
Subject: [Freeswitch-users] get error on skypopen module
In-Reply-To: <AANLkTinRG+qV-O_NJ2NnAn2f7xKo-3dm1m+Z+EjKLJVV@mail.gmail.com>
Message-ID: <861954.65794.qm@web32408.mail.mud.yahoo.com>

Hi Steven

yeah, there is some information of skypopen in event socket

Thank you
Ha`

--- On Tue, 12/7/10, Steven Ayre <steveayre at gmail.com> wrote:

> From: Steven Ayre <steveayre at gmail.com>
> Subject: Re: [Freeswitch-users] get error on skypopen module
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Tuesday, December 7, 2010, 12:42 PM
> > the skypopen module generate the
> events which are same as other modules
> 
> That's because they're channel events generated by the
> freeswitch
> core. But there may be variables in them that you can use
> to identify
> the Skypopen interface used.
> 
> > so i have to wait the 'sk list' command which maybe
> support to event socket in future :D
> 
> It's an API command so it's supported right now - just do
> "api sk
> list". You just won't get the response as XML.
> 
> -Steve
> 
> 
> 
> On 7 December 2010 15:00, ha do <haloha201 at yahoo.com>
> wrote:
> > Hi Steven
> >
> > the skypopen module generate the events which are same
> as other modules
> > so i have to wait the 'sk list' command which maybe
> support to event socket in future :D
> >
> >
> > Thank you
> > Ha`
> > --- On Tue, 12/7/10, Steven Ayre <steveayre at gmail.com>
> wrote:
> >
> >> From: Steven Ayre <steveayre at gmail.com>
> >> Subject: Re: [Freeswitch-users] get error on
> skypopen module
> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> >> Date: Tuesday, December 7, 2010, 6:48 AM
> >> > i have a question on event
> >> socket, what is the event name for skypopen to
> monitor the
> >> interface IDLE/not IDLE or ANSWER or IN
> PROGRESS...
> >>
> >> Full ESL event list: http://wiki.freeswitch.org/wiki/Event_list
> >>
> >> Skyopen doesn't generate events for state changes
> itself.
> >>
> >> Check for CHANNEL_ANSWER and
> >> CHANNEL_PROGRESS/CHANNEL_PROGRESS_MEDIA.
> >>
> >> For spotting an idle skype interface,
> CHANNEL_CREATE and
> >> CHANNEL_DESTROY might tell you that by telling you
> when the
> >> interface
> >> is in use.
> >>
> >> If you want to poll for the interface states you
> can use
> >> the sk list
> >> api function via ESL, although there's
> unfortunately no 'as
> >> xml'
> >> formatting at the moment so it'll be slightly
> trickier to
> >> parse.
> >>
> >> -Steve
> >>
> >> On 7 December 2010 12:15, ha do <haloha201 at yahoo.com>
> >> wrote:
> >> > Hi Meftab
> >> >
> >> >> did you autorised skypopen to access
> >> >> skype from the skype client?
> >> > you mean the skypopen_auth ??? if so, the
> answer is
> >> yes
> >> >
> >> >> did you configured the skypopen
> interfaces to each
> >> one
> >> >> of the skype
> >> > i dont understand, because i use multi skypy
> >> client(skype software) with 1 username and 1
> password
> >> > and use the sample config file in
> >>
> $source.../../configs/multiple-instances-same-skype-user/
> >> >
> >> > 2 skype clients run in difference folder
> >> > /home/cucku/multi/interfaces01
> >> > /home/cucku/multi/interfaces02
> >> >
> >> > so i only need to config the
> skypopen.conf.xml look
> >> like below:
> >> > <configuration name="skypopen.conf"
> >> description="Skypopen Configuration">
> >> > ?<global_settings>
> >> > ? ?<param name="debug" value="8"/>
> >> > ? ?<param name="dialplan"
> value="XML"/>
> >> > ? ?<param name="context"
> value="default"/>
> >> > ? ?<param name="destination"
> value="5000"/>
> >> > ? ?<param name="skype_user"
> >> value="do_nguyen_ha"/>
> >> > ? ?<param
> name="report_incoming_chatmessages"
> >> value="false"/>
> >> > ? ?<param name="silent_mode"
> value="false"/>
> >> > ? ?<param
> name="write_silence_when_idle"
> >> value="true"/>
> >> > ? ?<param name="setsockopt"
> value="false"/>
> >> > ?</global_settings>
> >> > ?<!-- one entry here per each skypopen
> interface
> >> -->
> >> > ?<per_interface_settings>
> >> > ? ?<interface id="1"
> name="interface1">
> >> > ? ? ? ?<param name="X11-display"
> >> value=":101"/>
> >> > ? ?</interface>
> >> > ? ?<interface id="2"
> name="interface2">
> >> > ? ? ? ?<param name="X11-display"
> >> value=":102"/>
> >> > ? ?</interface>
> >> > ?</per_interface_settings>
> >> > </configuration>
> >> >
> >> > does the conifg look ok?? if not, please
> guide me to
> >> make it right
> >> >
> >> > i have a question on event socket, what is
> the event
> >> name for skypopen to monitor the interface
> IDLE/not IDLE or
> >> ANSWER or IN PROGRESS...
> >> >
> >> > which event plain should i take care of
> >> >
> >> >
> >> > Thank you
> >> > Ha`
> >> > --- On Tue, 12/7/10, Meftah Tayeb <tayeb.meftah at gmail.com>
> >> wrote:
> >> >
> >> >> From: Meftah Tayeb <tayeb.meftah at gmail.com>
> >> >> Subject: Re: [Freeswitch-users] get error
> on
> >> skypopen module
> >> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> >> >> Cc: "ha do" <haloha201 at yahoo.com>
> >> >> Date: Tuesday, December 7, 2010, 4:43 AM
> >> >> did you autorised skypopen to access
> >> >> skype from the skype client?
> >> >> and did you configured the skypopen
> interfaces to
> >> each one
> >> >> of the skype
> >> >> clients?
> >> >> thanks
> >> >> Le 07/12/2010 05:34, ha do a ?crit :
> >> >> > Hi list
> >> >> >
> >> >> > i setup freeswitch and skypopen
> running fine
> >> >> >
> >> >> > there are 2 skype clients run on
> freeswitch
> >> >> > freeswitch at internal>? sk list
> >> >> > sk console is NOT yet assigned
> >> >> > F ID? ? ? ? Name
> >> >> ? ? IB (F/T)? ? OB (F/T)
> >> >> State???CallFlw
> >> >> ???UUID
> >> >> > = ====? ? ========
> >> >> =======? ???=======
> >> >> ???======? ============
> >> >> ======
> >> >> >? ? 1
> >> >> ???[interface1]? ? ? 0/1
> >> >> ? ? ? 3/7
> >> >> IDLE? ? IDLE
> >> >> >? ? 2
> >> >> ???[interface2]? ? ? 0/5
> >> >> ? ? ? 1/5
> >> >> IDLE? ? IDLE
> >> >> >
> >> >> > the skype clients are used the same
> username
> >> +
> >> >> password of skype account
> >> >> >
> >> >> >
> >> >> > the skypopen works fine but i get
> error below
> >> in the
> >> >> debug mode
> >> >> >
> >> >> > 2010-12-07 04:24:35.794611 [DEBUG]
> >> >> switch_core_state_machine.c:462
> >> (skypopen/interface1) State
> >> >> DESTROY going to sleep
> >> >> > 2010-12-07 04:24:35.795807 [DEBUG]
> >> >> skypopen_protocol.c:173? ???[|]
> >> >> [DEBUG_SKYPE? 173? ][interface1
> >> >> ???][IDLE,IDLE] READING: |||ERROR 559
> CALL:
> >> >> Action failed|||
> >> >> > 2010-12-07 04:24:35.795807 [DEBUG]
> >> >> skypopen_protocol.c:228? ???[|]
> >> >> [DEBUG_SKYPE? 228? ][interface1
> >> >> ???][IDLE,IDLE] Skype got ERROR about
> a
> >> >> failed action (probably TRYING to HANGUP
> A CALL),
> >> no
> >> >> problem: |||ERROR 559 CALL: Action
> failed|||
> >> >> > 2010-12-07 04:24:35.795807 [DEBUG]
> >> >> skypopen_protocol.c:173? ???[|]
> >> >> [DEBUG_SKYPE? 173? ][interface1
> >> >> ???][IDLE,IDLE] READING: |||ERROR 559
> CALL:
> >> >> Action failed|||
> >> >> > 2010-12-07 04:24:35.795807 [DEBUG]
> >> >> skypopen_protocol.c:228? ???[|]
> >> >> [DEBUG_SKYPE? 228? ][interface1
> >> >> ???][IDLE,IDLE] Skype got ERROR about
> a
> >> >> failed action (probably TRYING to HANGUP
> A CALL),
> >> no
> >> >> problem: |||ERROR 559 CALL: Action
> failed|||
> >> >> >
> >> >> > How to fix it
> >> >> >
> >> >> > Thank you
> >> >> > Ha`
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >>
> >> >>
> >> >> --
> >> >> Meftah Tayeb
> >> >> inum: +883510001288000
> >> >> Phone: +13602276297
> >> >> Fax: +12538020313
> >> >>
> >> >>
> >> >
> >> >
> >> >
> >> >
> >> >
> _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From haloha201 at yahoo.com  Wed Dec  8 05:46:24 2010
From: haloha201 at yahoo.com (ha do)
Date: Tue, 7 Dec 2010 18:46:24 -0800 (PST)
Subject: [Freeswitch-users] get error on skypopen module
In-Reply-To: <AANLkTimB+Nkc=S5gbOqrGFBnZo7rXKwVXs7uTc3FrteR@mail.gmail.com>
Message-ID: <358063.99637.qm@web32401.mail.mud.yahoo.com>

Hi Jian

so my freeswitch performance is normal state when run with skypopen :D

Thank you
Ha

--- On Tue, 12/7/10, Jian Ren <renjian at gmail.com> wrote:

From: Jian Ren <renjian at gmail.com>
Subject: Re: [Freeswitch-users] get error on skypopen module
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Tuesday, December 7, 2010, 2:14 PM

I am getting 5-6% for each skype call and 1% idle instance, on my Q6600. It's normal, 20 concurrent calls would be way enough for most people. If you want more, can scale to more servers.

Jian


On Tue, Dec 7, 2010 at 9:52 AM, ha do <haloha201 at yahoo.com> wrote:

Hi Steven



yes, i use the snd-dummy as guide in startskype.sh script





thank you for your guide, it works now :D

Ha`



--- On Tue, 12/7/10, Steven Ayre <steveayre at gmail.com> wrote:



> From: Steven Ayre <steveayre at gmail.com>

> Subject: Re: [Freeswitch-users] get error on skypopen module

> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

> Date: Tuesday, December 7, 2010, 6:40 AM

> > how to turn off the debug??

> remove the below line in skypopen.xml.conf

> >? ? <param name="debug" value="8"/>

>

> Yes, remove that line, then do reloadxml and reload

> mod_skypopen to

> pick up the new config. Or restart FS.

>

> > i do check the CPU performance so 1 call from/to skype

> client it will cost 5% CPU of server

> > if 2 calls, CPU will be 10%

> >

> > i dont know that is normal when freeswitch run with

> skypopen module

>

> Are you using the dummy sound driver? That's meant to

> reduce the CPU

> usage, but I don't know how much. % will also depend on

> what CPU

> you're using.

>

> > how do i change the

> > Would you like to receive list mail batched in a daily

> digest?

> > from yes to no? in http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

>

> 1) Visit that link

> 2) Enter your email and click 'Unsubscribe or edit options'

> near the bottom

> 3) Enter your password (there's a reminder button that'll

> email it to

> you if you don't know it)

> 4) Scroll down to the 'Your FreeSWITCH-users Subscription

> Options' section

> 5) Flip the 'Set Digest Mode' setting to Off and click

> 'Submit my changes'

>

> -Steve

>

>

>

>

>

>

> On 7 December 2010 12:29, ha do <haloha201 at yahoo.com>

> wrote:

> > Hi Giovanny

> >

> > how to turn off the debug?? remove the below line in

> skypopen.xml.conf

> > ? ?<param name="debug" value="8"/>

> >

> > the error i get when i run the

> /usr/local/freeswitch/bin/fs_cli

> >

> > i do check the CPU performance so 1 call from/to skype

> client it will cost 5% CPU of server

> > if 2 calls, CPU will be 10%

> >

> > i dont know that is normal when freeswitch run with

> skypopen module

> >

> >

> > how do i change the

> > Would you like to receive list mail batched in a daily

> digest?

> > from yes to no ?in http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

> >

> >

> > Thank you

> > Ha

> >

> >

> > On Tue, Dec 7, 2010 at 5:34 AM, ha do <haloha201 at yahoo.com>

> wrote:

> >> the skypopen works fine but i get error below in

> the debug mode

> >

> > don't run in debug mode if you are not debugging :)

> >

> >> [interface1 ? ? ][IDLE,IDLE] Skype got ERROR

> about a failed action (probably TRYING to HANGUP A CALL), no

> problem: |||ERROR 559 CALL: Action failed|||

> >

> > as is clearly written, this is not a problem. Is a

> debug message that

> > explains we tried to hangup a skype call that the

> skype client closed

> > before. Is just to be double sure that we try to close

> it anyway. The

> > debug message is just an explanation.

> >

> >>

> >> How to fix it

> >

> > nothing to fix, no problems here.

> >

> > -giovanni

> >

> > --

> > Sincerely,

> >

> > Giovanni Maruzzelli

> > Cell : +39-347-2665618

> >

> >

> >

> >

> > _______________________________________________

> > FreeSWITCH-users mailing list

> > FreeSWITCH-users at lists.freeswitch.org

> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

> > http://www.freeswitch.org

> >

>

> _______________________________________________

> FreeSWITCH-users mailing list

> FreeSWITCH-users at lists.freeswitch.org

> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

> http://www.freeswitch.org

>









_______________________________________________

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FreeSWITCH-users at lists.freeswitch.org

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

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From brian at freeswitch.org  Wed Dec  8 06:08:18 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 7 Dec 2010 21:08:18 -0600
Subject: [Freeswitch-users] Advertising and thread hijacking
Message-ID: <9852CCA5-6DF1-447C-B58B-AAEDA02F4745@freeswitch.org>

NOTICE:  Anyone posting commercial content on the dev or users list that isn't directly related to the project, cluecon or ostag will be promptly dealt with.  I do not wish these lists to become a mess.   And please do not hijack threads.  If you click reply then change the subject that is what causes the thread to be hijacked.  Click NEW then type in the list address.  

Thank you,
/b



From brian at freeswitch.org  Wed Dec  8 06:10:05 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 7 Dec 2010 21:10:05 -0600
Subject: [Freeswitch-users] I have good quality Bangladesh Mobile White
	route (8801)
In-Reply-To: <6b65470d0909181708t1e3cad2cm4e1b872d9b98f9c5@mail.gmail.com>
References: <4ab3d091.9453f10a.0891.2fa4@mx.google.com>
	<A810A7B7-F13F-4661-A95A-2DA832BBA99C@me.com>
	<BAY112-W12CAE675336A9A939AC654BFE00@phx.gbl>
	<6b65470d0909181708t1e3cad2cm4e1b872d9b98f9c5@mail.gmail.com>
Message-ID: <E64FBD56-393A-46D0-8B58-E1E87E72963B@freeswitch.org>

This is the guys company.  http://www.omega-tec.com/

I have banned his email address from our list and I hope that our moderators stop approving these types of posts.

I hate slime ball spammers.

/b

On Sep 18, 2009, at 7:08 PM, William Suffill wrote:

> Probably best to post these type of offers on the freeswitch-biz list
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz (Hopefully
> that list will eventually
> grow with business posts. Still pretty slow at present)
> 
> 
> Also best to create a new e-mail versus reply to another thread and
> changing the subject.
> 
> -- W

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From infos at madovsky.org  Wed Dec  8 06:53:04 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 7 Dec 2010 22:53:04 -0500
Subject: [Freeswitch-users] api_hangup_hook question
References: <4BEF6419594F4281874094369DC3289A@e1705>
	<AANLkTinNnpAyEAfsx56bciyjwUSfqAN=arJxbvhV-mAm@mail.gmail.com>
Message-ID: <2628065EC8D348DE8F47FE7C8B011F4B@e1705>

mmmhmm, weird in the logs I can see :

/usr/local/freeswitch/scripts/perl/faxStatus.pl(): ///////////// what is these parenthesis at the last?
INVALID COMMAND!

if I use this xml
<action application="set" data="api_hangup_hook=/usr/local/freeswitch/scripts/perl/faxStatus.pl"/>

and

system(usr/local/freeswitch/scripts/perl/faxStatus.pl()): 
INVALID COMMAND!

if

<action application="set" data="api_hangup_hook=system(/usr/local/freeswitch/scripts/perl/faxStatus.pl)"/>

if you think this log is not enough I will pastebin, but the rest of logs are ok....




  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, December 07, 2010 3:51 PM
  Subject: Re: [Freeswitch-users] api_hangup_hook question


  Can you pastebin your logs? There might be a clue as to what is happening.
  -MC


  On Tue, Dec 7, 2010 at 10:27 AM, Madovsky <infos at madovsky.org> wrote:

    I have this in dialplan/default

            <extension name="9999999999">
                    <condition field="destination_number" expression="^(9999999999)$">
                            <action application="set" data="domain_name=$${domain}"/>
                            <action application="export" data="dialed_extension=$1"/>
                            <action application="set" data="hangup_after_bridge=true"/>
                            <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,CALL_REJECTED,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
                            <action application="ring_ready"/>
                            <action application="tone_detect" data="fax 1100 r +5000 transfer 'receivefax XML features' 1" />
                            <action application="export" data="originate_timeout=15"/>
                            <action application="bridge" data="{absolute_codec_string='speex at 16000k,G729,GSM,G726,PCMU,PCMA',sip_from_uri=${sip_from_user}@${sip_from_host},origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number}}user/${dialed_extension}"/>
                            <action application="sleep" data="3000"/>
                            <action application="answer"/>
                            <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
                    </condition>
            </extension>


    and this in dialplan/features


        <extension name="receivefax">
          <condition field="destination_number" expression="^receivefax$">
            <action application="set" data="session_in_hangup_hook=true"/>
            <action application="set" data="fax_file=/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
            <action application="set" data="fax_mode=recv"/>
            <action application="set" data="api_hangup_hook=perl /usr/local/freeswitch/scripts/perl/faxStatus.pl"/>
            <action application="answer" />
            <action application="playback" data="silence_stream://2000"/>
            <action application="set" data="fax_enable_t38_request=true"/>
            <action application="set" data="fax_enable_t38=true"/>
            <action application="rxfax" data="/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
            <action application="hangup"/>
          </condition>
        </extension>


    the fax is received well, but the perl script is not executed.

    Am I missing anything ?

    Thanks

    Franck



    _______________________________________________
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    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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------------------------------------------------------------------------------


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Wed Dec  8 06:58:31 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 7 Dec 2010 22:58:31 -0500
Subject: [Freeswitch-users] api_hangup_hook question
Message-ID: <16DC3FEA00F8490FBB513677700428D3@e1705>

Sorry correction for the last code it shows

2010-12-07 22:57:25.262675 [DEBUG] switch_core_state_machine.c:492 Hangup Command with Session system(/usr/local/freeswitch/scripts/perl/faxStatus.pl)():
INVALID COMMAND!

for 

<action application="set" data="api_hangup_hook=system(/usr/local/freeswitch/scripts/perl/faxStatus.pl)"/>
  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, December 07, 2010 10:53 PM
  Subject: Re: [Freeswitch-users] api_hangup_hook question


  mmmhmm, weird in the logs I can see :

  /usr/local/freeswitch/scripts/perl/faxStatus.pl(): ///////////// what is these parenthesis at the last?
  INVALID COMMAND!

  if I use this xml
  <action application="set" data="api_hangup_hook=/usr/local/freeswitch/scripts/perl/faxStatus.pl"/>

  and

  system(usr/local/freeswitch/scripts/perl/faxStatus.pl()): 
  INVALID COMMAND!

  if

  <action application="set" data="api_hangup_hook=system(/usr/local/freeswitch/scripts/perl/faxStatus.pl)"/>

  if you think this log is not enough I will pastebin, but the rest of logs are ok....

   


    ----- Original Message ----- 
    From: Michael Collins 
    To: FreeSWITCH Users Help 
    Sent: Tuesday, December 07, 2010 3:51 PM
    Subject: Re: [Freeswitch-users] api_hangup_hook question


    Can you pastebin your logs? There might be a clue as to what is happening. 
    -MC


    On Tue, Dec 7, 2010 at 10:27 AM, Madovsky <infos at madovsky.org> wrote:

      I have this in dialplan/default

              <extension name="9999999999">
                      <condition field="destination_number" expression="^(9999999999)$">
                              <action application="set" data="domain_name=$${domain}"/>
                              <action application="export" data="dialed_extension=$1"/>
                              <action application="set" data="hangup_after_bridge=true"/>
                              <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,CALL_REJECTED,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
                              <action application="ring_ready"/>
                              <action application="tone_detect" data="fax 1100 r +5000 transfer 'receivefax XML features' 1" />
                              <action application="export" data="originate_timeout=15"/>
                              <action application="bridge" data="{absolute_codec_string='speex at 16000k,G729,GSM,G726,PCMU,PCMA',sip_from_uri=${sip_from_user}@${sip_from_host},origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number}}user/${dialed_extension}"/>
                              <action application="sleep" data="3000"/>
                              <action application="answer"/>
                              <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
                      </condition>
              </extension>


      and this in dialplan/features


          <extension name="receivefax">
            <condition field="destination_number" expression="^receivefax$">
              <action application="set" data="session_in_hangup_hook=true"/>
              <action application="set" data="fax_file=/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
              <action application="set" data="fax_mode=recv"/>
              <action application="set" data="api_hangup_hook=perl /usr/local/freeswitch/scripts/perl/faxStatus.pl"/>
              <action application="answer" />
              <action application="playback" data="silence_stream://2000"/>
              <action application="set" data="fax_enable_t38_request=true"/>
              <action application="set" data="fax_enable_t38=true"/>
              <action application="rxfax" data="/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
              <action application="hangup"/>
            </condition>
          </extension>


      the fax is received well, but the perl script is not executed.

      Am I missing anything ?

      Thanks

      Franck



      _______________________________________________
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      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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----------------------------------------------------------------------------


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From infos at madovsky.org  Wed Dec  8 07:04:11 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 7 Dec 2010 23:04:11 -0500
Subject: [Freeswitch-users] api_hangup_hook question
Message-ID: <A9D1B2DC5F764116927E5584C9383BF0@e1705>

arrrgh,

why I put parenthesis, this is the right syntax :

<action application="set" data="api_hangup_hook=system /usr/local/freeswitch/scripts/perl/faxStatus.pl"/>

thanks

Franck
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, December 07, 2010 10:58 PM
  Subject: Re: [Freeswitch-users] api_hangup_hook question


  Sorry correction for the last code it shows

  2010-12-07 22:57:25.262675 [DEBUG] switch_core_state_machine.c:492 Hangup Command with Session system(/usr/local/freeswitch/scripts/perl/faxStatus.pl)():
  INVALID COMMAND!

  for 

  <action application="set" data="api_hangup_hook=system(/usr/local/freeswitch/scripts/perl/faxStatus.pl)"/>
    ----- Original Message ----- 
    From: Madovsky 
    To: FreeSWITCH Users Help 
    Sent: Tuesday, December 07, 2010 10:53 PM
    Subject: Re: [Freeswitch-users] api_hangup_hook question


    mmmhmm, weird in the logs I can see :

    /usr/local/freeswitch/scripts/perl/faxStatus.pl(): ///////////// what is these parenthesis at the last?
    INVALID COMMAND!

    if I use this xml
    <action application="set" data="api_hangup_hook=/usr/local/freeswitch/scripts/perl/faxStatus.pl"/>

    and

    system(usr/local/freeswitch/scripts/perl/faxStatus.pl()): 
    INVALID COMMAND!

    if

    <action application="set" data="api_hangup_hook=system(/usr/local/freeswitch/scripts/perl/faxStatus.pl)"/>

    if you think this log is not enough I will pastebin, but the rest of logs are ok....

     


      ----- Original Message ----- 
      From: Michael Collins 
      To: FreeSWITCH Users Help 
      Sent: Tuesday, December 07, 2010 3:51 PM
      Subject: Re: [Freeswitch-users] api_hangup_hook question


      Can you pastebin your logs? There might be a clue as to what is happening. 
      -MC


      On Tue, Dec 7, 2010 at 10:27 AM, Madovsky <infos at madovsky.org> wrote:

        I have this in dialplan/default

                <extension name="9999999999">
                        <condition field="destination_number" expression="^(9999999999)$">
                                <action application="set" data="domain_name=$${domain}"/>
                                <action application="export" data="dialed_extension=$1"/>
                                <action application="set" data="hangup_after_bridge=true"/>
                                <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,CALL_REJECTED,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
                                <action application="ring_ready"/>
                                <action application="tone_detect" data="fax 1100 r +5000 transfer 'receivefax XML features' 1" />
                                <action application="export" data="originate_timeout=15"/>
                                <action application="bridge" data="{absolute_codec_string='speex at 16000k,G729,GSM,G726,PCMU,PCMA',sip_from_uri=${sip_from_user}@${sip_from_host},origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number}}user/${dialed_extension}"/>
                                <action application="sleep" data="3000"/>
                                <action application="answer"/>
                                <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
                        </condition>
                </extension>


        and this in dialplan/features


            <extension name="receivefax">
              <condition field="destination_number" expression="^receivefax$">
                <action application="set" data="session_in_hangup_hook=true"/>
                <action application="set" data="fax_file=/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
                <action application="set" data="fax_mode=recv"/>
                <action application="set" data="api_hangup_hook=perl /usr/local/freeswitch/scripts/perl/faxStatus.pl"/>
                <action application="answer" />
                <action application="playback" data="silence_stream://2000"/>
                <action application="set" data="fax_enable_t38_request=true"/>
                <action application="set" data="fax_enable_t38=true"/>
                <action application="rxfax" data="/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
                <action application="hangup"/>
              </condition>
            </extension>


        the fax is received well, but the perl script is not executed.

        Am I missing anything ?

        Thanks

        Franck



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






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


      _______________________________________________
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      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From renjian at gmail.com  Wed Dec  8 07:45:53 2010
From: renjian at gmail.com (Jian Ren)
Date: Tue, 7 Dec 2010 23:45:53 -0500
Subject: [Freeswitch-users] get error on skypopen module
In-Reply-To: <358063.99637.qm@web32401.mail.mud.yahoo.com>
References: <AANLkTimB+Nkc=S5gbOqrGFBnZo7rXKwVXs7uTc3FrteR@mail.gmail.com>
	<358063.99637.qm@web32401.mail.mud.yahoo.com>
Message-ID: <AANLkTim5qghatZP8CbYmFfV004PTNfwWx=P8u3Vvgk0A@mail.gmail.com>

Yes, I got the same. Unless Giovanni doesn't agree :-).
Jian

On Tue, Dec 7, 2010 at 9:46 PM, ha do <haloha201 at yahoo.com> wrote:

> Hi Jian
>
> so my freeswitch performance is normal state when run with skypopen :D
>
> Thank you
> Ha
>
> --- On *Tue, 12/7/10, Jian Ren <renjian at gmail.com>* wrote:
>
>
> From: Jian Ren <renjian at gmail.com>
>
> Subject: Re: [Freeswitch-users] get error on skypopen module
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Tuesday, December 7, 2010, 2:14 PM
>
>
> I am getting 5-6% for each skype call and 1% idle instance, on my Q6600.
> It's normal, 20 concurrent calls would be way enough for most people. If you
> want more, can scale to more servers.
>
> Jian
>
> On Tue, Dec 7, 2010 at 9:52 AM, ha do <haloha201 at yahoo.com<http://mc/compose?to=haloha201 at yahoo.com>
> > wrote:
>
> Hi Steven
>
> yes, i use the snd-dummy as guide in startskype.sh script
>
>
> thank you for your guide, it works now :D
> Ha`
>
> --- On Tue, 12/7/10, Steven Ayre <steveayre at gmail.com<http://mc/compose?to=steveayre at gmail.com>>
> wrote:
>
> > From: Steven Ayre <steveayre at gmail.com<http://mc/compose?to=steveayre at gmail.com>
> >
> > Subject: Re: [Freeswitch-users] get error on skypopen module
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users at lists.freeswitch.org>
> >
> > Date: Tuesday, December 7, 2010, 6:40 AM
> > > how to turn off the debug??
> > remove the below line in skypopen.xml.conf
> > >    <param name="debug" value="8"/>
> >
> > Yes, remove that line, then do reloadxml and reload
> > mod_skypopen to
> > pick up the new config. Or restart FS.
> >
> > > i do check the CPU performance so 1 call from/to skype
> > client it will cost 5% CPU of server
> > > if 2 calls, CPU will be 10%
> > >
> > > i dont know that is normal when freeswitch run with
> > skypopen module
> >
> > Are you using the dummy sound driver? That's meant to
> > reduce the CPU
> > usage, but I don't know how much. % will also depend on
> > what CPU
> > you're using.
> >
> > > how do i change the
> > > Would you like to receive list mail batched in a daily
> > digest?
> > > from yes to no  in
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > 1) Visit that link
> > 2) Enter your email and click 'Unsubscribe or edit options'
> > near the bottom
> > 3) Enter your password (there's a reminder button that'll
> > email it to
> > you if you don't know it)
> > 4) Scroll down to the 'Your FreeSWITCH-users Subscription
> > Options' section
> > 5) Flip the 'Set Digest Mode' setting to Off and click
> > 'Submit my changes'
> >
> > -Steve
> >
> >
> >
> >
> >
> >
> > On 7 December 2010 12:29, ha do <haloha201 at yahoo.com<http://mc/compose?to=haloha201 at yahoo.com>
> >
> > wrote:
> > > Hi Giovanny
> > >
> > > how to turn off the debug?? remove the below line in
> > skypopen.xml.conf
> > >    <param name="debug" value="8"/>
> > >
> > > the error i get when i run the
> > /usr/local/freeswitch/bin/fs_cli
> > >
> > > i do check the CPU performance so 1 call from/to skype
> > client it will cost 5% CPU of server
> > > if 2 calls, CPU will be 10%
> > >
> > > i dont know that is normal when freeswitch run with
> > skypopen module
> > >
> > >
> > > how do i change the
> > > Would you like to receive list mail batched in a daily
> > digest?
> > > from yes to no  in
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > >
> > > Thank you
> > > Ha
> > >
> > >
> > > On Tue, Dec 7, 2010 at 5:34 AM, ha do <haloha201 at yahoo.com<http://mc/compose?to=haloha201 at yahoo.com>
> >
> > wrote:
> > >> the skypopen works fine but i get error below in
> > the debug mode
> > >
> > > don't run in debug mode if you are not debugging :)
> > >
> > >> [interface1     ][IDLE,IDLE] Skype got ERROR
> > about a failed action (probably TRYING to HANGUP A CALL), no
> > problem: |||ERROR 559 CALL: Action failed|||
> > >
> > > as is clearly written, this is not a problem. Is a
> > debug message that
> > > explains we tried to hangup a skype call that the
> > skype client closed
> > > before. Is just to be double sure that we try to close
> > it anyway. The
> > > debug message is just an explanation.
> > >
> > >>
> > >> How to fix it
> > >
> > > nothing to fix, no problems here.
> > >
> > > -giovanni
> > >
> > > --
> > > Sincerely,
> > >
> > > Giovanni Maruzzelli
> > > Cell : +39-347-2665618
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -----Inline Attachment Follows-----
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Wed Dec  8 09:01:28 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 8 Dec 2010 01:01:28 -0500
Subject: [Freeswitch-users] api_hangup_hook question
Message-ID: <01DB1034C44B4E499B4C1A6402BA240B@e1705>

Ok now it seems that the perl script is called (even php script called inside perl script works)
but $session is empty, despite of session_in_hangup_hook=true

my Perl script is very simple :

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

#!/usr/local/bin/perl
use strict;
our $session;
my %VARS;

        sub fprint($) #Dump string to the console
        {
                my ($msg) = @_;
                freeswitch::consoleLog("CRIT",$msg . "\n");
        }


        sub GETV #takes one or more variables names to import in
        {
                my @arr = @_;
                foreach my $var (@arr)
                {
                        $VARS{$var} = $session->getVariable($var);
                }
                &QWPHP();
        }

        sub QWPHP
        {
                my $shell_result;
                if($VARS{fax_mode} eq "recv")
                {
                        $shell_result = `/usr/local/bin/php /usr/local/freeswitch/scripts/php/faxCallBack.php $VARS{account_email} $VARS{effective_caller_id_name} $VARS{caller_id_number} $VARS{fax_file} $VARS{fax_status} $VARS{fax_result_text}`;
                }
                elsif($VARS{fax_mode} eq "send")
                {
                        $shell_result = `/usr/local/bin/php /usr/local/freeswitch/scripts/php/faxCallBack.php $VARS{php_session} $VARS{fax_status} $VARS{fax_result_text}`;
                }
        }

GETV(qw/fax_mode destination_number effective_caller_id_name effective_caller_id_number caller_id_number fax_file fax_status fax_result_text/);

1;

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


Stucked.... help



  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, December 07, 2010 11:04 PM
  Subject: Re: [Freeswitch-users] api_hangup_hook question


  arrrgh,

  why I put parenthesis, this is the right syntax :

  <action application="set" data="api_hangup_hook=system /usr/local/freeswitch/scripts/perl/faxStatus.pl"/>

  thanks

  Franck
    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Tuesday, December 07, 2010 10:58 PM
    Subject: Re: [Freeswitch-users] api_hangup_hook question


    Sorry correction for the last code it shows

    2010-12-07 22:57:25.262675 [DEBUG] switch_core_state_machine.c:492 Hangup Command with Session system(/usr/local/freeswitch/scripts/perl/faxStatus.pl)():
    INVALID COMMAND!

    for 

    <action application="set" data="api_hangup_hook=system(/usr/local/freeswitch/scripts/perl/faxStatus.pl)"/>
      ----- Original Message ----- 
      From: Madovsky 
      To: FreeSWITCH Users Help 
      Sent: Tuesday, December 07, 2010 10:53 PM
      Subject: Re: [Freeswitch-users] api_hangup_hook question


      mmmhmm, weird in the logs I can see :

      /usr/local/freeswitch/scripts/perl/faxStatus.pl(): ///////////// what is these parenthesis at the last?
      INVALID COMMAND!

      if I use this xml
      <action application="set" data="api_hangup_hook=/usr/local/freeswitch/scripts/perl/faxStatus.pl"/>

      and

      system(usr/local/freeswitch/scripts/perl/faxStatus.pl()): 
      INVALID COMMAND!

      if

      <action application="set" data="api_hangup_hook=system(/usr/local/freeswitch/scripts/perl/faxStatus.pl)"/>

      if you think this log is not enough I will pastebin, but the rest of logs are ok....

       


        ----- Original Message ----- 
        From: Michael Collins 
        To: FreeSWITCH Users Help 
        Sent: Tuesday, December 07, 2010 3:51 PM
        Subject: Re: [Freeswitch-users] api_hangup_hook question


        Can you pastebin your logs? There might be a clue as to what is happening. 
        -MC


        On Tue, Dec 7, 2010 at 10:27 AM, Madovsky <infos at madovsky.org> wrote:

          I have this in dialplan/default

                  <extension name="9999999999">
                          <condition field="destination_number" expression="^(9999999999)$">
                                  <action application="set" data="domain_name=$${domain}"/>
                                  <action application="export" data="dialed_extension=$1"/>
                                  <action application="set" data="hangup_after_bridge=true"/>
                                  <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,CALL_REJECTED,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
                                  <action application="ring_ready"/>
                                  <action application="tone_detect" data="fax 1100 r +5000 transfer 'receivefax XML features' 1" />
                                  <action application="export" data="originate_timeout=15"/>
                                  <action application="bridge" data="{absolute_codec_string='speex at 16000k,G729,GSM,G726,PCMU,PCMA',sip_from_uri=${sip_from_user}@${sip_from_host},origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number}}user/${dialed_extension}"/>
                                  <action application="sleep" data="3000"/>
                                  <action application="answer"/>
                                  <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
                          </condition>
                  </extension>


          and this in dialplan/features


              <extension name="receivefax">
                <condition field="destination_number" expression="^receivefax$">
                  <action application="set" data="session_in_hangup_hook=true"/>
                  <action application="set" data="fax_file=/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
                  <action application="set" data="fax_mode=recv"/>
                  <action application="set" data="api_hangup_hook=perl /usr/local/freeswitch/scripts/perl/faxStatus.pl"/>
                  <action application="answer" />
                  <action application="playback" data="silence_stream://2000"/>
                  <action application="set" data="fax_enable_t38_request=true"/>
                  <action application="set" data="fax_enable_t38=true"/>
                  <action application="rxfax" data="/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
                  <action application="hangup"/>
                </condition>
              </extension>


          the fax is received well, but the perl script is not executed.

          Am I missing anything ?

          Thanks

          Franck



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






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


        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From ovvenkatesan at gmail.com  Wed Dec  8 10:11:25 2010
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Wed, 8 Dec 2010 12:41:25 +0530
Subject: [Freeswitch-users] TTS volume issue
Message-ID: <AANLkTi=RMr5L80amk4+MutNZ37io1RcHrfCecK=JC7Ku@mail.gmail.com>

Hi to All,

I need to increase the volume of TTS.
I mean, What ever I am getting output from the TTS,
Before playing to user, I need to increase the volume.

Is it possible??



-- 

Regards
Venkatesan OV.
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From tayeb.meftah at gmail.com  Wed Dec  8 11:11:34 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Wed, 08 Dec 2010 09:11:34 +0100
Subject: [Freeswitch-users] api_hangup_hook question
In-Reply-To: <2628065EC8D348DE8F47FE7C8B011F4B@e1705>
References: <4BEF6419594F4281874094369DC3289A@e1705>	<AANLkTinNnpAyEAfsx56bciyjwUSfqAN=arJxbvhV-mAm@mail.gmail.com>
	<2628065EC8D348DE8F47FE7C8B011F4B@e1705>
Message-ID: <4CFF3DB6.7090303@gmail.com>

lol,
do you have mod_perl compiled/loaded?
:P
Le 08/12/2010 04:53, Madovsky a ?crit :
> mmmhmm, weird in the logs I can see :
> /usr/local/freeswitch/scripts/perl/faxStatus.pl(): ///////////// what 
> is these parenthesis at the last?
> INVALID COMMAND!
> if I use this xml
> <action application="set" 
> data="api_hangup_hook=/usr/local/freeswitch/scripts/perl/faxStatus.pl"/>
> and
> system(usr/local/freeswitch/scripts/perl/faxStatus.pl()):
> INVALID COMMAND!
> if
> <action application="set" 
> data="api_hangup_hook=system(/usr/local/freeswitch/scripts/perl/faxStatus.pl)"/>
> if you think this log is not enough I will pastebin, but the rest of 
> logs are ok....
>
>
>     ----- Original Message -----
>     *From:* Michael Collins <mailto:msc at freeswitch.org>
>     *To:* FreeSWITCH Users Help
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     *Sent:* Tuesday, December 07, 2010 3:51 PM
>     *Subject:* Re: [Freeswitch-users] api_hangup_hook question
>
>     Can you pastebin your logs? There might be a clue as to what is
>     happening.
>     -MC
>
>     On Tue, Dec 7, 2010 at 10:27 AM, Madovsky <infos at madovsky.org
>     <mailto:infos at madovsky.org>> wrote:
>
>         I have this in dialplan/default
>         <extension name="9999999999">
>         <condition field="destination_number" expression="^(9999999999)$">
>         <action application="set" data="domain_name=$${domain}"/>
>         <action application="export" data="dialed_extension=$1"/>
>         <action application="set" data="hangup_after_bridge=true"/>
>         <action application="set"
>         data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,CALL_REJECTED,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
>         <action application="ring_ready"/>
>         <action application="tone_detect" data="fax 1100 r +5000
>         transfer 'receivefax XML features' 1" />
>         <action application="export" data="originate_timeout=15"/>
>         <action application="bridge"
>         data="{absolute_codec_string='speex at 16000k,G729,GSM,G726,PCMU,PCMA',sip_from_uri=${sip_from_user}@${sip_from_host},origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number}}user/${dialed_extension}"/
>         <mailto:%7Babsolute_codec_string=%27speex at 16000k,G729,GSM,G726,PCMU,PCMA%27,sip_from_uri=$%7Bsip_from_user%7D@$%7Bsip_from_host%7D,origination_caller_id_name=$%7Bcaller_id_name%7D,origination_caller_id_number=$%7Bcaller_id_number%7D%7Duser/$%7Bdialed_extension%7D%22/>>
>         <action application="sleep" data="3000"/>
>         <action application="answer"/>
>         <action application="voicemail" data="default ${domain_name}
>         ${dialed_extension}"/>
>         </condition>
>         </extension>
>         and this in dialplan/features
>         <extension name="receivefax">
>         <condition field="destination_number" expression="^receivefax$">
>         <action application="set" data="session_in_hangup_hook=true"/>
>         <action application="set"
>         data="fax_file=/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
>         <action application="set" data="fax_mode=recv"/>
>         <action application="set" data="api_hangup_hook=perl
>         /usr/local/freeswitch/scripts/perl/faxStatus.pl"/>
>         <action application="answer" />
>         <action application="playback" data="silence_stream://2000"/>
>         <action application="set" data="fax_enable_t38_request=true"/>
>         <action application="set" data="fax_enable_t38=true"/>
>         <action application="rxfax"
>         data="/usr/local/freeswitch/storage/fax/${caller_id_number}-${uuid}.rxfax.tiff"/>
>         <action application="hangup"/>
>         </condition>
>         </extension>
>         the fax is received well, but the perl script is not executed.
>         Am I missing anything ?
>         Thanks
>         Franck
>
>
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>     ------------------------------------------------------------------------
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313

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From tayeb.meftah at gmail.com  Wed Dec  8 11:16:07 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Wed, 08 Dec 2010 09:16:07 +0100
Subject: [Freeswitch-users] SIP MESSAGE requests
In-Reply-To: <004201cb9662$d770dad0$86529070$@comtex.net>
References: <004201cb9662$d770dad0$86529070$@comtex.net>
Message-ID: <4CFF3EC7.5030605@gmail.com>

freeswitch isn't a sip proxy
freeswitch is a media switching project growing up from a softphone up 
to a carrier grade softswitch/SBC but not a sip proxy
check:
Kamailio/Sip-Router Projects <http://www.kamailio.org>
OpenSips project <http://www.opensips.org>
don't by confused, this projects is both derived from openser that's 
becaume kamailio
thanks
Le 08/12/2010 00:02, John Rose a ?crit :
>
> Hello,
>
> Does anyone know if it is possible to construct, originate and receive 
> SIP MESSAGE requests in FreeSwitch or libfreeswitch? This would be in 
> a sessionless dialog as of RFC 3428 Page mode.
>
> Thanks,
>
> John
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313

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From nagalenoj at gmail.com  Wed Dec  8 14:45:24 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 8 Dec 2010 17:15:24 +0530
Subject: [Freeswitch-users] No media when bind_meta_app
Message-ID: <AANLkTimn03z=+Apzg-o5m1euiEeBdd6pNS-huVs6BA3A@mail.gmail.com>

Dear friends,
     I've tried using bind_meta_app for recording the call in my ESL script.
It works fine as intended, But I face a serious media breakage(a and b leg
doesn't hear the opposite leg's voice) for about 5 seconds once the person
enters the DTMF to record.

ESL script:
http://pastebin.freeswitch.org/14734

Freeswitch log:
http://pastebin.freeswitch.org/14735

I tried applications other than record_session and I face the same with all
applications. Have I done something?!
Help me to solve this.

-- 
Thanks.
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From lakindia89 at gmail.com  Wed Dec  8 16:01:13 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 8 Dec 2010 18:31:13 +0530
Subject: [Freeswitch-users] how bind_meta_app works?
In-Reply-To: <AANLkTi=m0ww373iFn0nrZQ4hofyAp6duN=-Y4G-wyPgt@mail.gmail.com>
References: <AANLkTi=m0ww373iFn0nrZQ4hofyAp6duN=-Y4G-wyPgt@mail.gmail.com>
Message-ID: <AANLkTi=O8WzT2a-NPCtaQ3ebgbTuzJ+YkTDMdXUsx7Ti@mail.gmail.com>

Hi all,
Is my understanding is right??. If so, is it possible to run applications in
A leg...


On Mon, Dec 6, 2010 at 12:19 PM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi all,
> I was experimenting the bind_mea_app and I have a doubt.
> There is a call from 1000 to FreeSwitch extension and it is connecting to
> event outbound socket.
> I've the following script in place.
>
> #!/usr/bin/perl
> use strict;
> use warnings;
> use Data::Dumper;
> use lib '/usr/src/freeswitch/libs/esl/perl/';
> use IO::Socket::INET;
> use ESL;
>
> my $SOCK = new IO::Socket::INET ( LocalHost => 'localhost',  LocalPort =>
> '8447',  Proto => 'tcp',  Listen => 10,  Reuse => 1 );
> unless ($SOCK)  {
>                 print("Could not create socket: $!");
>                 exit(2);
> }
>
> while (1)  {
>                 # Wait for any client through accept function in socket
> module.
>                 my $new_sock = $SOCK->accept();
>                 next if (not defined($new_sock));
>                 # Get socket host
>                 my $host = $new_sock->sockhost();
>                 print("Got a client accepted from $host\n");
>                 my $pid = fork();
>                 if ($pid) {
>                                 close($new_sock);
>                                 next;
>                 }
>                 my $fd = fileno($new_sock);
>                 print("Newly forked child, pid: $$\n");
>                 my $EslCon = new ESL::ESLconnection($fd);
>                 print "Connection created successfully\n";
>                 my $info = $EslCon->getInfo();
>                 $EslCon->setEventLock("true");
>                 my $uuid = $info->getHeader("unique-id");
>                 print Dumper $info->serialize();
>                 $EslCon->execute("set","bind_meta_key=#");
>                 $EslCon->execute("bind_meta_app","1 b o
> event::appli=testing");
>                 my $api =
> $EslCon->execute("bridge","{ignore_early_media=true}freetdm/1/a/xxxxxxxxx");
> }
>
> The script called to the number that was dialed. In that number I pressed
> #1, but the event was not sent,
> and I assume that the bridge application is executing in the A leg ( is it
> correct?? ) . So only after that bridge application gets completed the event
> gets triggered out.
>
> If I execute the application in the B leg, then it is working fine, since
> there is no application that is running at that time.
> Now I wanted to know if there is any way to run the application on A leg, I
> need that application to be executed even the bridge is not completed.
>
>
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From srinivas.ksvreddy at gmail.com  Wed Dec  8 16:33:56 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 8 Dec 2010 19:03:56 +0530
Subject: [Freeswitch-users] mod_managed.so Loading error:
	mono_class_from_name undefind symbol
Message-ID: <AANLkTik0_ZMPpvscmQzeG7VSLWfXJE-BOM3fVu2gP3Wx@mail.gmail.com>

HI All,

When we trying to load mod_managed.so in CentOS, We are facing a issue,

*mono_class_from_name undefind symbol.*
**
 mono2.8.1 we are using for this.
**
Prevously we got mono_thread_attach error in runtime, we commented the
mono_thread_attach in mod_managed.cpp then we are facing the above error.

Any Idea?

Thanks--
Srinivasula Reddy K
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From gmaruzz at celliax.org  Wed Dec  8 17:21:08 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 8 Dec 2010 15:21:08 +0100
Subject: [Freeswitch-users] get error on skypopen module
In-Reply-To: <AANLkTim5qghatZP8CbYmFfV004PTNfwWx=P8u3Vvgk0A@mail.gmail.com>
References: <AANLkTimB+Nkc=S5gbOqrGFBnZo7rXKwVXs7uTc3FrteR@mail.gmail.com>
	<358063.99637.qm@web32401.mail.mud.yahoo.com>
	<AANLkTim5qghatZP8CbYmFfV004PTNfwWx=P8u3Vvgk0A@mail.gmail.com>
Message-ID: <AANLkTik79k7p1h9HBJ=jvzKMGG+FugTp=9e0hjHVXr-B@mail.gmail.com>

Yep, I'm getting the same performances on a Q6600.

I'm trying since very longtime to come out with something to radically lower
the load on the machine, but until now I've wasted my time on false tracks.

As soon as I have a breakthrough, I'll communicate a lot ;), but maybe I
just hit the wall, and no further performance gains are possible.

On a Q6600 I believe you can have maybe 30 or 40 concurrent calls, but I
highly doubt you can have more than 40 with the current
skype-skypopen-alsadummy.

I'm experimenting with skype.static.oss and a new oss driver I wrote, but no
joy for now.

-giovanni

On Wed, Dec 8, 2010 at 5:45 AM, Jian Ren <renjian at gmail.com> wrote:

> Yes, I got the same. Unless Giovanni doesn't agree :-).
> Jian
>
>
> On Tue, Dec 7, 2010 at 9:46 PM, ha do <haloha201 at yahoo.com> wrote:
>
>> Hi Jian
>>
>> so my freeswitch performance is normal state when run with skypopen :D
>>
>> Thank you
>> Ha
>>
>> --- On *Tue, 12/7/10, Jian Ren <renjian at gmail.com>* wrote:
>>
>>
>> From: Jian Ren <renjian at gmail.com>
>>
>> Subject: Re: [Freeswitch-users] get error on skypopen module
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Date: Tuesday, December 7, 2010, 2:14 PM
>>
>>
>> I am getting 5-6% for each skype call and 1% idle instance, on my Q6600.
>> It's normal, 20 concurrent calls would be way enough for most people. If you
>> want more, can scale to more servers.
>>
>> Jian
>>
>> On Tue, Dec 7, 2010 at 9:52 AM, ha do <haloha201 at yahoo.com<http://mc/compose?to=haloha201 at yahoo.com>
>> > wrote:
>>
>> Hi Steven
>>
>> yes, i use the snd-dummy as guide in startskype.sh script
>>
>>
>> thank you for your guide, it works now :D
>> Ha`
>>
>> --- On Tue, 12/7/10, Steven Ayre <steveayre at gmail.com<http://mc/compose?to=steveayre at gmail.com>>
>> wrote:
>>
>> > From: Steven Ayre <steveayre at gmail.com<http://mc/compose?to=steveayre at gmail.com>
>> >
>> > Subject: Re: [Freeswitch-users] get error on skypopen module
>> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users at lists.freeswitch.org>
>> >
>> > Date: Tuesday, December 7, 2010, 6:40 AM
>> > > how to turn off the debug??
>> > remove the below line in skypopen.xml.conf
>> > >    <param name="debug" value="8"/>
>> >
>> > Yes, remove that line, then do reloadxml and reload
>> > mod_skypopen to
>> > pick up the new config. Or restart FS.
>> >
>> > > i do check the CPU performance so 1 call from/to skype
>> > client it will cost 5% CPU of server
>> > > if 2 calls, CPU will be 10%
>> > >
>> > > i dont know that is normal when freeswitch run with
>> > skypopen module
>> >
>> > Are you using the dummy sound driver? That's meant to
>> > reduce the CPU
>> > usage, but I don't know how much. % will also depend on
>> > what CPU
>> > you're using.
>> >
>> > > how do i change the
>> > > Would you like to receive list mail batched in a daily
>> > digest?
>> > > from yes to no  in
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> > 1) Visit that link
>> > 2) Enter your email and click 'Unsubscribe or edit options'
>> > near the bottom
>> > 3) Enter your password (there's a reminder button that'll
>> > email it to
>> > you if you don't know it)
>> > 4) Scroll down to the 'Your FreeSWITCH-users Subscription
>> > Options' section
>> > 5) Flip the 'Set Digest Mode' setting to Off and click
>> > 'Submit my changes'
>> >
>> > -Steve
>> >
>> >
>> >
>> >
>> >
>> >
>> > On 7 December 2010 12:29, ha do <haloha201 at yahoo.com<http://mc/compose?to=haloha201 at yahoo.com>
>> >
>> > wrote:
>> > > Hi Giovanny
>> > >
>> > > how to turn off the debug?? remove the below line in
>> > skypopen.xml.conf
>> > >    <param name="debug" value="8"/>
>> > >
>> > > the error i get when i run the
>> > /usr/local/freeswitch/bin/fs_cli
>> > >
>> > > i do check the CPU performance so 1 call from/to skype
>> > client it will cost 5% CPU of server
>> > > if 2 calls, CPU will be 10%
>> > >
>> > > i dont know that is normal when freeswitch run with
>> > skypopen module
>> > >
>> > >
>> > > how do i change the
>> > > Would you like to receive list mail batched in a daily
>> > digest?
>> > > from yes to no  in
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >
>> > >
>> > > Thank you
>> > > Ha
>> > >
>> > >
>> > > On Tue, Dec 7, 2010 at 5:34 AM, ha do <haloha201 at yahoo.com<http://mc/compose?to=haloha201 at yahoo.com>
>> >
>> > wrote:
>> > >> the skypopen works fine but i get error below in
>> > the debug mode
>> > >
>> > > don't run in debug mode if you are not debugging :)
>> > >
>> > >> [interface1     ][IDLE,IDLE] Skype got ERROR
>> > about a failed action (probably TRYING to HANGUP A CALL), no
>> > problem: |||ERROR 559 CALL: Action failed|||
>> > >
>> > > as is clearly written, this is not a problem. Is a
>> > debug message that
>> > > explains we tried to hangup a skype call that the
>> > skype client closed
>> > > before. Is just to be double sure that we try to close
>> > it anyway. The
>> > > debug message is just an explanation.
>> > >
>> > >>
>> > >> How to fix it
>> > >
>> > > nothing to fix, no problems here.
>> > >
>> > > -giovanni
>> > >
>> > > --
>> > > Sincerely,
>> > >
>> > > Giovanni Maruzzelli
>> > > Cell : +39-347-2665618
>> > >
>> > >
>> > >
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> -----Inline Attachment Follows-----
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From Nabble at slickdeals.endjunk.com  Wed Dec  8 17:23:04 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 8 Dec 2010 06:23:04 -0800 (PST)
Subject: [Freeswitch-users] OSTAG - How You Can Help
In-Reply-To: <AANLkTim+17x7vApY9Ag6-hsD6gPS1CoyWWGSr8f13zDr@mail.gmail.com>
References: <AANLkTim+17x7vApY9Ag6-hsD6gPS1CoyWWGSr8f13zDr@mail.gmail.com>
Message-ID: <1291818184359-5815375.post@n2.nabble.com>



Michael Collins-2 wrote:
> The OSTAG team would like to let everyone know that we are ready for
> non-profit work.
I thought FS is already a non-profit organization even sans OSTAG.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/OSTAG-How-You-Can-Help-tp5813907p5815375.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kris at kriskinc.com  Wed Dec  8 17:43:47 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 8 Dec 2010 09:43:47 -0500
Subject: [Freeswitch-users] SIP MESSAGE requests
In-Reply-To: <4CFF3EC7.5030605@gmail.com>
References: <004201cb9662$d770dad0$86529070$@comtex.net>
	<4CFF3EC7.5030605@gmail.com>
Message-ID: <AANLkTin3gep6cpQoV0b4NEV2Tqgxe6LGyVJE5ymu46NA@mail.gmail.com>

You don't have to be a proxy to handle the MESSAGE method.  The OP is
asking to handle MESSAGE requests within the capability of a SIP UA,
which FreeSWITCH is.

As more carriers and products offer SMS capabilities with SIP using
MESSAGE it will become more and more important for FreeSWITCH to have
support this feature in some way.

I think it would be very, very cool!

On Wed, Dec 8, 2010 at 3:16 AM, Meftah Tayeb <tayeb.meftah at gmail.com> wrote:
> freeswitch isn't a sip proxy
> freeswitch is a media switching project growing up from a softphone up to a
> carrier grade softswitch/SBC but not a sip proxy
> check:
> Kamailio/Sip-Router Projects
> OpenSips project
> don't by confused, this projects is both derived from openser that's becaume
> kamailio
> thanks
> Le 08/12/2010 00:02, John Rose a ?crit?:
>
> Hello,
>
>
>
> Does anyone know if it is possible to construct, originate and receive SIP
> MESSAGE requests in FreeSwitch or libfreeswitch? This would be in a
> sessionless dialog as of RFC 3428 Page mode.
>
>
>
> Thanks,
>
> John
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Meftah Tayeb
> inum: +883510001288000
> Phone: +13602276297
> Fax: +12538020313
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From gmaruzz at gmail.com  Wed Dec  8 17:48:25 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 8 Dec 2010 15:48:25 +0100
Subject: [Freeswitch-users] SIP MESSAGE requests
In-Reply-To: <AANLkTin3gep6cpQoV0b4NEV2Tqgxe6LGyVJE5ymu46NA@mail.gmail.com>
References: <004201cb9662$d770dad0$86529070$@comtex.net>
	<4CFF3EC7.5030605@gmail.com>
	<AANLkTin3gep6cpQoV0b4NEV2Tqgxe6LGyVJE5ymu46NA@mail.gmail.com>
Message-ID: <AANLkTikBonXPZ1TaQ+1cLjq3zdMgW0Rznq5RLkQ_bcgz@mail.gmail.com>

FS supports MESSAGE, both in SIP, in Jingle, in Skype, and in GSM (SMS).

The generic API for it called chat
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_chat

It generates MESSAGE events

-giovanni

On Wed, Dec 8, 2010 at 3:43 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> You don't have to be a proxy to handle the MESSAGE method. ?The OP is
> asking to handle MESSAGE requests within the capability of a SIP UA,
> which FreeSWITCH is.
>
> As more carriers and products offer SMS capabilities with SIP using
> MESSAGE it will become more and more important for FreeSWITCH to have
> support this feature in some way.
>
> I think it would be very, very cool!
>
> On Wed, Dec 8, 2010 at 3:16 AM, Meftah Tayeb <tayeb.meftah at gmail.com> wrote:
>> freeswitch isn't a sip proxy
>> freeswitch is a media switching project growing up from a softphone up to a
>> carrier grade softswitch/SBC but not a sip proxy
>> check:
>> Kamailio/Sip-Router Projects
>> OpenSips project
>> don't by confused, this projects is both derived from openser that's becaume
>> kamailio
>> thanks
>> Le 08/12/2010 00:02, John Rose a ?crit?:
>>
>> Hello,
>>
>>
>>
>> Does anyone know if it is possible to construct, originate and receive SIP
>> MESSAGE requests in FreeSwitch or libfreeswitch? This would be in a
>> sessionless dialog as of RFC 3428 Page mode.
>>
>>
>>
>> Thanks,
>>
>> John
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> Meftah Tayeb
>> inum: +883510001288000
>> Phone: +13602276297
>> Fax: +12538020313
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From mrene_lists at avgs.ca  Wed Dec  8 17:49:32 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 8 Dec 2010 09:49:32 -0500
Subject: [Freeswitch-users] SIP MESSAGE requests
In-Reply-To: <AANLkTin3gep6cpQoV0b4NEV2Tqgxe6LGyVJE5ymu46NA@mail.gmail.com>
References: <004201cb9662$d770dad0$86529070$@comtex.net>
	<4CFF3EC7.5030605@gmail.com>
	<AANLkTin3gep6cpQoV0b4NEV2Tqgxe6LGyVJE5ymu46NA@mail.gmail.com>
Message-ID: <B6823144-18AC-42BA-BF9E-9027255A45D0@avgs.ca>

Check out http://wiki.freeswitch.org/wiki/Mod_event_socket#sendevent 

There's an example on how to send a SIP MESSAGE using event socket's sendevent method.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-12-08, at 9:43 AM, Kristian Kielhofner wrote:

> You don't have to be a proxy to handle the MESSAGE method.  The OP is
> asking to handle MESSAGE requests within the capability of a SIP UA,
> which FreeSWITCH is.
> 
> As more carriers and products offer SMS capabilities with SIP using
> MESSAGE it will become more and more important for FreeSWITCH to have
> support this feature in some way.
> 
> I think it would be very, very cool!
> 
> On Wed, Dec 8, 2010 at 3:16 AM, Meftah Tayeb <tayeb.meftah at gmail.com> wrote:
>> freeswitch isn't a sip proxy
>> freeswitch is a media switching project growing up from a softphone up to a
>> carrier grade softswitch/SBC but not a sip proxy
>> check:
>> Kamailio/Sip-Router Projects
>> OpenSips project
>> don't by confused, this projects is both derived from openser that's becaume
>> kamailio
>> thanks
>> Le 08/12/2010 00:02, John Rose a ?crit :
>> 
>> Hello,
>> 
>> 
>> 
>> Does anyone know if it is possible to construct, originate and receive SIP
>> MESSAGE requests in FreeSwitch or libfreeswitch? This would be in a
>> sessionless dialog as of RFC 3428 Page mode.
>> 
>> 
>> 
>> Thanks,
>> 
>> John
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> --
>> Meftah Tayeb
>> inum: +883510001288000
>> Phone: +13602276297
>> Fax: +12538020313
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 
> 
> -- 
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kris at kriskinc.com  Wed Dec  8 18:00:20 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 8 Dec 2010 10:00:20 -0500
Subject: [Freeswitch-users] SIP MESSAGE requests
In-Reply-To: <B6823144-18AC-42BA-BF9E-9027255A45D0@avgs.ca>
References: <004201cb9662$d770dad0$86529070$@comtex.net>
	<4CFF3EC7.5030605@gmail.com>
	<AANLkTin3gep6cpQoV0b4NEV2Tqgxe6LGyVJE5ymu46NA@mail.gmail.com>
	<B6823144-18AC-42BA-BF9E-9027255A45D0@avgs.ca>
Message-ID: <AANLkTikP1qHa4-PtMjseJHoDxvJ5Fo7a5mfWa2aTA-Kc@mail.gmail.com>

While FS does handle MESSAGE the interface has always struck me as a bit odd...

For example, why don't FS messages hit the dialplan to provide for
some ability to route them if I want to?

In my case I typically use FS as an SBC facing carriers...  If a
carrier sends me a MESSAGE FS will generate an event.  While I *could*
do something with that, I suppose, I'd much rather have it hit the
dialplan where I can use some logic to handle it.  I could see at
least the following:

- Post to an HTTP server
- Route to another SIP endpoint
- LUA!
- Generate an event (as it does now)

Smells like a bounty...

On Wed, Dec 8, 2010 at 9:49 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> Check out?http://wiki.freeswitch.org/wiki/Mod_event_socket#sendevent
> There's an example on how to send a SIP MESSAGE using event socket's
> sendevent method.
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
> On 2010-12-08, at 9:43 AM, Kristian Kielhofner wrote:
>
> You don't have to be a proxy to handle the MESSAGE method. ?The OP is
> asking to handle MESSAGE requests within the capability of a SIP UA,
> which FreeSWITCH is.
>
> As more carriers and products offer SMS capabilities with SIP using
> MESSAGE it will become more and more important for FreeSWITCH to have
> support this feature in some way.
>
> I think it would be very, very cool!
>
> On Wed, Dec 8, 2010 at 3:16 AM, Meftah Tayeb <tayeb.meftah at gmail.com> wrote:
>
> freeswitch isn't a sip proxy
>
> freeswitch is a media switching project growing up from a softphone up to a
>
> carrier grade softswitch/SBC but not a sip proxy
>
> check:
>
> Kamailio/Sip-Router Projects
>
> OpenSips project
>
> don't by confused, this projects is both derived from openser that's becaume
>
> kamailio
>
> thanks
>
> Le 08/12/2010 00:02, John Rose a ?crit?:
>
> Hello,
>
>
>
> Does anyone know if it is possible to construct, originate and receive SIP
>
> MESSAGE requests in FreeSwitch or libfreeswitch? This would be in a
>
> sessionless dialog as of RFC 3428 Page mode.
>
>
>
> Thanks,
>
> John
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> --
>
> Meftah Tayeb
>
> inum: +883510001288000
>
> Phone: +13602276297
>
> Fax: +12538020313
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From mail at jankubr.com  Wed Dec  8 18:01:51 2010
From: mail at jankubr.com (Jan Kubr)
Date: Wed, 8 Dec 2010 16:01:51 +0100
Subject: [Freeswitch-users] SIP Registration DNS Error
In-Reply-To: <06502C073AD9394AADB3CA7FD94931BC0519DF46@okc1x1.Logixcom.com>
References: <ActqMzH/pjYEcWn1Q/KYdFPQ7356oA==>
	<06502C073AD9394AADB3CA7FD94931BC0519DF46@okc1x1.Logixcom.com>
Message-ID: <AANLkTi=10HbngBxzwzf=i5PpKMhG5tdzWvtcCdA_weJt@mail.gmail.com>

I have the same problem with one gateway. What also helps is restarting the
profile:

sofia profile external restart reloadxml

But that means dropping the calls as well.

Anyone knows how to fix this?

Thanks,
Jan

On Tue, Oct 12, 2010 at 7:27 PM, Joshua Foshee
<Joshua.Foshee at logixcom.com>wrote:

> I have setup up two sip providers that I can connect to fine but after a
> while I then start to get these messages.
>
>
>
> 2010-10-12 07:27:57.292742 [NOTICE] sofia_reg.c:342 Registering flowroute
>
> 2010-10-12 07:27:57.295138 [ERR] sofia_reg.c:1611 flowroute Registration
> Failed with status DNS Error [503]. failure #1391
>
> 2010-10-12 07:27:59.959022 [WARNING] sofia_reg.c:387 flowroute Failed
> Registration, setting retry to 30 seconds.
>
>
>
> Here is the output of the gateway status
>
>
>
> Name            flowroute
>
> Profile         external
>
> Scheme          Digest
>
> Realm           sip.flowroute.com
>
> Username        xxxxxxx
>
> Password        yes
>
> From            <sip:xxxxxxxxx at sip.flowroute.com<sip%3Axxxxxxxxx at sip.flowroute.com>
> ;transport=udp>
>
> Contact         <sip:gw+flowroute at xxxxxxxx
> :5080;transport=udp;gw=flowroute>
>
> Exten           xxxxxxx
>
> To              sip:xxxxxxxxx at sip.flowroute.com<sip%3Axxxxxxxxx at sip.flowroute.com>
>
> Proxy           sip:sip.flowroute.com
>
> Context         public
>
> Expires         600
>
> Freq            600
>
> Ping            1286835829
>
> PingFreq        25
>
> PingState       -1/0/1
>
> State           FAIL_WAIT
>
> Status          DOWN
>
> CallsIN         0
>
> CallsOUT        5
>
> FailedCallsIN   0
>
> FailedCallsOUT  5
>
>
>
>
>
> Name            broadvoice
>
> Profile         external
>
> Scheme          Digest
>
> Realm           BroadWorks
>
> Username        xxxxxxxx
>
> Password        yes
>
> From            <sip:xxxxxxxxxxxx at sip.broadvoice.com<sip%3Axxxxxxxxxxxx at sip.broadvoice.com>
> ;transport=udp>
>
> Contact         <sip:gw+broadvoice at xxxxxxxxxx
> :5080;transport=udp;gw=broadvoice>
>
> Exten           xxxxxxxx
>
> To              sip:xxxxxxxxxx at sip.broadvoice.com<sip%3Axxxxxxxxxx at sip.broadvoice.com>
>
> Proxy           sip:sip.broadvoice.com
>
> Context         public
>
> Expires         30
>
> Freq            30
>
> Ping            0
>
> PingFreq        0
>
> PingState       0/0/0
>
> State           FAIL_WAIT
>
> Status          DOWN
>
> CallsIN         0
>
> CallsOUT        2
>
> FailedCallsIN   0
>
> FailedCallsOUT  3
>
>
>
> If I restart Freeswitch process they both come up and Reg just fine for a
> while till it fails again.
>
>
>
> Thanks in advance,
>
> Josh
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mail at jankubr.com  Wed Dec  8 18:02:58 2010
From: mail at jankubr.com (Jan Kubr)
Date: Wed, 8 Dec 2010 16:02:58 +0100
Subject: [Freeswitch-users] mod_pocketsphinx
In-Reply-To: <AF932101-9B49-43F6-A488-8E1C5897E88F@freeswitch.org>
References: <4C1997A0.1080909@gmail.com>
	<BA0A1EAE-474B-4FE1-8BB3-03BC4B1C3655@freeswitch.org>
	<4C1ACD2A.70906@gmail.com>
	<D7122611-F34A-40B4-B0BE-C95A3D551E09@freeswitch.org>
	<4C1AE301.30405@gmail.com> <4C3D48A5.6030508@todandlorna.com>
	<C4BCDAB3-5371-48D5-810F-1A8F299529F6@freeswitch.org>
	<AANLkTi=aTwdANAkNR1V5HXnnrZD8FDeRVWKJktWYCGFa@mail.gmail.com>
	<AF932101-9B49-43F6-A488-8E1C5897E88F@freeswitch.org>
Message-ID: <AANLkTimOskco=-=uDGdU8OPTOnrU0LXD7iAQyKFo8-4e@mail.gmail.com>

Were you able to reproduce the problem? Not sure where to look to get rid of
this :(

Jan

On Mon, Dec 6, 2010 at 8:19 PM, Brian West <brian at freeswitch.org> wrote:

> No clue I'll try to compile it again today.
>
> /b
>
> On Dec 6, 2010, at 12:16 PM, Jan Kubr wrote:
>
> I've run into the same issue and can't find any pocketsphinx related
> packages on this machine.
> Did you guys figure out what the problem was? If it was a system package,
> what was its name?
>
> Thanks!
> Jan
>
> On Wed, Jul 14, 2010 at 7:39 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Yes it works fine if you don't use Ubuntu's packages and wipe them off the
>> system 100%.  We download and build a more bleeding edge version.
>>
>> /b
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Wed Dec  8 18:05:53 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 8 Dec 2010 16:05:53 +0100
Subject: [Freeswitch-users] SIP Registration DNS Error
In-Reply-To: <AANLkTi=10HbngBxzwzf=i5PpKMhG5tdzWvtcCdA_weJt@mail.gmail.com>
References: <ActqMzH/pjYEcWn1Q/KYdFPQ7356oA==>
	<06502C073AD9394AADB3CA7FD94931BC0519DF46@okc1x1.Logixcom.com>
	<AANLkTi=10HbngBxzwzf=i5PpKMhG5tdzWvtcCdA_weJt@mail.gmail.com>
Message-ID: <49E1E293-BE37-43AA-960C-56B921FEBC9D@ipeva.fr>

I think I remember that Flowroute has a DNS issue.
At least that was the case some months ago.
Use an IP, that should fix it.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 08/12/2010 ? 16:01, Jan Kubr a ?crit :

> I have the same problem with one gateway. What also helps is restarting the profile:
> 
> sofia profile external restart reloadxml
> 
> But that means dropping the calls as well. 
> 
> Anyone knows how to fix this?
> 
> Thanks,
> Jan
> 
> On Tue, Oct 12, 2010 at 7:27 PM, Joshua Foshee <Joshua.Foshee at logixcom.com> wrote:
> I have setup up two sip providers that I can connect to fine but after a while I then start to get these messages.
> 
>  
> 2010-10-12 07:27:57.292742 [NOTICE] sofia_reg.c:342 Registering flowroute
> 
> 2010-10-12 07:27:57.295138 [ERR] sofia_reg.c:1611 flowroute Registration Failed with status DNS Error [503]. failure #1391
> 
> 2010-10-12 07:27:59.959022 [WARNING] sofia_reg.c:387 flowroute Failed Registration, setting retry to 30 seconds.
> 
>  
> Here is the output of the gateway status
> 
>  
> Name            flowroute
> 
> Profile         external
> 
> Scheme          Digest
> 
> Realm           sip.flowroute.com
> 
> Username        xxxxxxx
> 
> Password        yes
> 
> From            <sip:xxxxxxxxx at sip.flowroute.com;transport=udp>
> 
> Contact         <sip:gw+flowroute at xxxxxxxx:5080;transport=udp;gw=flowroute>
> 
> Exten           xxxxxxx
> 
> To              sip:xxxxxxxxx at sip.flowroute.com
> 
> Proxy           sip:sip.flowroute.com
> 
> Context         public
> 
> Expires         600
> 
> Freq            600
> 
> Ping            1286835829
> 
> PingFreq        25
> 
> PingState       -1/0/1
> 
> State           FAIL_WAIT
> 
> Status          DOWN
> 
> CallsIN         0
> 
> CallsOUT        5
> 
> FailedCallsIN   0
> 
> FailedCallsOUT  5
> 
>  
>  
> Name            broadvoice
> 
> Profile         external
> 
> Scheme          Digest
> 
> Realm           BroadWorks
> 
> Username        xxxxxxxx
> 
> Password        yes
> 
> From            <sip:xxxxxxxxxxxx at sip.broadvoice.com;transport=udp>
> 
> Contact         <sip:gw+broadvoice at xxxxxxxxxx:5080;transport=udp;gw=broadvoice>
> 
> Exten           xxxxxxxx
> 
> To              sip:xxxxxxxxxx at sip.broadvoice.com
> 
> Proxy           sip:sip.broadvoice.com
> 
> Context         public
> 
> Expires         30
> 
> Freq            30
> 
> Ping            0
> 
> PingFreq        0
> 
> PingState       0/0/0
> 
> State           FAIL_WAIT
> 
> Status          DOWN
> 
> CallsIN         0
> 
> CallsOUT        2
> 
> FailedCallsIN   0
> 
> FailedCallsOUT  3
> 
>  
> If I restart Freeswitch process they both come up and Reg just fine for a while till it fails again.
> 
>  
> Thanks in advance,
> 
> Josh
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Wed Dec  8 18:08:34 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 8 Dec 2010 09:08:34 -0600
Subject: [Freeswitch-users] OSTAG - How You Can Help
In-Reply-To: <1291818184359-5815375.post@n2.nabble.com>
References: <AANLkTim+17x7vApY9Ag6-hsD6gPS1CoyWWGSr8f13zDr@mail.gmail.com>
	<1291818184359-5815375.post@n2.nabble.com>
Message-ID: <18CDEC8B-5EF7-4A0F-B06E-F43F901B1A99@freeswitch.org>

FreeSWITCH is a non-profit as well but we do not have the 501(c)(3) that allows companies to give to us and let them write it off on their taxes OSTAG does.

/b

On Dec 8, 2010, at 8:23 AM, mazilo wrote:

> 
> 
> Michael Collins-2 wrote:
>> The OSTAG team would like to let everyone know that we are ready for
>> non-profit work.
> I thought FS is already a non-profit organization even sans OSTAG.
> 
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> -- 
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/OSTAG-How-You-Can-Help-tp5813907p5815375.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.

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From brian at freeswitch.org  Wed Dec  8 18:16:15 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 8 Dec 2010 09:16:15 -0600
Subject: [Freeswitch-users] mod_pocketsphinx
In-Reply-To: <AANLkTimOskco=-=uDGdU8OPTOnrU0LXD7iAQyKFo8-4e@mail.gmail.com>
References: <4C1997A0.1080909@gmail.com>
	<BA0A1EAE-474B-4FE1-8BB3-03BC4B1C3655@freeswitch.org>
	<4C1ACD2A.70906@gmail.com>
	<D7122611-F34A-40B4-B0BE-C95A3D551E09@freeswitch.org>
	<4C1AE301.30405@gmail.com> <4C3D48A5.6030508@todandlorna.com>
	<C4BCDAB3-5371-48D5-810F-1A8F299529F6@freeswitch.org>
	<AANLkTi=aTwdANAkNR1V5HXnnrZD8FDeRVWKJktWYCGFa@mail.gmail.com>
	<AF932101-9B49-43F6-A488-8E1C5897E88F@freeswitch.org>
	<AANLkTimOskco=-=uDGdU8OPTOnrU0LXD7iAQyKFo8-4e@mail.gmail.com>
Message-ID: <CE8866A5-FAB1-429E-A911-C3EF0625651C@freeswitch.org>

Compiles fine on CentOS not sure what your issue is but someone that cares about Ubuntu should probably figure it out and post patches if possible.

/b

On Dec 8, 2010, at 9:02 AM, Jan Kubr wrote:

> Were you able to reproduce the problem? Not sure where to look to get rid of this :(
> 
> Jan
> 
> On Mon, Dec 6, 2010 at 8:19 PM, Brian West <brian at freeswitch.org> wrote:
> No clue I'll try to compile it again today.
> 
> /b
> 

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From brian at freeswitch.org  Wed Dec  8 18:17:14 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 8 Dec 2010 09:17:14 -0600
Subject: [Freeswitch-users] SIP MESSAGE requests
In-Reply-To: <AANLkTikP1qHa4-PtMjseJHoDxvJ5Fo7a5mfWa2aTA-Kc@mail.gmail.com>
References: <004201cb9662$d770dad0$86529070$@comtex.net>
	<4CFF3EC7.5030605@gmail.com>
	<AANLkTin3gep6cpQoV0b4NEV2Tqgxe6LGyVJE5ymu46NA@mail.gmail.com>
	<B6823144-18AC-42BA-BF9E-9027255A45D0@avgs.ca>
	<AANLkTikP1qHa4-PtMjseJHoDxvJ5Fo7a5mfWa2aTA-Kc@mail.gmail.com>
Message-ID: <18B23402-07D9-4FBF-9CF5-E7EE1F4D1F9E@freeswitch.org>

EXACTLY!  ;)

/b

On Dec 8, 2010, at 9:00 AM, Kristian Kielhofner wrote:

> Smells like a bounty...



From mrene_lists at avgs.ca  Wed Dec  8 18:34:10 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 8 Dec 2010 10:34:10 -0500
Subject: [Freeswitch-users] SIP MESSAGE requests
In-Reply-To: <AANLkTikP1qHa4-PtMjseJHoDxvJ5Fo7a5mfWa2aTA-Kc@mail.gmail.com>
References: <004201cb9662$d770dad0$86529070$@comtex.net>
	<4CFF3EC7.5030605@gmail.com>
	<AANLkTin3gep6cpQoV0b4NEV2Tqgxe6LGyVJE5ymu46NA@mail.gmail.com>
	<B6823144-18AC-42BA-BF9E-9027255A45D0@avgs.ca>
	<AANLkTikP1qHa4-PtMjseJHoDxvJ5Fo7a5mfWa2aTA-Kc@mail.gmail.com>
Message-ID: <9B0E926C-3238-4388-B135-9F1C7899EEF8@avgs.ca>

The problem with that concept, is that dialplan access requires an active session, and it doesn't make sense to create one for something as short-lived as a message. What I could see happening is some sort of customizable event routing that could be configured  to do some automated actions.

Yup, indeed smells like a bounty.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-12-08, at 10:00 AM, Kristian Kielhofner wrote:

> While FS does handle MESSAGE the interface has always struck me as a bit odd...
> 
> For example, why don't FS messages hit the dialplan to provide for
> some ability to route them if I want to?
> 
> In my case I typically use FS as an SBC facing carriers...  If a
> carrier sends me a MESSAGE FS will generate an event.  While I *could*
> do something with that, I suppose, I'd much rather have it hit the
> dialplan where I can use some logic to handle it.  I could see at
> least the following:
> 
> - Post to an HTTP server
> - Route to another SIP endpoint
> - LUA!
> - Generate an event (as it does now)
> 
> Smells like a bounty...
> 
> On Wed, Dec 8, 2010 at 9:49 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>> Check out http://wiki.freeswitch.org/wiki/Mod_event_socket#sendevent
>> There's an example on how to send a SIP MESSAGE using event socket's
>> sendevent method.
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>> 
>> 
>> 
>> On 2010-12-08, at 9:43 AM, Kristian Kielhofner wrote:
>> 
>> You don't have to be a proxy to handle the MESSAGE method.  The OP is
>> asking to handle MESSAGE requests within the capability of a SIP UA,
>> which FreeSWITCH is.
>> 
>> As more carriers and products offer SMS capabilities with SIP using
>> MESSAGE it will become more and more important for FreeSWITCH to have
>> support this feature in some way.
>> 
>> I think it would be very, very cool!
>> 
>> On Wed, Dec 8, 2010 at 3:16 AM, Meftah Tayeb <tayeb.meftah at gmail.com> wrote:
>> 
>> freeswitch isn't a sip proxy
>> 
>> freeswitch is a media switching project growing up from a softphone up to a
>> 
>> carrier grade softswitch/SBC but not a sip proxy
>> 
>> check:
>> 
>> Kamailio/Sip-Router Projects
>> 
>> OpenSips project
>> 
>> don't by confused, this projects is both derived from openser that's becaume
>> 
>> kamailio
>> 
>> thanks
>> 
>> Le 08/12/2010 00:02, John Rose a ?crit :
>> 
>> Hello,
>> 
>> 
>> 
>> Does anyone know if it is possible to construct, originate and receive SIP
>> 
>> MESSAGE requests in FreeSwitch or libfreeswitch? This would be in a
>> 
>> sessionless dialog as of RFC 3428 Page mode.
>> 
>> 
>> 
>> Thanks,
>> 
>> John
>> 
>> _______________________________________________
>> 
>> FreeSWITCH-users mailing list
>> 
>> FreeSWITCH-users at lists.freeswitch.org
>> 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 
>> http://www.freeswitch.org
>> 
>> 
>> --
>> 
>> Meftah Tayeb
>> 
>> inum: +883510001288000
>> 
>> Phone: +13602276297
>> 
>> Fax: +12538020313
>> 
>> _______________________________________________
>> 
>> FreeSWITCH-users mailing list
>> 
>> FreeSWITCH-users at lists.freeswitch.org
>> 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 
>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> 
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 
> 
> -- 
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From pbdlists at pinboard.com  Wed Dec  8 18:30:25 2010
From: pbdlists at pinboard.com (pbdlists at pinboard.com)
Date: Wed, 8 Dec 2010 16:30:25 +0100
Subject: [Freeswitch-users] fax files not saved
Message-ID: <20101208153025.GB17099@pinboard.com>

Hello!

I managed to solve most of my other problems sofar, but with this one I
have no clue at all. Has anybody an idea what is happening here?

 1 an inbound call to a number which is registered as fax (mod_spandsp)
 2 fax is detected and received according to the log (log entries see below)
 3 but _sometimes_ the tiff file is never written to disk (post processing
   script can't find it and it is really not there)

The destination filesystem has enough free space and free inodes,
permissions for writing are ok, the system is not under any kind of
heavy load, no other calls going on at the same time...

I don't have much data to test this (using www.freepopfax.com for
testing and they have a daily limit on faxes I can send), but it seems
to happen roughly for 30-50% of the incoming fax messages. From the same
fax provider it works fine one time, then not a couple of minutes later,
another couple of minutes later it may work again, all while nothing at
all is being changed on the freeswitch server. If it was something like
a dropped line or anything I could understand it, but the logs say
everything is ok and still _sometimes_ the fax is not written to disk...

Cheers,

Kurt

=======================================================================
entries in fax.conf.xml
=======================================================================
<configuration name="fax.conf" description="FAX application configuration">
    <settings>
        <param name="use-ecm"           value="true"/>
        <param name="verbose"           value="false"/>
        <param name="disable-v17"       value="false"/>
        <param name="ident"             value="SpanDSP Fax Ident"/>
        <param name="header"            value="SpanDSP Fax Header"/>

        <param name="spool-dir"         value="/tmp"/>
        <param name="file-prefix"       value="faxrx"/>
    </settings>
</configuration>


=======================================================================
the part from default.xml
=======================================================================
    <extension name="fax_receive">
        <condition field="destination_number" expression="^(1111)$">
            <action application="set" data="api_hangup_hook=system ${base_dir}/scripts/emailfax.sh $1 /tmp/FAX-${uuid}.tiff '\\\${fax_success}' '\\\${fax_result_code}' '\\\${fax_result_text}' '\\\${fax_local_station_id}' '\\\${fax_remote_station_id}' '\\\${fax_document_transferred_pages}' '\\\${fax_document_total_pages}' '\\\${fax__image_resolution}' '\\\${fax_image_size}' '\\\${fax_bad_rows}' '\\\${fax_transfer_rate}' '\\\${fax_v17_disabled}' '\\\${fax_ecm_requested}' '\\\${fax_filename}' "/>
            <action application="answer" />
            <action application="playback" data="silence_stream://2000"/>
<!--
            <action application="set" data="fax_enable_t38_request=true"/>
            <action application="set" data="fax_enable_t38=true"/>
-->
            <action application="set" data="fax_ident=+41xxxxxxxxx"/>
            <action application="set" data="fax_disable_v17=1"/>
            <action application="set" data="fax_verbose=0"/>
            <action application="rxfax" data="/tmp/FAX-${uuid}.tiff"/>
            <action application="hangup"/>
        </condition>
    </extension>


=======================================================================
log entry from an unsuccessful incoming fax:
=======================================================================

2010-12-07 23:33:51.157588 [DEBUG] sofia_glue.c:182 sofia/external/Anonymous at anonymous.invalid image media sdp:
v=0
o=FreeSWITCH 1291733462 1291733465 IN IP4 xxx.xxx.xxx.xxx
s=FreeSWITCH
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=image 31358 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:2000
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy

2010-12-07 23:33:51.157588 [DEBUG] sofia.c:4597 Channel sofia/external/Anonymous at anonymous.invalid entering state [completed][200]
2010-12-07 23:33:51.157588 [DEBUG] sofia.c:4597 Channel sofia/external/Anonymous at anonymous.invalid entering state [ready][200]
2010-12-07 23:33:58.814979 [DEBUG] switch_rtp.c:2544 Correct ip/port confirmed.
2010-12-07 23:35:09.772752 [DEBUG] switch_channel.c:2457 (sofia/external/Anonymous at anonymous.invalid) Callstate Change ACTIVE -> HANGUP
2010-12-07 23:35:09.772752 [NOTICE] sofia.c:528 Hangup sofia/external/Anonymous at anonymous.invalid [CS_EXECUTE] [NORMAL_CLEARING]
2010-12-07 23:35:09.772752 [DEBUG] switch_channel.c:2473 Send signal sofia/external/Anonymous at anonymous.invalid [KILL]
2010-12-07 23:35:09.772752 [DEBUG] switch_core_session.c:1057 Send signal sofia/external/Anonymous at anonymous.invalid [BREAK]
2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:323 ==============================================================================
2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:329 Fax successfully received.
2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:340 Remote station id: 
2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:341 Local station id:  +41xxxxxxxxx
2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:342 Pages transferred: 2
2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:344 Total fax pages:   2
2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:345 Image resolution:  8031x7700
2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate:     9600
2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:348 ECM status         on
2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:349 remote country:   
2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:350 remote vendor:    
2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:351 remote model:     
2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:353 ==============================================================================
2010-12-07 23:35:09.772752 [DEBUG] switch_core_codec.c:141 sofia/external/Anonymous at anonymous.invalid Restore previous codec PCMA:8.
2010-12-07 23:35:09.772752 [DEBUG] switch_core_session.c:1933 sofia/external/Anonymous at anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:366 (sofia/external/Anonymous at anonymous.invalid) State EXECUTE going to sleep
2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:320 (sofia/external/Anonymous at anonymous.invalid) Running State Change CS_HANGUP
2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:553 (sofia/external/Anonymous at anonymous.invalid) State HANGUP
2010-12-07 23:35:09.772752 [DEBUG] mod_sofia.c:453 sofia/external/Anonymous at anonymous.invalid Overriding SIP cause 480 with 200 from the other leg
2010-12-07 23:35:09.772752 [DEBUG] mod_sofia.c:459 Channel sofia/external/Anonymous at anonymous.invalid hanging up, cause: NORMAL_CLEARING
2010-12-07 23:35:09.821455 [DEBUG] switch_core_state_machine.c:46 sofia/external/Anonymous at anonymous.invalid Standard HANGUP, cause: NORMAL_CLEARING
2010-12-07 23:35:09.821455 [DEBUG] switch_core_state_machine.c:553 (sofia/external/Anonymous at anonymous.invalid) State HANGUP going to sleep
2010-12-07 23:35:09.822803 [NOTICE] mod_commands.c:4187 Executing command: /usr/local/freeswitch/scripts/emailfax.sh 1111 /tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff '1' '0' 'OK' '+41xxxxxxxxx' '' '2' '2
' '' '0' '0' '9600' '1' '1' '/tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff'
2010-12-07 23:35:10.076623 [DEBUG] switch_core_state_machine.c:488 Hangup Command with no Session system(/usr/local/freeswitch/scripts/emailfax.sh 1111 /tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff '1' '0'
 'OK' '+41xxxxxxxxx' '' '2' '2' '' '0' '0' '9600' '1' '1' '/tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff' ):
+OK


=======================================================================
log entry from a successful incoming fax:
=======================================================================

2010-12-07 23:54:59.082301 [DEBUG] sofia_glue.c:182 sofia/external/Anonymous at anonymous.invalid image media sdp:
v=0
o=FreeSWITCH 1291735801 1291735804 IN IP4 xxx.xxx.xxx.xxx
s=FreeSWITCH
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=image 30280 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:2000
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy

2010-12-07 23:54:59.082301 [DEBUG] sofia.c:4597 Channel sofia/external/Anonymous at anonymous.invalid entering state [completed][200]
2010-12-07 23:54:59.100811 [DEBUG] sofia.c:4597 Channel sofia/external/Anonymous at anonymous.invalid entering state [ready][200]
2010-12-07 23:55:06.783427 [DEBUG] switch_rtp.c:2544 Correct ip/port confirmed.
2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:323 ==============================================================================
2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:329 Fax successfully received.
2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:340 Remote station id: 
2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:341 Local station id:  +41xxxxxxxxx
2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:342 Pages transferred: 1
2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:344 Total fax pages:   1
2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:345 Image resolution:  8031x7700
2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate:     9600
2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:348 ECM status         on
2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:349 remote country:   
2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:350 remote vendor:    
2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:351 remote model:     
2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:353 ==============================================================================
2010-12-07 23:55:31.616038 [DEBUG] switch_channel.c:2457 (sofia/external/Anonymous at anonymous.invalid) Callstate Change ACTIVE -> HANGUP
2010-12-07 23:55:31.616038 [NOTICE] sofia.c:528 Hangup sofia/external/Anonymous at anonymous.invalid [CS_EXECUTE] [NORMAL_CLEARING]
2010-12-07 23:55:31.616038 [DEBUG] switch_channel.c:2473 Send signal sofia/external/Anonymous at anonymous.invalid [KILL]
2010-12-07 23:55:31.616038 [DEBUG] switch_core_session.c:1057 Send signal sofia/external/Anonymous at anonymous.invalid [BREAK]
2010-12-07 23:55:31.617356 [DEBUG] switch_core_codec.c:141 sofia/external/Anonymous at anonymous.invalid Restore previous codec PCMA:8.
2010-12-07 23:55:31.617356 [DEBUG] switch_core_session.c:1933 sofia/external/Anonymous at anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2010-12-07 23:55:31.617356 [DEBUG] switch_core_state_machine.c:366 (sofia/external/Anonymous at anonymous.invalid) State EXECUTE going to sleep
2010-12-07 23:55:31.617356 [DEBUG] switch_core_state_machine.c:320 (sofia/external/Anonymous at anonymous.invalid) Running State Change CS_HANGUP
2010-12-07 23:55:31.618542 [DEBUG] switch_core_state_machine.c:553 (sofia/external/Anonymous at anonymous.invalid) State HANGUP
2010-12-07 23:55:31.618542 [DEBUG] mod_sofia.c:453 sofia/external/Anonymous at anonymous.invalid Overriding SIP cause 480 with 200 from the other leg
2010-12-07 23:55:31.618542 [DEBUG] mod_sofia.c:459 Channel sofia/external/Anonymous at anonymous.invalid hanging up, cause: NORMAL_CLEARING
2010-12-07 23:55:31.721993 [DEBUG] switch_core_state_machine.c:46 sofia/external/Anonymous at anonymous.invalid Standard HANGUP, cause: NORMAL_CLEARING
2010-12-07 23:55:31.721993 [DEBUG] switch_core_state_machine.c:553 (sofia/external/Anonymous at anonymous.invalid) State HANGUP going to sleep
2010-12-07 23:55:31.721993 [NOTICE] mod_commands.c:4187 Executing command: /usr/local/freeswitch/scripts/emailfax.sh 1111 /tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff '1' '0' 'OK' '+41xxxxxxxxx' '' '1' '1' '' '0' '0' '9600' '1' '1' '/tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff'
2010-12-07 23:55:31.904090 [DEBUG] switch_core_state_machine.c:488 Hangup Command with no Session system(/usr/local/freeswitch/scripts/emailfax.sh 1111 /tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff '1' '0' 'OK' '+41xxxxxxxxx' '' '1' '1' '' '0' '0' '9600' '1' '1' '/tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff' ):
+OK



From mail at jankubr.com  Wed Dec  8 18:47:26 2010
From: mail at jankubr.com (Jan Kubr)
Date: Wed, 8 Dec 2010 16:47:26 +0100
Subject: [Freeswitch-users] SIP Registration DNS Error
In-Reply-To: <49E1E293-BE37-43AA-960C-56B921FEBC9D@ipeva.fr>
References: <06502C073AD9394AADB3CA7FD94931BC0519DF46@okc1x1.Logixcom.com>
	<AANLkTi=10HbngBxzwzf=i5PpKMhG5tdzWvtcCdA_weJt@mail.gmail.com>
	<49E1E293-BE37-43AA-960C-56B921FEBC9D@ipeva.fr>
Message-ID: <AANLkTins8Z6c=dHp=fDta8u9WEmdahz+vqeD7vfvuxOa@mail.gmail.com>

I'm sure they have a DNS issue (although it's a different provider), but I
was more interested in why the retries fail even when everything is working
again on their side. And only restart helps.

JK

On Wed, Dec 8, 2010 at 4:05 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> I think I remember that Flowroute has a DNS issue.
> At least that was the case some months ago.
> Use an IP, that should fix it.
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 08/12/2010 ? 16:01, Jan Kubr a ?crit :
>
> I have the same problem with one gateway. What also helps is restarting the
> profile:
>
> sofia profile external restart reloadxml
>
> But that means dropping the calls as well.
>
> Anyone knows how to fix this?
>
> Thanks,
> Jan
>
> On Tue, Oct 12, 2010 at 7:27 PM, Joshua Foshee <Joshua.Foshee at logixcom.com
> > wrote:
>
>> I have setup up two sip providers that I can connect to fine but after a
>> while I then start to get these messages.
>>
>>
>> 2010-10-12 07:27:57.292742 [NOTICE] sofia_reg.c:342 Registering flowroute
>>
>> 2010-10-12 07:27:57.295138 [ERR] sofia_reg.c:1611 flowroute Registration
>> Failed with status DNS Error [503]. failure #1391
>>
>> 2010-10-12 07:27:59.959022 [WARNING] sofia_reg.c:387 flowroute Failed
>> Registration, setting retry to 30 seconds.
>>
>>
>> Here is the output of the gateway status
>>
>>
>> Name            flowroute
>>
>> Profile         external
>>
>> Scheme          Digest
>>
>> Realm           sip.flowroute.com
>>
>> Username        xxxxxxx
>>
>> Password        yes
>>
>> From            <sip:xxxxxxxxx at sip.flowroute.com<sip%3Axxxxxxxxx at sip.flowroute.com>
>> ;transport=udp>
>>
>> Contact         <
>> sip:gw+flowroute at xxxxxxxx:5080;transport=udp;gw=flowroute>
>>
>> Exten           xxxxxxx
>>
>> To              sip:xxxxxxxxx at sip.flowroute.com<sip%3Axxxxxxxxx at sip.flowroute.com>
>>
>> Proxy           sip:sip.flowroute.com
>>
>> Context         public
>>
>> Expires         600
>>
>> Freq            600
>>
>> Ping            1286835829
>>
>> PingFreq        25
>>
>> PingState       -1/0/1
>>
>> State           FAIL_WAIT
>>
>> Status          DOWN
>>
>> CallsIN         0
>>
>> CallsOUT        5
>>
>> FailedCallsIN   0
>>
>> FailedCallsOUT  5
>>
>>
>>
>> Name            broadvoice
>>
>> Profile         external
>>
>> Scheme          Digest
>>
>> Realm           BroadWorks
>>
>> Username        xxxxxxxx
>>
>> Password        yes
>>
>> From            <sip:xxxxxxxxxxxx at sip.broadvoice.com<sip%3Axxxxxxxxxxxx at sip.broadvoice.com>
>> ;transport=udp>
>>
>> Contact         <
>> sip:gw+broadvoice at xxxxxxxxxx:5080;transport=udp;gw=broadvoice>
>>
>> Exten           xxxxxxxx
>>
>> To              sip:xxxxxxxxxx at sip.broadvoice.com<sip%3Axxxxxxxxxx at sip.broadvoice.com>
>>
>> Proxy           sip:sip.broadvoice.com
>>
>> Context         public
>>
>> Expires         30
>>
>> Freq            30
>>
>> Ping            0
>>
>> PingFreq        0
>>
>> PingState       0/0/0
>>
>> State           FAIL_WAIT
>>
>> Status          DOWN
>>
>> CallsIN         0
>>
>> CallsOUT        2
>>
>> FailedCallsIN   0
>>
>> FailedCallsOUT  3
>>
>>
>> If I restart Freeswitch process they both come up and Reg just fine for a
>> while till it fails again.
>>
>>
>> Thanks in advance,
>>
>> Josh
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Dec  8 18:57:28 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 8 Dec 2010 07:57:28 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <AANLkTikuhfo_AvkWXCW1WQeH9VJw9pbG+B7GKkx=cfXW@mail.gmail.com>

Don't forget about today's call!

http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_08

Our agenda is light today so bring your questions and suggestions and we'll
discuss them as a group.

-MC
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From kris at kriskinc.com  Wed Dec  8 19:04:10 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 8 Dec 2010 11:04:10 -0500
Subject: [Freeswitch-users] SIP MESSAGE requests
In-Reply-To: <9B0E926C-3238-4388-B135-9F1C7899EEF8@avgs.ca>
References: <004201cb9662$d770dad0$86529070$@comtex.net>
	<4CFF3EC7.5030605@gmail.com>
	<AANLkTin3gep6cpQoV0b4NEV2Tqgxe6LGyVJE5ymu46NA@mail.gmail.com>
	<B6823144-18AC-42BA-BF9E-9027255A45D0@avgs.ca>
	<AANLkTikP1qHa4-PtMjseJHoDxvJ5Fo7a5mfWa2aTA-Kc@mail.gmail.com>
	<9B0E926C-3238-4388-B135-9F1C7899EEF8@avgs.ca>
Message-ID: <AANLkTin9wJP-C-6q1KRh87RVFBztJbP5pTM9f-hYra6H@mail.gmail.com>

Mathieu,

  This is true for MESSAGE and I understand it may be a little
hairy...  However, SIMPLE (uh-oh) also allows for IM-like message
exchange using a real session established with INVITE.  As the OP said
MESSAGE is used for paging (SMS) type applications.  I understand that
creating a channel may be a bit overblown for this but something like
that would allow for much more flexibility.

On Wed, Dec 8, 2010 at 10:34 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> The problem with that concept, is that dialplan access requires an active session, and it doesn't make sense to create one for something as short-lived as a message. What I could see happening is some sort of customizable event routing that could be configured ?to do some automated actions.
>
> Yup, indeed smells like a bounty.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From Nabble at slickdeals.endjunk.com  Wed Dec  8 19:18:11 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 8 Dec 2010 08:18:11 -0800 (PST)
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
In-Reply-To: <AANLkTikuhfo_AvkWXCW1WQeH9VJw9pbG+B7GKkx=cfXW@mail.gmail.com>
References: <AANLkTikuhfo_AvkWXCW1WQeH9VJw9pbG+B7GKkx=cfXW@mail.gmail.com>
Message-ID: <1291825091569-5815757.post@n2.nabble.com>



mercutioviz wrote:
> 
> Don't forget about today's call!
> 
> http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_08
> 
> Our agenda is light today so bring your questions and suggestions and
> we'll
> discuss them as a group.
I read the above link on the Who's in your wallet? section. I haven't done
this, but I believe a better approach to counteract this cracking activity
is to append alphanumeric characters to the extension and let the dialplan
on ATA and/or IP Phone devices to automatically append the alphanumeric
characters to the dialed number to reach the extension. This way, the
extensions can be called from internally while it will take the crackers
more efforts to figure out the extension numbers.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-Conference-Call-Today-tp5815693p5815757.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Wed Dec  8 19:30:34 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 8 Dec 2010 11:30:34 -0500
Subject: [Freeswitch-users] fax files not saved
References: <20101208153025.GB17099@pinboard.com>
Message-ID: <EC54D2EA08B14F8BA13A0E50C94B1826@e1705>

you have first a typo
fax__image_resolution

the thing I don't understand is you didn't
set session_in hangup_hook=true and
you can retrieve the channel vars at api_hangup_hook...



----- Original Message ----- 
From: <pbdlists at pinboard.com>
To: "freeswitch-users" <FreeSWITCH-users at lists.freeswitch.org>
Sent: Wednesday, December 08, 2010 10:30 AM
Subject: [Freeswitch-users] fax files not saved


> Hello!
>
> I managed to solve most of my other problems sofar, but with this one I
> have no clue at all. Has anybody an idea what is happening here?
>
> 1 an inbound call to a number which is registered as fax (mod_spandsp)
> 2 fax is detected and received according to the log (log entries see 
> below)
> 3 but _sometimes_ the tiff file is never written to disk (post processing
>   script can't find it and it is really not there)
>
> The destination filesystem has enough free space and free inodes,
> permissions for writing are ok, the system is not under any kind of
> heavy load, no other calls going on at the same time...
>
> I don't have much data to test this (using www.freepopfax.com for
> testing and they have a daily limit on faxes I can send), but it seems
> to happen roughly for 30-50% of the incoming fax messages. From the same
> fax provider it works fine one time, then not a couple of minutes later,
> another couple of minutes later it may work again, all while nothing at
> all is being changed on the freeswitch server. If it was something like
> a dropped line or anything I could understand it, but the logs say
> everything is ok and still _sometimes_ the fax is not written to disk...
>
> Cheers,
>
> Kurt
>
> =======================================================================
> entries in fax.conf.xml
> =======================================================================
> <configuration name="fax.conf" description="FAX application 
> configuration">
>    <settings>
>        <param name="use-ecm"           value="true"/>
>        <param name="verbose"           value="false"/>
>        <param name="disable-v17"       value="false"/>
>        <param name="ident"             value="SpanDSP Fax Ident"/>
>        <param name="header"            value="SpanDSP Fax Header"/>
>
>        <param name="spool-dir"         value="/tmp"/>
>        <param name="file-prefix"       value="faxrx"/>
>    </settings>
> </configuration>
>
>
> =======================================================================
> the part from default.xml
> =======================================================================
>    <extension name="fax_receive">
>        <condition field="destination_number" expression="^(1111)$">
>            <action application="set" data="api_hangup_hook=system 
> ${base_dir}/scripts/emailfax.sh $1 /tmp/FAX-${uuid}.tiff 
> '\\\${fax_success}' '\\\${fax_result_code}' '\\\${fax_result_text}' 
> '\\\${fax_local_station_id}' '\\\${fax_remote_station_id}' 
> '\\\${fax_document_transferred_pages}' '\\\${fax_document_total_pages}' 
> '\\\${fax__image_resolution}' '\\\${fax_image_size}' '\\\${fax_bad_rows}' 
> '\\\${fax_transfer_rate}' '\\\${fax_v17_disabled}' 
> '\\\${fax_ecm_requested}' '\\\${fax_filename}' "/>
>            <action application="answer" />
>            <action application="playback" data="silence_stream://2000"/>
> <!--
>            <action application="set" data="fax_enable_t38_request=true"/>
>            <action application="set" data="fax_enable_t38=true"/>
> -->
>            <action application="set" data="fax_ident=+41xxxxxxxxx"/>
>            <action application="set" data="fax_disable_v17=1"/>
>            <action application="set" data="fax_verbose=0"/>
>            <action application="rxfax" data="/tmp/FAX-${uuid}.tiff"/>
>            <action application="hangup"/>
>        </condition>
>    </extension>
>
>
> =======================================================================
> log entry from an unsuccessful incoming fax:
> =======================================================================
>
> 2010-12-07 23:33:51.157588 [DEBUG] sofia_glue.c:182 
> sofia/external/Anonymous at anonymous.invalid image media sdp:
> v=0
> o=FreeSWITCH 1291733462 1291733465 IN IP4 xxx.xxx.xxx.xxx
> s=FreeSWITCH
> c=IN IP4 xxx.xxx.xxx.xxx
> t=0 0
> m=image 31358 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:9600
> a=T38FaxFillBitRemoval
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:2000
> a=T38FaxMaxDatagram:400
> a=T38FaxUdpEC:t38UDPRedundancy
>
> 2010-12-07 23:33:51.157588 [DEBUG] sofia.c:4597 Channel 
> sofia/external/Anonymous at anonymous.invalid entering state [completed][200]
> 2010-12-07 23:33:51.157588 [DEBUG] sofia.c:4597 Channel 
> sofia/external/Anonymous at anonymous.invalid entering state [ready][200]
> 2010-12-07 23:33:58.814979 [DEBUG] switch_rtp.c:2544 Correct ip/port 
> confirmed.
> 2010-12-07 23:35:09.772752 [DEBUG] switch_channel.c:2457 
> (sofia/external/Anonymous at anonymous.invalid) Callstate Change ACTIVE -> 
> HANGUP
> 2010-12-07 23:35:09.772752 [NOTICE] sofia.c:528 Hangup 
> sofia/external/Anonymous at anonymous.invalid [CS_EXECUTE] [NORMAL_CLEARING]
> 2010-12-07 23:35:09.772752 [DEBUG] switch_channel.c:2473 Send signal 
> sofia/external/Anonymous at anonymous.invalid [KILL]
> 2010-12-07 23:35:09.772752 [DEBUG] switch_core_session.c:1057 Send signal 
> sofia/external/Anonymous at anonymous.invalid [BREAK]
> 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:323 
> ==============================================================================
> 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:329 Fax successfully 
> received.
> 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:340 Remote station 
> id:
> 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:341 Local station id: 
> +41xxxxxxxxx
> 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:342 Pages 
> transferred: 2
> 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:344 Total fax pages: 
> 2
> 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:345 Image resolution: 
> 8031x7700
> 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate: 
> 9600
> 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:348 ECM status 
> on
> 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:349 remote country:
> 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:350 remote vendor:
> 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:351 remote model:
> 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:353 
> ==============================================================================
> 2010-12-07 23:35:09.772752 [DEBUG] switch_core_codec.c:141 
> sofia/external/Anonymous at anonymous.invalid Restore previous codec PCMA:8.
> 2010-12-07 23:35:09.772752 [DEBUG] switch_core_session.c:1933 
> sofia/external/Anonymous at anonymous.invalid skip receive message 
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:366 
> (sofia/external/Anonymous at anonymous.invalid) State EXECUTE going to sleep
> 2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:320 
> (sofia/external/Anonymous at anonymous.invalid) Running State Change 
> CS_HANGUP
> 2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:553 
> (sofia/external/Anonymous at anonymous.invalid) State HANGUP
> 2010-12-07 23:35:09.772752 [DEBUG] mod_sofia.c:453 
> sofia/external/Anonymous at anonymous.invalid Overriding SIP cause 480 with 
> 200 from the other leg
> 2010-12-07 23:35:09.772752 [DEBUG] mod_sofia.c:459 Channel 
> sofia/external/Anonymous at anonymous.invalid hanging up, cause: 
> NORMAL_CLEARING
> 2010-12-07 23:35:09.821455 [DEBUG] switch_core_state_machine.c:46 
> sofia/external/Anonymous at anonymous.invalid Standard HANGUP, cause: 
> NORMAL_CLEARING
> 2010-12-07 23:35:09.821455 [DEBUG] switch_core_state_machine.c:553 
> (sofia/external/Anonymous at anonymous.invalid) State HANGUP going to sleep
> 2010-12-07 23:35:09.822803 [NOTICE] mod_commands.c:4187 Executing command: 
> /usr/local/freeswitch/scripts/emailfax.sh 1111 
> /tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff '1' '0' 'OK' 
> '+41xxxxxxxxx' '' '2' '2
> ' '' '0' '0' '9600' '1' '1' 
> '/tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff'
> 2010-12-07 23:35:10.076623 [DEBUG] switch_core_state_machine.c:488 Hangup 
> Command with no Session system(/usr/local/freeswitch/scripts/emailfax.sh 
> 1111 /tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff '1' '0'
> 'OK' '+41xxxxxxxxx' '' '2' '2' '' '0' '0' '9600' '1' '1' 
> '/tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff' ):
> +OK
>
>
> =======================================================================
> log entry from a successful incoming fax:
> =======================================================================
>
> 2010-12-07 23:54:59.082301 [DEBUG] sofia_glue.c:182 
> sofia/external/Anonymous at anonymous.invalid image media sdp:
> v=0
> o=FreeSWITCH 1291735801 1291735804 IN IP4 xxx.xxx.xxx.xxx
> s=FreeSWITCH
> c=IN IP4 xxx.xxx.xxx.xxx
> t=0 0
> m=image 30280 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:9600
> a=T38FaxFillBitRemoval
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxBuffer:2000
> a=T38FaxMaxDatagram:400
> a=T38FaxUdpEC:t38UDPRedundancy
>
> 2010-12-07 23:54:59.082301 [DEBUG] sofia.c:4597 Channel 
> sofia/external/Anonymous at anonymous.invalid entering state [completed][200]
> 2010-12-07 23:54:59.100811 [DEBUG] sofia.c:4597 Channel 
> sofia/external/Anonymous at anonymous.invalid entering state [ready][200]
> 2010-12-07 23:55:06.783427 [DEBUG] switch_rtp.c:2544 Correct ip/port 
> confirmed.
> 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:323 
> ==============================================================================
> 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:329 Fax successfully 
> received.
> 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:340 Remote station 
> id:
> 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:341 Local station id: 
> +41xxxxxxxxx
> 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:342 Pages 
> transferred: 1
> 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:344 Total fax pages: 
> 1
> 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:345 Image resolution: 
> 8031x7700
> 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate: 
> 9600
> 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:348 ECM status 
> on
> 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:349 remote country:
> 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:350 remote vendor:
> 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:351 remote model:
> 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:353 
> ==============================================================================
> 2010-12-07 23:55:31.616038 [DEBUG] switch_channel.c:2457 
> (sofia/external/Anonymous at anonymous.invalid) Callstate Change ACTIVE -> 
> HANGUP
> 2010-12-07 23:55:31.616038 [NOTICE] sofia.c:528 Hangup 
> sofia/external/Anonymous at anonymous.invalid [CS_EXECUTE] [NORMAL_CLEARING]
> 2010-12-07 23:55:31.616038 [DEBUG] switch_channel.c:2473 Send signal 
> sofia/external/Anonymous at anonymous.invalid [KILL]
> 2010-12-07 23:55:31.616038 [DEBUG] switch_core_session.c:1057 Send signal 
> sofia/external/Anonymous at anonymous.invalid [BREAK]
> 2010-12-07 23:55:31.617356 [DEBUG] switch_core_codec.c:141 
> sofia/external/Anonymous at anonymous.invalid Restore previous codec PCMA:8.
> 2010-12-07 23:55:31.617356 [DEBUG] switch_core_session.c:1933 
> sofia/external/Anonymous at anonymous.invalid skip receive message 
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2010-12-07 23:55:31.617356 [DEBUG] switch_core_state_machine.c:366 
> (sofia/external/Anonymous at anonymous.invalid) State EXECUTE going to sleep
> 2010-12-07 23:55:31.617356 [DEBUG] switch_core_state_machine.c:320 
> (sofia/external/Anonymous at anonymous.invalid) Running State Change 
> CS_HANGUP
> 2010-12-07 23:55:31.618542 [DEBUG] switch_core_state_machine.c:553 
> (sofia/external/Anonymous at anonymous.invalid) State HANGUP
> 2010-12-07 23:55:31.618542 [DEBUG] mod_sofia.c:453 
> sofia/external/Anonymous at anonymous.invalid Overriding SIP cause 480 with 
> 200 from the other leg
> 2010-12-07 23:55:31.618542 [DEBUG] mod_sofia.c:459 Channel 
> sofia/external/Anonymous at anonymous.invalid hanging up, cause: 
> NORMAL_CLEARING
> 2010-12-07 23:55:31.721993 [DEBUG] switch_core_state_machine.c:46 
> sofia/external/Anonymous at anonymous.invalid Standard HANGUP, cause: 
> NORMAL_CLEARING
> 2010-12-07 23:55:31.721993 [DEBUG] switch_core_state_machine.c:553 
> (sofia/external/Anonymous at anonymous.invalid) State HANGUP going to sleep
> 2010-12-07 23:55:31.721993 [NOTICE] mod_commands.c:4187 Executing command: 
> /usr/local/freeswitch/scripts/emailfax.sh 1111 
> /tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff '1' '0' 'OK' 
> '+41xxxxxxxxx' '' '1' '1' '' '0' '0' '9600' '1' '1' 
> '/tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff'
> 2010-12-07 23:55:31.904090 [DEBUG] switch_core_state_machine.c:488 Hangup 
> Command with no Session system(/usr/local/freeswitch/scripts/emailfax.sh 
> 1111 /tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff '1' '0' 'OK' 
> '+41xxxxxxxxx' '' '1' '1' '' '0' '0' '9600' '1' '1' 
> '/tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff' ):
> +OK
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From melkybes at mail.ru  Wed Dec  8 20:02:15 2010
From: melkybes at mail.ru (=?koi8-r?Q?=ED=C9=C8=C1=C9=CC_=F3=C1=CC=D4=C1=CE=CF=D7?=)
Date: Wed, 08 Dec 2010 20:02:15 +0300
Subject: [Freeswitch-users] =?koi8-r?b?Y2hhbmdlINNhbGxlcl9pZF9uYW1lICYg?=
	=?koi8-r?b?bnVtYmVy?=
Message-ID: <E1PQNPD-0002tm-00.melkybes-mail-ru@f160.mail.ru>

Hello.

Can I change Caller_id_name &amp; caller_id_number in freeswitch during transfer?




? ?????????
?????? ????????



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From tayeb.meftah at gmail.com  Wed Dec  8 20:15:15 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Wed, 08 Dec 2010 18:15:15 +0100
Subject: [Freeswitch-users] =?koi8-r?b?Y2hhbmdlINNhbGxlcl9pZF9uYW1lICYg?=
 =?koi8-r?b?bnVtYmVy?=
In-Reply-To: <E1PQNPD-0002tm-00.melkybes-mail-ru@f160.mail.ru>
References: <E1PQNPD-0002tm-00.melkybes-mail-ru@f160.mail.ru>
Message-ID: <4CFFBD23.6010608@gmail.com>

<action application="set" data="effective_caller_id_name=russia"/>
<action application="set" data="effective_caller_id_number=+777777777777"/>
Le 08/12/2010 18:02, ?????? ???????? a e'crit :
> Hello.
>
> Can I change Caller_id_name & caller_id_number in freeswitch during 
> transfer?
>
>
>
>
> ? ?????????
> ?????? ????????
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313

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From davidjbrazier at gmail.com  Wed Dec  8 20:49:58 2010
From: davidjbrazier at gmail.com (David Brazier)
Date: Wed, 8 Dec 2010 17:49:58 +0000
Subject: [Freeswitch-users] mod_managed.so Loading error:
 mono_class_from_name undefind symbol
In-Reply-To: <AANLkTik0_ZMPpvscmQzeG7VSLWfXJE-BOM3fVu2gP3Wx@mail.gmail.com>
References: <AANLkTik0_ZMPpvscmQzeG7VSLWfXJE-BOM3fVu2gP3Wx@mail.gmail.com>
Message-ID: <AANLkTin5DG0aWycqPFTH67e2wap8B17gGn7xbCvgzE13@mail.gmail.com>

I suggest you get a fresh copy of FS sources and follow the instructions at

http://wiki.freeswitch.org/wiki/Mod_managed

On Wed, Dec 8, 2010 at 1:33 PM, srinivasula reddy
<srinivas.ksvreddy at gmail.com> wrote:
> HI All,
>
> When we trying to load mod_managed.so in CentOS, We are facing a issue,
>
> mono_class_from_name undefind symbol.
>
> ?mono2.8.1 we are using for this.
>
> Prevously we got mono_thread_attach error in runtime, we commented the
> mono_thread_attach in mod_managed.cpp then we are facing the above error.
>
> Any Idea?
>
> Thanks--
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From bwibowo at gmail.com  Thu Dec  9 01:44:45 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Thu, 9 Dec 2010 05:44:45 +0700
Subject: [Freeswitch-users] sip-t
Message-ID: <AANLkTikRctwUyPCrGOV0=r=K0L+Z=p4g0Pnz4eU9ADHX@mail.gmail.com>

hi
does freeswitch support sip-t protocol?


regards

budi
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From kris at kriskinc.com  Thu Dec  9 01:48:52 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 8 Dec 2010 17:48:52 -0500
Subject: [Freeswitch-users] sip-t
In-Reply-To: <AANLkTikRctwUyPCrGOV0=r=K0L+Z=p4g0Pnz4eU9ADHX@mail.gmail.com>
References: <AANLkTikRctwUyPCrGOV0=r=K0L+Z=p4g0Pnz4eU9ADHX@mail.gmail.com>
Message-ID: <AANLkTim+RdUGiHuwXr1mbf6pLi6ZWwYNqMrot2UOkAy2@mail.gmail.com>

FreeSWITCH will not decode the encapsulated ISUP but it will give you
control over the multipart message body:

http://wiki.freeswitch.org/wiki/Variable_sip_copy_multipart

On Wed, Dec 8, 2010 at 5:44 PM, budi wibowo <bwibowo at gmail.com> wrote:
> hi
> does freeswitch support sip-t protocol?
>
> regards
> budi
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From bwibowo at gmail.com  Thu Dec  9 04:35:02 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Thu, 9 Dec 2010 08:35:02 +0700
Subject: [Freeswitch-users] sip-t
In-Reply-To: <AANLkTim+RdUGiHuwXr1mbf6pLi6ZWwYNqMrot2UOkAy2@mail.gmail.com>
References: <AANLkTikRctwUyPCrGOV0=r=K0L+Z=p4g0Pnz4eU9ADHX@mail.gmail.com>
	<AANLkTim+RdUGiHuwXr1mbf6pLi6ZWwYNqMrot2UOkAy2@mail.gmail.com>
Message-ID: <AANLkTikqPCrhEeOyXa_6JUpa95J1OwZ1skY=CA9Y+DE9@mail.gmail.com>

based on experience connecting to sip-t device is possible, right?


On Thu, Dec 9, 2010 at 5:48 AM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> FreeSWITCH will not decode the encapsulated ISUP but it will give you
> control over the multipart message body:
>
> http://wiki.freeswitch.org/wiki/Variable_sip_copy_multipart
>
> On Wed, Dec 8, 2010 at 5:44 PM, budi wibowo <bwibowo at gmail.com> wrote:
> > hi
> > does freeswitch support sip-t protocol?
> >
> > regards
> > budi
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bwibowo at gmail.com  Thu Dec  9 04:41:26 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Thu, 9 Dec 2010 08:41:26 +0700
Subject: [Freeswitch-users] nibblebill multi tenant
Message-ID: <AANLkTin5TrwmpPRK-LNjOrAMdrcANJBFMo50-05c8VFF@mail.gmail.com>

i just finished configuring freeswitch to support multi tenant. how do i
implement mod_nibblebill to support multi tenant environment?
if i check the nibblebill database structure for accounts table, field id
has type int(11).
in my opinion for multi tenant environment, id must be varchar


regards

budi
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From william.suffill at gmail.com  Thu Dec  9 08:13:12 2010
From: william.suffill at gmail.com (William Suffill)
Date: Thu, 9 Dec 2010 00:13:12 -0500
Subject: [Freeswitch-users] nibblebill multi tenant
In-Reply-To: <AANLkTin5TrwmpPRK-LNjOrAMdrcANJBFMo50-05c8VFF@mail.gmail.com>
References: <AANLkTin5TrwmpPRK-LNjOrAMdrcANJBFMo50-05c8VFF@mail.gmail.com>
Message-ID: <AANLkTi=BYeiqmoF=n_aJ8ZQ8fDd_i=EvaFsnNQarmsSx@mail.gmail.com>

You can handle this many ways depending how  you wish to design your system.
Below is a simple concept based on the existing tables to give you an idea.
There are other ways to do it just what I came up with quickly.
If you didn't want to have to touch any of the core of nibblebill you could
add additional fields to accomplish your goal.
You could add a field such as tenantid type int(11).

Company A = tenantid 1
Company B = tenantid 2
....

id | tenantid | name | cash
1 | 1               | Bob   | 5.00
2 | 1               | Joe    | 6.00
3| 2                | George | 7.00



As far as varchar vs int  perhaps from a user conception varchar might be
more straight forward in a single table but should you start doing queries
across multiple tables to avoid data duplication numerical identification
fields are common.

Like my rough example above I'd need to keep another table store the details
on the tenants

Hope that helps get you started.

-- W
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From daniel-listas at gmx.net  Thu Dec  9 01:37:56 2010
From: daniel-listas at gmx.net (Daniel Bareiro)
Date: Wed, 8 Dec 2010 19:37:56 -0300
Subject: [Freeswitch-users] No sound on incomming calls
Message-ID: <slrnig028h.26s.daniel-listas@defiant.freesoftware>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi all!

I'm new in the list and I'm starting to test FreeSWITCH at home to learn
how to use it thinking as a alternative to Asterisk.

I've compiled FreeSwitch 1.0.6 and I have been doing some testing with
Iptel.org and FreeSWICH. Outgoing calls to Iptel.org accounts ran
without problems, however I am having trouble with incoming calls.

If, for example, I call from a softphone registered on Iptel.org with a
second account to the Iptel.org account that was registered from
FreeSWITCH, it answers the call but no sound.

The FreeSWITCH server is behind a firewall, but I do not think it's a
NAT problem because then there would be no sound for outgoing calls made
with the softphone registered on Iptel.org.

Could it be a codecs problem?

When FreeSWITCH receives the call, it is redirected to a Grandstream
BT200 phone. To answer the call in the BT200, the softphone indicates
that G711a is being used, but no sound.



Thanks in advance for your replies.

Regards,
Daniel

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From brian at freeswitch.org  Thu Dec  9 09:36:56 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 9 Dec 2010 00:36:56 -0600
Subject: [Freeswitch-users] No sound on incomming calls
In-Reply-To: <slrnig028h.26s.daniel-listas@defiant.freesoftware>
References: <slrnig028h.26s.daniel-listas@defiant.freesoftware>
Message-ID: <3501BDFE-1662-4BBB-995F-DB0D1902EA7E@freeswitch.org>

Please try SVN Git and stop your firewall or fix your firewall rules?

/b

On Dec 8, 2010, at 4:37 PM, Daniel Bareiro wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> Hi all!
> 
> I'm new in the list and I'm starting to test FreeSWITCH at home to learn
> how to use it thinking as a alternative to Asterisk.
> 
> I've compiled FreeSwitch 1.0.6 and I have been doing some testing with
> Iptel.org and FreeSWICH. Outgoing calls to Iptel.org accounts ran
> without problems, however I am having trouble with incoming calls.
> 
> If, for example, I call from a softphone registered on Iptel.org with a
> second account to the Iptel.org account that was registered from
> FreeSWITCH, it answers the call but no sound.
> 
> The FreeSWITCH server is behind a firewall, but I do not think it's a
> NAT problem because then there would be no sound for outgoing calls made
> with the softphone registered on Iptel.org.
> 
> Could it be a codecs problem?
> 
> When FreeSWITCH receives the call, it is redirected to a Grandstream
> BT200 phone. To answer the call in the BT200, the softphone indicates
> that G711a is being used, but no sound.
> 
> 
> 
> Thanks in advance for your replies.
> 
> Regards,
> Daniel
> 
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (GNU/Linux)
> 
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> aSgAn1rg4mjSqj8yzIGE77NDO5oOXKPq
> =+GKJ
> -----END PGP SIGNATURE-----
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From raison at chatsubo.net  Thu Dec  9 10:13:52 2010
From: raison at chatsubo.net (Kevin Raison)
Date: Wed, 08 Dec 2010 23:13:52 -0800
Subject: [Freeswitch-users] google talk issues
Message-ID: <4D0081B0.5070409@chatsubo.net>

I have followed the tutorials for setting up google talk with 
freeswitch, and yet I am unable to make or recieve calls. Here is a 
transcript from the logs for a failed outbound call:

EXECUTE sofia/internal/kraison at 192.168.0.2 
bridge(dingaling/xmppc/+1XXXXXXXXXX at voice.google.com)
2010-12-08 22:39:54.694557 [NOTICE] libdingaling.c:1307 SecSEND:
-------------------------------------------------------------------------------
<presence type="subscribe" to="+1XXXXXXXXXX at voice.google.com" 
from="kraison at gmail.com/talkECFD45DE">
   <status>Call Me!</status>
</presence>

2010-12-08 22:39:54.694557 [NOTICE] libdingaling.c:1307 SecSEND:
-------------------------------------------------------------------------------
<presence xmlns="jabber:client" type="probe" 
to="+1XXXXXXXXXX at voice.google.com" 
from="kraison at gmail.com/talkECFD45DE"></presence>

2010-12-08 22:39:54.938862 [INFO] libdingaling.c:1305 SecRECV:
-------------------------------------------------------------------------------
<presence type="error" to="kraison at gmail.com/talkECFD45DE" 
from="+1XXXXXXXXXX at voice.google.com">
   <error code="501" type="cancel">
     <feature-not-implemented 
xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></feature-not-implemented>
   </error>
</presence>
2010-12-08 22:39:54.999689 [DEBUG] mod_dingaling.c:1715 Unknown Recipient!


My current dingaling config looks like this:

<include>
   <profile type="client">
     <param name="name" value="$${xmpp_client_profile}"/>
     <param name="login" value="kraison at gmail.com/gtalk"/>
     <param name="password" value="XXXXXXXXXXXXXXXX"/>
     <param name="dialplan" value="XML"/>
     <param name="context" value="public"/>
     <param name="message" value="FreeSWITCH!"/>
     <param name="rtp-ip" value="auto"/>
     <param name="ext-rtp-ip" value="auto-nat"/>
     <param name="auto-login" value="true"/>
     <param name="sasl" value="plain"/>
     <param name="server" value="talk.google.com"/>
     <param name="tls" value="true"/>
     <param name="use-rtp-timer" value="true"/>
     <param name="exten" value="8908"/>
     <param name="vad" value="both"/>
     <param name="candidate-acl" value="wan.auto"/>
     <param name="local-network-acl" value="localnet.auto"/>
   </profile>
</include>

Any suggestions would be appreciated!

-Kevin


From david.ponzone at ipeva.fr  Thu Dec  9 10:37:36 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 9 Dec 2010 08:37:36 +0100
Subject: [Freeswitch-users] No sound on incomming calls
In-Reply-To: <slrnig028h.26s.daniel-listas@defiant.freesoftware>
References: <slrnig028h.26s.daniel-listas@defiant.freesoftware>
Message-ID: <265A287B-E9F4-4256-A80C-4F18578911A0@ipeva.fr>

Daniel,

if you expect people around to help you, you may try to pinpoint what works and what does not, like:

-is the softphone behind the same firewall ?
-for incoming call, is the sound missing both ways or only one way ?
-if you call from the BT200 to the softphone, is the sound ok ?
-if you register the 2nd account also with a softphone (not FS anymore), is an incoming call from the 1st softphone ok ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 08/12/2010 ? 23:37, Daniel Bareiro a ?crit :

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> Hi all!
> 
> I'm new in the list and I'm starting to test FreeSWITCH at home to learn
> how to use it thinking as a alternative to Asterisk.
> 
> I've compiled FreeSwitch 1.0.6 and I have been doing some testing with
> Iptel.org and FreeSWICH. Outgoing calls to Iptel.org accounts ran
> without problems, however I am having trouble with incoming calls.
> 
> If, for example, I call from a softphone registered on Iptel.org with a
> second account to the Iptel.org account that was registered from
> FreeSWITCH, it answers the call but no sound.
> 
> The FreeSWITCH server is behind a firewall, but I do not think it's a
> NAT problem because then there would be no sound for outgoing calls made
> with the softphone registered on Iptel.org.
> 
> Could it be a codecs problem?
> 
> When FreeSWITCH receives the call, it is redirected to a Grandstream
> BT200 phone. To answer the call in the BT200, the softphone indicates
> that G711a is being used, but no sound.
> 
> 
> 
> Thanks in advance for your replies.
> 
> Regards,
> Daniel
> 
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (GNU/Linux)
> 
> iEYEARECAAYFAk0ACREACgkQZpa/GxTmHTd9GQCcCoM4TzkSfZyVPthU7v9hzqw4
> aSgAn1rg4mjSqj8yzIGE77NDO5oOXKPq
> =+GKJ
> -----END PGP SIGNATURE-----
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From haloha201 at yahoo.com  Thu Dec  9 12:14:51 2010
From: haloha201 at yahoo.com (ha do)
Date: Thu, 9 Dec 2010 01:14:51 -0800 (PST)
Subject: [Freeswitch-users] get error on skypopen module
In-Reply-To: <AANLkTinRG+qV-O_NJ2NnAn2f7xKo-3dm1m+Z+EjKLJVV@mail.gmail.com>
Message-ID: <663837.94530.qm@web32405.mail.mud.yahoo.com>

Hi Steven

i just found some interesting thing in the freeswitch wiki "mod_xml_rpc"
and use the small of code to check the skypopen interface IDLE or not IDLE

import xmlrpclib 
from time import sleep
from os import system

server = xmlrpclib.Server('http://freeswitch:works at localhost:8080')
while True:
        print(server.freeswitch.api('sk', 'list'))
        sleep(5)
        system('clear')

it works fine

so there are a lot of ways to work with freeswitch and that makes me confuse

could you please tell me what difference between them:
    * mod_event_socket
    * mod_xmpp_event
    * mod_erlang_event
    * mod_xml_rpc 

which one is the best choice

Thank you
Ha`


--- On Tue, 12/7/10, Steven Ayre <steveayre at gmail.com> wrote:

> From: Steven Ayre <steveayre at gmail.com>
> Subject: Re: [Freeswitch-users] get error on skypopen module
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Tuesday, December 7, 2010, 12:42 PM
> > the skypopen module generate the
> events which are same as other modules
> 
> That's because they're channel events generated by the
> freeswitch
> core. But there may be variables in them that you can use
> to identify
> the Skypopen interface used.
> 
> > so i have to wait the 'sk list' command which maybe
> support to event socket in future :D
> 
> It's an API command so it's supported right now - just do
> "api sk
> list". You just won't get the response as XML.
> 
> -Steve
> 
> 
> 
> On 7 December 2010 15:00, ha do <haloha201 at yahoo.com>
> wrote:
> > Hi Steven
> >
> > the skypopen module generate the events which are same
> as other modules
> > so i have to wait the 'sk list' command which maybe
> support to event socket in future :D
> >
> >
> > Thank you
> > Ha`
> > --- On Tue, 12/7/10, Steven Ayre <steveayre at gmail.com>
> wrote:
> >
> >> From: Steven Ayre <steveayre at gmail.com>
> >> Subject: Re: [Freeswitch-users] get error on
> skypopen module
> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> >> Date: Tuesday, December 7, 2010, 6:48 AM
> >> > i have a question on event
> >> socket, what is the event name for skypopen to
> monitor the
> >> interface IDLE/not IDLE or ANSWER or IN
> PROGRESS...
> >>
> >> Full ESL event list: http://wiki.freeswitch.org/wiki/Event_list
> >>
> >> Skyopen doesn't generate events for state changes
> itself.
> >>
> >> Check for CHANNEL_ANSWER and
> >> CHANNEL_PROGRESS/CHANNEL_PROGRESS_MEDIA.
> >>
> >> For spotting an idle skype interface,
> CHANNEL_CREATE and
> >> CHANNEL_DESTROY might tell you that by telling you
> when the
> >> interface
> >> is in use.
> >>
> >> If you want to poll for the interface states you
> can use
> >> the sk list
> >> api function via ESL, although there's
> unfortunately no 'as
> >> xml'
> >> formatting at the moment so it'll be slightly
> trickier to
> >> parse.
> >>
> >> -Steve
> >>
> >> On 7 December 2010 12:15, ha do <haloha201 at yahoo.com>
> >> wrote:
> >> > Hi Meftab
> >> >
> >> >> did you autorised skypopen to access
> >> >> skype from the skype client?
> >> > you mean the skypopen_auth ??? if so, the
> answer is
> >> yes
> >> >
> >> >> did you configured the skypopen
> interfaces to each
> >> one
> >> >> of the skype
> >> > i dont understand, because i use multi skypy
> >> client(skype software) with 1 username and 1
> password
> >> > and use the sample config file in
> >>
> $source.../../configs/multiple-instances-same-skype-user/
> >> >
> >> > 2 skype clients run in difference folder
> >> > /home/cucku/multi/interfaces01
> >> > /home/cucku/multi/interfaces02
> >> >
> >> > so i only need to config the
> skypopen.conf.xml look
> >> like below:
> >> > <configuration name="skypopen.conf"
> >> description="Skypopen Configuration">
> >> > ?<global_settings>
> >> > ? ?<param name="debug" value="8"/>
> >> > ? ?<param name="dialplan"
> value="XML"/>
> >> > ? ?<param name="context"
> value="default"/>
> >> > ? ?<param name="destination"
> value="5000"/>
> >> > ? ?<param name="skype_user"
> >> value="do_nguyen_ha"/>
> >> > ? ?<param
> name="report_incoming_chatmessages"
> >> value="false"/>
> >> > ? ?<param name="silent_mode"
> value="false"/>
> >> > ? ?<param
> name="write_silence_when_idle"
> >> value="true"/>
> >> > ? ?<param name="setsockopt"
> value="false"/>
> >> > ?</global_settings>
> >> > ?<!-- one entry here per each skypopen
> interface
> >> -->
> >> > ?<per_interface_settings>
> >> > ? ?<interface id="1"
> name="interface1">
> >> > ? ? ? ?<param name="X11-display"
> >> value=":101"/>
> >> > ? ?</interface>
> >> > ? ?<interface id="2"
> name="interface2">
> >> > ? ? ? ?<param name="X11-display"
> >> value=":102"/>
> >> > ? ?</interface>
> >> > ?</per_interface_settings>
> >> > </configuration>
> >> >
> >> > does the conifg look ok?? if not, please
> guide me to
> >> make it right
> >> >
> >> > i have a question on event socket, what is
> the event
> >> name for skypopen to monitor the interface
> IDLE/not IDLE or
> >> ANSWER or IN PROGRESS...
> >> >
> >> > which event plain should i take care of
> >> >
> >> >
> >> > Thank you
> >> > Ha`
> >> > --- On Tue, 12/7/10, Meftah Tayeb <tayeb.meftah at gmail.com>
> >> wrote:
> >> >
> >> >> From: Meftah Tayeb <tayeb.meftah at gmail.com>
> >> >> Subject: Re: [Freeswitch-users] get error
> on
> >> skypopen module
> >> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> >> >> Cc: "ha do" <haloha201 at yahoo.com>
> >> >> Date: Tuesday, December 7, 2010, 4:43 AM
> >> >> did you autorised skypopen to access
> >> >> skype from the skype client?
> >> >> and did you configured the skypopen
> interfaces to
> >> each one
> >> >> of the skype
> >> >> clients?
> >> >> thanks
> >> >> Le 07/12/2010 05:34, ha do a ?crit :
> >> >> > Hi list
> >> >> >
> >> >> > i setup freeswitch and skypopen
> running fine
> >> >> >
> >> >> > there are 2 skype clients run on
> freeswitch
> >> >> > freeswitch at internal>? sk list
> >> >> > sk console is NOT yet assigned
> >> >> > F ID? ? ? ? Name
> >> >> ? ? IB (F/T)? ? OB (F/T)
> >> >> State???CallFlw
> >> >> ???UUID
> >> >> > = ====? ? ========
> >> >> =======? ???=======
> >> >> ???======? ============
> >> >> ======
> >> >> >? ? 1
> >> >> ???[interface1]? ? ? 0/1
> >> >> ? ? ? 3/7
> >> >> IDLE? ? IDLE
> >> >> >? ? 2
> >> >> ???[interface2]? ? ? 0/5
> >> >> ? ? ? 1/5
> >> >> IDLE? ? IDLE
> >> >> >
> >> >> > the skype clients are used the same
> username
> >> +
> >> >> password of skype account
> >> >> >
> >> >> >
> >> >> > the skypopen works fine but i get
> error below
> >> in the
> >> >> debug mode
> >> >> >
> >> >> > 2010-12-07 04:24:35.794611 [DEBUG]
> >> >> switch_core_state_machine.c:462
> >> (skypopen/interface1) State
> >> >> DESTROY going to sleep
> >> >> > 2010-12-07 04:24:35.795807 [DEBUG]
> >> >> skypopen_protocol.c:173? ???[|]
> >> >> [DEBUG_SKYPE? 173? ][interface1
> >> >> ???][IDLE,IDLE] READING: |||ERROR 559
> CALL:
> >> >> Action failed|||
> >> >> > 2010-12-07 04:24:35.795807 [DEBUG]
> >> >> skypopen_protocol.c:228? ???[|]
> >> >> [DEBUG_SKYPE? 228? ][interface1
> >> >> ???][IDLE,IDLE] Skype got ERROR about
> a
> >> >> failed action (probably TRYING to HANGUP
> A CALL),
> >> no
> >> >> problem: |||ERROR 559 CALL: Action
> failed|||
> >> >> > 2010-12-07 04:24:35.795807 [DEBUG]
> >> >> skypopen_protocol.c:173? ???[|]
> >> >> [DEBUG_SKYPE? 173? ][interface1
> >> >> ???][IDLE,IDLE] READING: |||ERROR 559
> CALL:
> >> >> Action failed|||
> >> >> > 2010-12-07 04:24:35.795807 [DEBUG]
> >> >> skypopen_protocol.c:228? ???[|]
> >> >> [DEBUG_SKYPE? 228? ][interface1
> >> >> ???][IDLE,IDLE] Skype got ERROR about
> a
> >> >> failed action (probably TRYING to HANGUP
> A CALL),
> >> no
> >> >> problem: |||ERROR 559 CALL: Action
> failed|||
> >> >> >
> >> >> > How to fix it
> >> >> >
> >> >> > Thank you
> >> >> > Ha`
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >>
> >> >>
> >> >> --
> >> >> Meftah Tayeb
> >> >> inum: +883510001288000
> >> >> Phone: +13602276297
> >> >> Fax: +12538020313
> >> >>
> >> >>
> >> >
> >> >
> >> >
> >> >
> >> >
> _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From steveayre at gmail.com  Thu Dec  9 13:26:35 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 9 Dec 2010 10:26:35 +0000
Subject: [Freeswitch-users] get error on skypopen module
In-Reply-To: <663837.94530.qm@web32405.mail.mud.yahoo.com>
References: <663837.94530.qm@web32405.mail.mud.yahoo.com>
Message-ID: <ECA62A12-3E59-4853-B24B-E4708F99C62D@gmail.com>

They're all different ways to connect to the api/events. All apis and events are available through each interface.

Use whichever you prefer/best suits your application. Personally I use ESL (mod_event_socket).

Steve on iPhone

On 9 Dec 2010, at 09:14, ha do <haloha201 at yahoo.com> wrote:

> Hi Steven
> 
> i just found some interesting thing in the freeswitch wiki "mod_xml_rpc"
> and use the small of code to check the skypopen interface IDLE or not IDLE
> 
> import xmlrpclib 
> from time import sleep
> from os import system
> 
> server = xmlrpclib.Server('http://freeswitch:works at localhost:8080')
> while True:
>        print(server.freeswitch.api('sk', 'list'))
>        sleep(5)
>        system('clear')
> 
> it works fine
> 
> so there are a lot of ways to work with freeswitch and that makes me confuse
> 
> could you please tell me what difference between them:
>    * mod_event_socket
>    * mod_xmpp_event
>    * mod_erlang_event
>    * mod_xml_rpc 
> 
> which one is the best choice
> 
> Thank you
> Ha`
> 
> 
> --- On Tue, 12/7/10, Steven Ayre <steveayre at gmail.com> wrote:
> 
>> From: Steven Ayre <steveayre at gmail.com>
>> Subject: Re: [Freeswitch-users] get error on skypopen module
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Date: Tuesday, December 7, 2010, 12:42 PM
>>> the skypopen module generate the
>> events which are same as other modules
>> 
>> That's because they're channel events generated by the
>> freeswitch
>> core. But there may be variables in them that you can use
>> to identify
>> the Skypopen interface used.
>> 
>>> so i have to wait the 'sk list' command which maybe
>> support to event socket in future :D
>> 
>> It's an API command so it's supported right now - just do
>> "api sk
>> list". You just won't get the response as XML.
>> 
>> -Steve
>> 
>> 
>> 
>> On 7 December 2010 15:00, ha do <haloha201 at yahoo.com>
>> wrote:
>>> Hi Steven
>>> 
>>> the skypopen module generate the events which are same
>> as other modules
>>> so i have to wait the 'sk list' command which maybe
>> support to event socket in future :D
>>> 
>>> 
>>> Thank you
>>> Ha`
>>> --- On Tue, 12/7/10, Steven Ayre <steveayre at gmail.com>
>> wrote:
>>> 
>>>> From: Steven Ayre <steveayre at gmail.com>
>>>> Subject: Re: [Freeswitch-users] get error on
>> skypopen module
>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>> Date: Tuesday, December 7, 2010, 6:48 AM
>>>>> i have a question on event
>>>> socket, what is the event name for skypopen to
>> monitor the
>>>> interface IDLE/not IDLE or ANSWER or IN
>> PROGRESS...
>>>> 
>>>> Full ESL event list: http://wiki.freeswitch.org/wiki/Event_list
>>>> 
>>>> Skyopen doesn't generate events for state changes
>> itself.
>>>> 
>>>> Check for CHANNEL_ANSWER and
>>>> CHANNEL_PROGRESS/CHANNEL_PROGRESS_MEDIA.
>>>> 
>>>> For spotting an idle skype interface,
>> CHANNEL_CREATE and
>>>> CHANNEL_DESTROY might tell you that by telling you
>> when the
>>>> interface
>>>> is in use.
>>>> 
>>>> If you want to poll for the interface states you
>> can use
>>>> the sk list
>>>> api function via ESL, although there's
>> unfortunately no 'as
>>>> xml'
>>>> formatting at the moment so it'll be slightly
>> trickier to
>>>> parse.
>>>> 
>>>> -Steve
>>>> 
>>>> On 7 December 2010 12:15, ha do <haloha201 at yahoo.com>
>>>> wrote:
>>>>> Hi Meftab
>>>>> 
>>>>>> did you autorised skypopen to access
>>>>>> skype from the skype client?
>>>>> you mean the skypopen_auth ??? if so, the
>> answer is
>>>> yes
>>>>> 
>>>>>> did you configured the skypopen
>> interfaces to each
>>>> one
>>>>>> of the skype
>>>>> i dont understand, because i use multi skypy
>>>> client(skype software) with 1 username and 1
>> password
>>>>> and use the sample config file in
>>>> 
>> $source.../../configs/multiple-instances-same-skype-user/
>>>>> 
>>>>> 2 skype clients run in difference folder
>>>>> /home/cucku/multi/interfaces01
>>>>> /home/cucku/multi/interfaces02
>>>>> 
>>>>> so i only need to config the
>> skypopen.conf.xml look
>>>> like below:
>>>>> <configuration name="skypopen.conf"
>>>> description="Skypopen Configuration">
>>>>>  <global_settings>
>>>>>    <param name="debug" value="8"/>
>>>>>    <param name="dialplan"
>> value="XML"/>
>>>>>    <param name="context"
>> value="default"/>
>>>>>    <param name="destination"
>> value="5000"/>
>>>>>    <param name="skype_user"
>>>> value="do_nguyen_ha"/>
>>>>>    <param
>> name="report_incoming_chatmessages"
>>>> value="false"/>
>>>>>    <param name="silent_mode"
>> value="false"/>
>>>>>    <param
>> name="write_silence_when_idle"
>>>> value="true"/>
>>>>>    <param name="setsockopt"
>> value="false"/>
>>>>>  </global_settings>
>>>>>  <!-- one entry here per each skypopen
>> interface
>>>> -->
>>>>>  <per_interface_settings>
>>>>>    <interface id="1"
>> name="interface1">
>>>>>        <param name="X11-display"
>>>> value=":101"/>
>>>>>    </interface>
>>>>>    <interface id="2"
>> name="interface2">
>>>>>        <param name="X11-display"
>>>> value=":102"/>
>>>>>    </interface>
>>>>>  </per_interface_settings>
>>>>> </configuration>
>>>>> 
>>>>> does the conifg look ok?? if not, please
>> guide me to
>>>> make it right
>>>>> 
>>>>> i have a question on event socket, what is
>> the event
>>>> name for skypopen to monitor the interface
>> IDLE/not IDLE or
>>>> ANSWER or IN PROGRESS...
>>>>> 
>>>>> which event plain should i take care of
>>>>> 
>>>>> 
>>>>> Thank you
>>>>> Ha`
>>>>> --- On Tue, 12/7/10, Meftah Tayeb <tayeb.meftah at gmail.com>
>>>> wrote:
>>>>> 
>>>>>> From: Meftah Tayeb <tayeb.meftah at gmail.com>
>>>>>> Subject: Re: [Freeswitch-users] get error
>> on
>>>> skypopen module
>>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>>> Cc: "ha do" <haloha201 at yahoo.com>
>>>>>> Date: Tuesday, December 7, 2010, 4:43 AM
>>>>>> did you autorised skypopen to access
>>>>>> skype from the skype client?
>>>>>> and did you configured the skypopen
>> interfaces to
>>>> each one
>>>>>> of the skype
>>>>>> clients?
>>>>>> thanks
>>>>>> Le 07/12/2010 05:34, ha do a ?crit :
>>>>>>> Hi list
>>>>>>> 
>>>>>>> i setup freeswitch and skypopen
>> running fine
>>>>>>> 
>>>>>>> there are 2 skype clients run on
>> freeswitch
>>>>>>> freeswitch at internal>  sk list
>>>>>>> sk console is NOT yet assigned
>>>>>>> F ID        Name
>>>>>>     IB (F/T)    OB (F/T)
>>>>>> State   CallFlw
>>>>>>    UUID
>>>>>>> = ====    ========
>>>>>> =======     =======
>>>>>>    ======  ============
>>>>>> ======
>>>>>>>     1
>>>>>>    [interface1]      0/1
>>>>>>       3/7
>>>>>> IDLE    IDLE
>>>>>>>     2
>>>>>>    [interface2]      0/5
>>>>>>       1/5
>>>>>> IDLE    IDLE
>>>>>>> 
>>>>>>> the skype clients are used the same
>> username
>>>> +
>>>>>> password of skype account
>>>>>>> 
>>>>>>> 
>>>>>>> the skypopen works fine but i get
>> error below
>>>> in the
>>>>>> debug mode
>>>>>>> 
>>>>>>> 2010-12-07 04:24:35.794611 [DEBUG]
>>>>>> switch_core_state_machine.c:462
>>>> (skypopen/interface1) State
>>>>>> DESTROY going to sleep
>>>>>>> 2010-12-07 04:24:35.795807 [DEBUG]
>>>>>> skypopen_protocol.c:173     [|]
>>>>>> [DEBUG_SKYPE  173  ][interface1
>>>>>>    ][IDLE,IDLE] READING: |||ERROR 559
>> CALL:
>>>>>> Action failed|||
>>>>>>> 2010-12-07 04:24:35.795807 [DEBUG]
>>>>>> skypopen_protocol.c:228     [|]
>>>>>> [DEBUG_SKYPE  228  ][interface1
>>>>>>    ][IDLE,IDLE] Skype got ERROR about
>> a
>>>>>> failed action (probably TRYING to HANGUP
>> A CALL),
>>>> no
>>>>>> problem: |||ERROR 559 CALL: Action
>> failed|||
>>>>>>> 2010-12-07 04:24:35.795807 [DEBUG]
>>>>>> skypopen_protocol.c:173     [|]
>>>>>> [DEBUG_SKYPE  173  ][interface1
>>>>>>    ][IDLE,IDLE] READING: |||ERROR 559
>> CALL:
>>>>>> Action failed|||
>>>>>>> 2010-12-07 04:24:35.795807 [DEBUG]
>>>>>> skypopen_protocol.c:228     [|]
>>>>>> [DEBUG_SKYPE  228  ][interface1
>>>>>>    ][IDLE,IDLE] Skype got ERROR about
>> a
>>>>>> failed action (probably TRYING to HANGUP
>> A CALL),
>>>> no
>>>>>> problem: |||ERROR 559 CALL: Action
>> failed|||
>>>>>>> 
>>>>>>> How to fix it
>>>>>>> 
>>>>>>> Thank you
>>>>>>> Ha`
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> --
>>>>>> Meftah Tayeb
>>>>>> inum: +883510001288000
>>>>>> Phone: +13602276297
>>>>>> Fax: +12538020313
>>>>>> 
>>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From peter.olsson at visionutveckling.se  Thu Dec  9 14:27:19 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 9 Dec 2010 12:27:19 +0100
Subject: [Freeswitch-users] Reliability of HEARTBEAT events every 20
 seconds? (in mod_event_socket)
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57ECC069FB@cooper>

Hello everyone,

I have a question about the reliability of heartbeat events, it should be sent from FreeSWITCH every 20 seconds - right? If you look below in my log sample (from my own process), you can see that it comes 12:07:01, then 21, then 41, but after this it suddenly takes 31 seconds for it to show up (12:08:12). I've also done a Wireshark trace, and it shows the same thing - so it is really delayed from FS. The load at the time was zero calls, and the server itself was using maybe 5% CPU. After the delayed event it continues to show up every 20 seconds.

[2010-12-09 12:07:01] Received HEARTBEAT event - we're alive.
[2010-12-09 12:07:21] Received HEARTBEAT event - we're alive.
[2010-12-09 12:07:41] Received HEARTBEAT event - we're alive.
[2010-12-09 12:08:12] Received HEARTBEAT event - we're alive.
[2010-12-09 12:08:32] Received HEARTBEAT event - we're alive.
[2010-12-09 12:08:52] Received HEARTBEAT event - we're alive.
[2010-12-09 12:09:12] Received HEARTBEAT event - we're alive.

Should the HEARTBEAT event be 100% reliable, or could something cause a delay like this? This server is not on latest GIT, but it's not more then one month old. I can debug this further myself, but I just want to know if it always should be triggered every 20 seconds, or if these events for some reasons might be delayed.

By the way - no clock skew or anything is detected on the machine, so that's not causing it.

Thanks,

Peter
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From steveayre at gmail.com  Thu Dec  9 14:39:39 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 9 Dec 2010 11:39:39 +0000
Subject: [Freeswitch-users] Reliability of HEARTBEAT events every 20
 seconds? (in mod_event_socket)
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57ECC069FB@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C57ECC069FB@cooper>
Message-ID: <AANLkTikjO6GiuzGRFtxcM+Ox04iwC0AKR2SktOcavLMg@mail.gmail.com>

What does the heartbeat event show as the uptime?

That should tell you whether the delay is occuring in the event being
generated, or whether the delay is when the event fires.

-Steve


On 9 December 2010 11:27, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> Hello everyone,
>
>
>
> I have a question about the reliability of heartbeat events, it should be
> sent from FreeSWITCH every 20 seconds ? right? If you look below in my log
> sample (from my own process), you can see that it comes 12:07:01, then 21,
> then 41, but after this it suddenly takes 31 seconds for it to show up
> (12:08:12). I?ve also done a Wireshark trace, and it shows the same thing ?
> so it is really delayed from FS. The load at the time was zero calls, and
> the server itself was using maybe 5% CPU. After the delayed event it
> continues to show up every 20 seconds.
>
>
>
> [2010-12-09 12:07:01] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:07:21] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:07:41] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:12] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:32] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:52] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:09:12] Received HEARTBEAT event - we're alive.
>
>
>
> Should the HEARTBEAT event be 100% reliable, or could something cause a
> delay like this? This server is not on latest GIT, but it?s not more then
> one month old. I can debug this further myself, but I just want to know if
> it always should be triggered every 20 seconds, or if these events for some
> reasons might be delayed.
>
>
>
> By the way ? no clock skew or anything is detected on the machine, so that?s
> not causing it.
>
>
>
> Thanks,
>
>
>
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Thu Dec  9 14:41:58 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 9 Dec 2010 11:41:58 +0000
Subject: [Freeswitch-users] Reliability of HEARTBEAT events every 20
 seconds? (in mod_event_socket)
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57ECC069FB@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C57ECC069FB@cooper>
Message-ID: <AANLkTik=6_OFuKKr4STS4fCLxbAt34s7b+Lkpn6QTnSW@mail.gmail.com>

Also do you have any warning messages like this in the fs log?
Task was executed late by %d seconds

(That should show up if the delay is in the scheduler, if the task
runs >1s  late).

-Steve


On 9 December 2010 11:27, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> Hello everyone,
>
>
>
> I have a question about the reliability of heartbeat events, it should be
> sent from FreeSWITCH every 20 seconds ? right? If you look below in my log
> sample (from my own process), you can see that it comes 12:07:01, then 21,
> then 41, but after this it suddenly takes 31 seconds for it to show up
> (12:08:12). I?ve also done a Wireshark trace, and it shows the same thing ?
> so it is really delayed from FS. The load at the time was zero calls, and
> the server itself was using maybe 5% CPU. After the delayed event it
> continues to show up every 20 seconds.
>
>
>
> [2010-12-09 12:07:01] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:07:21] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:07:41] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:12] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:32] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:52] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:09:12] Received HEARTBEAT event - we're alive.
>
>
>
> Should the HEARTBEAT event be 100% reliable, or could something cause a
> delay like this? This server is not on latest GIT, but it?s not more then
> one month old. I can debug this further myself, but I just want to know if
> it always should be triggered every 20 seconds, or if these events for some
> reasons might be delayed.
>
>
>
> By the way ? no clock skew or anything is detected on the machine, so that?s
> not causing it.
>
>
>
> Thanks,
>
>
>
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From peter.olsson at visionutveckling.se  Thu Dec  9 14:49:12 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 9 Dec 2010 12:49:12 +0100
Subject: [Freeswitch-users] Reliability of HEARTBEAT events every 20
 seconds? (in mod_event_socket)
In-Reply-To: <AANLkTik=6_OFuKKr4STS4fCLxbAt34s7b+Lkpn6QTnSW@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C57ECC069FB@cooper>
	<AANLkTik=6_OFuKKr4STS4fCLxbAt34s7b+Lkpn6QTnSW@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57ECC06A1B@cooper>

Nope, no warning message. Thanks for the tip of uptime, I will grab a log for that and wait for it to occur again.

It seems to occur once every 30-45 minutes, so not very often - but still very strange :)

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Steven Ayre
Skickat: den 9 december 2010 12:42
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Reliability of HEARTBEAT events every 20 seconds? (in mod_event_socket)

Also do you have any warning messages like this in the fs log?
Task was executed late by %d seconds

(That should show up if the delay is in the scheduler, if the task
runs >1s  late).

-Steve


On 9 December 2010 11:27, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> Hello everyone,
>
>
>
> I have a question about the reliability of heartbeat events, it should be
> sent from FreeSWITCH every 20 seconds - right? If you look below in my log
> sample (from my own process), you can see that it comes 12:07:01, then 21,
> then 41, but after this it suddenly takes 31 seconds for it to show up
> (12:08:12). I've also done a Wireshark trace, and it shows the same thing -
> so it is really delayed from FS. The load at the time was zero calls, and
> the server itself was using maybe 5% CPU. After the delayed event it
> continues to show up every 20 seconds.
>
>
>
> [2010-12-09 12:07:01] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:07:21] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:07:41] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:12] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:32] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:52] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:09:12] Received HEARTBEAT event - we're alive.
>
>
>
> Should the HEARTBEAT event be 100% reliable, or could something cause a
> delay like this? This server is not on latest GIT, but it's not more then
> one month old. I can debug this further myself, but I just want to know if
> it always should be triggered every 20 seconds, or if these events for some
> reasons might be delayed.
>
>
>
> By the way - no clock skew or anything is detected on the machine, so that's
> not causing it.
>
>
>
> Thanks,
>
>
>
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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

!DSPAM:4d00c1a432762107714629!



From peter.olsson at visionutveckling.se  Thu Dec  9 16:12:54 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 9 Dec 2010 14:12:54 +0100
Subject: [Freeswitch-users] Reliability of HEARTBEAT events every 20
 seconds? (in mod_event_socket)
In-Reply-To: <AANLkTik=6_OFuKKr4STS4fCLxbAt34s7b+Lkpn6QTnSW@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C57ECC069FB@cooper>
	<AANLkTik=6_OFuKKr4STS4fCLxbAt34s7b+Lkpn6QTnSW@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57ECC06A98@cooper>

Ok - I trapped the problem with full event info. The up-time also reports a difference, in this case 32 seconds. As you can see there is no load on the server at all by this time, no calls etc. Hmm, this is really strange I must say.

[2010-12-09 13:53:49] Received HEARTBEAT event - we're alive
content-length == 629
content-type == text/event-plain
core-uuid == 1550f368-b577-4304-bacd-2681d4067036
event-calling-file == switch_core.c
event-calling-function == send_heartbeat
event-calling-line-number == 65
event-date-gmt == Thu, 09 Dec 2010 12:53:53 GMT
event-date-local == 2010-12-09 13:53:53
event-date-timestamp == 1291899233256303
event-info == System Ready
event-name == HEARTBEAT
freeswitch-hostname == 79w3sipt4
freeswitch-ipv4 == 10.66.195.14
freeswitch-ipv6 == ::1
idle-cpu == 98.437500
session-count == 0
session-per-sec == 30
session-since-startup == 277
up-time == 0 years, 7 days, 1 hour, 1 minute, 1 second, 680 milliseconds, 750 microseconds

[2010-12-09 13:54:21] Received HEARTBEAT event - we're alive
content-length == 631
content-type == text/event-plain
core-uuid == 1550f368-b577-4304-bacd-2681d4067036
event-calling-file == switch_core.c
event-calling-function == send_heartbeat
event-calling-line-number == 65
event-date-gmt == Thu, 09 Dec 2010 12:54:25 GMT
event-date-local == 2010-12-09 13:54:25
event-date-timestamp == 1291899265259303
event-info == System Ready
event-name == HEARTBEAT
freeswitch-hostname == 79w3sipt4
freeswitch-ipv4 == 10.66.195.14
freeswitch-ipv6 == ::1
idle-cpu == 97.656250
session-count == 0
session-per-sec == 30
session-since-startup == 277
up-time == 0 years, 7 days, 1 hour, 1 minute, 33 seconds, 683 milliseconds, 750 microseconds

Mvh
Peter Olsson


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Steven Ayre
Skickat: den 9 december 2010 12:42
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Reliability of HEARTBEAT events every 20 seconds? (in mod_event_socket)

Also do you have any warning messages like this in the fs log?
Task was executed late by %d seconds

(That should show up if the delay is in the scheduler, if the task
runs >1s  late).

-Steve


On 9 December 2010 11:27, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> Hello everyone,
>
>
>
> I have a question about the reliability of heartbeat events, it should be
> sent from FreeSWITCH every 20 seconds - right? If you look below in my log
> sample (from my own process), you can see that it comes 12:07:01, then 21,
> then 41, but after this it suddenly takes 31 seconds for it to show up
> (12:08:12). I've also done a Wireshark trace, and it shows the same thing -
> so it is really delayed from FS. The load at the time was zero calls, and
> the server itself was using maybe 5% CPU. After the delayed event it
> continues to show up every 20 seconds.
>
>
>
> [2010-12-09 12:07:01] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:07:21] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:07:41] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:12] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:32] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:52] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:09:12] Received HEARTBEAT event - we're alive.
>
>
>
> Should the HEARTBEAT event be 100% reliable, or could something cause a
> delay like this? This server is not on latest GIT, but it's not more then
> one month old. I can debug this further myself, but I just want to know if
> it always should be triggered every 20 seconds, or if these events for some
> reasons might be delayed.
>
>
>
> By the way - no clock skew or anything is detected on the machine, so that's
> not causing it.
>
>
>
> Thanks,
>
>
>
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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

!DSPAM:4d00c1a432762107714629!



From melkybes at mail.ru  Thu Dec  9 16:31:42 2010
From: melkybes at mail.ru (=?UTF-8?Q?=D0=9C=D0=B8=D1=85=D0=B0=D0=B8=D0=BB_=D0=A1=D0=B0=D0=BB=D1=82=D0=B0=D0=BD=D0=BE=D0=B2?=)
Date: Thu, 09 Dec 2010 16:31:42 +0300
Subject: [Freeswitch-users]
	=?utf-8?b?Y2hhbmdlINGBYWxsZXJfaWRfbmFtZSAmIG51?= =?utf-8?q?mber?=
In-Reply-To: <4CFFBD23.6010608@gmail.com>
References: <E1PQNPD-0002tm-00.melkybes-mail-ru@f160.mail.ru>
	<4CFFBD23.6010608@gmail.com>
Message-ID: <E1PQgb0-0001QJ-00.melkybes-mail-ru@f252.mail.ru>


Don't work  :(



    <action application="set" data="effective_caller_id_name=russia"/>
<action application="set"data="effective_caller_id_number=+777777777777"/>
Le 08/12/2010 18:02, ?????? ???????? a ?crit :Hello.
  
Can I change Caller_id_name &amp; caller_id_number in freeswitch duringtransfer?
  
  
  
  
? ?????????  
?????? ????????  
  
  
  
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
  

-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313

? ?????????
?????? ????????



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From Nabble at slickdeals.endjunk.com  Thu Dec  9 17:02:40 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 9 Dec 2010 06:02:40 -0800 (PST)
Subject: [Freeswitch-users] Disabling Google Voice Voicemail
Message-ID: <1291903360189-5818816.post@n2.nabble.com>


My FS hosted on a Seagate 
http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
DockStar  works just fine to handle I/O calls with Google Voice (GV) service
using mod_dingaling. However, there is a simple problem with the GV
voicemail service that kicks in after two rings on my extension. unless
someone knows this, there seems to be no way to turn off and/or disable GV
voicemail. As such, I am looking for an alternate solution if anyone here
has one to share. This way, I don't have to rush to my FS extension whenever
it rings to avoid GV voicemail kicks in, let alone some of the calls from
not on GV services through mod_dingaling.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Disabling-Google-Voice-Voicemail-tp5818816p5818816.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mayamatakeshi at gmail.com  Thu Dec  9 17:11:55 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 9 Dec 2010 23:11:55 +0900
Subject: [Freeswitch-users] Refusing REFER requests
Message-ID: <AANLkTiks3YQm3Cj=M-WFYv+dA5g9QE4EJeyQmGEqE56O@mail.gmail.com>

Is there any way to ask FS to refuse a REFER request?
I mean, when a REFER is received, the first thing FS does is to disconnect
the channel of the referrer. Then, it sends the other channel to the
dialplan. But I want to prevent the unbridging of the call in some
circumstances (meaning: some destinations for transfer are not valid).
Obs: I don't want do disable support for REFER as I know it can be
configured in the sofia profile.

r,
takeshi
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From kris at kriskinc.com  Thu Dec  9 17:35:36 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 9 Dec 2010 09:35:36 -0500
Subject: [Freeswitch-users] sip-t
In-Reply-To: <AANLkTikqPCrhEeOyXa_6JUpa95J1OwZ1skY=CA9Y+DE9@mail.gmail.com>
References: <AANLkTikRctwUyPCrGOV0=r=K0L+Z=p4g0Pnz4eU9ADHX@mail.gmail.com>
	<AANLkTim+RdUGiHuwXr1mbf6pLi6ZWwYNqMrot2UOkAy2@mail.gmail.com>
	<AANLkTikqPCrhEeOyXa_6JUpa95J1OwZ1skY=CA9Y+DE9@mail.gmail.com>
Message-ID: <AANLkTimBty5LYj+=ELadyn2KQ7uN==AF7aQshCEi5VSd@mail.gmail.com>

I don't understand...

On Wed, Dec 8, 2010 at 8:35 PM, budi wibowo <bwibowo at gmail.com> wrote:
> based on experience connecting to sip-t device is possible, right?
>
> On Thu, Dec 9, 2010 at 5:48 AM, Kristian Kielhofner <kris at kriskinc.com>
> wrote:
>>
>> FreeSWITCH will not decode the encapsulated ISUP but it will give you
>> control over the multipart message body:
>>
>> http://wiki.freeswitch.org/wiki/Variable_sip_copy_multipart
>>
>> On Wed, Dec 8, 2010 at 5:44 PM, budi wibowo <bwibowo at gmail.com> wrote:
>> > hi
>> > does freeswitch support sip-t protocol?
>> >
>> > regards
>> > budi
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From brian at freeswitch.org  Thu Dec  9 17:40:19 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 9 Dec 2010 08:40:19 -0600
Subject: [Freeswitch-users] Refusing REFER requests
In-Reply-To: <AANLkTiks3YQm3Cj=M-WFYv+dA5g9QE4EJeyQmGEqE56O@mail.gmail.com>
References: <AANLkTiks3YQm3Cj=M-WFYv+dA5g9QE4EJeyQmGEqE56O@mail.gmail.com>
Message-ID: <F5A588F3-2B58-4069-BBCC-A33C56E649BE@freeswitch.org>

disable-transfer on the profile.

/b

On Dec 9, 2010, at 8:11 AM, mayamatakeshi wrote:

> Is there any way to ask FS to refuse a REFER request?
> I mean, when a REFER is received, the first thing FS does is to disconnect the channel of the referrer. Then, it sends the other channel to the dialplan. But I want to prevent the unbridging of the call in some circumstances (meaning: some destinations for transfer are not valid).
> Obs: I don't want do disable support for REFER as I know it can be configured in the sofia profile.
> 
> r,
> takeshi



From lloyd.aloysius at gmail.com  Thu Dec  9 18:27:27 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 9 Dec 2010 10:27:27 -0500
Subject: [Freeswitch-users] drops calls in the first 60 seconds
Message-ID: <AANLkTi=SsfAQfWP9GoH_dHi6+2rM9wCh+aBJFWV-P-Tq@mail.gmail.com>

Hi All,

How to trouble shoot drops calls in the first 60 seconds ?

*FreeSWITCH [Public IP / Hosted ] *<---------> Internet <--------> *Linksys
+ DD-WRT <------> Aastra 9143i IP Phone*

Any help appreciated

Thanks
Lloyd
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From Avi at aMarcus.com  Thu Dec  9 19:04:56 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Thu, 9 Dec 2010 18:04:56 +0200
Subject: [Freeswitch-users] Disabling Google Voice Voicemail
In-Reply-To: <1291903360189-5818816.post@n2.nabble.com>
References: <1291903360189-5818816.post@n2.nabble.com>
Message-ID: <AANLkTim_bp9Xk94R6uW2QJmp1dVPKOL8rxjhLS7s_K99@mail.gmail.com>

You can "answer" with FS and send a ringback until you pickup or send to
voicemail. This will start the billing for the other party, though.
-Avi

On Thu, Dec 9, 2010 at 4:02 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> My FS hosted on a Seagate
>
> http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
> DockStar  works just fine to handle I/O calls with Google Voice (GV)
> service
> using mod_dingaling. However, there is a simple problem with the GV
> voicemail service that kicks in after two rings on my extension. unless
> someone knows this, there seems to be no way to turn off and/or disable GV
> voicemail. As such, I am looking for an alternate solution if anyone here
> has one to share. This way, I don't have to rush to my FS extension
> whenever
> it rings to avoid GV voicemail kicks in, let alone some of the calls from
> not on GV services through mod_dingaling.
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Disabling-Google-Voice-Voicemail-tp5818816p5818816.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From djbinter at gmail.com  Thu Dec  9 19:16:35 2010
From: djbinter at gmail.com (DJB International)
Date: Thu, 9 Dec 2010 08:16:35 -0800
Subject: [Freeswitch-users] Reliability of HEARTBEAT events every 20
 seconds? (in mod_event_socket)
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57ECC06A98@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C57ECC069FB@cooper>
	<AANLkTik=6_OFuKKr4STS4fCLxbAt34s7b+Lkpn6QTnSW@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57ECC06A98@cooper>
Message-ID: <AANLkTim--+JD-yHOO8TuG+Y9MBwmo2XLsXY5Bqw3FPUz@mail.gmail.com>

Are you running FS with -nocal?

-djbinter

On Thu, Dec 9, 2010 at 5:12 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> Ok - I trapped the problem with full event info. The up-time also reports a
> difference, in this case 32 seconds. As you can see there is no load on the
> server at all by this time, no calls etc. Hmm, this is really strange I must
> say.
>
> [2010-12-09 13:53:49] Received HEARTBEAT event - we're alive
> content-length == 629
> content-type == text/event-plain
> core-uuid == 1550f368-b577-4304-bacd-2681d4067036
> event-calling-file == switch_core.c
> event-calling-function == send_heartbeat
> event-calling-line-number == 65
> event-date-gmt == Thu, 09 Dec 2010 12:53:53 GMT
> event-date-local == 2010-12-09 13:53:53
> event-date-timestamp == 1291899233256303
> event-info == System Ready
> event-name == HEARTBEAT
> freeswitch-hostname == 79w3sipt4
> freeswitch-ipv4 == 10.66.195.14
> freeswitch-ipv6 == ::1
> idle-cpu == 98.437500
> session-count == 0
> session-per-sec == 30
> session-since-startup == 277
> up-time == 0 years, 7 days, 1 hour, 1 minute, 1 second, 680 milliseconds,
> 750 microseconds
>
> [2010-12-09 13:54:21] Received HEARTBEAT event - we're alive
> content-length == 631
> content-type == text/event-plain
> core-uuid == 1550f368-b577-4304-bacd-2681d4067036
> event-calling-file == switch_core.c
> event-calling-function == send_heartbeat
> event-calling-line-number == 65
> event-date-gmt == Thu, 09 Dec 2010 12:54:25 GMT
> event-date-local == 2010-12-09 13:54:25
> event-date-timestamp == 1291899265259303
> event-info == System Ready
> event-name == HEARTBEAT
> freeswitch-hostname == 79w3sipt4
> freeswitch-ipv4 == 10.66.195.14
> freeswitch-ipv6 == ::1
> idle-cpu == 97.656250
> session-count == 0
> session-per-sec == 30
> session-since-startup == 277
> up-time == 0 years, 7 days, 1 hour, 1 minute, 33 seconds, 683 milliseconds,
> 750 microseconds
>
> Mvh
> Peter Olsson
>
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] F?r Steven Ayre
> Skickat: den 9 december 2010 12:42
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Reliability of HEARTBEAT events every 20
> seconds? (in mod_event_socket)
>
> Also do you have any warning messages like this in the fs log?
> Task was executed late by %d seconds
>
> (That should show up if the delay is in the scheduler, if the task
> runs >1s  late).
>
> -Steve
>
>
> On 9 December 2010 11:27, Peter Olsson <peter.olsson at visionutveckling.se>
> wrote:
> > Hello everyone,
> >
> >
> >
> > I have a question about the reliability of heartbeat events, it should be
> > sent from FreeSWITCH every 20 seconds - right? If you look below in my
> log
> > sample (from my own process), you can see that it comes 12:07:01, then
> 21,
> > then 41, but after this it suddenly takes 31 seconds for it to show up
> > (12:08:12). I've also done a Wireshark trace, and it shows the same thing
> -
> > so it is really delayed from FS. The load at the time was zero calls, and
> > the server itself was using maybe 5% CPU. After the delayed event it
> > continues to show up every 20 seconds.
> >
> >
> >
> > [2010-12-09 12:07:01] Received HEARTBEAT event - we're alive.
> >
> > [2010-12-09 12:07:21] Received HEARTBEAT event - we're alive.
> >
> > [2010-12-09 12:07:41] Received HEARTBEAT event - we're alive.
> >
> > [2010-12-09 12:08:12] Received HEARTBEAT event - we're alive.
> >
> > [2010-12-09 12:08:32] Received HEARTBEAT event - we're alive.
> >
> > [2010-12-09 12:08:52] Received HEARTBEAT event - we're alive.
> >
> > [2010-12-09 12:09:12] Received HEARTBEAT event - we're alive.
> >
> >
> >
> > Should the HEARTBEAT event be 100% reliable, or could something cause a
> > delay like this? This server is not on latest GIT, but it's not more then
> > one month old. I can debug this further myself, but I just want to know
> if
> > it always should be triggered every 20 seconds, or if these events for
> some
> > reasons might be delayed.
> >
> >
> >
> > By the way - no clock skew or anything is detected on the machine, so
> that's
> > not causing it.
> >
> >
> >
> > Thanks,
> >
> >
> >
> > Peter
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4d00c1a432762107714629!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tayeb.meftah at gmail.com  Thu Dec  9 19:31:23 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Thu, 09 Dec 2010 17:31:23 +0100
Subject: [Freeswitch-users] Refusing REFER requests
In-Reply-To: <AANLkTiks3YQm3Cj=M-WFYv+dA5g9QE4EJeyQmGEqE56O@mail.gmail.com>
References: <AANLkTiks3YQm3Cj=M-WFYv+dA5g9QE4EJeyQmGEqE56O@mail.gmail.com>
Message-ID: <4D01045B.1090704@gmail.com>

do you want to disable transfer completly?
or you want to twick it?

Le 09/12/2010 15:11, mayamatakeshi a ?crit :
> Is there any way to ask FS to refuse a REFER request?
> I mean, when a REFER is received, the first thing FS does is to 
> disconnect the channel of the referrer. Then, it sends the other 
> channel to the dialplan. But I want to prevent the unbridging of the 
> call in some circumstances (meaning: some destinations for transfer 
> are not valid).
> Obs: I don't want do disable support for REFER as I know it can be 
> configured in the sofia profile.
>
> r,
> takeshi
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313

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From tayeb.meftah at gmail.com  Thu Dec  9 19:33:20 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Thu, 09 Dec 2010 17:33:20 +0100
Subject: [Freeswitch-users] Disabling Google Voice Voicemail
In-Reply-To: <1291903360189-5818816.post@n2.nabble.com>
References: <1291903360189-5818816.post@n2.nabble.com>
Message-ID: <4D0104D0.4000801@gmail.com>

change the call timeout to a lower value like 10secs and bridge it to 
your vm?

Le 09/12/2010 15:02, mazilo a ?crit :
> My FS hosted on a Seagate
> http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
> DockStar  works just fine to handle I/O calls with Google Voice (GV) service
> using mod_dingaling. However, there is a simple problem with the GV
> voicemail service that kicks in after two rings on my extension. unless
> someone knows this, there seems to be no way to turn off and/or disable GV
> voicemail. As such, I am looking for an alternate solution if anyone here
> has one to share. This way, I don't have to rush to my FS extension whenever
> it rings to avoid GV voicemail kicks in, let alone some of the calls from
> not on GV services through mod_dingaling.
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313



From victor.chukalovskiy at utoronto.ca  Thu Dec  9 18:33:12 2010
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Thu, 09 Dec 2010 10:33:12 -0500
Subject: [Freeswitch-users] drops calls in the first 60 seconds
In-Reply-To: <AANLkTi=SsfAQfWP9GoH_dHi6+2rM9wCh+aBJFWV-P-Tq@mail.gmail.com>
References: <AANLkTi=SsfAQfWP9GoH_dHi6+2rM9wCh+aBJFWV-P-Tq@mail.gmail.com>
Message-ID: <4D00F6B8.3080006@utoronto.ca>

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From peter.olsson at visionutveckling.se  Thu Dec  9 19:44:25 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 9 Dec 2010 17:44:25 +0100
Subject: [Freeswitch-users] Reliability of HEARTBEAT events every 20
 seconds? (in mod_event_socket)
Message-ID: <C2628CCB-21C4-4C66-957F-4BB4AC1AF37F@visionutveckling.se>

No, and all timing seems ok (both good audio and timer_test returns with good results). And we're actually talking about 10 seconds here :)

This is on Windows, and I'm starting with -hp, I wonder if I might have some kind of thread priority issue - I can't remember seeing this before adding -hp.. hmm, I will check that out.


Peter Olsson

----- Reply message -----
Fr?n: "DJB International" <djbinter at gmail.com>
Datum: tors, dec 9, 2010 17:25
Rubrik: [Freeswitch-users] Reliability of HEARTBEAT events every 20 seconds? (in mod_event_socket)
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

Are you running FS with -nocal?

-djbinter

On Thu, Dec 9, 2010 at 5:12 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Ok - I trapped the problem with full event info. The up-time also reports a difference, in this case 32 seconds. As you can see there is no load on the server at all by this time, no calls etc. Hmm, this is really strange I must say.

[2010-12-09 13:53:49] Received HEARTBEAT event - we're alive
content-length == 629
content-type == text/event-plain
core-uuid == 1550f368-b577-4304-bacd-2681d4067036
event-calling-file == switch_core.c
event-calling-function == send_heartbeat
event-calling-line-number == 65
event-date-gmt == Thu, 09 Dec 2010 12:53:53 GMT
event-date-local == 2010-12-09 13:53:53
event-date-timestamp == 1291899233256303
event-info == System Ready
event-name == HEARTBEAT
freeswitch-hostname == 79w3sipt4
freeswitch-ipv4 == 10.66.195.14
freeswitch-ipv6 == ::1
idle-cpu == 98.437500
session-count == 0
session-per-sec == 30
session-since-startup == 277
up-time == 0 years, 7 days, 1 hour, 1 minute, 1 second, 680 milliseconds, 750 microseconds

[2010-12-09 13:54:21] Received HEARTBEAT event - we're alive
content-length == 631
content-type == text/event-plain
core-uuid == 1550f368-b577-4304-bacd-2681d4067036
event-calling-file == switch_core.c
event-calling-function == send_heartbeat
event-calling-line-number == 65
event-date-gmt == Thu, 09 Dec 2010 12:54:25 GMT
event-date-local == 2010-12-09 13:54:25
event-date-timestamp == 1291899265259303
event-info == System Ready
event-name == HEARTBEAT
freeswitch-hostname == 79w3sipt4
freeswitch-ipv4 == 10.66.195.14
freeswitch-ipv6 == ::1
idle-cpu == 97.656250
session-count == 0
session-per-sec == 30
session-since-startup == 277
up-time == 0 years, 7 days, 1 hour, 1 minute, 33 seconds, 683 milliseconds, 750 microseconds

Mvh
Peter Olsson


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Steven Ayre
Skickat: den 9 december 2010 12:42
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Reliability of HEARTBEAT events every 20 seconds? (in mod_event_socket)

Also do you have any warning messages like this in the fs log?
Task was executed late by %d seconds

(That should show up if the delay is in the scheduler, if the task
runs >1s  late).

-Steve


On 9 December 2010 11:27, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
> Hello everyone,
>
>
>
> I have a question about the reliability of heartbeat events, it should be
> sent from FreeSWITCH every 20 seconds - right? If you look below in my log
> sample (from my own process), you can see that it comes 12:07:01, then 21,
> then 41, but after this it suddenly takes 31 seconds for it to show up
> (12:08:12). I've also done a Wireshark trace, and it shows the same thing -
> so it is really delayed from FS. The load at the time was zero calls, and
> the server itself was using maybe 5% CPU. After the delayed event it
> continues to show up every 20 seconds.
>
>
>
> [2010-12-09 12:07:01] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:07:21] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:07:41] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:12] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:32] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:52] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:09:12] Received HEARTBEAT event - we're alive.
>
>
>
> Should the HEARTBEAT event be 100% reliable, or could something cause a
> delay like this? This server is not on latest GIT, but it's not more then
> one month old. I can debug this further myself, but I just want to know if
> it always should be triggered every 20 seconds, or if these events for some
> reasons might be delayed.
>
>
>
> By the way - no clock skew or anything is detected on the machine, so that's
> not causing it.
>
>
>
> Thanks,
>
>
>
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4d0102cf32763135019700!


From mario_fs at mgtech.com  Thu Dec  9 20:32:48 2010
From: mario_fs at mgtech.com (Mario G)
Date: Thu, 9 Dec 2010 09:32:48 -0800
Subject: [Freeswitch-users] NOTE: OS X memory leak info update
Message-ID: <2AA007DC-8FE1-483A-A193-3759ECE55406@mgtech.com>

I have been collecting data for Anthony for a month on this problem, but it changed twice after Git updates. There were 2 different leaks and it seems both may be fixed. I am posting for Anthony and anyone with FreeSWITCH on OS X who may want to update:

LEAK 1 The Big Leak - noted in jira 2008
FreeSWITCH allocated 2.3 to 2.3 meg per hour with no calls in or out. This required daily restarts. This leak fixed after Nov 22 Git.

LEAK 2
After November 22 2010 Git - I updated before I sent the observations since developers want the latest Git tested. Good thing I did, the previous big leak disappeared!  However, each in or out call added 400-500k to FreeSWITCH. It was never released in any 24 hour period.

LEAK 3?
After December 7 Git - Only a small amount (150-200k) is added per call. But.... I also tested in Linux at the same Git level and it did the same thing so this may be normal.

I updated the OS X Wiki with the recommended Git version.

Mario G

From danb.lists at googlemail.com  Thu Dec  9 20:39:00 2010
From: danb.lists at googlemail.com (Dan-Cristian Bogos)
Date: Thu, 9 Dec 2010 18:39:00 +0100
Subject: [Freeswitch-users] G729B and mod_com_g729
Message-ID: <AANLkTikwmhjavRRVznTDh0iW=nLUX3tDni3=pRPrJcKP@mail.gmail.com>

Hey Guys,

When using mod_com_g729 I found out that FreeSWITCH is rejecting G729B
in this form. I know about annexa=no parameter, however have seen
quite some UAs sending it in this form. Any work-around I can make
FreeSWITCH accepting it?

My FreeSWITCH version:
FreeSWITCH version: 1.0.head (git-) checked out somewhere in the
middle of October.
mod_com_g729: fsg729-167-installer

Debug of such call:

2010-12-07 13:56:05.371029 [DEBUG] sofia.c:4559 Channel
sofia/Ip-Switch.Net/7809163901 at 2.3.4.5 entering state [received][100]
2010-12-07 13:56:05.371029 [DEBUG] switch_core_state_machine.c:318
(sofia/Ip-Switch.Net/7809163901 at 2.3.4.5) Running State Change CS_NEW
2010-12-07 13:56:05.371029 [DEBUG] sofia.c:4570 Remote SDP:
v=0
o=- 24240 28749 IN IP4 1.2.3.4
s=session
c=IN IP4 1.2.3.4
t=0 0
m=audio 35892 RTP/AVP 18 4 56
a=rtpmap:18 G729B/8000
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3
a=rtpmap:56 telephone-event/8000
a=ptime:30
a=nortpproxy:yes
a=nortpproxy:yes
a=direction:active

2010-12-07 13:56:05.371029 [DEBUG] switch_core_state_machine.c:324
(sofia/Ip-Switch.Net/7809163901 at 2.3.4.5) State NEW
2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec
Compare [G729B:18:8000:30:8000]/[PCMU:0:8000:20:64000]
2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec
Compare [G723:4:8000:30:6300]/[PCMU:0:8000:20:64000]
2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec
Compare [telephone-event:56:8000:30:0]/[PCMU:0:8000:20:64000]
2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4455 Set 2833 dtmf
send/recv payload to 56
2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec
Compare [G729B:18:8000:30:8000]/[PCMA:8:8000:20:64000]
2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec
Compare [G723:4:8000:30:6300]/[PCMA:8:8000:20:64000]
2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec
Compare [telephone-event:56:8000:30:0]/[PCMA:8:8000:20:64000]
2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4455 Set 2833 dtmf
send/recv payload to 56
2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec
Compare [G729B:18:8000:30:8000]/[G729:18:8000:20:8000]
2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec
Compare [G723:4:8000:30:6300]/[G729:18:8000:20:8000]
2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4351 Audio Codec
Compare [telephone-event:56:8000:30:0]/[G729:18:8000:20:8000]
2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4455 Set 2833 dtmf
send/recv payload to 56
2010-12-07 13:56:05.371029 [DEBUG] sofia_glue.c:4455 Set 2833 dtmf
send/recv payload to 56
2010-12-07 13:56:05.371029 [DEBUG] switch_channel.c:2457
(sofia/Ip-Switch.Net/7809163901 at 2.3.4.5) Callstate Change DOWN ->
HANGUP
2010-12-07 13:56:05.371029 [NOTICE] sofia.c:4776 Hangup
sofia/Ip-Switch.Net/7809163901 at 2.3.4.5 [CS_NEW]
[INCOMPATIBLE_DESTINATION]
2010-12-07 13:56:05.371029 [DEBUG] switch_channel.c:2473 Send signal
sofia/Ip-Switch.Net/7809163901 at 2.3.4.5 [KILL]
2010-12-07 13:56:05.371029 [DEBUG] switch_core_session.c:1057 Send
signal sofia/Ip-Switch.Net/7809163901 at 2.3.4.5 [BREAK]
2010-12-07 13:56:05.371971 [DEBUG] switch_core_state_machine.c:318
(sofia/Ip-Switch.Net/7809163901 at 2.3.4.5) Running State Change
CS_HANGUP
2010-12-07 13:56:05.371971 [DEBUG] switch_core_state_machine.c:539
(sofia/Ip-Switch.Net/7809163901 at 2.3.4.5) State HANGUP
2010-12-07 13:56:05.371971 [DEBUG] mod_sofia.c:456 Channel
sofia/Ip-Switch.Net/7809163901 at 2.3.4.5 hanging up, cause:
INCOMPATIBLE_DESTINATION
2010-12-07 13:56:05.371971 [DEBUG] mod_sofia.c:518 Responding to
INVITE with: 488
2010-12-07 13:56:05.371971 [DEBUG] switch_core_state_machine.c:46
sofia/Ip-Switch.Net/7809163901 at 2.3.4.5 Standard HANGUP, cause:
INCOMPATIBLE_DESTINATION
2010-12-07 13:56:05.372880 [DEBUG] switch_core_state_machine.c:539
(sofia/Ip-Switch.Net/7809163901 at 2.3.4.5) State HANGUP going to sleep
2010-12-07 13:56:05.372880 [DEBUG] switch_core_state_machine.c:337
(sofia/Ip-Switch.Net/7809163901 at 2.3.4.5) State Change CS_HANGUP ->
CS_REPORTING
2010-12-07 13:56:05.372880 [DEBUG] switch_core_session.c:1057 Send
signal sofia/Ip-Switch.Net/7809163901 at 2.3.4.5 [BREAK]
2010-12-07 13:56:05.372880 [DEBUG] switch_core_state_machine.c:318
(sofia/Ip-Switch.Net/7809163901 at 2.3.4.5) Running State Change
CS_REPORTING
2010-12-07 13:56:05.372880 [DEBUG] switch_core_state_machine.c:599
(sofia/Ip-Switch.Net/7809163901 at 2.3.4.5) State REPORTING
2010-12-07 13:56:05.372880 [DEBUG] switch_core_state_machine.c:53
sofia/Ip-Switch.Net/7809163901 at 2.3.4.5 Standard REPORTING, cause:
INCOMPATIBLE_DESTINATION
2010-12-07 13:56:05.372880 [DEBUG] switch_core_state_machine.c:599
(sofia/Ip-Switch.Net/7809163901 at 2.3.4.5) State REPORTING going to
sleep
2010-12-07 13:56:05.372880 [DEBUG] switch_core_state_machine.c:331
(sofia/Ip-Switch.Net/7809163901 at 2.3.4.5) State Change CS_REPORTING ->
CS_DESTROY
2010-12-07 13:56:05.372880 [DEBUG] switch_core_session.c:1057 Send
signal sofia/Ip-Switch.Net/7809163901 at 2.3.4.5 [BREAK]
2010-12-07 13:56:05.372880 [DEBUG] switch_core_session.c:1224 Session
959 (sofia/Ip-Switch.Net/7809163901 at 2.3.4.5) Locked, Waiting on
external entities
2010-12-07 13:56:05.372880 [NOTICE] switch_core_session.c:1242 Session
959 (sofia/Ip-Switch.Net/7809163901 at 2.3.4.5) Ended


Ta in advance for any kind of tip!

DanB


From brian at freeswitch.org  Thu Dec  9 20:42:28 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 9 Dec 2010 11:42:28 -0600
Subject: [Freeswitch-users] G729B and mod_com_g729
In-Reply-To: <AANLkTikwmhjavRRVznTDh0iW=nLUX3tDni3=pRPrJcKP@mail.gmail.com>
References: <AANLkTikwmhjavRRVznTDh0iW=nLUX3tDni3=pRPrJcKP@mail.gmail.com>
Message-ID: <AF1F2258-0F3F-4735-B45A-F2535D6BA9A2@freeswitch.org>

Its rejecting it because saying G729B in your sdp is illegal. 

/b

On Dec 9, 2010, at 11:39 AM, Dan-Cristian Bogos wrote:

> Hey Guys,
> 
> When using mod_com_g729 I found out that FreeSWITCH is rejecting G729B
> in this form. I know about annexa=no parameter, however have seen
> quite some UAs sending it in this form. Any work-around I can make
> FreeSWITCH accepting it?
> 
> My FreeSWITCH version:
> FreeSWITCH version: 1.0.head (git-) checked out somewhere in the
> middle of October.
> mod_com_g729: fsg729-167-installer
> 
> Debug of such call:
> 
> 2010-12-07 13:56:05.371029 [DEBUG] sofia.c:4559 Channel
> sofia/Ip-Switch.Net/7809163901 at 2.3.4.5 entering state [received][100]
> 2010-12-07 13:56:05.371029 [DEBUG] switch_core_state_machine.c:318
> (sofia/Ip-Switch.Net/7809163901 at 2.3.4.5) Running State Change CS_NEW
> 2010-12-07 13:56:05.371029 [DEBUG] sofia.c:4570 Remote SDP:
> v=0
> o=- 24240 28749 IN IP4 1.2.3.4
> s=session
> c=IN IP4 1.2.3.4
> t=0 0
> m=audio 35892 RTP/AVP 18 4 56
> a=rtpmap:18 G729B/8000
> a=rtpmap:4 G723/8000
> a=fmtp:4 bitrate=6.3
> a=rtpmap:56 telephone-event/8000
> a=ptime:30
> a=nortpproxy:yes
> a=nortpproxy:yes
> a=direction:active



From mario_fs at mgtech.com  Thu Dec  9 20:47:52 2010
From: mario_fs at mgtech.com (Mario G)
Date: Thu, 9 Dec 2010 09:47:52 -0800
Subject: [Freeswitch-users] OS X lower FS idle processor if launchd used
Message-ID: <559EA6C2-785D-4403-8CD6-E7E411FE9306@mgtech.com>

Note for OS X users and developers:
While working on the memory leak problem I noticed that on OS X FreeSWITCH used more processor utilization at idle while running in background mode. The difference was 2.9 percent vs 4.3 percent. I found that if started with Terminal with or without -nc the utilization was lower than using launchd. I discovered that when i added -nc to the FS plist for launchd processor is lowered to the same level as if started from Terminal. I updated the OS X wiki.

Mario G

From mrene_lists at avgs.ca  Thu Dec  9 20:50:06 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 9 Dec 2010 12:50:06 -0500
Subject: [Freeswitch-users] NOTE: OS X memory leak info update
In-Reply-To: <2AA007DC-8FE1-483A-A193-3759ECE55406@mgtech.com>
References: <2AA007DC-8FE1-483A-A193-3759ECE55406@mgtech.com>
Message-ID: <D21D895A-87CA-4D5C-9C32-BEE65C556CA5@avgs.ca>

Can you give me a bit more details on what the calls are doing, maybe post the log of a typical call so we can lab it up under a memory debugger?

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-12-09, at 12:32 PM, Mario G wrote:

> I have been collecting data for Anthony for a month on this problem, but it changed twice after Git updates. There were 2 different leaks and it seems both may be fixed. I am posting for Anthony and anyone with FreeSWITCH on OS X who may want to update:
> 
> LEAK 1 The Big Leak - noted in jira 2008
> FreeSWITCH allocated 2.3 to 2.3 meg per hour with no calls in or out. This required daily restarts. This leak fixed after Nov 22 Git.
> 
> LEAK 2
> After November 22 2010 Git - I updated before I sent the observations since developers want the latest Git tested. Good thing I did, the previous big leak disappeared!  However, each in or out call added 400-500k to FreeSWITCH. It was never released in any 24 hour period.
> 
> LEAK 3?
> After December 7 Git - Only a small amount (150-200k) is added per call. But.... I also tested in Linux at the same Git level and it did the same thing so this may be normal.
> 
> I updated the OS X Wiki with the recommended Git version.
> 
> Mario G
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mario_fs at mgtech.com  Thu Dec  9 20:56:28 2010
From: mario_fs at mgtech.com (Mario G)
Date: Thu, 9 Dec 2010 09:56:28 -0800
Subject: [Freeswitch-users] Email Voicemail Volume Mystery
Message-ID: <A3633C87-61A4-41BB-B6BD-B9477F9AB5DB@mgtech.com>

This is consistent and baffling.This volume for emailed voicemails is not the same for different callers. This is an ONLY and issue for emailed voicemails, listening locally is fine. Whether the volume is good or bad is always consistent for a given caller.

For all cell phone callers so far:
The call volume AND local voicemail volume is good AND the emailed voicemail volume is good.

For almost all landline callers
The call volume AND local voicemail volume is good BUT the emailed voicemail volume is so low that it's unusable unless you are in a dead quite room.

The problem is NOT the device you listening on since I tested the file with an audio analyzer and the it's clear for the bad ones the volume is about 1/5 normal.

I could not find anything to try, any ideas?

Mario G




From mario_fs at mgtech.com  Thu Dec  9 21:01:29 2010
From: mario_fs at mgtech.com (Mario G)
Date: Thu, 9 Dec 2010 10:01:29 -0800
Subject: [Freeswitch-users] NOTE: OS X memory leak info update
In-Reply-To: <D21D895A-87CA-4D5C-9C32-BEE65C556CA5@avgs.ca>
References: <2AA007DC-8FE1-483A-A193-3759ECE55406@mgtech.com>
	<D21D895A-87CA-4D5C-9C32-BEE65C556CA5@avgs.ca>
Message-ID: <3D1E9F89-4C73-4EE2-8566-2CD816AE4A7E@mgtech.com>

This is meant as a notice for OS X users, not a request for help.

On Dec 9, 2010, at 9:50 AM, Mathieu Rene wrote:

> Can you give me a bit more details on what the calls are doing, maybe post the log of a typical call so we can lab it up under a memory debugger?
> 
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
> 
> 
> 
> 
> On 2010-12-09, at 12:32 PM, Mario G wrote:
> 
>> I have been collecting data for Anthony for a month on this problem, but it changed twice after Git updates. There were 2 different leaks and it seems both may be fixed. I am posting for Anthony and anyone with FreeSWITCH on OS X who may want to update:
>> 
>> LEAK 1 The Big Leak - noted in jira 2008
>> FreeSWITCH allocated 2.3 to 2.3 meg per hour with no calls in or out. This required daily restarts. This leak fixed after Nov 22 Git.
>> 
>> LEAK 2
>> After November 22 2010 Git - I updated before I sent the observations since developers want the latest Git tested. Good thing I did, the previous big leak disappeared!  However, each in or out call added 400-500k to FreeSWITCH. It was never released in any 24 hour period.
>> 
>> LEAK 3?
>> After December 7 Git - Only a small amount (150-200k) is added per call. But.... I also tested in Linux at the same Git level and it did the same thing so this may be normal.
>> 
>> I updated the OS X Wiki with the recommended Git version.
>> 
>> Mario G
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Thu Dec  9 21:10:40 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 9 Dec 2010 10:10:40 -0800
Subject: [Freeswitch-users] TTS volume issue
In-Reply-To: <AANLkTi=RMr5L80amk4+MutNZ37io1RcHrfCecK=JC7Ku@mail.gmail.com>
References: <AANLkTi=RMr5L80amk4+MutNZ37io1RcHrfCecK=JC7Ku@mail.gmail.com>
Message-ID: <AANLkTimxX6DEpz_33HF4ZqP6=jN+JnPwDKHewEDPo8tB@mail.gmail.com>

Which TTS engine are you using? Sometimes the TTS engine has a volume
control feature.
-MC

On Tue, Dec 7, 2010 at 11:11 PM, ovvenkat <ovvenkatesan at gmail.com> wrote:

> Hi to All,
>
> I need to increase the volume of TTS.
> I mean, What ever I am getting output from the TTS,
> Before playing to user, I need to increase the volume.
>
> Is it possible??
>
>
>
> --
>
> Regards
> Venkatesan OV.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Dec  9 21:12:35 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 9 Dec 2010 10:12:35 -0800
Subject: [Freeswitch-users] how bind_meta_app works?
In-Reply-To: <AANLkTi=O8WzT2a-NPCtaQ3ebgbTuzJ+YkTDMdXUsx7Ti@mail.gmail.com>
References: <AANLkTi=m0ww373iFn0nrZQ4hofyAp6duN=-Y4G-wyPgt@mail.gmail.com>
	<AANLkTi=O8WzT2a-NPCtaQ3ebgbTuzJ+YkTDMdXUsx7Ti@mail.gmail.com>
Message-ID: <AANLkTinaNJtjCNbORiK_3gWB+fRTU17C_aRL0LYtqkMf@mail.gmail.com>

Did you try enabling on the A leg also?
$EslCon->execute("bind_meta_app","1 ab o event::appli=testing")

Just curious as I've never actually tried this before.
-MC

On Wed, Dec 8, 2010 at 5:01 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi all,
> Is my understanding is right??. If so, is it possible to run applications
> in A leg...
>
>
> On Mon, Dec 6, 2010 at 12:19 PM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Hi all,
>> I was experimenting the bind_mea_app and I have a doubt.
>> There is a call from 1000 to FreeSwitch extension and it is connecting to
>> event outbound socket.
>> I've the following script in place.
>>
>> #!/usr/bin/perl
>> use strict;
>> use warnings;
>> use Data::Dumper;
>> use lib '/usr/src/freeswitch/libs/esl/perl/';
>> use IO::Socket::INET;
>> use ESL;
>>
>> my $SOCK = new IO::Socket::INET ( LocalHost => 'localhost',  LocalPort =>
>> '8447',  Proto => 'tcp',  Listen => 10,  Reuse => 1 );
>> unless ($SOCK)  {
>>                 print("Could not create socket: $!");
>>                 exit(2);
>> }
>>
>> while (1)  {
>>                 # Wait for any client through accept function in socket
>> module.
>>                 my $new_sock = $SOCK->accept();
>>                 next if (not defined($new_sock));
>>                 # Get socket host
>>                 my $host = $new_sock->sockhost();
>>                 print("Got a client accepted from $host\n");
>>                 my $pid = fork();
>>                 if ($pid) {
>>                                 close($new_sock);
>>                                 next;
>>                 }
>>                 my $fd = fileno($new_sock);
>>                 print("Newly forked child, pid: $$\n");
>>                 my $EslCon = new ESL::ESLconnection($fd);
>>                 print "Connection created successfully\n";
>>                 my $info = $EslCon->getInfo();
>>                 $EslCon->setEventLock("true");
>>                 my $uuid = $info->getHeader("unique-id");
>>                 print Dumper $info->serialize();
>>                 $EslCon->execute("set","bind_meta_key=#");
>>                 $EslCon->execute("bind_meta_app","1 b o
>> event::appli=testing");
>>                 my $api =
>> $EslCon->execute("bridge","{ignore_early_media=true}freetdm/1/a/xxxxxxxxx");
>> }
>>
>> The script called to the number that was dialed. In that number I pressed
>> #1, but the event was not sent,
>> and I assume that the bridge application is executing in the A leg ( is it
>> correct?? ) . So only after that bridge application gets completed the event
>> gets triggered out.
>>
>> If I execute the application in the B leg, then it is working fine, since
>> there is no application that is running at that time.
>> Now I wanted to know if there is any way to run the application on A leg,
>> I need that application to be executed even the bridge is not completed.
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.olsson at visionutveckling.se  Thu Dec  9 21:25:10 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 9 Dec 2010 19:25:10 +0100
Subject: [Freeswitch-users] Reliability of HEARTBEAT events every 20
 seconds? (in mod_event_socket)
In-Reply-To: <C2628CCB-21C4-4C66-957F-4BB4AC1AF37F@visionutveckling.se>
References: <C2628CCB-21C4-4C66-957F-4BB4AC1AF37F@visionutveckling.se>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57EC81C469@cooper>

That didn't make a difference either.. I will have to look into this deeper, and add some more debug logging for when the event fires within FS.

No more suggestions anyone? :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Peter Olsson [peter.olsson at visionutveckling.se]
Skickat: den 9 december 2010 17:44
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Reliability of HEARTBEAT events every 20 seconds? (in mod_event_socket)

No, and all timing seems ok (both good audio and timer_test returns with good results). And we're actually talking about 10 seconds here :)

This is on Windows, and I'm starting with -hp, I wonder if I might have some kind of thread priority issue - I can't remember seeing this before adding -hp.. hmm, I will check that out.


Peter Olsson

----- Reply message -----
Fr?n: "DJB International" <djbinter at gmail.com>
Datum: tors, dec 9, 2010 17:25
Rubrik: [Freeswitch-users] Reliability of HEARTBEAT events every 20 seconds? (in mod_event_socket)
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

Are you running FS with -nocal?

-djbinter

On Thu, Dec 9, 2010 at 5:12 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Ok - I trapped the problem with full event info. The up-time also reports a difference, in this case 32 seconds. As you can see there is no load on the server at all by this time, no calls etc. Hmm, this is really strange I must say.

[2010-12-09 13:53:49] Received HEARTBEAT event - we're alive
content-length == 629
content-type == text/event-plain
core-uuid == 1550f368-b577-4304-bacd-2681d4067036
event-calling-file == switch_core.c
event-calling-function == send_heartbeat
event-calling-line-number == 65
event-date-gmt == Thu, 09 Dec 2010 12:53:53 GMT
event-date-local == 2010-12-09 13:53:53
event-date-timestamp == 1291899233256303
event-info == System Ready
event-name == HEARTBEAT
freeswitch-hostname == 79w3sipt4
freeswitch-ipv4 == 10.66.195.14
freeswitch-ipv6 == ::1
idle-cpu == 98.437500
session-count == 0
session-per-sec == 30
session-since-startup == 277
up-time == 0 years, 7 days, 1 hour, 1 minute, 1 second, 680 milliseconds, 750 microseconds

[2010-12-09 13:54:21] Received HEARTBEAT event - we're alive
content-length == 631
content-type == text/event-plain
core-uuid == 1550f368-b577-4304-bacd-2681d4067036
event-calling-file == switch_core.c
event-calling-function == send_heartbeat
event-calling-line-number == 65
event-date-gmt == Thu, 09 Dec 2010 12:54:25 GMT
event-date-local == 2010-12-09 13:54:25
event-date-timestamp == 1291899265259303
event-info == System Ready
event-name == HEARTBEAT
freeswitch-hostname == 79w3sipt4
freeswitch-ipv4 == 10.66.195.14
freeswitch-ipv6 == ::1
idle-cpu == 97.656250
session-count == 0
session-per-sec == 30
session-since-startup == 277
up-time == 0 years, 7 days, 1 hour, 1 minute, 33 seconds, 683 milliseconds, 750 microseconds

Mvh
Peter Olsson


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Steven Ayre
Skickat: den 9 december 2010 12:42
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Reliability of HEARTBEAT events every 20 seconds? (in mod_event_socket)

Also do you have any warning messages like this in the fs log?
Task was executed late by %d seconds

(That should show up if the delay is in the scheduler, if the task
runs >1s  late).

-Steve


On 9 December 2010 11:27, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
> Hello everyone,
>
>
>
> I have a question about the reliability of heartbeat events, it should be
> sent from FreeSWITCH every 20 seconds - right? If you look below in my log
> sample (from my own process), you can see that it comes 12:07:01, then 21,
> then 41, but after this it suddenly takes 31 seconds for it to show up
> (12:08:12). I've also done a Wireshark trace, and it shows the same thing -
> so it is really delayed from FS. The load at the time was zero calls, and
> the server itself was using maybe 5% CPU. After the delayed event it
> continues to show up every 20 seconds.
>
>
>
> [2010-12-09 12:07:01] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:07:21] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:07:41] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:12] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:32] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:08:52] Received HEARTBEAT event - we're alive.
>
> [2010-12-09 12:09:12] Received HEARTBEAT event - we're alive.
>
>
>
> Should the HEARTBEAT event be 100% reliable, or could something cause a
> delay like this? This server is not on latest GIT, but it's not more then
> one month old. I can debug this further myself, but I just want to know if
> it always should be triggered every 20 seconds, or if these events for some
> reasons might be delayed.
>
>
>
> By the way - no clock skew or anything is detected on the machine, so that's
> not causing it.
>
>
>
> Thanks,
>
>
>
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



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

!DSPAM:4d01088c32762074620997!



From brian at freeswitch.org  Thu Dec  9 21:35:43 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 9 Dec 2010 12:35:43 -0600
Subject: [Freeswitch-users] Reliability of HEARTBEAT events every 20
	seconds? (in mod_event_socket)
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57EC81C469@cooper>
References: <C2628CCB-21C4-4C66-957F-4BB4AC1AF37F@visionutveckling.se>
	<549CFEF87AEDE841A38E9D15EAB4C04C57EC81C469@cooper>
Message-ID: <934A2ECD-3013-4598-8740-F20C6162FB2B@freeswitch.org>

You do realize the time in the log is when its submitted.  And you have some time between then the task is run then rescheduled... Are we really going to be that picky?

/b

On Dec 9, 2010, at 12:25 PM, Peter Olsson wrote:

> That didn't make a difference either.. I will have to look into this deeper, and add some more debug logging for when the event fires within FS.
> 
> No more suggestions anyone? :)
> 
> /Peter



From lloyd.aloysius at sunteltech.ca  Thu Dec  9 21:39:01 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Thu, 9 Dec 2010 13:39:01 -0500
Subject: [Freeswitch-users] drops calls in the first 60 seconds
In-Reply-To: <4D00F6B8.3080006@utoronto.ca>
References: <AANLkTi=SsfAQfWP9GoH_dHi6+2rM9wCh+aBJFWV-P-Tq@mail.gmail.com>
	<4D00F6B8.3080006@utoronto.ca>
Message-ID: <AANLkTimEEpAsYvZpSQeHqNnAv8QN7BhaG56aZb2a0SsY@mail.gmail.com>

Chris,

I just switch the customer from Asterisk to FreeSWITCH. The same DD-WRT
firmware run smoothly for a long time with the Asterisk.I do not think there
is no issue with DD-WRT firmware. I am thinking the Aastra phone.

Because of calls Drop , I do not want to switch the router which is working
fine. In your case Asterisk/ FreeeSWITCH in the same network. My case
FreeSWITCH in the Data center( Public Network) , Phones behind the NAT

I work from my home and I use the same DD-WRT firmware to connect same
FreeSWITCH, but using Linksys SPA942 phone. Never have any problem.

Victor,

Thank you for the suggestion. I am going to capture and analyse.


My question is Does anyone have similar experience with Aastra Phones &
FreeSWITCH.

Thanks
Lloyd




On Thu, Dec 9, 2010 at 10:33 AM, Victor Chukalovskiy <
victor.chukalovskiy at utoronto.ca> wrote:

>  Hi Lloyd,
>
> I'd start by looking on a hangup cause from FS CDR.
>
> You can also do packet capture on your DD-WRT with tcpdump and libpcap.
> Or do packet capture on your FS machine.
> Then analyze packets with WireShark on your computer.
>
> Regards,
> Victor
>
>
> On 09/12/10 10:27 AM, Aloysius Lloyd wrote:
>
> Hi All,
>
> How to trouble shoot drops calls in the first 60 seconds ?
>
> *FreeSWITCH [Public IP / Hosted ] *<---------> Internet <--------> *Linksys
> + DD-WRT <------> Aastra 9143i IP Phone*
>
> Any help appreciated
>
> Thanks
> Lloyd
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Thu Dec  9 21:43:52 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 9 Dec 2010 12:43:52 -0600
Subject: [Freeswitch-users] drops calls in the first 60 seconds
In-Reply-To: <AANLkTimEEpAsYvZpSQeHqNnAv8QN7BhaG56aZb2a0SsY@mail.gmail.com>
References: <AANLkTi=SsfAQfWP9GoH_dHi6+2rM9wCh+aBJFWV-P-Tq@mail.gmail.com>
	<4D00F6B8.3080006@utoronto.ca>
	<AANLkTimEEpAsYvZpSQeHqNnAv8QN7BhaG56aZb2a0SsY@mail.gmail.com>
Message-ID: <3DA48A34-E06E-4D67-BE05-E05B800699DC@freeswitch.org>

First off you need to change the phone side port.. you can not have two phones on the same port behind the same nat... secondly enable rport on the phone's config.

/b

On Dec 9, 2010, at 12:39 PM, Aloysius Lloyd wrote:

> Chris,
> 
> I just switch the customer from Asterisk to FreeSWITCH. The same DD-WRT firmware run smoothly for a long time with the Asterisk.I do not think there is no issue with DD-WRT firmware. I am thinking the Aastra phone.
> 
> Because of calls Drop , I do not want to switch the router which is working fine. In your case Asterisk/ FreeeSWITCH in the same network. My case FreeSWITCH in the Data center( Public Network) , Phones behind the NAT
> 
> I work from my home and I use the same DD-WRT firmware to connect same FreeSWITCH, but using Linksys SPA942 phone. Never have any problem.
> 
> Victor,
> 
> Thank you for the suggestion. I am going to capture and analyse.
> 
> 
> My question is Does anyone have similar experience with Aastra Phones & FreeSWITCH.
> 
> Thanks
> Lloyd

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From peter.olsson at visionutveckling.se  Thu Dec  9 21:42:09 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 9 Dec 2010 19:42:09 +0100
Subject: [Freeswitch-users] Reliability of HEARTBEAT events every
	20	seconds? (in mod_event_socket)
In-Reply-To: <934A2ECD-3013-4598-8740-F20C6162FB2B@freeswitch.org>
References: <C2628CCB-21C4-4C66-957F-4BB4AC1AF37F@visionutveckling.se>
	<549CFEF87AEDE841A38E9D15EAB4C04C57EC81C469@cooper>,
	<934A2ECD-3013-4598-8740-F20C6162FB2B@freeswitch.org>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57EC81C46C@cooper>

Yes I realize this, but I just think 10-12 seconds (30-32 instead of 20) seems like a little too much to be correct - or should I consider this as normal?

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Brian West [brian at freeswitch.org]
Skickat: den 9 december 2010 19:35
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Reliability of HEARTBEAT events every 20   seconds? (in mod_event_socket)

You do realize the time in the log is when its submitted.  And you have some time between then the task is run then rescheduled... Are we really going to be that picky?

/b

On Dec 9, 2010, at 12:25 PM, Peter Olsson wrote:

> That didn't make a difference either.. I will have to look into this deeper, and add some more debug logging for when the event fires within FS.
>
> No more suggestions anyone? :)
>
> /Peter


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

!DSPAM:4d0122b032761938095686!



From lloyd.aloysius at sunteltech.ca  Thu Dec  9 21:48:59 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Thu, 9 Dec 2010 13:48:59 -0500
Subject: [Freeswitch-users] drops calls in the first 60 seconds
In-Reply-To: <3DA48A34-E06E-4D67-BE05-E05B800699DC@freeswitch.org>
References: <AANLkTi=SsfAQfWP9GoH_dHi6+2rM9wCh+aBJFWV-P-Tq@mail.gmail.com>
	<4D00F6B8.3080006@utoronto.ca>
	<AANLkTimEEpAsYvZpSQeHqNnAv8QN7BhaG56aZb2a0SsY@mail.gmail.com>
	<3DA48A34-E06E-4D67-BE05-E05B800699DC@freeswitch.org>
Message-ID: <AANLkTi=61oeCQwcJH00-uFVeWhQNNEOX02fsVLXXgvU0@mail.gmail.com>

Thanks Brian,

1. Phone side port - I will give a try.

2. rport - already enabled.

Thanks
Lloyd


On Thu, Dec 9, 2010 at 1:43 PM, Brian West <brian at freeswitch.org> wrote:

> First off you need to change the phone side port.. you can not have two
> phones on the same port behind the same nat... secondly enable rport on the
> phone's config.
>
> /b
>
> On Dec 9, 2010, at 12:39 PM, Aloysius Lloyd wrote:
>
> Chris,
>
> I just switch the customer from Asterisk to FreeSWITCH. The same DD-WRT
> firmware run smoothly for a long time with the Asterisk.I do not think
> there is no issue with DD-WRT firmware. I am thinking the Aastra phone.
>
> Because of calls Drop , I do not want to switch the router which is working
> fine. In your case Asterisk/ FreeeSWITCH in the same network. My case
> FreeSWITCH in the Data center( Public Network) , Phones behind the NAT
>
> I work from my home and I use the same DD-WRT firmware to connect same
> FreeSWITCH, but using Linksys SPA942 phone. Never have any problem.
>
> Victor,
>
> Thank you for the suggestion. I am going to capture and analyse.
>
>
> My question is Does anyone have similar experience with Aastra Phones &
> FreeSWITCH.
>
> Thanks
> Lloyd
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at sunteltech.ca  Thu Dec  9 21:57:47 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Thu, 9 Dec 2010 13:57:47 -0500
Subject: [Freeswitch-users] drops calls in the first 60 seconds
In-Reply-To: <AANLkTi=61oeCQwcJH00-uFVeWhQNNEOX02fsVLXXgvU0@mail.gmail.com>
References: <AANLkTi=SsfAQfWP9GoH_dHi6+2rM9wCh+aBJFWV-P-Tq@mail.gmail.com>
	<4D00F6B8.3080006@utoronto.ca>
	<AANLkTimEEpAsYvZpSQeHqNnAv8QN7BhaG56aZb2a0SsY@mail.gmail.com>
	<3DA48A34-E06E-4D67-BE05-E05B800699DC@freeswitch.org>
	<AANLkTi=61oeCQwcJH00-uFVeWhQNNEOX02fsVLXXgvU0@mail.gmail.com>
Message-ID: <AANLkTimOWEWQjbC_EamMbm3i1CKTRVzM3Y=JiPcYUOYH@mail.gmail.com>

Brian,

In the current scenario Asterisk and Aastra phones working smoothly without
any additional settings.
All I need to enter username , password and sip server address.

But FreeSWITCH and Aastra phone - I need to enter username , password , sip
server address , rport enable. *Without rport enabled, phone never register
to FreeSSWITCH.*

Why FreeSWITCH and Nat behaving like this? What is difference between
FreeSWITCH NAT and Asterisk NAT.

I do not want to go back Asterisk. I want to educate myself. Also want to
switch more customers to FreeSWITCH.

Any help is appreciated.

Thanks
Lloyd

On Thu, Dec 9, 2010 at 1:48 PM, Aloysius Lloyd <lloyd.aloysius at sunteltech.ca
> wrote:

> Thanks Brian,
>
> 1. Phone side port - I will give a try.
>
> 2. rport - already enabled.
>
> Thanks
> Lloyd
>
>
> On Thu, Dec 9, 2010 at 1:43 PM, Brian West <brian at freeswitch.org> wrote:
>
>> First off you need to change the phone side port.. you can not have two
>> phones on the same port behind the same nat... secondly enable rport on the
>> phone's config.
>>
>> /b
>>
>> On Dec 9, 2010, at 12:39 PM, Aloysius Lloyd wrote:
>>
>> Chris,
>>
>> I just switch the customer from Asterisk to FreeSWITCH. The same DD-WRT
>> firmware run smoothly for a long time with the Asterisk.I do not think
>> there is no issue with DD-WRT firmware. I am thinking the Aastra phone.
>>
>> Because of calls Drop , I do not want to switch the router which is
>> working fine. In your case Asterisk/ FreeeSWITCH in the same network. My
>> case FreeSWITCH in the Data center( Public Network) , Phones behind the NAT
>>
>> I work from my home and I use the same DD-WRT firmware to connect same
>> FreeSWITCH, but using Linksys SPA942 phone. Never have any problem.
>>
>> Victor,
>>
>> Thank you for the suggestion. I am going to capture and analyse.
>>
>>
>> My question is Does anyone have similar experience with Aastra Phones &
>> FreeSWITCH.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From freeswitch at peely.com  Thu Dec  9 22:10:15 2010
From: freeswitch at peely.com (peely)
Date: Thu, 9 Dec 2010 11:10:15 -0800 (PST)
Subject: [Freeswitch-users] Stop offering video on outbound if no video
	offered on inbound
Message-ID: <1291921815608-5820193.post@n2.nabble.com>


Hi,

I have a "registrar" sip profile which I have set H263 and H264 in the
codec-prefs to allow video.

Now, I see that when a call is offered which only provides audio codecs, FS
still offers video codecs on the outbound call. Interestingly I have also
occasionally seen this on session timer updates, where an audio codec had
been negotiated but then video offered suddenly in the timer refresh, which
has confused a few devices.

Is there an option to not offer video codecs on the egress if no video codec
is offered on the ingress, and possibly to allow audio transcoding but no
video transcoding?

I guess I could have a stab at a set of swanky dialplan entries but wondered
if there's a specific profile option.


Thanks,


Neil.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Stop-offering-video-on-outbound-if-no-video-offered-on-inbound-tp5820193p5820193.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Thu Dec  9 22:28:05 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 9 Dec 2010 13:28:05 -0600
Subject: [Freeswitch-users] Stop offering video on outbound if no video
	offered on inbound
In-Reply-To: <1291921815608-5820193.post@n2.nabble.com>
References: <1291921815608-5820193.post@n2.nabble.com>
Message-ID: <610548AF-200D-4B3A-9AEE-F1901ACE6A3F@freeswitch.org>

Its only going to offer what you have allowed in the codec prefs.  Check your profile to ensure that its not still there... sofia status profile xxxx and you'll see it in the list i'm sure.

/b

On Dec 9, 2010, at 1:10 PM, peely wrote:

> 
> Hi,
> 
> I have a "registrar" sip profile which I have set H263 and H264 in the
> codec-prefs to allow video.
> 
> Now, I see that when a call is offered which only provides audio codecs, FS
> still offers video codecs on the outbound call. Interestingly I have also
> occasionally seen this on session timer updates, where an audio codec had
> been negotiated but then video offered suddenly in the timer refresh, which
> has confused a few devices.
> 
> Is there an option to not offer video codecs on the egress if no video codec
> is offered on the ingress, and possibly to allow audio transcoding but no
> video transcoding?
> 
> I guess I could have a stab at a set of swanky dialplan entries but wondered
> if there's a specific profile option.
> 
> 
> Thanks,
> 
> 
> Neil.



From brian at freeswitch.org  Thu Dec  9 22:31:11 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 9 Dec 2010 13:31:11 -0600
Subject: [Freeswitch-users] drops calls in the first 60 seconds
In-Reply-To: <AANLkTimOWEWQjbC_EamMbm3i1CKTRVzM3Y=JiPcYUOYH@mail.gmail.com>
References: <AANLkTi=SsfAQfWP9GoH_dHi6+2rM9wCh+aBJFWV-P-Tq@mail.gmail.com>
	<4D00F6B8.3080006@utoronto.ca>
	<AANLkTimEEpAsYvZpSQeHqNnAv8QN7BhaG56aZb2a0SsY@mail.gmail.com>
	<3DA48A34-E06E-4D67-BE05-E05B800699DC@freeswitch.org>
	<AANLkTi=61oeCQwcJH00-uFVeWhQNNEOX02fsVLXXgvU0@mail.gmail.com>
	<AANLkTimOWEWQjbC_EamMbm3i1CKTRVzM3Y=JiPcYUOYH@mail.gmail.com>
Message-ID: <8ED4B2C0-085C-4E02-BF38-75A028EB5173@freeswitch.org>

Get sip traces of it not working and revert your sofia profiles back to what is in git and I bet it will just work.  

So you can take your pick.. make your device do its job and not force the registrar to overcome your devices stupidness and you can have call density out the wazzo or you can pick to have FS overcome it for them chewing up more cpu and processing power trying to overcome your stupid device and then only have a handful of calls on a machine.  Its really what it boils down to.

/b

On Dec 9, 2010, at 12:57 PM, Aloysius Lloyd wrote:

> Brian,
> 
> In the current scenario Asterisk and Aastra phones working smoothly without any additional settings.
> All I need to enter username , password and sip server address.
> 
> But FreeSWITCH and Aastra phone - I need to enter username , password , sip server address , rport enable. Without rport enabled, phone never register to FreeSSWITCH.
> 
> Why FreeSWITCH and Nat behaving like this? What is difference between FreeSWITCH NAT and Asterisk NAT.
> 
> I do not want to go back Asterisk. I want to educate myself. Also want to switch more customers to FreeSWITCH.
> 
> Any help is appreciated.
> 
> Thanks
> Lloyd

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From Nabble at slickdeals.endjunk.com  Thu Dec  9 22:45:16 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 9 Dec 2010 11:45:16 -0800 (PST)
Subject: [Freeswitch-users] Disabling Google Voice Voicemail
In-Reply-To: <AANLkTim_bp9Xk94R6uW2QJmp1dVPKOL8rxjhLS7s_K99@mail.gmail.com>
References: <1291903360189-5818816.post@n2.nabble.com>
	<AANLkTim_bp9Xk94R6uW2QJmp1dVPKOL8rxjhLS7s_K99@mail.gmail.com>
Message-ID: <1291923916094-5820341.post@n2.nabble.com>



Avi Marcus wrote:
> 
> You can "answer" with FS and send a ringback until you pickup or send to
> voicemail. This will start the billing for the other party, though.
This is OK since Google Voice has already implemented 
http://www.voip-info.org/wiki/view/Fake+False+Answer+Supervision+%28FAS%29+service
FAS . But, the question is how to implement this in FS? Can you kindly show
some examples?

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Disabling-Google-Voice-Voicemail-tp5818816p5820341.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Thu Dec  9 22:45:53 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 9 Dec 2010 11:45:53 -0800 (PST)
Subject: [Freeswitch-users] Disabling Google Voice Voicemail
In-Reply-To: <4D0104D0.4000801@gmail.com>
References: <1291903360189-5818816.post@n2.nabble.com>
	<4D0104D0.4000801@gmail.com>
Message-ID: <1291923953990-5820345.post@n2.nabble.com>



Meftah Tayeb wrote:
> 
> change the call timeout to a lower value like 10secs and bridge it to 
> your vm?
This will be fine. However, can you show some examples, please?

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Disabling-Google-Voice-Voicemail-tp5818816p5820345.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Avi at aMarcus.com  Thu Dec  9 23:49:00 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Thu, 9 Dec 2010 22:49:00 +0200
Subject: [Freeswitch-users] Disabling Google Voice Voicemail
In-Reply-To: <1291923916094-5820341.post@n2.nabble.com>
References: <1291903360189-5818816.post@n2.nabble.com>
	<AANLkTim_bp9Xk94R6uW2QJmp1dVPKOL8rxjhLS7s_K99@mail.gmail.com>
	<1291923916094-5820341.post@n2.nabble.com>
Message-ID: <AANLkTinndUGCzh+gsoyZLjQ8LR+phQLuBMsW1LvMSH4+@mail.gmail.com>

<action application="answer"/>
<action application="send_dtmf" data="1"/> (if needed for call screening)

Hmm, the ringing part is harder than I thought. I don't think ring_ready
will work...
Well try that, if not, maybe a transfer or something after - see
http://wiki.freeswitch.org/wiki/Dialplan_FollowMe#FollowMe_Dialplan_Example_3

-Avi


On Thu, Dec 9, 2010 at 9:45 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
>
> Avi Marcus wrote:
> >
> > You can "answer" with FS and send a ringback until you pickup or send to
> > voicemail. This will start the billing for the other party, though.
> This is OK since Google Voice has already implemented
>
> http://www.voip-info.org/wiki/view/Fake+False+Answer+Supervision+%28FAS%29+service
> FAS . But, the question is how to implement this in FS? Can you kindly show
> some examples?
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Disabling-Google-Voice-Voicemail-tp5818816p5820341.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Dec  9 23:53:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 9 Dec 2010 12:53:16 -0800
Subject: [Freeswitch-users] =?koi8-r?b?Y2hhbmdlINNhbGxlcl9pZF9uYW1lICYg?=
	=?koi8-r?b?bnVtYmVy?=
In-Reply-To: <E1PQgb0-0001QJ-00.melkybes-mail-ru@f252.mail.ru>
References: <E1PQNPD-0002tm-00.melkybes-mail-ru@f160.mail.ru>
	<4CFFBD23.6010608@gmail.com>
	<E1PQgb0-0001QJ-00.melkybes-mail-ru@f252.mail.ru>
Message-ID: <AANLkTi=OBmDKBwmx3p0z-RQDa8bOsW3jeHTiKPPxO-cD@mail.gmail.com>

On Thu, Dec 9, 2010 at 5:31 AM, ?????? ???????? <melkybes at mail.ru> wrote:

>
> Don't work  :(
>
Please supply more information. Best thing to do is to use pastebin (
pastebin.freeswitch.org)
Things to put in there:
Your setup - what phones are you using and what kind of telephone service?
(VoIP, PRI, analog, etc.)
Your dialplan for this extension
The console debug for this call
If you don't have a lot of traffic you can also include a SIP trace
(assuming you are using SIP)
At the console do "sofia global siptrace on"
NOTE: the siptrace goes to the console but not to the log file, so you need
to capture the output at the console or use the logger.pl script in
/scripts/perl under the fs source dir.

Put the pastebin URL in this thread.

Also, check this page for helpful tips on collecting information for
debugging:
http://wiki.freeswitch.org/wiki/Reporting_Bugs

-MC

>
>
>
>
>  <action application="set" data="effective_caller_id_name=russia"/>
> <action application="set" data="effective_caller_id_number=+777777777777"/>
> Le 08/12/2010 18:02, ?????? ???????? a ?crit :
>
> Hello.
>
> Can I change Caller_id_name & caller_id_number in freeswitch during
> transfer?
>
>
>
>
> ? ?????????
> ?????? ????????
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <http://sentmsg?compose&To=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> --
> Meftah Tayeb
> inum: +883510001288000
> Phone: +13602276297
> Fax: +12538020313
>
>
>
> ? ?????????
> ?????? ????????
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Dec 10 00:30:11 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 9 Dec 2010 15:30:11 -0600
Subject: [Freeswitch-users] User Interface Programmer Job Opportunity
Message-ID: <AANLkTim6d5tTO6J=XVowAGcKCAOh6cmLTQjddyoCmUUM@mail.gmail.com>

If anyone who lives in the US especially [MI WI OK CA] or would like
to move to one of those places and is good at UI design.
Contact jobs at freeswitch.org with a resume.
We need some programmers to work with us on the CudaTel UI.

Its a demanding yet rewarding job in software development and you get
to help with FreeSWITCH as well.

Jquery/Js/Comet type skills are the most important.


-- 
Anthony Minessale II

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

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

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


From mayamatakeshi at gmail.com  Fri Dec 10 01:05:39 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Fri, 10 Dec 2010 07:05:39 +0900
Subject: [Freeswitch-users] Refusing REFER requests
In-Reply-To: <AANLkTiks3YQm3Cj=M-WFYv+dA5g9QE4EJeyQmGEqE56O@mail.gmail.com>
References: <AANLkTiks3YQm3Cj=M-WFYv+dA5g9QE4EJeyQmGEqE56O@mail.gmail.com>
Message-ID: <AANLkTi=x2rsehx6obU2dZnHXoqnOKbqzkp0uT_9Nn5fs@mail.gmail.com>

On Thu, Dec 9, 2010 at 11:11 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

> Is there any way to ask FS to refuse a REFER request?
> I mean, when a REFER is received, the first thing FS does is to disconnect
> the channel of the referrer. Then, it sends the other channel to the
> dialplan. But I want to prevent the unbridging of the call in some
> circumstances (meaning: some destinations for transfer are not valid).
> Obs: I don't want do disable support for REFER as I know it can be
> configured in the sofia profile.
>

Thanks for the answers.
I checked the mod_sofia code (sofia.c) and there is no way to do this.
But I can see that if I really needed it, I can check the URI in the REFER
and refuse the call at mod_sofia level without completely disabling
transfer.

br,
takeshi
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From brian at freeswitch.org  Fri Dec 10 01:08:47 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 9 Dec 2010 16:08:47 -0600
Subject: [Freeswitch-users] Refusing REFER requests
In-Reply-To: <AANLkTi=x2rsehx6obU2dZnHXoqnOKbqzkp0uT_9Nn5fs@mail.gmail.com>
References: <AANLkTiks3YQm3Cj=M-WFYv+dA5g9QE4EJeyQmGEqE56O@mail.gmail.com>
	<AANLkTi=x2rsehx6obU2dZnHXoqnOKbqzkp0uT_9Nn5fs@mail.gmail.com>
Message-ID: <6C9B5526-88B9-4AB5-B1A2-36A4751751FB@freeswitch.org>

Is this thing on?  Did you not see my email where I said use disable-transfer option on the profile?

/b

On Dec 9, 2010, at 4:05 PM, mayamatakeshi wrote:

> 
> On Thu, Dec 9, 2010 at 11:11 PM, mayamatakeshi <mayamatakeshi at gmail.com> wrote:
> Is there any way to ask FS to refuse a REFER request?
> I mean, when a REFER is received, the first thing FS does is to disconnect the channel of the referrer. Then, it sends the other channel to the dialplan. But I want to prevent the unbridging of the call in some circumstances (meaning: some destinations for transfer are not valid).
> Obs: I don't want do disable support for REFER as I know it can be configured in the sofia profile.
> 
> Thanks for the answers.
> I checked the mod_sofia code (sofia.c) and there is no way to do this. 
> But I can see that if I really needed it, I can check the URI in the REFER and refuse the call at mod_sofia level without completely disabling transfer.
> 
> br,
> takeshi

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From covici at ccs.covici.com  Fri Dec 10 01:10:21 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 09 Dec 2010 17:10:21 -0500
Subject: [Freeswitch-users]
 =?koi8-r?b?Y2hhbmdlINNhbGxlcl9pZF9uYW1lICYg?= =?koi8-r?b?bnVtYmVy?=
In-Reply-To: <AANLkTi=OBmDKBwmx3p0z-RQDa8bOsW3jeHTiKPPxO-cD@mail.gmail.com>
References: <E1PQNPD-0002tm-00.melkybes-mail-ru@f160.mail.ru>
	<4CFFBD23.6010608@gmail.com>
	<E1PQgb0-0001QJ-00.melkybes-mail-ru@f252.mail.ru>
	<AANLkTi=OBmDKBwmx3p0z-RQDa8bOsW3jeHTiKPPxO-cD@mail.gmail.com>
Message-ID: <25600.1291932621@ccs.covici.com>

I don't see a logger.pl in that directory.  I have used script to
capture such things, except that it puts color escape codes in.

Michael Collins <msc at freeswitch.org> wrote:

> On Thu, Dec 9, 2010 at 5:31 AM, ?????? ???????? <melkybes at mail.ru> wrote:
> 
> >
> > Don't work  :(
> >
> Please supply more information. Best thing to do is to use pastebin (
> pastebin.freeswitch.org)
> Things to put in there:
> Your setup - what phones are you using and what kind of telephone service?
> (VoIP, PRI, analog, etc.)
> Your dialplan for this extension
> The console debug for this call
> If you don't have a lot of traffic you can also include a SIP trace
> (assuming you are using SIP)
> At the console do "sofia global siptrace on"
> NOTE: the siptrace goes to the console but not to the log file, so you need
> to capture the output at the console or use the logger.pl script in
> /scripts/perl under the fs source dir.
> 
> Put the pastebin URL in this thread.
> 
> Also, check this page for helpful tips on collecting information for
> debugging:
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
> 
> -MC
> 
> >
> >
> >
> >
> >  <action application="set" data="effective_caller_id_name=russia"/>
> > <action application="set" data="effective_caller_id_number=+777777777777"/>
> > Le 08/12/2010 18:02, ?????? ???????? a ?crit :
> >
> > Hello.
> >
> > Can I change Caller_id_name & caller_id_number in freeswitch during
> > transfer?
> >
> >
> >
> >
> > ? ?????????
> > ?????? ????????
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org <http://sentmsg?compose&To=FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > --
> > Meftah Tayeb
> > inum: +883510001288000
> > Phone: +13602276297
> > Fax: +12538020313
> >
> >
> >
> > ? ?????????
> > ?????? ????????
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From johnrose at comtex.net  Fri Dec 10 00:09:09 2010
From: johnrose at comtex.net (John Rose)
Date: Thu, 9 Dec 2010 14:09:09 -0700
Subject: [Freeswitch-users] SIP MESSAGE requests
In-Reply-To: <AANLkTin9wJP-C-6q1KRh87RVFBztJbP5pTM9f-hYra6H@mail.gmail.com>
References: <004201cb9662$d770dad0$86529070$@comtex.net>	<4CFF3EC7.5030605@gmail.com>	<AANLkTin3gep6cpQoV0b4NEV2Tqgxe6LGyVJE5ymu46NA@mail.gmail.com>	<B6823144-18AC-42BA-BF9E-9027255A45D0@avgs.ca>	<AANLkTikP1qHa4-PtMjseJHoDxvJ5Fo7a5mfWa2aTA-Kc@mail.gmail.com>	<9B0E926C-3238-4388-B135-9F1C7899EEF8@avgs.ca>
	<AANLkTin9wJP-C-6q1KRh87RVFBztJbP5pTM9f-hYra6H@mail.gmail.com>
Message-ID: <000201cb97e5$52fa9550$f8efbff0$@comtex.net>

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner
> 
> Mathieu,
> 
>   This is true for MESSAGE and I understand it may be a little hairy...
> However, SIMPLE (uh-oh) also allows for IM-like message exchange using a
> real session established with INVITE.  As the OP said MESSAGE is used for
> paging (SMS) type applications.  I understand that creating a channel may
be
> a bit overblown for this but something like that would allow for much more
> flexibility.
> 


Well thanks for the help on this y'all, I'm going to test using managed ESL
to receive the MESSAGE generated events once I try sending some...

John



From pbdlists at pinboard.com  Thu Dec  9 22:57:30 2010
From: pbdlists at pinboard.com (pbdlists at pinboard.com)
Date: Thu, 9 Dec 2010 20:57:30 +0100
Subject: [Freeswitch-users] fax files not saved
In-Reply-To: <EC54D2EA08B14F8BA13A0E50C94B1826@e1705>
References: <20101208153025.GB17099@pinboard.com>
	<EC54D2EA08B14F8BA13A0E50C94B1826@e1705>
Message-ID: <20101209195730.GA11419@pinboard.com>

Oops, didn't catch the double underbar (the called script is not
evaluating that parameter). Thanks, fixed it.

You got me puzzled with your remark about session_in_hangup_hook. Had
to look it up on the wiki. My guess is that with session_in_hangup_hook
the variables get exported into the environment of the called
script/application and can be queried there directly. But I'm calling
the script with all the variables as parameters; the called script is
not evaluating the environment, but the command line parameters it was
passed.

Meanwhile, anybody any idea about why the fax file is only written to
disk sometimes?

Cheers,

Kurt

On Wed, Dec 08, 2010 at 11:30:34AM -0500, Madovsky wrote:
> you have first a typo
> fax__image_resolution
> 
> the thing I don't understand is you didn't
> set session_in hangup_hook=true and
> you can retrieve the channel vars at api_hangup_hook...
> 
> 
> 
> ----- Original Message ----- 
> From: <pbdlists at pinboard.com>
> To: "freeswitch-users" <FreeSWITCH-users at lists.freeswitch.org>
> Sent: Wednesday, December 08, 2010 10:30 AM
> Subject: [Freeswitch-users] fax files not saved
> 
> 
> > Hello!
> >
> > I managed to solve most of my other problems sofar, but with this one I
> > have no clue at all. Has anybody an idea what is happening here?
> >
> > 1 an inbound call to a number which is registered as fax (mod_spandsp)
> > 2 fax is detected and received according to the log (log entries see 
> > below)
> > 3 but _sometimes_ the tiff file is never written to disk (post processing
> >   script can't find it and it is really not there)
> >
> > The destination filesystem has enough free space and free inodes,
> > permissions for writing are ok, the system is not under any kind of
> > heavy load, no other calls going on at the same time...
> >
> > I don't have much data to test this (using www.freepopfax.com for
> > testing and they have a daily limit on faxes I can send), but it seems
> > to happen roughly for 30-50% of the incoming fax messages. From the same
> > fax provider it works fine one time, then not a couple of minutes later,
> > another couple of minutes later it may work again, all while nothing at
> > all is being changed on the freeswitch server. If it was something like
> > a dropped line or anything I could understand it, but the logs say
> > everything is ok and still _sometimes_ the fax is not written to disk...
> >
> > Cheers,
> >
> > Kurt
> >
> > =======================================================================
> > entries in fax.conf.xml
> > =======================================================================
> > <configuration name="fax.conf" description="FAX application 
> > configuration">
> >    <settings>
> >        <param name="use-ecm"           value="true"/>
> >        <param name="verbose"           value="false"/>
> >        <param name="disable-v17"       value="false"/>
> >        <param name="ident"             value="SpanDSP Fax Ident"/>
> >        <param name="header"            value="SpanDSP Fax Header"/>
> >
> >        <param name="spool-dir"         value="/tmp"/>
> >        <param name="file-prefix"       value="faxrx"/>
> >    </settings>
> > </configuration>
> >
> >
> > =======================================================================
> > the part from default.xml
> > =======================================================================
> >    <extension name="fax_receive">
> >        <condition field="destination_number" expression="^(1111)$">
> >            <action application="set" data="api_hangup_hook=system 
> > ${base_dir}/scripts/emailfax.sh $1 /tmp/FAX-${uuid}.tiff 
> > '\\\${fax_success}' '\\\${fax_result_code}' '\\\${fax_result_text}' 
> > '\\\${fax_local_station_id}' '\\\${fax_remote_station_id}' 
> > '\\\${fax_document_transferred_pages}' '\\\${fax_document_total_pages}' 
> > '\\\${fax__image_resolution}' '\\\${fax_image_size}' '\\\${fax_bad_rows}' 
> > '\\\${fax_transfer_rate}' '\\\${fax_v17_disabled}' 
> > '\\\${fax_ecm_requested}' '\\\${fax_filename}' "/>
> >            <action application="answer" />
> >            <action application="playback" data="silence_stream://2000"/>
> > <!--
> >            <action application="set" data="fax_enable_t38_request=true"/>
> >            <action application="set" data="fax_enable_t38=true"/>
> > -->
> >            <action application="set" data="fax_ident=+41xxxxxxxxx"/>
> >            <action application="set" data="fax_disable_v17=1"/>
> >            <action application="set" data="fax_verbose=0"/>
> >            <action application="rxfax" data="/tmp/FAX-${uuid}.tiff"/>
> >            <action application="hangup"/>
> >        </condition>
> >    </extension>
> >
> >
> > =======================================================================
> > log entry from an unsuccessful incoming fax:
> > =======================================================================
> >
> > 2010-12-07 23:33:51.157588 [DEBUG] sofia_glue.c:182 
> > sofia/external/Anonymous at anonymous.invalid image media sdp:
> > v=0
> > o=FreeSWITCH 1291733462 1291733465 IN IP4 xxx.xxx.xxx.xxx
> > s=FreeSWITCH
> > c=IN IP4 xxx.xxx.xxx.xxx
> > t=0 0
> > m=image 31358 udptl t38
> > a=T38FaxVersion:0
> > a=T38MaxBitRate:9600
> > a=T38FaxFillBitRemoval
> > a=T38FaxRateManagement:transferredTCF
> > a=T38FaxMaxBuffer:2000
> > a=T38FaxMaxDatagram:400
> > a=T38FaxUdpEC:t38UDPRedundancy
> >
> > 2010-12-07 23:33:51.157588 [DEBUG] sofia.c:4597 Channel 
> > sofia/external/Anonymous at anonymous.invalid entering state [completed][200]
> > 2010-12-07 23:33:51.157588 [DEBUG] sofia.c:4597 Channel 
> > sofia/external/Anonymous at anonymous.invalid entering state [ready][200]
> > 2010-12-07 23:33:58.814979 [DEBUG] switch_rtp.c:2544 Correct ip/port 
> > confirmed.
> > 2010-12-07 23:35:09.772752 [DEBUG] switch_channel.c:2457 
> > (sofia/external/Anonymous at anonymous.invalid) Callstate Change ACTIVE -> 
> > HANGUP
> > 2010-12-07 23:35:09.772752 [NOTICE] sofia.c:528 Hangup 
> > sofia/external/Anonymous at anonymous.invalid [CS_EXECUTE] [NORMAL_CLEARING]
> > 2010-12-07 23:35:09.772752 [DEBUG] switch_channel.c:2473 Send signal 
> > sofia/external/Anonymous at anonymous.invalid [KILL]
> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_session.c:1057 Send signal 
> > sofia/external/Anonymous at anonymous.invalid [BREAK]
> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:323 
> > ==============================================================================
> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:329 Fax successfully 
> > received.
> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:340 Remote station 
> > id:
> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:341 Local station id: 
> > +41xxxxxxxxx
> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:342 Pages 
> > transferred: 2
> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:344 Total fax pages: 
> > 2
> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:345 Image resolution: 
> > 8031x7700
> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate: 
> > 9600
> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:348 ECM status 
> > on
> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:349 remote country:
> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:350 remote vendor:
> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:351 remote model:
> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:353 
> > ==============================================================================
> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_codec.c:141 
> > sofia/external/Anonymous at anonymous.invalid Restore previous codec PCMA:8.
> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_session.c:1933 
> > sofia/external/Anonymous at anonymous.invalid skip receive message 
> > [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:366 
> > (sofia/external/Anonymous at anonymous.invalid) State EXECUTE going to sleep
> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:320 
> > (sofia/external/Anonymous at anonymous.invalid) Running State Change 
> > CS_HANGUP
> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:553 
> > (sofia/external/Anonymous at anonymous.invalid) State HANGUP
> > 2010-12-07 23:35:09.772752 [DEBUG] mod_sofia.c:453 
> > sofia/external/Anonymous at anonymous.invalid Overriding SIP cause 480 with 
> > 200 from the other leg
> > 2010-12-07 23:35:09.772752 [DEBUG] mod_sofia.c:459 Channel 
> > sofia/external/Anonymous at anonymous.invalid hanging up, cause: 
> > NORMAL_CLEARING
> > 2010-12-07 23:35:09.821455 [DEBUG] switch_core_state_machine.c:46 
> > sofia/external/Anonymous at anonymous.invalid Standard HANGUP, cause: 
> > NORMAL_CLEARING
> > 2010-12-07 23:35:09.821455 [DEBUG] switch_core_state_machine.c:553 
> > (sofia/external/Anonymous at anonymous.invalid) State HANGUP going to sleep
> > 2010-12-07 23:35:09.822803 [NOTICE] mod_commands.c:4187 Executing command: 
> > /usr/local/freeswitch/scripts/emailfax.sh 1111 
> > /tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff '1' '0' 'OK' 
> > '+41xxxxxxxxx' '' '2' '2
> > ' '' '0' '0' '9600' '1' '1' 
> > '/tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff'
> > 2010-12-07 23:35:10.076623 [DEBUG] switch_core_state_machine.c:488 Hangup 
> > Command with no Session system(/usr/local/freeswitch/scripts/emailfax.sh 
> > 1111 /tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff '1' '0'
> > 'OK' '+41xxxxxxxxx' '' '2' '2' '' '0' '0' '9600' '1' '1' 
> > '/tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff' ):
> > +OK
> >
> >
> > =======================================================================
> > log entry from a successful incoming fax:
> > =======================================================================
> >
> > 2010-12-07 23:54:59.082301 [DEBUG] sofia_glue.c:182 
> > sofia/external/Anonymous at anonymous.invalid image media sdp:
> > v=0
> > o=FreeSWITCH 1291735801 1291735804 IN IP4 xxx.xxx.xxx.xxx
> > s=FreeSWITCH
> > c=IN IP4 xxx.xxx.xxx.xxx
> > t=0 0
> > m=image 30280 udptl t38
> > a=T38FaxVersion:0
> > a=T38MaxBitRate:9600
> > a=T38FaxFillBitRemoval
> > a=T38FaxRateManagement:transferredTCF
> > a=T38FaxMaxBuffer:2000
> > a=T38FaxMaxDatagram:400
> > a=T38FaxUdpEC:t38UDPRedundancy
> >
> > 2010-12-07 23:54:59.082301 [DEBUG] sofia.c:4597 Channel 
> > sofia/external/Anonymous at anonymous.invalid entering state [completed][200]
> > 2010-12-07 23:54:59.100811 [DEBUG] sofia.c:4597 Channel 
> > sofia/external/Anonymous at anonymous.invalid entering state [ready][200]
> > 2010-12-07 23:55:06.783427 [DEBUG] switch_rtp.c:2544 Correct ip/port 
> > confirmed.
> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:323 
> > ==============================================================================
> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:329 Fax successfully 
> > received.
> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:340 Remote station 
> > id:
> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:341 Local station id: 
> > +41xxxxxxxxx
> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:342 Pages 
> > transferred: 1
> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:344 Total fax pages: 
> > 1
> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:345 Image resolution: 
> > 8031x7700
> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate: 
> > 9600
> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:348 ECM status 
> > on
> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:349 remote country:
> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:350 remote vendor:
> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:351 remote model:
> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:353 
> > ==============================================================================
> > 2010-12-07 23:55:31.616038 [DEBUG] switch_channel.c:2457 
> > (sofia/external/Anonymous at anonymous.invalid) Callstate Change ACTIVE -> 
> > HANGUP
> > 2010-12-07 23:55:31.616038 [NOTICE] sofia.c:528 Hangup 
> > sofia/external/Anonymous at anonymous.invalid [CS_EXECUTE] [NORMAL_CLEARING]
> > 2010-12-07 23:55:31.616038 [DEBUG] switch_channel.c:2473 Send signal 
> > sofia/external/Anonymous at anonymous.invalid [KILL]
> > 2010-12-07 23:55:31.616038 [DEBUG] switch_core_session.c:1057 Send signal 
> > sofia/external/Anonymous at anonymous.invalid [BREAK]
> > 2010-12-07 23:55:31.617356 [DEBUG] switch_core_codec.c:141 
> > sofia/external/Anonymous at anonymous.invalid Restore previous codec PCMA:8.
> > 2010-12-07 23:55:31.617356 [DEBUG] switch_core_session.c:1933 
> > sofia/external/Anonymous at anonymous.invalid skip receive message 
> > [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> > 2010-12-07 23:55:31.617356 [DEBUG] switch_core_state_machine.c:366 
> > (sofia/external/Anonymous at anonymous.invalid) State EXECUTE going to sleep
> > 2010-12-07 23:55:31.617356 [DEBUG] switch_core_state_machine.c:320 
> > (sofia/external/Anonymous at anonymous.invalid) Running State Change 
> > CS_HANGUP
> > 2010-12-07 23:55:31.618542 [DEBUG] switch_core_state_machine.c:553 
> > (sofia/external/Anonymous at anonymous.invalid) State HANGUP
> > 2010-12-07 23:55:31.618542 [DEBUG] mod_sofia.c:453 
> > sofia/external/Anonymous at anonymous.invalid Overriding SIP cause 480 with 
> > 200 from the other leg
> > 2010-12-07 23:55:31.618542 [DEBUG] mod_sofia.c:459 Channel 
> > sofia/external/Anonymous at anonymous.invalid hanging up, cause: 
> > NORMAL_CLEARING
> > 2010-12-07 23:55:31.721993 [DEBUG] switch_core_state_machine.c:46 
> > sofia/external/Anonymous at anonymous.invalid Standard HANGUP, cause: 
> > NORMAL_CLEARING
> > 2010-12-07 23:55:31.721993 [DEBUG] switch_core_state_machine.c:553 
> > (sofia/external/Anonymous at anonymous.invalid) State HANGUP going to sleep
> > 2010-12-07 23:55:31.721993 [NOTICE] mod_commands.c:4187 Executing command: 
> > /usr/local/freeswitch/scripts/emailfax.sh 1111 
> > /tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff '1' '0' 'OK' 
> > '+41xxxxxxxxx' '' '1' '1' '' '0' '0' '9600' '1' '1' 
> > '/tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff'
> > 2010-12-07 23:55:31.904090 [DEBUG] switch_core_state_machine.c:488 Hangup 
> > Command with no Session system(/usr/local/freeswitch/scripts/emailfax.sh 
> > 1111 /tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff '1' '0' 'OK' 
> > '+41xxxxxxxxx' '' '1' '1' '' '0' '0' '9600' '1' '1' 
> > '/tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff' ):
> > +OK
> >


From mayamatakeshi at gmail.com  Fri Dec 10 01:25:49 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Fri, 10 Dec 2010 07:25:49 +0900
Subject: [Freeswitch-users] Refusing REFER requests
In-Reply-To: <6C9B5526-88B9-4AB5-B1A2-36A4751751FB@freeswitch.org>
References: <AANLkTiks3YQm3Cj=M-WFYv+dA5g9QE4EJeyQmGEqE56O@mail.gmail.com>
	<AANLkTi=x2rsehx6obU2dZnHXoqnOKbqzkp0uT_9Nn5fs@mail.gmail.com>
	<6C9B5526-88B9-4AB5-B1A2-36A4751751FB@freeswitch.org>
Message-ID: <AANLkTi=mOgm8fWA+F+JvSDDc9kNntXE6OygWe4=cHHSa@mail.gmail.com>

On Fri, Dec 10, 2010 at 7:08 AM, Brian West <brian at freeswitch.org> wrote:

> Is this thing on?  Did you not see my email where I said
> use disable-transfer option on the profile?
>

Yes, but I don't want to refuse all REFER request (that's what
disable-transfer do).
I only want to refuse some REFER requests when I detect the destination is
invalid.


>
>
> On Dec 9, 2010, at 4:05 PM, mayamatakeshi wrote:
>
>
> On Thu, Dec 9, 2010 at 11:11 PM, mayamatakeshi <mayamatakeshi at gmail.com>
> wrote:
>
>> Is there any way to ask FS to refuse a REFER request?
>> I mean, when a REFER is received, the first thing FS does is to disconnect
>> the channel of the referrer. Then, it sends the other channel to the
>> dialplan. But I want to prevent the unbridging of the call in some
>> circumstances (meaning: some destinations for transfer are not valid).
>> Obs: I don't want do disable support for REFER as I know it can be
>> configured in the sofia profile.
>>
>
> Thanks for the answers.
> I checked the mod_sofia code (sofia.c) and there is no way to do this.
> But I can see that if I really needed it, I can check the URI in the REFER
> and refuse the call at mod_sofia level without completely disabling
> transfer.
>
> br,
> takeshi
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Fri Dec 10 01:32:50 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 9 Dec 2010 16:32:50 -0600
Subject: [Freeswitch-users] Refusing REFER requests
In-Reply-To: <AANLkTi=mOgm8fWA+F+JvSDDc9kNntXE6OygWe4=cHHSa@mail.gmail.com>
References: <AANLkTiks3YQm3Cj=M-WFYv+dA5g9QE4EJeyQmGEqE56O@mail.gmail.com>
	<AANLkTi=x2rsehx6obU2dZnHXoqnOKbqzkp0uT_9Nn5fs@mail.gmail.com>
	<6C9B5526-88B9-4AB5-B1A2-36A4751751FB@freeswitch.org>
	<AANLkTi=mOgm8fWA+F+JvSDDc9kNntXE6OygWe4=cHHSa@mail.gmail.com>
Message-ID: <56E80050-72E6-4960-866E-7B20E42B0C06@freeswitch.org>

TAG_IF((profile->mflags & MFLAG_REFER), NUTAG_ALLOW("REFER")),

This removes refer from the allowed methods in sofia so we will say that the method is NOT allowed.

If you want to refuse when the destination is invalid then give up now because you will need to know before the refer is sent to know that.

/b

On Dec 9, 2010, at 4:25 PM, mayamatakeshi wrote:

> 
> On Fri, Dec 10, 2010 at 7:08 AM, Brian West <brian at freeswitch.org> wrote:
> Is this thing on?  Did you not see my email where I said use disable-transfer option on the profile?
> 
> Yes, but I don't want to refuse all REFER request (that's what disable-transfer do).
> I only want to refuse some REFER requests when I detect the destination is invalid.
>  
> 
> 
> On

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From mayamatakeshi at gmail.com  Fri Dec 10 01:49:29 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Fri, 10 Dec 2010 07:49:29 +0900
Subject: [Freeswitch-users] Refusing REFER requests
In-Reply-To: <56E80050-72E6-4960-866E-7B20E42B0C06@freeswitch.org>
References: <AANLkTiks3YQm3Cj=M-WFYv+dA5g9QE4EJeyQmGEqE56O@mail.gmail.com>
	<AANLkTi=x2rsehx6obU2dZnHXoqnOKbqzkp0uT_9Nn5fs@mail.gmail.com>
	<6C9B5526-88B9-4AB5-B1A2-36A4751751FB@freeswitch.org>
	<AANLkTi=mOgm8fWA+F+JvSDDc9kNntXE6OygWe4=cHHSa@mail.gmail.com>
	<56E80050-72E6-4960-866E-7B20E42B0C06@freeswitch.org>
Message-ID: <AANLkTi=s4pXFSc+anKUPW37jdfgaF2O51PRedEDZ7Q+N@mail.gmail.com>

On Fri, Dec 10, 2010 at 7:32 AM, Brian West <brian at freeswitch.org> wrote:

> TAG_IF((profile->mflags & MFLAG_REFER), NUTAG_ALLOW("REFER")),
>
> This removes refer from the allowed methods in sofia so we will say that
> the method is NOT allowed.
>
> If you want to refuse when the destination is invalid then give up now
> because you will need to know before the refer is sent to know that.
>

I am not sure what you mean.
I was thinking in adding a new parameter for mod_sofia like
"forbidden_transfer_target" and add code in sofia.c right after that MFLAG
check:

if (!(profile->mflags & MFLAG_REFER)) {
  nua_respond(nh, SIP_403_FORBIDDEN, NUTAG_WITH_THIS(nua), TAG_END());
  goto done;
}

// my pseudo-code:

if (REFER_URI =~ forbidded_transfer_target) {
  nua_respond(nh, SIP_403_FORBIDDEN, NUTAG_WITH_THIS(nua), TAG_END());
  goto done;
}
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From msc at freeswitch.org  Fri Dec 10 02:52:31 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 9 Dec 2010 15:52:31 -0800
Subject: [Freeswitch-users] =?gb2312?b?Y2hhbmdlIKfjYWxsZXJfaWRfbmFtZSAm?=
	=?gb2312?b?IG51bWJlcg==?=
In-Reply-To: <25600.1291932621@ccs.covici.com>
References: <E1PQNPD-0002tm-00.melkybes-mail-ru@f160.mail.ru>
	<4CFFBD23.6010608@gmail.com>
	<E1PQgb0-0001QJ-00.melkybes-mail-ru@f252.mail.ru>
	<AANLkTi=OBmDKBwmx3p0z-RQDa8bOsW3jeHTiKPPxO-cD@mail.gmail.com>
	<25600.1291932621@ccs.covici.com>
Message-ID: <AANLkTik_EwZ1eqjfwd0ahc=0DtEyyWg8Muv8Ay1+J7tp@mail.gmail.com>

2010/12/9 <covici at ccs.covici.com>

> I don't see a logger.pl in that directory.  I have used script to
> capture such things, except that it puts color escape codes in.
>
> Doh!

It's in libs/esl/perl/logger.pl

My bad.
-MC
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From msc at freeswitch.org  Fri Dec 10 02:53:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 9 Dec 2010 15:53:32 -0800
Subject: [Freeswitch-users] Email Voicemail Volume Mystery
In-Reply-To: <A3633C87-61A4-41BB-B6BD-B9477F9AB5DB@mgtech.com>
References: <A3633C87-61A4-41BB-B6BD-B9477F9AB5DB@mgtech.com>
Message-ID: <AANLkTinjBY9Qb6gWUXJQMrp6_FX3Bj2WSOsZ4cJw-ODo@mail.gmail.com>

What format are you using?
-MC

On Thu, Dec 9, 2010 at 9:56 AM, Mario G <mario_fs at mgtech.com> wrote:

> This is consistent and baffling.This volume for emailed voicemails is not
> the same for different callers. This is an ONLY and issue for emailed
> voicemails, listening locally is fine. Whether the volume is good or bad is
> always consistent for a given caller.
>
> For all cell phone callers so far:
> The call volume AND local voicemail volume is good AND the emailed
> voicemail volume is good.
>
> For almost all landline callers
> The call volume AND local voicemail volume is good BUT the emailed
> voicemail volume is so low that it's unusable unless you are in a dead quite
> room.
>
> The problem is NOT the device you listening on since I tested the file with
> an audio analyzer and the it's clear for the bad ones the volume is about
> 1/5 normal.
>
> I could not find anything to try, any ideas?
>
> Mario G
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mario_fs at mgtech.com  Fri Dec 10 03:12:30 2010
From: mario_fs at mgtech.com (Mario G)
Date: Thu, 9 Dec 2010 16:12:30 -0800
Subject: [Freeswitch-users] Email Voicemail Volume Mystery
In-Reply-To: <AANLkTinjBY9Qb6gWUXJQMrp6_FX3Bj2WSOsZ4cJw-ODo@mail.gmail.com>
References: <A3633C87-61A4-41BB-B6BD-B9477F9AB5DB@mgtech.com>
	<AANLkTinjBY9Qb6gWUXJQMrp6_FX3Bj2WSOsZ4cJw-ODo@mail.gmail.com>
Message-ID: <ACC02B70-651D-47D5-9FC2-36662EC7A5F8@mgtech.com>

Aiff because the iPhone will no play all formats, but I also tried mp3 to see if it made any difference and it did not. Both types of files had the same very low volume.
MG

On Dec 9, 2010, at 3:53 PM, Michael Collins wrote:

> What format are you using?
> -MC
> 
> On Thu, Dec 9, 2010 at 9:56 AM, Mario G <mario_fs at mgtech.com> wrote:
> This is consistent and baffling.This volume for emailed voicemails is not the same for different callers. This is an ONLY and issue for emailed voicemails, listening locally is fine. Whether the volume is good or bad is always consistent for a given caller.
> 
> For all cell phone callers so far:
> The call volume AND local voicemail volume is good AND the emailed voicemail volume is good.
> 
> For almost all landline callers
> The call volume AND local voicemail volume is good BUT the emailed voicemail volume is so low that it's unusable unless you are in a dead quite room.
> 
> The problem is NOT the device you listening on since I tested the file with an audio analyzer and the it's clear for the bad ones the volume is about 1/5 normal.
> 
> I could not find anything to try, any ideas?
> 
> Mario G
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From Nabble at slickdeals.endjunk.com  Fri Dec 10 03:37:51 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 9 Dec 2010 16:37:51 -0800 (PST)
Subject: [Freeswitch-users] Disabling Google Voice Voicemail
In-Reply-To: <AANLkTinndUGCzh+gsoyZLjQ8LR+phQLuBMsW1LvMSH4+@mail.gmail.com>
References: <1291903360189-5818816.post@n2.nabble.com>
	<AANLkTim_bp9Xk94R6uW2QJmp1dVPKOL8rxjhLS7s_K99@mail.gmail.com>
	<1291923916094-5820341.post@n2.nabble.com>
	<AANLkTinndUGCzh+gsoyZLjQ8LR+phQLuBMsW1LvMSH4+@mail.gmail.com>
Message-ID: <1291941471978-5821494.post@n2.nabble.com>



Avi Marcus wrote:
> 
> <action application="answer"/>
> <action application="send_dtmf" data="1"/> (if needed for call screening)
> 
> Hmm, the ringing part is harder than I thought. I don't think ring_ready
> will work...
Thanks for the above sample. I injected it in the dialplan, but it doesn't
work. In other words, the call went to GV voicemail after the 4th ring. 


Well try that, if not, maybe a transfer or something after - see
> http://wiki.freeswitch.org/wiki/Dialplan_FollowMe#FollowMe_Dialplan_Example_3
Thanls for the above link. One of the examples shows how to implement the
call timeout as suggested by the 
http://freeswitch-users.2379917.n2.nabble.com/Disabling-Google-Voice-Voicemail-tp5818816p5819684.html
above post . And, this too doesn't work, unfortunately. 

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Disabling-Google-Voice-Voicemail-tp5818816p5821494.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From david.ponzone at ipeva.fr  Fri Dec 10 04:36:12 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 10 Dec 2010 02:36:12 +0100
Subject: [Freeswitch-users]
 =?utf-8?b?Y2hhbmdlINGBYWxsZXJfaWRfbmFtZSAmIG51?= =?utf-8?q?mber?=
In-Reply-To: <25600.1291932621@ccs.covici.com>
References: <E1PQNPD-0002tm-00.melkybes-mail-ru@f160.mail.ru>
	<4CFFBD23.6010608@gmail.com>
	<E1PQgb0-0001QJ-00.melkybes-mail-ru@f252.mail.ru>
	<AANLkTi=OBmDKBwmx3p0z-RQDa8bOsW3jeHTiKPPxO-cD@mail.gmail.com>
	<25600.1291932621@ccs.covici.com>
Message-ID: <55A2F129-618B-42BD-BB6D-E8271164FC6D@ipeva.fr>

find . -name logger.pl

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 09/12/2010 ? 23:10, covici at ccs.covici.com a ?crit :

> I don't see a logger.pl in that directory.  I have used script to
> capture such things, except that it puts color escape codes in.
> 
> Michael Collins <msc at freeswitch.org> wrote:
> 
>> On Thu, Dec 9, 2010 at 5:31 AM, ?????? ???????? <melkybes at mail.ru> wrote:
>> 
>>> 
>>> Don't work  :(
>>> 
>> Please supply more information. Best thing to do is to use pastebin (
>> pastebin.freeswitch.org)
>> Things to put in there:
>> Your setup - what phones are you using and what kind of telephone service?
>> (VoIP, PRI, analog, etc.)
>> Your dialplan for this extension
>> The console debug for this call
>> If you don't have a lot of traffic you can also include a SIP trace
>> (assuming you are using SIP)
>> At the console do "sofia global siptrace on"
>> NOTE: the siptrace goes to the console but not to the log file, so you need
>> to capture the output at the console or use the logger.pl script in
>> /scripts/perl under the fs source dir.
>> 
>> Put the pastebin URL in this thread.
>> 
>> Also, check this page for helpful tips on collecting information for
>> debugging:
>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>> 
>> -MC
>> 
>>> 
>>> 
>>> 
>>> 
>>> <action application="set" data="effective_caller_id_name=russia"/>
>>> <action application="set" data="effective_caller_id_number=+777777777777"/>
>>> Le 08/12/2010 18:02, ?????? ???????? a ?crit :
>>> 
>>> Hello.
>>> 
>>> Can I change Caller_id_name & caller_id_number in freeswitch during
>>> transfer?
>>> 
>>> 
>>> 
>>> 
>>> ? ?????????
>>> ?????? ????????
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <http://sentmsg?compose&To=FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> --
>>> Meftah Tayeb
>>> inum: +883510001288000
>>> Phone: +13602276297
>>> Fax: +12538020313
>>> 
>>> 
>>> 
>>> ? ?????????
>>> ?????? ????????
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>> 
>> ----------------------------------------------------
>> Alternatives:
>> 
>> ----------------------------------------------------
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>         John Covici
>         covici at ccs.covici.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From fraserredmond at gmail.com  Fri Dec 10 05:26:25 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Thu, 9 Dec 2010 21:26:25 -0500
Subject: [Freeswitch-users] No sip BYE from gateway at hangup
Message-ID: <AANLkTi=qjdZHgEbZG1WTUCATcPDE-t93m0V3-_-Bfthz@mail.gmail.com>

I've got a gateway that seems to be sending the sip BYE really late.

If the local (softphone) party hangs up then everything happens as normal as
I'd expect it to. If the end terminated through the gateway hangs up then
it's another 15-40 sec before Freeswitch finds out it's hung up and
processes the end of the call.

I did a "sofia profile external siptrace on" and during that 15-40 sec there
was no activity until the end when the BYE turned up.

I had the gateway run a sip trace their end and they said everything looked
normal to them, and they said:
"We have reviewed all of the calls specified both on the SIP signaling side
and the SS7 side.  Under certain circumstances, some calls can take longer
to fully tear from the PSTN than others.  "
Which I interpret as a polite shrug of the shoulders :-)

Any ideas of what I could look at? Or is it just one of those things that
happens?

The other gateways I have registered don't have the same problem.

Cheers,
Fraser
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From infos at madovsky.org  Fri Dec 10 04:06:13 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 9 Dec 2010 20:06:13 -0500
Subject: [Freeswitch-users] fax files not saved
References: <20101208153025.GB17099@pinboard.com><EC54D2EA08B14F8BA13A0E50C94B1826@e1705>
	<20101209195730.GA11419@pinboard.com>
Message-ID: <603F45904E8D4922A0EAF575BAA327D9@e1705>

do you have any function or command that delete your fax after sent email ?
if not, maybe try to delay the hangup...


----- Original Message ----- 
From: <pbdlists at pinboard.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, December 09, 2010 2:57 PM
Subject: Re: [Freeswitch-users] fax files not saved


> Oops, didn't catch the double underbar (the called script is not
> evaluating that parameter). Thanks, fixed it.
>
> You got me puzzled with your remark about session_in_hangup_hook. Had
> to look it up on the wiki. My guess is that with session_in_hangup_hook
> the variables get exported into the environment of the called
> script/application and can be queried there directly. But I'm calling
> the script with all the variables as parameters; the called script is
> not evaluating the environment, but the command line parameters it was
> passed.
>
> Meanwhile, anybody any idea about why the fax file is only written to
> disk sometimes?
>
> Cheers,
>
> Kurt
>
> On Wed, Dec 08, 2010 at 11:30:34AM -0500, Madovsky wrote:
>> you have first a typo
>> fax__image_resolution
>>
>> the thing I don't understand is you didn't
>> set session_in hangup_hook=true and
>> you can retrieve the channel vars at api_hangup_hook...
>>
>>
>>
>> ----- Original Message ----- 
>> From: <pbdlists at pinboard.com>
>> To: "freeswitch-users" <FreeSWITCH-users at lists.freeswitch.org>
>> Sent: Wednesday, December 08, 2010 10:30 AM
>> Subject: [Freeswitch-users] fax files not saved
>>
>>
>> > Hello!
>> >
>> > I managed to solve most of my other problems sofar, but with this one I
>> > have no clue at all. Has anybody an idea what is happening here?
>> >
>> > 1 an inbound call to a number which is registered as fax (mod_spandsp)
>> > 2 fax is detected and received according to the log (log entries see
>> > below)
>> > 3 but _sometimes_ the tiff file is never written to disk (post 
>> > processing
>> >   script can't find it and it is really not there)
>> >
>> > The destination filesystem has enough free space and free inodes,
>> > permissions for writing are ok, the system is not under any kind of
>> > heavy load, no other calls going on at the same time...
>> >
>> > I don't have much data to test this (using www.freepopfax.com for
>> > testing and they have a daily limit on faxes I can send), but it seems
>> > to happen roughly for 30-50% of the incoming fax messages. From the 
>> > same
>> > fax provider it works fine one time, then not a couple of minutes 
>> > later,
>> > another couple of minutes later it may work again, all while nothing at
>> > all is being changed on the freeswitch server. If it was something like
>> > a dropped line or anything I could understand it, but the logs say
>> > everything is ok and still _sometimes_ the fax is not written to 
>> > disk...
>> >
>> > Cheers,
>> >
>> > Kurt
>> >
>> > =======================================================================
>> > entries in fax.conf.xml
>> > =======================================================================
>> > <configuration name="fax.conf" description="FAX application
>> > configuration">
>> >    <settings>
>> >        <param name="use-ecm"           value="true"/>
>> >        <param name="verbose"           value="false"/>
>> >        <param name="disable-v17"       value="false"/>
>> >        <param name="ident"             value="SpanDSP Fax Ident"/>
>> >        <param name="header"            value="SpanDSP Fax Header"/>
>> >
>> >        <param name="spool-dir"         value="/tmp"/>
>> >        <param name="file-prefix"       value="faxrx"/>
>> >    </settings>
>> > </configuration>
>> >
>> >
>> > =======================================================================
>> > the part from default.xml
>> > =======================================================================
>> >    <extension name="fax_receive">
>> >        <condition field="destination_number" expression="^(1111)$">
>> >            <action application="set" data="api_hangup_hook=system
>> > ${base_dir}/scripts/emailfax.sh $1 /tmp/FAX-${uuid}.tiff
>> > '\\\${fax_success}' '\\\${fax_result_code}' '\\\${fax_result_text}'
>> > '\\\${fax_local_station_id}' '\\\${fax_remote_station_id}'
>> > '\\\${fax_document_transferred_pages}' '\\\${fax_document_total_pages}'
>> > '\\\${fax__image_resolution}' '\\\${fax_image_size}' 
>> > '\\\${fax_bad_rows}'
>> > '\\\${fax_transfer_rate}' '\\\${fax_v17_disabled}'
>> > '\\\${fax_ecm_requested}' '\\\${fax_filename}' "/>
>> >            <action application="answer" />
>> >            <action application="playback" 
>> > data="silence_stream://2000"/>
>> > <!--
>> >            <action application="set" 
>> > data="fax_enable_t38_request=true"/>
>> >            <action application="set" data="fax_enable_t38=true"/>
>> > -->
>> >            <action application="set" data="fax_ident=+41xxxxxxxxx"/>
>> >            <action application="set" data="fax_disable_v17=1"/>
>> >            <action application="set" data="fax_verbose=0"/>
>> >            <action application="rxfax" data="/tmp/FAX-${uuid}.tiff"/>
>> >            <action application="hangup"/>
>> >        </condition>
>> >    </extension>
>> >
>> >
>> > =======================================================================
>> > log entry from an unsuccessful incoming fax:
>> > =======================================================================
>> >
>> > 2010-12-07 23:33:51.157588 [DEBUG] sofia_glue.c:182
>> > sofia/external/Anonymous at anonymous.invalid image media sdp:
>> > v=0
>> > o=FreeSWITCH 1291733462 1291733465 IN IP4 xxx.xxx.xxx.xxx
>> > s=FreeSWITCH
>> > c=IN IP4 xxx.xxx.xxx.xxx
>> > t=0 0
>> > m=image 31358 udptl t38
>> > a=T38FaxVersion:0
>> > a=T38MaxBitRate:9600
>> > a=T38FaxFillBitRemoval
>> > a=T38FaxRateManagement:transferredTCF
>> > a=T38FaxMaxBuffer:2000
>> > a=T38FaxMaxDatagram:400
>> > a=T38FaxUdpEC:t38UDPRedundancy
>> >
>> > 2010-12-07 23:33:51.157588 [DEBUG] sofia.c:4597 Channel
>> > sofia/external/Anonymous at anonymous.invalid entering state 
>> > [completed][200]
>> > 2010-12-07 23:33:51.157588 [DEBUG] sofia.c:4597 Channel
>> > sofia/external/Anonymous at anonymous.invalid entering state [ready][200]
>> > 2010-12-07 23:33:58.814979 [DEBUG] switch_rtp.c:2544 Correct ip/port
>> > confirmed.
>> > 2010-12-07 23:35:09.772752 [DEBUG] switch_channel.c:2457
>> > (sofia/external/Anonymous at anonymous.invalid) Callstate Change ACTIVE ->
>> > HANGUP
>> > 2010-12-07 23:35:09.772752 [NOTICE] sofia.c:528 Hangup
>> > sofia/external/Anonymous at anonymous.invalid [CS_EXECUTE] 
>> > [NORMAL_CLEARING]
>> > 2010-12-07 23:35:09.772752 [DEBUG] switch_channel.c:2473 Send signal
>> > sofia/external/Anonymous at anonymous.invalid [KILL]
>> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_session.c:1057 Send 
>> > signal
>> > sofia/external/Anonymous at anonymous.invalid [BREAK]
>> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:323
>> > ==============================================================================
>> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:329 Fax 
>> > successfully
>> > received.
>> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:340 Remote station
>> > id:
>> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:341 Local station 
>> > id:
>> > +41xxxxxxxxx
>> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:342 Pages
>> > transferred: 2
>> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:344 Total fax 
>> > pages:
>> > 2
>> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:345 Image 
>> > resolution:
>> > 8031x7700
>> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate:
>> > 9600
>> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:348 ECM status
>> > on
>> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:349 remote 
>> > country:
>> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:350 remote vendor:
>> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:351 remote model:
>> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:353
>> > ==============================================================================
>> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_codec.c:141
>> > sofia/external/Anonymous at anonymous.invalid Restore previous codec 
>> > PCMA:8.
>> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_session.c:1933
>> > sofia/external/Anonymous at anonymous.invalid skip receive message
>> > [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:366
>> > (sofia/external/Anonymous at anonymous.invalid) State EXECUTE going to 
>> > sleep
>> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:320
>> > (sofia/external/Anonymous at anonymous.invalid) Running State Change
>> > CS_HANGUP
>> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:553
>> > (sofia/external/Anonymous at anonymous.invalid) State HANGUP
>> > 2010-12-07 23:35:09.772752 [DEBUG] mod_sofia.c:453
>> > sofia/external/Anonymous at anonymous.invalid Overriding SIP cause 480 
>> > with
>> > 200 from the other leg
>> > 2010-12-07 23:35:09.772752 [DEBUG] mod_sofia.c:459 Channel
>> > sofia/external/Anonymous at anonymous.invalid hanging up, cause:
>> > NORMAL_CLEARING
>> > 2010-12-07 23:35:09.821455 [DEBUG] switch_core_state_machine.c:46
>> > sofia/external/Anonymous at anonymous.invalid Standard HANGUP, cause:
>> > NORMAL_CLEARING
>> > 2010-12-07 23:35:09.821455 [DEBUG] switch_core_state_machine.c:553
>> > (sofia/external/Anonymous at anonymous.invalid) State HANGUP going to 
>> > sleep
>> > 2010-12-07 23:35:09.822803 [NOTICE] mod_commands.c:4187 Executing 
>> > command:
>> > /usr/local/freeswitch/scripts/emailfax.sh 1111
>> > /tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff '1' '0' 'OK'
>> > '+41xxxxxxxxx' '' '2' '2
>> > ' '' '0' '0' '9600' '1' '1'
>> > '/tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff'
>> > 2010-12-07 23:35:10.076623 [DEBUG] switch_core_state_machine.c:488 
>> > Hangup
>> > Command with no Session 
>> > system(/usr/local/freeswitch/scripts/emailfax.sh
>> > 1111 /tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff '1' '0'
>> > 'OK' '+41xxxxxxxxx' '' '2' '2' '' '0' '0' '9600' '1' '1'
>> > '/tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff' ):
>> > +OK
>> >
>> >
>> > =======================================================================
>> > log entry from a successful incoming fax:
>> > =======================================================================
>> >
>> > 2010-12-07 23:54:59.082301 [DEBUG] sofia_glue.c:182
>> > sofia/external/Anonymous at anonymous.invalid image media sdp:
>> > v=0
>> > o=FreeSWITCH 1291735801 1291735804 IN IP4 xxx.xxx.xxx.xxx
>> > s=FreeSWITCH
>> > c=IN IP4 xxx.xxx.xxx.xxx
>> > t=0 0
>> > m=image 30280 udptl t38
>> > a=T38FaxVersion:0
>> > a=T38MaxBitRate:9600
>> > a=T38FaxFillBitRemoval
>> > a=T38FaxRateManagement:transferredTCF
>> > a=T38FaxMaxBuffer:2000
>> > a=T38FaxMaxDatagram:400
>> > a=T38FaxUdpEC:t38UDPRedundancy
>> >
>> > 2010-12-07 23:54:59.082301 [DEBUG] sofia.c:4597 Channel
>> > sofia/external/Anonymous at anonymous.invalid entering state 
>> > [completed][200]
>> > 2010-12-07 23:54:59.100811 [DEBUG] sofia.c:4597 Channel
>> > sofia/external/Anonymous at anonymous.invalid entering state [ready][200]
>> > 2010-12-07 23:55:06.783427 [DEBUG] switch_rtp.c:2544 Correct ip/port
>> > confirmed.
>> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:323
>> > ==============================================================================
>> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:329 Fax 
>> > successfully
>> > received.
>> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:340 Remote station
>> > id:
>> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:341 Local station 
>> > id:
>> > +41xxxxxxxxx
>> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:342 Pages
>> > transferred: 1
>> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:344 Total fax 
>> > pages:
>> > 1
>> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:345 Image 
>> > resolution:
>> > 8031x7700
>> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate:
>> > 9600
>> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:348 ECM status
>> > on
>> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:349 remote 
>> > country:
>> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:350 remote vendor:
>> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:351 remote model:
>> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:353
>> > ==============================================================================
>> > 2010-12-07 23:55:31.616038 [DEBUG] switch_channel.c:2457
>> > (sofia/external/Anonymous at anonymous.invalid) Callstate Change ACTIVE ->
>> > HANGUP
>> > 2010-12-07 23:55:31.616038 [NOTICE] sofia.c:528 Hangup
>> > sofia/external/Anonymous at anonymous.invalid [CS_EXECUTE] 
>> > [NORMAL_CLEARING]
>> > 2010-12-07 23:55:31.616038 [DEBUG] switch_channel.c:2473 Send signal
>> > sofia/external/Anonymous at anonymous.invalid [KILL]
>> > 2010-12-07 23:55:31.616038 [DEBUG] switch_core_session.c:1057 Send 
>> > signal
>> > sofia/external/Anonymous at anonymous.invalid [BREAK]
>> > 2010-12-07 23:55:31.617356 [DEBUG] switch_core_codec.c:141
>> > sofia/external/Anonymous at anonymous.invalid Restore previous codec 
>> > PCMA:8.
>> > 2010-12-07 23:55:31.617356 [DEBUG] switch_core_session.c:1933
>> > sofia/external/Anonymous at anonymous.invalid skip receive message
>> > [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> > 2010-12-07 23:55:31.617356 [DEBUG] switch_core_state_machine.c:366
>> > (sofia/external/Anonymous at anonymous.invalid) State EXECUTE going to 
>> > sleep
>> > 2010-12-07 23:55:31.617356 [DEBUG] switch_core_state_machine.c:320
>> > (sofia/external/Anonymous at anonymous.invalid) Running State Change
>> > CS_HANGUP
>> > 2010-12-07 23:55:31.618542 [DEBUG] switch_core_state_machine.c:553
>> > (sofia/external/Anonymous at anonymous.invalid) State HANGUP
>> > 2010-12-07 23:55:31.618542 [DEBUG] mod_sofia.c:453
>> > sofia/external/Anonymous at anonymous.invalid Overriding SIP cause 480 
>> > with
>> > 200 from the other leg
>> > 2010-12-07 23:55:31.618542 [DEBUG] mod_sofia.c:459 Channel
>> > sofia/external/Anonymous at anonymous.invalid hanging up, cause:
>> > NORMAL_CLEARING
>> > 2010-12-07 23:55:31.721993 [DEBUG] switch_core_state_machine.c:46
>> > sofia/external/Anonymous at anonymous.invalid Standard HANGUP, cause:
>> > NORMAL_CLEARING
>> > 2010-12-07 23:55:31.721993 [DEBUG] switch_core_state_machine.c:553
>> > (sofia/external/Anonymous at anonymous.invalid) State HANGUP going to 
>> > sleep
>> > 2010-12-07 23:55:31.721993 [NOTICE] mod_commands.c:4187 Executing 
>> > command:
>> > /usr/local/freeswitch/scripts/emailfax.sh 1111
>> > /tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff '1' '0' 'OK'
>> > '+41xxxxxxxxx' '' '1' '1' '' '0' '0' '9600' '1' '1'
>> > '/tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff'
>> > 2010-12-07 23:55:31.904090 [DEBUG] switch_core_state_machine.c:488 
>> > Hangup
>> > Command with no Session 
>> > system(/usr/local/freeswitch/scripts/emailfax.sh
>> > 1111 /tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff '1' '0' 'OK'
>> > '+41xxxxxxxxx' '' '1' '1' '' '0' '0' '9600' '1' '1'
>> > '/tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff' ):
>> > +OK
>> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From brian at freeswitch.org  Fri Dec 10 06:53:11 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 9 Dec 2010 21:53:11 -0600
Subject: [Freeswitch-users] Email Voicemail Volume Mystery
In-Reply-To: <ACC02B70-651D-47D5-9FC2-36662EC7A5F8@mgtech.com>
References: <A3633C87-61A4-41BB-B6BD-B9477F9AB5DB@mgtech.com>
	<AANLkTinjBY9Qb6gWUXJQMrp6_FX3Bj2WSOsZ4cJw-ODo@mail.gmail.com>
	<ACC02B70-651D-47D5-9FC2-36662EC7A5F8@mgtech.com>
Message-ID: <ADD52DC0-8496-45C1-942F-E72FE15DD1D4@freeswitch.org>

Welcome to voip... google a little this is a common loss problem going to voicemail in some cases... 

/b

On Dec 9, 2010, at 6:12 PM, Mario G wrote:

> Aiff because the iPhone will no play all formats, but I also tried mp3 to see if it made any difference and it did not. Both types of files had the same very low volume.
> MG
> 



From covici at ccs.covici.com  Fri Dec 10 06:55:58 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 09 Dec 2010 22:55:58 -0500
Subject: [Freeswitch-users] =?gb2312?b?Y2hhbmdlIKfjYWxsZXJfaWRfbmFtZSAm?=
	=?gb2312?b?IG51bWJlcg==?=
In-Reply-To: <AANLkTik_EwZ1eqjfwd0ahc=0DtEyyWg8Muv8Ay1+J7tp@mail.gmail.com>
References: <E1PQNPD-0002tm-00.melkybes-mail-ru@f160.mail.ru>
	<4CFFBD23.6010608@gmail.com>
	<E1PQgb0-0001QJ-00.melkybes-mail-ru@f252.mail.ru>
	<AANLkTi=OBmDKBwmx3p0z-RQDa8bOsW3jeHTiKPPxO-cD@mail.gmail.com>
	<25600.1291932621@ccs.covici.com>
	<AANLkTik_EwZ1eqjfwd0ahc=0DtEyyWg8Muv8Ay1+J7tp@mail.gmail.com>
Message-ID: <31387.1291953358@ccs.covici.com>

OK, found it -- thanks.

Michael Collins <msc at freeswitch.org> wrote:

> 2010/12/9 <covici at ccs.covici.com>
> 
> > I don't see a logger.pl in that directory.  I have used script to
> > capture such things, except that it puts color escape codes in.
> >
> > Doh!
> 
> It's in libs/esl/perl/logger.pl
> 
> My bad.
> -MC
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From lakindia89 at gmail.com  Fri Dec 10 07:16:06 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 10 Dec 2010 09:46:06 +0530
Subject: [Freeswitch-users] how bind_meta_app works?
In-Reply-To: <AANLkTinaNJtjCNbORiK_3gWB+fRTU17C_aRL0LYtqkMf@mail.gmail.com>
References: <AANLkTi=m0ww373iFn0nrZQ4hofyAp6duN=-Y4G-wyPgt@mail.gmail.com>
	<AANLkTi=O8WzT2a-NPCtaQ3ebgbTuzJ+YkTDMdXUsx7Ti@mail.gmail.com>
	<AANLkTinaNJtjCNbORiK_3gWB+fRTU17C_aRL0LYtqkMf@mail.gmail.com>
Message-ID: <AANLkTikqBqTbTDofsE2_Mg4kAAzSoOPjOmvSikQ6419O@mail.gmail.com>

Hi Collins,
Thanks for your suggestion. I've just now figured out what was the problem.

I believe, It is because of setting the setEventLock to true. I've set that
to false and after that it works as expected.

Here is my understanding.
SetEventLock=true.
In A leg, currently bridge application is running. When B leg presses the
#1, since SetEventLock is set to true, it was queued for execution.
When B hangup the call ( bridge application got completed ) , the queued
application got executed.

Somebody can correct me if I am wrong.



On Thu, Dec 9, 2010 at 11:42 PM, Michael Collins <msc at freeswitch.org> wrote:

> Did you try enabling on the A leg also?
> $EslCon->execute("bind_meta_app","1 ab o event::appli=testing")
>
> Just curious as I've never actually tried this before.
> -MC
>
> On Wed, Dec 8, 2010 at 5:01 AM, lakshmanan ganapathy <lakindia89 at gmail.com
> > wrote:
>
>> Hi all,
>> Is my understanding is right??. If so, is it possible to run applications
>> in A leg...
>>
>>
>> On Mon, Dec 6, 2010 at 12:19 PM, lakshmanan ganapathy <
>> lakindia89 at gmail.com> wrote:
>>
>>> Hi all,
>>> I was experimenting the bind_mea_app and I have a doubt.
>>> There is a call from 1000 to FreeSwitch extension and it is connecting to
>>> event outbound socket.
>>> I've the following script in place.
>>>
>>> #!/usr/bin/perl
>>> use strict;
>>> use warnings;
>>> use Data::Dumper;
>>> use lib '/usr/src/freeswitch/libs/esl/perl/';
>>> use IO::Socket::INET;
>>> use ESL;
>>>
>>> my $SOCK = new IO::Socket::INET ( LocalHost => 'localhost',  LocalPort =>
>>> '8447',  Proto => 'tcp',  Listen => 10,  Reuse => 1 );
>>> unless ($SOCK)  {
>>>                 print("Could not create socket: $!");
>>>                 exit(2);
>>> }
>>>
>>> while (1)  {
>>>                 # Wait for any client through accept function in socket
>>> module.
>>>                 my $new_sock = $SOCK->accept();
>>>                 next if (not defined($new_sock));
>>>                 # Get socket host
>>>                 my $host = $new_sock->sockhost();
>>>                 print("Got a client accepted from $host\n");
>>>                 my $pid = fork();
>>>                 if ($pid) {
>>>                                 close($new_sock);
>>>                                 next;
>>>                 }
>>>                 my $fd = fileno($new_sock);
>>>                 print("Newly forked child, pid: $$\n");
>>>                 my $EslCon = new ESL::ESLconnection($fd);
>>>                 print "Connection created successfully\n";
>>>                 my $info = $EslCon->getInfo();
>>>                 $EslCon->setEventLock("true");
>>>                 my $uuid = $info->getHeader("unique-id");
>>>                 print Dumper $info->serialize();
>>>                 $EslCon->execute("set","bind_meta_key=#");
>>>                 $EslCon->execute("bind_meta_app","1 b o
>>> event::appli=testing");
>>>                 my $api =
>>> $EslCon->execute("bridge","{ignore_early_media=true}freetdm/1/a/xxxxxxxxx");
>>> }
>>>
>>> The script called to the number that was dialed. In that number I pressed
>>> #1, but the event was not sent,
>>> and I assume that the bridge application is executing in the A leg ( is
>>> it correct?? ) . So only after that bridge application gets completed the
>>> event gets triggered out.
>>>
>>> If I execute the application in the B leg, then it is working fine, since
>>> there is no application that is running at that time.
>>> Now I wanted to know if there is any way to run the application on A leg,
>>> I need that application to be executed even the bridge is not completed.
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Fri Dec 10 10:01:48 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 10 Dec 2010 07:01:48 +0000
Subject: [Freeswitch-users] Refusing REFER requests
In-Reply-To: <AANLkTi=s4pXFSc+anKUPW37jdfgaF2O51PRedEDZ7Q+N@mail.gmail.com>
References: <AANLkTiks3YQm3Cj=M-WFYv+dA5g9QE4EJeyQmGEqE56O@mail.gmail.com>
	<AANLkTi=x2rsehx6obU2dZnHXoqnOKbqzkp0uT_9Nn5fs@mail.gmail.com>
	<6C9B5526-88B9-4AB5-B1A2-36A4751751FB@freeswitch.org>
	<AANLkTi=mOgm8fWA+F+JvSDDc9kNntXE6OygWe4=cHHSa@mail.gmail.com>
	<56E80050-72E6-4960-866E-7B20E42B0C06@freeswitch.org>
	<AANLkTi=s4pXFSc+anKUPW37jdfgaF2O51PRedEDZ7Q+N@mail.gmail.com>
Message-ID: <F1F8B7A4-0878-437C-8C8A-BABC37021076@gmail.com>

I think what Brian is getting at is that if you refuse/ignore the REFER you'll probably end up with a dropped call.

Steve on iPhone

On 9 Dec 2010, at 22:49, mayamatakeshi <mayamatakeshi at gmail.com> wrote:

> 
> On Fri, Dec 10, 2010 at 7:32 AM, Brian West <brian at freeswitch.org> wrote:
> TAG_IF((profile->mflags & MFLAG_REFER), NUTAG_ALLOW("REFER")),
> 
> This removes refer from the allowed methods in sofia so we will say that the method is NOT allowed.
> 
> If you want to refuse when the destination is invalid then give up now because you will need to know before the refer is sent to know that.
> 
> I am not sure what you mean.
> I was thinking in adding a new parameter for mod_sofia like "forbidden_transfer_target" and add code in sofia.c right after that MFLAG check:
> 
> if (!(profile->mflags & MFLAG_REFER)) {
>   nua_respond(nh, SIP_403_FORBIDDEN, NUTAG_WITH_THIS(nua), TAG_END());
>   goto done;
> }
> 
> // my pseudo-code:
> 
> if (REFER_URI =~ forbidded_transfer_target) {
>   nua_respond(nh, SIP_403_FORBIDDEN, NUTAG_WITH_THIS(nua), TAG_END());
>   goto done;
> }
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From msc at freeswitch.org  Fri Dec 10 10:43:04 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 9 Dec 2010 23:43:04 -0800
Subject: [Freeswitch-users] Asterisk migrations
Message-ID: <AANLkTim48KyCq8GfsqJ=fmjfeb81g0UZs6MDkhhcAvRS@mail.gmail.com>

Hey gang,

I was just curious if anyone out there had created any tools to assist with
migrating from Asterisk (or Trixbox, etc.) to FreeSWITCH. We're starting to
see this scenario more and I would like to find out how our community
members are handling it.

Thanks,
MC
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From haloha201 at yahoo.com  Fri Dec 10 11:14:18 2010
From: haloha201 at yahoo.com (ha do)
Date: Fri, 10 Dec 2010 00:14:18 -0800 (PST)
Subject: [Freeswitch-users] need help to install and use the Event socket
	library
Message-ID: <125236.27594.qm@web32407.mail.mud.yahoo.com>

Hi all

i follow the wiki page
http://wiki.freeswitch.org/wiki/Event_Socket_Library

the steps:
cd ../libs/esl

then vi python/Makefile and edit
from
LOCAL_CFLAGS=`python ./python-config --includes`
LOCAL_LDFLAGS=`python ./python-config --ldflags`

to 
LOCAL_CFLAGS=`python2.5 ./python-config --includes`
LOCAL_LDFLAGS=`python2.5 ./python-config --ldflags`

make pymod
done

then i switch to python mode
do from freeswitch import * and get error:
>>> from freeswitch import *
Traceback (most recent call last):
  File "<stdin>", line 1, in <module>
  File "/usr/lib/python2.5/site-packages/freeswitch.py", line 7, in <module>
    import _freeswitch
ImportError: No module named _freeswitch


how do i install the esl for python properly

thank you
Ha`



      


From haloha201 at yahoo.com  Fri Dec 10 11:14:19 2010
From: haloha201 at yahoo.com (ha do)
Date: Fri, 10 Dec 2010 00:14:19 -0800 (PST)
Subject: [Freeswitch-users] need help to install and use the Event socket
	library
Message-ID: <503729.57489.qm@web32405.mail.mud.yahoo.com>

Hi all

i follow the wiki page
http://wiki.freeswitch.org/wiki/Event_Socket_Library

the steps:
cd ../libs/esl

then vi python/Makefile and edit
from
LOCAL_CFLAGS=`python ./python-config --includes`
LOCAL_LDFLAGS=`python ./python-config --ldflags`

to 
LOCAL_CFLAGS=`python2.5 ./python-config --includes`
LOCAL_LDFLAGS=`python2.5 ./python-config --ldflags`

make pymod
done

then i switch to python mode
do from freeswitch import * and get error:
>>> from freeswitch import *
Traceback (most recent call last):
  File "<stdin>", line 1, in <module>
  File "/usr/lib/python2.5/site-packages/freeswitch.py", line 7, in <module>
    import _freeswitch
ImportError: No module named _freeswitch


how do i install the esl for python properly

thank you
Ha`



      


From mayamatakeshi at gmail.com  Fri Dec 10 11:49:49 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Fri, 10 Dec 2010 17:49:49 +0900
Subject: [Freeswitch-users] Refusing REFER requests
In-Reply-To: <F1F8B7A4-0878-437C-8C8A-BABC37021076@gmail.com>
References: <AANLkTiks3YQm3Cj=M-WFYv+dA5g9QE4EJeyQmGEqE56O@mail.gmail.com>
	<AANLkTi=x2rsehx6obU2dZnHXoqnOKbqzkp0uT_9Nn5fs@mail.gmail.com>
	<6C9B5526-88B9-4AB5-B1A2-36A4751751FB@freeswitch.org>
	<AANLkTi=mOgm8fWA+F+JvSDDc9kNntXE6OygWe4=cHHSa@mail.gmail.com>
	<56E80050-72E6-4960-866E-7B20E42B0C06@freeswitch.org>
	<AANLkTi=s4pXFSc+anKUPW37jdfgaF2O51PRedEDZ7Q+N@mail.gmail.com>
	<F1F8B7A4-0878-437C-8C8A-BABC37021076@gmail.com>
Message-ID: <AANLkTikJx5gEYo5U_T=8JDTNCQP9tWBz5O2+OVKF9bcb@mail.gmail.com>

Thanks,
actually, I don't want to go with this, but I need to list my options trying
to solve an issue with a terminal (and replacing the terminal is not a
option).

On Fri, Dec 10, 2010 at 4:01 PM, Steven Ayre <steveayre at gmail.com> wrote:

> I think what Brian is getting at is that if you refuse/ignore the REFER
> you'll probably end up with a dropped call.
>
> Steve on iPhone
>
> On 9 Dec 2010, at 22:49, mayamatakeshi <mayamatakeshi at gmail.com> wrote:
>
>
> On Fri, Dec 10, 2010 at 7:32 AM, Brian West <brian at freeswitch.org> wrote:
>
>> TAG_IF((profile->mflags & MFLAG_REFER), NUTAG_ALLOW("REFER")),
>>
>> This removes refer from the allowed methods in sofia so we will say that
>> the method is NOT allowed.
>>
>> If you want to refuse when the destination is invalid then give up now
>> because you will need to know before the refer is sent to know that.
>>
>
> I am not sure what you mean.
> I was thinking in adding a new parameter for mod_sofia like
> "forbidden_transfer_target" and add code in sofia.c right after that MFLAG
> check:
>
> if (!(profile->mflags & MFLAG_REFER)) {
>   nua_respond(nh, SIP_403_FORBIDDEN, NUTAG_WITH_THIS(nua), TAG_END());
>   goto done;
> }
>
> // my pseudo-code:
>
> if (REFER_URI =~ forbidded_transfer_target) {
>   nua_respond(nh, SIP_403_FORBIDDEN, NUTAG_WITH_THIS(nua), TAG_END());
>   goto done;
> }
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From melkybes at mail.ru  Fri Dec 10 14:33:30 2010
From: melkybes at mail.ru (=?koi8-r?Q?=ED=C9=C8=C1=C9=CC_=F3=C1=CC=D4=C1=CE=CF=D7?=)
Date: Fri, 10 Dec 2010 14:33:30 +0300
Subject: [Freeswitch-users] =?koi8-r?b?YmluZF9tZXRhX2FwcCB3aGF0IGNhbGwg?=
	=?koi8-r?b?bGVnPw==?=
In-Reply-To: <1291941471978-5821494.post@n2.nabble.com>
References: <AANLkTinndUGCzh+gsoyZLjQ8LR+phQLuBMsW1LvMSH4+@mail.gmail.com>
	<1291903360189-5818816.post@n2.nabble.com>
	<1291941471978-5821494.post@n2.nabble.com>
Message-ID: <E1PR1EA-00025a-00.melkybes-mail-ru@f82.mail.ru>

Hello
Can I  receive  information what side of the call (A or B) initiate bind_meta_app ?

? ?????????
?????? ????????





From peter.olsson at visionutveckling.se  Fri Dec 10 16:45:32 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 10 Dec 2010 14:45:32 +0100
Subject: [Freeswitch-users] Reliability of HEARTBEAT events
	every	20	seconds? (in mod_event_socket)
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57EC81C46C@cooper>
References: <C2628CCB-21C4-4C66-957F-4BB4AC1AF37F@visionutveckling.se>
	<549CFEF87AEDE841A38E9D15EAB4C04C57EC81C469@cooper>,
	<934A2ECD-3013-4598-8740-F20C6162FB2B@freeswitch.org>,
	<549CFEF87AEDE841A38E9D15EAB4C04C57EC81C46C@cooper>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57EC81C46E@cooper>

Ok, I found the reason for the delays.

check_ip() (which is called after heartbeat has been sent) sometimes took up to 12 seconds to complete. This seems to be related with some slow/semi-working DNS-servers in my lab network.

I've created a patch that adds a setting that makes it possible to deactivate detection of IP address change (default behaviour will be the way it works now). I know this is kind of a special case, but still, the check_ip() is not needed on most servers I would say.

I will create a jira and post the patch there (later on today).

Thanks for all suggestions about this.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Peter Olsson [peter.olsson at visionutveckling.se]
Skickat: den 9 december 2010 19:42
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Reliability of HEARTBEAT events every      20      seconds? (in mod_event_socket)

Yes I realize this, but I just think 10-12 seconds (30-32 instead of 20) seems like a little too much to be correct - or should I consider this as normal?

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Brian West [brian at freeswitch.org]
Skickat: den 9 december 2010 19:35
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Reliability of HEARTBEAT events every 20   seconds? (in mod_event_socket)

You do realize the time in the log is when its submitted.  And you have some time between then the task is run then rescheduled... Are we really going to be that picky?

/b

On Dec 9, 2010, at 12:25 PM, Peter Olsson wrote:

> That didn't make a difference either.. I will have to look into this deeper, and add some more debug logging for when the event fires within FS.
>
> No more suggestions anyone? :)
>
> /Peter


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




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

!DSPAM:4d0124e632764708731795!



From freeswitch at peely.com  Fri Dec 10 16:56:57 2010
From: freeswitch at peely.com (peely)
Date: Fri, 10 Dec 2010 05:56:57 -0800 (PST)
Subject: [Freeswitch-users] Stop offering video on outbound if no video
	offered on inbound
In-Reply-To: <610548AF-200D-4B3A-9AEE-F1901ACE6A3F@freeswitch.org>
References: <1291921815608-5820193.post@n2.nabble.com>
	<610548AF-200D-4B3A-9AEE-F1901ACE6A3F@freeswitch.org>
Message-ID: <1291989417349-5823130.post@n2.nabble.com>


Hi Brian,

Yes, the codecs are present in the profile, as I want to support video for
when two user agent support video.

What I mean is that I see video offered on the outbound when the inbound
does not offer video, I wanted to stop this behaviour.


Regards,


Neil.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Stop-offering-video-on-outbound-if-no-video-offered-on-inbound-tp5820193p5823130.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Fri Dec 10 17:14:57 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 10 Dec 2010 14:14:57 +0000
Subject: [Freeswitch-users] Stop offering video on outbound if no video
 offered on inbound
In-Reply-To: <1291989417349-5823130.post@n2.nabble.com>
References: <1291921815608-5820193.post@n2.nabble.com>
	<610548AF-200D-4B3A-9AEE-F1901ACE6A3F@freeswitch.org>
	<1291989417349-5823130.post@n2.nabble.com>
Message-ID: <AANLkTinFhEL-w94-fEEOyf0tFQKKgg0HQtq1vQu2yH9x@mail.gmail.com>

Not ideal, but you could adjust from dialplan with absolute_codec_string.

Steve


On 10 December 2010 13:56, peely <freeswitch at peely.com> wrote:
>
> Hi Brian,
>
> Yes, the codecs are present in the profile, as I want to support video for
> when two user agent support video.
>
> What I mean is that I see video offered on the outbound when the inbound
> does not offer video, I wanted to stop this behaviour.
>
>
> Regards,
>
>
> Neil.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Stop-offering-video-on-outbound-if-no-video-offered-on-inbound-tp5820193p5823130.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From Nabble at slickdeals.endjunk.com  Fri Dec 10 18:05:44 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 10 Dec 2010 07:05:44 -0800 (PST)
Subject: [Freeswitch-users] bind_meta_app what call leg?
In-Reply-To: <E1PR1EA-00025a-00.melkybes-mail-ru@f82.mail.ru>
References: <1291903360189-5818816.post@n2.nabble.com>
	<AANLkTim_bp9Xk94R6uW2QJmp1dVPKOL8rxjhLS7s_K99@mail.gmail.com>
	<1291923916094-5820341.post@n2.nabble.com>
	<AANLkTinndUGCzh+gsoyZLjQ8LR+phQLuBMsW1LvMSH4+@mail.gmail.com>
	<1291941471978-5821494.post@n2.nabble.com>
	<E1PR1EA-00025a-00.melkybes-mail-ru@f82.mail.ru>
Message-ID: <1291993544270-5823341.post@n2.nabble.com>



?????? ???????? wrote:
> 
> Can I  receive  information what side of the call (A or B) initiate
> bind_meta_app ?
Being a newbie in FS, I am not sure what exactly you meant; however, I have 
http://pastebin.com/XBuNYSKS pastebin  the output from fs_cli when the call
was initiated (if that will help).

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Disabling-Google-Voice-Voicemail-tp5818816p5823341.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Fri Dec 10 18:38:19 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 10 Dec 2010 09:38:19 -0600
Subject: [Freeswitch-users] Stop offering video on outbound if no video
	offered on inbound
In-Reply-To: <AANLkTinFhEL-w94-fEEOyf0tFQKKgg0HQtq1vQu2yH9x@mail.gmail.com>
References: <1291921815608-5820193.post@n2.nabble.com>
	<610548AF-200D-4B3A-9AEE-F1901ACE6A3F@freeswitch.org>
	<1291989417349-5823130.post@n2.nabble.com>
	<AANLkTinFhEL-w94-fEEOyf0tFQKKgg0HQtq1vQu2yH9x@mail.gmail.com>
Message-ID: <16A21EFC-1774-4599-AAB9-BDA41DFE2015@freeswitch.org>

yah absolute_codec_string is the best way to resolve this on a case by case basis.
/b

On Dec 10, 2010, at 8:14 AM, Steven Ayre wrote:

> Not ideal, but you could adjust from dialplan with absolute_codec_string.
> 
> Steve
> 
> 
> On 10 December 2010 13:56, peely <freeswitch at peely.com> wrote:
>> 
>> Hi Brian,
>> 
>> Yes, the codecs are present in the profile, as I want to support video for
>> when two user agent support video.
>> 
>> What I mean is that I see video offered on the outbound when the inbound
>> does not offer video, I wanted to stop this behaviour.
>> 
>> 
>> Regards,
>> 
>> 
>> Neil.
>> --

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From danb.lists at googlemail.com  Fri Dec 10 19:51:51 2010
From: danb.lists at googlemail.com (Dan-Cristian Bogos)
Date: Fri, 10 Dec 2010 17:51:51 +0100
Subject: [Freeswitch-users] G729B and mod_com_g729
Message-ID: <AANLkTikK-aNyUWX_qVSAzVJvhBsX9KHZSRPbLwuVTQwM@mail.gmail.com>

Hey Brian,

Thanks for coming back to me so fast.

For the sake of records, I found out that you are perfectly right,
G729B is not standard and it is a proprietary method of Teles UA for
G729 handling.

Have a good one!

DanB


From Nabble at slickdeals.endjunk.com  Fri Dec 10 20:15:30 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 10 Dec 2010 09:15:30 -0800 (PST)
Subject: [Freeswitch-users] Asterisk migrations
In-Reply-To: <AANLkTim48KyCq8GfsqJ=fmjfeb81g0UZs6MDkhhcAvRS@mail.gmail.com>
References: <AANLkTim48KyCq8GfsqJ=fmjfeb81g0UZs6MDkhhcAvRS@mail.gmail.com>
Message-ID: <1292001330480-5823851.post@n2.nabble.com>


Even though I only use a plain-vanilla Asterisk PBX System hosted on a
discontinued Netgear  http://kb.netgear.com/app/products/model/a_id/2598
WGT634U  flashed with a self-built OpenWRT firmware, I sure would like to
know the answer to this too.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Asterisk-migrations-tp5822248p5823851.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Fri Dec 10 20:29:31 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 10 Dec 2010 17:29:31 +0000
Subject: [Freeswitch-users] G729B and mod_com_g729
In-Reply-To: <AANLkTikK-aNyUWX_qVSAzVJvhBsX9KHZSRPbLwuVTQwM@mail.gmail.com>
References: <AANLkTikK-aNyUWX_qVSAzVJvhBsX9KHZSRPbLwuVTQwM@mail.gmail.com>
Message-ID: <AANLkTik+y_+LkoOsg9ytnyPtO72LU7PWBF1zbQCrEL9s@mail.gmail.com>

Brian, just out of curiosity (and I know this is bad bad bad bad) why
not have a compatibility setting so that codecs identified by static
IANA-assigned numbers (his example uses G729's assigned 18) can still
function. Or are there any broken devices that use the wrong numbers
that that might cause even problems with?

*ducks*

-Steve


On 10 December 2010 16:51, Dan-Cristian Bogos <danb.lists at googlemail.com> wrote:
> Hey Brian,
>
> Thanks for coming back to me so fast.
>
> For the sake of records, I found out that you are perfectly right,
> G729B is not standard and it is a proprietary method of Teles UA for
> G729 handling.
>
> Have a good one!
>
> DanB
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From steveayre at gmail.com  Fri Dec 10 20:30:04 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 10 Dec 2010 17:30:04 +0000
Subject: [Freeswitch-users] G729B and mod_com_g729
In-Reply-To: <AANLkTik+y_+LkoOsg9ytnyPtO72LU7PWBF1zbQCrEL9s@mail.gmail.com>
References: <AANLkTikK-aNyUWX_qVSAzVJvhBsX9KHZSRPbLwuVTQwM@mail.gmail.com>
	<AANLkTik+y_+LkoOsg9ytnyPtO72LU7PWBF1zbQCrEL9s@mail.gmail.com>
Message-ID: <AANLkTinAQf7JzqaiGH+-UH7gRPbfEngnOSU+eoSviUsy@mail.gmail.com>

s/even problems/even more problems/


On 10 December 2010 17:29, Steven Ayre <steveayre at gmail.com> wrote:
> Brian, just out of curiosity (and I know this is bad bad bad bad) why
> not have a compatibility setting so that codecs identified by static
> IANA-assigned numbers (his example uses G729's assigned 18) can still
> function. Or are there any broken devices that use the wrong numbers
> that that might cause even problems with?
>
> *ducks*
>
> -Steve
>
>
> On 10 December 2010 16:51, Dan-Cristian Bogos <danb.lists at googlemail.com> wrote:
>> Hey Brian,
>>
>> Thanks for coming back to me so fast.
>>
>> For the sake of records, I found out that you are perfectly right,
>> G729B is not standard and it is a proprietary method of Teles UA for
>> G729 handling.
>>
>> Have a good one!
>>
>> DanB
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From lloyd.aloysius at gmail.com  Fri Dec 10 20:37:50 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 10 Dec 2010 12:37:50 -0500
Subject: [Freeswitch-users] Asterisk migrations
In-Reply-To: <1292001330480-5823851.post@n2.nabble.com>
References: <AANLkTim48KyCq8GfsqJ=fmjfeb81g0UZs6MDkhhcAvRS@mail.gmail.com>
	<1292001330480-5823851.post@n2.nabble.com>
Message-ID: <AANLkTikHy5OJsquX0C+yHUemgsu+EN9LOb=aZ-Pw2H39@mail.gmail.com>

Hi Michael,

Recently I shutdown 7 Asterisk boxes and move all customers to a
Single Multi tenant FreeSWITCH box.

I did not create any tools . All I did is create custom dial plan for each
domain. But here is couple of areas I find more difficulties

1. Find Me / Follow Me + Confirm
2. NAT Handling
3. Call Forwarding on System Level not the Phone level
4. Toggle Dial plan conditions . Eg: Day/Night mode
5. Voice mail features

Other than that FreeSWITCH is more scalable and have lots of flexibility.

Thanks
Lloyd
On Fri, Dec 10, 2010 at 12:15 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> Even though I only use a plain-vanilla Asterisk PBX System hosted on a
> discontinued Netgear  http://kb.netgear.com/app/products/model/a_id/2598
> WGT634U  flashed with a self-built OpenWRT firmware, I sure would like to
> know the answer to this too.
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Asterisk-migrations-tp5822248p5823851.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Fri Dec 10 20:38:49 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 10 Dec 2010 11:38:49 -0600
Subject: [Freeswitch-users] G729B and mod_com_g729
In-Reply-To: <AANLkTik+y_+LkoOsg9ytnyPtO72LU7PWBF1zbQCrEL9s@mail.gmail.com>
References: <AANLkTikK-aNyUWX_qVSAzVJvhBsX9KHZSRPbLwuVTQwM@mail.gmail.com>
	<AANLkTik+y_+LkoOsg9ytnyPtO72LU7PWBF1zbQCrEL9s@mail.gmail.com>
Message-ID: <9FD71C2B-FDA6-43F9-BD3D-2762701C9620@freeswitch.org>

NDLB-allow-bad-iananame is the option you want on the profile.

The issue with the continued acceptance of broken things is what keeps getting us into this hole that we will never climb out of if we don't STOP IT.  If its wrong do NOT accept it and make the vendor fix their non-compliant crap otherwise this is only going to continue to get worse!

/b

On Dec 10, 2010, at 11:29 AM, Steven Ayre wrote:

> Brian, just out of curiosity (and I know this is bad bad bad bad) why
> not have a compatibility setting so that codecs identified by static
> IANA-assigned numbers (his example uses G729's assigned 18) can still
> function. Or are there any broken devices that use the wrong numbers
> that that might cause even problems with?
> 
> *ducks*
> 
> -Steve



From msc at freeswitch.org  Fri Dec 10 21:17:23 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 10 Dec 2010 10:17:23 -0800
Subject: [Freeswitch-users] G729B and mod_com_g729
In-Reply-To: <9FD71C2B-FDA6-43F9-BD3D-2762701C9620@freeswitch.org>
References: <AANLkTikK-aNyUWX_qVSAzVJvhBsX9KHZSRPbLwuVTQwM@mail.gmail.com>
	<AANLkTik+y_+LkoOsg9ytnyPtO72LU7PWBF1zbQCrEL9s@mail.gmail.com>
	<9FD71C2B-FDA6-43F9-BD3D-2762701C9620@freeswitch.org>
Message-ID: <AANLkTi=V68LqFKwKZXiZiZpaHLNHE7k0oH_3ZA-CFmrd@mail.gmail.com>

Amen, brothah!

On Fri, Dec 10, 2010 at 9:38 AM, Brian West <brian at freeswitch.org> wrote:

> NDLB-allow-bad-iananame is the option you want on the profile.
>
> The issue with the continued acceptance of broken things is what keeps
> getting us into this hole that we will never climb out of if we don't STOP
> IT.  If its wrong do NOT accept it and make the vendor fix their
> non-compliant crap otherwise this is only going to continue to get worse!
>
> /b
>
> On Dec 10, 2010, at 11:29 AM, Steven Ayre wrote:
>
> > Brian, just out of curiosity (and I know this is bad bad bad bad) why
> > not have a compatibility setting so that codecs identified by static
> > IANA-assigned numbers (his example uses G729's assigned 18) can still
> > function. Or are there any broken devices that use the wrong numbers
> > that that might cause even problems with?
> >
> > *ducks*
> >
> > -Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Dec 10 21:21:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 10 Dec 2010 10:21:50 -0800
Subject: [Freeswitch-users] No sip BYE from gateway at hangup
In-Reply-To: <AANLkTi=qjdZHgEbZG1WTUCATcPDE-t93m0V3-_-Bfthz@mail.gmail.com>
References: <AANLkTi=qjdZHgEbZG1WTUCATcPDE-t93m0V3-_-Bfthz@mail.gmail.com>
Message-ID: <AANLkTi==fotz9ppM1bnTA4sm1RkPV2AEm5A-7e7=jYrn@mail.gmail.com>

On Thu, Dec 9, 2010 at 6:26 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> I've got a gateway that seems to be sending the sip BYE really late.
>
> If the local (softphone) party hangs up then everything happens as normal
> as I'd expect it to. If the end terminated through the gateway hangs up then
> it's another 15-40 sec before Freeswitch finds out it's hung up and
> processes the end of the call.
>
> I did a "sofia profile external siptrace on" and during that 15-40 sec
> there was no activity until the end when the BYE turned up.
>
> I had the gateway run a sip trace their end and they said everything looked
> normal to them, and they said:
> "We have reviewed all of the calls specified both on the SIP signaling side
> and the SS7 side.  Under certain circumstances, some calls can take longer
> to fully tear from the PSTN than others.  "
> Which I interpret as a polite shrug of the shoulders :-)
>
I would ask which conditions cause calls to take longer to fully tear down
so I could do my best to avoid those conditions.
-MC
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From brian at freeswitch.org  Fri Dec 10 21:24:22 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 10 Dec 2010 12:24:22 -0600
Subject: [Freeswitch-users] No sip BYE from gateway at hangup
In-Reply-To: <AANLkTi==fotz9ppM1bnTA4sm1RkPV2AEm5A-7e7=jYrn@mail.gmail.com>
References: <AANLkTi=qjdZHgEbZG1WTUCATcPDE-t93m0V3-_-Bfthz@mail.gmail.com>
	<AANLkTi==fotz9ppM1bnTA4sm1RkPV2AEm5A-7e7=jYrn@mail.gmail.com>
Message-ID: <098189B0-2F62-4E03-97DC-B67DE211070A@freeswitch.org>

Full sip trace would be more helpful.

/b

On Dec 10, 2010, at 12:21 PM, Michael Collins wrote:

> I would ask which conditions cause calls to take longer to fully tear down so I could do my best to avoid those conditions.
> -MC
> 



From mario_fs at mgtech.com  Fri Dec 10 23:02:07 2010
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 10 Dec 2010 12:02:07 -0800
Subject: [Freeswitch-users] Email Voicemail Volume Mystery
In-Reply-To: <ADD52DC0-8496-45C1-942F-E72FE15DD1D4@freeswitch.org>
References: <A3633C87-61A4-41BB-B6BD-B9477F9AB5DB@mgtech.com>
	<AANLkTinjBY9Qb6gWUXJQMrp6_FX3Bj2WSOsZ4cJw-ODo@mail.gmail.com>
	<ACC02B70-651D-47D5-9FC2-36662EC7A5F8@mgtech.com>
	<ADD52DC0-8496-45C1-942F-E72FE15DD1D4@freeswitch.org>
Message-ID: <A82B230D-65DE-4A04-929E-2C2D0BA8BEDB@mgtech.com>

I googled a lot trying to figure this out, the strange part is the voicemail volume is always fine, it's only the emailed voicemail that is affected for particular callers.

On Dec 9, 2010, at 7:53 PM, Brian West wrote:

> Welcome to voip... google a little this is a common loss problem going to voicemail in some cases... 
> 
> /b
> 
> On Dec 9, 2010, at 6:12 PM, Mario G wrote:
> 
>> Aiff because the iPhone will no play all formats, but I also tried mp3 to see if it made any difference and it did not. Both types of files had the same very low volume.
>> MG
>> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Fri Dec 10 23:30:02 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 10 Dec 2010 14:30:02 -0600
Subject: [Freeswitch-users] Email Voicemail Volume Mystery
In-Reply-To: <A82B230D-65DE-4A04-929E-2C2D0BA8BEDB@mgtech.com>
References: <A3633C87-61A4-41BB-B6BD-B9477F9AB5DB@mgtech.com>
	<AANLkTinjBY9Qb6gWUXJQMrp6_FX3Bj2WSOsZ4cJw-ODo@mail.gmail.com>
	<ACC02B70-651D-47D5-9FC2-36662EC7A5F8@mgtech.com>
	<ADD52DC0-8496-45C1-942F-E72FE15DD1D4@freeswitch.org>
	<A82B230D-65DE-4A04-929E-2C2D0BA8BEDB@mgtech.com>
Message-ID: <2D5B3151-A236-461C-8B30-BCCBAE7F6B2D@freeswitch.org>

Its an on going issue for some... not sure what causes the loss but it does happen.  The alternative fix is to turn the volume knob up.  ;)

/b

On Dec 10, 2010, at 2:02 PM, Mario G wrote:

> I googled a lot trying to figure this out, the strange part is the voicemail volume is always fine, it's only the emailed voicemail that is affected for particular callers.
> 
> On Dec 9, 2010, at 7:53 PM, Brian West wrote:
> 
>> Welcome to voip... google a little this is a common loss problem going to voicemail in some cases... 
>> 
>> /b
> 



From steveayre at gmail.com  Fri Dec 10 23:39:11 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 10 Dec 2010 20:39:11 +0000
Subject: [Freeswitch-users] G729B and mod_com_g729
In-Reply-To: <9FD71C2B-FDA6-43F9-BD3D-2762701C9620@freeswitch.org>
References: <AANLkTikK-aNyUWX_qVSAzVJvhBsX9KHZSRPbLwuVTQwM@mail.gmail.com>
	<AANLkTik+y_+LkoOsg9ytnyPtO72LU7PWBF1zbQCrEL9s@mail.gmail.com>
	<9FD71C2B-FDA6-43F9-BD3D-2762701C9620@freeswitch.org>
Message-ID: <549031E8-1165-457B-8A38-E70F9405211B@gmail.com>

Cool. That's why I was asking about a parameter. I was never suggesting making it default behaviour.

Steve on iPhone

On 10 Dec 2010, at 17:38, Brian West <brian at freeswitch.org> wrote:

> NDLB-allow-bad-iananame is the option you want on the profile.
> 
> The issue with the continued acceptance of broken things is what keeps getting us into this hole that we will never climb out of if we don't STOP IT.  If its wrong do NOT accept it and make the vendor fix their non-compliant crap otherwise this is only going to continue to get worse!
> 
> /b
> 
> On Dec 10, 2010, at 11:29 AM, Steven Ayre wrote:
> 
>> Brian, just out of curiosity (and I know this is bad bad bad bad) why
>> not have a compatibility setting so that codecs identified by static
>> IANA-assigned numbers (his example uses G729's assigned 18) can still
>> function. Or are there any broken devices that use the wrong numbers
>> that that might cause even problems with?
>> 
>> *ducks*
>> 
>> -Steve
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From gavin.henry at gmail.com  Sat Dec 11 00:47:54 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 10 Dec 2010 21:47:54 +0000
Subject: [Freeswitch-users] mod_easyroute and dialling via a gateway (not
 appending an IP to the return route)
Message-ID: <AANLkTimBB+u0xAcOcLk69=+0rXZD-MM8r67qFT3o8o9O@mail.gmail.com>

Hi all,

With easyroute how do I get it *not* to stick the gateway IP at the
end of THE ROUTE, i.e. to dial out of a gateway? I just want
sofia/gateway/suretec/441224279484 returned for entries that *are not*
in the database number table. It's fine to have @XX.XX.XX.XX for
entries *in* the database. I've checked mod_easyroute.c and I don't
think you can?

[DEBUG] mod_easyroute.c:242 THE ROUTE
[sofia/gateway/suretec/441224279484 at XX.XX.XX.XX]

My tables are like:

freeswitch=# select * from gateways;
 gateway_id | gateway_ip  |  group   | limit |      techprofile
------------+-------------+----------+-------+------------------------
          0 | XX.XX.XX.XX | suretec |   300 | sofia/gateway/suretec
(1 row)

freeswitch=# select * from numbers;
 number_id | gateway_id |    number    | acctcode | translated
-----------+------------+--------------+----------+------------
         0 |          0 | 441224279484 | 100      |
(1 row)


easyroute config:

    <!-- Default Technology and profile -->
    <param name="default-techprofile" value="sofia/gateway/suretec"/>

    <!-- IP or Hostname of Default Route -->
    <param name="default-gateway" value="suretec"/>


FS CLI results:

*In* the database (which is where I'm happy with an IP on the end as a
client will be getting the DID/DDI delivered to their IP address):

freeswitch at internal> easyroute 441224279484
Number    	Limit     	Group    	AcctCode  	Dialstring
441224279484	300       	surevoip  	100
	sofia/gateway/suretec/441224279484 at XX.XX.XX.XX


*Not* in the database (which is where *I'm not* happy with an IP on
the end as it needs to go through the gateway definition):

freeswitch at internal> easyroute 441224279485
Number    	Limit     	Group    	AcctCode  	Dialstring
441224279485	9999      	          	
	sofia/gateway/suretec/441224279485 at suretec


Thanks.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk


From krice at freeswitch.org  Sat Dec 11 01:27:19 2010
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 10 Dec 2010 16:27:19 -0600
Subject: [Freeswitch-users] mod_easyroute and dialling via a gateway
 (not appending an IP to the return route)
In-Reply-To: <AANLkTimBB+u0xAcOcLk69=+0rXZD-MM8r67qFT3o8o9O@mail.gmail.com>
Message-ID: <C9280567.5853%krice@freeswitch.org>

It was never designed to do that from the beginning however you might be
able the noat option and leave the gateway IP field blank to get the desired
result

However keep in mind that you probably don't even need to use a gateway
unless the gateway requires sip challenge/response authentication...

Easyroute was intended as a way for itsps and the like easily route incoming
calls to the appropriate proxy for handling incoming calls where IP
authentication is the authentication of choice


On 12/10/10 3:47 PM, "Gavin Henry" <gavin.henry at gmail.com> wrote:

> Hi all,
> 
> With easyroute how do I get it *not* to stick the gateway IP at the
> end of THE ROUTE, i.e. to dial out of a gateway? I just want
> sofia/gateway/suretec/441224279484 returned for entries that *are not*
> in the database number table. It's fine to have @XX.XX.XX.XX for
> entries *in* the database. I've checked mod_easyroute.c and I don't
> think you can?
> 
> [DEBUG] mod_easyroute.c:242 THE ROUTE
> [sofia/gateway/suretec/441224279484 at XX.XX.XX.XX]
> 
> My tables are like:
> 
> freeswitch=# select * from gateways;
>  gateway_id | gateway_ip  |  group   | limit |      techprofile
> ------------+-------------+----------+-------+------------------------
>           0 | XX.XX.XX.XX | suretec |   300 | sofia/gateway/suretec
> (1 row)
> 
> freeswitch=# select * from numbers;
>  number_id | gateway_id |    number    | acctcode | translated
> -----------+------------+--------------+----------+------------
>          0 |          0 | 441224279484 | 100      |
> (1 row)
> 
> 
> easyroute config:
> 
>     <!-- Default Technology and profile -->
>     <param name="default-techprofile" value="sofia/gateway/suretec"/>
> 
>     <!-- IP or Hostname of Default Route -->
>     <param name="default-gateway" value="suretec"/>
> 
> 
> FS CLI results:
> 
> *In* the database (which is where I'm happy with an IP on the end as a
> client will be getting the DID/DDI delivered to their IP address):
> 
> freeswitch at internal> easyroute 441224279484
> Number     Limit      Group     AcctCode   Dialstring
> 441224279484 300        surevoip   100
> sofia/gateway/suretec/441224279484 at XX.XX.XX.XX
> 
> 
> *Not* in the database (which is where *I'm not* happy with an IP on
> the end as it needs to go through the gateway definition):
> 
> freeswitch at internal> easyroute 441224279485
> Number     Limit      Group     AcctCode   Dialstring
> 441224279485 9999                  
> sofia/gateway/suretec/441224279485 at suretec
> 
> 
> Thanks.




From gavin.henry at gmail.com  Sat Dec 11 02:06:43 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 10 Dec 2010 23:06:43 +0000
Subject: [Freeswitch-users] mod_easyroute and dialling via a gateway
 (not appending an IP to the return route)
In-Reply-To: <C9280567.5853%krice@freeswitch.org>
References: <AANLkTimBB+u0xAcOcLk69=+0rXZD-MM8r67qFT3o8o9O@mail.gmail.com>
	<C9280567.5853%krice@freeswitch.org>
Message-ID: <AANLkTikG-X33+-DXD-iHAuSgpCxNAdQs2v6qSXg6CVVn@mail.gmail.com>

On 10 December 2010 22:27, Ken Rice <krice at freeswitch.org> wrote:
> It was never designed to do that from the beginning however you might be
> able the noat option and leave the gateway IP field blank to get the desired
> result

OK, thanks.

> However keep in mind that you probably don't even need to use a gateway
> unless the gateway requires sip challenge/response authentication...
>
> Easyroute was intended as a way for itsps and the like easily route incoming
> calls to the appropriate proxy for handling incoming calls where IP
> authentication is the authentication of choice

Understood. Will do some tests.

Thanks.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk


From steveu at coppice.org  Sat Dec 11 06:57:02 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 11 Dec 2010 11:57:02 +0800
Subject: [Freeswitch-users] G729B and mod_com_g729
In-Reply-To: <9FD71C2B-FDA6-43F9-BD3D-2762701C9620@freeswitch.org>
References: <AANLkTikK-aNyUWX_qVSAzVJvhBsX9KHZSRPbLwuVTQwM@mail.gmail.com>	<AANLkTik+y_+LkoOsg9ytnyPtO72LU7PWBF1zbQCrEL9s@mail.gmail.com>
	<9FD71C2B-FDA6-43F9-BD3D-2762701C9620@freeswitch.org>
Message-ID: <4D02F68E.9020407@coppice.org>

It is worth looking at Internet Explorer as an example of the sad 
consequences of tolerance. Microsoft made it so tolerant of wrongly 
tagged languages, file types, etc. that it now practically ignores all 
the tagging, even when the tagging is right. Perfectly good web pages 
get misinterpreted very frequently.

The tolerance of IE has made web page designers so sloppy that about 90% 
of the world's Chinese language web pages only display correctly if you 
set your locale to the appropriate type of Chinese (traditional or 
simplified), or manually force each page you visit. The designer has 
their locale set to Chinese, and never bothers checking beyond that.

If you have worked with things like webmail, you will know that the 
handling of files, like attached voice mails, in the displayed mail is 
very quirky in IE. By trying to be super tolerant it ends up with near 
random behaviour.

So, a short term move in the 90s by MS, which must have made people 
greatly thank them in the short term, has turned out to be a HUGE PITA.

Steve


On 12/11/2010 01:38 AM, Brian West wrote:
> NDLB-allow-bad-iananame is the option you want on the profile.
>
> The issue with the continued acceptance of broken things is what keeps getting us into this hole that we will never climb out of if we don't STOP IT.  If its wrong do NOT accept it and make the vendor fix their non-compliant crap otherwise this is only going to continue to get worse!
>
> /b
>
> On Dec 10, 2010, at 11:29 AM, Steven Ayre wrote:
>
>> Brian, just out of curiosity (and I know this is bad bad bad bad) why
>> not have a compatibility setting so that codecs identified by static
>> IANA-assigned numbers (his example uses G729's assigned 18) can still
>> function. Or are there any broken devices that use the wrong numbers
>> that that might cause even problems with?
>>
>> *ducks*
>>
>> -Steve



From phone.bytes at gmail.com  Sat Dec 11 11:04:13 2010
From: phone.bytes at gmail.com (phone.bytes)
Date: Sat, 11 Dec 2010 01:04:13 -0700
Subject: [Freeswitch-users] cepstral problem
In-Reply-To: <AANLkTinwWwfSY2uPhs5c8SuyxnwOn9myWv2u2WiKqGj8@mail.gmail.com>
References: <AANLkTi=tp_JK2H4cHtK8129TnWyYhArxaS2R1jmyPMPr@mail.gmail.com>	<4CFEA333.1090706@gmail.com>
	<AANLkTinwWwfSY2uPhs5c8SuyxnwOn9myWv2u2WiKqGj8@mail.gmail.com>
Message-ID: <4D03307D.9000104@gmail.com>

okay MC, mod_tts_commandline has run for a several days without a 
problem.  Good.

However, now we are trying to get this to speak from a text file instead 
of putting text on the command line.
Cepstral Swift has a -f option to do this, but it does not work here.

I have looked through the source, and am wondering if this is not 
supported in this mod.

Here is a partial trace that I think supports my conclusions below.

EXECUTE FreeTDM/1:1/7878030 speak(tts_commandline|Callie|-f/tmp/test.txt)
2010-12-11 00:26:58.683522 [DEBUG] switch_ivr_play_say.c:2237 OPEN TTS 
tts_commandline
2010-12-11 00:26:58.683522 [DEBUG] switch_ivr_play_say.c:2246 Raw Codec 
Activated
2010-12-11 00:26:58.683522 [DEBUG] mod_tts_commandline.c:147 Executing: 
swift -p audio/sampling-rate=8000 -n 'Callie' '-f/tmp/test.txt'  -o 
'/tmp/bd5bb8db-ffc2-4830-a971-24a1454c725c.tmp.wav'
2010-12-11 00:26:58.696525 [ERR] mod_sndfile.c:194 Error Opening File 
[/tmp/bd5bb8db-ffc2-4830-a971-24a1454c725c.tmp.wav] [System error : No 
such file or directory.]
2010-12-11 00:26:58.696525 [ERR] mod_tts_commandline.c:157 Failed to 
open file: /tmp/bd5bb8db-ffc2-4830-a971-24a1454c725c.tmp.wav
2010-12-11 00:26:58.696525 [DEBUG] switch_ivr_play_say.c:1935 Speaking 
text: -f/tmp/test.txt
2010-12-11 00:26:59.097509 [DEBUG] switch_ivr_play_say.c:2127 done 
speaking text

It seems to almost build the swift command correctly.  However, the -f 
options is inside the single quotes, instead of outside of them like the 
-o (output file) is.

I don't think it is finding the .txt file, as the log later shows that 
it is trying to speak the filename including the -f option, instead of 
the contents of the file.

What do you think?

Thanks



On 12/7/2010 2:17 PM, Michael Collins wrote:
> Can you try the mod_tts_commandline way of doing it? The wiki talks 
> about using Cepstral IIRC.
> -MC
>
> On Tue, Dec 7, 2010 at 1:12 PM, Phone <phone.bytes at gmail.com 
> <mailto:phone.bytes at gmail.com>> wrote:
>
>     Just wondering if you have had any luck in resolving this issue.
>
>     We are running Cent 5.5 on FreeSWITCH Version 1.0.head (git-8825b6e
>     2010-11-28 17-15-39 -0500) with Cepstral 5.1 with the Callie voice.
>
>     We have good audio for a short period of time, then suddenly the audio
>     is gone.  Watching the logs, it show that it is trying to play the
>     correct TTS, but it is silent.
>
>     We have tried to unload and reload mod_cepstral, but the only way we
>     have been able to restore the audio is with a FreeSWITCH restart.
>
>     These voices sound good, but with these issues, it is really not
>     useable.
>
>     Any Ideas?  Thanks
>
>
>

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From abid_freeswitch at live.com  Sat Dec 11 16:28:13 2010
From: abid_freeswitch at live.com (Abid Saleem)
Date: Sat, 11 Dec 2010 18:28:13 +0500
Subject: [Freeswitch-users] Radius AAA
Message-ID: <COL109-W556C528971964A0ED8308BF4100@phx.gbl>






Hi,
Sorry I was away from work. Just recalling the radius AAA and credit-time stuff, I am wondering if there is a way to announce the remaining time of 1 minute or 30 seconds before the call has to disconnect.
Please help if you FS supports this feature.
Regards-----------Abid Saleem

Date: Mon, 15 Nov 2010 08:53:23 +0100
From: tculjaga at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Radius AAA





     <action application="log" data="INFO  credit_time=${credit_time}"/>

     <action application="sched_hangup" data="+${credit_time:17:-1} ${Core-UUID}"/>



In this scenario you don't have to check for return code, because you will get hangup

if authentication fails.



Nice one Nazim :P

T.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From saeedahmad1981 at gmail.com  Sat Dec 11 16:48:19 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Sat, 11 Dec 2010 14:48:19 +0100
Subject: [Freeswitch-users] sipgate.de
In-Reply-To: <4CFEC3F5.2030600@gmail.com>
References: <104F5003-373A-465E-85E3-F6C154026B81@freeswitch.org>
	<4CFEC3F5.2030600@gmail.com>
Message-ID: <AANLkTikDt+G=ZoTVa2pgELvHyMX+NYKvE-sfeEhPMakx@mail.gmail.com>

yup i've their DID configured and working fine.


On Wed, Dec 8, 2010 at 12:32 AM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:

> i do, brian
> anything needed?
> thanks
> Le 08/12/2010 00:00, Brian West a ?crit :
> > Anyone have FS working with sipgate?
> >
> > /b
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Meftah Tayeb
> inum: +883510001288000
> Phone: +13602276297
> Fax: +12538020313
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From saeedahmad1981 at gmail.com  Sat Dec 11 17:00:53 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Sat, 11 Dec 2010 15:00:53 +0100
Subject: [Freeswitch-users] Scale UP Freeswitch
In-Reply-To: <AANLkTi=N+6=AcELMtbyje3O7W0k7qm3nz8z0CoO3LmLo@mail.gmail.com>
References: <AANLkTi=4n_wMSDE7KA9JTDhP+ToGggt-nJzf=EYwTd=6@mail.gmail.com>
	<AANLkTi=N+6=AcELMtbyje3O7W0k7qm3nz8z0CoO3LmLo@mail.gmail.com>
Message-ID: <AANLkTikkCTMpa-mwD79Fv3W8+1CwS4cMs8Hovdi0sfVS@mail.gmail.com>

Hi,

1. i am thinking to use kamailo in front of FS boxes, is there any
difference between kamailo and opensips?

2. if kamailo or opensips is running in front of FS, then will it send call
to FS with original customer ip? so i can do billing etc on FS box

-> actually i do IP based authentication and also ip based billing on FS
box, so in case, i recieve kamailo ip on FS box then i'll loose the original
customer overview.

thanks

On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre at gmail.com> wrote:

> There are a few performance tweaking tips at
> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
>
> Yes a Sangoma card will reduce your CPU load since transcoding won't
> be done on the CPU any longer, that will then mean there's more CPU
> available so you'll be able to handle more calls.
>
> However, if you're looking to increase your number of calls then you
> probably want a cluster of servers as Juan pointed out.
>
> It'll mean you can increase the capacity by adding extra servers, so
> there'd no longer be a limit to the number of calls you could handle
> (just add another server).
>
> It'll also make maintenance easier, as you'll be able to pull a server
> from service for updates etc while traffic continues to run on the
> other servers. Maintenance won't mean a service outage.
>
> If you're handling that many calls then additional servers would make
> your service more reliable. If a server crashes you'll still have the
> calls running on the other servers while you're fixing the problem so
> you won't have a complete outage. If FS is behind a load balancer then
> your customers might not even notice anything apart from a few dropped
> calls.
>
> There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will
> attempt to continue calls if FS crashes and restarts, but I think
> that's only for SIP-SIP not SIP-ISDN.
>
> -Steve
>
>
>
>
> On 7 December 2010 12:26, Stephen Wilde <wstephen80 at gmail.com> wrote:
> > Hi,
> > I have one server running Freeswitch with some ISDN connections (via
> > FreeTDM+Sangoma boards) and some SIP connections with service providers
> and
> > customer.
> > The usage of Freeswitch is as switching so it "bridge" each incoming call
> to
> > a new outgoing call.
> > SIP calls use G.729 and ISDN calls use ALaw for voice encoding.
> > Now the number of call is grow up and also the CPU load is a little high
> so
> > I have the necessity to scale UP my Freeswitch to handle more calls: what
> is
> > the best way to do that?
> > My first idea is to use a Sangoma D500 board to reduce the CPU load. Can
> be
> > this a solution?
> > There are different way to scale UP?
> > Thanks in advance,
> > Stephen
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From saeedahmad1981 at gmail.com  Sat Dec 11 17:09:16 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Sat, 11 Dec 2010 15:09:16 +0100
Subject: [Freeswitch-users] core dump
Message-ID: <AANLkTi=BgFiLtDwH5fhdoqe1aPHJLqhpsgR6GRCrAOV3@mail.gmail.com>

Dear List,

today my FS crashed with core dump

i used support-d to generate trace etc..

http://pastebin.freeswitch.org/14757

please let me know if i should open the jira.

thanks
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From acosgrov at gmail.com  Sat Dec 11 18:44:27 2010
From: acosgrov at gmail.com (Anthony Cosgrove)
Date: Sat, 11 Dec 2010 10:44:27 -0500
Subject: [Freeswitch-users] Email Voicemail Volume Mystery
In-Reply-To: <2D5B3151-A236-461C-8B30-BCCBAE7F6B2D@freeswitch.org>
References: <A3633C87-61A4-41BB-B6BD-B9477F9AB5DB@mgtech.com>
	<AANLkTinjBY9Qb6gWUXJQMrp6_FX3Bj2WSOsZ4cJw-ODo@mail.gmail.com>
	<ACC02B70-651D-47D5-9FC2-36662EC7A5F8@mgtech.com>
	<ADD52DC0-8496-45C1-942F-E72FE15DD1D4@freeswitch.org>
	<A82B230D-65DE-4A04-929E-2C2D0BA8BEDB@mgtech.com>
	<2D5B3151-A236-461C-8B30-BCCBAE7F6B2D@freeswitch.org>
Message-ID: <B2F5FEE4-1CDD-4F45-8D7F-0F331671C62E@gmail.com>

After seeing this I started thinking of a trick used in disasterisk.... using sox to normalize the volume. Maybe it will help here:

http://www.voip-info.org/wiki/view/sox

Anthony C




On Dec 10, 2010, at 3:30 PM, Brian West wrote:

> Its an on going issue for some... not sure what causes the loss but it does happen.  The alternative fix is to turn the volume knob up.  ;)
> 
> /b
> 
> On Dec 10, 2010, at 2:02 PM, Mario G wrote:
> 
>> I googled a lot trying to figure this out, the strange part is the voicemail volume is always fine, it's only the emailed voicemail that is affected for particular callers.
>> 
>> On Dec 9, 2010, at 7:53 PM, Brian West wrote:
>> 
>>> Welcome to voip... google a little this is a common loss problem going to voicemail in some cases... 
>>> 
>>> /b
>> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mario_fs at mgtech.com  Sat Dec 11 19:21:57 2010
From: mario_fs at mgtech.com (Mario G)
Date: Sat, 11 Dec 2010 08:21:57 -0800
Subject: [Freeswitch-users] Email Voicemail Volume Mystery
In-Reply-To: <B2F5FEE4-1CDD-4F45-8D7F-0F331671C62E@gmail.com>
References: <A3633C87-61A4-41BB-B6BD-B9477F9AB5DB@mgtech.com>
	<AANLkTinjBY9Qb6gWUXJQMrp6_FX3Bj2WSOsZ4cJw-ODo@mail.gmail.com>
	<ACC02B70-651D-47D5-9FC2-36662EC7A5F8@mgtech.com>
	<ADD52DC0-8496-45C1-942F-E72FE15DD1D4@freeswitch.org>
	<A82B230D-65DE-4A04-929E-2C2D0BA8BEDB@mgtech.com>
	<2D5B3151-A236-461C-8B30-BCCBAE7F6B2D@freeswitch.org>
	<B2F5FEE4-1CDD-4F45-8D7F-0F331671C62E@gmail.com>
Message-ID: <B03E8374-E823-4AFA-8453-5F89F6F8AF6B@mgtech.com>

Thanks, I will look into that. If I ever figure a way to fix it I will post. BTW, the volume is all the way up and the sound is barely audible.

On Dec 11, 2010, at 7:44 AM, Anthony Cosgrove wrote:

> After seeing this I started thinking of a trick used in disasterisk.... using sox to normalize the volume. Maybe it will help here:
> 
> http://www.voip-info.org/wiki/view/sox
> 
> Anthony C
> 
> 
> 
> 
> On Dec 10, 2010, at 3:30 PM, Brian West wrote:
> 
>> Its an on going issue for some... not sure what causes the loss but it does happen.  The alternative fix is to turn the volume knob up.  ;)
>> 
>> /b
>> 
>> On Dec 10, 2010, at 2:02 PM, Mario G wrote:
>> 
>>> I googled a lot trying to figure this out, the strange part is the voicemail volume is always fine, it's only the emailed voicemail that is affected for particular callers.
>>> 
>>> On Dec 9, 2010, at 7:53 PM, Brian West wrote:
>>> 
>>>> Welcome to voip... google a little this is a common loss problem going to voicemail in some cases... 
>>>> 
>>>> /b
>>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Sat Dec 11 19:37:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 11 Dec 2010 10:37:43 -0600
Subject: [Freeswitch-users] core dump
In-Reply-To: <AANLkTi=BgFiLtDwH5fhdoqe1aPHJLqhpsgR6GRCrAOV3@mail.gmail.com>
References: <AANLkTi=BgFiLtDwH5fhdoqe1aPHJLqhpsgR6GRCrAOV3@mail.gmail.com>
Message-ID: <AANLkTi=PcbxNSN6KNmz8uS9gSQ3VXoEtQeiiHXVFHV5=@mail.gmail.com>

report this to http://jira.freeswitch.org and be sure to categorize it
under the h323 module

On Sat, Dec 11, 2010 at 8:09 AM, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> Dear List,
> today my FS crashed with core dump
> i used support-d to generate trace etc..
> http://pastebin.freeswitch.org/14757
> please let me know if i should open the jira.
> thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From pbdlists at pinboard.com  Sun Dec 12 01:53:55 2010
From: pbdlists at pinboard.com (pbdlists at pinboard.com)
Date: Sat, 11 Dec 2010 23:53:55 +0100
Subject: [Freeswitch-users] possibly fixed: fax files not saved
In-Reply-To: <603F45904E8D4922A0EAF575BAA327D9@e1705>
References: <20101209195730.GA11419@pinboard.com>
	<603F45904E8D4922A0EAF575BAA327D9@e1705>
Message-ID: <20101211225355.GA16230@pinboard.com>

Today I spent a couple of hours testing various things in regard to this
problem. As far as I can tell, the files do not go missing if I have
    <action application="answer" />
    <action application="playback" data="silence_stream://2000"/>
coming before 
    <action application="set" data="api_hangup_hook=system ...

But if the api_hangup_hook comes before answer, then it happens. Why?
Well, that's beyond me.

Hope this info helps others too.

Kurt


On Thu, Dec 09, 2010 at 08:06:13PM -0500, Madovsky wrote:
> do you have any function or command that delete your fax after sent email ?
> if not, maybe try to delay the hangup...
> 
> 
> ----- Original Message ----- 
> From: <pbdlists at pinboard.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, December 09, 2010 2:57 PM
> Subject: Re: [Freeswitch-users] fax files not saved
> 
> 
> > Oops, didn't catch the double underbar (the called script is not
> > evaluating that parameter). Thanks, fixed it.
> >
> > You got me puzzled with your remark about session_in_hangup_hook. Had
> > to look it up on the wiki. My guess is that with session_in_hangup_hook
> > the variables get exported into the environment of the called
> > script/application and can be queried there directly. But I'm calling
> > the script with all the variables as parameters; the called script is
> > not evaluating the environment, but the command line parameters it was
> > passed.
> >
> > Meanwhile, anybody any idea about why the fax file is only written to
> > disk sometimes?
> >
> > Cheers,
> >
> > Kurt
> >
> > On Wed, Dec 08, 2010 at 11:30:34AM -0500, Madovsky wrote:
> >> you have first a typo
> >> fax__image_resolution
> >>
> >> the thing I don't understand is you didn't
> >> set session_in hangup_hook=true and
> >> you can retrieve the channel vars at api_hangup_hook...
> >>
> >>
> >>
> >> ----- Original Message ----- 
> >> From: <pbdlists at pinboard.com>
> >> To: "freeswitch-users" <FreeSWITCH-users at lists.freeswitch.org>
> >> Sent: Wednesday, December 08, 2010 10:30 AM
> >> Subject: [Freeswitch-users] fax files not saved
> >>
> >>
> >> > Hello!
> >> >
> >> > I managed to solve most of my other problems sofar, but with this one I
> >> > have no clue at all. Has anybody an idea what is happening here?
> >> >
> >> > 1 an inbound call to a number which is registered as fax (mod_spandsp)
> >> > 2 fax is detected and received according to the log (log entries see
> >> > below)
> >> > 3 but _sometimes_ the tiff file is never written to disk (post 
> >> > processing
> >> >   script can't find it and it is really not there)
> >> >
> >> > The destination filesystem has enough free space and free inodes,
> >> > permissions for writing are ok, the system is not under any kind of
> >> > heavy load, no other calls going on at the same time...
> >> >
> >> > I don't have much data to test this (using www.freepopfax.com for
> >> > testing and they have a daily limit on faxes I can send), but it seems
> >> > to happen roughly for 30-50% of the incoming fax messages. From the 
> >> > same
> >> > fax provider it works fine one time, then not a couple of minutes 
> >> > later,
> >> > another couple of minutes later it may work again, all while nothing at
> >> > all is being changed on the freeswitch server. If it was something like
> >> > a dropped line or anything I could understand it, but the logs say
> >> > everything is ok and still _sometimes_ the fax is not written to 
> >> > disk...
> >> >
> >> > Cheers,
> >> >
> >> > Kurt
> >> >
> >> > =======================================================================
> >> > entries in fax.conf.xml
> >> > =======================================================================
> >> > <configuration name="fax.conf" description="FAX application
> >> > configuration">
> >> >    <settings>
> >> >        <param name="use-ecm"           value="true"/>
> >> >        <param name="verbose"           value="false"/>
> >> >        <param name="disable-v17"       value="false"/>
> >> >        <param name="ident"             value="SpanDSP Fax Ident"/>
> >> >        <param name="header"            value="SpanDSP Fax Header"/>
> >> >
> >> >        <param name="spool-dir"         value="/tmp"/>
> >> >        <param name="file-prefix"       value="faxrx"/>
> >> >    </settings>
> >> > </configuration>
> >> >
> >> >
> >> > =======================================================================
> >> > the part from default.xml
> >> > =======================================================================
> >> >    <extension name="fax_receive">
> >> >        <condition field="destination_number" expression="^(1111)$">
> >> >            <action application="set" data="api_hangup_hook=system
> >> > ${base_dir}/scripts/emailfax.sh $1 /tmp/FAX-${uuid}.tiff
> >> > '\\\${fax_success}' '\\\${fax_result_code}' '\\\${fax_result_text}'
> >> > '\\\${fax_local_station_id}' '\\\${fax_remote_station_id}'
> >> > '\\\${fax_document_transferred_pages}' '\\\${fax_document_total_pages}'
> >> > '\\\${fax__image_resolution}' '\\\${fax_image_size}' 
> >> > '\\\${fax_bad_rows}'
> >> > '\\\${fax_transfer_rate}' '\\\${fax_v17_disabled}'
> >> > '\\\${fax_ecm_requested}' '\\\${fax_filename}' "/>
> >> >            <action application="answer" />
> >> >            <action application="playback" 
> >> > data="silence_stream://2000"/>
> >> > <!--
> >> >            <action application="set" 
> >> > data="fax_enable_t38_request=true"/>
> >> >            <action application="set" data="fax_enable_t38=true"/>
> >> > -->
> >> >            <action application="set" data="fax_ident=+41xxxxxxxxx"/>
> >> >            <action application="set" data="fax_disable_v17=1"/>
> >> >            <action application="set" data="fax_verbose=0"/>
> >> >            <action application="rxfax" data="/tmp/FAX-${uuid}.tiff"/>
> >> >            <action application="hangup"/>
> >> >        </condition>
> >> >    </extension>
> >> >
> >> >
> >> > =======================================================================
> >> > log entry from an unsuccessful incoming fax:
> >> > =======================================================================
> >> >
> >> > 2010-12-07 23:33:51.157588 [DEBUG] sofia_glue.c:182
> >> > sofia/external/Anonymous at anonymous.invalid image media sdp:
> >> > v=0
> >> > o=FreeSWITCH 1291733462 1291733465 IN IP4 xxx.xxx.xxx.xxx
> >> > s=FreeSWITCH
> >> > c=IN IP4 xxx.xxx.xxx.xxx
> >> > t=0 0
> >> > m=image 31358 udptl t38
> >> > a=T38FaxVersion:0
> >> > a=T38MaxBitRate:9600
> >> > a=T38FaxFillBitRemoval
> >> > a=T38FaxRateManagement:transferredTCF
> >> > a=T38FaxMaxBuffer:2000
> >> > a=T38FaxMaxDatagram:400
> >> > a=T38FaxUdpEC:t38UDPRedundancy
> >> >
> >> > 2010-12-07 23:33:51.157588 [DEBUG] sofia.c:4597 Channel
> >> > sofia/external/Anonymous at anonymous.invalid entering state 
> >> > [completed][200]
> >> > 2010-12-07 23:33:51.157588 [DEBUG] sofia.c:4597 Channel
> >> > sofia/external/Anonymous at anonymous.invalid entering state [ready][200]
> >> > 2010-12-07 23:33:58.814979 [DEBUG] switch_rtp.c:2544 Correct ip/port
> >> > confirmed.
> >> > 2010-12-07 23:35:09.772752 [DEBUG] switch_channel.c:2457
> >> > (sofia/external/Anonymous at anonymous.invalid) Callstate Change ACTIVE ->
> >> > HANGUP
> >> > 2010-12-07 23:35:09.772752 [NOTICE] sofia.c:528 Hangup
> >> > sofia/external/Anonymous at anonymous.invalid [CS_EXECUTE] 
> >> > [NORMAL_CLEARING]
> >> > 2010-12-07 23:35:09.772752 [DEBUG] switch_channel.c:2473 Send signal
> >> > sofia/external/Anonymous at anonymous.invalid [KILL]
> >> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_session.c:1057 Send 
> >> > signal
> >> > sofia/external/Anonymous at anonymous.invalid [BREAK]
> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:323
> >> > ==============================================================================
> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:329 Fax 
> >> > successfully
> >> > received.
> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:340 Remote station
> >> > id:
> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:341 Local station 
> >> > id:
> >> > +41xxxxxxxxx
> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:342 Pages
> >> > transferred: 2
> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:344 Total fax 
> >> > pages:
> >> > 2
> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:345 Image 
> >> > resolution:
> >> > 8031x7700
> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate:
> >> > 9600
> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:348 ECM status
> >> > on
> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:349 remote 
> >> > country:
> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:350 remote vendor:
> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:351 remote model:
> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:353
> >> > ==============================================================================
> >> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_codec.c:141
> >> > sofia/external/Anonymous at anonymous.invalid Restore previous codec 
> >> > PCMA:8.
> >> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_session.c:1933
> >> > sofia/external/Anonymous at anonymous.invalid skip receive message
> >> > [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> >> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:366
> >> > (sofia/external/Anonymous at anonymous.invalid) State EXECUTE going to 
> >> > sleep
> >> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:320
> >> > (sofia/external/Anonymous at anonymous.invalid) Running State Change
> >> > CS_HANGUP
> >> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:553
> >> > (sofia/external/Anonymous at anonymous.invalid) State HANGUP
> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_sofia.c:453
> >> > sofia/external/Anonymous at anonymous.invalid Overriding SIP cause 480 
> >> > with
> >> > 200 from the other leg
> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_sofia.c:459 Channel
> >> > sofia/external/Anonymous at anonymous.invalid hanging up, cause:
> >> > NORMAL_CLEARING
> >> > 2010-12-07 23:35:09.821455 [DEBUG] switch_core_state_machine.c:46
> >> > sofia/external/Anonymous at anonymous.invalid Standard HANGUP, cause:
> >> > NORMAL_CLEARING
> >> > 2010-12-07 23:35:09.821455 [DEBUG] switch_core_state_machine.c:553
> >> > (sofia/external/Anonymous at anonymous.invalid) State HANGUP going to 
> >> > sleep
> >> > 2010-12-07 23:35:09.822803 [NOTICE] mod_commands.c:4187 Executing 
> >> > command:
> >> > /usr/local/freeswitch/scripts/emailfax.sh 1111
> >> > /tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff '1' '0' 'OK'
> >> > '+41xxxxxxxxx' '' '2' '2
> >> > ' '' '0' '0' '9600' '1' '1'
> >> > '/tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff'
> >> > 2010-12-07 23:35:10.076623 [DEBUG] switch_core_state_machine.c:488 
> >> > Hangup
> >> > Command with no Session 
> >> > system(/usr/local/freeswitch/scripts/emailfax.sh
> >> > 1111 /tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff '1' '0'
> >> > 'OK' '+41xxxxxxxxx' '' '2' '2' '' '0' '0' '9600' '1' '1'
> >> > '/tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff' ):
> >> > +OK
> >> >
> >> >
> >> > =======================================================================
> >> > log entry from a successful incoming fax:
> >> > =======================================================================
> >> >
> >> > 2010-12-07 23:54:59.082301 [DEBUG] sofia_glue.c:182
> >> > sofia/external/Anonymous at anonymous.invalid image media sdp:
> >> > v=0
> >> > o=FreeSWITCH 1291735801 1291735804 IN IP4 xxx.xxx.xxx.xxx
> >> > s=FreeSWITCH
> >> > c=IN IP4 xxx.xxx.xxx.xxx
> >> > t=0 0
> >> > m=image 30280 udptl t38
> >> > a=T38FaxVersion:0
> >> > a=T38MaxBitRate:9600
> >> > a=T38FaxFillBitRemoval
> >> > a=T38FaxRateManagement:transferredTCF
> >> > a=T38FaxMaxBuffer:2000
> >> > a=T38FaxMaxDatagram:400
> >> > a=T38FaxUdpEC:t38UDPRedundancy
> >> >
> >> > 2010-12-07 23:54:59.082301 [DEBUG] sofia.c:4597 Channel
> >> > sofia/external/Anonymous at anonymous.invalid entering state 
> >> > [completed][200]
> >> > 2010-12-07 23:54:59.100811 [DEBUG] sofia.c:4597 Channel
> >> > sofia/external/Anonymous at anonymous.invalid entering state [ready][200]
> >> > 2010-12-07 23:55:06.783427 [DEBUG] switch_rtp.c:2544 Correct ip/port
> >> > confirmed.
> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:323
> >> > ==============================================================================
> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:329 Fax 
> >> > successfully
> >> > received.
> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:340 Remote station
> >> > id:
> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:341 Local station 
> >> > id:
> >> > +41xxxxxxxxx
> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:342 Pages
> >> > transferred: 1
> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:344 Total fax 
> >> > pages:
> >> > 1
> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:345 Image 
> >> > resolution:
> >> > 8031x7700
> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate:
> >> > 9600
> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:348 ECM status
> >> > on
> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:349 remote 
> >> > country:
> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:350 remote vendor:
> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:351 remote model:
> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:353
> >> > ==============================================================================
> >> > 2010-12-07 23:55:31.616038 [DEBUG] switch_channel.c:2457
> >> > (sofia/external/Anonymous at anonymous.invalid) Callstate Change ACTIVE ->
> >> > HANGUP
> >> > 2010-12-07 23:55:31.616038 [NOTICE] sofia.c:528 Hangup
> >> > sofia/external/Anonymous at anonymous.invalid [CS_EXECUTE] 
> >> > [NORMAL_CLEARING]
> >> > 2010-12-07 23:55:31.616038 [DEBUG] switch_channel.c:2473 Send signal
> >> > sofia/external/Anonymous at anonymous.invalid [KILL]
> >> > 2010-12-07 23:55:31.616038 [DEBUG] switch_core_session.c:1057 Send 
> >> > signal
> >> > sofia/external/Anonymous at anonymous.invalid [BREAK]
> >> > 2010-12-07 23:55:31.617356 [DEBUG] switch_core_codec.c:141
> >> > sofia/external/Anonymous at anonymous.invalid Restore previous codec 
> >> > PCMA:8.
> >> > 2010-12-07 23:55:31.617356 [DEBUG] switch_core_session.c:1933
> >> > sofia/external/Anonymous at anonymous.invalid skip receive message
> >> > [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> >> > 2010-12-07 23:55:31.617356 [DEBUG] switch_core_state_machine.c:366
> >> > (sofia/external/Anonymous at anonymous.invalid) State EXECUTE going to 
> >> > sleep
> >> > 2010-12-07 23:55:31.617356 [DEBUG] switch_core_state_machine.c:320
> >> > (sofia/external/Anonymous at anonymous.invalid) Running State Change
> >> > CS_HANGUP
> >> > 2010-12-07 23:55:31.618542 [DEBUG] switch_core_state_machine.c:553
> >> > (sofia/external/Anonymous at anonymous.invalid) State HANGUP
> >> > 2010-12-07 23:55:31.618542 [DEBUG] mod_sofia.c:453
> >> > sofia/external/Anonymous at anonymous.invalid Overriding SIP cause 480 
> >> > with
> >> > 200 from the other leg
> >> > 2010-12-07 23:55:31.618542 [DEBUG] mod_sofia.c:459 Channel
> >> > sofia/external/Anonymous at anonymous.invalid hanging up, cause:
> >> > NORMAL_CLEARING
> >> > 2010-12-07 23:55:31.721993 [DEBUG] switch_core_state_machine.c:46
> >> > sofia/external/Anonymous at anonymous.invalid Standard HANGUP, cause:
> >> > NORMAL_CLEARING
> >> > 2010-12-07 23:55:31.721993 [DEBUG] switch_core_state_machine.c:553
> >> > (sofia/external/Anonymous at anonymous.invalid) State HANGUP going to 
> >> > sleep
> >> > 2010-12-07 23:55:31.721993 [NOTICE] mod_commands.c:4187 Executing 
> >> > command:
> >> > /usr/local/freeswitch/scripts/emailfax.sh 1111
> >> > /tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff '1' '0' 'OK'
> >> > '+41xxxxxxxxx' '' '1' '1' '' '0' '0' '9600' '1' '1'
> >> > '/tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff'
> >> > 2010-12-07 23:55:31.904090 [DEBUG] switch_core_state_machine.c:488 
> >> > Hangup
> >> > Command with no Session 
> >> > system(/usr/local/freeswitch/scripts/emailfax.sh
> >> > 1111 /tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff '1' '0' 'OK'
> >> > '+41xxxxxxxxx' '' '1' '1' '' '0' '0' '9600' '1' '1'
> >> > '/tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff' ):
> >> > +OK
> >> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From steveayre at gmail.com  Sun Dec 12 14:23:02 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 12 Dec 2010 11:23:02 +0000
Subject: [Freeswitch-users] Scale UP Freeswitch
In-Reply-To: <AANLkTikkCTMpa-mwD79Fv3W8+1CwS4cMs8Hovdi0sfVS@mail.gmail.com>
References: <AANLkTi=4n_wMSDE7KA9JTDhP+ToGggt-nJzf=EYwTd=6@mail.gmail.com>
	<AANLkTi=N+6=AcELMtbyje3O7W0k7qm3nz8z0CoO3LmLo@mail.gmail.com>
	<AANLkTikkCTMpa-mwD79Fv3W8+1CwS4cMs8Hovdi0sfVS@mail.gmail.com>
Message-ID: <AANLkTikG=rPb-Lz7rMySa3gzJpjo42ftuthsPReD6ShJ@mail.gmail.com>

> 1. i am thinking to use kamailo in front of FS boxes, is there any
> difference between kamailo and opensips?

They're both forks of OpenSER so for the most part there's little difference.

There are some small differences though since the fork. For example,
opensips has a load_balancer module which kamalio does not (kamalio
can still do load balancing but has a different interface to do so).

> 2. if kamailo or opensips is running in front of FS, then will it send call
> to FS with original customer ip? so i can do billing etc on FS box
> -> actually i do IP based authentication and also ip based billing on FS
> box, so in case, i recieve kamailo ip on FS box then i'll loose the original
> customer overview.

It will appear coming from the proxy IP. But there is a workaround.
Configure a proxy ACL on the SIP profile and add your proxy IP to it.
Then adjust your proxy routing rules so that it adds a X-Auth-IP
header that contains the original IP.
Anything coming from anything in the proxy ACL is trusted and FS will
use the value from X-Auth-IP (if it exists).

-Steve




On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> Hi,
>
> 1. i am thinking to use kamailo in front of FS boxes, is there any
> difference between kamailo and opensips?
>
> 2. if kamailo or opensips is running in front of FS, then will it send call
> to FS with original customer ip? so i can do billing etc on FS box
> -> actually i do IP based authentication and also ip based billing on FS
> box, so in case, i recieve kamailo ip on FS box then i'll loose the original
> customer overview.
>
> thanks
> On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> There are a few performance tweaking tips at
>> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
>>
>> Yes a Sangoma card will reduce your CPU load since transcoding won't
>> be done on the CPU any longer, that will then mean there's more CPU
>> available so you'll be able to handle more calls.
>>
>> However, if you're looking to increase your number of calls then you
>> probably want a cluster of servers as Juan pointed out.
>>
>> It'll mean you can increase the capacity by adding extra servers, so
>> there'd no longer be a limit to the number of calls you could handle
>> (just add another server).
>>
>> It'll also make maintenance easier, as you'll be able to pull a server
>> from service for updates etc while traffic continues to run on the
>> other servers. Maintenance won't mean a service outage.
>>
>> If you're handling that many calls then additional servers would make
>> your service more reliable. If a server crashes you'll still have the
>> calls running on the other servers while you're fixing the problem so
>> you won't have a complete outage. If FS is behind a load balancer then
>> your customers might not even notice anything apart from a few dropped
>> calls.
>>
>> There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will
>> attempt to continue calls if FS crashes and restarts, but I think
>> that's only for SIP-SIP not SIP-ISDN.
>>
>> -Steve
>>
>>
>>
>>
>> On 7 December 2010 12:26, Stephen Wilde <wstephen80 at gmail.com> wrote:
>> > Hi,
>> > I have one server running Freeswitch with some ISDN connections (via
>> > FreeTDM+Sangoma boards) and some SIP connections with service providers
>> > and
>> > customer.
>> > The usage of Freeswitch is as switching so it "bridge" each incoming
>> > call to
>> > a new outgoing call.
>> > SIP calls use G.729 and ISDN calls use ALaw for voice encoding.
>> > Now the number of call is grow up and also the CPU load is a little high
>> > so
>> > I have the necessity to scale UP my Freeswitch to handle more calls:
>> > what is
>> > the best way to do that?
>> > My first idea is to use a Sangoma D500 board to reduce the CPU load. Can
>> > be
>> > this a solution?
>> > There are different way to scale UP?
>> > Thanks in advance,
>> > Stephen
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From a.afzali2003 at gmail.com  Sun Dec 12 14:38:54 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sun, 12 Dec 2010 15:08:54 +0330
Subject: [Freeswitch-users] Removing FIFO's on-hook agent in case of
	unsuccessful bridge
Message-ID: <AANLkTi=etZhfGPTVBrwNqx37i-6TatqomdM6sq_ky5Wr@mail.gmail.com>

Hi Guys,

There are some situations which logged in on-hook agents don't respond to
assigned calls by FIFO (USER_BUSY, NO_ANSWER, ...). I want to remove those
agents automatically. I've considered it already by using loop backing dial
string to another dial-plan and track what will happen to bridge
application. In this technique I have troubles with additional xml-cdr
posts, bridge_hangup_cause and transfer_after_bridge variables. I'll
appreciate if know there are some other techniques except this one.

BEST,
-- afshin
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From Nabble at slickdeals.endjunk.com  Sun Dec 12 16:48:34 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 12 Dec 2010 05:48:34 -0800 (PST)
Subject: [Freeswitch-users] libsofia: 1 thread/profile or 1 profile/thread
Message-ID: <1292161714030-5828064.post@n2.nabble.com>


When I read 
http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations#Recommended_SIP_settings
this , it says libsofia only handles 1 thread per profile. Is that correct
or is it more like 1 profile/thread?

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/libsofia-1-thread-profile-or-1-profile-thread-tp5828064p5828064.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Sun Dec 12 17:19:55 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 12 Dec 2010 08:19:55 -0600
Subject: [Freeswitch-users] libsofia: 1 thread/profile or 1
	profile/thread
In-Reply-To: <1292161714030-5828064.post@n2.nabble.com>
References: <1292161714030-5828064.post@n2.nabble.com>
Message-ID: <21F60DF4-EFEB-413B-B569-CB635777DFAC@freeswitch.org>

You're saying the exact same thing in both statements.

/b

On Dec 12, 2010, at 7:48 AM, mazilo wrote:

> this , it says libsofia only handles 1 thread per profile. Is that correct
> or is it more like 1 profile/thread?



From steveayre at gmail.com  Sun Dec 12 18:14:06 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 12 Dec 2010 15:14:06 +0000
Subject: [Freeswitch-users] libsofia: 1 thread/profile or 1
	profile/thread
In-Reply-To: <1292161714030-5828064.post@n2.nabble.com>
References: <1292161714030-5828064.post@n2.nabble.com>
Message-ID: <AANLkTi=Wc-YAXi5cydzucFeURXRU4EZ9U6BhW34TeC8i@mail.gmail.com>

Each profile runs in its own thread.

On 12 December 2010 13:48, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
> When I read
> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations#Recommended_SIP_settings
> this , it says libsofia only handles 1 thread per profile. Is that correct
> or is it more like 1 profile/thread?
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/libsofia-1-thread-profile-or-1-profile-thread-tp5828064p5828064.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From haloha201 at yahoo.com  Sun Dec 12 18:25:24 2010
From: haloha201 at yahoo.com (ha do)
Date: Sun, 12 Dec 2010 07:25:24 -0800 (PST)
Subject: [Freeswitch-users] what ports should open on firewall for skypopen
Message-ID: <822650.75966.qm@web32404.mail.mud.yahoo.com>

Hi All

my network topology:
    Freeswitch(skypopen module with multiple interfaces)---firewall ----internet

so which ports should i open on firewall to make the skypopen works properly
which ports for signaling  of skype
which ports for media of skype


Thank you
Ha`


      


From tayeb.meftah at gmail.com  Sun Dec 12 18:38:23 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sun, 12 Dec 2010 16:38:23 +0100
Subject: [Freeswitch-users] what ports should open on firewall for
	skypopen
In-Reply-To: <822650.75966.qm@web32404.mail.mud.yahoo.com>
References: <822650.75966.qm@web32404.mail.mud.yahoo.com>
Message-ID: <4D04EC6F.1050303@gmail.com>

no port required
skype can bypass firewalls
thanks
Le 12/12/2010 16:25, ha do a ?crit :
> Hi All
>
> my network topology:
>      Freeswitch(skypopen module with multiple interfaces)---firewall ----internet
>
> so which ports should i open on firewall to make the skypopen works properly
> which ports for signaling  of skype
> which ports for media of skype
>
>
> Thank you
> Ha`
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313



From infos at madovsky.org  Sun Dec 12 19:27:46 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 12 Dec 2010 11:27:46 -0500
Subject: [Freeswitch-users] possibly fixed: fax files not saved
References: <20101209195730.GA11419@pinboard.com><603F45904E8D4922A0EAF575BAA327D9@e1705>
	<20101211225355.GA16230@pinboard.com>
Message-ID: <7CB30055BC4243F4BBCA8380B574425C@e1705>

I think the api_hangup_hook can't come before if you don't answer ;)

----- Original Message ----- 
From: <pbdlists at pinboard.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, December 11, 2010 5:53 PM
Subject: [Freeswitch-users] possibly fixed: fax files not saved


> Today I spent a couple of hours testing various things in regard to this
> problem. As far as I can tell, the files do not go missing if I have
>    <action application="answer" />
>    <action application="playback" data="silence_stream://2000"/>
> coming before
>    <action application="set" data="api_hangup_hook=system ...
>
> But if the api_hangup_hook comes before answer, then it happens. Why?
> Well, that's beyond me.
>
> Hope this info helps others too.
>
> Kurt
>
>
> On Thu, Dec 09, 2010 at 08:06:13PM -0500, Madovsky wrote:
>> do you have any function or command that delete your fax after sent email 
>> ?
>> if not, maybe try to delay the hangup...
>>
>>
>> ----- Original Message ----- 
>> From: <pbdlists at pinboard.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Thursday, December 09, 2010 2:57 PM
>> Subject: Re: [Freeswitch-users] fax files not saved
>>
>>
>> > Oops, didn't catch the double underbar (the called script is not
>> > evaluating that parameter). Thanks, fixed it.
>> >
>> > You got me puzzled with your remark about session_in_hangup_hook. Had
>> > to look it up on the wiki. My guess is that with session_in_hangup_hook
>> > the variables get exported into the environment of the called
>> > script/application and can be queried there directly. But I'm calling
>> > the script with all the variables as parameters; the called script is
>> > not evaluating the environment, but the command line parameters it was
>> > passed.
>> >
>> > Meanwhile, anybody any idea about why the fax file is only written to
>> > disk sometimes?
>> >
>> > Cheers,
>> >
>> > Kurt
>> >
>> > On Wed, Dec 08, 2010 at 11:30:34AM -0500, Madovsky wrote:
>> >> you have first a typo
>> >> fax__image_resolution
>> >>
>> >> the thing I don't understand is you didn't
>> >> set session_in hangup_hook=true and
>> >> you can retrieve the channel vars at api_hangup_hook...
>> >>
>> >>
>> >>
>> >> ----- Original Message ----- 
>> >> From: <pbdlists at pinboard.com>
>> >> To: "freeswitch-users" <FreeSWITCH-users at lists.freeswitch.org>
>> >> Sent: Wednesday, December 08, 2010 10:30 AM
>> >> Subject: [Freeswitch-users] fax files not saved
>> >>
>> >>
>> >> > Hello!
>> >> >
>> >> > I managed to solve most of my other problems sofar, but with this 
>> >> > one I
>> >> > have no clue at all. Has anybody an idea what is happening here?
>> >> >
>> >> > 1 an inbound call to a number which is registered as fax 
>> >> > (mod_spandsp)
>> >> > 2 fax is detected and received according to the log (log entries see
>> >> > below)
>> >> > 3 but _sometimes_ the tiff file is never written to disk (post
>> >> > processing
>> >> >   script can't find it and it is really not there)
>> >> >
>> >> > The destination filesystem has enough free space and free inodes,
>> >> > permissions for writing are ok, the system is not under any kind of
>> >> > heavy load, no other calls going on at the same time...
>> >> >
>> >> > I don't have much data to test this (using www.freepopfax.com for
>> >> > testing and they have a daily limit on faxes I can send), but it 
>> >> > seems
>> >> > to happen roughly for 30-50% of the incoming fax messages. From the
>> >> > same
>> >> > fax provider it works fine one time, then not a couple of minutes
>> >> > later,
>> >> > another couple of minutes later it may work again, all while nothing 
>> >> > at
>> >> > all is being changed on the freeswitch server. If it was something 
>> >> > like
>> >> > a dropped line or anything I could understand it, but the logs say
>> >> > everything is ok and still _sometimes_ the fax is not written to
>> >> > disk...
>> >> >
>> >> > Cheers,
>> >> >
>> >> > Kurt
>> >> >
>> >> > =======================================================================
>> >> > entries in fax.conf.xml
>> >> > =======================================================================
>> >> > <configuration name="fax.conf" description="FAX application
>> >> > configuration">
>> >> >    <settings>
>> >> >        <param name="use-ecm"           value="true"/>
>> >> >        <param name="verbose"           value="false"/>
>> >> >        <param name="disable-v17"       value="false"/>
>> >> >        <param name="ident"             value="SpanDSP Fax Ident"/>
>> >> >        <param name="header"            value="SpanDSP Fax Header"/>
>> >> >
>> >> >        <param name="spool-dir"         value="/tmp"/>
>> >> >        <param name="file-prefix"       value="faxrx"/>
>> >> >    </settings>
>> >> > </configuration>
>> >> >
>> >> >
>> >> > =======================================================================
>> >> > the part from default.xml
>> >> > =======================================================================
>> >> >    <extension name="fax_receive">
>> >> >        <condition field="destination_number" expression="^(1111)$">
>> >> >            <action application="set" data="api_hangup_hook=system
>> >> > ${base_dir}/scripts/emailfax.sh $1 /tmp/FAX-${uuid}.tiff
>> >> > '\\\${fax_success}' '\\\${fax_result_code}' '\\\${fax_result_text}'
>> >> > '\\\${fax_local_station_id}' '\\\${fax_remote_station_id}'
>> >> > '\\\${fax_document_transferred_pages}' 
>> >> > '\\\${fax_document_total_pages}'
>> >> > '\\\${fax__image_resolution}' '\\\${fax_image_size}'
>> >> > '\\\${fax_bad_rows}'
>> >> > '\\\${fax_transfer_rate}' '\\\${fax_v17_disabled}'
>> >> > '\\\${fax_ecm_requested}' '\\\${fax_filename}' "/>
>> >> >            <action application="answer" />
>> >> >            <action application="playback"
>> >> > data="silence_stream://2000"/>
>> >> > <!--
>> >> >            <action application="set"
>> >> > data="fax_enable_t38_request=true"/>
>> >> >            <action application="set" data="fax_enable_t38=true"/>
>> >> > -->
>> >> >            <action application="set" data="fax_ident=+41xxxxxxxxx"/>
>> >> >            <action application="set" data="fax_disable_v17=1"/>
>> >> >            <action application="set" data="fax_verbose=0"/>
>> >> >            <action application="rxfax" 
>> >> > data="/tmp/FAX-${uuid}.tiff"/>
>> >> >            <action application="hangup"/>
>> >> >        </condition>
>> >> >    </extension>
>> >> >
>> >> >
>> >> > =======================================================================
>> >> > log entry from an unsuccessful incoming fax:
>> >> > =======================================================================
>> >> >
>> >> > 2010-12-07 23:33:51.157588 [DEBUG] sofia_glue.c:182
>> >> > sofia/external/Anonymous at anonymous.invalid image media sdp:
>> >> > v=0
>> >> > o=FreeSWITCH 1291733462 1291733465 IN IP4 xxx.xxx.xxx.xxx
>> >> > s=FreeSWITCH
>> >> > c=IN IP4 xxx.xxx.xxx.xxx
>> >> > t=0 0
>> >> > m=image 31358 udptl t38
>> >> > a=T38FaxVersion:0
>> >> > a=T38MaxBitRate:9600
>> >> > a=T38FaxFillBitRemoval
>> >> > a=T38FaxRateManagement:transferredTCF
>> >> > a=T38FaxMaxBuffer:2000
>> >> > a=T38FaxMaxDatagram:400
>> >> > a=T38FaxUdpEC:t38UDPRedundancy
>> >> >
>> >> > 2010-12-07 23:33:51.157588 [DEBUG] sofia.c:4597 Channel
>> >> > sofia/external/Anonymous at anonymous.invalid entering state
>> >> > [completed][200]
>> >> > 2010-12-07 23:33:51.157588 [DEBUG] sofia.c:4597 Channel
>> >> > sofia/external/Anonymous at anonymous.invalid entering state 
>> >> > [ready][200]
>> >> > 2010-12-07 23:33:58.814979 [DEBUG] switch_rtp.c:2544 Correct ip/port
>> >> > confirmed.
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] switch_channel.c:2457
>> >> > (sofia/external/Anonymous at anonymous.invalid) Callstate Change 
>> >> > ACTIVE ->
>> >> > HANGUP
>> >> > 2010-12-07 23:35:09.772752 [NOTICE] sofia.c:528 Hangup
>> >> > sofia/external/Anonymous at anonymous.invalid [CS_EXECUTE]
>> >> > [NORMAL_CLEARING]
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] switch_channel.c:2473 Send signal
>> >> > sofia/external/Anonymous at anonymous.invalid [KILL]
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_session.c:1057 Send
>> >> > signal
>> >> > sofia/external/Anonymous at anonymous.invalid [BREAK]
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:323
>> >> > ==============================================================================
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:329 Fax
>> >> > successfully
>> >> > received.
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:340 Remote 
>> >> > station
>> >> > id:
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:341 Local 
>> >> > station
>> >> > id:
>> >> > +41xxxxxxxxx
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:342 Pages
>> >> > transferred: 2
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:344 Total fax
>> >> > pages:
>> >> > 2
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:345 Image
>> >> > resolution:
>> >> > 8031x7700
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:346 Transfer 
>> >> > Rate:
>> >> > 9600
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:348 ECM status
>> >> > on
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:349 remote
>> >> > country:
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:350 remote 
>> >> > vendor:
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:351 remote 
>> >> > model:
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_spandsp_fax.c:353
>> >> > ==============================================================================
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_codec.c:141
>> >> > sofia/external/Anonymous at anonymous.invalid Restore previous codec
>> >> > PCMA:8.
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_session.c:1933
>> >> > sofia/external/Anonymous at anonymous.invalid skip receive message
>> >> > [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:366
>> >> > (sofia/external/Anonymous at anonymous.invalid) State EXECUTE going to
>> >> > sleep
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:320
>> >> > (sofia/external/Anonymous at anonymous.invalid) Running State Change
>> >> > CS_HANGUP
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] switch_core_state_machine.c:553
>> >> > (sofia/external/Anonymous at anonymous.invalid) State HANGUP
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_sofia.c:453
>> >> > sofia/external/Anonymous at anonymous.invalid Overriding SIP cause 480
>> >> > with
>> >> > 200 from the other leg
>> >> > 2010-12-07 23:35:09.772752 [DEBUG] mod_sofia.c:459 Channel
>> >> > sofia/external/Anonymous at anonymous.invalid hanging up, cause:
>> >> > NORMAL_CLEARING
>> >> > 2010-12-07 23:35:09.821455 [DEBUG] switch_core_state_machine.c:46
>> >> > sofia/external/Anonymous at anonymous.invalid Standard HANGUP, cause:
>> >> > NORMAL_CLEARING
>> >> > 2010-12-07 23:35:09.821455 [DEBUG] switch_core_state_machine.c:553
>> >> > (sofia/external/Anonymous at anonymous.invalid) State HANGUP going to
>> >> > sleep
>> >> > 2010-12-07 23:35:09.822803 [NOTICE] mod_commands.c:4187 Executing
>> >> > command:
>> >> > /usr/local/freeswitch/scripts/emailfax.sh 1111
>> >> > /tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff '1' '0' 'OK'
>> >> > '+41xxxxxxxxx' '' '2' '2
>> >> > ' '' '0' '0' '9600' '1' '1'
>> >> > '/tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff'
>> >> > 2010-12-07 23:35:10.076623 [DEBUG] switch_core_state_machine.c:488
>> >> > Hangup
>> >> > Command with no Session
>> >> > system(/usr/local/freeswitch/scripts/emailfax.sh
>> >> > 1111 /tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff '1' '0'
>> >> > 'OK' '+41xxxxxxxxx' '' '2' '2' '' '0' '0' '9600' '1' '1'
>> >> > '/tmp/FAX-6bf84d88-025a-11e0-b236-7391745d448b.tiff' ):
>> >> > +OK
>> >> >
>> >> >
>> >> > =======================================================================
>> >> > log entry from a successful incoming fax:
>> >> > =======================================================================
>> >> >
>> >> > 2010-12-07 23:54:59.082301 [DEBUG] sofia_glue.c:182
>> >> > sofia/external/Anonymous at anonymous.invalid image media sdp:
>> >> > v=0
>> >> > o=FreeSWITCH 1291735801 1291735804 IN IP4 xxx.xxx.xxx.xxx
>> >> > s=FreeSWITCH
>> >> > c=IN IP4 xxx.xxx.xxx.xxx
>> >> > t=0 0
>> >> > m=image 30280 udptl t38
>> >> > a=T38FaxVersion:0
>> >> > a=T38MaxBitRate:9600
>> >> > a=T38FaxFillBitRemoval
>> >> > a=T38FaxRateManagement:transferredTCF
>> >> > a=T38FaxMaxBuffer:2000
>> >> > a=T38FaxMaxDatagram:400
>> >> > a=T38FaxUdpEC:t38UDPRedundancy
>> >> >
>> >> > 2010-12-07 23:54:59.082301 [DEBUG] sofia.c:4597 Channel
>> >> > sofia/external/Anonymous at anonymous.invalid entering state
>> >> > [completed][200]
>> >> > 2010-12-07 23:54:59.100811 [DEBUG] sofia.c:4597 Channel
>> >> > sofia/external/Anonymous at anonymous.invalid entering state 
>> >> > [ready][200]
>> >> > 2010-12-07 23:55:06.783427 [DEBUG] switch_rtp.c:2544 Correct ip/port
>> >> > confirmed.
>> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:323
>> >> > ==============================================================================
>> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:329 Fax
>> >> > successfully
>> >> > received.
>> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:340 Remote 
>> >> > station
>> >> > id:
>> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:341 Local 
>> >> > station
>> >> > id:
>> >> > +41xxxxxxxxx
>> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:342 Pages
>> >> > transferred: 1
>> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:344 Total fax
>> >> > pages:
>> >> > 1
>> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:345 Image
>> >> > resolution:
>> >> > 8031x7700
>> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:346 Transfer 
>> >> > Rate:
>> >> > 9600
>> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:348 ECM status
>> >> > on
>> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:349 remote
>> >> > country:
>> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:350 remote 
>> >> > vendor:
>> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:351 remote 
>> >> > model:
>> >> > 2010-12-07 23:55:31.563208 [DEBUG] mod_spandsp_fax.c:353
>> >> > ==============================================================================
>> >> > 2010-12-07 23:55:31.616038 [DEBUG] switch_channel.c:2457
>> >> > (sofia/external/Anonymous at anonymous.invalid) Callstate Change 
>> >> > ACTIVE ->
>> >> > HANGUP
>> >> > 2010-12-07 23:55:31.616038 [NOTICE] sofia.c:528 Hangup
>> >> > sofia/external/Anonymous at anonymous.invalid [CS_EXECUTE]
>> >> > [NORMAL_CLEARING]
>> >> > 2010-12-07 23:55:31.616038 [DEBUG] switch_channel.c:2473 Send signal
>> >> > sofia/external/Anonymous at anonymous.invalid [KILL]
>> >> > 2010-12-07 23:55:31.616038 [DEBUG] switch_core_session.c:1057 Send
>> >> > signal
>> >> > sofia/external/Anonymous at anonymous.invalid [BREAK]
>> >> > 2010-12-07 23:55:31.617356 [DEBUG] switch_core_codec.c:141
>> >> > sofia/external/Anonymous at anonymous.invalid Restore previous codec
>> >> > PCMA:8.
>> >> > 2010-12-07 23:55:31.617356 [DEBUG] switch_core_session.c:1933
>> >> > sofia/external/Anonymous at anonymous.invalid skip receive message
>> >> > [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> >> > 2010-12-07 23:55:31.617356 [DEBUG] switch_core_state_machine.c:366
>> >> > (sofia/external/Anonymous at anonymous.invalid) State EXECUTE going to
>> >> > sleep
>> >> > 2010-12-07 23:55:31.617356 [DEBUG] switch_core_state_machine.c:320
>> >> > (sofia/external/Anonymous at anonymous.invalid) Running State Change
>> >> > CS_HANGUP
>> >> > 2010-12-07 23:55:31.618542 [DEBUG] switch_core_state_machine.c:553
>> >> > (sofia/external/Anonymous at anonymous.invalid) State HANGUP
>> >> > 2010-12-07 23:55:31.618542 [DEBUG] mod_sofia.c:453
>> >> > sofia/external/Anonymous at anonymous.invalid Overriding SIP cause 480
>> >> > with
>> >> > 200 from the other leg
>> >> > 2010-12-07 23:55:31.618542 [DEBUG] mod_sofia.c:459 Channel
>> >> > sofia/external/Anonymous at anonymous.invalid hanging up, cause:
>> >> > NORMAL_CLEARING
>> >> > 2010-12-07 23:55:31.721993 [DEBUG] switch_core_state_machine.c:46
>> >> > sofia/external/Anonymous at anonymous.invalid Standard HANGUP, cause:
>> >> > NORMAL_CLEARING
>> >> > 2010-12-07 23:55:31.721993 [DEBUG] switch_core_state_machine.c:553
>> >> > (sofia/external/Anonymous at anonymous.invalid) State HANGUP going to
>> >> > sleep
>> >> > 2010-12-07 23:55:31.721993 [NOTICE] mod_commands.c:4187 Executing
>> >> > command:
>> >> > /usr/local/freeswitch/scripts/emailfax.sh 1111
>> >> > /tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff '1' '0' 'OK'
>> >> > '+41xxxxxxxxx' '' '1' '1' '' '0' '0' '9600' '1' '1'
>> >> > '/tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff'
>> >> > 2010-12-07 23:55:31.904090 [DEBUG] switch_core_state_machine.c:488
>> >> > Hangup
>> >> > Command with no Session
>> >> > system(/usr/local/freeswitch/scripts/emailfax.sh
>> >> > 1111 /tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff '1' '0' 'OK'
>> >> > '+41xxxxxxxxx' '' '1' '1' '' '0' '0' '9600' '1' '1'
>> >> > '/tmp/FAX-5bd5649c-025d-11e0-b242-7391745d448b.tiff' ):
>> >> > +OK
>> >> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From steveayre at gmail.com  Sun Dec 12 21:52:01 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 12 Dec 2010 18:52:01 +0000
Subject: [Freeswitch-users] what ports should open on firewall for
	skypopen
In-Reply-To: <822650.75966.qm@web32404.mail.mud.yahoo.com>
References: <822650.75966.qm@web32404.mail.mud.yahoo.com>
Message-ID: <AANLkTinH5h5e4uP+2-GkZfxCW4TofEp2vkMdLKkahLPM@mail.gmail.com>

The same ports as Skype itself would require.

They have a firewall guide on their website:
http://www.skype.com/intl/en-us/support/user-guides/firewalls/

Almost all firewalls will allow Skype through by default, since it
uses outgoing connections on port 443. Everything else can then run
through that connection. You should only get problems on extremely
restrictive firewalls that filter outgoing traffic (most don't) and
which enforce the protocols using each port (most don't).

>From the website:
    *  Ideally, outgoing TCP connections to all ports (1.65535) should
be opened. This option results in Skype working most reliably. This is
only necessary for your Skype connection to be able to connect to the
Skype network and will not make your network any less secure.
    * If the above is not possible, open up outgoing TCP connections
to port 443. This will only work if you are using Skype version 0.97
and above.
    * If the above does not solve the problem, open up outgoing TCP
connections to port 80. Some firewalls restrict traffic to port 80 to
HTTP protocol, and in this case Skype can not use it since Skype does
not use HTTP. In some firewalls it is possible to open up all traffic
to port 80, not just HTTP, and in this case Skype will work.
    * If the above is not possible, Skype versions 0.97 and above can
use a HTTPS/SSL proxy. In order to do that, you have to configure the
proxy address in Internet Explorer options. Skype will then be able to
use it as well.
    * Please use our problem reporting form to report details of all
instances when you have experienced a problem with Skype and a
firewall.

-Steve



On 12 December 2010 15:25, ha do <haloha201 at yahoo.com> wrote:
> Hi All
>
> my network topology:
> ? ?Freeswitch(skypopen module with multiple interfaces)---firewall ----internet
>
> so which ports should i open on firewall to make the skypopen works properly
> which ports for signaling ?of skype
> which ports for media of skype
>
>
> Thank you
> Ha`
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From saeedahmad1981 at gmail.com  Sun Dec 12 22:28:55 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Sun, 12 Dec 2010 20:28:55 +0100
Subject: [Freeswitch-users] Scale UP Freeswitch
In-Reply-To: <AANLkTikG=rPb-Lz7rMySa3gzJpjo42ftuthsPReD6ShJ@mail.gmail.com>
References: <AANLkTi=4n_wMSDE7KA9JTDhP+ToGggt-nJzf=EYwTd=6@mail.gmail.com>
	<AANLkTi=N+6=AcELMtbyje3O7W0k7qm3nz8z0CoO3LmLo@mail.gmail.com>
	<AANLkTikkCTMpa-mwD79Fv3W8+1CwS4cMs8Hovdi0sfVS@mail.gmail.com>
	<AANLkTikG=rPb-Lz7rMySa3gzJpjo42ftuthsPReD6ShJ@mail.gmail.com>
Message-ID: <AANLkTinBf2Zc1o3Vagnha2C1YBba6YemjJjsf-UfCNht@mail.gmail.com>

Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me.

Since we are talking about HA options... Is it practically doable use it:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2

 <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2>The
idea is to run one FS box (Redirect-FS) in front of several FS boxes which
redirect the call to active/available FS. If we make some script on redirect
FS to count the active calls on media FSes and rearrange the order of
redirect then loadbalacing can also be done.

...possible?

On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre at gmail.com> wrote:

> > 1. i am thinking to use kamailo in front of FS boxes, is there any
> > difference between kamailo and opensips?
>
> They're both forks of OpenSER so for the most part there's little
> difference.
>
> There are some small differences though since the fork. For example,
> opensips has a load_balancer module which kamalio does not (kamalio
> can still do load balancing but has a different interface to do so).
>
> > 2. if kamailo or opensips is running in front of FS, then will it send
> call
> > to FS with original customer ip? so i can do billing etc on FS box
> > -> actually i do IP based authentication and also ip based billing on FS
> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
> original
> > customer overview.
>
> It will appear coming from the proxy IP. But there is a workaround.
> Configure a proxy ACL on the SIP profile and add your proxy IP to it.
> Then adjust your proxy routing rules so that it adds a X-Auth-IP
> header that contains the original IP.
> Anything coming from anything in the proxy ACL is trusted and FS will
> use the value from X-Auth-IP (if it exists).
>
> -Steve
>
>
>
>
> On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> > Hi,
> >
> > 1. i am thinking to use kamailo in front of FS boxes, is there any
> > difference between kamailo and opensips?
> >
> > 2. if kamailo or opensips is running in front of FS, then will it send
> call
> > to FS with original customer ip? so i can do billing etc on FS box
> > -> actually i do IP based authentication and also ip based billing on FS
> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
> original
> > customer overview.
> >
> > thanks
> > On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre at gmail.com> wrote:
> >>
> >> There are a few performance tweaking tips at
> >> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
> >>
> >> Yes a Sangoma card will reduce your CPU load since transcoding won't
> >> be done on the CPU any longer, that will then mean there's more CPU
> >> available so you'll be able to handle more calls.
> >>
> >> However, if you're looking to increase your number of calls then you
> >> probably want a cluster of servers as Juan pointed out.
> >>
> >> It'll mean you can increase the capacity by adding extra servers, so
> >> there'd no longer be a limit to the number of calls you could handle
> >> (just add another server).
> >>
> >> It'll also make maintenance easier, as you'll be able to pull a server
> >> from service for updates etc while traffic continues to run on the
> >> other servers. Maintenance won't mean a service outage.
> >>
> >> If you're handling that many calls then additional servers would make
> >> your service more reliable. If a server crashes you'll still have the
> >> calls running on the other servers while you're fixing the problem so
> >> you won't have a complete outage. If FS is behind a load balancer then
> >> your customers might not even notice anything apart from a few dropped
> >> calls.
> >>
> >> There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will
> >> attempt to continue calls if FS crashes and restarts, but I think
> >> that's only for SIP-SIP not SIP-ISDN.
> >>
> >> -Steve
> >>
> >>
> >>
> >>
> >> On 7 December 2010 12:26, Stephen Wilde <wstephen80 at gmail.com> wrote:
> >> > Hi,
> >> > I have one server running Freeswitch with some ISDN connections (via
> >> > FreeTDM+Sangoma boards) and some SIP connections with service
> providers
> >> > and
> >> > customer.
> >> > The usage of Freeswitch is as switching so it "bridge" each incoming
> >> > call to
> >> > a new outgoing call.
> >> > SIP calls use G.729 and ISDN calls use ALaw for voice encoding.
> >> > Now the number of call is grow up and also the CPU load is a little
> high
> >> > so
> >> > I have the necessity to scale UP my Freeswitch to handle more calls:
> >> > what is
> >> > the best way to do that?
> >> > My first idea is to use a Sangoma D500 board to reduce the CPU load.
> Can
> >> > be
> >> > this a solution?
> >> > There are different way to scale UP?
> >> > Thanks in advance,
> >> > Stephen
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rafonline at hotmail.com  Sun Dec 12 23:02:54 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Sun, 12 Dec 2010 20:02:54 +0000
Subject: [Freeswitch-users] mod_nibblebill ODBC issue
In-Reply-To: <COL122-W3739FC7490DFB018B97B5EB5110@phx.gbl>
References: <COL122-W5418BECBF3C852E0219589B5110@phx.gbl>,
	<COL122-W612DEC932A8C1F6F07FA5EB5110@phx.gbl>,
	<COL122-W3739FC7490DFB018B97B5EB5110@phx.gbl>
Message-ID: <COL122-W47AEDB0EEEFBE7D2CB91EFB5110@phx.gbl>



Hi,

I have a problem trying to get mod_nibblebill to work.  I have thought I had done everything to get it to work but obviously not:

2010-12-12 18:01:24.112241 [ERR] switch_odbc.c:313 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified

2010-12-12 18:01:24.112413 [CRIT] mod_nibblebill.c:223 Cannot connect to ODBC driver/database bandwidth.com (user: bandwidth.com / pass password)!

Do I need to update more than just the nibblebill conf file? 

Any help is much appreciated

CHeers

Raf


My /etc/odbc.ini file:

[FreeSWITCH]
Driver   = MySQL
SERVER   = localhost
PORT     = 3306
DATABASE = FreeSWITCH
OPTION  = 67108864
Socket   = /var/lib/mysql/mysql.sock

My /etc/odbcinst.ini file:

# Example driver definitinions
#
#
# Included in the unixODBC package
#[PostgreSQL]
#Description    = ODBC for PostgreSQL
#Driver         = /usr/lib/libodbcpsql.so
#Setup          = /usr/lib/libodbcpsqlS.so
#FileUsage      = 1

# Driver from the MyODBC package
# Setup from the unixODBC package
[MySQL]
Description     = ODBC for MySQL
Driver            = /usr/lib64/libmyodbc3.so
Setup            = /usr/lib64/libodbcmyS.so
FileUsage      = 1

Snippet of my nibblebill.conf.xml file:

<!-- Information for connecting to your database -->
<param name="db_username" value="rafqat"/>
<param name="db_password" value="?????"/>
<param name="db_dsn" value="FreeSWITCH"/>


I have even setup the following links:
mkdir /usr/local/freeswitch/etc
ln -s /etc/odbcinst.ini /usr/local/freeswitch/etc/odbcinst.ini
ln -s /etc/odbc.ini /usr/local/freeswitch/etc/odbc.ini

I can connect successfully using isql:

isql FreeSWITCH rafqat ??????







 		 	   		  
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From infos at madovsky.org  Sun Dec 12 23:11:32 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 12 Dec 2010 15:11:32 -0500
Subject: [Freeswitch-users] mod_nibblebill ODBC issue
References: <COL122-W5418BECBF3C852E0219589B5110@phx.gbl>,
	<COL122-W612DEC932A8C1F6F07FA5EB5110@phx.gbl>,
	<COL122-W3739FC7490DFB018B97B5EB5110@phx.gbl>
	<COL122-W47AEDB0EEEFBE7D2CB91EFB5110@phx.gbl>
Message-ID: <1F070BD848664E029DA01D63EC529A8E@e1705>

you have to specify a unix odbc drivers,
usually in /etc/odbc.ini /etc/odbcinst.ini
  ----- Original Message ----- 
  From: Rafqat . 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, December 12, 2010 3:02 PM
  Subject: [Freeswitch-users] mod_nibblebill ODBC issue



  Hi,

  I have a problem trying to get mod_nibblebill to work.  I have thought I had done everything to get it to work but obviously not:

  2010-12-12 18:01:24.112241 [ERR] switch_odbc.c:313 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified

  2010-12-12 18:01:24.112413 [CRIT] mod_nibblebill.c:223 Cannot connect to ODBC driver/database bandwidth.com (user: bandwidth.com / pass password)!

  Do I need to update more than just the nibblebill conf file? 

  Any help is much appreciated

  CHeers

  Raf


  My /etc/odbc.ini file:

  [FreeSWITCH]
  Driver   = MySQL
  SERVER   = localhost
  PORT     = 3306
  DATABASE = FreeSWITCH
  OPTION  = 67108864
  Socket   = /var/lib/mysql/mysql.sock

  My /etc/odbcinst.ini file:

  # Example driver definitinions
  #
  #
  # Included in the unixODBC package
  #[PostgreSQL]
  #Description    = ODBC for PostgreSQL
  #Driver         = /usr/lib/libodbcpsql.so
  #Setup          = /usr/lib/libodbcpsqlS.so
  #FileUsage      = 1

  # Driver from the MyODBC package
  # Setup from the unixODBC package
  [MySQL]
  Description     = ODBC for MySQL
  Driver            = /usr/lib64/libmyodbc3.so
  Setup            = /usr/lib64/libodbcmyS.so
  FileUsage      = 1

  Snippet of my nibblebill.conf.xml file:

  <!-- Information for connecting to your database -->
  <param name="db_username" value="rafqat"/>
  <param name="db_password" value="?????"/>
  <param name="db_dsn" value="FreeSWITCH"/>


  I have even setup the following links:

mkdir /usr/local/freeswitch/etcln -s /etc/odbcinst.ini /usr/local/freeswitch/etc/odbcinst.iniln -s /etc/odbc.ini /usr/local/freeswitch/etc/odbc.iniI can connect successfully using isql:isql FreeSWITCH rafqat ??????








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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From rafonline at hotmail.com  Sun Dec 12 23:19:18 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Sun, 12 Dec 2010 20:19:18 +0000
Subject: [Freeswitch-users] mod_nibblebill ODBC issue
In-Reply-To: <1F070BD848664E029DA01D63EC529A8E@e1705>
References: <COL122-W5418BECBF3C852E0219589B5110@phx.gbl>, ,
	<COL122-W612DEC932A8C1F6F07FA5EB5110@phx.gbl>, ,
	<COL122-W3739FC7490DFB018B97B5EB5110@phx.gbl>,
	<COL122-W47AEDB0EEEFBE7D2CB91EFB5110@phx.gbl>,
	<1F070BD848664E029DA01D63EC529A8E@e1705>
Message-ID: <COL122-W1312F9EEB16A140EA52334B5110@phx.gbl>



More than what I have already specified?

My /etc/odbc.ini 
  file:

[FreeSWITCH]
Driver   = MySQL
SERVER   
  = localhost
PORT     = 3306
DATABASE = 
  FreeSWITCH
OPTION  = 67108864
Socket   = 
  /var/lib/mysql/mysql.sock

My /etc/odbcinst.ini file:

# Example 
  driver definitinions
#
#
# Included in the unixODBC 
  package
#[PostgreSQL]
#Description    = ODBC for 
  PostgreSQL
#Driver         = 
  /usr/lib/libodbcpsql.so
#Setup          
  = /usr/lib/libodbcpsqlS.so
#FileUsage      = 
  1

# Driver from the MyODBC package
# Setup from the unixODBC 
  package
[MySQL]
Description     = ODBC for 
  MySQL
Driver            
  = 
  /usr/lib64/libmyodbc3.so
Setup            
  = /usr/lib64/libodbcmyS.so
FileUsage      = 
  1

From: infos at madovsky.org
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 12 Dec 2010 15:11:32 -0500
Subject: Re: [Freeswitch-users] mod_nibblebill ODBC issue










you have to specify a unix odbc drivers,
usually in /etc/odbc.ini /etc/odbcinst.ini

  ----- Original Message ----- 
  From: 
  Rafqat 
  . 
  To: freeswitch-users at lists.freeswitch.org 
  
  Sent: Sunday, December 12, 2010 3:02 
  PM
  Subject: [Freeswitch-users] 
  mod_nibblebill ODBC issue
  

Hi,

I have a problem trying to get mod_nibblebill to 
  work.  I have thought I had done everything to get it to work but 
  obviously not:

2010-12-12 18:01:24.112241 [ERR] switch_odbc.c:313 
  STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not 
  found, and no default driver specified

2010-12-12 18:01:24.112413 
  [CRIT] mod_nibblebill.c:223 Cannot connect to ODBC driver/database 
  bandwidth.com (user: bandwidth.com / pass password)!

Do I need to 
  update more than just the nibblebill conf file? 

Any help is much 
  appreciated

CHeers

Raf


My /etc/odbc.ini 
  file:

[FreeSWITCH]
Driver   = MySQL
SERVER   
  = localhost
PORT     = 3306
DATABASE = 
  FreeSWITCH
OPTION  = 67108864
Socket   = 
  /var/lib/mysql/mysql.sock

My /etc/odbcinst.ini file:

# Example 
  driver definitinions
#
#
# Included in the unixODBC 
  package
#[PostgreSQL]
#Description    = ODBC for 
  PostgreSQL
#Driver         = 
  /usr/lib/libodbcpsql.so
#Setup          
  = /usr/lib/libodbcpsqlS.so
#FileUsage      = 
  1

# Driver from the MyODBC package
# Setup from the unixODBC 
  package
[MySQL]
Description     = ODBC for 
  MySQL
Driver            
  = 
  /usr/lib64/libmyodbc3.so
Setup            
  = /usr/lib64/libodbcmyS.so
FileUsage      = 
  1

Snippet of my nibblebill.conf.xml file:

<!-- Information 
  for connecting to your database -->
<param name="db_username" 
  value="rafqat"/>
<param name="db_password" 
  value="?????"/>
<param name="db_dsn" 
  value="FreeSWITCH"/>


I have even setup the following links:
mkdir /usr/local/freeswitch/etc
ln -s /etc/odbcinst.ini /usr/local/freeswitch/etc/odbcinst.ini
ln -s /etc/odbc.ini /usr/local/freeswitch/etc/odbc.ini

I can connect successfully using isql:

isql FreeSWITCH rafqat ??????








  
  

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


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
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From infos at madovsky.org  Sun Dec 12 23:27:39 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 12 Dec 2010 15:27:39 -0500
Subject: [Freeswitch-users] mod_nibblebill ODBC issue
References: <COL122-W5418BECBF3C852E0219589B5110@phx.gbl>, ,
	<COL122-W612DEC932A8C1F6F07FA5EB5110@phx.gbl>, ,
	<COL122-W3739FC7490DFB018B97B5EB5110@phx.gbl>,
	<COL122-W47AEDB0EEEFBE7D2CB91EFB5110@phx.gbl>,
	<1F070BD848664E029DA01D63EC529A8E@e1705>
	<COL122-W1312F9EEB16A140EA52334B5110@phx.gbl>
Message-ID: <0281F1808ACB4AA1A39EFCB3035A1055@e1705>

what did you write on every
db_dsn in Freeswitch config ?
  ----- Original Message ----- 
  From: Rafqat . 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, December 12, 2010 3:19 PM
  Subject: Re: [Freeswitch-users] mod_nibblebill ODBC issue



  More than what I have already specified?

  My /etc/odbc.ini file:

  [FreeSWITCH]
  Driver   = MySQL
  SERVER   = localhost
  PORT     = 3306
  DATABASE = FreeSWITCH
  OPTION  = 67108864
  Socket   = /var/lib/mysql/mysql.sock

  My /etc/odbcinst.ini file:

  # Example driver definitinions
  #
  #
  # Included in the unixODBC package
  #[PostgreSQL]
  #Description    = ODBC for PostgreSQL
  #Driver         = /usr/lib/libodbcpsql.so
  #Setup          = /usr/lib/libodbcpsqlS.so
  #FileUsage      = 1

  # Driver from the MyODBC package
  # Setup from the unixODBC package
  [MySQL]
  Description     = ODBC for MySQL
  Driver            = /usr/lib64/libmyodbc3.so
  Setup            = /usr/lib64/libodbcmyS.so
  FileUsage      = 1


------------------------------------------------------------------------------
  From: infos at madovsky.org
  To: freeswitch-users at lists.freeswitch.org
  Date: Sun, 12 Dec 2010 15:11:32 -0500
  Subject: Re: [Freeswitch-users] mod_nibblebill ODBC issue


  you have to specify a unix odbc drivers,
  usually in /etc/odbc.ini /etc/odbcinst.ini
    ----- Original Message ----- 
    From: Rafqat . 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Sunday, December 12, 2010 3:02 PM
    Subject: [Freeswitch-users] mod_nibblebill ODBC issue



    Hi,

    I have a problem trying to get mod_nibblebill to work.  I have thought I had done everything to get it to work but obviously not:

    2010-12-12 18:01:24.112241 [ERR] switch_odbc.c:313 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified

    2010-12-12 18:01:24.112413 [CRIT] mod_nibblebill.c:223 Cannot connect to ODBC driver/database bandwidth.com (user: bandwidth.com / pass password)!

    Do I need to update more than just the nibblebill conf file? 

    Any help is much appreciated

    CHeers

    Raf


    My /etc/odbc.ini file:

    [FreeSWITCH]
    Driver   = MySQL
    SERVER   = localhost
    PORT     = 3306
    DATABASE = FreeSWITCH
    OPTION  = 67108864
    Socket   = /var/lib/mysql/mysql.sock

    My /etc/odbcinst.ini file:

    # Example driver definitinions
    #
    #
    # Included in the unixODBC package
    #[PostgreSQL]
    #Description    = ODBC for PostgreSQL
    #Driver         = /usr/lib/libodbcpsql.so
    #Setup          = /usr/lib/libodbcpsqlS.so
    #FileUsage      = 1

    # Driver from the MyODBC package
    # Setup from the unixODBC package
    [MySQL]
    Description     = ODBC for MySQL
    Driver            = /usr/lib64/libmyodbc3.so
    Setup            = /usr/lib64/libodbcmyS.so
    FileUsage      = 1

    Snippet of my nibblebill.conf.xml file:

    <!-- Information for connecting to your database -->
    <param name="db_username" value="rafqat"/>
    <param name="db_password" value="?????"/>
    <param name="db_dsn" value="FreeSWITCH"/>


    I have even setup the following links:

mkdir /usr/local/freeswitch/etcln -s /etc/odbcinst.ini /usr/local/freeswitch/etc/odbcinst.iniln -s /etc/odbc.ini /usr/local/freeswitch/etc/odbc.iniI can connect successfully using isql:isql FreeSWITCH rafqat ??????







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

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
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From infos at madovsky.org  Sun Dec 12 23:30:24 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 12 Dec 2010 15:30:24 -0500
Subject: [Freeswitch-users] send dtmf on one leg bridge
Message-ID: <D10B61781EFA4CFB913F2FEAC7579B27@e1705>

Is it possible to send dtfmf via
send_dtmf from CLI on a one leg bridge ?
it's for enter a pin conference and I can't use
any RFC on my sip phone.

Thanks

Franck
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From rafonline at hotmail.com  Sun Dec 12 23:36:03 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Sun, 12 Dec 2010 20:36:03 +0000
Subject: [Freeswitch-users] mod_nibblebill ODBC issue
In-Reply-To: <0281F1808ACB4AA1A39EFCB3035A1055@e1705>
References: <COL122-W5418BECBF3C852E0219589B5110@phx.gbl>,
	,,<COL122-W612DEC932A8C1F6F07FA5EB5110@phx.gbl>,
	, , <COL122-W3739FC7490DFB018B97B5EB5110@phx.gbl>, ,
	<COL122-W47AEDB0EEEFBE7D2CB91EFB5110@phx.gbl>, ,
	<1F070BD848664E029DA01D63EC529A8E@e1705>,
	<COL122-W1312F9EEB16A140EA52334B5110@phx.gbl>,
	<0281F1808ACB4AA1A39EFCB3035A1055@e1705>
Message-ID: <COL122-W5179D1FE2DB91024F97067B5110@phx.gbl>


I only updated the nibblebill config file with the following: 
<!-- Information for connecting to your database -->
<param name="db_username" value="rafqat"/>
<param name="db_password" value="?????"/>
<param name="db_dsn" value="FreeSWITCH"/>

I did not change any of the dsn related stuff in any of the other FreeSWITCH config files.

I dont see why I should be changing other freeswitch config files, can't one use a different database for the billing side of things?  It seems odd to allow dsn config to be added to the nibblebill conf file but not actually get used?

or am i missing the point?

Cheers

Raf


From: infos at madovsky.org
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 12 Dec 2010 15:27:39 -0500
Subject: Re: [Freeswitch-users] mod_nibblebill ODBC issue










what did you write on every
db_dsn in Freeswitch config ?

  ----- Original Message ----- 
  From: 
  Rafqat 
  . 
  To: freeswitch-users at lists.freeswitch.org 
  
  Sent: Sunday, December 12, 2010 3:19 
  PM
  Subject: Re: [Freeswitch-users] 
  mod_nibblebill ODBC issue
  

More than what I have already specified?

My 
  /etc/odbc.ini file:

[FreeSWITCH]
Driver   = 
  MySQL
SERVER   = localhost
PORT     = 
  3306
DATABASE = FreeSWITCH
OPTION  = 67108864
Socket   
  = /var/lib/mysql/mysql.sock

My /etc/odbcinst.ini file:

# Example 
  driver definitinions
#
#
# Included in the unixODBC 
  package
#[PostgreSQL]
#Description    = ODBC for 
  PostgreSQL
#Driver         = 
  /usr/lib/libodbcpsql.so
#Setup          
  = /usr/lib/libodbcpsqlS.so
#FileUsage      = 
  1

# Driver from the MyODBC package
# Setup from the unixODBC 
  package
[MySQL]
Description     = ODBC for 
  MySQL
Driver            
  = 
  /usr/lib64/libmyodbc3.so
Setup            
  = /usr/lib64/libodbcmyS.so
FileUsage      = 
  1


  
  From: infos at madovsky.org
To: freeswitch-users at lists.freeswitch.org
Date: 
  Sun, 12 Dec 2010 15:11:32 -0500
Subject: Re: [Freeswitch-users] 
  mod_nibblebill ODBC issue


  
  

  you have to specify a unix odbc drivers,
  usually in /etc/odbc.ini /etc/odbcinst.ini
  
    ----- Original Message ----- 
    From: Rafqat . 
    
    To: freeswitch-users at lists.freeswitch.org 
    
    Sent: Sunday, December 12, 2010 3:02 
    PM
    Subject: [Freeswitch-users] 
    mod_nibblebill ODBC issue
    

Hi,

I have a problem trying to get mod_nibblebill 
    to work.  I have thought I had done everything to get it to work but 
    obviously not:

2010-12-12 18:01:24.112241 [ERR] switch_odbc.c:313 
    STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not 
    found, and no default driver specified

2010-12-12 18:01:24.112413 
    [CRIT] mod_nibblebill.c:223 Cannot connect to ODBC driver/database 
    bandwidth.com (user: bandwidth.com / pass password)!

Do I need to 
    update more than just the nibblebill conf file? 

Any help is much 
    appreciated

CHeers

Raf


My /etc/odbc.ini 
    file:

[FreeSWITCH]
Driver   = 
    MySQL
SERVER   = localhost
PORT     = 
    3306
DATABASE = FreeSWITCH
OPTION  = 
    67108864
Socket   = /var/lib/mysql/mysql.sock

My 
    /etc/odbcinst.ini file:

# Example driver definitinions
#
#
# 
    Included in the unixODBC 
    package
#[PostgreSQL]
#Description    = ODBC for 
    PostgreSQL
#Driver         = 
    /usr/lib/libodbcpsql.so
#Setup          
    = /usr/lib/libodbcpsqlS.so
#FileUsage      = 
    1

# Driver from the MyODBC package
# Setup from the unixODBC 
    package
[MySQL]
Description     = ODBC for 
    MySQL
Driver            
    = 
    /usr/lib64/libmyodbc3.so
Setup            
    = /usr/lib64/libodbcmyS.so
FileUsage      = 
    1

Snippet of my nibblebill.conf.xml file:

<!-- Information 
    for connecting to your database -->
<param name="db_username" 
    value="rafqat"/>
<param name="db_password" 
    value="?????"/>
<param name="db_dsn" 
    value="FreeSWITCH"/>


I have even setup the following 
links:
mkdir /usr/local/freeswitch/etc
ln -s /etc/odbcinst.ini /usr/local/freeswitch/etc/odbcinst.ini
ln -s /etc/odbc.ini /usr/local/freeswitch/etc/odbc.ini

I can connect successfully using isql:

isql FreeSWITCH rafqat ??????









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

_______________________________________________ 
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
  http://www.freeswitch.org 
  
  

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


_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Sun Dec 12 23:50:20 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 12 Dec 2010 15:50:20 -0500
Subject: [Freeswitch-users] mod_nibblebill ODBC issue
References: <COL122-W5418BECBF3C852E0219589B5110@phx.gbl>, , ,
	<COL122-W612DEC932A8C1F6F07FA5EB5110@phx.gbl>, , ,
	<COL122-W3739FC7490DFB018B97B5EB5110@phx.gbl>, ,
	<COL122-W47AEDB0EEEFBE7D2CB91EFB5110@phx.gbl>, ,
	<1F070BD848664E029DA01D63EC529A8E@e1705>,
	<COL122-W1312F9EEB16A140EA52334B5110@phx.gbl>,
	<0281F1808ACB4AA1A39EFCB3035A1055@e1705>
	<COL122-W5179D1FE2DB91024F97067B5110@phx.gbl>
Message-ID: <DC864B84C89C47C9B1404DEA84D6DC1F@e1705>

Mabe your system env variables are not set.
it looks like odbc can't find the path...
  ----- Original Message ----- 
  From: Rafqat . 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, December 12, 2010 3:36 PM
  Subject: Re: [Freeswitch-users] mod_nibblebill ODBC issue


  I only updated the nibblebill config file with the following: 

<!-- Information for connecting to your database --><param name="db_username" value="rafqat"/><param name="db_password" value="?????"/><param name="db_dsn" value="FreeSWITCH"/>I did not change any of the dsn related stuff in any of the other FreeSWITCH config files.I dont see why I should be changing other freeswitch config files, can't one use a different database for the billing side of things?  It seems odd to allow dsn config to be added to the nibblebill conf file but not actually get used?or am i missing the point?CheersRaf


------------------------------------------------------------------------------
  From: infos at madovsky.org
  To: freeswitch-users at lists.freeswitch.org
  Date: Sun, 12 Dec 2010 15:27:39 -0500
  Subject: Re: [Freeswitch-users] mod_nibblebill ODBC issue


  what did you write on every
  db_dsn in Freeswitch config ?
    ----- Original Message ----- 
    From: Rafqat . 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Sunday, December 12, 2010 3:19 PM
    Subject: Re: [Freeswitch-users] mod_nibblebill ODBC issue



    More than what I have already specified?

    My /etc/odbc.ini file:

    [FreeSWITCH]
    Driver   = MySQL
    SERVER   = localhost
    PORT     = 3306
    DATABASE = FreeSWITCH
    OPTION  = 67108864
    Socket   = /var/lib/mysql/mysql.sock

    My /etc/odbcinst.ini file:

    # Example driver definitinions
    #
    #
    # Included in the unixODBC package
    #[PostgreSQL]
    #Description    = ODBC for PostgreSQL
    #Driver         = /usr/lib/libodbcpsql.so
    #Setup          = /usr/lib/libodbcpsqlS.so
    #FileUsage      = 1

    # Driver from the MyODBC package
    # Setup from the unixODBC package
    [MySQL]
    Description     = ODBC for MySQL
    Driver            = /usr/lib64/libmyodbc3.so
    Setup            = /usr/lib64/libodbcmyS.so
    FileUsage      = 1


----------------------------------------------------------------------------
    From: infos at madovsky.org
    To: freeswitch-users at lists.freeswitch.org
    Date: Sun, 12 Dec 2010 15:11:32 -0500
    Subject: Re: [Freeswitch-users] mod_nibblebill ODBC issue


    you have to specify a unix odbc drivers,
    usually in /etc/odbc.ini /etc/odbcinst.ini
      ----- Original Message ----- 
      From: Rafqat . 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Sunday, December 12, 2010 3:02 PM
      Subject: [Freeswitch-users] mod_nibblebill ODBC issue



      Hi,

      I have a problem trying to get mod_nibblebill to work.  I have thought I had done everything to get it to work but obviously not:

      2010-12-12 18:01:24.112241 [ERR] switch_odbc.c:313 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified

      2010-12-12 18:01:24.112413 [CRIT] mod_nibblebill.c:223 Cannot connect to ODBC driver/database bandwidth.com (user: bandwidth.com / pass password)!

      Do I need to update more than just the nibblebill conf file? 

      Any help is much appreciated

      CHeers

      Raf


      My /etc/odbc.ini file:

      [FreeSWITCH]
      Driver   = MySQL
      SERVER   = localhost
      PORT     = 3306
      DATABASE = FreeSWITCH
      OPTION  = 67108864
      Socket   = /var/lib/mysql/mysql.sock

      My /etc/odbcinst.ini file:

      # Example driver definitinions
      #
      #
      # Included in the unixODBC package
      #[PostgreSQL]
      #Description    = ODBC for PostgreSQL
      #Driver         = /usr/lib/libodbcpsql.so
      #Setup          = /usr/lib/libodbcpsqlS.so
      #FileUsage      = 1

      # Driver from the MyODBC package
      # Setup from the unixODBC package
      [MySQL]
      Description     = ODBC for MySQL
      Driver            = /usr/lib64/libmyodbc3.so
      Setup            = /usr/lib64/libodbcmyS.so
      FileUsage      = 1

      Snippet of my nibblebill.conf.xml file:

      <!-- Information for connecting to your database -->
      <param name="db_username" value="rafqat"/>
      <param name="db_password" value="?????"/>
      <param name="db_dsn" value="FreeSWITCH"/>


      I have even setup the following links:

mkdir /usr/local/freeswitch/etcln -s /etc/odbcinst.ini /usr/local/freeswitch/etc/odbcinst.iniln -s /etc/odbc.ini /usr/local/freeswitch/etc/odbc.iniI can connect successfully using isql:isql FreeSWITCH rafqat ??????







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

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From steveayre at gmail.com  Sun Dec 12 23:51:01 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 12 Dec 2010 20:51:01 +0000
Subject: [Freeswitch-users] Scale UP Freeswitch
In-Reply-To: <AANLkTinBf2Zc1o3Vagnha2C1YBba6YemjJjsf-UfCNht@mail.gmail.com>
References: <AANLkTi=4n_wMSDE7KA9JTDhP+ToGggt-nJzf=EYwTd=6@mail.gmail.com>
	<AANLkTi=N+6=AcELMtbyje3O7W0k7qm3nz8z0CoO3LmLo@mail.gmail.com>
	<AANLkTikkCTMpa-mwD79Fv3W8+1CwS4cMs8Hovdi0sfVS@mail.gmail.com>
	<AANLkTikG=rPb-Lz7rMySa3gzJpjo42ftuthsPReD6ShJ@mail.gmail.com>
	<AANLkTinBf2Zc1o3Vagnha2C1YBba6YemjJjsf-UfCNht@mail.gmail.com>
Message-ID: <AANLkTikHQtEM1f4sU6Ar0Gd5yeRsor0Gy_KOD7btQYW4@mail.gmail.com>

It is, but it relies on the caller supporting 3xx. They might not
handle the redirect.

A lot won't because you could redirect them to anywhere, so lots of
implementations will ignore the 3xx. FreeSWITCH for instance can
either ignore a 3xx or will send the call back into the dialplan.

I think you'll have more success having a FS server in front of the
others and bridging the call through to each server. If you set
inbound_bypass_media=true on the SIP profile, the RTP media will
bypass that server and go directly between the caller and the other FS
box. That means that the call won't be using any CPU since it'll only
wake up when a SIP packet is being sent/received. You'll still be
creating a session through so it'll still be allocating memory to the
call, a SIP proxy would use fewer resources.

-Steve


On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> Thanks Steve for suggestion, i'll check?X-Auth-IP, its new for me.
> Since we are talking about HA options... Is it practically doable use it:
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2
> The idea is to run one FS box (Redirect-FS) in front of several FS boxes
> which redirect the call to active/available FS. If we make some script on
> redirect FS to count the active calls on media FSes and rearrange the order
> of redirect then loadbalacing can also be done.
> ...possible?
>
> On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> > 1. i am thinking to use kamailo in front of FS boxes, is there any
>> > difference between kamailo and opensips?
>>
>> They're both forks of OpenSER so for the most part there's little
>> difference.
>>
>> There are some small differences though since the fork. For example,
>> opensips has a load_balancer module which kamalio does not (kamalio
>> can still do load balancing but has a different interface to do so).
>>
>> > 2. if kamailo or opensips is running in front of FS, then will it send
>> > call
>> > to FS with original customer ip? so i can do billing etc on FS box
>> > -> actually i do IP based authentication and also ip based billing on FS
>> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
>> > original
>> > customer overview.
>>
>> It will appear coming from the proxy IP. But there is a workaround.
>> Configure a proxy ACL on the SIP profile and add your proxy IP to it.
>> Then adjust your proxy routing rules so that it adds a X-Auth-IP
>> header that contains the original IP.
>> Anything coming from anything in the proxy ACL is trusted and FS will
>> use the value from X-Auth-IP (if it exists).
>>
>> -Steve
>>
>>
>>
>>
>> On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
>> > Hi,
>> >
>> > 1. i am thinking to use kamailo in front of FS boxes, is there any
>> > difference between kamailo and opensips?
>> >
>> > 2. if kamailo or opensips is running in front of FS, then will it send
>> > call
>> > to FS with original customer ip? so i can do billing etc on FS box
>> > -> actually i do IP based authentication and also ip based billing on FS
>> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
>> > original
>> > customer overview.
>> >
>> > thanks
>> > On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> >>
>> >> There are a few performance tweaking tips at
>> >> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
>> >>
>> >> Yes a Sangoma card will reduce your CPU load since transcoding won't
>> >> be done on the CPU any longer, that will then mean there's more CPU
>> >> available so you'll be able to handle more calls.
>> >>
>> >> However, if you're looking to increase your number of calls then you
>> >> probably want a cluster of servers as Juan pointed out.
>> >>
>> >> It'll mean you can increase the capacity by adding extra servers, so
>> >> there'd no longer be a limit to the number of calls you could handle
>> >> (just add another server).
>> >>
>> >> It'll also make maintenance easier, as you'll be able to pull a server
>> >> from service for updates etc while traffic continues to run on the
>> >> other servers. Maintenance won't mean a service outage.
>> >>
>> >> If you're handling that many calls then additional servers would make
>> >> your service more reliable. If a server crashes you'll still have the
>> >> calls running on the other servers while you're fixing the problem so
>> >> you won't have a complete outage. If FS is behind a load balancer then
>> >> your customers might not even notice anything apart from a few dropped
>> >> calls.
>> >>
>> >> There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will
>> >> attempt to continue calls if FS crashes and restarts, but I think
>> >> that's only for SIP-SIP not SIP-ISDN.
>> >>
>> >> -Steve
>> >>
>> >>
>> >>
>> >>
>> >> On 7 December 2010 12:26, Stephen Wilde <wstephen80 at gmail.com> wrote:
>> >> > Hi,
>> >> > I have one server running Freeswitch with some ISDN connections (via
>> >> > FreeTDM+Sangoma boards) and some SIP connections with service
>> >> > providers
>> >> > and
>> >> > customer.
>> >> > The usage of Freeswitch is as switching so it "bridge" each incoming
>> >> > call to
>> >> > a new outgoing call.
>> >> > SIP calls use G.729 and ISDN calls use ALaw for voice encoding.
>> >> > Now the number of call is grow up and also the CPU load is a little
>> >> > high
>> >> > so
>> >> > I have the necessity to scale UP my Freeswitch to handle more calls:
>> >> > what is
>> >> > the best way to do that?
>> >> > My first idea is to use a Sangoma D500 board to reduce the CPU load.
>> >> > Can
>> >> > be
>> >> > this a solution?
>> >> > There are different way to scale UP?
>> >> > Thanks in advance,
>> >> > Stephen
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Sun Dec 12 23:58:10 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 12 Dec 2010 20:58:10 +0000
Subject: [Freeswitch-users] mod_nibblebill ODBC issue
In-Reply-To: <COL122-W47AEDB0EEEFBE7D2CB91EFB5110@phx.gbl>
References: <COL122-W5418BECBF3C852E0219589B5110@phx.gbl>
	<COL122-W612DEC932A8C1F6F07FA5EB5110@phx.gbl>
	<COL122-W3739FC7490DFB018B97B5EB5110@phx.gbl>
	<COL122-W47AEDB0EEEFBE7D2CB91EFB5110@phx.gbl>
Message-ID: <AANLkTimqA5MuEHKiabq_UVHA6gyezS+QcghV-LwyQPqd@mail.gmail.com>

2010-12-12 18:01:24.112413 [CRIT] mod_nibblebill.c:223 Cannot connect
to ODBC driver/database bandwidth.com (user: bandwidth.com / pass
password)!

Those are the default settings. I think you've probably loaded the
module with the default options before you saved the new nibblebill
config file.

Try running these commands at the console/cli:
reloadxml
reload mod_nibblebill

Also, check you're editing the correct nibblebill.conf.xml file, which
will be at
/usr/local/freeswitch/conf/autoload_configs/nibblebill.conf.xml

You won't need to check the DSN settings for any other files - nibble
bill will only use its own DSN setting.

-Steve


On 12 December 2010 20:02, Rafqat . <rafonline at hotmail.com> wrote:
>
> Hi,
>
> I have a problem trying to get mod_nibblebill to work.? I have thought I had
> done everything to get it to work but obviously not:
>
> 2010-12-12 18:01:24.112241 [ERR] switch_odbc.c:313 STATE: IM002 CODE 0
> ERROR: [unixODBC][Driver Manager]Data source name not found, and no default
> driver specified
>
> 2010-12-12 18:01:24.112413 [CRIT] mod_nibblebill.c:223 Cannot connect to
> ODBC driver/database bandwidth.com (user: bandwidth.com / pass password)!
>
> Do I need to update more than just the nibblebill conf file?
>
> Any help is much appreciated
>
> CHeers
>
> Raf
>
>
> My /etc/odbc.ini file:
>
> [FreeSWITCH]
> Driver?? = MySQL
> SERVER?? = localhost
> PORT???? = 3306
> DATABASE = FreeSWITCH
> OPTION? = 67108864
> Socket?? = /var/lib/mysql/mysql.sock
>
> My /etc/odbcinst.ini file:
>
> # Example driver definitinions
> #
> #
> # Included in the unixODBC package
> #[PostgreSQL]
> #Description??? = ODBC for PostgreSQL
> #Driver???????? = /usr/lib/libodbcpsql.so
> #Setup????????? = /usr/lib/libodbcpsqlS.so
> #FileUsage????? = 1
>
> # Driver from the MyODBC package
> # Setup from the unixODBC package
> [MySQL]
> Description???? = ODBC for MySQL
> Driver??????????? = /usr/lib64/libmyodbc3.so
> Setup??????????? = /usr/lib64/libodbcmyS.so
> FileUsage????? = 1
>
> Snippet of my nibblebill.conf.xml file:
>
> <!-- Information for connecting to your database -->
> <param name="db_username" value="rafqat"/>
> <param name="db_password" value="?????"/>
> <param name="db_dsn" value="FreeSWITCH"/>
>
>
> I have even setup the following links:
>
> mkdir /usr/local/freeswitch/etc
> ln -s /etc/odbcinst.ini /usr/local/freeswitch/etc/odbcinst.ini
> ln -s /etc/odbc.ini /usr/local/freeswitch/etc/odbc.ini
>
> I can connect successfully using isql:
>
> isql FreeSWITCH rafqat ??????
>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From rafonline at hotmail.com  Mon Dec 13 00:04:00 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Sun, 12 Dec 2010 21:04:00 +0000
Subject: [Freeswitch-users] mod_nibblebill ODBC issue
In-Reply-To: <DC864B84C89C47C9B1404DEA84D6DC1F@e1705>
References: <COL122-W5418BECBF3C852E0219589B5110@phx.gbl>, ,
	,,<COL122-W612DEC932A8C1F6F07FA5EB5110@phx.gbl>, ,
	,,<COL122-W3739FC7490DFB018B97B5EB5110@phx.gbl>,
	,,<COL122-W47AEDB0EEEFBE7D2CB91EFB5110@phx.gbl>,
	, , <1F070BD848664E029DA01D63EC529A8E@e1705>, ,
	<COL122-W1312F9EEB16A140EA52334B5110@phx.gbl>, ,
	<0281F1808ACB4AA1A39EFCB3035A1055@e1705>,
	<COL122-W5179D1FE2DB91024F97067B5110@phx.gbl>,
	<DC864B84C89C47C9B1404DEA84D6DC1F@e1705>
Message-ID: <COL122-W1500980CD74C691565F8E5B5110@phx.gbl>



Which variables do I need to set?

I am suprised to see the following line in my error output:

2010-12-12 18:01:24.112413 [CRIT] mod_nibblebill.c:223 
      Cannot connect to ODBC driver/database bandwidth.com (user: bandwidth.com 
      / pass password)!

I looked at the mod_nibblebill.c source code to see when the values for the variables used in this error message get set.  They should be getting set from the values in the nibblebill config file, so I have no idea how 'bandwidth.com' gets to be in the variable globals.db_dsn.

All seems very odd.

Please help.

Cheers

Raf

From: infos at madovsky.org
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 12 Dec 2010 15:50:20 -0500
Subject: Re: [Freeswitch-users] mod_nibblebill ODBC issue










Mabe your system env variables are not set.
it looks like odbc can't find the path...

  ----- Original Message ----- 
  From: 
  Rafqat 
  . 
  To: freeswitch-users at lists.freeswitch.org 
  
  Sent: Sunday, December 12, 2010 3:36 
  PM
  Subject: Re: [Freeswitch-users] 
  mod_nibblebill ODBC issue
  
I only updated the nibblebill config file with the following: 
  
<!-- Information for connecting to your database -->
<param name="db_username" value="rafqat"/>
<param name="db_password" value="?????"/>
<param name="db_dsn" value="FreeSWITCH"/>

I did not change any of the dsn related stuff in any of the other FreeSWITCH config files.

I dont see why I should be changing other freeswitch config files, can't one use a different database for the billing side of things?  It seems odd to allow dsn config to be added to the nibblebill conf file but not actually get used?

or am i missing the point?

Cheers

Raf



  
  From: infos at madovsky.org
To: freeswitch-users at lists.freeswitch.org
Date: 
  Sun, 12 Dec 2010 15:27:39 -0500
Subject: Re: [Freeswitch-users] 
  mod_nibblebill ODBC issue


  
  

  what did you write on every
  db_dsn in Freeswitch config ?
  
    ----- Original Message ----- 
    From: Rafqat . 
    
    To: freeswitch-users at lists.freeswitch.org 
    
    Sent: Sunday, December 12, 2010 3:19 
    PM
    Subject: Re: [Freeswitch-users] 
    mod_nibblebill ODBC issue
    

More than what I have already specified?

My 
    /etc/odbc.ini file:

[FreeSWITCH]
Driver   = 
    MySQL
SERVER   = localhost
PORT     = 
    3306
DATABASE = FreeSWITCH
OPTION  = 
    67108864
Socket   = /var/lib/mysql/mysql.sock

My 
    /etc/odbcinst.ini file:

# Example driver definitinions
#
#
# 
    Included in the unixODBC 
    package
#[PostgreSQL]
#Description    = ODBC for 
    PostgreSQL
#Driver         = 
    /usr/lib/libodbcpsql.so
#Setup          
    = /usr/lib/libodbcpsqlS.so
#FileUsage      = 
    1

# Driver from the MyODBC package
# Setup from the unixODBC 
    package
[MySQL]
Description     = ODBC for 
    MySQL
Driver            
    = 
    /usr/lib64/libmyodbc3.so
Setup            
    = /usr/lib64/libodbcmyS.so
FileUsage      = 
    1


    
    From: infos at madovsky.org
To: 
    freeswitch-users at lists.freeswitch.org
Date: Sun, 12 Dec 2010 15:11:32 
    -0500
Subject: Re: [Freeswitch-users] mod_nibblebill ODBC issue


    

    you have to specify a unix odbc drivers,
    usually in /etc/odbc.ini 
/etc/odbcinst.ini
    
      ----- Original Message ----- 
      From: 
      Rafqat 
      . 
      To: freeswitch-users at lists.freeswitch.org 
      
      Sent: Sunday, December 12, 2010 3:02 
      PM
      Subject: [Freeswitch-users] 
      mod_nibblebill ODBC issue
      

Hi,

I have a problem trying to get 
      mod_nibblebill to work.  I have thought I had done everything to get 
      it to work but obviously not:

2010-12-12 18:01:24.112241 [ERR] 
      switch_odbc.c:313 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver 
      Manager]Data source name not found, and no default driver 
      specified

2010-12-12 18:01:24.112413 [CRIT] mod_nibblebill.c:223 
      Cannot connect to ODBC driver/database bandwidth.com (user: bandwidth.com 
      / pass password)!

Do I need to update more than just the nibblebill 
      conf file? 

Any help is much 
      appreciated

CHeers

Raf


My /etc/odbc.ini 
      file:

[FreeSWITCH]
Driver   = 
      MySQL
SERVER   = localhost
PORT     = 
      3306
DATABASE = FreeSWITCH
OPTION  = 
      67108864
Socket   = /var/lib/mysql/mysql.sock

My 
      /etc/odbcinst.ini file:

# Example driver 
      definitinions
#
#
# Included in the unixODBC 
      package
#[PostgreSQL]
#Description    = ODBC for 
      PostgreSQL
#Driver         = 
      /usr/lib/libodbcpsql.so
#Setup          
      = /usr/lib/libodbcpsqlS.so
#FileUsage      = 
      1

# Driver from the MyODBC package
# Setup from the unixODBC 
      package
[MySQL]
Description     = ODBC for 
      MySQL
Driver            
      = 
      /usr/lib64/libmyodbc3.so
Setup            
      = /usr/lib64/libodbcmyS.so
FileUsage      = 
      1

Snippet of my nibblebill.conf.xml file:

<!-- 
      Information for connecting to your database -->
<param 
      name="db_username" value="rafqat"/>
<param name="db_password" 
      value="?????"/>
<param name="db_dsn" 
      value="FreeSWITCH"/>


I have even setup the following 
      links:
mkdir /usr/local/freeswitch/etc
ln -s /etc/odbcinst.ini /usr/local/freeswitch/etc/odbcinst.ini
ln -s /etc/odbc.ini /usr/local/freeswitch/etc/odbc.ini

I can connect successfully using isql:

isql FreeSWITCH rafqat ??????









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

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

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

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

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


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
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From wstephen80 at gmail.com  Mon Dec 13 00:09:51 2010
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Sun, 12 Dec 2010 22:09:51 +0100
Subject: [Freeswitch-users] Scale UP Freeswitch
In-Reply-To: <AANLkTikHQtEM1f4sU6Ar0Gd5yeRsor0Gy_KOD7btQYW4@mail.gmail.com>
References: <AANLkTi=4n_wMSDE7KA9JTDhP+ToGggt-nJzf=EYwTd=6@mail.gmail.com>
	<AANLkTi=N+6=AcELMtbyje3O7W0k7qm3nz8z0CoO3LmLo@mail.gmail.com>
	<AANLkTikkCTMpa-mwD79Fv3W8+1CwS4cMs8Hovdi0sfVS@mail.gmail.com>
	<AANLkTikG=rPb-Lz7rMySa3gzJpjo42ftuthsPReD6ShJ@mail.gmail.com>
	<AANLkTinBf2Zc1o3Vagnha2C1YBba6YemjJjsf-UfCNht@mail.gmail.com>
	<AANLkTikHQtEM1f4sU6Ar0Gd5yeRsor0Gy_KOD7btQYW4@mail.gmail.com>
Message-ID: <AANLkTikMXxDkyB=46-u-__jUaQU5FUhawLz0Anf5_pNs@mail.gmail.com>

The FS in front that doesn't handle the media is a good solution for me!
The incoming SIP call can be bridged with a simple round robin rule to "FS
media server" and a called prefix number can be added by front FS (and
removed by FS media server) to propagate the "source IP address".
Thank you for this info: I'll do some test on this solution.

Stephen

On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <steveayre at gmail.com> wrote:

> It is, but it relies on the caller supporting 3xx. They might not
> handle the redirect.
>
> A lot won't because you could redirect them to anywhere, so lots of
> implementations will ignore the 3xx. FreeSWITCH for instance can
> either ignore a 3xx or will send the call back into the dialplan.
>
> I think you'll have more success having a FS server in front of the
> others and bridging the call through to each server. If you set
> inbound_bypass_media=true on the SIP profile, the RTP media will
> bypass that server and go directly between the caller and the other FS
> box. That means that the call won't be using any CPU since it'll only
> wake up when a SIP packet is being sent/received. You'll still be
> creating a session through so it'll still be allocating memory to the
> call, a SIP proxy would use fewer resources.
>
> -Steve
>
>
> On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> > Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me.
> > Since we are talking about HA options... Is it practically doable use it:
> >
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2
> > The idea is to run one FS box (Redirect-FS) in front of several FS boxes
> > which redirect the call to active/available FS. If we make some script on
> > redirect FS to count the active calls on media FSes and rearrange the
> order
> > of redirect then loadbalacing can also be done.
> > ...possible?
> >
> > On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>
> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any
> >> > difference between kamailo and opensips?
> >>
> >> They're both forks of OpenSER so for the most part there's little
> >> difference.
> >>
> >> There are some small differences though since the fork. For example,
> >> opensips has a load_balancer module which kamalio does not (kamalio
> >> can still do load balancing but has a different interface to do so).
> >>
> >> > 2. if kamailo or opensips is running in front of FS, then will it send
> >> > call
> >> > to FS with original customer ip? so i can do billing etc on FS box
> >> > -> actually i do IP based authentication and also ip based billing on
> FS
> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
> >> > original
> >> > customer overview.
> >>
> >> It will appear coming from the proxy IP. But there is a workaround.
> >> Configure a proxy ACL on the SIP profile and add your proxy IP to it.
> >> Then adjust your proxy routing rules so that it adds a X-Auth-IP
> >> header that contains the original IP.
> >> Anything coming from anything in the proxy ACL is trusted and FS will
> >> use the value from X-Auth-IP (if it exists).
> >>
> >> -Steve
> >>
> >>
> >>
> >>
> >> On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981 at gmail.com>
> wrote:
> >> > Hi,
> >> >
> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any
> >> > difference between kamailo and opensips?
> >> >
> >> > 2. if kamailo or opensips is running in front of FS, then will it send
> >> > call
> >> > to FS with original customer ip? so i can do billing etc on FS box
> >> > -> actually i do IP based authentication and also ip based billing on
> FS
> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
> >> > original
> >> > customer overview.
> >> >
> >> > thanks
> >> > On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >> >>
> >> >> There are a few performance tweaking tips at
> >> >>
> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
> >> >>
> >> >> Yes a Sangoma card will reduce your CPU load since transcoding won't
> >> >> be done on the CPU any longer, that will then mean there's more CPU
> >> >> available so you'll be able to handle more calls.
> >> >>
> >> >> However, if you're looking to increase your number of calls then you
> >> >> probably want a cluster of servers as Juan pointed out.
> >> >>
> >> >> It'll mean you can increase the capacity by adding extra servers, so
> >> >> there'd no longer be a limit to the number of calls you could handle
> >> >> (just add another server).
> >> >>
> >> >> It'll also make maintenance easier, as you'll be able to pull a
> server
> >> >> from service for updates etc while traffic continues to run on the
> >> >> other servers. Maintenance won't mean a service outage.
> >> >>
> >> >> If you're handling that many calls then additional servers would make
> >> >> your service more reliable. If a server crashes you'll still have the
> >> >> calls running on the other servers while you're fixing the problem so
> >> >> you won't have a complete outage. If FS is behind a load balancer
> then
> >> >> your customers might not even notice anything apart from a few
> dropped
> >> >> calls.
> >> >>
> >> >> There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will
> >> >> attempt to continue calls if FS crashes and restarts, but I think
> >> >> that's only for SIP-SIP not SIP-ISDN.
> >> >>
> >> >> -Steve
> >> >>
> >> >>
> >> >>
> >> >>
> >> >> On 7 December 2010 12:26, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> >> >> > Hi,
> >> >> > I have one server running Freeswitch with some ISDN connections
> (via
> >> >> > FreeTDM+Sangoma boards) and some SIP connections with service
> >> >> > providers
> >> >> > and
> >> >> > customer.
> >> >> > The usage of Freeswitch is as switching so it "bridge" each
> incoming
> >> >> > call to
> >> >> > a new outgoing call.
> >> >> > SIP calls use G.729 and ISDN calls use ALaw for voice encoding.
> >> >> > Now the number of call is grow up and also the CPU load is a little
> >> >> > high
> >> >> > so
> >> >> > I have the necessity to scale UP my Freeswitch to handle more
> calls:
> >> >> > what is
> >> >> > the best way to do that?
> >> >> > My first idea is to use a Sangoma D500 board to reduce the CPU
> load.
> >> >> > Can
> >> >> > be
> >> >> > this a solution?
> >> >> > There are different way to scale UP?
> >> >> > Thanks in advance,
> >> >> > Stephen
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rafonline at hotmail.com  Mon Dec 13 00:10:10 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Sun, 12 Dec 2010 21:10:10 +0000
Subject: [Freeswitch-users] mod_nibblebill ODBC issue
In-Reply-To: <COL122-W1500980CD74C691565F8E5B5110@phx.gbl>
References: <COL122-W5418BECBF3C852E0219589B5110@phx.gbl>,
	,,,,<COL122-W612DEC932A8C1F6F07FA5EB5110@phx.gbl>,
	, , , , <COL122-W3739FC7490DFB018B97B5EB5110@phx.gbl>, , , ,
	<COL122-W47AEDB0EEEFBE7D2CB91EFB5110@phx.gbl>, , , 
	, <1F070BD848664E029DA01D63EC529A8E@e1705>,
	,,<COL122-W1312F9EEB16A140EA52334B5110@phx.gbl>,
	, , <0281F1808ACB4AA1A39EFCB3035A1055@e1705>, ,
	<COL122-W5179D1FE2DB91024F97067B5110@phx.gbl>, ,
	<DC864B84C89C47C9B1404DEA84D6DC1F@e1705>,
	<COL122-W1500980CD74C691565F8E5B5110@phx.gbl>
Message-ID: <COL122-W14A1AD99CFFDA63F624D16B5110@phx.gbl>



I am a total idiot! I was changing the wrong nibblebill conf file!

Thanks very much guys.

Sorry for wasting your time.

Cheers

Raf

From: rafonline at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 12 Dec 2010 21:04:00 +0000
Subject: Re: [Freeswitch-users] mod_nibblebill ODBC issue









Which variables do I need to set?

I am suprised to see the following line in my error output:

2010-12-12 18:01:24.112413 [CRIT] mod_nibblebill.c:223 
      Cannot connect to ODBC driver/database bandwidth.com (user: bandwidth.com 
      / pass password)!

I looked at the mod_nibblebill.c source code to see when the values for the variables used in this error message get set.  They should be getting set from the values in the nibblebill config file, so I have no idea how 'bandwidth.com' gets to be in the variable globals.db_dsn.

All seems very odd.

Please help.

Cheers

Raf

From: infos at madovsky.org
To: freeswitch-users at lists.freeswitch.org
Date: Sun, 12 Dec 2010 15:50:20 -0500
Subject: Re: [Freeswitch-users] mod_nibblebill ODBC issue










Mabe your system env variables are not set.
it looks like odbc can't find the path...

  ----- Original Message ----- 
  From: 
  Rafqat 
  . 
  To: freeswitch-users at lists.freeswitch.org 
  
  Sent: Sunday, December 12, 2010 3:36 
  PM
  Subject: Re: [Freeswitch-users] 
  mod_nibblebill ODBC issue
  
I only updated the nibblebill config file with the following: 
  
<!-- Information for connecting to your database -->
<param name="db_username" value="rafqat"/>
<param name="db_password" value="?????"/>
<param name="db_dsn" value="FreeSWITCH"/>

I did not change any of the dsn related stuff in any of the other FreeSWITCH config files.

I dont see why I should be changing other freeswitch config files, can't one use a different database for the billing side of things?  It seems odd to allow dsn config to be added to the nibblebill conf file but not actually get used?

or am i missing the point?

Cheers

Raf



  
  From: infos at madovsky.org
To: freeswitch-users at lists.freeswitch.org
Date: 
  Sun, 12 Dec 2010 15:27:39 -0500
Subject: Re: [Freeswitch-users] 
  mod_nibblebill ODBC issue


  
  

  what did you write on every
  db_dsn in Freeswitch config ?
  
    ----- Original Message ----- 
    From: Rafqat . 
    
    To: freeswitch-users at lists.freeswitch.org 
    
    Sent: Sunday, December 12, 2010 3:19 
    PM
    Subject: Re: [Freeswitch-users] 
    mod_nibblebill ODBC issue
    

More than what I have already specified?

My 
    /etc/odbc.ini file:

[FreeSWITCH]
Driver   = 
    MySQL
SERVER   = localhost
PORT     = 
    3306
DATABASE = FreeSWITCH
OPTION  = 
    67108864
Socket   = /var/lib/mysql/mysql.sock

My 
    /etc/odbcinst.ini file:

# Example driver definitinions
#
#
# 
    Included in the unixODBC 
    package
#[PostgreSQL]
#Description    = ODBC for 
    PostgreSQL
#Driver         = 
    /usr/lib/libodbcpsql.so
#Setup          
    = /usr/lib/libodbcpsqlS.so
#FileUsage      = 
    1

# Driver from the MyODBC package
# Setup from the unixODBC 
    package
[MySQL]
Description     = ODBC for 
    MySQL
Driver            
    = 
    /usr/lib64/libmyodbc3.so
Setup            
    = /usr/lib64/libodbcmyS.so
FileUsage      = 
    1


    
    From: infos at madovsky.org
To: 
    freeswitch-users at lists.freeswitch.org
Date: Sun, 12 Dec 2010 15:11:32 
    -0500
Subject: Re: [Freeswitch-users] mod_nibblebill ODBC issue


    

    you have to specify a unix odbc drivers,
    usually in /etc/odbc.ini 
/etc/odbcinst.ini
    
      ----- Original Message ----- 
      From: 
      Rafqat 
      . 
      To: freeswitch-users at lists.freeswitch.org 
      
      Sent: Sunday, December 12, 2010 3:02 
      PM
      Subject: [Freeswitch-users] 
      mod_nibblebill ODBC issue
      

Hi,

I have a problem trying to get 
      mod_nibblebill to work.  I have thought I had done everything to get 
      it to work but obviously not:

2010-12-12 18:01:24.112241 [ERR] 
      switch_odbc.c:313 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver 
      Manager]Data source name not found, and no default driver 
      specified

2010-12-12 18:01:24.112413 [CRIT] mod_nibblebill.c:223 
      Cannot connect to ODBC driver/database bandwidth.com (user: bandwidth.com 
      / pass password)!

Do I need to update more than just the nibblebill 
      conf file? 

Any help is much 
      appreciated

CHeers

Raf


My /etc/odbc.ini 
      file:

[FreeSWITCH]
Driver   = 
      MySQL
SERVER   = localhost
PORT     = 
      3306
DATABASE = FreeSWITCH
OPTION  = 
      67108864
Socket   = /var/lib/mysql/mysql.sock

My 
      /etc/odbcinst.ini file:

# Example driver 
      definitinions
#
#
# Included in the unixODBC 
      package
#[PostgreSQL]
#Description    = ODBC for 
      PostgreSQL
#Driver         = 
      /usr/lib/libodbcpsql.so
#Setup          
      = /usr/lib/libodbcpsqlS.so
#FileUsage      = 
      1

# Driver from the MyODBC package
# Setup from the unixODBC 
      package
[MySQL]
Description     = ODBC for 
      MySQL
Driver            
      = 
      /usr/lib64/libmyodbc3.so
Setup            
      = /usr/lib64/libodbcmyS.so
FileUsage      = 
      1

Snippet of my nibblebill.conf.xml file:

<!-- 
      Information for connecting to your database -->
<param 
      name="db_username" value="rafqat"/>
<param name="db_password" 
      value="?????"/>
<param name="db_dsn" 
      value="FreeSWITCH"/>


I have even setup the following 
      links:
mkdir /usr/local/freeswitch/etc
ln -s /etc/odbcinst.ini /usr/local/freeswitch/etc/odbcinst.ini
ln -s /etc/odbc.ini /usr/local/freeswitch/etc/odbc.ini

I can connect successfully using isql:

isql FreeSWITCH rafqat ??????









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

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

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

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


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


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

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
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From saeedahmad1981 at gmail.com  Mon Dec 13 00:32:40 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Sun, 12 Dec 2010 22:32:40 +0100
Subject: [Freeswitch-users] Scale UP Freeswitch
In-Reply-To: <AANLkTikHQtEM1f4sU6Ar0Gd5yeRsor0Gy_KOD7btQYW4@mail.gmail.com>
References: <AANLkTi=4n_wMSDE7KA9JTDhP+ToGggt-nJzf=EYwTd=6@mail.gmail.com>
	<AANLkTi=N+6=AcELMtbyje3O7W0k7qm3nz8z0CoO3LmLo@mail.gmail.com>
	<AANLkTikkCTMpa-mwD79Fv3W8+1CwS4cMs8Hovdi0sfVS@mail.gmail.com>
	<AANLkTikG=rPb-Lz7rMySa3gzJpjo42ftuthsPReD6ShJ@mail.gmail.com>
	<AANLkTinBf2Zc1o3Vagnha2C1YBba6YemjJjsf-UfCNht@mail.gmail.com>
	<AANLkTikHQtEM1f4sU6Ar0Gd5yeRsor0Gy_KOD7btQYW4@mail.gmail.com>
Message-ID: <AANLkTimS5ko3Yt155+z-vhpLUA9DG43y2vyUPMOFECj2@mail.gmail.com>

hmmm... so doing that will also require X-Auth-IP, right or something more
tricky can be done?

On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <steveayre at gmail.com> wrote:

> It is, but it relies on the caller supporting 3xx. They might not
> handle the redirect.
>
> A lot won't because you could redirect them to anywhere, so lots of
> implementations will ignore the 3xx. FreeSWITCH for instance can
> either ignore a 3xx or will send the call back into the dialplan.
>
> I think you'll have more success having a FS server in front of the
> others and bridging the call through to each server. If you set
> inbound_bypass_media=true on the SIP profile, the RTP media will
> bypass that server and go directly between the caller and the other FS
> box. That means that the call won't be using any CPU since it'll only
> wake up when a SIP packet is being sent/received. You'll still be
> creating a session through so it'll still be allocating memory to the
> call, a SIP proxy would use fewer resources.
>
> -Steve
>
>
> On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> > Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me.
> > Since we are talking about HA options... Is it practically doable use it:
> >
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2
> > The idea is to run one FS box (Redirect-FS) in front of several FS boxes
> > which redirect the call to active/available FS. If we make some script on
> > redirect FS to count the active calls on media FSes and rearrange the
> order
> > of redirect then loadbalacing can also be done.
> > ...possible?
> >
> > On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>
> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any
> >> > difference between kamailo and opensips?
> >>
> >> They're both forks of OpenSER so for the most part there's little
> >> difference.
> >>
> >> There are some small differences though since the fork. For example,
> >> opensips has a load_balancer module which kamalio does not (kamalio
> >> can still do load balancing but has a different interface to do so).
> >>
> >> > 2. if kamailo or opensips is running in front of FS, then will it send
> >> > call
> >> > to FS with original customer ip? so i can do billing etc on FS box
> >> > -> actually i do IP based authentication and also ip based billing on
> FS
> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
> >> > original
> >> > customer overview.
> >>
> >> It will appear coming from the proxy IP. But there is a workaround.
> >> Configure a proxy ACL on the SIP profile and add your proxy IP to it.
> >> Then adjust your proxy routing rules so that it adds a X-Auth-IP
> >> header that contains the original IP.
> >> Anything coming from anything in the proxy ACL is trusted and FS will
> >> use the value from X-Auth-IP (if it exists).
> >>
> >> -Steve
> >>
> >>
> >>
> >>
> >> On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981 at gmail.com>
> wrote:
> >> > Hi,
> >> >
> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any
> >> > difference between kamailo and opensips?
> >> >
> >> > 2. if kamailo or opensips is running in front of FS, then will it send
> >> > call
> >> > to FS with original customer ip? so i can do billing etc on FS box
> >> > -> actually i do IP based authentication and also ip based billing on
> FS
> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
> >> > original
> >> > customer overview.
> >> >
> >> > thanks
> >> > On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >> >>
> >> >> There are a few performance tweaking tips at
> >> >>
> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
> >> >>
> >> >> Yes a Sangoma card will reduce your CPU load since transcoding won't
> >> >> be done on the CPU any longer, that will then mean there's more CPU
> >> >> available so you'll be able to handle more calls.
> >> >>
> >> >> However, if you're looking to increase your number of calls then you
> >> >> probably want a cluster of servers as Juan pointed out.
> >> >>
> >> >> It'll mean you can increase the capacity by adding extra servers, so
> >> >> there'd no longer be a limit to the number of calls you could handle
> >> >> (just add another server).
> >> >>
> >> >> It'll also make maintenance easier, as you'll be able to pull a
> server
> >> >> from service for updates etc while traffic continues to run on the
> >> >> other servers. Maintenance won't mean a service outage.
> >> >>
> >> >> If you're handling that many calls then additional servers would make
> >> >> your service more reliable. If a server crashes you'll still have the
> >> >> calls running on the other servers while you're fixing the problem so
> >> >> you won't have a complete outage. If FS is behind a load balancer
> then
> >> >> your customers might not even notice anything apart from a few
> dropped
> >> >> calls.
> >> >>
> >> >> There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will
> >> >> attempt to continue calls if FS crashes and restarts, but I think
> >> >> that's only for SIP-SIP not SIP-ISDN.
> >> >>
> >> >> -Steve
> >> >>
> >> >>
> >> >>
> >> >>
> >> >> On 7 December 2010 12:26, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> >> >> > Hi,
> >> >> > I have one server running Freeswitch with some ISDN connections
> (via
> >> >> > FreeTDM+Sangoma boards) and some SIP connections with service
> >> >> > providers
> >> >> > and
> >> >> > customer.
> >> >> > The usage of Freeswitch is as switching so it "bridge" each
> incoming
> >> >> > call to
> >> >> > a new outgoing call.
> >> >> > SIP calls use G.729 and ISDN calls use ALaw for voice encoding.
> >> >> > Now the number of call is grow up and also the CPU load is a little
> >> >> > high
> >> >> > so
> >> >> > I have the necessity to scale UP my Freeswitch to handle more
> calls:
> >> >> > what is
> >> >> > the best way to do that?
> >> >> > My first idea is to use a Sangoma D500 board to reduce the CPU
> load.
> >> >> > Can
> >> >> > be
> >> >> > this a solution?
> >> >> > There are different way to scale UP?
> >> >> > Thanks in advance,
> >> >> > Stephen
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From haloha201 at yahoo.com  Mon Dec 13 04:55:20 2010
From: haloha201 at yahoo.com (ha do)
Date: Sun, 12 Dec 2010 17:55:20 -0800 (PST)
Subject: [Freeswitch-users] what ports should open on firewall for
	skypopen
In-Reply-To: <AANLkTinH5h5e4uP+2-GkZfxCW4TofEp2vkMdLKkahLPM@mail.gmail.com>
Message-ID: <499169.27092.qm@web32408.mail.mud.yahoo.com>

got it

Thank you
Ha`

--- On Sun, 12/12/10, Steven Ayre <steveayre at gmail.com> wrote:

> From: Steven Ayre <steveayre at gmail.com>
> Subject: Re: [Freeswitch-users] what ports should open on firewall for skypopen
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Sunday, December 12, 2010, 11:52 AM
> The same ports as Skype itself would
> require.
> 
> They have a firewall guide on their website:
> http://www.skype.com/intl/en-us/support/user-guides/firewalls/
> 
> Almost all firewalls will allow Skype through by default,
> since it
> uses outgoing connections on port 443. Everything else can
> then run
> through that connection. You should only get problems on
> extremely
> restrictive firewalls that filter outgoing traffic (most
> don't) and
> which enforce the protocols using each port (most don't).
> 
> >From the website:
> ? ? *? Ideally, outgoing TCP connections to
> all ports (1.65535) should
> be opened. This option results in Skype working most
> reliably. This is
> only necessary for your Skype connection to be able to
> connect to the
> Skype network and will not make your network any less
> secure.
> ? ? * If the above is not possible, open up
> outgoing TCP connections
> to port 443. This will only work if you are using Skype
> version 0.97
> and above.
> ? ? * If the above does not solve the problem,
> open up outgoing TCP
> connections to port 80. Some firewalls restrict traffic to
> port 80 to
> HTTP protocol, and in this case Skype can not use it since
> Skype does
> not use HTTP. In some firewalls it is possible to open up
> all traffic
> to port 80, not just HTTP, and in this case Skype will
> work.
> ? ? * If the above is not possible, Skype
> versions 0.97 and above can
> use a HTTPS/SSL proxy. In order to do that, you have to
> configure the
> proxy address in Internet Explorer options. Skype will then
> be able to
> use it as well.
> ? ? * Please use our problem reporting form to
> report details of all
> instances when you have experienced a problem with Skype
> and a
> firewall.
> 
> -Steve
> 
> 
> 
> On 12 December 2010 15:25, ha do <haloha201 at yahoo.com>
> wrote:
> > Hi All
> >
> > my network topology:
> > ? ?Freeswitch(skypopen module with multiple
> interfaces)---firewall ----internet
> >
> > so which ports should i open on firewall to make the
> skypopen works properly
> > which ports for signaling ?of skype
> > which ports for media of skype
> >
> >
> > Thank you
> > Ha`
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From haloha201 at yahoo.com  Mon Dec 13 04:56:03 2010
From: haloha201 at yahoo.com (ha do)
Date: Sun, 12 Dec 2010 17:56:03 -0800 (PST)
Subject: [Freeswitch-users] what ports should open on firewall for
	skypopen
In-Reply-To: <4D04EC6F.1050303@gmail.com>
Message-ID: <517430.92929.qm@web32407.mail.mud.yahoo.com>

Thank you
Ha`

--- On Sun, 12/12/10, Meftah Tayeb <tayeb.meftah at gmail.com> wrote:

> From: Meftah Tayeb <tayeb.meftah at gmail.com>
> Subject: Re: [Freeswitch-users] what ports should open on firewall for skypopen
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Cc: "ha do" <haloha201 at yahoo.com>
> Date: Sunday, December 12, 2010, 8:38 AM
> no port required
> skype can bypass firewalls
> thanks
> Le 12/12/2010 16:25, ha do a ?crit :
> > Hi All
> >
> > my network topology:
> >? ? ? Freeswitch(skypopen module with
> multiple interfaces)---firewall ----internet
> >
> > so which ports should i open on firewall to make the
> skypopen works properly
> > which ports for signaling? of skype
> > which ports for media of skype
> >
> >
> > Thank you
> > Ha`
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >? ? 
> 
> 
> -- 
> Meftah Tayeb
> inum: +883510001288000
> Phone: +13602276297
> Fax: +12538020313
> 
> 


      


From mel0torme at gmail.com  Mon Dec 13 10:10:07 2010
From: mel0torme at gmail.com (Tom C)
Date: Sun, 12 Dec 2010 23:10:07 -0800
Subject: [Freeswitch-users] DockStar compile failure, with possible fix
Message-ID: <AANLkTi=-1xKh21fR5PADAQs2YPRKHjssCLQMRWNikSu6@mail.gmail.com>

I was attempting to build FreeSwitch on a Dockstar running Debian Squeeze,
and ran into the sofia.c logger() compile problem as described here:
http://jira.freeswitch.org/browse/FS-802

The offered solution was to downgrade GCC, but being a Linux newbie, I
didn't know how to do that. So I decided to figure out what the problem
really is.

After learning waaaay too much about va_list, I see that the logger()
function in sofia.c assumes that va_list will always be a pointer.  But on
different platforms, va_list can be an array or a struct.  On the Dockstar
(with Debian Squeeze and GCC 4.4), va_list is apparently a struct.

I made the following modifications to logger() to make it truly portable,
and it compiles and runs fine on my DockStar, my Debian Lenny x86.  (My
Windows build is having bigger problems.)  But I'm no C guru, so someone
needs to thoroughly examine my code before anyone else takes it as gospel.
Am I actually checking what needs to be checked with the va_arg() macro?
Changing it to if(1) worked too, after all.


ORIGINAL logger() in sofia.c, to make it easy to see what I changed.
static void logger(void *logarg, char const *fmt, va_list ap)
{
/* gcc 4.4 gets mad at us for testing if (ap) so let's try to work around
it....*/
void *ap_ptr = (void *) (intptr_t) ap;  //Error now occurs here, because ap
is a struct.
if (!fmt) return;
if (ap_ptr) {  //Error used to occur here, before attempted fix above.
   switch_log_vprintf(SWITCH_CHANNEL_LOG_CLEAN,
mod_sofia_globals.tracelevel, fmt, ap);
} else {
   switch_log_printf(SWITCH_CHANNEL_LOG_CLEAN, mod_sofia_globals.tracelevel,
"%s", fmt);
}
}


My MODIFIED logger() function, additions and changes labelled with DOCKSTAR:

static void logger(void *logarg, char const *fmt, va_list ap)
{
va_list temp_ap;  //DOCKSTAR: Added line (replacing previous ap_ptr cast).

if (!fmt) return;

va_copy(temp_ap, ap); //DOCKSTAR: Added Line. Make copy of "ap" so va_arg()
macro doesn't move pointer.

if (va_arg(temp_ap, int)) {  //DOCKSTAR: Modified, get first argument from
ap, check non-null.
     switch_log_vprintf(SWITCH_CHANNEL_LOG_CLEAN,
mod_sofia_globals.tracelevel, fmt, ap);
} else {
     switch_log_printf(SWITCH_CHANNEL_LOG_CLEAN,
mod_sofia_globals.tracelevel, "%s", fmt);
}

va_end(temp_ap);  //DOCKSTAR: Added line.  Release our copy of "ap".
}
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From steveayre at gmail.com  Mon Dec 13 12:41:19 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 13 Dec 2010 09:41:19 +0000
Subject: [Freeswitch-users] Scale UP Freeswitch
In-Reply-To: <AANLkTimS5ko3Yt155+z-vhpLUA9DG43y2vyUPMOFECj2@mail.gmail.com>
References: <AANLkTi=4n_wMSDE7KA9JTDhP+ToGggt-nJzf=EYwTd=6@mail.gmail.com>
	<AANLkTi=N+6=AcELMtbyje3O7W0k7qm3nz8z0CoO3LmLo@mail.gmail.com>
	<AANLkTikkCTMpa-mwD79Fv3W8+1CwS4cMs8Hovdi0sfVS@mail.gmail.com>
	<AANLkTikG=rPb-Lz7rMySa3gzJpjo42ftuthsPReD6ShJ@mail.gmail.com>
	<AANLkTinBf2Zc1o3Vagnha2C1YBba6YemjJjsf-UfCNht@mail.gmail.com>
	<AANLkTikHQtEM1f4sU6Ar0Gd5yeRsor0Gy_KOD7btQYW4@mail.gmail.com>
	<AANLkTimS5ko3Yt155+z-vhpLUA9DG43y2vyUPMOFECj2@mail.gmail.com>
Message-ID: <AANLkTi=YtFWGrP_AYfGOkfeMR6Z0OWMwju2ySTt5y6ua@mail.gmail.com>

You can use X-Auth-IP with a FS-FS call too:

Customer --> FS1 --> FS2
FS1 = front FS
FS2 = media server

1. Create a proxy ACL on FS2
2. Add the IP of FS1 to that ACL
3. On FS1 do this in the dialplan:

<extension ...>
  <condition ...>
    <action application="set" data="sip_h_X-Auth-IP=${network_addr}" />
    <action application="bridge" data="sofia/gateway/fs2/..." />
  </condition>
</extension>

FS2 will then be able to use the customer's IP in ACLs, user directory, etc.

Remember to either set inbound_bypass_media=true on the sip profile,
or <action application="set" data="bypass_media=true" /> in dialplan
before the bridge.

-Steve



On 12 December 2010 21:32, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> hmmm... so doing that will also require?X-Auth-IP, right or something more
> tricky can be done?
> On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> It is, but it relies on the caller supporting 3xx. They might not
>> handle the redirect.
>>
>> A lot won't because you could redirect them to anywhere, so lots of
>> implementations will ignore the 3xx. FreeSWITCH for instance can
>> either ignore a 3xx or will send the call back into the dialplan.
>>
>> I think you'll have more success having a FS server in front of the
>> others and bridging the call through to each server. If you set
>> inbound_bypass_media=true on the SIP profile, the RTP media will
>> bypass that server and go directly between the caller and the other FS
>> box. That means that the call won't be using any CPU since it'll only
>> wake up when a SIP packet is being sent/received. You'll still be
>> creating a session through so it'll still be allocating memory to the
>> call, a SIP proxy would use fewer resources.
>>
>> -Steve
>>
>>
>> On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
>> > Thanks Steve for suggestion, i'll check?X-Auth-IP, its new for me.
>> > Since we are talking about HA options... Is it practically doable use
>> > it:
>> >
>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2
>> > The idea is to run one FS box (Redirect-FS) in front of several FS boxes
>> > which redirect the call to active/available FS. If we make some script
>> > on
>> > redirect FS to count the active calls on media FSes and rearrange the
>> > order
>> > of redirect then loadbalacing can also be done.
>> > ...possible?
>> >
>> > On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre at gmail.com>
>> > wrote:
>> >>
>> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any
>> >> > difference between kamailo and opensips?
>> >>
>> >> They're both forks of OpenSER so for the most part there's little
>> >> difference.
>> >>
>> >> There are some small differences though since the fork. For example,
>> >> opensips has a load_balancer module which kamalio does not (kamalio
>> >> can still do load balancing but has a different interface to do so).
>> >>
>> >> > 2. if kamailo or opensips is running in front of FS, then will it
>> >> > send
>> >> > call
>> >> > to FS with original customer ip? so i can do billing etc on FS box
>> >> > -> actually i do IP based authentication and also ip based billing on
>> >> > FS
>> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
>> >> > original
>> >> > customer overview.
>> >>
>> >> It will appear coming from the proxy IP. But there is a workaround.
>> >> Configure a proxy ACL on the SIP profile and add your proxy IP to it.
>> >> Then adjust your proxy routing rules so that it adds a X-Auth-IP
>> >> header that contains the original IP.
>> >> Anything coming from anything in the proxy ACL is trusted and FS will
>> >> use the value from X-Auth-IP (if it exists).
>> >>
>> >> -Steve
>> >>
>> >>
>> >>
>> >>
>> >> On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981 at gmail.com>
>> >> wrote:
>> >> > Hi,
>> >> >
>> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any
>> >> > difference between kamailo and opensips?
>> >> >
>> >> > 2. if kamailo or opensips is running in front of FS, then will it
>> >> > send
>> >> > call
>> >> > to FS with original customer ip? so i can do billing etc on FS box
>> >> > -> actually i do IP based authentication and also ip based billing on
>> >> > FS
>> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
>> >> > original
>> >> > customer overview.
>> >> >
>> >> > thanks
>> >> > On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> There are a few performance tweaking tips at
>> >> >>
>> >> >> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
>> >> >>
>> >> >> Yes a Sangoma card will reduce your CPU load since transcoding won't
>> >> >> be done on the CPU any longer, that will then mean there's more CPU
>> >> >> available so you'll be able to handle more calls.
>> >> >>
>> >> >> However, if you're looking to increase your number of calls then you
>> >> >> probably want a cluster of servers as Juan pointed out.
>> >> >>
>> >> >> It'll mean you can increase the capacity by adding extra servers, so
>> >> >> there'd no longer be a limit to the number of calls you could handle
>> >> >> (just add another server).
>> >> >>
>> >> >> It'll also make maintenance easier, as you'll be able to pull a
>> >> >> server
>> >> >> from service for updates etc while traffic continues to run on the
>> >> >> other servers. Maintenance won't mean a service outage.
>> >> >>
>> >> >> If you're handling that many calls then additional servers would
>> >> >> make
>> >> >> your service more reliable. If a server crashes you'll still have
>> >> >> the
>> >> >> calls running on the other servers while you're fixing the problem
>> >> >> so
>> >> >> you won't have a complete outage. If FS is behind a load balancer
>> >> >> then
>> >> >> your customers might not even notice anything apart from a few
>> >> >> dropped
>> >> >> calls.
>> >> >>
>> >> >> There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will
>> >> >> attempt to continue calls if FS crashes and restarts, but I think
>> >> >> that's only for SIP-SIP not SIP-ISDN.
>> >> >>
>> >> >> -Steve
>> >> >>
>> >> >>
>> >> >>
>> >> >>
>> >> >> On 7 December 2010 12:26, Stephen Wilde <wstephen80 at gmail.com>
>> >> >> wrote:
>> >> >> > Hi,
>> >> >> > I have one server running Freeswitch with some ISDN connections
>> >> >> > (via
>> >> >> > FreeTDM+Sangoma boards) and some SIP connections with service
>> >> >> > providers
>> >> >> > and
>> >> >> > customer.
>> >> >> > The usage of Freeswitch is as switching so it "bridge" each
>> >> >> > incoming
>> >> >> > call to
>> >> >> > a new outgoing call.
>> >> >> > SIP calls use G.729 and ISDN calls use ALaw for voice encoding.
>> >> >> > Now the number of call is grow up and also the CPU load is a
>> >> >> > little
>> >> >> > high
>> >> >> > so
>> >> >> > I have the necessity to scale UP my Freeswitch to handle more
>> >> >> > calls:
>> >> >> > what is
>> >> >> > the best way to do that?
>> >> >> > My first idea is to use a Sangoma D500 board to reduce the CPU
>> >> >> > load.
>> >> >> > Can
>> >> >> > be
>> >> >> > this a solution?
>> >> >> > There are different way to scale UP?
>> >> >> > Thanks in advance,
>> >> >> > Stephen
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From melkybes at mail.ru  Mon Dec 13 13:12:35 2010
From: melkybes at mail.ru (=?koi8-r?Q?=ED=C9=C8=C1=C9=CC_=F3=C1=CC=D4=C1=CE=CF=D7?=)
Date: Mon, 13 Dec 2010 13:12:35 +0300
Subject: [Freeswitch-users] =?koi8-r?b?YmluZF9tZXRhX2FwcCB3aGF0IGNhbGwg?=
	=?koi8-r?b?bGVnPy1kZWNpc2lvbg==?=
Message-ID: <E1PS5OV-00074j-00.melkybes-mail-ru@f47.mail.ru>

Hello
The decision of this problem

in default.xml

<------><action application="set" data="bridge_pre_execute_aleg_app=bind_meta_app"/>
<------><action application="set" data="bridge_pre_execute_aleg_data=1 ab s execute_extension::dxa XML features"/>
<------><action application="set" data="bridge_pre_execute_bleg_app=bind_meta_app"/>
<------><action application="set" data="bridge_pre_execute_bleg_data=1 ab s execute_extension::dxb XML features"/>


in features.xml

    <!-- In call Transfer for phones without a transfer button -->
    <extension name="dxa">
      <condition field="destination_number" expression="^dxa$">
<------><action application="answer"/>
<------><action application="set" data="who_make_transfer=${caller_id_number}"/>
<------><action application="read" data="11 11 'tone_stream://%(10000,0,350,440)' digits 5000 #"/>
<------><action application="execute_extension" data="is_transfer XML features"/>
      </condition>
    </extension>
....
    <extension name="dxb">
      <condition field="destination_number" expression="^dxb$">
        <action application="set" data="who_make_transfer=${destination_number}"/>
      <><action application="answer"/>
<------><action application="read" data="11 11 'tone_stream://%(10000,0,350,440)' digits 5000 #"/>
<------><action application="set" data="accountcode=${who_make_transfer}"/>
<------><action application="execute_extension" data="is_transfer XML features"/>
      </condition>
    </extension>



? ?????????
?????? ????????



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From rafonline at hotmail.com  Mon Dec 13 14:16:03 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Mon, 13 Dec 2010 11:16:03 +0000
Subject: [Freeswitch-users] a2billing and freeswitch
Message-ID: <COL122-W4030C97574705F9B9DC69FB5120@phx.gbl>



Hi

I was wondering if a2billing can be setup to work with freeswitch?  

If so, can anyone please point me to some documentation on how this can be done?

Cheers

Raf
 		 	   		  
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From srinivas.ksvreddy at gmail.com  Mon Dec 13 14:24:08 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 13 Dec 2010 16:54:08 +0530
Subject: [Freeswitch-users] Error Loading mod_java.so
Message-ID: <AANLkTikz9gbLdKKx=aGJHxUZgwt4MqnLvVfad86PdyWV@mail.gmail.com>

Hi All,

Currently i am using JDK1.06.0_23 and jre also same version for using
mod_java module in freeswith,

i can able to generate mod_java.so and freeswitch.jar. and building and
installation had succussful without any proble.

when i am trying to run the freeswitch, i am gettig the following error.


2010-12-13 16:43:56.988097 [NOTICE] modjava.c:342 Java Framework Loading...
2010-12-13 16:43:57.042698 [CRIT] switch_loadable_module.c:928 Error Loading
module /usr/local/freeswitch/mod/mod_java.so
**Module load routine returned an error**
2010-12-13 16:43:57.043603 [CONSOLE] switch_loadable_module.c:946
Successfully Loaded [mod_lua]


and please find the following java.conf.xml file.

<configuration name="java.conf" description="Java Plug-Ins">
  <!-- Path to the Java 1.6 virtual machine to use -->
 <!--  <javavm
path="/usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre/lib/x86_64/client/libjvm.so"/>
-->
  <javavm
path="/usr/lib/jvm/java-1.6.0-openjdk-1.6.0.0.x86_64/jre/lib/amd64/server/libjvm.so"/>
  <!-- Options to pass to Java -->
  <options>
    <!-- Your class path (make sure freeswitch.jar is on it) -->
    <option
value="-Djava.class.path=/usr/local/freeswitch/scripts/freeswitch.jar
"/>
    <!-- Enable remote debugging -->
    <!-- <option
value="-agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address=
127.0.0.1:8001"/> -->
  </options>
  <startup class="net/cog/fs/system/Control" method="startup" arg="start up
arg"/>
  <shutdown class="net/cog/fs/system/Control" method="shutdown"
arg="shutdown arg"/>
</configuration>


Any Idea why i am getting the error?

Thanks
Srinivasula Reddy K
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From srinivas.ksvreddy at gmail.com  Mon Dec 13 14:39:35 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 13 Dec 2010 17:09:35 +0530
Subject: [Freeswitch-users] Fwd: Error Loading mod_java.so
In-Reply-To: <AANLkTikz9gbLdKKx=aGJHxUZgwt4MqnLvVfad86PdyWV@mail.gmail.com>
References: <AANLkTikz9gbLdKKx=aGJHxUZgwt4MqnLvVfad86PdyWV@mail.gmail.com>
Message-ID: <AANLkTimYd+7vrObud9iJ5t81u0pdO9+LtT2GC5Nar2KE@mail.gmail.com>

Hi All,

Currently i am using JDK1.06.0_23 and jre also same version for using
mod_java module in freeswith,

i can able to generate mod_java.so and freeswitch.jar. and building and
installation had succussful without any proble.

when i am trying to run the freeswitch, i am gettig the following error.


Exception in thread "Thread-0" java.lang.NoClassDefFoundError:
net/cog/fs/system/Control
Caused by: java.lang.ClassNotFoundException: net.cog.fs.system.Control
        at java.net.URLClassLoader$1.run(URLClassLoader.java:217)
        at java.security.AccessController.doPrivileged(Native Method)
        at java.net.URLClassLoader.findClass(URLClassLoader.java:205)
        at java.lang.ClassLoader.loadClass(ClassLoader.java:319)
        at sun.misc.Launcher$AppClassLoader.loadClass(Launcher.java:294)
        at java.lang.ClassLoader.loadClass(ClassLoader.java:264)
        at java.lang.ClassLoader.loadClassInternal(ClassLoader.java:332)
2010-12-13 17:07:58.193373 [CRIT] switch_loadable_module.c:928 Error Loading
module /usr/local/freeswitch/mod/mod_java.so
**Module load routine returned an error**



and please find the following java.conf.xml file.

<configuration name="java.conf" description="Java Plug-Ins">
  <!-- Path to the Java 1.6 virtual machine to use -->
 <!--  <javavm
path="/usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre/lib/x86_64/client/libjvm.so"/>
-->
  <javavm
path="/usr/lib/jvm/java-1.6.0-openjdk-1.6.0.0.x86_64/jre/lib/amd64/server/libjvm.so"/>
  <!-- Options to pass to Java -->
  <options>
    <!-- Your class path (make sure freeswitch.jar is on it) -->
    <option
value="-Djava.class.path=/usr/local/freeswitch/scripts/freeswitch.jar
"/>
    <!-- Enable remote debugging -->
    <!-- <option
value="-agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address=
127.0.0.1:8001"/> -->
  </options>
  <startup class="net/cog/fs/system/Control" method="startup" arg="start up
arg"/>
  <shutdown class="net/cog/fs/system/Control" method="shutdown"
arg="shutdown arg"/>
</configuration>


Any Idea why i am getting the error?

Thanks
Srinivasula Reddy K



-- 
Srinivasula Reddy K
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From Avi at aMarcus.com  Mon Dec 13 16:26:08 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Mon, 13 Dec 2010 15:26:08 +0200
Subject: [Freeswitch-users] Scale UP Freeswitch
In-Reply-To: <AANLkTi=YtFWGrP_AYfGOkfeMR6Z0OWMwju2ySTt5y6ua@mail.gmail.com>
References: <AANLkTi=4n_wMSDE7KA9JTDhP+ToGggt-nJzf=EYwTd=6@mail.gmail.com>
	<AANLkTi=N+6=AcELMtbyje3O7W0k7qm3nz8z0CoO3LmLo@mail.gmail.com>
	<AANLkTikkCTMpa-mwD79Fv3W8+1CwS4cMs8Hovdi0sfVS@mail.gmail.com>
	<AANLkTikG=rPb-Lz7rMySa3gzJpjo42ftuthsPReD6ShJ@mail.gmail.com>
	<AANLkTinBf2Zc1o3Vagnha2C1YBba6YemjJjsf-UfCNht@mail.gmail.com>
	<AANLkTikHQtEM1f4sU6Ar0Gd5yeRsor0Gy_KOD7btQYW4@mail.gmail.com>
	<AANLkTimS5ko3Yt155+z-vhpLUA9DG43y2vyUPMOFECj2@mail.gmail.com>
	<AANLkTi=YtFWGrP_AYfGOkfeMR6Z0OWMwju2ySTt5y6ua@mail.gmail.com>
Message-ID: <AANLkTikrrTGwn9Frz8msKD7p_rGuzSWwow72Z0i=wq9z@mail.gmail.com>

Can someone explain what the difference is between using FS with bypass
media and opensips? I've heard that opensips can handle a much higher CPS.
-Avi

On Mon, Dec 13, 2010 at 11:41 AM, Steven Ayre <steveayre at gmail.com> wrote:

> You can use X-Auth-IP with a FS-FS call too:
>
> Customer --> FS1 --> FS2
> FS1 = front FS
> FS2 = media server
>
> 1. Create a proxy ACL on FS2
> 2. Add the IP of FS1 to that ACL
> 3. On FS1 do this in the dialplan:
>
> <extension ...>
>  <condition ...>
>    <action application="set" data="sip_h_X-Auth-IP=${network_addr}" />
>    <action application="bridge" data="sofia/gateway/fs2/..." />
>  </condition>
> </extension>
>
> FS2 will then be able to use the customer's IP in ACLs, user directory,
> etc.
>
> Remember to either set inbound_bypass_media=true on the sip profile,
> or <action application="set" data="bypass_media=true" /> in dialplan
> before the bridge.
>
> -Steve
>
>
>
> On 12 December 2010 21:32, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> > hmmm... so doing that will also require X-Auth-IP, right or something
> more
> > tricky can be done?
> > On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>
> >> It is, but it relies on the caller supporting 3xx. They might not
> >> handle the redirect.
> >>
> >> A lot won't because you could redirect them to anywhere, so lots of
> >> implementations will ignore the 3xx. FreeSWITCH for instance can
> >> either ignore a 3xx or will send the call back into the dialplan.
> >>
> >> I think you'll have more success having a FS server in front of the
> >> others and bridging the call through to each server. If you set
> >> inbound_bypass_media=true on the SIP profile, the RTP media will
> >> bypass that server and go directly between the caller and the other FS
> >> box. That means that the call won't be using any CPU since it'll only
> >> wake up when a SIP packet is being sent/received. You'll still be
> >> creating a session through so it'll still be allocating memory to the
> >> call, a SIP proxy would use fewer resources.
> >>
> >> -Steve
> >>
> >>
> >> On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981 at gmail.com>
> wrote:
> >> > Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me.
> >> > Since we are talking about HA options... Is it practically doable use
> >> > it:
> >> >
> >> >
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2
> >> > The idea is to run one FS box (Redirect-FS) in front of several FS
> boxes
> >> > which redirect the call to active/available FS. If we make some script
> >> > on
> >> > redirect FS to count the active calls on media FSes and rearrange the
> >> > order
> >> > of redirect then loadbalacing can also be done.
> >> > ...possible?
> >> >
> >> > On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre at gmail.com>
> >> > wrote:
> >> >>
> >> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any
> >> >> > difference between kamailo and opensips?
> >> >>
> >> >> They're both forks of OpenSER so for the most part there's little
> >> >> difference.
> >> >>
> >> >> There are some small differences though since the fork. For example,
> >> >> opensips has a load_balancer module which kamalio does not (kamalio
> >> >> can still do load balancing but has a different interface to do so).
> >> >>
> >> >> > 2. if kamailo or opensips is running in front of FS, then will it
> >> >> > send
> >> >> > call
> >> >> > to FS with original customer ip? so i can do billing etc on FS box
> >> >> > -> actually i do IP based authentication and also ip based billing
> on
> >> >> > FS
> >> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
> >> >> > original
> >> >> > customer overview.
> >> >>
> >> >> It will appear coming from the proxy IP. But there is a workaround.
> >> >> Configure a proxy ACL on the SIP profile and add your proxy IP to it.
> >> >> Then adjust your proxy routing rules so that it adds a X-Auth-IP
> >> >> header that contains the original IP.
> >> >> Anything coming from anything in the proxy ACL is trusted and FS will
> >> >> use the value from X-Auth-IP (if it exists).
> >> >>
> >> >> -Steve
> >> >>
> >> >>
> >> >>
> >> >>
> >> >> On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981 at gmail.com>
> >> >> wrote:
> >> >> > Hi,
> >> >> >
> >> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any
> >> >> > difference between kamailo and opensips?
> >> >> >
> >> >> > 2. if kamailo or opensips is running in front of FS, then will it
> >> >> > send
> >> >> > call
> >> >> > to FS with original customer ip? so i can do billing etc on FS box
> >> >> > -> actually i do IP based authentication and also ip based billing
> on
> >> >> > FS
> >> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
> >> >> > original
> >> >> > customer overview.
> >> >> >
> >> >> > thanks
> >> >> > On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >> There are a few performance tweaking tips at
> >> >> >>
> >> >> >>
> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
> >> >> >>
> >> >> >> Yes a Sangoma card will reduce your CPU load since transcoding
> won't
> >> >> >> be done on the CPU any longer, that will then mean there's more
> CPU
> >> >> >> available so you'll be able to handle more calls.
> >> >> >>
> >> >> >> However, if you're looking to increase your number of calls then
> you
> >> >> >> probably want a cluster of servers as Juan pointed out.
> >> >> >>
> >> >> >> It'll mean you can increase the capacity by adding extra servers,
> so
> >> >> >> there'd no longer be a limit to the number of calls you could
> handle
> >> >> >> (just add another server).
> >> >> >>
> >> >> >> It'll also make maintenance easier, as you'll be able to pull a
> >> >> >> server
> >> >> >> from service for updates etc while traffic continues to run on the
> >> >> >> other servers. Maintenance won't mean a service outage.
> >> >> >>
> >> >> >> If you're handling that many calls then additional servers would
> >> >> >> make
> >> >> >> your service more reliable. If a server crashes you'll still have
> >> >> >> the
> >> >> >> calls running on the other servers while you're fixing the problem
> >> >> >> so
> >> >> >> you won't have a complete outage. If FS is behind a load balancer
> >> >> >> then
> >> >> >> your customers might not even notice anything apart from a few
> >> >> >> dropped
> >> >> >> calls.
> >> >> >>
> >> >> >> There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will
> >> >> >> attempt to continue calls if FS crashes and restarts, but I think
> >> >> >> that's only for SIP-SIP not SIP-ISDN.
> >> >> >>
> >> >> >> -Steve
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >> On 7 December 2010 12:26, Stephen Wilde <wstephen80 at gmail.com>
> >> >> >> wrote:
> >> >> >> > Hi,
> >> >> >> > I have one server running Freeswitch with some ISDN connections
> >> >> >> > (via
> >> >> >> > FreeTDM+Sangoma boards) and some SIP connections with service
> >> >> >> > providers
> >> >> >> > and
> >> >> >> > customer.
> >> >> >> > The usage of Freeswitch is as switching so it "bridge" each
> >> >> >> > incoming
> >> >> >> > call to
> >> >> >> > a new outgoing call.
> >> >> >> > SIP calls use G.729 and ISDN calls use ALaw for voice encoding.
> >> >> >> > Now the number of call is grow up and also the CPU load is a
> >> >> >> > little
> >> >> >> > high
> >> >> >> > so
> >> >> >> > I have the necessity to scale UP my Freeswitch to handle more
> >> >> >> > calls:
> >> >> >> > what is
> >> >> >> > the best way to do that?
> >> >> >> > My first idea is to use a Sangoma D500 board to reduce the CPU
> >> >> >> > load.
> >> >> >> > Can
> >> >> >> > be
> >> >> >> > this a solution?
> >> >> >> > There are different way to scale UP?
> >> >> >> > Thanks in advance,
> >> >> >> > Stephen
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.ponzone at ipeva.fr  Mon Dec 13 16:36:15 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 13 Dec 2010 14:36:15 +0100
Subject: [Freeswitch-users] Scale UP Freeswitch
In-Reply-To: <AANLkTikrrTGwn9Frz8msKD7p_rGuzSWwow72Z0i=wq9z@mail.gmail.com>
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	<AANLkTinBf2Zc1o3Vagnha2C1YBba6YemjJjsf-UfCNht@mail.gmail.com>
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	<AANLkTikrrTGwn9Frz8msKD7p_rGuzSWwow72Z0i=wq9z@mail.gmail.com>
Message-ID: <DD8368B3-5074-4781-9DF2-D90992488DC8@ipeva.fr>

Well OpenSIPS is a proxy and FreeSWITCH is a B2BUA.
If you just need to load-balance calls to your main FS boxes (which are used as B2BUA), to use another B2BUA for that task is not really useful.
A SIP Proxy is generally more efficient for this, as they are dedicated to handle lots of REGISTER & INVITE, but if you want to avoid the trouble to deal with OpenSIPS config, which I find far less intuitive than FreeSWITCH, you may use FreeSWITCH in bypass-media as proposed in this thread.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

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Le 13/12/2010 ? 14:26, Avi Marcus a ?crit :

> Can someone explain what the difference is between using FS with bypass media and opensips? I've heard that opensips can handle a much higher CPS.
> -Avi
> 
> On Mon, Dec 13, 2010 at 11:41 AM, Steven Ayre <steveayre at gmail.com> wrote:
> You can use X-Auth-IP with a FS-FS call too:
> 
> Customer --> FS1 --> FS2
> FS1 = front FS
> FS2 = media server
> 
> 1. Create a proxy ACL on FS2
> 2. Add the IP of FS1 to that ACL
> 3. On FS1 do this in the dialplan:
> 
> <extension ...>
>  <condition ...>
>    <action application="set" data="sip_h_X-Auth-IP=${network_addr}" />
>    <action application="bridge" data="sofia/gateway/fs2/..." />
>  </condition>
> </extension>
> 
> FS2 will then be able to use the customer's IP in ACLs, user directory, etc.
> 
> Remember to either set inbound_bypass_media=true on the sip profile,
> or <action application="set" data="bypass_media=true" /> in dialplan
> before the bridge.
> 
> -Steve
> 
> 
> 
> On 12 December 2010 21:32, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> > hmmm... so doing that will also require X-Auth-IP, right or something more
> > tricky can be done?
> > On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <steveayre at gmail.com> wrote:
> >>
> >> It is, but it relies on the caller supporting 3xx. They might not
> >> handle the redirect.
> >>
> >> A lot won't because you could redirect them to anywhere, so lots of
> >> implementations will ignore the 3xx. FreeSWITCH for instance can
> >> either ignore a 3xx or will send the call back into the dialplan.
> >>
> >> I think you'll have more success having a FS server in front of the
> >> others and bridging the call through to each server. If you set
> >> inbound_bypass_media=true on the SIP profile, the RTP media will
> >> bypass that server and go directly between the caller and the other FS
> >> box. That means that the call won't be using any CPU since it'll only
> >> wake up when a SIP packet is being sent/received. You'll still be
> >> creating a session through so it'll still be allocating memory to the
> >> call, a SIP proxy would use fewer resources.
> >>
> >> -Steve
> >>
> >>
> >> On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> >> > Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me.
> >> > Since we are talking about HA options... Is it practically doable use
> >> > it:
> >> >
> >> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2
> >> > The idea is to run one FS box (Redirect-FS) in front of several FS boxes
> >> > which redirect the call to active/available FS. If we make some script
> >> > on
> >> > redirect FS to count the active calls on media FSes and rearrange the
> >> > order
> >> > of redirect then loadbalacing can also be done.
> >> > ...possible?
> >> >
> >> > On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre at gmail.com>
> >> > wrote:
> >> >>
> >> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any
> >> >> > difference between kamailo and opensips?
> >> >>
> >> >> They're both forks of OpenSER so for the most part there's little
> >> >> difference.
> >> >>
> >> >> There are some small differences though since the fork. For example,
> >> >> opensips has a load_balancer module which kamalio does not (kamalio
> >> >> can still do load balancing but has a different interface to do so).
> >> >>
> >> >> > 2. if kamailo or opensips is running in front of FS, then will it
> >> >> > send
> >> >> > call
> >> >> > to FS with original customer ip? so i can do billing etc on FS box
> >> >> > -> actually i do IP based authentication and also ip based billing on
> >> >> > FS
> >> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
> >> >> > original
> >> >> > customer overview.
> >> >>
> >> >> It will appear coming from the proxy IP. But there is a workaround.
> >> >> Configure a proxy ACL on the SIP profile and add your proxy IP to it.
> >> >> Then adjust your proxy routing rules so that it adds a X-Auth-IP
> >> >> header that contains the original IP.
> >> >> Anything coming from anything in the proxy ACL is trusted and FS will
> >> >> use the value from X-Auth-IP (if it exists).
> >> >>
> >> >> -Steve
> >> >>
> >> >>
> >> >>
> >> >>
> >> >> On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981 at gmail.com>
> >> >> wrote:
> >> >> > Hi,
> >> >> >
> >> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any
> >> >> > difference between kamailo and opensips?
> >> >> >
> >> >> > 2. if kamailo or opensips is running in front of FS, then will it
> >> >> > send
> >> >> > call
> >> >> > to FS with original customer ip? so i can do billing etc on FS box
> >> >> > -> actually i do IP based authentication and also ip based billing on
> >> >> > FS
> >> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
> >> >> > original
> >> >> > customer overview.
> >> >> >
> >> >> > thanks
> >> >> > On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >> There are a few performance tweaking tips at
> >> >> >>
> >> >> >> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
> >> >> >>
> >> >> >> Yes a Sangoma card will reduce your CPU load since transcoding won't
> >> >> >> be done on the CPU any longer, that will then mean there's more CPU
> >> >> >> available so you'll be able to handle more calls.
> >> >> >>
> >> >> >> However, if you're looking to increase your number of calls then you
> >> >> >> probably want a cluster of servers as Juan pointed out.
> >> >> >>
> >> >> >> It'll mean you can increase the capacity by adding extra servers, so
> >> >> >> there'd no longer be a limit to the number of calls you could handle
> >> >> >> (just add another server).
> >> >> >>
> >> >> >> It'll also make maintenance easier, as you'll be able to pull a
> >> >> >> server
> >> >> >> from service for updates etc while traffic continues to run on the
> >> >> >> other servers. Maintenance won't mean a service outage.
> >> >> >>
> >> >> >> If you're handling that many calls then additional servers would
> >> >> >> make
> >> >> >> your service more reliable. If a server crashes you'll still have
> >> >> >> the
> >> >> >> calls running on the other servers while you're fixing the problem
> >> >> >> so
> >> >> >> you won't have a complete outage. If FS is behind a load balancer
> >> >> >> then
> >> >> >> your customers might not even notice anything apart from a few
> >> >> >> dropped
> >> >> >> calls.
> >> >> >>
> >> >> >> There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will
> >> >> >> attempt to continue calls if FS crashes and restarts, but I think
> >> >> >> that's only for SIP-SIP not SIP-ISDN.
> >> >> >>
> >> >> >> -Steve
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >> On 7 December 2010 12:26, Stephen Wilde <wstephen80 at gmail.com>
> >> >> >> wrote:
> >> >> >> > Hi,
> >> >> >> > I have one server running Freeswitch with some ISDN connections
> >> >> >> > (via
> >> >> >> > FreeTDM+Sangoma boards) and some SIP connections with service
> >> >> >> > providers
> >> >> >> > and
> >> >> >> > customer.
> >> >> >> > The usage of Freeswitch is as switching so it "bridge" each
> >> >> >> > incoming
> >> >> >> > call to
> >> >> >> > a new outgoing call.
> >> >> >> > SIP calls use G.729 and ISDN calls use ALaw for voice encoding.
> >> >> >> > Now the number of call is grow up and also the CPU load is a
> >> >> >> > little
> >> >> >> > high
> >> >> >> > so
> >> >> >> > I have the necessity to scale UP my Freeswitch to handle more
> >> >> >> > calls:
> >> >> >> > what is
> >> >> >> > the best way to do that?
> >> >> >> > My first idea is to use a Sangoma D500 board to reduce the CPU
> >> >> >> > load.
> >> >> >> > Can
> >> >> >> > be
> >> >> >> > this a solution?
> >> >> >> > There are different way to scale UP?
> >> >> >> > Thanks in advance,
> >> >> >> > Stephen
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From bernhard.suttner at winet.ch  Mon Dec 13 16:40:12 2010
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Mon, 13 Dec 2010 14:40:12 +0100
Subject: [Freeswitch-users] Scale UP Freeswitch
In-Reply-To: <AANLkTikrrTGwn9Frz8msKD7p_rGuzSWwow72Z0i=wq9z@mail.gmail.com>
References: <AANLkTi=4n_wMSDE7KA9JTDhP+ToGggt-nJzf=EYwTd=6@mail.gmail.com>
	<AANLkTi=N+6=AcELMtbyje3O7W0k7qm3nz8z0CoO3LmLo@mail.gmail.com>
	<AANLkTikkCTMpa-mwD79Fv3W8+1CwS4cMs8Hovdi0sfVS@mail.gmail.com>
	<AANLkTikG=rPb-Lz7rMySa3gzJpjo42ftuthsPReD6ShJ@mail.gmail.com>
	<AANLkTinBf2Zc1o3Vagnha2C1YBba6YemjJjsf-UfCNht@mail.gmail.com>
	<AANLkTikHQtEM1f4sU6Ar0Gd5yeRsor0Gy_KOD7btQYW4@mail.gmail.com>
	<AANLkTimS5ko3Yt155+z-vhpLUA9DG43y2vyUPMOFECj2@mail.gmail.com>
	<AANLkTi=YtFWGrP_AYfGOkfeMR6Z0OWMwju2ySTt5y6ua@mail.gmail.com>
	<AANLkTikrrTGwn9Frz8msKD7p_rGuzSWwow72Z0i=wq9z@mail.gmail.com>
Message-ID: <5bd44d16-ed8a-43fc-92c7-36312ba077f8@winet.ch>

OpenSIPS / Kamailio / OpenSER is just a SIP Proxy. Proxy nearly in the same way than a HTTP proxy. It will normally just forward messages from A to B. OpenSIPS could also be used as a registrar. OpenSIPS does NEVER handle media- this task could be done within a scenario with mediaproxy/rtpproxy.
 
FreeSWITCH with bypass_media enabled does not handle media but does still handle all the SIP requests. Simple example:
 
OpenSIPS could forward a REGISTER to another peer without a check if the user does exists.
FreeSWITCH would handle the REGISTER and all the other SIP messages.
 
 
 
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Avi Marcus
Gesendet: Montag, 13. Dezember 2010 14:26
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Scale UP Freeswitch
 
Can someone explain what the difference is between using FS with bypass media and opensips? I've heard that opensips can handle a much higher CPS.
-Avi
On Mon, Dec 13, 2010 at 11:41 AM, Steven Ayre <steveayre at gmail.com> wrote:
You can use X-Auth-IP with a FS-FS call too:

Customer --> FS1 --> FS2
FS1 = front FS
FS2 = media server

1. Create a proxy ACL on FS2
2. Add the IP of FS1 to that ACL
3. On FS1 do this in the dialplan:

<extension ...>
 <condition ...>
   <action application="set" data="sip_h_X-Auth-IP=${network_addr}" />
   <action application="bridge" data="sofia/gateway/fs2/..." />
 </condition>
</extension>

FS2 will then be able to use the customer's IP in ACLs, user directory, etc.

Remember to either set inbound_bypass_media=true on the sip profile,
or <action application="set" data="bypass_media=true" /> in dialplan
before the bridge.

-Steve



On 12 December 2010 21:32, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> hmmm... so doing that will also require X-Auth-IP, right or something more
> tricky can be done?
> On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> It is, but it relies on the caller supporting 3xx. They might not
>> handle the redirect.
>>
>> A lot won't because you could redirect them to anywhere, so lots of
>> implementations will ignore the 3xx. FreeSWITCH for instance can
>> either ignore a 3xx or will send the call back into the dialplan.
>>
>> I think you'll have more success having a FS server in front of the
>> others and bridging the call through to each server. If you set
>> inbound_bypass_media=true on the SIP profile, the RTP media will
>> bypass that server and go directly between the caller and the other FS
>> box. That means that the call won't be using any CPU since it'll only
>> wake up when a SIP packet is being sent/received. You'll still be
>> creating a session through so it'll still be allocating memory to the
>> call, a SIP proxy would use fewer resources.
>>
>> -Steve
>>
>>
>> On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
>> > Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me.
>> > Since we are talking about HA options... Is it practically doable use
>> > it:
>> >
>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2
>> > The idea is to run one FS box (Redirect-FS) in front of several FS boxes
>> > which redirect the call to active/available FS. If we make some script
>> > on
>> > redirect FS to count the active calls on media FSes and rearrange the
>> > order
>> > of redirect then loadbalacing can also be done.
>> > ...possible?
>> >
>> > On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre at gmail.com>
>> > wrote:
>> >>
>> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any
>> >> > difference between kamailo and opensips?
>> >>
>> >> They're both forks of OpenSER so for the most part there's little
>> >> difference.
>> >>
>> >> There are some small differences though since the fork. For example,
>> >> opensips has a load_balancer module which kamalio does not (kamalio
>> >> can still do load balancing but has a different interface to do so).
>> >>
>> >> > 2. if kamailo or opensips is running in front of FS, then will it
>> >> > send
>> >> > call
>> >> > to FS with original customer ip? so i can do billing etc on FS box
>> >> > -> actually i do IP based authentication and also ip based billing on
>> >> > FS
>> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
>> >> > original
>> >> > customer overview.
>> >>
>> >> It will appear coming from the proxy IP. But there is a workaround.
>> >> Configure a proxy ACL on the SIP profile and add your proxy IP to it.
>> >> Then adjust your proxy routing rules so that it adds a X-Auth-IP
>> >> header that contains the original IP.
>> >> Anything coming from anything in the proxy ACL is trusted and FS will
>> >> use the value from X-Auth-IP (if it exists).
>> >>
>> >> -Steve
>> >>
>> >>
>> >>
>> >>
>> >> On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981 at gmail.com>
>> >> wrote:
>> >> > Hi,
>> >> >
>> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any
>> >> > difference between kamailo and opensips?
>> >> >
>> >> > 2. if kamailo or opensips is running in front of FS, then will it
>> >> > send
>> >> > call
>> >> > to FS with original customer ip? so i can do billing etc on FS box
>> >> > -> actually i do IP based authentication and also ip based billing on
>> >> > FS
>> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
>> >> > original
>> >> > customer overview.
>> >> >
>> >> > thanks
>> >> > On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> There are a few performance tweaking tips at
>> >> >>
>> >> >> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
>> >> >>
>> >> >> Yes a Sangoma card will reduce your CPU load since transcoding won't
>> >> >> be done on the CPU any longer, that will then mean there's more CPU
>> >> >> available so you'll be able to handle more calls.
>> >> >>
>> >> >> However, if you're looking to increase your number of calls then you
>> >> >> probably want a cluster of servers as Juan pointed out.
>> >> >>
>> >> >> It'll mean you can increase the capacity by adding extra servers, so
>> >> >> there'd no longer be a limit to the number of calls you could handle
>> >> >> (just add another server).
>> >> >>
>> >> >> It'll also make maintenance easier, as you'll be able to pull a
>> >> >> server
>> >> >> from service for updates etc while traffic continues to run on the
>> >> >> other servers. Maintenance won't mean a service outage.
>> >> >>
>> >> >> If you're handling that many calls then additional servers would
>> >> >> make
>> >> >> your service more reliable. If a server crashes you'll still have
>> >> >> the
>> >> >> calls running on the other servers while you're fixing the problem
>> >> >> so
>> >> >> you won't have a complete outage. If FS is behind a load balancer
>> >> >> then
>> >> >> your customers might not even notice anything apart from a few
>> >> >> dropped
>> >> >> calls.
>> >> >>
>> >> >> There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will
>> >> >> attempt to continue calls if FS crashes and restarts, but I think
>> >> >> that's only for SIP-SIP not SIP-ISDN.
>> >> >>
>> >> >> -Steve
>> >> >>
>> >> >>
>> >> >>
>> >> >>
>> >> >> On 7 December 2010 12:26, Stephen Wilde <wstephen80 at gmail.com>
>> >> >> wrote:
>> >> >> > Hi,
>> >> >> > I have one server running Freeswitch with some ISDN connections
>> >> >> > (via
>> >> >> > FreeTDM+Sangoma boards) and some SIP connections with service
>> >> >> > providers
>> >> >> > and
>> >> >> > customer.
>> >> >> > The usage of Freeswitch is as switching so it "bridge" each
>> >> >> > incoming
>> >> >> > call to
>> >> >> > a new outgoing call.
>> >> >> > SIP calls use G.729 and ISDN calls use ALaw for voice encoding.
>> >> >> > Now the number of call is grow up and also the CPU load is a
>> >> >> > little
>> >> >> > high
>> >> >> > so
>> >> >> > I have the necessity to scale UP my Freeswitch to handle more
>> >> >> > calls:
>> >> >> > what is
>> >> >> > the best way to do that?
>> >> >> > My first idea is to use a Sangoma D500 board to reduce the CPU
>> >> >> > load.
>> >> >> > Can
>> >> >> > be
>> >> >> > this a solution?
>> >> >> > There are different way to scale UP?
>> >> >> > Thanks in advance,
>> >> >> > Stephen
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From Nabble at slickdeals.endjunk.com  Mon Dec 13 17:12:21 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 13 Dec 2010 06:12:21 -0800 (PST)
Subject: [Freeswitch-users] DockStar compile failure, with possible fix
In-Reply-To: <AANLkTi=-1xKh21fR5PADAQs2YPRKHjssCLQMRWNikSu6@mail.gmail.com>
References: <AANLkTi=-1xKh21fR5PADAQs2YPRKHjssCLQMRWNikSu6@mail.gmail.com>
Message-ID: <1292249541197-5830855.post@n2.nabble.com>


If you want, please take a look at posts on this 
http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-td5252219.html
thread .

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/DockStar-compile-failure-with-possible-fix-tp5829897p5830855.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From marcdecorny at gmail.com  Mon Dec 13 17:18:22 2010
From: marcdecorny at gmail.com (Marc de Corny)
Date: Mon, 13 Dec 2010 14:18:22 +0000
Subject: [Freeswitch-users] Test - Please ignore
Message-ID: <AANLkTi=WsCgi++Da72CrtUeQ71nV6JDBe5oZsiu1TcFj@mail.gmail.com>

Test - Please ignore

thanks
Marc
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From gustavo.espeche at upper-soft.com  Mon Dec 13 17:24:12 2010
From: gustavo.espeche at upper-soft.com (Gustavo Espeche)
Date: Mon, 13 Dec 2010 11:24:12 -0300
Subject: [Freeswitch-users] opal to sip problem
Message-ID: <1292250252.2159.14.camel@gustavo-laptop>

Hello list, we are try to do inter-worker between h323 and sip using
opal in freeswitch, but FS don't send the call to our sip gw, follow is
the  call flow:

h323 endpoint -->FS->sip gateway

attached a FS debug of call.

if something know or work with opal in freeSwitch and have some tips we
appreciate a lot if advice about it.
Best Regards.
-----------------------------------------------------------------------------------------------------------------------------
2010-12-13 11:58:21.325242 [DEBUG] transports.cxx:1200 Started
connection to 200.117.192.17:19270 (if=72.51.47.100:1720)
2010-12-13 11:58:21.325242 [DEBUG] osutil.cxx:189 File handle high water
mark set: 67 Thread unblock pipe
2010-12-13 11:58:21.325242 [DEBUG] tlibthrd.cxx:587 Thread high water
mark set: 8
2010-12-13 11:58:21.325242 [DEBUG] transports.cxx:666 Waiting on socket
accept on tcp$72.51.47.100:1720
2010-12-13 11:58:21.325242 [DEBUG] h323ep.cxx:501 Awaiting first PDU
2010-12-13 11:58:21.805203 [DEBUG] h323pdu.cxx:80 Receiving PDU:
  {
    q931pdu = {
      protocolDiscriminator = 8
      callReference = 22716
      from = originator
      messageType = Setup
      IE: Bearer-Capability = {
        88 93 a5                            
               ...
      }
      IE: Display = {
        67 75 73 74 61 76 6f 00                            gustavo.
      }
      IE: Called-Party-Number = {
        81 32 33 31 35 34 33 35  31 34 32 38 30 36 33
33   .231543514280633
      }
      
IE: User-User = {
        20 b8 06 00 08 91 4a 00  06 01 40 06 00 67 00
75    .....J... at ..g.u
        00 73 00 74 00 61 00 76  00 6f 22 c0 09 00 00
3d   .s.t.a.v.o"....=
        ...
      }
    }
    h225pdu = {
      h323_uu_pdu = {
        h323_me
ssage_body = setup {
          protocolIdentifier = 0.0.8.2250.0.6
          sourceAddress = 1 entries {
            [0]=h323_ID  7 characters {
              0067 0075 0073 0074 0061 0076 006f        gustavo
            }
          }
          sour
ceInfo = {
            vendor = {
              vendor = {
                t35CountryCode = 9
                t35Extension = 0
                manufacturerCode = 61
              }
              productId =  3 octets {
                00 00 00      
(OPAL v3.6
                2e 36 29 00 00                                     .6
)..
              }
            }
            terminal = {
            }
            mc = false
            undefinedNode = false
          }
          destinationAddress = 1 entries {
            [0]=dialedDigits "231543514280633"
          }
     
Log-Func: 
Log-Line: 0
User-Data: 

     destCallSignalAddress = ipAddress {
            ip =  4 octets {
              48 33 2f 64                                        H3/d
            }
            port = 1720
          }
          activeMC = falets {
            4e 76 b2 f2 1d 05 e0 11  9b b7 00 1e 64 1f 33 fe
Nv..........d.3.
          }
          conferenceGoal = create <<null>>
          callType = pointToPoint <<null>>
          sourceCallSignalAddress = ipAddress {
            ip = 
Log-Func: 
Log-Line: 0
User-Data: 

 4 octets {
              c8 75 c0 11                                        .u..
            }
            port = 19270
          }
          callIdentifier = {
            guid =  16 octets {
              3a 76 f 33 fe   :v..........d.3.
            }
          }
          fastStart = 8 entries {
            [0]= 29 octets {
              40 00 00 06 04 01 00 4c  20 13 80 11 1c 00 01 00
@......L .......
              c8 75 c0 11 13 de 00 c8  75 c0 11 13 
df            .u......u....
            }
            [1]= 19 octets {
              00 00 64 0c 20 13 80 0b  0d 00 01 00 c8 75 c0
11   ..d. ........u..
              13 df 00                                           ...
            }
            [
2]= 35 octets {
              40 00 00 06 04 01 00 48  78 00 4a ff 00 80 01 00
@......Hx.J.....
              80 11 1c 00 02 00 c8 75  c0 11 13 e0 00 c8 75
c0   .......u......u.
              ...
            }
            [3]= 25 octets {
        
      00 00 65 08 78 00 4a ff  00 80 01 00 80 0b 0d
00   ..e.x.J.........
              02 00 c8 75 c0 11 13 e1
00                        ...u.....
            }
            [4]= 34 octets {
              40 00 00 06 04 01 00 48  68 4a ff 00 80 01 
00 80   @......HhJ......
              11 1c 00 02 00 c8 75 c0  11 13 e0 00 c8 75 c0
11   ......u......u..
              ...
            }
            [5]= 24 octets {
              00 00 66 08 68 4a ff 00  80 01 00 80 0b 0d 00
02   ..f.hJ..........

              00 c8 75 c0 11 13 e1
00                            ..u.....
            }
            [6]= 34 octets {
              40 00 00 06 04 01 00 48  70 4a ff 00 80 01 00 80
@......HpJ......
              11 1c 00 02 00 c8 75 c0  11 13 e0 0
0 c8 75 c0 11   ......u......u..
              ...
            }
            [7]= 24 octets {
              00 00 67 08 70 4a ff 00  80 01 00 80 0b 0d 00
02   ..g.pJ..........
              00 c8 75 c0 11 13 e1
00                            ..u.....

            }
          }
          mediaWaitForConnect = false
          canOverlapSend = false
          multipleCalls = false
          maintainConnection = false
          parallelH245Control = 2 entries {
            [0]= 125 octets {
        
      02 70 01 06 00 08 81 75  00 0d 80 13 80 01 f4
00   .p.....u........
              01 00 00 01 00 00 01 00  00 0c c0 01 00 01 80
07   ................
              ...
            }
            [1]= 7 octets {
              01 00 32 80 5d 41 2
b                               ..2.]A+
            }
          }
        }
        h245Tunneling = true
      }
    }
  }
2010-12-13 11:58:21.805203 [DEBUG] h323ep.cxx:510 Incoming call, first
PDU: callReference=22716
2010-12-13 11:58:21.805203 [DEBUG] call.cxx:72 Created Call[Cc70e3cf91]
2010-12-13 11:58:21.805203 [DEBUG] connection.cxx:262 Created connection
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
2010-12-13 11:58:21.805203 [DEBUG] rfc2833.cxx:140 Handler created
2010-12-13 11:58:21.805203 [DEBUG] rfc2833.cxx:328 Rx capability set to
"0-16,32,36"
2010-12-13 11:58:21.805203 [DEBUG] rfc2833.cxx:140 Handler created
2010-12-13 11:58:21.805203 [DEBUG] rfc2833.cxx:328 Rx capability set to
"192,193"
2010-12-13 11:58:21.805203 [DEBUG] osutil.cxx:189 File handle high water
mark set: 68 PUDPSocket
2010-12-13 11:58:21.808186 [DEBUG] h4601.cxx:1583 Endpoint Attached
2010-12-13 11:58:21.808186 [DEBUG] h323ep.cxx:552 Created new
connection: tcp$200.117.192.17:19270/22716
2010-12-13 11:58:21.808186 [DEBUG] h323.cxx:559 Handling PDU: Setup
callRef=22716
2010-12-13 11:58:21.808186 [DEBUG] connection.cxx:1516 SetPhase from
UninitialisedPhase to SetUpPhase for Call[Cc70e3cf91]-EP<h323>[tcp
$200.117.192.17:19270/22716]
2010-12-13 11:58:21.808186 [DEBUG] h323.cxx:943 Set protocol version to
6 and implying H.245 version 13
2010-12-13 11:58:21.808186 [DEBUG] h323.cxx:1240 Set remote application
name: " 3.2.6 (OPAL v3.6.6) 9/61 "
2010-12-13 11:58:21.808186 [DEBUG] manager.cxx:1392 Checking incoming
call for NAT: local=72.51.47.100, peer=200.117.192.17,
sig=200.117.192.17
2010-12-13 11:58:21.808186 [DEBUG] h323.cxx:1041 Sending call proceeding
PDU
2010-12-13 11:58:21.811196 [DEBUG] h323pdu.cxx:80 Sending PDU:
  {
    q931pdu = {
      protocolDiscriminator = 8
      callReference = 22716
      from = destination
      messageType = CallProceeding
      IE: Display = {
        72 6f 6f 74 00                        
             root.
      }
      IE: User-User = {
        21 80 06 00 08 91 4a 00  06 22 c0 09 00 00 3d
1e   !.....J.."....=.
        56 6f 78 20 4c 75 63 69  64 61 20 50 74 79 2e 20   Vox Lucida
Pty. 
        ...
      }
    }
    h225pdu = {
    
  h323_uu_pdu = {
        h323_message_body = callProceeding {
          protocolIdentifier = 0.0.8.2250.0.6
          destinationInfo = {
            vendor = {
              vendor = {
                t35CountryCode = 9
                t35Extensio
n = 0
                manufacturerCode = 61
              }
              productId =  31 octets {
                56 6f 78 20 4c 75 63 69  64 61 20 50 74 79 2e 20   Vox
Lucida Pty. 
                4c 74 64 2e 20 6d 6f 64  5f 6f 70 61 6c 00 00     
Log-Line: 0
UINCOMING DATA [(null)]

RECV EVENT
Event-Name: SOCKET_DATA
ser-Data: _undef_


 }
            }
            terminal = {
            }
            mc = false
            undefinedNode = false
          }
          callIdentifier = {
            guid =  16 octets {
              3a 76 b2 f2 1d 05 e0 11  9b b7 00 1e 64 1f 33 fe 
  :v..........d.3.
            }
          }
          multipleCalls = false
          maintainConnection = false
        }
        h245Tunneling = true
      }
    }
  }
2010-12-13 11:58:21.811196 [DEBUG] manager.cxx:600 OnIncoming connection
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
2010-12-13 11:58:21.811196 [DEBUG] call.cxx:288 GetOtherPartyConnection
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
2010-12-13 11:58:21.811196 [DEBUG] manager.cxx:1200 Searching for route
"h323:root 231543514280633"
2010-12-13 11:58:21.811196 [DEBUG] manager.cxx:1228 Matched regex
"h323:.*"
2010-12-13 11:58:21.811196 [DEBUG] manager.cxx:573 Set up connection to
"local:231543514280633"
2010-12-13 11:58:21.811196 [DEBUG] connection.cxx:262 Created connection
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.811196 [DEBUG] localep.cxx:205 Created connection
with token "Ldc8d6c462"
2010-12-13 11:58:21.811196 [DEBUG] h323.cxx:1075 Incoming call accepted
2010-12-13 11:58:21.811196 [DEBUG] mod_opal.cpp:770 Matched FS codec
PCMA to OPAL media format G.711-ALaw-64k
2010-12-13 11:58:21.811196 [DEBUG] mod_opal.cpp:770 Matched FS codec
G729 to OPAL media format G.729
2010-12-13 11:58:21.811196 [DEBUG] mod_opal.cpp:770 Matched FS codec
PCMU to OPAL media format G.711-uLaw-64k
2010-12-13 11:58:21.811196 [DEBUG] mod_opal.cpp:770 Matched FS codec
G723 to OPAL media format G.723.1
2010-12-13 11:58:21.814188 [DEBUG] call.cxx:425 GetMediaFormats for
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
G.723.1
G.729
G.711-uLaw-64k
G.711-ALaw-64k

2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2381 Could not find
capability: "G.723.1"
2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
G.723.1 <1>
2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2381 Could not find
capability: "G.729"
2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
G.729 <2>
2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2381 Could not find
capability: "G.711-uLaw-64k"
2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
G.711-uLaw-64k <3>
2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2381 Could not find
capability: "G.711-ALaw-64k"
2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
G.711-ALaw-64k <4>
2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2376 Found capability:
G.723.1 <1>
2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2376 Found capability:
G.729 <2>
2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2376 Found capability:
G.711-uLaw-64k <3>
2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2376 Found capability:
G.711-ALaw-64k <4>
2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
UserInput/hookflash <5>
2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
UserInput/basicString <6>
2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2284 Added capability:
UserInput/dtmf <7>
2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2284 Added capability:
UserInput/RFC2833 <8>
2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2376 Found capability:
UserInput/RFC2833 <8>
2010-12-13 11:58:21.817197 [DEBUG] call.cxx:288 GetOtherPartyConnection
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
2010-12-13 11:58:21.817197 [DEBUG] h323.cxx:3704 SetLocalCapabilities:
 Table:
   G.723.1 <1>
   G.729 <2>
   G.711-uLaw-64k <3>
   G.711-ALaw-64k <4>
   UserInput/hookflash <5>
   UserInput/basicString <6>
   UserInput/dtmf <7>
   UserInput/RFC2833 <8>
 Set:
   0:
  

Log-Func: 
Log-Line: 0
User-Data: 

   0:
       G.723.1 <1>
       G.729 <2>
       G.711-uLaw-64k <3>
       G.711-ALaw-64k <4>
     1:
       UserInput/hookflash <5>
     2:
       UserInput/basicString <6>
       UserInput/dtINCOMING DATA [(null)]

RECV EVENT
Event-Name: SOCKET_DATA


2010-12-13 11:58:21.817197 [DEBUG] h323.cxx:1121 Fast start detected
2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2594 Could not find
capability: audioData, type g711Alaw64k
2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:845 Capability tx frames
left at 20 as remote allows 20
2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2294 Added capability:
G.711-ALaw-64k <4>
2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:845 Capability tx frames
left at 20 as remote allows 20
2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2594 Could not find
capability: videoData, type h261VideoCapability
2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2594 Could not find
capability: videoData, type h261VideoCapability
2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2594 Could not find
capability: videoData, type h261VideoCapability
2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2594 Could not find
capability: videoData, type h261VideoCapability
2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
capability: videoData, type h261VideoCapability
2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
capability: videoData, type h261VideoCapability
2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
capability: videoData, type h261VideoCapability
2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
capability: videoData, type h261VideoCapability
2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
capability: videoData, type h261VideoCapability
2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
capability: videoData, type h261VideoCapability
2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
capability: videoData, type h261VideoCapability
2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
capability: videoData, type h261VideoCapability
2010-12-13 11:58:21.820187 [DEBUG] call.cxx:288 GetOtherPartyConnection
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.820187 [DEBUG] call.cxx:169 OnSetUp
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
2010-12-13 11:58:21.820187 [DEBUG] localep.cxx:240 Incoming call from
gustavo [200.117.192.17]
2010-12-13 11:58:21.820187 [DEBUG] mod_opal.cpp:641 Created switch
caller profile:
  username       = 
  dialplan       = XML
  caller_id_name     = gustavo [200.117.192.17]
  caller_id_number   = 0000000000
  network_addr   = 
  source         = opal
  context       
Log-Func: 
Log-Line: 0
User-Data: 

 = default
  2010-12-13 11:58:21.823196 [NOTICE] switch_channel.c:784 New Channel
opal/in:231543514280633 [7b4a89ac-b5d7-4fb4-bab5-ddaa8e73485c]
2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:647
(opal/in:231543514280633) State Change CS_NEW -> CS_INIT
2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:1092 State changed on
connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.823196 [DEBUG] switch_core_session.c:1083 Send
signal opal/in:231543514280633 [BREAK]
2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:931 Kill 3 on connection
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.823196 [DEBUG] connection.cxx:1516 SetPhase from
UninitialisedPhase to AlertingPhase for
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.823196 [NOTICE] mod_opal.cpp:676 Ring-Ready
opal/in:231543514280633!
2010-12-13 11:58:21.823196 [DEBUG] manager.cxx:678 OnAlerting
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.823196 [DEBUG] call.cxx:196 OnAlerting
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.823196 [DEBUG] h323.cxx:2067 SetAlerting
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:320
(opal/in:231543514280633) Running State Change CS_INIT
2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:356
(opal/in:231543514280633) State INIT
2010-12-13 11:58:21.823196 [DEBUG] osutil.cxx:189 File handle high water
mark set: 72 Thread unblock pipe
2010-12-13 11:58:21.823196 [DEBUG] tlibthrd.cxx:587 Thread high water
mark set: 10
2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:842 Started routing for
connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:843
(opal/in:231543514280633) State Change CS_INIT -> CS_ROUTING
2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:1092 State changed on
connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.823196 [DEBUG] switch_core_session.c:1083 Send
signal opal/in:231543514280633 [BREAK]
2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:931 Kill 3 on connection
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:356
(opal/in:231543514280633) State INIT going to sleep
2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:320
(opal/in:231543514280633) Running State Change CS_ROUTING
2010-12-13 11:58:21.823196 [DEBUG] switch_channel.c:1615
(opal/in:231543514280633) Callstate Change DOWN -> RINGING
2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:359
(opal/in:231543514280633) State ROUTING
2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:850 Routing connection
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:77
opal/in:231543514280633 Standard ROUTING
2010-12-13 11:58:21.823196 [INFO] mod_dialplan_xml.c:331 Processing
gustavo [200.117.192.17] <0000000000>->231543514280633 in context
default
2010-12-13 11:58:21.826206 [DEBUG] h323pdu.cxx:80 Receiving PDU:
  request terminalCapabilitySet {
    sequenceNumber = 1
    protocolIdentifier = 0.0.8.245.0.13
    multiplexCapability = h2250Capability {
      maximumAudioDelayJitter = 500
      receiveMultipointCapabi
lity = {
        multicastCapability = false
        multiUniCastConference = false
        mediaDistributionCapability = 1 entries {
          [0]={
            centralizedControl = false
            distributedControl = false
            centraliz
edAudio = false
            distributedAudio = false
            centralizedVideo = false
            distributedVideo = false
          }
        }
      }
      transmitMultipointCapability = {
        multicastCapability = false
        multiUniC
astConference = false
        mediaDistributionCapability = 1 entries {
          [0]={
            centralizedControl = false
            distributedControl = false
            centralizedAudio = false
            distributedAudio = false
         
   centralizedVideo = false
            distributedVideo = false
          }
        }
      }
      receiveAndTransmitMultipointCapability = {
        multicastCapability = false
        multiUniCastConference = false
        mediaDistributionCapab
ility = 1 entries {
          [0]={
            centralizedControl = false
            distributedControl = false
            centralizedAudio = false
            distributedAudio = false
            centralizedVideo = false
            distributedV
ideo = false
          }
        }
      }
      mcCapability = {
        centralizedConferenceMC = false
        decentralizedConferenceMC = false
      }
      rtcpVideoControlCapability = false
      mediaPacketizationCapability = {
        h261a
VideoPacketization = false
      }
      logicalChannelSwitchingCapability = false
      t120DynamicPortCapability = true
    }
    capabilityTable = 8 entries {
      [0]={
        capabilityTableEntryNumber = 1
        capability = receiveAudioCap
ability g711Alaw64k 240
      }
      [1]={
        capabilityTableEntryNumber = 2
        capability = receiveVideoCapability h261VideoCapability {
          qcifMPI = 1
          cifMPI = 1
          temporalSpatialTradeOffCapability = false
     
     maxBitRate = 19200
          stillImageTransmission = false
          videoBadMBsCap = false
        }
      }
      [2]={
        capabilityTableEntryNumber = 3
        capability = receiveVideoCapability h261VideoCapability {
          cifMPI
 = 1
          temporalSpatialTradeOffCapability = false
          maxBitRate = 19200
          stillImageTransmission = false
          videoBadMBsCap = false
        }
      }
      [3]={
        capabilityTableEntryNumber = 4
        capability =
 receiveVideoCapability h261VideoCapability {
          qcifMPI = 1
          temporalSpatialTradeOffCapability = false
          maxBitRate = 19200
          stillImageTransmission = false
          videoBadMBsCap = false
        }
      }
      [4
]={
        capabilityTableEntryNumber = 5
        capability = receiveUserInputCapability hookflash <<null>>
      }
      [5]={
        capabilityTableEntryNumber = 6
        capability = receiveUserInputCapability basicString <<null>>
      }
   
   [6]={
        capabilityTableEntryNumber = 7
        capability = receiveUserInputCapability dtmf <<null>>
      }
      [7]={
        capabilityTableEntryNumber = 8
        capability = receiveRTPAudioTelephonyEventCapability {
          dynamic
RTPPayloadType = 101
          audioTelephoneEvent = "0-16"
        }
      }
    }
    capabilityDescriptors = 1 entries {
      [0]={
        capabilityDescriptorNumber = 1
        simultaneousCapabilities = 4 entries {
          [0]=1 entries {
 
           [0]=1
          }
          [1]=3 entries {
            [0]=2
            [1]=3
            [2]=4
          }
          [2]=1 entries {
            [0]=5
          }
          [3]=3 entries {
            [0]=6
            [1]=7
          
Log-Func: 
Log-Line: 0
User-Data: 

  [2]=8
    2010-12-13 11:58:21.826206 [DEBUG] h323.cxx:2792 Set protocol
version to 13
2010-12-13 11:58:21.826206 [DEBUG] h323neg.cxx:378 Received
TerminalCapabilitySet: state=Idle pduSeq=1 inSeq=4294967295
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:1988
H323Capabilities(ctor)
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
G.723.1 <1>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
G.729 <2>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
G.711-uLaw-64k <3>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
G.711-ALaw-64k <4>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
UserInput/hookflash <5>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
UserInput/basicString <6>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
UserInput/dtmf <7>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
UserInput/RFC2833 <8>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
G.723.1 <1>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
G.729 <2>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
G.711-uLaw-64k <3>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
G.711-ALaw-64k <4>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
UserInput/hookflash <5>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
UserInput/basicString <6>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
UserInput/dtmf <7>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
UserInput/RFC2833 <8>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2376 Found capability:
G.711-ALaw-64k <4>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2376 Found capability:
G.711-uLaw-64k <3>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2376 Found capability:
G.723.1 <1>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
capability: "G.723.1(5.3k)"
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
G.723.1 <9>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
capability: "G.723.1A(5.3k)"
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
G.723.1 <10>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
capability: "G.723.1A(6.3k)"
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
G.723.1 <11>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
capability: "G.728"
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
G.728 <12>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2376 Found capability:
G.729 <2>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
capability: "G.729A"
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
G.729A <13>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
capability: "G.729A/B"
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
G.729A/B <14>
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
capability: "G.729B"
2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
G.729B <15>
2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2381 Could not find
capability: "GSM-06.10"
2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2284 Added capability:
GSM-06.10 <16>
2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2381 Could not find
capability: "GSM-AMR"
2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2284 Added capability:
GSM-AMR <17>
2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2381 Could not find
capability: "T.38"
2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2284 Added capability:
T.38 <18>
2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2376 Found capability:
UserInput/RFC2833 <8>
2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2376 Found capability:
UserInput/basicString <6>
2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2376 Found capability:
UserInput/dtmf <7>
2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2381 Could not find
capability: "UserInput/generalString"
2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
UserInput/generalString <19>
2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2376 Found capability:
UserInput/hookflash <5>
2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2381 Could not find
capability: "UserInput/iA5String"
2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
UserInput/iA5String <20>
2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2381 Could not find
capability: "iLBC"
2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
iLBC <21>
2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
UserInput/hookflash <22>
2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
UserInput/basicString <23>
2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
UserInput/dtmf <24>
2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
UserInput/RFC2833 <25>
2010-12-13 11:58:21.834192 [CONSOLE] mod_xml_curl.c:312 XML response is
in /tmp/90f1dd7d-a63c-4e15-981c-6eed9e70313b.tmp.xml
Dialplan: opal/in:231543514280633 parsing [default->external]
continue=false
Dialplan: opal/in:231543514280633 Regex (PASS) [external]
destination_number(231543514280633) =~ /^231543514280633/ break=on-false
Dialplan: opal/in:231543514280633 Action set(continue_on_fail=true) 
Dialplan: opal/in:231543514280633 Action set(hangup_after_bridge=true) 
Dialplan: opal/in:231543514280633 Action set(progress_timeout=15) 
Dialplan: opal/in:231543514280633 Action set(proxy_media=false) 
Dialplan: opal/in:231543514280633 Action set(bypass_media=true) 
Dialplan: opal/in:231543514280633 Action
set(absolute_codec_string=PCMA) 
Dialplan: opal/in:231543514280633 Action
bridge(sofia/external/21543514280633 at 200.35.145.149) 
2010-12-13 11:58:21.834192 [DEBUG] switch_core_state_machine.c:119
(opal/in:231543514280633) State Change CS_ROUTING -> CS_EXECUTE
2010-12-13 11:58:21.834192 [DEBUG] mod_opal.cpp:1092 State changed on
connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.834192 [DEBUG] switch_core_session.c:1083 Send
signal opal/in:231543514280633 [BREAK]
2010-12-13 11:58:21.834192 [DEBUG] mod_opal.cpp:931 Kill 3 on connection
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.834192 [DEBUG] switch_core_state_machine.c:359
(opal/in:231543514280633) State ROUTING going to sleep
2010-12-13 11:58:21.834192 [DEBUG] switch_core_state_machine.c:320
(opal/in:231543514280633) Running State Change CS_EXECUTE
2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
H.239-Video+H.239-Video <26>
2010-12-13 11:58:21.834192 [DEBUG] switch_core_state_machine.c:366
(opal/in:231543514280633) State EXECUTE
2010-12-13 11:58:21.834192 [DEBUG] mod_opal.cpp:857 Executing connection
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.834192 [DEBUG] switch_core_state_machine.c:157
opal/in:231543514280633 Standard EXECUTE
EXECUTE opal/in:231543514280633 set(continue_on_fail=true)
2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2284 Added capability:
H.239-Control <27>
2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2029 Parsing remote
capabilities
2010-12-13 11:58:21.837191 [DEBUG] mod_opal.cpp:1076 Received message 26
on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.837191 [DEBUG] mod_dptools.c:1028
opal/in:231543514280633 SET [continue_on_fail]=[true]
2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:819 Capability tx frames
left at 20 as remote allows 240
2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2529 Could not find
capability: receiveVideoCapability, type h261VideoCapability
2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2529 Could not find
capability: receiveVideoCapability, type h261VideoCapability
2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2529 Could not find
capability: receiveVideoCapability, type h261VideoCapability
2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2671 Found capability:
UserInput/RFC2833 <8>
2010-12-13 11:58:21.837191 [DEBUG] mod_opal.cpp:1076 Received message 27
on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
EXECUTE opal/in:231543514280633 set(hangup_after_bridge=true)
2010-12-13 11:58:21.837191 [DEBUG] mod_opal.cpp:1076 Received message 26
on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.837191 [DEBUG] mod_dptools.c:1028
opal/in:231543514280633 SET [hangup_after_bridge]=[true]
2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2411 Could not find
capability: G.711-ALaw-64k <1>
2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2294 Added capability:
G.711-ALaw-64k <1>
2010-12-13 11:58:21.837191 [DEBUG] mod_opal.cpp:1076 Received message 27
on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
EXECUTE opal/in:231543514280633 set(progress_timeout=15)
2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2411 Could not find
capability: UserInput/hookflash <5>
2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2294 Added capability:
UserInput/hookflash <5>
2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2411 Could not find
capability: UserInput/basicString <6>
2010-12-13 11:58:21.837191 [DEBUG] mod_opal.cpp:1076 Received message 26
on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.837191 [DEBUG] mod_dptools.c:1028
opal/in:231543514280633 SET [progress_timeout]=[15]
2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2294 Added capability:
UserInput/basicString <6>
2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2411 Could not find
capability: UserInput/dtmf <7>
2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2294 Added capability:
UserInput/dtmf <7>
2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2411 Could not find
capability: UserInput/RFC2833 <8>
2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2294 Added capability:
UserInput/RFC2833 <8>
2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2355 Found capability:
G.711-ALaw-64k <1>
2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2355 Found capability:
UserInput/hookflash <5>
2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2355 Found capability:
UserInput/basicString <6>
2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2355 Found capability:
UserInput/dtmf <7>
2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2355 Found capability:
UserInput/RFC2833 <8>
2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received message 27
on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
EXECUTE opal/in:231543514280633 set(proxy_media=false)
2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2777 Capability merge
result:
     Table:
       G.711-ALaw-64k <1>
       UserInput/hookflash <5>
       UserInput/basicString <6>
       UserInput/dtmf <7>
       UserInput/RFC2833 <8>
     Set:
       0:
         0:
     

Log-Func: 
Log-Line: 0
User-Data: 

      G.711-ALaw-64k <1>
         1:
         2:
           UserInput/hookflash <5>
         3:
           UserInput/basicString <6>
           UserInput/dtmf <INCOMING DATA [(null)]

RECV EVENT
Event-Name: SOCKET_DATA


2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2778 Received capability
set, is accepted
2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received message 26
on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.840189 [DEBUG] mod_dptools.c:1028
opal/in:231543514280633 SET [proxy_media]=[false]
2010-12-13 11:58:21.840189 [DEBUG] h323neg.cxx:341 Sending
TerminalCapabilitySet: outSeq=1
2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received message 27
on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
EXECUTE opal/in:231543514280633 set(bypass_media=true)
2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received message 26
on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.840189 [DEBUG] mod_dptools.c:1028
opal/in:231543514280633 SET [bypass_media]=[true]
2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received message 27
on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
EXECUTE opal/in:231543514280633 set(absolute_codec_string=PCMA)
2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received message 26
on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.840189 [DEBUG] mod_dptools.c:1028
opal/in:231543514280633 SET [absolute_codec_string]=[PCMA]
2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received message 27
on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
EXECUTE opal/in:231543514280633
bridge(sofia/external/21543514280633 at 200.35.145.149)
2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received message 26
on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.840189 [NOTICE] switch_channel.c:784 New Channel
sofia/external/21543514280633 at 200.35.145.149
[f9bf5b9e-a273-400c-b18c-e2c98c84750f]
2010-12-13 11:58:21.840189 [DEBUG] mod_sofia.c:3995
(sofia/external/21543514280633 at 200.35.145.149) State Change CS_NEW ->
CS_INIT
2010-12-13 11:58:21.840189 [DEBUG] switch_core_session.c:1083 Send
signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
2010-12-13 11:58:21.843189 [DEBUG] h323pdu.cxx:80 Sending PDU:
  request terminalCapabilitySet {
    sequenceNumber = 1
    protocolIdentifier = 0.0.8.245.0.13
    multiplexCapability = h2250Capability {
      maximumAudioDelayJitter = 250
      receiveMultipointCapabili
ty = {
        multicastCapability = false
        multiUniCastConference = false
        mediaDistributionCapability = 1 entries {
          [0]={
            centralizedControl = false
            distributedControl = false
            centralized
Audio = false
            distributedAudio = false
            centralizedVideo = false
            distributedVideo = false
          }
        }
      }
      transmitMultipointCapability = {
        multicastCapability = false
        multiUniCas
tConference = false
        mediaDistributionCapability = 1 entries {
          [0]={
            centralizedControl = false
            distributedControl = false
            centralizedAudio = false
            distributedAudio = false
           
 centralizedVideo = false
            distributedVideo = false
          }
        }
      }
      receiveAndTransmitMultipointCapability = {
        multicastCapability = false
        multiUniCastConference = false
        mediaDistributionCapabil
ity = 1 entries {
          [0]={
            centralizedControl = false
            distributedControl = false
            centralizedAudio = false
            distributedAudio = false
            centralizedVideo = false
            distributedVid
eo = false
          }
        }
      }
      mcCapability = {
        centralizedConferenceMC = false
        decentralizedConferenceMC = false
      }
      rtcpVideoControlCapability = false
      mediaPacketizationCapability = {
        h261aVi
deoPacketization = false
      }
      logicalChannelSwitchingCapability = false
      t120DynamicPortCapability = true
    }
    capabilityTable = 8 entries {
      [0]={
        capabilityTableEntryNumber = 1
        capability = receiveAudioCapab
ility g7231 {
          maxAl_sduAudioFrames = 8
          silenceSuppression = false
        }
      }
      [1]={
        capabilityTableEntryNumber = 2
        capability = receiveAudioCapability g729 24
      }
      [2]={
        capabilityTabl
eEntryNumber = 3
        capability = receiveAudioCapability g711Ulaw64k 240
      }
      [3]={
        capabilityTableEntryNumber = 4
        capability = receiveAudioCapability g711Alaw64k 240
      }
      [4]={
        capabilityTableEntryNumbe
r = 5
        capability = receiveUserInputCapability hookflash <<null>>
      }
      [5]={
        capabilityTableEntryNumber = 6
        capability = receiveUserInputCapability basicString <<null>>
      }
      [6]={
        capabilityTableEntry
Number = 7
        capability = receiveUserInputCapability dtmf <<null>>
      }
      [7]={
        capabilityTableEntryNumber = 8
        capability = receiveRTPAudioTelephonyEventCapability {
          dynamicRTPPayloadType = 101
          audioT
elephoneEvent = "0-16"
        }
      }
    }
    capabilityDescriptors = 1 entries {
      [0]={
        capabilityDescriptorNumber = 1
        simultaneousCapabilities = 3 entries {
          [0]=4 entries {
            [0]=1
            [1]=2
  
          [2]=3
            [3]=4
          }
          [1]=1 entries {
            [0]=5
          }
          [2]=3 entries {
            [0]=6
            [1]=7
            [2]=8
          }
        }
      }
    }
  }
2010-12-13 11:58:21.843189 [DEBUG] h323caps.cxx:2376 Found capability:
UserInput/RFC2833 <8>
2010-12-13 11:58:21.843189 [DEBUG] h323.cxx:3535 User Input RFC2833
payload type set to [pt=101]
2010-12-13 11:58:21.843189 [DEBUG] h323pdu.cxx:80 Sending PDU:
  response terminalCapabilitySetAck {
    sequenceNumber = 1
  }
2010-12-13 11:58:21.843189 [DEBUG] switch_core_state_machine.c:320
(sofia/external/21543514280633 at 200.35.145.149) Running State Change
CS_INIT
2010-12-13 11:58:21.843189 [DEBUG] switch_core_state_machine.c:356
(sofia/external/21543514280633 at 200.35.145.149) State INIT
2010-12-13 11:58:21.843189 [DEBUG] mod_sofia.c:86
sofia/external/21543514280633 at 200.35.145.149 SOFIA INIT
2010-12-13 11:58:21.843189 [DEBUG] h323.cxx:3771
InternalEstablishedConnectionCheck: connectionState=AwaitingLocalAnswer
fastStartState=FastStartResponse H.245 is unavailable
2010-12-13 11:58:21.843189 [DEBUG] h323pdu.cxx:80 Receiving PDU:
  request masterSlaveDetermination {
    terminalType = 50
    statusDeterminationNumber = 6111531
  }
2010-12-13 11:58:21.843189 [DEBUG] h323neg.cxx:129 Received
MasterSlaveDetermination: state=Idle
2010-12-13 11:58:21.843189 [DEBUG] h323neg.cxx:160
MasterSlaveDetermination: local is master
2010-12-13 11:58:21.843189 [DEBUG] h323pdu.cxx:80 Sending PDU:
  response masterSlaveDeterminationAck {
    decision = slave <<null>>
  }
2010-12-13 11:58:21.843189 [DEBUG] h323.cxx:3771
InternalEstablishedConnectionCheck: connectionState=AwaitingLocalAnswer
fastStartState=FastStartResponse H.245 is ready
2010-12-13 11:58:21.843189 [DEBUG] mod_sofia.c:126
(sofia/external/21543514280633 at 200.35.145.149) State Change CS_INIT ->
CS_ROUTING
2010-12-13 11:58:21.843189 [DEBUG] switch_core_session.c:1083 Send
signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
2010-12-13 11:58:21.843189 [DEBUG] sofia.c:4604 Channel
sofia/external/21543514280633 at 200.35.145.149 entering state
[terminated][900]
2010-12-13 11:58:21.843189 [DEBUG] h323ep.cxx:1033 OnSendAlerting conn =
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
2010-12-13 11:58:21.843189 [DEBUG] switch_channel.c:2455
(sofia/external/21543514280633 at 200.35.145.149) Callstate Change DOWN ->
HANGUP
2010-12-13 11:58:21.843189 [DEBUG] h323.cxx:2110 SetAlerting sending
Alerting PDU
2010-12-13 11:58:21.843189 [NOTICE] sofia.c:5244 Hangup
sofia/external/21543514280633 at 200.35.145.149 [CS_ROUTING]
[NORMAL_UNSPECIFIED]
2010-12-13 11:58:21.843189 [DEBUG] switch_channel.c:2471 Send signal
sofia/external/21543514280633 at 200.35.145.149 [KILL]
2010-12-13 11:58:21.843189 [DEBUG] switch_core_session.c:1083 Send
signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
2010-12-13 11:58:21.843189 [DEBUG] switch_core_state_machine.c:356
(sofia/external/21543514280633 at 200.35.145.149) State INIT going to sleep
2010-12-13 11:58:21.843189 [DEBUG] switch_core_state_machine.c:320
(sofia/external/21543514280633 at 200.35.145.149) Running State Change
CS_HANGUP
2010-12-13 11:58:21.846211 [DEBUG] h323pdu.cxx:80 Sending PDU:
  {
    q931pdu = {
      protocolDiscriminator = 8
      callReference = 22716
      from = destination
      messageType = Alerting
      IE: Display = {
        72 6f 6f 74 00                              
       root.
      }
      IE: User-User = {
        23 80 06 00 08 91 4a 00  06 22 c0 09 00 00 3d 1e
#.....J.."....=.
        56 6f 78 20 4c 75 63 69  64 61 20 50 74 79 2e 20   Vox Lucida
Pty. 
        ...
      }
    }
    h225pdu = {
      h323
_uu_pdu = {
        h323_message_body = alerting {
          protocolIdentifier = 0.0.8.2250.0.6
          destinationInfo = {
            vendor = {
              vendor = {
                t35CountryCode = 9
                t35Extension = 0
      
          manufacturerCode = 61
              }
              productId =  31 octets {
                56 6f 78 20 4c 75 63 69  64 61 20 50 74 79 2e 20   Vox
Lucida Pty. 
                4c 74 64 2e 20 6d 6f 64  5f 6f 70 61 6c 00 00      Ltd.
mod_op
Log-Line: 0
User-Data: 

   }
            terminal = {
            }
            mc = false
            undefinedNode = false
          }
          callIdentifier = {
            guid =  16 octets {
              3a 76 b2 f2 1d 05 e0 11  9b b7 00 1e 64 1f 33
fe   :v........
..d.3.
            }
          }
          multipleCalls = false
          maintainConnection = false
        }
        h245Tunneling = true
        h245Control = 3 entries {
          [0]= 108 octets {
            02 70 01 06 00 08 81 75  00 0d 80 
13 80 00 fa 00   .p.....u........
            01 00 00 01 00 00 01 00  00 0c c0 01 00 01 80
07   ................
            ...
          }
          [1]= 3 octets {
            21 80 01                                           !..
          }
  
        [2]= 2 octets {
            20 a0                                               .
          }
        }
      }
    }
  }
2010-12-13 11:58:21.846211 [DEBUG] h323ep.cxx:1039 OnSentAlerting conn =
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
2010-12-13 11:58:21.846211 [DEBUG] h323.cxx:3771
InternalEstablishedConnectionCheck: connectionState=AwaitingLocalAnswer
fastStartState=FastStartResponse H.245 is ready
2010-12-13 11:58:21.846211 [DEBUG] connection.cxx:361
OnSetUpConnectionCall[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.846211 [DEBUG] endpoint.cxx:408 OnSetUpConnection
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.846211 [DEBUG] h323.cxx:1701 OnAnswerCall
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716], caller =
gustavo [200.117.192.17]
2010-12-13 11:58:21.846211 [DEBUG] call.cxx:219 OnAnswerCall
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716] caller
"gustavo [200.117.192.17]"
2010-12-13 11:58:21.846211 [DEBUG] h323.cxx:1713 Answering call:
AnswerCallDeferred
2010-12-13 11:58:21.846211 [DEBUG] connection.cxx:541 Answering call:
AnswerCallDeferred
2010-12-13 11:58:21.846211 [DEBUG] h323.cxx:3771
InternalEstablishedConnectionCheck: connectionState=AwaitingLocalAnswer
fastStartState=FastStartResponse H.245 is ready
2010-12-13 11:58:21.846211 [DEBUG] h323.cxx:506 Reading PDUs:
callRef=22716
2010-12-13 11:58:21.846211 [DEBUG] switch_ivr_originate.c:3448 Originate
Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED]
2010-12-13 11:58:21.849192 [INFO] mod_dptools.c:2579 Originate Failed.
Cause: NORMAL_UNSPECIFIED
2010-12-13 11:58:21.849192 [DEBUG] mod_opal.cpp:1076 Received message 27
on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.849192 [NOTICE] switch_core_state_machine.c:189
opal/in:231543514280633 has executed the last dialplan instruction,
hanging up.
2010-12-13 11:58:21.849192 [DEBUG] switch_channel.c:2455
(opal/in:231543514280633) Callstate Change RINGING -> HANGUP
2010-12-13 11:58:21.849192 [NOTICE] switch_core_state_machine.c:191
Hangup opal/in:231543514280633 [CS_EXECUTE] [NORMAL_CLEARING]
2010-12-13 11:58:21.849192 [DEBUG] switch_channel.c:2471 Send signal
opal/in:231543514280633 [KILL]
2010-12-13 11:58:21.849192 [DEBUG] mod_opal.cpp:931 Kill 1 on connection
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.849192 [DEBUG] mod_opal.cpp:1092 State changed on
connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.849192 [DEBUG] switch_core_session.c:1083 Send
signal opal/in:231543514280633 [BREAK]
2010-12-13 11:58:21.849192 [DEBUG] mod_opal.cpp:931 Kill 3 on connection
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:366
(opal/in:231543514280633) State EXECUTE going to sleep
2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:320
(opal/in:231543514280633) Running State Change CS_HANGUP
2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:557
(sofia/external/21543514280633 at 200.35.145.149) State HANGUP
2010-12-13 11:58:21.849192 [DEBUG] mod_sofia.c:453
sofia/external/21543514280633 at 200.35.145.149 Overriding SIP cause 480
with 900 from the other leg
2010-12-13 11:58:21.849192 [DEBUG] mod_sofia.c:459 Channel
sofia/external/21543514280633 at 200.35.145.149 hanging up, cause:
NORMAL_UNSPECIFIED
2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:46
sofia/external/21543514280633 at 200.35.145.149 Standard HANGUP, cause:
NORMAL_UNSPECIFIED
2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:557
(sofia/external/21543514280633 at 200.35.145.149) State HANGUP going to
sleep
2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:351
(sofia/external/21543514280633 at 200.35.145.149) State Change CS_HANGUP ->
CS_REPORTING
2010-12-13 11:58:21.849192 [DEBUG] switch_core_session.c:1083 Send
signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:320
(sofia/external/21543514280633 at 200.35.145.149) Running State Change
CS_REPORTING
2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:617
(sofia/external/21543514280633 at 200.35.145.149) State REPORTING
2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:557
(opal/in:231543514280633) State HANGUP
2010-12-13 11:58:21.849192 [DEBUG] connection.cxx:401 Call end reason
for Call[Cc70e3cf91]-EP<local>[Ldc8d6c462] set to EndedByRemoteUser
2010-12-13 11:58:21.849192 [DEBUG] call.cxx:112 Clearing
Call[Cc70e3cf91] reason=EndedByRemoteUser
2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:1516 SetPhase from
SetUpPhase to ReleasingPhase for Call[Cc70e3cf91]-EP<h323>[tcp
$200.117.192.17:19270/22716]
2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:459 Releasing
asynchronously Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
2010-12-13 11:58:21.852189 [DEBUG] osutil.cxx:196 File handle low water
mark set: 69 Thread unblock pipe
2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:1516 SetPhase from
AlertingPhase to ReleasingPhase for
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:448 Releasing
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:489 OnReleased
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:808 Media streams
closed.
2010-12-13 11:58:21.852189 [DEBUG] endpoint.cxx:450 OnReleased
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.852189 [DEBUG] manager.cxx:709 OnReleased
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.852189 [DEBUG] call.cxx:716 OnReleased
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:436 Already released
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
2010-12-13 11:58:21.856029 [DEBUG] connection.cxx:1516 SetPhase from
ReleasingPhase to ReleasedPhase for
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:46
opal/in:231543514280633 Standard HANGUP, cause: NORMAL_CLEARING
2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:557
(opal/in:231543514280633) State HANGUP going to sleep
2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:351
(opal/in:231543514280633) State Change CS_HANGUP -> CS_REPORTING
2010-12-13 11:58:21.856029 [DEBUG] switch_core_session.c:1083 Send
signal opal/in:231543514280633 [BREAK]
2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:320
(opal/in:231543514280633) Running State Change CS_REPORTING
2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:617
(opal/in:231543514280633) State REPORTING
2010-12-13 11:58:21.856029 [DEBUG] connection.cxx:401 Call end reason
for Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716] set to
EndedByRemoteUser
2010-12-13 11:58:21.856029 [DEBUG] h323.cxx:349 OnReleased: tcp
$200.117.192.17:19270/22716, connectionState=AwaitingLocalAnswer
2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:53
sofia/external/21543514280633 at 200.35.145.149 Standard REPORTING, cause:
NORMAL_UNSPECIFIED
2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:617
(sofia/external/21543514280633 at 200.35.145.149) State REPORTING going to
sleep
2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:345
(sofia/external/21543514280633 at 200.35.145.149) State Change CS_REPORTING
-> CS_DESTROY
2010-12-13 11:58:21.856029 [DEBUG] switch_core_session.c:1083 Send
signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
2010-12-13 11:58:21.856029 [DEBUG] switch_core_session.c:1250 Session 2
(sofia/external/21543514280633 at 200.35.145.149) Locked, Waiting on
external entities
2010-12-13 11:58:21.856029 [NOTICE] switch_core_session.c:1268 Session 2
(sofia/external/21543514280633 at 200.35.145.149) Ended
2010-12-13 11:58:21.856029 [NOTICE] switch_core_session.c:1270 Close
Channel sofia/external/21543514280633 at 200.35.145.149 [CS_DESTROY]
2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:449
(sofia/external/21543514280633 at 200.35.145.149) Callstate Change HANGUP
-> DOWN
2010-12-13 11:58:21.858228 [DEBUG] switch_core_state_machine.c:452
(sofia/external/21543514280633 at 200.35.145.149) Running State Change
CS_DESTROY
2010-12-13 11:58:21.858228 [DEBUG] switch_core_state_machine.c:462
(sofia/external/21543514280633 at 200.35.145.149) State DESTROY
2010-12-13 11:58:21.858228 [DEBUG] mod_sofia.c:364
sofia/external/21543514280633 at 200.35.145.149 SOFIA DESTROY
2010-12-13 11:58:21.858228 [DEBUG] switch_core_state_machine.c:60
sofia/external/21543514280633 at 200.35.145.149 Standard DESTROY
2010-12-13 11:58:21.858228 [DEBUG] switch_core_state_machine.c:462
(sofia/external/21543514280633 at 200.35.145.149) State DESTROY going to
sleep
2010-12-13 11:58:21.856029 [DEBUG] h323.cxx:356 Sending release complete
PDU: callRef=22716
2010-12-13 11:58:21.858228 [DEBUG] h323pdu.cxx:80 Sending PDU:
  command endSessionCommand disconnect <<null>>
2010-12-13 11:58:21.858228 [DEBUG] h323pdu.cxx:80 Sending PDU:
  {
    q931pdu = {
      protocolDiscriminator = 8
      callReference = 22716
      from = destination
      messageType = ReleaseComplete
      IE: Cause - Normal call clearing = {
        80 90           
                                   ..
      }
      IE: Display = {
        72 6f 6f 74 00                                     root.
      }
      IE: User-User = {
        25 80 06 00 08 91 4a 00  06 01 11 00 3a 76 b2 f2
%.....J.....:v..
        
Log-Func: 
Log-Line: 0
User-Data: 

1d 05 e0 11 9b b7 00 1e  64 1f 33 fe 04 c0 01 80   ........d.3.....
        ...
      }
    }
    h225pdu = {
      h323_uu_pdu = {
        h323_message_body = releaseComplete {
          protocolIdentifier = 0.0.8= {
            guid =  16 octets {
              3a 76 b2 f2 1d 05 e0 11  9b b7 00 1e 64 1f 33
fe   :v..........d.3.
            }
          }
        }
        h245Tunneling = true
        h245Control = 1 entries {
          [0]= 2 octets {
      
      4a 40                                              J@
          }
        }
      }
    }
  }
2010-12-13 11:58:21.861189 [DEBUG] h323.cxx:412 Awaiting end session
from remote for 9.997 seconds
2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:53
opal/in:231543514280633 Standard REPORTING, cause: NORMAL_CLEARING
2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:617
(opal/in:231543514280633) State REPORTING going to sleep
2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:345
(opal/in:231543514280633) State Change CS_REPORTING -> CS_DESTROY
2010-12-13 11:58:21.861189 [DEBUG] switch_core_session.c:1083 Send
signal opal/in:231543514280633 [BREAK]
2010-12-13 11:58:21.861189 [DEBUG] switch_core_session.c:1250 Session 1
(opal/in:231543514280633) Locked, Waiting on external entities
2010-12-13 11:58:21.861189 [NOTICE] switch_core_session.c:1268 Session 1
(opal/in:231543514280633) Ended
2010-12-13 11:58:21.861189 [NOTICE] switch_core_session.c:1270 Close
Channel opal/in:231543514280633 [CS_DESTROY]
2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:449
(opal/in:231543514280633) Callstate Change HANGUP -> DOWN
2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:452
(opal/in:231543514280633) Running State Change CS_DESTROY
2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:462
(opal/in:231543514280633) State DESTROY
2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:60
opal/in:231543514280633 Standard DESTROY
2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:462
(opal/in:231543514280633) State DESTROY going to sleep
2010-12-13 11:58:22.087167 [DEBUG] h323pdu.cxx:80 Receiving PDU:
  {
    q931pdu = {
      protocolDiscriminator = 8
      callReference = 22716
      from = originator
      messageType = Facility
      IE: Facility = {

      }
      IE: Display = {
        67 75 73 74
(..........!... 
        80                                                 .
      }
    }
    h225pdu = {
 
     h323_uu_pdu = {
        h323_message_body = empty <<null>>
        h245Tunneling = true
        h245Control = 2 entries {
          [0]= 3 octets {
            21 80 01                                           !..
          }
          [1]= 2 
octets {
            20 80                                               .
          }
        }
      }
    }
  }
2010-12-13 11:58:22.087167 [DEBUG] h323.cxx:559 Handling PDU: Facility
callRef=22716
2010-12-13 11:58:22.087167 [DEBUG] h323.cxx:2709 Checking for end
session on PDU: response terminalCapabilitySetAck
2010-12-13 11:58:22.087167 [DEBUG] h323.cxx:2709 Checking for end
session on PDU: response masterSlaveDeterminationAck
2010-12-13 11:58:22.087167 [DEBUG] connection.cxx:436 Already released
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
2010-12-13 11:58:22.087167 [DEBUG] h323.cxx:549 Signal channel closed.
2010-12-13 11:58:22.087167 [DEBUG] transports.cxx:900 Transport clean up
on termination
2010-12-13 11:58:22.087167 [DEBUG] transports.cxx:885 Transport Close
2010-12-13 11:58:22.090169 [DEBUG] osutil.cxx:196 File handle low water
mark set: 65 PTextFile
2010-12-13 11:58:22.090169 [DEBUG] tlibthrd.cxx:1020 Could not parse
thread stat file /proc/21145/task/21190/stat
2010-12-13 11:58:22.306147 [DEBUG] connection.cxx:489 OnReleased
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
2010-12-13 11:58:22.306147 [DEBUG] connection.cxx:808 Media streams
closed.
2010-12-13 11:58:22.306147 [DEBUG] endpoint.cxx:450 OnReleased
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
2010-12-13 11:58:22.306147 [DEBUG] manager.cxx:709 OnReleased
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
2010-12-13 11:58:22.306147 [DEBUG] call.cxx:716 OnReleased
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
2010-12-13 11:58:22.306147 [DEBUG] manager.cxx:535 OnClearedCall
Call[Cc70e3cf91] from "h323:200.117.192.17" to "h323:231543514280633"
2010-12-13 11:58:22.306147 [DEBUG] connection.cxx:1516 SetPhase from
ReleasingPhase to ReleasedPhase for Call[Cc70e3cf91]-EP<h323>[tcp
$200.117.192.17:19270/22716]
2010-12-13 11:58:22.306147 [DEBUG] connection.cxx:480 OnRelease thread
completed for Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
2010-12-13 11:58:22.308152 [DEBUG] tlibthrd.cxx:1020 Could not parse
thread stat file /proc/21145/task/21194/stat
2010-12-13 11:58:22.588122 [DEBUG] transports.cxx:900 Transport clean up
on termination
2010-12-13 11:58:22.588122 [DEBUG] transports.cxx:885 Transport Close
2010-12-13 11:58:22.588122 [DEBUG] transports.cxx:1045 Deleted transport
tcp$200.117.192.17:19270<if=tcp$72.51.47.100:1720>
2010-12-13 11:58:22.590122 [DEBUG] h323.cxx:330 Connection tcp
$200.117.192.17:19270/22716 deleted.
2010-12-13 11:58:22.590122 [DEBUG] connection.cxx:343 Connection
Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716] destroyed.
2010-12-13 11:58:22.590122 [DEBUG] localep.cxx:211 Deleted connection.
2010-12-13 11:58:22.590122 [DEBUG] connection.cxx:343 Connection
Call[Cc70e3cf91]-EP<local>[Ldc8d6c462] destroyed.
2010-12-13 11:58:23.592039 [DEBUG] call.cxx:86 Call[Cc70e3cf91]
destroyed.

---------------------------------------------------------------------------------------------------------------------------------------------------------------------
Gustavo Espeche
www.easyipcall.com








From kond at nstel.ru  Mon Dec 13 17:48:08 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Mon, 13 Dec 2010 17:48:08 +0300
Subject: [Freeswitch-users] opal to sip problem
In-Reply-To: <1292250252.2159.14.camel@gustavo-laptop>
Message-ID: <20101213144808.A674D114F0@mail.nstel.ru>

I can't help you with mod_opal, but I use FS with mod_h323 ad sip<->h.323
gateway successfully.
Rgds,
Nikolay.

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On 
> Behalf Of Gustavo Espeche
> Sent: Monday, December 13, 2010 5:24 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] opal to sip problem
> 
> Hello list, we are try to do inter-worker between h323 and sip using
> opal in freeswitch, but FS don't send the call to our sip gw, 
> follow is
> the  call flow:
> 
> h323 endpoint -->FS->sip gateway
> 
> attached a FS debug of call.
> 
> if something know or work with opal in freeSwitch and have 
> some tips we
> appreciate a lot if advice about it.
> Best Regards.
> --------------------------------------------------------------
> ---------------------------------------------------------------
> 2010-12-13 11:58:21.325242 [DEBUG] transports.cxx:1200 Started
> connection to 200.117.192.17:19270 (if=72.51.47.100:1720)
> 2010-12-13 11:58:21.325242 [DEBUG] osutil.cxx:189 File handle 
> high water
> mark set: 67 Thread unblock pipe
> 2010-12-13 11:58:21.325242 [DEBUG] tlibthrd.cxx:587 Thread high water
> mark set: 8
> 2010-12-13 11:58:21.325242 [DEBUG] transports.cxx:666 Waiting 
> on socket
> accept on tcp$72.51.47.100:1720
> 2010-12-13 11:58:21.325242 [DEBUG] h323ep.cxx:501 Awaiting first PDU
> 2010-12-13 11:58:21.805203 [DEBUG] h323pdu.cxx:80 Receiving PDU:
>   {
>     q931pdu = {
>       protocolDiscriminator = 8
>       callReference = 22716
>       from = originator
>       messageType = Setup
>       IE: Bearer-Capability = {
>         88 93 a5                            
>                ...
>       }
>       IE: Display = {
>         67 75 73 74 61 76 6f 00                            gustavo.
>       }
>       IE: Called-Party-Number = {
>         81 32 33 31 35 34 33 35  31 34 32 38 30 36 33
> 33   .231543514280633
>       }
>       
> IE: User-User = {
>         20 b8 06 00 08 91 4a 00  06 01 40 06 00 67 00
> 75    .....J... at ..g.u
>         00 73 00 74 00 61 00 76  00 6f 22 c0 09 00 00
> 3d   .s.t.a.v.o"....=
>         ...
>       }
>     }
>     h225pdu = {
>       h323_uu_pdu = {
>         h323_me
> ssage_body = setup {
>           protocolIdentifier = 0.0.8.2250.0.6
>           sourceAddress = 1 entries {
>             [0]=h323_ID  7 characters {
>               0067 0075 0073 0074 0061 0076 006f        gustavo
>             }
>           }
>           sour
> ceInfo = {
>             vendor = {
>               vendor = {
>                 t35CountryCode = 9
>                 t35Extension = 0
>                 manufacturerCode = 61
>               }
>               productId =  3 octets {
>                 00 00 00      
> (OPAL v3.6
>                 2e 36 29 00 00                                     .6
> )..
>               }
>             }
>             terminal = {
>             }
>             mc = false
>             undefinedNode = false
>           }
>           destinationAddress = 1 entries {
>             [0]=dialedDigits "231543514280633"
>           }
>      
> Log-Func: 
> Log-Line: 0
> User-Data: 
> 
>      destCallSignalAddress = ipAddress {
>             ip =  4 octets {
>               48 33 2f 64                                        H3/d
>             }
>             port = 1720
>           }
>           activeMC = falets {
>             4e 76 b2 f2 1d 05 e0 11  9b b7 00 1e 64 1f 33 fe
> Nv..........d.3.
>           }
>           conferenceGoal = create <<null>>
>           callType = pointToPoint <<null>>
>           sourceCallSignalAddress = ipAddress {
>             ip = 
> Log-Func: 
> Log-Line: 0
> User-Data: 
> 
>  4 octets {
>               c8 75 c0 11                                        .u..
>             }
>             port = 19270
>           }
>           callIdentifier = {
>             guid =  16 octets {
>               3a 76 f 33 fe   :v..........d.3.
>             }
>           }
>           fastStart = 8 entries {
>             [0]= 29 octets {
>               40 00 00 06 04 01 00 4c  20 13 80 11 1c 00 01 00
> @......L .......
>               c8 75 c0 11 13 de 00 c8  75 c0 11 13 
> df            .u......u....
>             }
>             [1]= 19 octets {
>               00 00 64 0c 20 13 80 0b  0d 00 01 00 c8 75 c0
> 11   ..d. ........u..
>               13 df 00                                           ...
>             }
>             [
> 2]= 35 octets {
>               40 00 00 06 04 01 00 48  78 00 4a ff 00 80 01 00
> @......Hx.J.....
>               80 11 1c 00 02 00 c8 75  c0 11 13 e0 00 c8 75
> c0   .......u......u.
>               ...
>             }
>             [3]= 25 octets {
>         
>       00 00 65 08 78 00 4a ff  00 80 01 00 80 0b 0d
> 00   ..e.x.J.........
>               02 00 c8 75 c0 11 13 e1
> 00                        ...u.....
>             }
>             [4]= 34 octets {
>               40 00 00 06 04 01 00 48  68 4a ff 00 80 01 
> 00 80   @......HhJ......
>               11 1c 00 02 00 c8 75 c0  11 13 e0 00 c8 75 c0
> 11   ......u......u..
>               ...
>             }
>             [5]= 24 octets {
>               00 00 66 08 68 4a ff 00  80 01 00 80 0b 0d 00
> 02   ..f.hJ..........
> 
>               00 c8 75 c0 11 13 e1
> 00                            ..u.....
>             }
>             [6]= 34 octets {
>               40 00 00 06 04 01 00 48  70 4a ff 00 80 01 00 80
> @......HpJ......
>               11 1c 00 02 00 c8 75 c0  11 13 e0 0
> 0 c8 75 c0 11   ......u......u..
>               ...
>             }
>             [7]= 24 octets {
>               00 00 67 08 70 4a ff 00  80 01 00 80 0b 0d 00
> 02   ..g.pJ..........
>               00 c8 75 c0 11 13 e1
> 00                            ..u.....
> 
>             }
>           }
>           mediaWaitForConnect = false
>           canOverlapSend = false
>           multipleCalls = false
>           maintainConnection = false
>           parallelH245Control = 2 entries {
>             [0]= 125 octets {
>         
>       02 70 01 06 00 08 81 75  00 0d 80 13 80 01 f4
> 00   .p.....u........
>               01 00 00 01 00 00 01 00  00 0c c0 01 00 01 80
> 07   ................
>               ...
>             }
>             [1]= 7 octets {
>               01 00 32 80 5d 41 2
> b                               ..2.]A+
>             }
>           }
>         }
>         h245Tunneling = true
>       }
>     }
>   }
> 2010-12-13 11:58:21.805203 [DEBUG] h323ep.cxx:510 Incoming call, first
> PDU: callReference=22716
> 2010-12-13 11:58:21.805203 [DEBUG] call.cxx:72 Created 
> Call[Cc70e3cf91]
> 2010-12-13 11:58:21.805203 [DEBUG] connection.cxx:262 Created 
> connection
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> 2010-12-13 11:58:21.805203 [DEBUG] rfc2833.cxx:140 Handler created
> 2010-12-13 11:58:21.805203 [DEBUG] rfc2833.cxx:328 Rx 
> capability set to
> "0-16,32,36"
> 2010-12-13 11:58:21.805203 [DEBUG] rfc2833.cxx:140 Handler created
> 2010-12-13 11:58:21.805203 [DEBUG] rfc2833.cxx:328 Rx 
> capability set to
> "192,193"
> 2010-12-13 11:58:21.805203 [DEBUG] osutil.cxx:189 File handle 
> high water
> mark set: 68 PUDPSocket
> 2010-12-13 11:58:21.808186 [DEBUG] h4601.cxx:1583 Endpoint Attached
> 2010-12-13 11:58:21.808186 [DEBUG] h323ep.cxx:552 Created new
> connection: tcp$200.117.192.17:19270/22716
> 2010-12-13 11:58:21.808186 [DEBUG] h323.cxx:559 Handling PDU: Setup
> callRef=22716
> 2010-12-13 11:58:21.808186 [DEBUG] connection.cxx:1516 SetPhase from
> UninitialisedPhase to SetUpPhase for Call[Cc70e3cf91]-EP<h323>[tcp
> $200.117.192.17:19270/22716]
> 2010-12-13 11:58:21.808186 [DEBUG] h323.cxx:943 Set protocol 
> version to
> 6 and implying H.245 version 13
> 2010-12-13 11:58:21.808186 [DEBUG] h323.cxx:1240 Set remote 
> application
> name: " 3.2.6 (OPAL v3.6.6) 9/61 "
> 2010-12-13 11:58:21.808186 [DEBUG] manager.cxx:1392 Checking incoming
> call for NAT: local=72.51.47.100, peer=200.117.192.17,
> sig=200.117.192.17
> 2010-12-13 11:58:21.808186 [DEBUG] h323.cxx:1041 Sending call 
> proceeding
> PDU
> 2010-12-13 11:58:21.811196 [DEBUG] h323pdu.cxx:80 Sending PDU:
>   {
>     q931pdu = {
>       protocolDiscriminator = 8
>       callReference = 22716
>       from = destination
>       messageType = CallProceeding
>       IE: Display = {
>         72 6f 6f 74 00                        
>              root.
>       }
>       IE: User-User = {
>         21 80 06 00 08 91 4a 00  06 22 c0 09 00 00 3d
> 1e   !.....J.."....=.
>         56 6f 78 20 4c 75 63 69  64 61 20 50 74 79 2e 20   Vox Lucida
> Pty. 
>         ...
>       }
>     }
>     h225pdu = {
>     
>   h323_uu_pdu = {
>         h323_message_body = callProceeding {
>           protocolIdentifier = 0.0.8.2250.0.6
>           destinationInfo = {
>             vendor = {
>               vendor = {
>                 t35CountryCode = 9
>                 t35Extensio
> n = 0
>                 manufacturerCode = 61
>               }
>               productId =  31 octets {
>                 56 6f 78 20 4c 75 63 69  64 61 20 50 74 79 2e 20   Vox
> Lucida Pty. 
>                 4c 74 64 2e 20 6d 6f 64  5f 6f 70 61 6c 00 00     
> Log-Line: 0
> UINCOMING DATA [(null)]
> 
> RECV EVENT
> Event-Name: SOCKET_DATA
> ser-Data: _undef_
> 
> 
>  }
>             }
>             terminal = {
>             }
>             mc = false
>             undefinedNode = false
>           }
>           callIdentifier = {
>             guid =  16 octets {
>               3a 76 b2 f2 1d 05 e0 11  9b b7 00 1e 64 1f 33 fe 
>   :v..........d.3.
>             }
>           }
>           multipleCalls = false
>           maintainConnection = false
>         }
>         h245Tunneling = true
>       }
>     }
>   }
> 2010-12-13 11:58:21.811196 [DEBUG] manager.cxx:600 OnIncoming 
> connection
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> 2010-12-13 11:58:21.811196 [DEBUG] call.cxx:288 
> GetOtherPartyConnection
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> 2010-12-13 11:58:21.811196 [DEBUG] manager.cxx:1200 Searching 
> for route
> "h323:root 231543514280633"
> 2010-12-13 11:58:21.811196 [DEBUG] manager.cxx:1228 Matched regex
> "h323:.*"
> 2010-12-13 11:58:21.811196 [DEBUG] manager.cxx:573 Set up 
> connection to
> "local:231543514280633"
> 2010-12-13 11:58:21.811196 [DEBUG] connection.cxx:262 Created 
> connection
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.811196 [DEBUG] localep.cxx:205 Created connection
> with token "Ldc8d6c462"
> 2010-12-13 11:58:21.811196 [DEBUG] h323.cxx:1075 Incoming 
> call accepted
> 2010-12-13 11:58:21.811196 [DEBUG] mod_opal.cpp:770 Matched FS codec
> PCMA to OPAL media format G.711-ALaw-64k
> 2010-12-13 11:58:21.811196 [DEBUG] mod_opal.cpp:770 Matched FS codec
> G729 to OPAL media format G.729
> 2010-12-13 11:58:21.811196 [DEBUG] mod_opal.cpp:770 Matched FS codec
> PCMU to OPAL media format G.711-uLaw-64k
> 2010-12-13 11:58:21.811196 [DEBUG] mod_opal.cpp:770 Matched FS codec
> G723 to OPAL media format G.723.1
> 2010-12-13 11:58:21.814188 [DEBUG] call.cxx:425 GetMediaFormats for
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> G.723.1
> G.729
> G.711-uLaw-64k
> G.711-ALaw-64k
> 
> 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2381 Could not find
> capability: "G.723.1"
> 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
> G.723.1 <1>
> 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2381 Could not find
> capability: "G.729"
> 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
> G.729 <2>
> 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2381 Could not find
> capability: "G.711-uLaw-64k"
> 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
> G.711-uLaw-64k <3>
> 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2381 Could not find
> capability: "G.711-ALaw-64k"
> 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
> G.711-ALaw-64k <4>
> 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2376 Found capability:
> G.723.1 <1>
> 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2376 Found capability:
> G.729 <2>
> 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2376 Found capability:
> G.711-uLaw-64k <3>
> 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2376 Found capability:
> G.711-ALaw-64k <4>
> 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
> UserInput/hookflash <5>
> 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
> UserInput/basicString <6>
> 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2284 Added capability:
> UserInput/dtmf <7>
> 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2284 Added capability:
> UserInput/RFC2833 <8>
> 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2376 Found capability:
> UserInput/RFC2833 <8>
> 2010-12-13 11:58:21.817197 [DEBUG] call.cxx:288 
> GetOtherPartyConnection
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> 2010-12-13 11:58:21.817197 [DEBUG] h323.cxx:3704 SetLocalCapabilities:
>  Table:
>    G.723.1 <1>
>    G.729 <2>
>    G.711-uLaw-64k <3>
>    G.711-ALaw-64k <4>
>    UserInput/hookflash <5>
>    UserInput/basicString <6>
>    UserInput/dtmf <7>
>    UserInput/RFC2833 <8>
>  Set:
>    0:
>   
> 
> Log-Func: 
> Log-Line: 0
> User-Data: 
> 
>    0:
>        G.723.1 <1>
>        G.729 <2>
>        G.711-uLaw-64k <3>
>        G.711-ALaw-64k <4>
>      1:
>        UserInput/hookflash <5>
>      2:
>        UserInput/basicString <6>
>        UserInput/dtINCOMING DATA [(null)]
> 
> RECV EVENT
> Event-Name: SOCKET_DATA
> 
> 
> 2010-12-13 11:58:21.817197 [DEBUG] h323.cxx:1121 Fast start detected
> 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2594 Could not find
> capability: audioData, type g711Alaw64k
> 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:845 
> Capability tx frames
> left at 20 as remote allows 20
> 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2294 Added capability:
> G.711-ALaw-64k <4>
> 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:845 
> Capability tx frames
> left at 20 as remote allows 20
> 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2594 Could not find
> capability: videoData, type h261VideoCapability
> 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2594 Could not find
> capability: videoData, type h261VideoCapability
> 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2594 Could not find
> capability: videoData, type h261VideoCapability
> 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2594 Could not find
> capability: videoData, type h261VideoCapability
> 2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
> capability: videoData, type h261VideoCapability
> 2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
> capability: videoData, type h261VideoCapability
> 2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
> capability: videoData, type h261VideoCapability
> 2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
> capability: videoData, type h261VideoCapability
> 2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
> capability: videoData, type h261VideoCapability
> 2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
> capability: videoData, type h261VideoCapability
> 2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
> capability: videoData, type h261VideoCapability
> 2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
> capability: videoData, type h261VideoCapability
> 2010-12-13 11:58:21.820187 [DEBUG] call.cxx:288 
> GetOtherPartyConnection
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.820187 [DEBUG] call.cxx:169 OnSetUp
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> 2010-12-13 11:58:21.820187 [DEBUG] localep.cxx:240 Incoming call from
> gustavo [200.117.192.17]
> 2010-12-13 11:58:21.820187 [DEBUG] mod_opal.cpp:641 Created switch
> caller profile:
>   username       = 
>   dialplan       = XML
>   caller_id_name     = gustavo [200.117.192.17]
>   caller_id_number   = 0000000000
>   network_addr   = 
>   source         = opal
>   context       
> Log-Func: 
> Log-Line: 0
> User-Data: 
> 
>  = default
>   2010-12-13 11:58:21.823196 [NOTICE] switch_channel.c:784 New Channel
> opal/in:231543514280633 [7b4a89ac-b5d7-4fb4-bab5-ddaa8e73485c]
> 2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:647
> (opal/in:231543514280633) State Change CS_NEW -> CS_INIT
> 2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:1092 State changed on
> connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.823196 [DEBUG] switch_core_session.c:1083 Send
> signal opal/in:231543514280633 [BREAK]
> 2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:931 Kill 3 on 
> connection
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.823196 [DEBUG] connection.cxx:1516 SetPhase from
> UninitialisedPhase to AlertingPhase for
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.823196 [NOTICE] mod_opal.cpp:676 Ring-Ready
> opal/in:231543514280633!
> 2010-12-13 11:58:21.823196 [DEBUG] manager.cxx:678 OnAlerting
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.823196 [DEBUG] call.cxx:196 OnAlerting
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.823196 [DEBUG] h323.cxx:2067 SetAlerting
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> 2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:320
> (opal/in:231543514280633) Running State Change CS_INIT
> 2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:356
> (opal/in:231543514280633) State INIT
> 2010-12-13 11:58:21.823196 [DEBUG] osutil.cxx:189 File handle 
> high water
> mark set: 72 Thread unblock pipe
> 2010-12-13 11:58:21.823196 [DEBUG] tlibthrd.cxx:587 Thread high water
> mark set: 10
> 2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:842 Started 
> routing for
> connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:843
> (opal/in:231543514280633) State Change CS_INIT -> CS_ROUTING
> 2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:1092 State changed on
> connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.823196 [DEBUG] switch_core_session.c:1083 Send
> signal opal/in:231543514280633 [BREAK]
> 2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:931 Kill 3 on 
> connection
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:356
> (opal/in:231543514280633) State INIT going to sleep
> 2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:320
> (opal/in:231543514280633) Running State Change CS_ROUTING
> 2010-12-13 11:58:21.823196 [DEBUG] switch_channel.c:1615
> (opal/in:231543514280633) Callstate Change DOWN -> RINGING
> 2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:359
> (opal/in:231543514280633) State ROUTING
> 2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:850 Routing connection
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:77
> opal/in:231543514280633 Standard ROUTING
> 2010-12-13 11:58:21.823196 [INFO] mod_dialplan_xml.c:331 Processing
> gustavo [200.117.192.17] <0000000000>->231543514280633 in context
> default
> 2010-12-13 11:58:21.826206 [DEBUG] h323pdu.cxx:80 Receiving PDU:
>   request terminalCapabilitySet {
>     sequenceNumber = 1
>     protocolIdentifier = 0.0.8.245.0.13
>     multiplexCapability = h2250Capability {
>       maximumAudioDelayJitter = 500
>       receiveMultipointCapabi
> lity = {
>         multicastCapability = false
>         multiUniCastConference = false
>         mediaDistributionCapability = 1 entries {
>           [0]={
>             centralizedControl = false
>             distributedControl = false
>             centraliz
> edAudio = false
>             distributedAudio = false
>             centralizedVideo = false
>             distributedVideo = false
>           }
>         }
>       }
>       transmitMultipointCapability = {
>         multicastCapability = false
>         multiUniC
> astConference = false
>         mediaDistributionCapability = 1 entries {
>           [0]={
>             centralizedControl = false
>             distributedControl = false
>             centralizedAudio = false
>             distributedAudio = false
>          
>    centralizedVideo = false
>             distributedVideo = false
>           }
>         }
>       }
>       receiveAndTransmitMultipointCapability = {
>         multicastCapability = false
>         multiUniCastConference = false
>         mediaDistributionCapab
> ility = 1 entries {
>           [0]={
>             centralizedControl = false
>             distributedControl = false
>             centralizedAudio = false
>             distributedAudio = false
>             centralizedVideo = false
>             distributedV
> ideo = false
>           }
>         }
>       }
>       mcCapability = {
>         centralizedConferenceMC = false
>         decentralizedConferenceMC = false
>       }
>       rtcpVideoControlCapability = false
>       mediaPacketizationCapability = {
>         h261a
> VideoPacketization = false
>       }
>       logicalChannelSwitchingCapability = false
>       t120DynamicPortCapability = true
>     }
>     capabilityTable = 8 entries {
>       [0]={
>         capabilityTableEntryNumber = 1
>         capability = receiveAudioCap
> ability g711Alaw64k 240
>       }
>       [1]={
>         capabilityTableEntryNumber = 2
>         capability = receiveVideoCapability h261VideoCapability {
>           qcifMPI = 1
>           cifMPI = 1
>           temporalSpatialTradeOffCapability = false
>      
>      maxBitRate = 19200
>           stillImageTransmission = false
>           videoBadMBsCap = false
>         }
>       }
>       [2]={
>         capabilityTableEntryNumber = 3
>         capability = receiveVideoCapability h261VideoCapability {
>           cifMPI
>  = 1
>           temporalSpatialTradeOffCapability = false
>           maxBitRate = 19200
>           stillImageTransmission = false
>           videoBadMBsCap = false
>         }
>       }
>       [3]={
>         capabilityTableEntryNumber = 4
>         capability =
>  receiveVideoCapability h261VideoCapability {
>           qcifMPI = 1
>           temporalSpatialTradeOffCapability = false
>           maxBitRate = 19200
>           stillImageTransmission = false
>           videoBadMBsCap = false
>         }
>       }
>       [4
> ]={
>         capabilityTableEntryNumber = 5
>         capability = receiveUserInputCapability hookflash <<null>>
>       }
>       [5]={
>         capabilityTableEntryNumber = 6
>         capability = receiveUserInputCapability basicString <<null>>
>       }
>    
>    [6]={
>         capabilityTableEntryNumber = 7
>         capability = receiveUserInputCapability dtmf <<null>>
>       }
>       [7]={
>         capabilityTableEntryNumber = 8
>         capability = receiveRTPAudioTelephonyEventCapability {
>           dynamic
> RTPPayloadType = 101
>           audioTelephoneEvent = "0-16"
>         }
>       }
>     }
>     capabilityDescriptors = 1 entries {
>       [0]={
>         capabilityDescriptorNumber = 1
>         simultaneousCapabilities = 4 entries {
>           [0]=1 entries {
>  
>            [0]=1
>           }
>           [1]=3 entries {
>             [0]=2
>             [1]=3
>             [2]=4
>           }
>           [2]=1 entries {
>             [0]=5
>           }
>           [3]=3 entries {
>             [0]=6
>             [1]=7
>           
> Log-Func: 
> Log-Line: 0
> User-Data: 
> 
>   [2]=8
>     2010-12-13 11:58:21.826206 [DEBUG] h323.cxx:2792 Set protocol
> version to 13
> 2010-12-13 11:58:21.826206 [DEBUG] h323neg.cxx:378 Received
> TerminalCapabilitySet: state=Idle pduSeq=1 inSeq=4294967295
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:1988
> H323Capabilities(ctor)
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
> G.723.1 <1>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
> G.729 <2>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
> G.711-uLaw-64k <3>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
> G.711-ALaw-64k <4>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
> UserInput/hookflash <5>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
> UserInput/basicString <6>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
> UserInput/dtmf <7>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
> UserInput/RFC2833 <8>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
> G.723.1 <1>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
> G.729 <2>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
> G.711-uLaw-64k <3>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
> G.711-ALaw-64k <4>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
> UserInput/hookflash <5>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
> UserInput/basicString <6>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
> UserInput/dtmf <7>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
> UserInput/RFC2833 <8>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2376 Found capability:
> G.711-ALaw-64k <4>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2376 Found capability:
> G.711-uLaw-64k <3>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2376 Found capability:
> G.723.1 <1>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
> capability: "G.723.1(5.3k)"
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
> G.723.1 <9>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
> capability: "G.723.1A(5.3k)"
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
> G.723.1 <10>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
> capability: "G.723.1A(6.3k)"
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
> G.723.1 <11>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
> capability: "G.728"
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
> G.728 <12>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2376 Found capability:
> G.729 <2>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
> capability: "G.729A"
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
> G.729A <13>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
> capability: "G.729A/B"
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
> G.729A/B <14>
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
> capability: "G.729B"
> 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
> G.729B <15>
> 2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2381 Could not find
> capability: "GSM-06.10"
> 2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2284 Added capability:
> GSM-06.10 <16>
> 2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2381 Could not find
> capability: "GSM-AMR"
> 2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2284 Added capability:
> GSM-AMR <17>
> 2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2381 Could not find
> capability: "T.38"
> 2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2284 Added capability:
> T.38 <18>
> 2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2376 Found capability:
> UserInput/RFC2833 <8>
> 2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2376 Found capability:
> UserInput/basicString <6>
> 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2376 Found capability:
> UserInput/dtmf <7>
> 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2381 Could not find
> capability: "UserInput/generalString"
> 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
> UserInput/generalString <19>
> 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2376 Found capability:
> UserInput/hookflash <5>
> 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2381 Could not find
> capability: "UserInput/iA5String"
> 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
> UserInput/iA5String <20>
> 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2381 Could not find
> capability: "iLBC"
> 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
> iLBC <21>
> 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
> UserInput/hookflash <22>
> 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
> UserInput/basicString <23>
> 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
> UserInput/dtmf <24>
> 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
> UserInput/RFC2833 <25>
> 2010-12-13 11:58:21.834192 [CONSOLE] mod_xml_curl.c:312 XML 
> response is
> in /tmp/90f1dd7d-a63c-4e15-981c-6eed9e70313b.tmp.xml
> Dialplan: opal/in:231543514280633 parsing [default->external]
> continue=false
> Dialplan: opal/in:231543514280633 Regex (PASS) [external]
> destination_number(231543514280633) =~ /^231543514280633/ 
> break=on-false
> Dialplan: opal/in:231543514280633 Action set(continue_on_fail=true) 
> Dialplan: opal/in:231543514280633 Action 
> set(hangup_after_bridge=true) 
> Dialplan: opal/in:231543514280633 Action set(progress_timeout=15) 
> Dialplan: opal/in:231543514280633 Action set(proxy_media=false) 
> Dialplan: opal/in:231543514280633 Action set(bypass_media=true) 
> Dialplan: opal/in:231543514280633 Action
> set(absolute_codec_string=PCMA) 
> Dialplan: opal/in:231543514280633 Action
> bridge(sofia/external/21543514280633 at 200.35.145.149) 
> 2010-12-13 11:58:21.834192 [DEBUG] switch_core_state_machine.c:119
> (opal/in:231543514280633) State Change CS_ROUTING -> CS_EXECUTE
> 2010-12-13 11:58:21.834192 [DEBUG] mod_opal.cpp:1092 State changed on
> connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.834192 [DEBUG] switch_core_session.c:1083 Send
> signal opal/in:231543514280633 [BREAK]
> 2010-12-13 11:58:21.834192 [DEBUG] mod_opal.cpp:931 Kill 3 on 
> connection
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.834192 [DEBUG] switch_core_state_machine.c:359
> (opal/in:231543514280633) State ROUTING going to sleep
> 2010-12-13 11:58:21.834192 [DEBUG] switch_core_state_machine.c:320
> (opal/in:231543514280633) Running State Change CS_EXECUTE
> 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
> H.239-Video+H.239-Video <26>
> 2010-12-13 11:58:21.834192 [DEBUG] switch_core_state_machine.c:366
> (opal/in:231543514280633) State EXECUTE
> 2010-12-13 11:58:21.834192 [DEBUG] mod_opal.cpp:857 Executing 
> connection
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.834192 [DEBUG] switch_core_state_machine.c:157
> opal/in:231543514280633 Standard EXECUTE
> EXECUTE opal/in:231543514280633 set(continue_on_fail=true)
> 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2284 Added capability:
> H.239-Control <27>
> 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2029 Parsing remote
> capabilities
> 2010-12-13 11:58:21.837191 [DEBUG] mod_opal.cpp:1076 Received 
> message 26
> on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.837191 [DEBUG] mod_dptools.c:1028
> opal/in:231543514280633 SET [continue_on_fail]=[true]
> 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:819 
> Capability tx frames
> left at 20 as remote allows 240
> 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2529 Could not find
> capability: receiveVideoCapability, type h261VideoCapability
> 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2529 Could not find
> capability: receiveVideoCapability, type h261VideoCapability
> 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2529 Could not find
> capability: receiveVideoCapability, type h261VideoCapability
> 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2671 Found capability:
> UserInput/RFC2833 <8>
> 2010-12-13 11:58:21.837191 [DEBUG] mod_opal.cpp:1076 Received 
> message 27
> on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> EXECUTE opal/in:231543514280633 set(hangup_after_bridge=true)
> 2010-12-13 11:58:21.837191 [DEBUG] mod_opal.cpp:1076 Received 
> message 26
> on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.837191 [DEBUG] mod_dptools.c:1028
> opal/in:231543514280633 SET [hangup_after_bridge]=[true]
> 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2411 Could not find
> capability: G.711-ALaw-64k <1>
> 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2294 Added capability:
> G.711-ALaw-64k <1>
> 2010-12-13 11:58:21.837191 [DEBUG] mod_opal.cpp:1076 Received 
> message 27
> on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> EXECUTE opal/in:231543514280633 set(progress_timeout=15)
> 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2411 Could not find
> capability: UserInput/hookflash <5>
> 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2294 Added capability:
> UserInput/hookflash <5>
> 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2411 Could not find
> capability: UserInput/basicString <6>
> 2010-12-13 11:58:21.837191 [DEBUG] mod_opal.cpp:1076 Received 
> message 26
> on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.837191 [DEBUG] mod_dptools.c:1028
> opal/in:231543514280633 SET [progress_timeout]=[15]
> 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2294 Added capability:
> UserInput/basicString <6>
> 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2411 Could not find
> capability: UserInput/dtmf <7>
> 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2294 Added capability:
> UserInput/dtmf <7>
> 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2411 Could not find
> capability: UserInput/RFC2833 <8>
> 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2294 Added capability:
> UserInput/RFC2833 <8>
> 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2355 Found capability:
> G.711-ALaw-64k <1>
> 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2355 Found capability:
> UserInput/hookflash <5>
> 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2355 Found capability:
> UserInput/basicString <6>
> 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2355 Found capability:
> UserInput/dtmf <7>
> 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2355 Found capability:
> UserInput/RFC2833 <8>
> 2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received 
> message 27
> on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> EXECUTE opal/in:231543514280633 set(proxy_media=false)
> 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2777 Capability merge
> result:
>      Table:
>        G.711-ALaw-64k <1>
>        UserInput/hookflash <5>
>        UserInput/basicString <6>
>        UserInput/dtmf <7>
>        UserInput/RFC2833 <8>
>      Set:
>        0:
>          0:
>      
> 
> Log-Func: 
> Log-Line: 0
> User-Data: 
> 
>       G.711-ALaw-64k <1>
>          1:
>          2:
>            UserInput/hookflash <5>
>          3:
>            UserInput/basicString <6>
>            UserInput/dtmf <INCOMING DATA [(null)]
> 
> RECV EVENT
> Event-Name: SOCKET_DATA
> 
> 
> 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2778 Received 
> capability
> set, is accepted
> 2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received 
> message 26
> on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.840189 [DEBUG] mod_dptools.c:1028
> opal/in:231543514280633 SET [proxy_media]=[false]
> 2010-12-13 11:58:21.840189 [DEBUG] h323neg.cxx:341 Sending
> TerminalCapabilitySet: outSeq=1
> 2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received 
> message 27
> on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> EXECUTE opal/in:231543514280633 set(bypass_media=true)
> 2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received 
> message 26
> on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.840189 [DEBUG] mod_dptools.c:1028
> opal/in:231543514280633 SET [bypass_media]=[true]
> 2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received 
> message 27
> on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> EXECUTE opal/in:231543514280633 set(absolute_codec_string=PCMA)
> 2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received 
> message 26
> on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.840189 [DEBUG] mod_dptools.c:1028
> opal/in:231543514280633 SET [absolute_codec_string]=[PCMA]
> 2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received 
> message 27
> on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> EXECUTE opal/in:231543514280633
> bridge(sofia/external/21543514280633 at 200.35.145.149)
> 2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received 
> message 26
> on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.840189 [NOTICE] switch_channel.c:784 New Channel
> sofia/external/21543514280633 at 200.35.145.149
> [f9bf5b9e-a273-400c-b18c-e2c98c84750f]
> 2010-12-13 11:58:21.840189 [DEBUG] mod_sofia.c:3995
> (sofia/external/21543514280633 at 200.35.145.149) State Change CS_NEW ->
> CS_INIT
> 2010-12-13 11:58:21.840189 [DEBUG] switch_core_session.c:1083 Send
> signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
> 2010-12-13 11:58:21.843189 [DEBUG] h323pdu.cxx:80 Sending PDU:
>   request terminalCapabilitySet {
>     sequenceNumber = 1
>     protocolIdentifier = 0.0.8.245.0.13
>     multiplexCapability = h2250Capability {
>       maximumAudioDelayJitter = 250
>       receiveMultipointCapabili
> ty = {
>         multicastCapability = false
>         multiUniCastConference = false
>         mediaDistributionCapability = 1 entries {
>           [0]={
>             centralizedControl = false
>             distributedControl = false
>             centralized
> Audio = false
>             distributedAudio = false
>             centralizedVideo = false
>             distributedVideo = false
>           }
>         }
>       }
>       transmitMultipointCapability = {
>         multicastCapability = false
>         multiUniCas
> tConference = false
>         mediaDistributionCapability = 1 entries {
>           [0]={
>             centralizedControl = false
>             distributedControl = false
>             centralizedAudio = false
>             distributedAudio = false
>            
>  centralizedVideo = false
>             distributedVideo = false
>           }
>         }
>       }
>       receiveAndTransmitMultipointCapability = {
>         multicastCapability = false
>         multiUniCastConference = false
>         mediaDistributionCapabil
> ity = 1 entries {
>           [0]={
>             centralizedControl = false
>             distributedControl = false
>             centralizedAudio = false
>             distributedAudio = false
>             centralizedVideo = false
>             distributedVid
> eo = false
>           }
>         }
>       }
>       mcCapability = {
>         centralizedConferenceMC = false
>         decentralizedConferenceMC = false
>       }
>       rtcpVideoControlCapability = false
>       mediaPacketizationCapability = {
>         h261aVi
> deoPacketization = false
>       }
>       logicalChannelSwitchingCapability = false
>       t120DynamicPortCapability = true
>     }
>     capabilityTable = 8 entries {
>       [0]={
>         capabilityTableEntryNumber = 1
>         capability = receiveAudioCapab
> ility g7231 {
>           maxAl_sduAudioFrames = 8
>           silenceSuppression = false
>         }
>       }
>       [1]={
>         capabilityTableEntryNumber = 2
>         capability = receiveAudioCapability g729 24
>       }
>       [2]={
>         capabilityTabl
> eEntryNumber = 3
>         capability = receiveAudioCapability g711Ulaw64k 240
>       }
>       [3]={
>         capabilityTableEntryNumber = 4
>         capability = receiveAudioCapability g711Alaw64k 240
>       }
>       [4]={
>         capabilityTableEntryNumbe
> r = 5
>         capability = receiveUserInputCapability hookflash <<null>>
>       }
>       [5]={
>         capabilityTableEntryNumber = 6
>         capability = receiveUserInputCapability basicString <<null>>
>       }
>       [6]={
>         capabilityTableEntry
> Number = 7
>         capability = receiveUserInputCapability dtmf <<null>>
>       }
>       [7]={
>         capabilityTableEntryNumber = 8
>         capability = receiveRTPAudioTelephonyEventCapability {
>           dynamicRTPPayloadType = 101
>           audioT
> elephoneEvent = "0-16"
>         }
>       }
>     }
>     capabilityDescriptors = 1 entries {
>       [0]={
>         capabilityDescriptorNumber = 1
>         simultaneousCapabilities = 3 entries {
>           [0]=4 entries {
>             [0]=1
>             [1]=2
>   
>           [2]=3
>             [3]=4
>           }
>           [1]=1 entries {
>             [0]=5
>           }
>           [2]=3 entries {
>             [0]=6
>             [1]=7
>             [2]=8
>           }
>         }
>       }
>     }
>   }
> 2010-12-13 11:58:21.843189 [DEBUG] h323caps.cxx:2376 Found capability:
> UserInput/RFC2833 <8>
> 2010-12-13 11:58:21.843189 [DEBUG] h323.cxx:3535 User Input RFC2833
> payload type set to [pt=101]
> 2010-12-13 11:58:21.843189 [DEBUG] h323pdu.cxx:80 Sending PDU:
>   response terminalCapabilitySetAck {
>     sequenceNumber = 1
>   }
> 2010-12-13 11:58:21.843189 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/21543514280633 at 200.35.145.149) Running State Change
> CS_INIT
> 2010-12-13 11:58:21.843189 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/21543514280633 at 200.35.145.149) State INIT
> 2010-12-13 11:58:21.843189 [DEBUG] mod_sofia.c:86
> sofia/external/21543514280633 at 200.35.145.149 SOFIA INIT
> 2010-12-13 11:58:21.843189 [DEBUG] h323.cxx:3771
> InternalEstablishedConnectionCheck: 
> connectionState=AwaitingLocalAnswer
> fastStartState=FastStartResponse H.245 is unavailable
> 2010-12-13 11:58:21.843189 [DEBUG] h323pdu.cxx:80 Receiving PDU:
>   request masterSlaveDetermination {
>     terminalType = 50
>     statusDeterminationNumber = 6111531
>   }
> 2010-12-13 11:58:21.843189 [DEBUG] h323neg.cxx:129 Received
> MasterSlaveDetermination: state=Idle
> 2010-12-13 11:58:21.843189 [DEBUG] h323neg.cxx:160
> MasterSlaveDetermination: local is master
> 2010-12-13 11:58:21.843189 [DEBUG] h323pdu.cxx:80 Sending PDU:
>   response masterSlaveDeterminationAck {
>     decision = slave <<null>>
>   }
> 2010-12-13 11:58:21.843189 [DEBUG] h323.cxx:3771
> InternalEstablishedConnectionCheck: 
> connectionState=AwaitingLocalAnswer
> fastStartState=FastStartResponse H.245 is ready
> 2010-12-13 11:58:21.843189 [DEBUG] mod_sofia.c:126
> (sofia/external/21543514280633 at 200.35.145.149) State Change CS_INIT ->
> CS_ROUTING
> 2010-12-13 11:58:21.843189 [DEBUG] switch_core_session.c:1083 Send
> signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
> 2010-12-13 11:58:21.843189 [DEBUG] sofia.c:4604 Channel
> sofia/external/21543514280633 at 200.35.145.149 entering state
> [terminated][900]
> 2010-12-13 11:58:21.843189 [DEBUG] h323ep.cxx:1033 
> OnSendAlerting conn =
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> 2010-12-13 11:58:21.843189 [DEBUG] switch_channel.c:2455
> (sofia/external/21543514280633 at 200.35.145.149) Callstate 
> Change DOWN ->
> HANGUP
> 2010-12-13 11:58:21.843189 [DEBUG] h323.cxx:2110 SetAlerting sending
> Alerting PDU
> 2010-12-13 11:58:21.843189 [NOTICE] sofia.c:5244 Hangup
> sofia/external/21543514280633 at 200.35.145.149 [CS_ROUTING]
> [NORMAL_UNSPECIFIED]
> 2010-12-13 11:58:21.843189 [DEBUG] switch_channel.c:2471 Send signal
> sofia/external/21543514280633 at 200.35.145.149 [KILL]
> 2010-12-13 11:58:21.843189 [DEBUG] switch_core_session.c:1083 Send
> signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
> 2010-12-13 11:58:21.843189 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/21543514280633 at 200.35.145.149) State INIT 
> going to sleep
> 2010-12-13 11:58:21.843189 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/21543514280633 at 200.35.145.149) Running State Change
> CS_HANGUP
> 2010-12-13 11:58:21.846211 [DEBUG] h323pdu.cxx:80 Sending PDU:
>   {
>     q931pdu = {
>       protocolDiscriminator = 8
>       callReference = 22716
>       from = destination
>       messageType = Alerting
>       IE: Display = {
>         72 6f 6f 74 00                              
>        root.
>       }
>       IE: User-User = {
>         23 80 06 00 08 91 4a 00  06 22 c0 09 00 00 3d 1e
> #.....J.."....=.
>         56 6f 78 20 4c 75 63 69  64 61 20 50 74 79 2e 20   Vox Lucida
> Pty. 
>         ...
>       }
>     }
>     h225pdu = {
>       h323
> _uu_pdu = {
>         h323_message_body = alerting {
>           protocolIdentifier = 0.0.8.2250.0.6
>           destinationInfo = {
>             vendor = {
>               vendor = {
>                 t35CountryCode = 9
>                 t35Extension = 0
>       
>           manufacturerCode = 61
>               }
>               productId =  31 octets {
>                 56 6f 78 20 4c 75 63 69  64 61 20 50 74 79 2e 20   Vox
> Lucida Pty. 
>                 4c 74 64 2e 20 6d 6f 64  5f 6f 70 61 6c 00 00 
>      Ltd.
> mod_op
> Log-Line: 0
> User-Data: 
> 
>    }
>             terminal = {
>             }
>             mc = false
>             undefinedNode = false
>           }
>           callIdentifier = {
>             guid =  16 octets {
>               3a 76 b2 f2 1d 05 e0 11  9b b7 00 1e 64 1f 33
> fe   :v........
> ..d.3.
>             }
>           }
>           multipleCalls = false
>           maintainConnection = false
>         }
>         h245Tunneling = true
>         h245Control = 3 entries {
>           [0]= 108 octets {
>             02 70 01 06 00 08 81 75  00 0d 80 
> 13 80 00 fa 00   .p.....u........
>             01 00 00 01 00 00 01 00  00 0c c0 01 00 01 80
> 07   ................
>             ...
>           }
>           [1]= 3 octets {
>             21 80 01                                           !..
>           }
>   
>         [2]= 2 octets {
>             20 a0                                               .
>           }
>         }
>       }
>     }
>   }
> 2010-12-13 11:58:21.846211 [DEBUG] h323ep.cxx:1039 
> OnSentAlerting conn =
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> 2010-12-13 11:58:21.846211 [DEBUG] h323.cxx:3771
> InternalEstablishedConnectionCheck: 
> connectionState=AwaitingLocalAnswer
> fastStartState=FastStartResponse H.245 is ready
> 2010-12-13 11:58:21.846211 [DEBUG] connection.cxx:361
> OnSetUpConnectionCall[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.846211 [DEBUG] endpoint.cxx:408 OnSetUpConnection
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.846211 [DEBUG] h323.cxx:1701 OnAnswerCall
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716], caller =
> gustavo [200.117.192.17]
> 2010-12-13 11:58:21.846211 [DEBUG] call.cxx:219 OnAnswerCall
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716] caller
> "gustavo [200.117.192.17]"
> 2010-12-13 11:58:21.846211 [DEBUG] h323.cxx:1713 Answering call:
> AnswerCallDeferred
> 2010-12-13 11:58:21.846211 [DEBUG] connection.cxx:541 Answering call:
> AnswerCallDeferred
> 2010-12-13 11:58:21.846211 [DEBUG] h323.cxx:3771
> InternalEstablishedConnectionCheck: 
> connectionState=AwaitingLocalAnswer
> fastStartState=FastStartResponse H.245 is ready
> 2010-12-13 11:58:21.846211 [DEBUG] h323.cxx:506 Reading PDUs:
> callRef=22716
> 2010-12-13 11:58:21.846211 [DEBUG] 
> switch_ivr_originate.c:3448 Originate
> Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED]
> 2010-12-13 11:58:21.849192 [INFO] mod_dptools.c:2579 Originate Failed.
> Cause: NORMAL_UNSPECIFIED
> 2010-12-13 11:58:21.849192 [DEBUG] mod_opal.cpp:1076 Received 
> message 27
> on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.849192 [NOTICE] switch_core_state_machine.c:189
> opal/in:231543514280633 has executed the last dialplan instruction,
> hanging up.
> 2010-12-13 11:58:21.849192 [DEBUG] switch_channel.c:2455
> (opal/in:231543514280633) Callstate Change RINGING -> HANGUP
> 2010-12-13 11:58:21.849192 [NOTICE] switch_core_state_machine.c:191
> Hangup opal/in:231543514280633 [CS_EXECUTE] [NORMAL_CLEARING]
> 2010-12-13 11:58:21.849192 [DEBUG] switch_channel.c:2471 Send signal
> opal/in:231543514280633 [KILL]
> 2010-12-13 11:58:21.849192 [DEBUG] mod_opal.cpp:931 Kill 1 on 
> connection
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.849192 [DEBUG] mod_opal.cpp:1092 State changed on
> connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.849192 [DEBUG] switch_core_session.c:1083 Send
> signal opal/in:231543514280633 [BREAK]
> 2010-12-13 11:58:21.849192 [DEBUG] mod_opal.cpp:931 Kill 3 on 
> connection
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:366
> (opal/in:231543514280633) State EXECUTE going to sleep
> 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:320
> (opal/in:231543514280633) Running State Change CS_HANGUP
> 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:557
> (sofia/external/21543514280633 at 200.35.145.149) State HANGUP
> 2010-12-13 11:58:21.849192 [DEBUG] mod_sofia.c:453
> sofia/external/21543514280633 at 200.35.145.149 Overriding SIP cause 480
> with 900 from the other leg
> 2010-12-13 11:58:21.849192 [DEBUG] mod_sofia.c:459 Channel
> sofia/external/21543514280633 at 200.35.145.149 hanging up, cause:
> NORMAL_UNSPECIFIED
> 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:46
> sofia/external/21543514280633 at 200.35.145.149 Standard HANGUP, cause:
> NORMAL_UNSPECIFIED
> 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:557
> (sofia/external/21543514280633 at 200.35.145.149) State HANGUP going to
> sleep
> 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:351
> (sofia/external/21543514280633 at 200.35.145.149) State Change 
> CS_HANGUP ->
> CS_REPORTING
> 2010-12-13 11:58:21.849192 [DEBUG] switch_core_session.c:1083 Send
> signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
> 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/21543514280633 at 200.35.145.149) Running State Change
> CS_REPORTING
> 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:617
> (sofia/external/21543514280633 at 200.35.145.149) State REPORTING
> 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:557
> (opal/in:231543514280633) State HANGUP
> 2010-12-13 11:58:21.849192 [DEBUG] connection.cxx:401 Call end reason
> for Call[Cc70e3cf91]-EP<local>[Ldc8d6c462] set to EndedByRemoteUser
> 2010-12-13 11:58:21.849192 [DEBUG] call.cxx:112 Clearing
> Call[Cc70e3cf91] reason=EndedByRemoteUser
> 2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:1516 SetPhase from
> SetUpPhase to ReleasingPhase for Call[Cc70e3cf91]-EP<h323>[tcp
> $200.117.192.17:19270/22716]
> 2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:459 Releasing
> asynchronously 
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> 2010-12-13 11:58:21.852189 [DEBUG] osutil.cxx:196 File handle 
> low water
> mark set: 69 Thread unblock pipe
> 2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:1516 SetPhase from
> AlertingPhase to ReleasingPhase for
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:448 Releasing
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:489 OnReleased
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:808 Media streams
> closed.
> 2010-12-13 11:58:21.852189 [DEBUG] endpoint.cxx:450 OnReleased
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.852189 [DEBUG] manager.cxx:709 OnReleased
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.852189 [DEBUG] call.cxx:716 OnReleased
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:436 Already released
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> 2010-12-13 11:58:21.856029 [DEBUG] connection.cxx:1516 SetPhase from
> ReleasingPhase to ReleasedPhase for
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:46
> opal/in:231543514280633 Standard HANGUP, cause: NORMAL_CLEARING
> 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:557
> (opal/in:231543514280633) State HANGUP going to sleep
> 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:351
> (opal/in:231543514280633) State Change CS_HANGUP -> CS_REPORTING
> 2010-12-13 11:58:21.856029 [DEBUG] switch_core_session.c:1083 Send
> signal opal/in:231543514280633 [BREAK]
> 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:320
> (opal/in:231543514280633) Running State Change CS_REPORTING
> 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:617
> (opal/in:231543514280633) State REPORTING
> 2010-12-13 11:58:21.856029 [DEBUG] connection.cxx:401 Call end reason
> for Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716] set to
> EndedByRemoteUser
> 2010-12-13 11:58:21.856029 [DEBUG] h323.cxx:349 OnReleased: tcp
> $200.117.192.17:19270/22716, connectionState=AwaitingLocalAnswer
> 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:53
> sofia/external/21543514280633 at 200.35.145.149 Standard 
> REPORTING, cause:
> NORMAL_UNSPECIFIED
> 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:617
> (sofia/external/21543514280633 at 200.35.145.149) State 
> REPORTING going to
> sleep
> 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:345
> (sofia/external/21543514280633 at 200.35.145.149) State Change 
> CS_REPORTING
> -> CS_DESTROY
> 2010-12-13 11:58:21.856029 [DEBUG] switch_core_session.c:1083 Send
> signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
> 2010-12-13 11:58:21.856029 [DEBUG] switch_core_session.c:1250 
> Session 2
> (sofia/external/21543514280633 at 200.35.145.149) Locked, Waiting on
> external entities
> 2010-12-13 11:58:21.856029 [NOTICE] 
> switch_core_session.c:1268 Session 2
> (sofia/external/21543514280633 at 200.35.145.149) Ended
> 2010-12-13 11:58:21.856029 [NOTICE] switch_core_session.c:1270 Close
> Channel sofia/external/21543514280633 at 200.35.145.149 [CS_DESTROY]
> 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:449
> (sofia/external/21543514280633 at 200.35.145.149) Callstate Change HANGUP
> -> DOWN
> 2010-12-13 11:58:21.858228 [DEBUG] switch_core_state_machine.c:452
> (sofia/external/21543514280633 at 200.35.145.149) Running State Change
> CS_DESTROY
> 2010-12-13 11:58:21.858228 [DEBUG] switch_core_state_machine.c:462
> (sofia/external/21543514280633 at 200.35.145.149) State DESTROY
> 2010-12-13 11:58:21.858228 [DEBUG] mod_sofia.c:364
> sofia/external/21543514280633 at 200.35.145.149 SOFIA DESTROY
> 2010-12-13 11:58:21.858228 [DEBUG] switch_core_state_machine.c:60
> sofia/external/21543514280633 at 200.35.145.149 Standard DESTROY
> 2010-12-13 11:58:21.858228 [DEBUG] switch_core_state_machine.c:462
> (sofia/external/21543514280633 at 200.35.145.149) State DESTROY going to
> sleep
> 2010-12-13 11:58:21.856029 [DEBUG] h323.cxx:356 Sending 
> release complete
> PDU: callRef=22716
> 2010-12-13 11:58:21.858228 [DEBUG] h323pdu.cxx:80 Sending PDU:
>   command endSessionCommand disconnect <<null>>
> 2010-12-13 11:58:21.858228 [DEBUG] h323pdu.cxx:80 Sending PDU:
>   {
>     q931pdu = {
>       protocolDiscriminator = 8
>       callReference = 22716
>       from = destination
>       messageType = ReleaseComplete
>       IE: Cause - Normal call clearing = {
>         80 90           
>                                    ..
>       }
>       IE: Display = {
>         72 6f 6f 74 00                                     root.
>       }
>       IE: User-User = {
>         25 80 06 00 08 91 4a 00  06 01 11 00 3a 76 b2 f2
> %.....J.....:v..
>         
> Log-Func: 
> Log-Line: 0
> User-Data: 
> 
> 1d 05 e0 11 9b b7 00 1e  64 1f 33 fe 04 c0 01 80   ........d.3.....
>         ...
>       }
>     }
>     h225pdu = {
>       h323_uu_pdu = {
>         h323_message_body = releaseComplete {
>           protocolIdentifier = 0.0.8= {
>             guid =  16 octets {
>               3a 76 b2 f2 1d 05 e0 11  9b b7 00 1e 64 1f 33
> fe   :v..........d.3.
>             }
>           }
>         }
>         h245Tunneling = true
>         h245Control = 1 entries {
>           [0]= 2 octets {
>       
>       4a 40                                              J@
>           }
>         }
>       }
>     }
>   }
> 2010-12-13 11:58:21.861189 [DEBUG] h323.cxx:412 Awaiting end session
> from remote for 9.997 seconds
> 2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:53
> opal/in:231543514280633 Standard REPORTING, cause: NORMAL_CLEARING
> 2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:617
> (opal/in:231543514280633) State REPORTING going to sleep
> 2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:345
> (opal/in:231543514280633) State Change CS_REPORTING -> CS_DESTROY
> 2010-12-13 11:58:21.861189 [DEBUG] switch_core_session.c:1083 Send
> signal opal/in:231543514280633 [BREAK]
> 2010-12-13 11:58:21.861189 [DEBUG] switch_core_session.c:1250 
> Session 1
> (opal/in:231543514280633) Locked, Waiting on external entities
> 2010-12-13 11:58:21.861189 [NOTICE] 
> switch_core_session.c:1268 Session 1
> (opal/in:231543514280633) Ended
> 2010-12-13 11:58:21.861189 [NOTICE] switch_core_session.c:1270 Close
> Channel opal/in:231543514280633 [CS_DESTROY]
> 2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:449
> (opal/in:231543514280633) Callstate Change HANGUP -> DOWN
> 2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:452
> (opal/in:231543514280633) Running State Change CS_DESTROY
> 2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:462
> (opal/in:231543514280633) State DESTROY
> 2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:60
> opal/in:231543514280633 Standard DESTROY
> 2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:462
> (opal/in:231543514280633) State DESTROY going to sleep
> 2010-12-13 11:58:22.087167 [DEBUG] h323pdu.cxx:80 Receiving PDU:
>   {
>     q931pdu = {
>       protocolDiscriminator = 8
>       callReference = 22716
>       from = originator
>       messageType = Facility
>       IE: Facility = {
> 
>       }
>       IE: Display = {
>         67 75 73 74
> (..........!... 
>         80                                                 .
>       }
>     }
>     h225pdu = {
>  
>      h323_uu_pdu = {
>         h323_message_body = empty <<null>>
>         h245Tunneling = true
>         h245Control = 2 entries {
>           [0]= 3 octets {
>             21 80 01                                           !..
>           }
>           [1]= 2 
> octets {
>             20 80                                               .
>           }
>         }
>       }
>     }
>   }
> 2010-12-13 11:58:22.087167 [DEBUG] h323.cxx:559 Handling PDU: Facility
> callRef=22716
> 2010-12-13 11:58:22.087167 [DEBUG] h323.cxx:2709 Checking for end
> session on PDU: response terminalCapabilitySetAck
> 2010-12-13 11:58:22.087167 [DEBUG] h323.cxx:2709 Checking for end
> session on PDU: response masterSlaveDeterminationAck
> 2010-12-13 11:58:22.087167 [DEBUG] connection.cxx:436 Already released
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> 2010-12-13 11:58:22.087167 [DEBUG] h323.cxx:549 Signal channel closed.
> 2010-12-13 11:58:22.087167 [DEBUG] transports.cxx:900 
> Transport clean up
> on termination
> 2010-12-13 11:58:22.087167 [DEBUG] transports.cxx:885 Transport Close
> 2010-12-13 11:58:22.090169 [DEBUG] osutil.cxx:196 File handle 
> low water
> mark set: 65 PTextFile
> 2010-12-13 11:58:22.090169 [DEBUG] tlibthrd.cxx:1020 Could not parse
> thread stat file /proc/21145/task/21190/stat
> 2010-12-13 11:58:22.306147 [DEBUG] connection.cxx:489 OnReleased
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> 2010-12-13 11:58:22.306147 [DEBUG] connection.cxx:808 Media streams
> closed.
> 2010-12-13 11:58:22.306147 [DEBUG] endpoint.cxx:450 OnReleased
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> 2010-12-13 11:58:22.306147 [DEBUG] manager.cxx:709 OnReleased
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> 2010-12-13 11:58:22.306147 [DEBUG] call.cxx:716 OnReleased
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> 2010-12-13 11:58:22.306147 [DEBUG] manager.cxx:535 OnClearedCall
> Call[Cc70e3cf91] from "h323:200.117.192.17" to "h323:231543514280633"
> 2010-12-13 11:58:22.306147 [DEBUG] connection.cxx:1516 SetPhase from
> ReleasingPhase to ReleasedPhase for Call[Cc70e3cf91]-EP<h323>[tcp
> $200.117.192.17:19270/22716]
> 2010-12-13 11:58:22.306147 [DEBUG] connection.cxx:480 OnRelease thread
> completed for 
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> 2010-12-13 11:58:22.308152 [DEBUG] tlibthrd.cxx:1020 Could not parse
> thread stat file /proc/21145/task/21194/stat
> 2010-12-13 11:58:22.588122 [DEBUG] transports.cxx:900 
> Transport clean up
> on termination
> 2010-12-13 11:58:22.588122 [DEBUG] transports.cxx:885 Transport Close
> 2010-12-13 11:58:22.588122 [DEBUG] transports.cxx:1045 
> Deleted transport
> tcp$200.117.192.17:19270<if=tcp$72.51.47.100:1720>
> 2010-12-13 11:58:22.590122 [DEBUG] h323.cxx:330 Connection tcp
> $200.117.192.17:19270/22716 deleted.
> 2010-12-13 11:58:22.590122 [DEBUG] connection.cxx:343 Connection
> Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716] destroyed.
> 2010-12-13 11:58:22.590122 [DEBUG] localep.cxx:211 Deleted connection.
> 2010-12-13 11:58:22.590122 [DEBUG] connection.cxx:343 Connection
> Call[Cc70e3cf91]-EP<local>[Ldc8d6c462] destroyed.
> 2010-12-13 11:58:23.592039 [DEBUG] call.cxx:86 Call[Cc70e3cf91]
> destroyed.
> 
> --------------------------------------------------------------
> --------------------------------------------------------------
> -----------------------------------------
> Gustavo Espeche
> www.easyipcall.com
> 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
> itch-users
> http://www.freeswitch.org



From davidjbrazier at gmail.com  Mon Dec 13 17:55:34 2010
From: davidjbrazier at gmail.com (David Brazier)
Date: Mon, 13 Dec 2010 14:55:34 +0000
Subject: [Freeswitch-users] Cepstral + FS question
In-Reply-To: <AANLkTi=N4yG-z5FLUo_hn8YCqDZezSpTcmn74=Qv7c8u@mail.gmail.com>
References: <AANLkTi=N4yG-z5FLUo_hn8YCqDZezSpTcmn74=Qv7c8u@mail.gmail.com>
Message-ID: <AANLkTimt4N9jGQ9aNWGCGYMk3L0pxJWhSOSPzuJ4h=N4@mail.gmail.com>

On Tue, Dec 7, 2010 at 12:38 AM, Malay Thakershi <mthakershi at gmail.com> wrote:
> Hello, it would be great help if someone who has used Cepstral from FS can
> share their views.
> 1. I just have one Cepstral Allison voice license (1 port) on my FS server.
> I use swift command to convert text files to WAV which are then played by
> mod_managed in FS call process. First question is regarding limitations on
> simultaneous conversions (TXT to WAV) using swift command. If I have
> multiple threads doing this, will there be any degradation because of
> Cepstral? I tried running two BAT files with 3 commands each. But running
> them simultaneously or separate produced same outcome. Does anyone know when
> Cepstral licensing kicks in and starts degrading quality ( or worse
> inserting "not licensed") prompt?

The synthesis via the swift command is much faster than real time
(i.e. the time it takes to play the WAV) and is only restricted by CPU
speed and licenses.  You'd need to ask Cepstral the details, but my
simple tests have shown that it only uses one CPU per command though
some of the processing of multiple command is on multiple CPUs.  But I
think the licensing mechanism prevents multiple commands running
completely simultaneously on multiple CPUs.  I think the "not
licensed" speech is only inserted when you have no license at all -
all that happens if you try to run multiple commands is that they are
just delayed until  there is a free license.  I don't think quality
will ever be degraded - it's just a question of CPUs and licenses that
will determine your throughput.

> 2. When I call session speak from mod_managed (or stream file) after
> selecting Allison / Cepstral as my voice, does Cepstral engine interfere
> with quality of the playback? If yes, when will I see it and how can I
> produce their effects?

Not unless something in Cepstral or FS is going badly wrong!

> 3. What is the sensible number of ports (from Cepstral) I should be prepared
> to buy if findings in the previous points imposes significant limitations?

Depends on the length of your prompts and number and speed of CPUs and
anticipated simultaneous calls.  Try testing the time for a typical
prompt and work it out from there.

David


From brian at freeswitch.org  Mon Dec 13 18:14:52 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 13 Dec 2010 09:14:52 -0600
Subject: [Freeswitch-users] NOTICE about posting.
Message-ID: <614CC9AD-F85E-4AE3-823D-8232D9D1E36C@freeswitch.org>

If you've never posted to our mailing list you are first post moderated.  If you send the same message to the list multiple times while its in the moderator queue I'll reject them all and you'll have to resend it.  I know this sounds mean but when you send the same message 10 times within 3 minutes of each wondering why it doesn't make it to the list it just creates more work for our team so please be patient on weekends if your first posts do not make it to the list.

Thanks,
Brian



From sfrippiat at dti-be.com  Mon Dec 13 18:22:53 2010
From: sfrippiat at dti-be.com (=?ISO-8859-1?Q?S=E9bastien_Frippiat?=)
Date: Mon, 13 Dec 2010 16:22:53 +0100
Subject: [Freeswitch-users] NOTICE about posting.
In-Reply-To: <614CC9AD-F85E-4AE3-823D-8232D9D1E36C@freeswitch.org>
References: <614CC9AD-F85E-4AE3-823D-8232D9D1E36C@freeswitch.org>
Message-ID: <4D063A4D.4060205@dti-be.com>

Ok, no problem, I understand. I didn't know the first post(s) was 
moderated and I thought there was a problem when sending my mail.

I'll re-send it then.

Thank you,
Sebastien Frippiat

Le 13/12/2010 16:14, Brian West a ?crit :
> If you've never posted to our mailing list you are first post moderated.  If you send the same message to the list multiple times while its in the moderator queue I'll reject them all and you'll have to resend it.  I know this sounds mean but when you send the same message 10 times within 3 minutes of each wondering why it doesn't make it to the list it just creates more work for our team so please be patient on weekends if your first posts do not make it to the list.
>
> Thanks,
> Brian
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From sfrippiat at dti-be.com  Mon Dec 13 18:23:02 2010
From: sfrippiat at dti-be.com (=?ISO-8859-1?Q?S=E9bastien_Frippiat?=)
Date: Mon, 13 Dec 2010 16:23:02 +0100
Subject: [Freeswitch-users] originate call generates weird noise and freezes
	phone
Message-ID: <4D063A56.9000801@dti-be.com>

Hello.

I was using FreeSwitch from 2010/09/09 (don't have the exact git version 
sorry) but I recently updated it with version from 2010/12/09 
(git-e680c82 2010-12-09 08-59-06 -0600). I run an application that 
handles web requests and convert them to originate calls. Everything was 
working fine but now, anytime I try an originate call, everything goes 
wrong.

Here is what I do (to originate a call from 26 to 27):
- bgapi originate {{parameters }}sofia/internal/26% 27
   with the following parameters:
     origination_caller_id_name='{0}',
     origination_caller_id_number='{1}',
     originate_timeout=20,
     effective_caller_id_name='{2}',
     effective_caller_id_number='{3}',
     allow_outside_calls={0},
     outside_calls_gateway={1}
   (the last two parameters are user parameters I use in my dialplan)
- phone 26 starts ringing and I pick it up (silence on the line)
- phone 27 starts ringing and I pick it up (lots of deafening noise + 
phone freezes after a few seconds)

As I said, it was working fine with version from early september. I did 
not find anything useful in the logs or in the config. Has anything 
changed ? What can I do to solve this problem ?

I tested with a Grandstream GXP2000 and a SNOM M3 and only the 
Grandstream has the problem. I suspect a codec issue but what puzzles me 
is that it was previously working and that it completely freezes the phone.

Thanks,
Sebastien
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From zahid.maqbool at gmail.com  Mon Dec 13 18:23:38 2010
From: zahid.maqbool at gmail.com (Zahid Maqbool)
Date: Mon, 13 Dec 2010 15:23:38 +0000
Subject: [Freeswitch-users] NOTICE about posting.
In-Reply-To: <614CC9AD-F85E-4AE3-823D-8232D9D1E36C@freeswitch.org>
References: <614CC9AD-F85E-4AE3-823D-8232D9D1E36C@freeswitch.org>
Message-ID: <AANLkTi=sfZniedwgyFAg7jQyBUhrSttjVUgBkDAGQVZC@mail.gmail.com>

Brian,

Thanks for your prompt reply. I am sorry about that, but I forget to put
some info, so I had edited my original post. I didn't knew it will resend
it. I am not so familiar with mailing lists I thought its more like a forum.
Apologies for the incovenience. I will keep it in mind now.

Thanks a lot and I'm enjoying Freeswitch by the way :)

Regards,

Zahid

On Mon, Dec 13, 2010 at 3:14 PM, Brian West <brian at freeswitch.org> wrote:

> If you've never posted to our mailing list you are first post moderated.
>  If you send the same message to the list multiple times while its in the
> moderator queue I'll reject them all and you'll have to resend it.  I know
> this sounds mean but when you send the same message 10 times within 3
> minutes of each wondering why it doesn't make it to the list it just creates
> more work for our team so please be patient on weekends if your first posts
> do not make it to the list.
>
> Thanks,
> Brian
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kind Regards,

Zahid Maqbool
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From juanito1982 at gmail.com  Mon Dec 13 18:50:15 2010
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Mon, 13 Dec 2010 16:50:15 +0100
Subject: [Freeswitch-users] a2billing and freeswitch
In-Reply-To: <COL122-W4030C97574705F9B9DC69FB5120@phx.gbl>
References: <COL122-W4030C97574705F9B9DC69FB5120@phx.gbl>
Message-ID: <AANLkTinu3fVVGb2fB=h_SVsfd06_A9ce1wYHxR0V3jug@mail.gmail.com>

A2billing is Asterisk + PHP AGI so it must work. You could search docs about
Asterisk + Freeswitch.

Regards

2010/12/13 Rafqat . <rafonline at hotmail.com>

>
> Hi
>
> I was wondering if a2billing can be setup to work with freeswitch?
>
> If so, can anyone please point me to some documentation on how this can be
> done?
>
> Cheers
>
> Raf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Dec 13 19:10:36 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 13 Dec 2010 10:10:36 -0600
Subject: [Freeswitch-users] originate call generates weird noise and
 freezes phone
In-Reply-To: <4D063A56.9000801@dti-be.com>
References: <4D063A56.9000801@dti-be.com>
Message-ID: <AANLkTi=2oSETkRnT88S4kVOELfXNKMjMB0-MzBPOkfCf@mail.gmail.com>

try setting the global variable verbose_sdp=true in vars.xml
if that doesn't work, post a trace at debug log level + sip trace with
sofia global siptrace on



On Mon, Dec 13, 2010 at 9:23 AM, S?bastien Frippiat
<sfrippiat at dti-be.com> wrote:
> Hello.
>
> I was using FreeSwitch from 2010/09/09 (don't have the exact git version
> sorry) but I recently updated it with version from 2010/12/09 (git-e680c82
> 2010-12-09 08-59-06 -0600). I run an application that handles web requests
> and convert them to originate calls. Everything was working fine but now,
> anytime I try an originate call, everything goes wrong.
>
> Here is what I do (to originate a call from 26 to 27):
> - bgapi originate {{parameters }}sofia/internal/26% 27
> ? with the following parameters:
> ??? origination_caller_id_name='{0}',
> ??? origination_caller_id_number='{1}',
> ??? originate_timeout=20,
> ??? effective_caller_id_name='{2}',
> ??? effective_caller_id_number='{3}',
> ??? allow_outside_calls={0},
> ??? outside_calls_gateway={1}
> ? (the last two parameters are user parameters I use in my dialplan)
> - phone 26 starts ringing and I pick it up (silence on the line)
> - phone 27 starts ringing and I pick it up (lots of deafening noise + phone
> freezes after a few seconds)
>
> As I said, it was working fine with version from early september. I did not
> find anything useful in the logs or in the config. Has anything changed ?
> What can I do to solve this problem ?
>
> I tested with a Grandstream GXP2000 and a SNOM M3 and only the Grandstream
> has the problem. I suspect a codec issue but what puzzles me is that it was
> previously working and that it completely freezes the phone.
>
> Thanks,
> Sebastien
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From lloyd.aloysius at gmail.com  Mon Dec 13 19:14:52 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 13 Dec 2010 11:14:52 -0500
Subject: [Freeswitch-users] a2billing and freeswitch
In-Reply-To: <AANLkTinu3fVVGb2fB=h_SVsfd06_A9ce1wYHxR0V3jug@mail.gmail.com>
References: <COL122-W4030C97574705F9B9DC69FB5120@phx.gbl>
	<AANLkTinu3fVVGb2fB=h_SVsfd06_A9ce1wYHxR0V3jug@mail.gmail.com>
Message-ID: <AANLkTimeFcw4p=BUvXb965zOrFn1ap9zpiOgAghho6E8@mail.gmail.com>

A2Billing consists of two parts

1. Web Interface
2. Asterisk - PHP-AGI

You can use the Web interface without any problem. But you need rewrite the
PHP-AGI to a FreeSWITCH Application Development Language. I would say LUA

Thanks
Lloyd



2010/12/13 Juan Antonio Iba?ez Santorum <juanito1982 at gmail.com>

> A2billing is Asterisk + PHP AGI so it must work. You could search docs
> about Asterisk + Freeswitch.
>
> Regards
>
> 2010/12/13 Rafqat . <rafonline at hotmail.com>
>
>>
>> Hi
>>
>> I was wondering if a2billing can be setup to work with freeswitch?
>>
>> If so, can anyone please point me to some documentation on how this can be
>> done?
>>
>> Cheers
>>
>> Raf
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From saeedahmad1981 at gmail.com  Mon Dec 13 19:39:41 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Mon, 13 Dec 2010 17:39:41 +0100
Subject: [Freeswitch-users] Scale UP Freeswitch
In-Reply-To: <AANLkTi=YtFWGrP_AYfGOkfeMR6Z0OWMwju2ySTt5y6ua@mail.gmail.com>
References: <AANLkTi=4n_wMSDE7KA9JTDhP+ToGggt-nJzf=EYwTd=6@mail.gmail.com>
	<AANLkTi=N+6=AcELMtbyje3O7W0k7qm3nz8z0CoO3LmLo@mail.gmail.com>
	<AANLkTikkCTMpa-mwD79Fv3W8+1CwS4cMs8Hovdi0sfVS@mail.gmail.com>
	<AANLkTikG=rPb-Lz7rMySa3gzJpjo42ftuthsPReD6ShJ@mail.gmail.com>
	<AANLkTinBf2Zc1o3Vagnha2C1YBba6YemjJjsf-UfCNht@mail.gmail.com>
	<AANLkTikHQtEM1f4sU6Ar0Gd5yeRsor0Gy_KOD7btQYW4@mail.gmail.com>
	<AANLkTimS5ko3Yt155+z-vhpLUA9DG43y2vyUPMOFECj2@mail.gmail.com>
	<AANLkTi=YtFWGrP_AYfGOkfeMR6Z0OWMwju2ySTt5y6ua@mail.gmail.com>
Message-ID: <AANLkTik6icBa9ZTyN=vtEQ-1q8hN-uFSndCup7AH0sWH@mail.gmail.com>

Thanks Steve.

I'll try it but as per other suggestion.. i'll try opensips in front.

Regarding your example below, i have two concerns:

1. on FS2 (media FS) i am using xml_curl to authenticate the customer ip and
then generate the bridge (depends on customer and called number etc..). So
in that case i don't have ACL involved. FS2 also don't deal with sip
registrations etc.. its used just for ip 2 ip communication. So i feel that
if i send x-auth-ip to FS2 i can still use it and can follow my current
implementation with xml_curl, right? but:

2. Even i use FS or opensips to inject xauth ip, and also use it on media FS
to authenticate my original customer.. but what about if someone inject my
real customer ip in xauth ip? that way anyone call send calls, right?



On Mon, Dec 13, 2010 at 10:41 AM, Steven Ayre <steveayre at gmail.com> wrote:

> You can use X-Auth-IP with a FS-FS call too:
>
> Customer --> FS1 --> FS2
> FS1 = front FS
> FS2 = media server
>
> 1. Create a proxy ACL on FS2
> 2. Add the IP of FS1 to that ACL
> 3. On FS1 do this in the dialplan:
>
> <extension ...>
>  <condition ...>
>    <action application="set" data="sip_h_X-Auth-IP=${network_addr}" />
>    <action application="bridge" data="sofia/gateway/fs2/..." />
>  </condition>
> </extension>
>
> FS2 will then be able to use the customer's IP in ACLs, user directory,
> etc.
>
> Remember to either set inbound_bypass_media=true on the sip profile,
> or <action application="set" data="bypass_media=true" /> in dialplan
> before the bridge.
>
> -Steve
>
>
>
> On 12 December 2010 21:32, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> > hmmm... so doing that will also require X-Auth-IP, right or something
> more
> > tricky can be done?
> > On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>
> >> It is, but it relies on the caller supporting 3xx. They might not
> >> handle the redirect.
> >>
> >> A lot won't because you could redirect them to anywhere, so lots of
> >> implementations will ignore the 3xx. FreeSWITCH for instance can
> >> either ignore a 3xx or will send the call back into the dialplan.
> >>
> >> I think you'll have more success having a FS server in front of the
> >> others and bridging the call through to each server. If you set
> >> inbound_bypass_media=true on the SIP profile, the RTP media will
> >> bypass that server and go directly between the caller and the other FS
> >> box. That means that the call won't be using any CPU since it'll only
> >> wake up when a SIP packet is being sent/received. You'll still be
> >> creating a session through so it'll still be allocating memory to the
> >> call, a SIP proxy would use fewer resources.
> >>
> >> -Steve
> >>
> >>
> >> On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981 at gmail.com>
> wrote:
> >> > Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me.
> >> > Since we are talking about HA options... Is it practically doable use
> >> > it:
> >> >
> >> >
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2
> >> > The idea is to run one FS box (Redirect-FS) in front of several FS
> boxes
> >> > which redirect the call to active/available FS. If we make some script
> >> > on
> >> > redirect FS to count the active calls on media FSes and rearrange the
> >> > order
> >> > of redirect then loadbalacing can also be done.
> >> > ...possible?
> >> >
> >> > On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre at gmail.com>
> >> > wrote:
> >> >>
> >> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any
> >> >> > difference between kamailo and opensips?
> >> >>
> >> >> They're both forks of OpenSER so for the most part there's little
> >> >> difference.
> >> >>
> >> >> There are some small differences though since the fork. For example,
> >> >> opensips has a load_balancer module which kamalio does not (kamalio
> >> >> can still do load balancing but has a different interface to do so).
> >> >>
> >> >> > 2. if kamailo or opensips is running in front of FS, then will it
> >> >> > send
> >> >> > call
> >> >> > to FS with original customer ip? so i can do billing etc on FS box
> >> >> > -> actually i do IP based authentication and also ip based billing
> on
> >> >> > FS
> >> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
> >> >> > original
> >> >> > customer overview.
> >> >>
> >> >> It will appear coming from the proxy IP. But there is a workaround.
> >> >> Configure a proxy ACL on the SIP profile and add your proxy IP to it.
> >> >> Then adjust your proxy routing rules so that it adds a X-Auth-IP
> >> >> header that contains the original IP.
> >> >> Anything coming from anything in the proxy ACL is trusted and FS will
> >> >> use the value from X-Auth-IP (if it exists).
> >> >>
> >> >> -Steve
> >> >>
> >> >>
> >> >>
> >> >>
> >> >> On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981 at gmail.com>
> >> >> wrote:
> >> >> > Hi,
> >> >> >
> >> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any
> >> >> > difference between kamailo and opensips?
> >> >> >
> >> >> > 2. if kamailo or opensips is running in front of FS, then will it
> >> >> > send
> >> >> > call
> >> >> > to FS with original customer ip? so i can do billing etc on FS box
> >> >> > -> actually i do IP based authentication and also ip based billing
> on
> >> >> > FS
> >> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose the
> >> >> > original
> >> >> > customer overview.
> >> >> >
> >> >> > thanks
> >> >> > On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >> There are a few performance tweaking tips at
> >> >> >>
> >> >> >>
> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
> >> >> >>
> >> >> >> Yes a Sangoma card will reduce your CPU load since transcoding
> won't
> >> >> >> be done on the CPU any longer, that will then mean there's more
> CPU
> >> >> >> available so you'll be able to handle more calls.
> >> >> >>
> >> >> >> However, if you're looking to increase your number of calls then
> you
> >> >> >> probably want a cluster of servers as Juan pointed out.
> >> >> >>
> >> >> >> It'll mean you can increase the capacity by adding extra servers,
> so
> >> >> >> there'd no longer be a limit to the number of calls you could
> handle
> >> >> >> (just add another server).
> >> >> >>
> >> >> >> It'll also make maintenance easier, as you'll be able to pull a
> >> >> >> server
> >> >> >> from service for updates etc while traffic continues to run on the
> >> >> >> other servers. Maintenance won't mean a service outage.
> >> >> >>
> >> >> >> If you're handling that many calls then additional servers would
> >> >> >> make
> >> >> >> your service more reliable. If a server crashes you'll still have
> >> >> >> the
> >> >> >> calls running on the other servers while you're fixing the problem
> >> >> >> so
> >> >> >> you won't have a complete outage. If FS is behind a load balancer
> >> >> >> then
> >> >> >> your customers might not even notice anything apart from a few
> >> >> >> dropped
> >> >> >> calls.
> >> >> >>
> >> >> >> There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will
> >> >> >> attempt to continue calls if FS crashes and restarts, but I think
> >> >> >> that's only for SIP-SIP not SIP-ISDN.
> >> >> >>
> >> >> >> -Steve
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >> On 7 December 2010 12:26, Stephen Wilde <wstephen80 at gmail.com>
> >> >> >> wrote:
> >> >> >> > Hi,
> >> >> >> > I have one server running Freeswitch with some ISDN connections
> >> >> >> > (via
> >> >> >> > FreeTDM+Sangoma boards) and some SIP connections with service
> >> >> >> > providers
> >> >> >> > and
> >> >> >> > customer.
> >> >> >> > The usage of Freeswitch is as switching so it "bridge" each
> >> >> >> > incoming
> >> >> >> > call to
> >> >> >> > a new outgoing call.
> >> >> >> > SIP calls use G.729 and ISDN calls use ALaw for voice encoding.
> >> >> >> > Now the number of call is grow up and also the CPU load is a
> >> >> >> > little
> >> >> >> > high
> >> >> >> > so
> >> >> >> > I have the necessity to scale UP my Freeswitch to handle more
> >> >> >> > calls:
> >> >> >> > what is
> >> >> >> > the best way to do that?
> >> >> >> > My first idea is to use a Sangoma D500 board to reduce the CPU
> >> >> >> > load.
> >> >> >> > Can
> >> >> >> > be
> >> >> >> > this a solution?
> >> >> >> > There are different way to scale UP?
> >> >> >> > Thanks in advance,
> >> >> >> > Stephen
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mel0torme at gmail.com  Mon Dec 13 19:47:23 2010
From: mel0torme at gmail.com (Tom C)
Date: Mon, 13 Dec 2010 08:47:23 -0800
Subject: [Freeswitch-users] DockStar compile failure, with possible fix
In-Reply-To: <1292249541197-5830855.post@n2.nabble.com>
References: <AANLkTi=-1xKh21fR5PADAQs2YPRKHjssCLQMRWNikSu6@mail.gmail.com>
	<1292249541197-5830855.post@n2.nabble.com>
Message-ID: <AANLkTikWqGB3An3jDJLVXSeiUDWSKNHkJCMD3uHfzo5f@mail.gmail.com>

Yes, I saw that thread when I was researching the problem.

Since parameters are passed in different ways on different platforms, it
makes sense for va_list to be a different memory structure on different
platforms.  So it really is not a compiler bug.  It is a portability issue
in FreeSwitch that va_list is assumed to be a char * in this one location.

Is there something more I should do, to re-open the issue and submit my
suggested fix?



On Mon, Dec 13, 2010 at 6:12 AM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> If you want, please take a look at posts on this
>
> http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-td5252219.html
> thread .
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/DockStar-compile-failure-with-possible-fix-tp5829897p5830855.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Mon Dec 13 20:06:47 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 13 Dec 2010 17:06:47 +0000
Subject: [Freeswitch-users] Scale UP Freeswitch
In-Reply-To: <AANLkTik6icBa9ZTyN=vtEQ-1q8hN-uFSndCup7AH0sWH@mail.gmail.com>
References: <AANLkTi=4n_wMSDE7KA9JTDhP+ToGggt-nJzf=EYwTd=6@mail.gmail.com>
	<AANLkTi=N+6=AcELMtbyje3O7W0k7qm3nz8z0CoO3LmLo@mail.gmail.com>
	<AANLkTikkCTMpa-mwD79Fv3W8+1CwS4cMs8Hovdi0sfVS@mail.gmail.com>
	<AANLkTikG=rPb-Lz7rMySa3gzJpjo42ftuthsPReD6ShJ@mail.gmail.com>
	<AANLkTinBf2Zc1o3Vagnha2C1YBba6YemjJjsf-UfCNht@mail.gmail.com>
	<AANLkTikHQtEM1f4sU6Ar0Gd5yeRsor0Gy_KOD7btQYW4@mail.gmail.com>
	<AANLkTimS5ko3Yt155+z-vhpLUA9DG43y2vyUPMOFECj2@mail.gmail.com>
	<AANLkTi=YtFWGrP_AYfGOkfeMR6Z0OWMwju2ySTt5y6ua@mail.gmail.com>
	<AANLkTik6icBa9ZTyN=vtEQ-1q8hN-uFSndCup7AH0sWH@mail.gmail.com>
Message-ID: <AANLkTi=9DOtEBLcf4yrSODpwKouu3JaAqyC_CnC04eRR@mail.gmail.com>

> 1. on FS2 (media FS) i am using xml_curl to authenticate the customer ip and
> then generate the bridge (depends on customer and called number etc..). So
> in that case i don't have ACL involved. FS2 also don't deal with sip
> registrations etc.. its used just for ip 2 ip communication. So i feel that
> if i send x-auth-ip to FS2 i can still use it and can follow my current
> implementation with xml_curl, right? but:

Yes, you can still use it. You should be able to check both
network_addr (which I believe will store the IP of FS1/OpenSIPS) and
sip_h_X-Auth-IP variables. The sip_h_X-Auth-IP variable will contain
the IP set in X-Auth-IP by FS1/OpenSIPS.

It will be up to you in your application to check that network_addr is
a trusted proxy and check X-Auth-IP against the list of customer IPs.

> 2. Even i use FS or opensips to inject xauth ip, and also use it on media FS
> to authenticate my original customer.. but what about if someone inject my
> real customer ip in xauth ip? that way anyone call send calls, right?

That is why you check the network_addr is coming from a trusted IP
(FS1/OpenSIPS) (which mod_sofia does using the proxy ACL). You only
allow that IP to set X-Auth-IP, and ignore that header if it comes
from any other IP.

- Steve

On 13 December 2010 16:39, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> Thanks Steve.
>
> I'll try it but as per other suggestion.. i'll try opensips in front.
>
> Regarding your example below, i have two concerns:
>
> 1. on FS2 (media FS) i am using xml_curl to authenticate the customer ip and
> then generate the bridge (depends on customer and called number etc..). So
> in that case i don't have ACL involved. FS2 also don't deal with sip
> registrations etc.. its used just for ip 2 ip communication. So i feel that
> if i send x-auth-ip to FS2 i can still use it and can follow my current
> implementation with xml_curl, right? but:
>
> 2. Even i use FS or opensips to inject xauth ip, and also use it on media FS
> to authenticate my original customer.. but what about if someone inject my
> real customer ip in xauth ip? that way anyone call send calls, right?
>
>
>
> On Mon, Dec 13, 2010 at 10:41 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> You can use X-Auth-IP with a FS-FS call too:
>>
>> Customer --> FS1 --> FS2
>> FS1 = front FS
>> FS2 = media server
>>
>> 1. Create a proxy ACL on FS2
>> 2. Add the IP of FS1 to that ACL
>> 3. On FS1 do this in the dialplan:
>>
>> <extension ...>
>> ?<condition ...>
>> ? ?<action application="set" data="sip_h_X-Auth-IP=${network_addr}" />
>> ? ?<action application="bridge" data="sofia/gateway/fs2/..." />
>> ?</condition>
>> </extension>
>>
>> FS2 will then be able to use the customer's IP in ACLs, user directory,
>> etc.
>>
>> Remember to either set inbound_bypass_media=true on the sip profile,
>> or <action application="set" data="bypass_media=true" /> in dialplan
>> before the bridge.
>>
>> -Steve
>>
>>
>>
>> On 12 December 2010 21:32, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
>> > hmmm... so doing that will also require?X-Auth-IP, right or something
>> > more
>> > tricky can be done?
>> > On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <steveayre at gmail.com>
>> > wrote:
>> >>
>> >> It is, but it relies on the caller supporting 3xx. They might not
>> >> handle the redirect.
>> >>
>> >> A lot won't because you could redirect them to anywhere, so lots of
>> >> implementations will ignore the 3xx. FreeSWITCH for instance can
>> >> either ignore a 3xx or will send the call back into the dialplan.
>> >>
>> >> I think you'll have more success having a FS server in front of the
>> >> others and bridging the call through to each server. If you set
>> >> inbound_bypass_media=true on the SIP profile, the RTP media will
>> >> bypass that server and go directly between the caller and the other FS
>> >> box. That means that the call won't be using any CPU since it'll only
>> >> wake up when a SIP packet is being sent/received. You'll still be
>> >> creating a session through so it'll still be allocating memory to the
>> >> call, a SIP proxy would use fewer resources.
>> >>
>> >> -Steve
>> >>
>> >>
>> >> On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981 at gmail.com>
>> >> wrote:
>> >> > Thanks Steve for suggestion, i'll check?X-Auth-IP, its new for me.
>> >> > Since we are talking about HA options... Is it practically doable use
>> >> > it:
>> >> >
>> >> >
>> >> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2
>> >> > The idea is to run one FS box (Redirect-FS) in front of several FS
>> >> > boxes
>> >> > which redirect the call to active/available FS. If we make some
>> >> > script
>> >> > on
>> >> > redirect FS to count the active calls on media FSes and rearrange the
>> >> > order
>> >> > of redirect then loadbalacing can also be done.
>> >> > ...possible?
>> >> >
>> >> > On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any
>> >> >> > difference between kamailo and opensips?
>> >> >>
>> >> >> They're both forks of OpenSER so for the most part there's little
>> >> >> difference.
>> >> >>
>> >> >> There are some small differences though since the fork. For example,
>> >> >> opensips has a load_balancer module which kamalio does not (kamalio
>> >> >> can still do load balancing but has a different interface to do so).
>> >> >>
>> >> >> > 2. if kamailo or opensips is running in front of FS, then will it
>> >> >> > send
>> >> >> > call
>> >> >> > to FS with original customer ip? so i can do billing etc on FS box
>> >> >> > -> actually i do IP based authentication and also ip based billing
>> >> >> > on
>> >> >> > FS
>> >> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose
>> >> >> > the
>> >> >> > original
>> >> >> > customer overview.
>> >> >>
>> >> >> It will appear coming from the proxy IP. But there is a workaround.
>> >> >> Configure a proxy ACL on the SIP profile and add your proxy IP to
>> >> >> it.
>> >> >> Then adjust your proxy routing rules so that it adds a X-Auth-IP
>> >> >> header that contains the original IP.
>> >> >> Anything coming from anything in the proxy ACL is trusted and FS
>> >> >> will
>> >> >> use the value from X-Auth-IP (if it exists).
>> >> >>
>> >> >> -Steve
>> >> >>
>> >> >>
>> >> >>
>> >> >>
>> >> >> On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981 at gmail.com>
>> >> >> wrote:
>> >> >> > Hi,
>> >> >> >
>> >> >> > 1. i am thinking to use kamailo in front of FS boxes, is there any
>> >> >> > difference between kamailo and opensips?
>> >> >> >
>> >> >> > 2. if kamailo or opensips is running in front of FS, then will it
>> >> >> > send
>> >> >> > call
>> >> >> > to FS with original customer ip? so i can do billing etc on FS box
>> >> >> > -> actually i do IP based authentication and also ip based billing
>> >> >> > on
>> >> >> > FS
>> >> >> > box, so in case, i recieve kamailo ip on FS box then i'll loose
>> >> >> > the
>> >> >> > original
>> >> >> > customer overview.
>> >> >> >
>> >> >> > thanks
>> >> >> > On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre at gmail.com>
>> >> >> > wrote:
>> >> >> >>
>> >> >> >> There are a few performance tweaking tips at
>> >> >> >>
>> >> >> >>
>> >> >> >> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
>> >> >> >>
>> >> >> >> Yes a Sangoma card will reduce your CPU load since transcoding
>> >> >> >> won't
>> >> >> >> be done on the CPU any longer, that will then mean there's more
>> >> >> >> CPU
>> >> >> >> available so you'll be able to handle more calls.
>> >> >> >>
>> >> >> >> However, if you're looking to increase your number of calls then
>> >> >> >> you
>> >> >> >> probably want a cluster of servers as Juan pointed out.
>> >> >> >>
>> >> >> >> It'll mean you can increase the capacity by adding extra servers,
>> >> >> >> so
>> >> >> >> there'd no longer be a limit to the number of calls you could
>> >> >> >> handle
>> >> >> >> (just add another server).
>> >> >> >>
>> >> >> >> It'll also make maintenance easier, as you'll be able to pull a
>> >> >> >> server
>> >> >> >> from service for updates etc while traffic continues to run on
>> >> >> >> the
>> >> >> >> other servers. Maintenance won't mean a service outage.
>> >> >> >>
>> >> >> >> If you're handling that many calls then additional servers would
>> >> >> >> make
>> >> >> >> your service more reliable. If a server crashes you'll still have
>> >> >> >> the
>> >> >> >> calls running on the other servers while you're fixing the
>> >> >> >> problem
>> >> >> >> so
>> >> >> >> you won't have a complete outage. If FS is behind a load balancer
>> >> >> >> then
>> >> >> >> your customers might not even notice anything apart from a few
>> >> >> >> dropped
>> >> >> >> calls.
>> >> >> >>
>> >> >> >> There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will
>> >> >> >> attempt to continue calls if FS crashes and restarts, but I think
>> >> >> >> that's only for SIP-SIP not SIP-ISDN.
>> >> >> >>
>> >> >> >> -Steve
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> On 7 December 2010 12:26, Stephen Wilde <wstephen80 at gmail.com>
>> >> >> >> wrote:
>> >> >> >> > Hi,
>> >> >> >> > I have one server running Freeswitch with some ISDN connections
>> >> >> >> > (via
>> >> >> >> > FreeTDM+Sangoma boards) and some SIP connections with service
>> >> >> >> > providers
>> >> >> >> > and
>> >> >> >> > customer.
>> >> >> >> > The usage of Freeswitch is as switching so it "bridge" each
>> >> >> >> > incoming
>> >> >> >> > call to
>> >> >> >> > a new outgoing call.
>> >> >> >> > SIP calls use G.729 and ISDN calls use ALaw for voice encoding.
>> >> >> >> > Now the number of call is grow up and also the CPU load is a
>> >> >> >> > little
>> >> >> >> > high
>> >> >> >> > so
>> >> >> >> > I have the necessity to scale UP my Freeswitch to handle more
>> >> >> >> > calls:
>> >> >> >> > what is
>> >> >> >> > the best way to do that?
>> >> >> >> > My first idea is to use a Sangoma D500 board to reduce the CPU
>> >> >> >> > load.
>> >> >> >> > Can
>> >> >> >> > be
>> >> >> >> > this a solution?
>> >> >> >> > There are different way to scale UP?
>> >> >> >> > Thanks in advance,
>> >> >> >> > Stephen
>> >> >> >> >
>> >> >> >> > _______________________________________________
>> >> >> >> > FreeSWITCH-users mailing list
>> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> > http://www.freeswitch.org
>> >> >> >> >
>> >> >> >> >
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Mon Dec 13 20:30:42 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 13 Dec 2010 11:30:42 -0600
Subject: [Freeswitch-users] DockStar compile failure, with possible fix
In-Reply-To: <AANLkTi=-1xKh21fR5PADAQs2YPRKHjssCLQMRWNikSu6@mail.gmail.com>
References: <AANLkTi=-1xKh21fR5PADAQs2YPRKHjssCLQMRWNikSu6@mail.gmail.com>
Message-ID: <AANLkTikiJ1hoGWSVcL7-dg4At=7SKRy+2hrPSu763StC@mail.gmail.com>

va_copy is not portable either =D welcome to C

I think this code has evolved to the point at which i don't think its
even possible for this to happen anyway.
I am going to try running it without the test at all.

commit dfecc914876b164ce64c53c4f048aa38ed65d9c5
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Mon Dec 13 11:20:12 2010 -0600


On Mon, Dec 13, 2010 at 1:10 AM, Tom C <mel0torme at gmail.com> wrote:
> I was attempting to build FreeSwitch on a Dockstar running Debian Squeeze,
> and ran into?the sofia.c logger()?compile problem as described here:
> http://jira.freeswitch.org/browse/FS-802
>
> The offered solution was to downgrade GCC, but being a Linux newbie, I
> didn't know how to do that. So I decided to figure out what the problem
> really is.
>
> After learning waaaay too much about va_list, I see that the logger()
> function in sofia.c assumes that va_list will always be a pointer.? But on
> different platforms,?va_list can be an array or a struct.? On the Dockstar
> (with Debian Squeeze and GCC 4.4), va_list is apparently a struct.
>
> I made the following modifications to logger() to make it truly portable,
> and it compiles and runs fine on my DockStar, my Debian Lenny x86.? (My
> Windows build is having bigger problems.)? But I'm no C guru, so someone
> needs to thoroughly examine my code before anyone?else takes it as gospel.
> Am I actually checking what needs to be checked with the va_arg() macro?
> Changing it to if(1) worked too, after all.
>
>
> ORIGINAL logger() in sofia.c, to make it easy to see what I changed.
> static void logger(void *logarg, char const *fmt, va_list ap)
> {
> /* gcc 4.4 gets mad at us for testing if (ap) so let's try to work around
> it....*/
> void *ap_ptr = (void *) (intptr_t) ap;? //Error now occurs here, because ap
> is a struct.
> if (!fmt) return;
> if (ap_ptr) {??//Error used to occur here, before attempted fix above.
> ?? switch_log_vprintf(SWITCH_CHANNEL_LOG_CLEAN,
> mod_sofia_globals.tracelevel, fmt, ap);
> } else {
> ???switch_log_printf(SWITCH_CHANNEL_LOG_CLEAN, mod_sofia_globals.tracelevel,
> "%s", fmt);
> }
> }
>
>
> My MODIFIED logger() function, additions and changes labelled with DOCKSTAR:
>
> static
>
> void logger(void *logarg, char const *fmt, va_list ap)
> {
> va_list temp_ap;? //DOCKSTAR: Added line (replacing previous ap_ptr cast).
>
> if (!fmt) return;
>
> va_copy(temp_ap, ap); //DOCKSTAR: Added Line. Make copy of "ap" so va_arg()
> macro?doesn't move pointer.
>
> if (va_arg(temp_ap, int)) {? //DOCKSTAR: Modified, get first argument from
> ap, check non-null.
> ???? switch_log_vprintf(SWITCH_CHANNEL_LOG_CLEAN,
> mod_sofia_globals.tracelevel, fmt, ap);
> } else {
> ???? switch_log_printf(SWITCH_CHANNEL_LOG_CLEAN,
> mod_sofia_globals.tracelevel, "%s", fmt);
> }
>
> va_end(temp_ap);? //DOCKSTAR: Added line.? Release our copy of "ap".
> }
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From mthakershi at gmail.com  Mon Dec 13 20:32:09 2010
From: mthakershi at gmail.com (Malay Thakershi)
Date: Mon, 13 Dec 2010 11:32:09 -0600
Subject: [Freeswitch-users] Cepstral + FS question
In-Reply-To: <AANLkTimt4N9jGQ9aNWGCGYMk3L0pxJWhSOSPzuJ4h=N4@mail.gmail.com>
References: <AANLkTi=N4yG-z5FLUo_hn8YCqDZezSpTcmn74=Qv7c8u@mail.gmail.com>
	<AANLkTimt4N9jGQ9aNWGCGYMk3L0pxJWhSOSPzuJ4h=N4@mail.gmail.com>
Message-ID: <AANLkTi=agVz=fgCv+EN5PT0k5QRq8Cn=7AV2J3ZE70pT@mail.gmail.com>

Thank you for your response.

By licenses, do you mean CPU license or Cepstral ports?

Say if I have 30 simultaneous calls (some part using direct WAV and some
using session speak), what algorithm I use to determine # or ports?

My server configuration:
Windows 2008 server 32-bit
4 GB RAM
Intel Xeon CPU X3220 @ 2.40GHz 2.40GHz

Malay


On Mon, Dec 13, 2010 at 8:55 AM, David Brazier <davidjbrazier at gmail.com>wrote:

> On Tue, Dec 7, 2010 at 12:38 AM, Malay Thakershi <mthakershi at gmail.com>
> wrote:
> > Hello, it would be great help if someone who has used Cepstral from FS
> can
> > share their views.
> > 1. I just have one Cepstral Allison voice license (1 port) on my FS
> server.
> > I use swift command to convert text files to WAV which are then played by
> > mod_managed in FS call process. First question is regarding limitations
> on
> > simultaneous conversions (TXT to WAV) using swift command. If I have
> > multiple threads doing this, will there be any degradation because of
> > Cepstral? I tried running two BAT files with 3 commands each. But running
> > them simultaneously or separate produced same outcome. Does anyone know
> when
> > Cepstral licensing kicks in and starts degrading quality ( or worse
> > inserting "not licensed") prompt?
>
> The synthesis via the swift command is much faster than real time
> (i.e. the time it takes to play the WAV) and is only restricted by CPU
> speed and licenses.  You'd need to ask Cepstral the details, but my
> simple tests have shown that it only uses one CPU per command though
> some of the processing of multiple command is on multiple CPUs.  But I
> think the licensing mechanism prevents multiple commands running
> completely simultaneously on multiple CPUs.  I think the "not
> licensed" speech is only inserted when you have no license at all -
> all that happens if you try to run multiple commands is that they are
> just delayed until  there is a free license.  I don't think quality
> will ever be degraded - it's just a question of CPUs and licenses that
> will determine your throughput.
>
> > 2. When I call session speak from mod_managed (or stream file) after
> > selecting Allison / Cepstral as my voice, does Cepstral engine interfere
> > with quality of the playback? If yes, when will I see it and how can I
> > produce their effects?
>
> Not unless something in Cepstral or FS is going badly wrong!
>
> > 3. What is the sensible number of ports (from Cepstral) I should be
> prepared
> > to buy if findings in the previous points imposes significant
> limitations?
>
> Depends on the length of your prompts and number and speed of CPUs and
> anticipated simultaneous calls.  Try testing the time for a typical
> prompt and work it out from there.
>
> David
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From gustavo.espeche at upper-soft.com  Mon Dec 13 20:32:31 2010
From: gustavo.espeche at upper-soft.com (Gustavo Espeche)
Date: Mon, 13 Dec 2010 14:32:31 -0300
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 54, Issue 87
In-Reply-To: <mailman.470.1292251727.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.470.1292251727.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <1292261551.2202.5.camel@gustavo-laptop>

Hi i try to start mod_h323 but i have follow issue.

2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting on socket
accept on ip$10.0.0.1:1720
2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to open
transport, connection not started.
On Mon, 2010-12-13 at 17:48 +0300,
freeswitch-users-request at lists.freeswitch.org wrote:
> Send FreeSWITCH-users mailing list submissions to
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> 
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> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
> Today's Topics:
> 
>    1. Re: opal to sip problem (Nikolay Kondratyev)
> MHTML Document attachment
> > -------- Forwarded Message --------
> > From: Nikolay Kondratyev <kond at nstel.ru>
> > Reply-to: FreeSWITCH Users Help
> > <freeswitch-users at lists.freeswitch.org>
> > To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
> > Subject: Re: [Freeswitch-users] opal to sip problem
> > Date: Mon, 13 Dec 2010 17:48:08 +0300
> > 
> > I can't help you with mod_opal, but I use FS with mod_h323 ad sip<->h.323
> > gateway successfully.
> > Rgds,
> > Nikolay.
> > 
> > > -----Original Message-----
> > > From: freeswitch-users-bounces at lists.freeswitch.org 
> > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On 
> > > Behalf Of Gustavo Espeche
> > > Sent: Monday, December 13, 2010 5:24 PM
> > > To: freeswitch-users at lists.freeswitch.org
> > > Subject: [Freeswitch-users] opal to sip problem
> > > 
> > > Hello list, we are try to do inter-worker between h323 and sip using
> > > opal in freeswitch, but FS don't send the call to our sip gw, 
> > > follow is
> > > the  call flow:
> > > 
> > > h323 endpoint -->FS->sip gateway
> > > 
> > > attached a FS debug of call.
> > > 
> > > if something know or work with opal in freeSwitch and have 
> > > some tips we
> > > appreciate a lot if advice about it.
> > > Best Regards.
> > > --------------------------------------------------------------
> > > ---------------------------------------------------------------
> > > 2010-12-13 11:58:21.325242 [DEBUG] transports.cxx:1200 Started
> > > connection to 200.117.192.17:19270 (if=10.0.0.1:1720)
> > > 2010-12-13 11:58:21.325242 [DEBUG] osutil.cxx:189 File handle 
> > > high water
> > > mark set: 67 Thread unblock pipe
> > > 2010-12-13 11:58:21.325242 [DEBUG] tlibthrd.cxx:587 Thread high water
> > > mark set: 8
> > > 2010-12-13 11:58:21.325242 [DEBUG] transports.cxx:666 Waiting 
> > > on socket
> > > accept on tcp$10.0.0.1:1720
> > > 2010-12-13 11:58:21.325242 [DEBUG] h323ep.cxx:501 Awaiting first PDU
> > > 2010-12-13 11:58:21.805203 [DEBUG] h323pdu.cxx:80 Receiving PDU:
> > >   {
> > >     q931pdu = {
> > >       protocolDiscriminator = 8
> > >       callReference = 22716
> > >       from = originator
> > >       messageType = Setup
> > >       IE: Bearer-Capability = {
> > >         88 93 a5                            
> > >                ...
> > >       }
> > >       IE: Display = {
> > >         67 75 73 74 61 76 6f 00                            gustavo.
> > >       }
> > >       IE: Called-Party-Number = {
> > >         81 32 33 31 35 34 33 35  31 34 32 38 30 36 33
> > > 33   .231543514280633
> > >       }
> > >       
> > > IE: User-User = {
> > >         20 b8 06 00 08 91 4a 00  06 01 40 06 00 67 00
> > > 75    .....J... at ..g.u
> > >         00 73 00 74 00 61 00 76  00 6f 22 c0 09 00 00
> > > 3d   .s.t.a.v.o"....=
> > >         ...
> > >       }
> > >     }
> > >     h225pdu = {
> > >       h323_uu_pdu = {
> > >         h323_me
> > > ssage_body = setup {
> > >           protocolIdentifier = 0.0.8.2250.0.6
> > >           sourceAddress = 1 entries {
> > >             [0]=h323_ID  7 characters {
> > >               0067 0075 0073 0074 0061 0076 006f        gustavo
> > >             }
> > >           }
> > >           sour
> > > ceInfo = {
> > >             vendor = {
> > >               vendor = {
> > >                 t35CountryCode = 9
> > >                 t35Extension = 0
> > >                 manufacturerCode = 61
> > >               }
> > >               productId =  3 octets {
> > >                 00 00 00      
> > > (OPAL v3.6
> > >                 2e 36 29 00 00                                     .6
> > > )..
> > >               }
> > >             }
> > >             terminal = {
> > >             }
> > >             mc = false
> > >             undefinedNode = false
> > >           }
> > >           destinationAddress = 1 entries {
> > >             [0]=dialedDigits "231543514280633"
> > >           }
> > >      
> > > Log-Func: 
> > > Log-Line: 0
> > > User-Data: 
> > > 
> > >      destCallSignalAddress = ipAddress {
> > >             ip =  4 octets {
> > >               48 33 2f 64                                        H3/d
> > >             }
> > >             port = 1720
> > >           }
> > >           activeMC = falets {
> > >             4e 76 b2 f2 1d 05 e0 11  9b b7 00 1e 64 1f 33 fe
> > > Nv..........d.3.
> > >           }
> > >           conferenceGoal = create <<null>>
> > >           callType = pointToPoint <<null>>
> > >           sourceCallSignalAddress = ipAddress {
> > >             ip = 
> > > Log-Func: 
> > > Log-Line: 0
> > > User-Data: 
> > > 
> > >  4 octets {
> > >               c8 75 c0 11                                        .u..
> > >             }
> > >             port = 19270
> > >           }
> > >           callIdentifier = {
> > >             guid =  16 octets {
> > >               3a 76 f 33 fe   :v..........d.3.
> > >             }
> > >           }
> > >           fastStart = 8 entries {
> > >             [0]= 29 octets {
> > >               40 00 00 06 04 01 00 4c  20 13 80 11 1c 00 01 00
> > > @......L .......
> > >               c8 75 c0 11 13 de 00 c8  75 c0 11 13 
> > > df            .u......u....
> > >             }
> > >             [1]= 19 octets {
> > >               00 00 64 0c 20 13 80 0b  0d 00 01 00 c8 75 c0
> > > 11   ..d. ........u..
> > >               13 df 00                                           ...
> > >             }
> > >             [
> > > 2]= 35 octets {
> > >               40 00 00 06 04 01 00 48  78 00 4a ff 00 80 01 00
> > > @......Hx.J.....
> > >               80 11 1c 00 02 00 c8 75  c0 11 13 e0 00 c8 75
> > > c0   .......u......u.
> > >               ...
> > >             }
> > >             [3]= 25 octets {
> > >         
> > >       00 00 65 08 78 00 4a ff  00 80 01 00 80 0b 0d
> > > 00   ..e.x.J.........
> > >               02 00 c8 75 c0 11 13 e1
> > > 00                        ...u.....
> > >             }
> > >             [4]= 34 octets {
> > >               40 00 00 06 04 01 00 48  68 4a ff 00 80 01 
> > > 00 80   @......HhJ......
> > >               11 1c 00 02 00 c8 75 c0  11 13 e0 00 c8 75 c0
> > > 11   ......u......u..
> > >               ...
> > >             }
> > >             [5]= 24 octets {
> > >               00 00 66 08 68 4a ff 00  80 01 00 80 0b 0d 00
> > > 02   ..f.hJ..........
> > > 
> > >               00 c8 75 c0 11 13 e1
> > > 00                            ..u.....
> > >             }
> > >             [6]= 34 octets {
> > >               40 00 00 06 04 01 00 48  70 4a ff 00 80 01 00 80
> > > @......HpJ......
> > >               11 1c 00 02 00 c8 75 c0  11 13 e0 0
> > > 0 c8 75 c0 11   ......u......u..
> > >               ...
> > >             }
> > >             [7]= 24 octets {
> > >               00 00 67 08 70 4a ff 00  80 01 00 80 0b 0d 00
> > > 02   ..g.pJ..........
> > >               00 c8 75 c0 11 13 e1
> > > 00                            ..u.....
> > > 
> > >             }
> > >           }
> > >           mediaWaitForConnect = false
> > >           canOverlapSend = false
> > >           multipleCalls = false
> > >           maintainConnection = false
> > >           parallelH245Control = 2 entries {
> > >             [0]= 125 octets {
> > >         
> > >       02 70 01 06 00 08 81 75  00 0d 80 13 80 01 f4
> > > 00   .p.....u........
> > >               01 00 00 01 00 00 01 00  00 0c c0 01 00 01 80
> > > 07   ................
> > >               ...
> > >             }
> > >             [1]= 7 octets {
> > >               01 00 32 80 5d 41 2
> > > b                               ..2.]A+
> > >             }
> > >           }
> > >         }
> > >         h245Tunneling = true
> > >       }
> > >     }
> > >   }
> > > 2010-12-13 11:58:21.805203 [DEBUG] h323ep.cxx:510 Incoming call, first
> > > PDU: callReference=22716
> > > 2010-12-13 11:58:21.805203 [DEBUG] call.cxx:72 Created 
> > > Call[Cc70e3cf91]
> > > 2010-12-13 11:58:21.805203 [DEBUG] connection.cxx:262 Created 
> > > connection
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:21.805203 [DEBUG] rfc2833.cxx:140 Handler created
> > > 2010-12-13 11:58:21.805203 [DEBUG] rfc2833.cxx:328 Rx 
> > > capability set to
> > > "0-16,32,36"
> > > 2010-12-13 11:58:21.805203 [DEBUG] rfc2833.cxx:140 Handler created
> > > 2010-12-13 11:58:21.805203 [DEBUG] rfc2833.cxx:328 Rx 
> > > capability set to
> > > "192,193"
> > > 2010-12-13 11:58:21.805203 [DEBUG] osutil.cxx:189 File handle 
> > > high water
> > > mark set: 68 PUDPSocket
> > > 2010-12-13 11:58:21.808186 [DEBUG] h4601.cxx:1583 Endpoint Attached
> > > 2010-12-13 11:58:21.808186 [DEBUG] h323ep.cxx:552 Created new
> > > connection: tcp$200.117.192.17:19270/22716
> > > 2010-12-13 11:58:21.808186 [DEBUG] h323.cxx:559 Handling PDU: Setup
> > > callRef=22716
> > > 2010-12-13 11:58:21.808186 [DEBUG] connection.cxx:1516 SetPhase from
> > > UninitialisedPhase to SetUpPhase for Call[Cc70e3cf91]-EP<h323>[tcp
> > > $200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:21.808186 [DEBUG] h323.cxx:943 Set protocol 
> > > version to
> > > 6 and implying H.245 version 13
> > > 2010-12-13 11:58:21.808186 [DEBUG] h323.cxx:1240 Set remote 
> > > application
> > > name: " 3.2.6 (OPAL v3.6.6) 9/61 "
> > > 2010-12-13 11:58:21.808186 [DEBUG] manager.cxx:1392 Checking incoming
> > > call for NAT: local=72.51.47.100, peer=200.117.192.17,
> > > sig=200.117.192.17
> > > 2010-12-13 11:58:21.808186 [DEBUG] h323.cxx:1041 Sending call 
> > > proceeding
> > > PDU
> > > 2010-12-13 11:58:21.811196 [DEBUG] h323pdu.cxx:80 Sending PDU:
> > >   {
> > >     q931pdu = {
> > >       protocolDiscriminator = 8
> > >       callReference = 22716
> > >       from = destination
> > >       messageType = CallProceeding
> > >       IE: Display = {
> > >         72 6f 6f 74 00                        
> > >              root.
> > >       }
> > >       IE: User-User = {
> > >         21 80 06 00 08 91 4a 00  06 22 c0 09 00 00 3d
> > > 1e   !.....J.."....=.
> > >         56 6f 78 20 4c 75 63 69  64 61 20 50 74 79 2e 20   Vox Lucida
> > > Pty. 
> > >         ...
> > >       }
> > >     }
> > >     h225pdu = {
> > >     
> > >   h323_uu_pdu = {
> > >         h323_message_body = callProceeding {
> > >           protocolIdentifier = 0.0.8.2250.0.6
> > >           destinationInfo = {
> > >             vendor = {
> > >               vendor = {
> > >                 t35CountryCode = 9
> > >                 t35Extensio
> > > n = 0
> > >                 manufacturerCode = 61
> > >               }
> > >               productId =  31 octets {
> > >                 56 6f 78 20 4c 75 63 69  64 61 20 50 74 79 2e 20   Vox
> > > Lucida Pty. 
> > >                 4c 74 64 2e 20 6d 6f 64  5f 6f 70 61 6c 00 00     
> > > Log-Line: 0
> > > UINCOMING DATA [(null)]
> > > 
> > > RECV EVENT
> > > Event-Name: SOCKET_DATA
> > > ser-Data: _undef_
> > > 
> > > 
> > >  }
> > >             }
> > >             terminal = {
> > >             }
> > >             mc = false
> > >             undefinedNode = false
> > >           }
> > >           callIdentifier = {
> > >             guid =  16 octets {
> > >               3a 76 b2 f2 1d 05 e0 11  9b b7 00 1e 64 1f 33 fe 
> > >   :v..........d.3.
> > >             }
> > >           }
> > >           multipleCalls = false
> > >           maintainConnection = false
> > >         }
> > >         h245Tunneling = true
> > >       }
> > >     }
> > >   }
> > > 2010-12-13 11:58:21.811196 [DEBUG] manager.cxx:600 OnIncoming 
> > > connection
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:21.811196 [DEBUG] call.cxx:288 
> > > GetOtherPartyConnection
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:21.811196 [DEBUG] manager.cxx:1200 Searching 
> > > for route
> > > "h323:root 231543514280633"
> > > 2010-12-13 11:58:21.811196 [DEBUG] manager.cxx:1228 Matched regex
> > > "h323:.*"
> > > 2010-12-13 11:58:21.811196 [DEBUG] manager.cxx:573 Set up 
> > > connection to
> > > "local:231543514280633"
> > > 2010-12-13 11:58:21.811196 [DEBUG] connection.cxx:262 Created 
> > > connection
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.811196 [DEBUG] localep.cxx:205 Created connection
> > > with token "Ldc8d6c462"
> > > 2010-12-13 11:58:21.811196 [DEBUG] h323.cxx:1075 Incoming 
> > > call accepted
> > > 2010-12-13 11:58:21.811196 [DEBUG] mod_opal.cpp:770 Matched FS codec
> > > PCMA to OPAL media format G.711-ALaw-64k
> > > 2010-12-13 11:58:21.811196 [DEBUG] mod_opal.cpp:770 Matched FS codec
> > > G729 to OPAL media format G.729
> > > 2010-12-13 11:58:21.811196 [DEBUG] mod_opal.cpp:770 Matched FS codec
> > > PCMU to OPAL media format G.711-uLaw-64k
> > > 2010-12-13 11:58:21.811196 [DEBUG] mod_opal.cpp:770 Matched FS codec
> > > G723 to OPAL media format G.723.1
> > > 2010-12-13 11:58:21.814188 [DEBUG] call.cxx:425 GetMediaFormats for
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> > > G.723.1
> > > G.729
> > > G.711-uLaw-64k
> > > G.711-ALaw-64k
> > > 
> > > 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2381 Could not find
> > > capability: "G.723.1"
> > > 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
> > > G.723.1 <1>
> > > 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2381 Could not find
> > > capability: "G.729"
> > > 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
> > > G.729 <2>
> > > 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2381 Could not find
> > > capability: "G.711-uLaw-64k"
> > > 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
> > > G.711-uLaw-64k <3>
> > > 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2381 Could not find
> > > capability: "G.711-ALaw-64k"
> > > 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
> > > G.711-ALaw-64k <4>
> > > 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2376 Found capability:
> > > G.723.1 <1>
> > > 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2376 Found capability:
> > > G.729 <2>
> > > 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2376 Found capability:
> > > G.711-uLaw-64k <3>
> > > 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2376 Found capability:
> > > G.711-ALaw-64k <4>
> > > 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
> > > UserInput/hookflash <5>
> > > 2010-12-13 11:58:21.814188 [DEBUG] h323caps.cxx:2284 Added capability:
> > > UserInput/basicString <6>
> > > 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2284 Added capability:
> > > UserInput/dtmf <7>
> > > 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2284 Added capability:
> > > UserInput/RFC2833 <8>
> > > 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2376 Found capability:
> > > UserInput/RFC2833 <8>
> > > 2010-12-13 11:58:21.817197 [DEBUG] call.cxx:288 
> > > GetOtherPartyConnection
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:21.817197 [DEBUG] h323.cxx:3704 SetLocalCapabilities:
> > >  Table:
> > >    G.723.1 <1>
> > >    G.729 <2>
> > >    G.711-uLaw-64k <3>
> > >    G.711-ALaw-64k <4>
> > >    UserInput/hookflash <5>
> > >    UserInput/basicString <6>
> > >    UserInput/dtmf <7>
> > >    UserInput/RFC2833 <8>
> > >  Set:
> > >    0:
> > >   
> > > 
> > > Log-Func: 
> > > Log-Line: 0
> > > User-Data: 
> > > 
> > >    0:
> > >        G.723.1 <1>
> > >        G.729 <2>
> > >        G.711-uLaw-64k <3>
> > >        G.711-ALaw-64k <4>
> > >      1:
> > >        UserInput/hookflash <5>
> > >      2:
> > >        UserInput/basicString <6>
> > >        UserInput/dtINCOMING DATA [(null)]
> > > 
> > > RECV EVENT
> > > Event-Name: SOCKET_DATA
> > > 
> > > 
> > > 2010-12-13 11:58:21.817197 [DEBUG] h323.cxx:1121 Fast start detected
> > > 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2594 Could not find
> > > capability: audioData, type g711Alaw64k
> > > 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:845 
> > > Capability tx frames
> > > left at 20 as remote allows 20
> > > 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2294 Added capability:
> > > G.711-ALaw-64k <4>
> > > 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:845 
> > > Capability tx frames
> > > left at 20 as remote allows 20
> > > 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2594 Could not find
> > > capability: videoData, type h261VideoCapability
> > > 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2594 Could not find
> > > capability: videoData, type h261VideoCapability
> > > 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2594 Could not find
> > > capability: videoData, type h261VideoCapability
> > > 2010-12-13 11:58:21.817197 [DEBUG] h323caps.cxx:2594 Could not find
> > > capability: videoData, type h261VideoCapability
> > > 2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
> > > capability: videoData, type h261VideoCapability
> > > 2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
> > > capability: videoData, type h261VideoCapability
> > > 2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
> > > capability: videoData, type h261VideoCapability
> > > 2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
> > > capability: videoData, type h261VideoCapability
> > > 2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
> > > capability: videoData, type h261VideoCapability
> > > 2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
> > > capability: videoData, type h261VideoCapability
> > > 2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
> > > capability: videoData, type h261VideoCapability
> > > 2010-12-13 11:58:21.820187 [DEBUG] h323caps.cxx:2594 Could not find
> > > capability: videoData, type h261VideoCapability
> > > 2010-12-13 11:58:21.820187 [DEBUG] call.cxx:288 
> > > GetOtherPartyConnection
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.820187 [DEBUG] call.cxx:169 OnSetUp
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:21.820187 [DEBUG] localep.cxx:240 Incoming call from
> > > gustavo [200.117.192.17]
> > > 2010-12-13 11:58:21.820187 [DEBUG] mod_opal.cpp:641 Created switch
> > > caller profile:
> > >   username       = 
> > >   dialplan       = XML
> > >   caller_id_name     = gustavo [200.117.192.17]
> > >   caller_id_number   = 0000000000
> > >   network_addr   = 
> > >   source         = opal
> > >   context       
> > > Log-Func: 
> > > Log-Line: 0
> > > User-Data: 
> > > 
> > >  = default
> > >   2010-12-13 11:58:21.823196 [NOTICE] switch_channel.c:784 New Channel
> > > opal/in:231543514280633 [7b4a89ac-b5d7-4fb4-bab5-ddaa8e73485c]
> > > 2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:647
> > > (opal/in:231543514280633) State Change CS_NEW -> CS_INIT
> > > 2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:1092 State changed on
> > > connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.823196 [DEBUG] switch_core_session.c:1083 Send
> > > signal opal/in:231543514280633 [BREAK]
> > > 2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:931 Kill 3 on 
> > > connection
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.823196 [DEBUG] connection.cxx:1516 SetPhase from
> > > UninitialisedPhase to AlertingPhase for
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.823196 [NOTICE] mod_opal.cpp:676 Ring-Ready
> > > opal/in:231543514280633!
> > > 2010-12-13 11:58:21.823196 [DEBUG] manager.cxx:678 OnAlerting
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.823196 [DEBUG] call.cxx:196 OnAlerting
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.823196 [DEBUG] h323.cxx:2067 SetAlerting
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:320
> > > (opal/in:231543514280633) Running State Change CS_INIT
> > > 2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:356
> > > (opal/in:231543514280633) State INIT
> > > 2010-12-13 11:58:21.823196 [DEBUG] osutil.cxx:189 File handle 
> > > high water
> > > mark set: 72 Thread unblock pipe
> > > 2010-12-13 11:58:21.823196 [DEBUG] tlibthrd.cxx:587 Thread high water
> > > mark set: 10
> > > 2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:842 Started 
> > > routing for
> > > connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:843
> > > (opal/in:231543514280633) State Change CS_INIT -> CS_ROUTING
> > > 2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:1092 State changed on
> > > connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.823196 [DEBUG] switch_core_session.c:1083 Send
> > > signal opal/in:231543514280633 [BREAK]
> > > 2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:931 Kill 3 on 
> > > connection
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:356
> > > (opal/in:231543514280633) State INIT going to sleep
> > > 2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:320
> > > (opal/in:231543514280633) Running State Change CS_ROUTING
> > > 2010-12-13 11:58:21.823196 [DEBUG] switch_channel.c:1615
> > > (opal/in:231543514280633) Callstate Change DOWN -> RINGING
> > > 2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:359
> > > (opal/in:231543514280633) State ROUTING
> > > 2010-12-13 11:58:21.823196 [DEBUG] mod_opal.cpp:850 Routing connection
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.823196 [DEBUG] switch_core_state_machine.c:77
> > > opal/in:231543514280633 Standard ROUTING
> > > 2010-12-13 11:58:21.823196 [INFO] mod_dialplan_xml.c:331 Processing
> > > gustavo [200.117.192.17] <0000000000>->231543514280633 in context
> > > default
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323pdu.cxx:80 Receiving PDU:
> > >   request terminalCapabilitySet {
> > >     sequenceNumber = 1
> > >     protocolIdentifier = 0.0.8.245.0.13
> > >     multiplexCapability = h2250Capability {
> > >       maximumAudioDelayJitter = 500
> > >       receiveMultipointCapabi
> > > lity = {
> > >         multicastCapability = false
> > >         multiUniCastConference = false
> > >         mediaDistributionCapability = 1 entries {
> > >           [0]={
> > >             centralizedControl = false
> > >             distributedControl = false
> > >             centraliz
> > > edAudio = false
> > >             distributedAudio = false
> > >             centralizedVideo = false
> > >             distributedVideo = false
> > >           }
> > >         }
> > >       }
> > >       transmitMultipointCapability = {
> > >         multicastCapability = false
> > >         multiUniC
> > > astConference = false
> > >         mediaDistributionCapability = 1 entries {
> > >           [0]={
> > >             centralizedControl = false
> > >             distributedControl = false
> > >             centralizedAudio = false
> > >             distributedAudio = false
> > >          
> > >    centralizedVideo = false
> > >             distributedVideo = false
> > >           }
> > >         }
> > >       }
> > >       receiveAndTransmitMultipointCapability = {
> > >         multicastCapability = false
> > >         multiUniCastConference = false
> > >         mediaDistributionCapab
> > > ility = 1 entries {
> > >           [0]={
> > >             centralizedControl = false
> > >             distributedControl = false
> > >             centralizedAudio = false
> > >             distributedAudio = false
> > >             centralizedVideo = false
> > >             distributedV
> > > ideo = false
> > >           }
> > >         }
> > >       }
> > >       mcCapability = {
> > >         centralizedConferenceMC = false
> > >         decentralizedConferenceMC = false
> > >       }
> > >       rtcpVideoControlCapability = false
> > >       mediaPacketizationCapability = {
> > >         h261a
> > > VideoPacketization = false
> > >       }
> > >       logicalChannelSwitchingCapability = false
> > >       t120DynamicPortCapability = true
> > >     }
> > >     capabilityTable = 8 entries {
> > >       [0]={
> > >         capabilityTableEntryNumber = 1
> > >         capability = receiveAudioCap
> > > ability g711Alaw64k 240
> > >       }
> > >       [1]={
> > >         capabilityTableEntryNumber = 2
> > >         capability = receiveVideoCapability h261VideoCapability {
> > >           qcifMPI = 1
> > >           cifMPI = 1
> > >           temporalSpatialTradeOffCapability = false
> > >      
> > >      maxBitRate = 19200
> > >           stillImageTransmission = false
> > >           videoBadMBsCap = false
> > >         }
> > >       }
> > >       [2]={
> > >         capabilityTableEntryNumber = 3
> > >         capability = receiveVideoCapability h261VideoCapability {
> > >           cifMPI
> > >  = 1
> > >           temporalSpatialTradeOffCapability = false
> > >           maxBitRate = 19200
> > >           stillImageTransmission = false
> > >           videoBadMBsCap = false
> > >         }
> > >       }
> > >       [3]={
> > >         capabilityTableEntryNumber = 4
> > >         capability =
> > >  receiveVideoCapability h261VideoCapability {
> > >           qcifMPI = 1
> > >           temporalSpatialTradeOffCapability = false
> > >           maxBitRate = 19200
> > >           stillImageTransmission = false
> > >           videoBadMBsCap = false
> > >         }
> > >       }
> > >       [4
> > > ]={
> > >         capabilityTableEntryNumber = 5
> > >         capability = receiveUserInputCapability hookflash <<null>>
> > >       }
> > >       [5]={
> > >         capabilityTableEntryNumber = 6
> > >         capability = receiveUserInputCapability basicString <<null>>
> > >       }
> > >    
> > >    [6]={
> > >         capabilityTableEntryNumber = 7
> > >         capability = receiveUserInputCapability dtmf <<null>>
> > >       }
> > >       [7]={
> > >         capabilityTableEntryNumber = 8
> > >         capability = receiveRTPAudioTelephonyEventCapability {
> > >           dynamic
> > > RTPPayloadType = 101
> > >           audioTelephoneEvent = "0-16"
> > >         }
> > >       }
> > >     }
> > >     capabilityDescriptors = 1 entries {
> > >       [0]={
> > >         capabilityDescriptorNumber = 1
> > >         simultaneousCapabilities = 4 entries {
> > >           [0]=1 entries {
> > >  
> > >            [0]=1
> > >           }
> > >           [1]=3 entries {
> > >             [0]=2
> > >             [1]=3
> > >             [2]=4
> > >           }
> > >           [2]=1 entries {
> > >             [0]=5
> > >           }
> > >           [3]=3 entries {
> > >             [0]=6
> > >             [1]=7
> > >           
> > > Log-Func: 
> > > Log-Line: 0
> > > User-Data: 
> > > 
> > >   [2]=8
> > >     2010-12-13 11:58:21.826206 [DEBUG] h323.cxx:2792 Set protocol
> > > version to 13
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323neg.cxx:378 Received
> > > TerminalCapabilitySet: state=Idle pduSeq=1 inSeq=4294967295
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:1988
> > > H323Capabilities(ctor)
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
> > > G.723.1 <1>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
> > > G.729 <2>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
> > > G.711-uLaw-64k <3>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
> > > G.711-ALaw-64k <4>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
> > > UserInput/hookflash <5>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
> > > UserInput/basicString <6>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
> > > UserInput/dtmf <7>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2294 Added capability:
> > > UserInput/RFC2833 <8>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
> > > G.723.1 <1>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
> > > G.729 <2>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
> > > G.711-uLaw-64k <3>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
> > > G.711-ALaw-64k <4>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
> > > UserInput/hookflash <5>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
> > > UserInput/basicString <6>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
> > > UserInput/dtmf <7>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2355 Found capability:
> > > UserInput/RFC2833 <8>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2376 Found capability:
> > > G.711-ALaw-64k <4>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2376 Found capability:
> > > G.711-uLaw-64k <3>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2376 Found capability:
> > > G.723.1 <1>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
> > > capability: "G.723.1(5.3k)"
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
> > > G.723.1 <9>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
> > > capability: "G.723.1A(5.3k)"
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
> > > G.723.1 <10>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
> > > capability: "G.723.1A(6.3k)"
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
> > > G.723.1 <11>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
> > > capability: "G.728"
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
> > > G.728 <12>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2376 Found capability:
> > > G.729 <2>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
> > > capability: "G.729A"
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
> > > G.729A <13>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
> > > capability: "G.729A/B"
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
> > > G.729A/B <14>
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2381 Could not find
> > > capability: "G.729B"
> > > 2010-12-13 11:58:21.826206 [DEBUG] h323caps.cxx:2284 Added capability:
> > > G.729B <15>
> > > 2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2381 Could not find
> > > capability: "GSM-06.10"
> > > 2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2284 Added capability:
> > > GSM-06.10 <16>
> > > 2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2381 Could not find
> > > capability: "GSM-AMR"
> > > 2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2284 Added capability:
> > > GSM-AMR <17>
> > > 2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2381 Could not find
> > > capability: "T.38"
> > > 2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2284 Added capability:
> > > T.38 <18>
> > > 2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2376 Found capability:
> > > UserInput/RFC2833 <8>
> > > 2010-12-13 11:58:21.831146 [DEBUG] h323caps.cxx:2376 Found capability:
> > > UserInput/basicString <6>
> > > 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2376 Found capability:
> > > UserInput/dtmf <7>
> > > 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2381 Could not find
> > > capability: "UserInput/generalString"
> > > 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
> > > UserInput/generalString <19>
> > > 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2376 Found capability:
> > > UserInput/hookflash <5>
> > > 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2381 Could not find
> > > capability: "UserInput/iA5String"
> > > 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
> > > UserInput/iA5String <20>
> > > 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2381 Could not find
> > > capability: "iLBC"
> > > 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
> > > iLBC <21>
> > > 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
> > > UserInput/hookflash <22>
> > > 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
> > > UserInput/basicString <23>
> > > 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
> > > UserInput/dtmf <24>
> > > 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
> > > UserInput/RFC2833 <25>
> > > 2010-12-13 11:58:21.834192 [CONSOLE] mod_xml_curl.c:312 XML 
> > > response is
> > > in /tmp/90f1dd7d-a63c-4e15-981c-6eed9e70313b.tmp.xml
> > > Dialplan: opal/in:231543514280633 parsing [default->external]
> > > continue=false
> > > Dialplan: opal/in:231543514280633 Regex (PASS) [external]
> > > destination_number(231543514280633) =~ /^231543514280633/ 
> > > break=on-false
> > > Dialplan: opal/in:231543514280633 Action set(continue_on_fail=true) 
> > > Dialplan: opal/in:231543514280633 Action 
> > > set(hangup_after_bridge=true) 
> > > Dialplan: opal/in:231543514280633 Action set(progress_timeout=15) 
> > > Dialplan: opal/in:231543514280633 Action set(proxy_media=false) 
> > > Dialplan: opal/in:231543514280633 Action set(bypass_media=true) 
> > > Dialplan: opal/in:231543514280633 Action
> > > set(absolute_codec_string=PCMA) 
> > > Dialplan: opal/in:231543514280633 Action
> > > bridge(sofia/external/21543514280633 at 200.35.145.149) 
> > > 2010-12-13 11:58:21.834192 [DEBUG] switch_core_state_machine.c:119
> > > (opal/in:231543514280633) State Change CS_ROUTING -> CS_EXECUTE
> > > 2010-12-13 11:58:21.834192 [DEBUG] mod_opal.cpp:1092 State changed on
> > > connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.834192 [DEBUG] switch_core_session.c:1083 Send
> > > signal opal/in:231543514280633 [BREAK]
> > > 2010-12-13 11:58:21.834192 [DEBUG] mod_opal.cpp:931 Kill 3 on 
> > > connection
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.834192 [DEBUG] switch_core_state_machine.c:359
> > > (opal/in:231543514280633) State ROUTING going to sleep
> > > 2010-12-13 11:58:21.834192 [DEBUG] switch_core_state_machine.c:320
> > > (opal/in:231543514280633) Running State Change CS_EXECUTE
> > > 2010-12-13 11:58:21.834192 [DEBUG] h323caps.cxx:2284 Added capability:
> > > H.239-Video+H.239-Video <26>
> > > 2010-12-13 11:58:21.834192 [DEBUG] switch_core_state_machine.c:366
> > > (opal/in:231543514280633) State EXECUTE
> > > 2010-12-13 11:58:21.834192 [DEBUG] mod_opal.cpp:857 Executing 
> > > connection
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.834192 [DEBUG] switch_core_state_machine.c:157
> > > opal/in:231543514280633 Standard EXECUTE
> > > EXECUTE opal/in:231543514280633 set(continue_on_fail=true)
> > > 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2284 Added capability:
> > > H.239-Control <27>
> > > 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2029 Parsing remote
> > > capabilities
> > > 2010-12-13 11:58:21.837191 [DEBUG] mod_opal.cpp:1076 Received 
> > > message 26
> > > on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.837191 [DEBUG] mod_dptools.c:1028
> > > opal/in:231543514280633 SET [continue_on_fail]=[true]
> > > 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:819 
> > > Capability tx frames
> > > left at 20 as remote allows 240
> > > 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2529 Could not find
> > > capability: receiveVideoCapability, type h261VideoCapability
> > > 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2529 Could not find
> > > capability: receiveVideoCapability, type h261VideoCapability
> > > 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2529 Could not find
> > > capability: receiveVideoCapability, type h261VideoCapability
> > > 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2671 Found capability:
> > > UserInput/RFC2833 <8>
> > > 2010-12-13 11:58:21.837191 [DEBUG] mod_opal.cpp:1076 Received 
> > > message 27
> > > on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > EXECUTE opal/in:231543514280633 set(hangup_after_bridge=true)
> > > 2010-12-13 11:58:21.837191 [DEBUG] mod_opal.cpp:1076 Received 
> > > message 26
> > > on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.837191 [DEBUG] mod_dptools.c:1028
> > > opal/in:231543514280633 SET [hangup_after_bridge]=[true]
> > > 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2411 Could not find
> > > capability: G.711-ALaw-64k <1>
> > > 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2294 Added capability:
> > > G.711-ALaw-64k <1>
> > > 2010-12-13 11:58:21.837191 [DEBUG] mod_opal.cpp:1076 Received 
> > > message 27
> > > on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > EXECUTE opal/in:231543514280633 set(progress_timeout=15)
> > > 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2411 Could not find
> > > capability: UserInput/hookflash <5>
> > > 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2294 Added capability:
> > > UserInput/hookflash <5>
> > > 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2411 Could not find
> > > capability: UserInput/basicString <6>
> > > 2010-12-13 11:58:21.837191 [DEBUG] mod_opal.cpp:1076 Received 
> > > message 26
> > > on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.837191 [DEBUG] mod_dptools.c:1028
> > > opal/in:231543514280633 SET [progress_timeout]=[15]
> > > 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2294 Added capability:
> > > UserInput/basicString <6>
> > > 2010-12-13 11:58:21.837191 [DEBUG] h323caps.cxx:2411 Could not find
> > > capability: UserInput/dtmf <7>
> > > 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2294 Added capability:
> > > UserInput/dtmf <7>
> > > 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2411 Could not find
> > > capability: UserInput/RFC2833 <8>
> > > 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2294 Added capability:
> > > UserInput/RFC2833 <8>
> > > 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2355 Found capability:
> > > G.711-ALaw-64k <1>
> > > 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2355 Found capability:
> > > UserInput/hookflash <5>
> > > 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2355 Found capability:
> > > UserInput/basicString <6>
> > > 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2355 Found capability:
> > > UserInput/dtmf <7>
> > > 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2355 Found capability:
> > > UserInput/RFC2833 <8>
> > > 2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received 
> > > message 27
> > > on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > EXECUTE opal/in:231543514280633 set(proxy_media=false)
> > > 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2777 Capability merge
> > > result:
> > >      Table:
> > >        G.711-ALaw-64k <1>
> > >        UserInput/hookflash <5>
> > >        UserInput/basicString <6>
> > >        UserInput/dtmf <7>
> > >        UserInput/RFC2833 <8>
> > >      Set:
> > >        0:
> > >          0:
> > >      
> > > 
> > > Log-Func: 
> > > Log-Line: 0
> > > User-Data: 
> > > 
> > >       G.711-ALaw-64k <1>
> > >          1:
> > >          2:
> > >            UserInput/hookflash <5>
> > >          3:
> > >            UserInput/basicString <6>
> > >            UserInput/dtmf <INCOMING DATA [(null)]
> > > 
> > > RECV EVENT
> > > Event-Name: SOCKET_DATA
> > > 
> > > 
> > > 2010-12-13 11:58:21.840189 [DEBUG] h323caps.cxx:2778 Received 
> > > capability
> > > set, is accepted
> > > 2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received 
> > > message 26
> > > on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.840189 [DEBUG] mod_dptools.c:1028
> > > opal/in:231543514280633 SET [proxy_media]=[false]
> > > 2010-12-13 11:58:21.840189 [DEBUG] h323neg.cxx:341 Sending
> > > TerminalCapabilitySet: outSeq=1
> > > 2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received 
> > > message 27
> > > on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > EXECUTE opal/in:231543514280633 set(bypass_media=true)
> > > 2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received 
> > > message 26
> > > on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.840189 [DEBUG] mod_dptools.c:1028
> > > opal/in:231543514280633 SET [bypass_media]=[true]
> > > 2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received 
> > > message 27
> > > on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > EXECUTE opal/in:231543514280633 set(absolute_codec_string=PCMA)
> > > 2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received 
> > > message 26
> > > on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.840189 [DEBUG] mod_dptools.c:1028
> > > opal/in:231543514280633 SET [absolute_codec_string]=[PCMA]
> > > 2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received 
> > > message 27
> > > on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > EXECUTE opal/in:231543514280633
> > > bridge(sofia/external/21543514280633 at 200.35.145.149)
> > > 2010-12-13 11:58:21.840189 [DEBUG] mod_opal.cpp:1076 Received 
> > > message 26
> > > on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.840189 [NOTICE] switch_channel.c:784 New Channel
> > > sofia/external/21543514280633 at 200.35.145.149
> > > [f9bf5b9e-a273-400c-b18c-e2c98c84750f]
> > > 2010-12-13 11:58:21.840189 [DEBUG] mod_sofia.c:3995
> > > (sofia/external/21543514280633 at 200.35.145.149) State Change CS_NEW ->
> > > CS_INIT
> > > 2010-12-13 11:58:21.840189 [DEBUG] switch_core_session.c:1083 Send
> > > signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
> > > 2010-12-13 11:58:21.843189 [DEBUG] h323pdu.cxx:80 Sending PDU:
> > >   request terminalCapabilitySet {
> > >     sequenceNumber = 1
> > >     protocolIdentifier = 0.0.8.245.0.13
> > >     multiplexCapability = h2250Capability {
> > >       maximumAudioDelayJitter = 250
> > >       receiveMultipointCapabili
> > > ty = {
> > >         multicastCapability = false
> > >         multiUniCastConference = false
> > >         mediaDistributionCapability = 1 entries {
> > >           [0]={
> > >             centralizedControl = false
> > >             distributedControl = false
> > >             centralized
> > > Audio = false
> > >             distributedAudio = false
> > >             centralizedVideo = false
> > >             distributedVideo = false
> > >           }
> > >         }
> > >       }
> > >       transmitMultipointCapability = {
> > >         multicastCapability = false
> > >         multiUniCas
> > > tConference = false
> > >         mediaDistributionCapability = 1 entries {
> > >           [0]={
> > >             centralizedControl = false
> > >             distributedControl = false
> > >             centralizedAudio = false
> > >             distributedAudio = false
> > >            
> > >  centralizedVideo = false
> > >             distributedVideo = false
> > >           }
> > >         }
> > >       }
> > >       receiveAndTransmitMultipointCapability = {
> > >         multicastCapability = false
> > >         multiUniCastConference = false
> > >         mediaDistributionCapabil
> > > ity = 1 entries {
> > >           [0]={
> > >             centralizedControl = false
> > >             distributedControl = false
> > >             centralizedAudio = false
> > >             distributedAudio = false
> > >             centralizedVideo = false
> > >             distributedVid
> > > eo = false
> > >           }
> > >         }
> > >       }
> > >       mcCapability = {
> > >         centralizedConferenceMC = false
> > >         decentralizedConferenceMC = false
> > >       }
> > >       rtcpVideoControlCapability = false
> > >       mediaPacketizationCapability = {
> > >         h261aVi
> > > deoPacketization = false
> > >       }
> > >       logicalChannelSwitchingCapability = false
> > >       t120DynamicPortCapability = true
> > >     }
> > >     capabilityTable = 8 entries {
> > >       [0]={
> > >         capabilityTableEntryNumber = 1
> > >         capability = receiveAudioCapab
> > > ility g7231 {
> > >           maxAl_sduAudioFrames = 8
> > >           silenceSuppression = false
> > >         }
> > >       }
> > >       [1]={
> > >         capabilityTableEntryNumber = 2
> > >         capability = receiveAudioCapability g729 24
> > >       }
> > >       [2]={
> > >         capabilityTabl
> > > eEntryNumber = 3
> > >         capability = receiveAudioCapability g711Ulaw64k 240
> > >       }
> > >       [3]={
> > >         capabilityTableEntryNumber = 4
> > >         capability = receiveAudioCapability g711Alaw64k 240
> > >       }
> > >       [4]={
> > >         capabilityTableEntryNumbe
> > > r = 5
> > >         capability = receiveUserInputCapability hookflash <<null>>
> > >       }
> > >       [5]={
> > >         capabilityTableEntryNumber = 6
> > >         capability = receiveUserInputCapability basicString <<null>>
> > >       }
> > >       [6]={
> > >         capabilityTableEntry
> > > Number = 7
> > >         capability = receiveUserInputCapability dtmf <<null>>
> > >       }
> > >       [7]={
> > >         capabilityTableEntryNumber = 8
> > >         capability = receiveRTPAudioTelephonyEventCapability {
> > >           dynamicRTPPayloadType = 101
> > >           audioT
> > > elephoneEvent = "0-16"
> > >         }
> > >       }
> > >     }
> > >     capabilityDescriptors = 1 entries {
> > >       [0]={
> > >         capabilityDescriptorNumber = 1
> > >         simultaneousCapabilities = 3 entries {
> > >           [0]=4 entries {
> > >             [0]=1
> > >             [1]=2
> > >   
> > >           [2]=3
> > >             [3]=4
> > >           }
> > >           [1]=1 entries {
> > >             [0]=5
> > >           }
> > >           [2]=3 entries {
> > >             [0]=6
> > >             [1]=7
> > >             [2]=8
> > >           }
> > >         }
> > >       }
> > >     }
> > >   }
> > > 2010-12-13 11:58:21.843189 [DEBUG] h323caps.cxx:2376 Found capability:
> > > UserInput/RFC2833 <8>
> > > 2010-12-13 11:58:21.843189 [DEBUG] h323.cxx:3535 User Input RFC2833
> > > payload type set to [pt=101]
> > > 2010-12-13 11:58:21.843189 [DEBUG] h323pdu.cxx:80 Sending PDU:
> > >   response terminalCapabilitySetAck {
> > >     sequenceNumber = 1
> > >   }
> > > 2010-12-13 11:58:21.843189 [DEBUG] switch_core_state_machine.c:320
> > > (sofia/external/21543514280633 at 200.35.145.149) Running State Change
> > > CS_INIT
> > > 2010-12-13 11:58:21.843189 [DEBUG] switch_core_state_machine.c:356
> > > (sofia/external/21543514280633 at 200.35.145.149) State INIT
> > > 2010-12-13 11:58:21.843189 [DEBUG] mod_sofia.c:86
> > > sofia/external/21543514280633 at 200.35.145.149 SOFIA INIT
> > > 2010-12-13 11:58:21.843189 [DEBUG] h323.cxx:3771
> > > InternalEstablishedConnectionCheck: 
> > > connectionState=AwaitingLocalAnswer
> > > fastStartState=FastStartResponse H.245 is unavailable
> > > 2010-12-13 11:58:21.843189 [DEBUG] h323pdu.cxx:80 Receiving PDU:
> > >   request masterSlaveDetermination {
> > >     terminalType = 50
> > >     statusDeterminationNumber = 6111531
> > >   }
> > > 2010-12-13 11:58:21.843189 [DEBUG] h323neg.cxx:129 Received
> > > MasterSlaveDetermination: state=Idle
> > > 2010-12-13 11:58:21.843189 [DEBUG] h323neg.cxx:160
> > > MasterSlaveDetermination: local is master
> > > 2010-12-13 11:58:21.843189 [DEBUG] h323pdu.cxx:80 Sending PDU:
> > >   response masterSlaveDeterminationAck {
> > >     decision = slave <<null>>
> > >   }
> > > 2010-12-13 11:58:21.843189 [DEBUG] h323.cxx:3771
> > > InternalEstablishedConnectionCheck: 
> > > connectionState=AwaitingLocalAnswer
> > > fastStartState=FastStartResponse H.245 is ready
> > > 2010-12-13 11:58:21.843189 [DEBUG] mod_sofia.c:126
> > > (sofia/external/21543514280633 at 200.35.145.149) State Change CS_INIT ->
> > > CS_ROUTING
> > > 2010-12-13 11:58:21.843189 [DEBUG] switch_core_session.c:1083 Send
> > > signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
> > > 2010-12-13 11:58:21.843189 [DEBUG] sofia.c:4604 Channel
> > > sofia/external/21543514280633 at 200.35.145.149 entering state
> > > [terminated][900]
> > > 2010-12-13 11:58:21.843189 [DEBUG] h323ep.cxx:1033 
> > > OnSendAlerting conn =
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:21.843189 [DEBUG] switch_channel.c:2455
> > > (sofia/external/21543514280633 at 200.35.145.149) Callstate 
> > > Change DOWN ->
> > > HANGUP
> > > 2010-12-13 11:58:21.843189 [DEBUG] h323.cxx:2110 SetAlerting sending
> > > Alerting PDU
> > > 2010-12-13 11:58:21.843189 [NOTICE] sofia.c:5244 Hangup
> > > sofia/external/21543514280633 at 200.35.145.149 [CS_ROUTING]
> > > [NORMAL_UNSPECIFIED]
> > > 2010-12-13 11:58:21.843189 [DEBUG] switch_channel.c:2471 Send signal
> > > sofia/external/21543514280633 at 200.35.145.149 [KILL]
> > > 2010-12-13 11:58:21.843189 [DEBUG] switch_core_session.c:1083 Send
> > > signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
> > > 2010-12-13 11:58:21.843189 [DEBUG] switch_core_state_machine.c:356
> > > (sofia/external/21543514280633 at 200.35.145.149) State INIT 
> > > going to sleep
> > > 2010-12-13 11:58:21.843189 [DEBUG] switch_core_state_machine.c:320
> > > (sofia/external/21543514280633 at 200.35.145.149) Running State Change
> > > CS_HANGUP
> > > 2010-12-13 11:58:21.846211 [DEBUG] h323pdu.cxx:80 Sending PDU:
> > >   {
> > >     q931pdu = {
> > >       protocolDiscriminator = 8
> > >       callReference = 22716
> > >       from = destination
> > >       messageType = Alerting
> > >       IE: Display = {
> > >         72 6f 6f 74 00                              
> > >        root.
> > >       }
> > >       IE: User-User = {
> > >         23 80 06 00 08 91 4a 00  06 22 c0 09 00 00 3d 1e
> > > #.....J.."....=.
> > >         56 6f 78 20 4c 75 63 69  64 61 20 50 74 79 2e 20   Vox Lucida
> > > Pty. 
> > >         ...
> > >       }
> > >     }
> > >     h225pdu = {
> > >       h323
> > > _uu_pdu = {
> > >         h323_message_body = alerting {
> > >           protocolIdentifier = 0.0.8.2250.0.6
> > >           destinationInfo = {
> > >             vendor = {
> > >               vendor = {
> > >                 t35CountryCode = 9
> > >                 t35Extension = 0
> > >       
> > >           manufacturerCode = 61
> > >               }
> > >               productId =  31 octets {
> > >                 56 6f 78 20 4c 75 63 69  64 61 20 50 74 79 2e 20   Vox
> > > Lucida Pty. 
> > >                 4c 74 64 2e 20 6d 6f 64  5f 6f 70 61 6c 00 00 
> > >      Ltd.
> > > mod_op
> > > Log-Line: 0
> > > User-Data: 
> > > 
> > >    }
> > >             terminal = {
> > >             }
> > >             mc = false
> > >             undefinedNode = false
> > >           }
> > >           callIdentifier = {
> > >             guid =  16 octets {
> > >               3a 76 b2 f2 1d 05 e0 11  9b b7 00 1e 64 1f 33
> > > fe   :v........
> > > ..d.3.
> > >             }
> > >           }
> > >           multipleCalls = false
> > >           maintainConnection = false
> > >         }
> > >         h245Tunneling = true
> > >         h245Control = 3 entries {
> > >           [0]= 108 octets {
> > >             02 70 01 06 00 08 81 75  00 0d 80 
> > > 13 80 00 fa 00   .p.....u........
> > >             01 00 00 01 00 00 01 00  00 0c c0 01 00 01 80
> > > 07   ................
> > >             ...
> > >           }
> > >           [1]= 3 octets {
> > >             21 80 01                                           !..
> > >           }
> > >   
> > >         [2]= 2 octets {
> > >             20 a0                                               .
> > >           }
> > >         }
> > >       }
> > >     }
> > >   }
> > > 2010-12-13 11:58:21.846211 [DEBUG] h323ep.cxx:1039 
> > > OnSentAlerting conn =
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:21.846211 [DEBUG] h323.cxx:3771
> > > InternalEstablishedConnectionCheck: 
> > > connectionState=AwaitingLocalAnswer
> > > fastStartState=FastStartResponse H.245 is ready
> > > 2010-12-13 11:58:21.846211 [DEBUG] connection.cxx:361
> > > OnSetUpConnectionCall[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.846211 [DEBUG] endpoint.cxx:408 OnSetUpConnection
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.846211 [DEBUG] h323.cxx:1701 OnAnswerCall
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716], caller =
> > > gustavo [200.117.192.17]
> > > 2010-12-13 11:58:21.846211 [DEBUG] call.cxx:219 OnAnswerCall
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716] caller
> > > "gustavo [200.117.192.17]"
> > > 2010-12-13 11:58:21.846211 [DEBUG] h323.cxx:1713 Answering call:
> > > AnswerCallDeferred
> > > 2010-12-13 11:58:21.846211 [DEBUG] connection.cxx:541 Answering call:
> > > AnswerCallDeferred
> > > 2010-12-13 11:58:21.846211 [DEBUG] h323.cxx:3771
> > > InternalEstablishedConnectionCheck: 
> > > connectionState=AwaitingLocalAnswer
> > > fastStartState=FastStartResponse H.245 is ready
> > > 2010-12-13 11:58:21.846211 [DEBUG] h323.cxx:506 Reading PDUs:
> > > callRef=22716
> > > 2010-12-13 11:58:21.846211 [DEBUG] 
> > > switch_ivr_originate.c:3448 Originate
> > > Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED]
> > > 2010-12-13 11:58:21.849192 [INFO] mod_dptools.c:2579 Originate Failed.
> > > Cause: NORMAL_UNSPECIFIED
> > > 2010-12-13 11:58:21.849192 [DEBUG] mod_opal.cpp:1076 Received 
> > > message 27
> > > on connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.849192 [NOTICE] switch_core_state_machine.c:189
> > > opal/in:231543514280633 has executed the last dialplan instruction,
> > > hanging up.
> > > 2010-12-13 11:58:21.849192 [DEBUG] switch_channel.c:2455
> > > (opal/in:231543514280633) Callstate Change RINGING -> HANGUP
> > > 2010-12-13 11:58:21.849192 [NOTICE] switch_core_state_machine.c:191
> > > Hangup opal/in:231543514280633 [CS_EXECUTE] [NORMAL_CLEARING]
> > > 2010-12-13 11:58:21.849192 [DEBUG] switch_channel.c:2471 Send signal
> > > opal/in:231543514280633 [KILL]
> > > 2010-12-13 11:58:21.849192 [DEBUG] mod_opal.cpp:931 Kill 1 on 
> > > connection
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.849192 [DEBUG] mod_opal.cpp:1092 State changed on
> > > connection Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.849192 [DEBUG] switch_core_session.c:1083 Send
> > > signal opal/in:231543514280633 [BREAK]
> > > 2010-12-13 11:58:21.849192 [DEBUG] mod_opal.cpp:931 Kill 3 on 
> > > connection
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:366
> > > (opal/in:231543514280633) State EXECUTE going to sleep
> > > 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:320
> > > (opal/in:231543514280633) Running State Change CS_HANGUP
> > > 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:557
> > > (sofia/external/21543514280633 at 200.35.145.149) State HANGUP
> > > 2010-12-13 11:58:21.849192 [DEBUG] mod_sofia.c:453
> > > sofia/external/21543514280633 at 200.35.145.149 Overriding SIP cause 480
> > > with 900 from the other leg
> > > 2010-12-13 11:58:21.849192 [DEBUG] mod_sofia.c:459 Channel
> > > sofia/external/21543514280633 at 200.35.145.149 hanging up, cause:
> > > NORMAL_UNSPECIFIED
> > > 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:46
> > > sofia/external/21543514280633 at 200.35.145.149 Standard HANGUP, cause:
> > > NORMAL_UNSPECIFIED
> > > 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:557
> > > (sofia/external/21543514280633 at 200.35.145.149) State HANGUP going to
> > > sleep
> > > 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:351
> > > (sofia/external/21543514280633 at 200.35.145.149) State Change 
> > > CS_HANGUP ->
> > > CS_REPORTING
> > > 2010-12-13 11:58:21.849192 [DEBUG] switch_core_session.c:1083 Send
> > > signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
> > > 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:320
> > > (sofia/external/21543514280633 at 200.35.145.149) Running State Change
> > > CS_REPORTING
> > > 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:617
> > > (sofia/external/21543514280633 at 200.35.145.149) State REPORTING
> > > 2010-12-13 11:58:21.849192 [DEBUG] switch_core_state_machine.c:557
> > > (opal/in:231543514280633) State HANGUP
> > > 2010-12-13 11:58:21.849192 [DEBUG] connection.cxx:401 Call end reason
> > > for Call[Cc70e3cf91]-EP<local>[Ldc8d6c462] set to EndedByRemoteUser
> > > 2010-12-13 11:58:21.849192 [DEBUG] call.cxx:112 Clearing
> > > Call[Cc70e3cf91] reason=EndedByRemoteUser
> > > 2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:1516 SetPhase from
> > > SetUpPhase to ReleasingPhase for Call[Cc70e3cf91]-EP<h323>[tcp
> > > $200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:459 Releasing
> > > asynchronously 
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:21.852189 [DEBUG] osutil.cxx:196 File handle 
> > > low water
> > > mark set: 69 Thread unblock pipe
> > > 2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:1516 SetPhase from
> > > AlertingPhase to ReleasingPhase for
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:448 Releasing
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:489 OnReleased
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:808 Media streams
> > > closed.
> > > 2010-12-13 11:58:21.852189 [DEBUG] endpoint.cxx:450 OnReleased
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.852189 [DEBUG] manager.cxx:709 OnReleased
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.852189 [DEBUG] call.cxx:716 OnReleased
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.852189 [DEBUG] connection.cxx:436 Already released
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:21.856029 [DEBUG] connection.cxx:1516 SetPhase from
> > > ReleasingPhase to ReleasedPhase for
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462]
> > > 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:46
> > > opal/in:231543514280633 Standard HANGUP, cause: NORMAL_CLEARING
> > > 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:557
> > > (opal/in:231543514280633) State HANGUP going to sleep
> > > 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:351
> > > (opal/in:231543514280633) State Change CS_HANGUP -> CS_REPORTING
> > > 2010-12-13 11:58:21.856029 [DEBUG] switch_core_session.c:1083 Send
> > > signal opal/in:231543514280633 [BREAK]
> > > 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:320
> > > (opal/in:231543514280633) Running State Change CS_REPORTING
> > > 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:617
> > > (opal/in:231543514280633) State REPORTING
> > > 2010-12-13 11:58:21.856029 [DEBUG] connection.cxx:401 Call end reason
> > > for Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716] set to
> > > EndedByRemoteUser
> > > 2010-12-13 11:58:21.856029 [DEBUG] h323.cxx:349 OnReleased: tcp
> > > $200.117.192.17:19270/22716, connectionState=AwaitingLocalAnswer
> > > 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:53
> > > sofia/external/21543514280633 at 200.35.145.149 Standard 
> > > REPORTING, cause:
> > > NORMAL_UNSPECIFIED
> > > 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:617
> > > (sofia/external/21543514280633 at 200.35.145.149) State 
> > > REPORTING going to
> > > sleep
> > > 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:345
> > > (sofia/external/21543514280633 at 200.35.145.149) State Change 
> > > CS_REPORTING
> > > -> CS_DESTROY
> > > 2010-12-13 11:58:21.856029 [DEBUG] switch_core_session.c:1083 Send
> > > signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
> > > 2010-12-13 11:58:21.856029 [DEBUG] switch_core_session.c:1250 
> > > Session 2
> > > (sofia/external/21543514280633 at 200.35.145.149) Locked, Waiting on
> > > external entities
> > > 2010-12-13 11:58:21.856029 [NOTICE] 
> > > switch_core_session.c:1268 Session 2
> > > (sofia/external/21543514280633 at 200.35.145.149) Ended
> > > 2010-12-13 11:58:21.856029 [NOTICE] switch_core_session.c:1270 Close
> > > Channel sofia/external/21543514280633 at 200.35.145.149 [CS_DESTROY]
> > > 2010-12-13 11:58:21.856029 [DEBUG] switch_core_state_machine.c:449
> > > (sofia/external/21543514280633 at 200.35.145.149) Callstate Change HANGUP
> > > -> DOWN
> > > 2010-12-13 11:58:21.858228 [DEBUG] switch_core_state_machine.c:452
> > > (sofia/external/21543514280633 at 200.35.145.149) Running State Change
> > > CS_DESTROY
> > > 2010-12-13 11:58:21.858228 [DEBUG] switch_core_state_machine.c:462
> > > (sofia/external/21543514280633 at 200.35.145.149) State DESTROY
> > > 2010-12-13 11:58:21.858228 [DEBUG] mod_sofia.c:364
> > > sofia/external/21543514280633 at 200.35.145.149 SOFIA DESTROY
> > > 2010-12-13 11:58:21.858228 [DEBUG] switch_core_state_machine.c:60
> > > sofia/external/21543514280633 at 200.35.145.149 Standard DESTROY
> > > 2010-12-13 11:58:21.858228 [DEBUG] switch_core_state_machine.c:462
> > > (sofia/external/21543514280633 at 200.35.145.149) State DESTROY going to
> > > sleep
> > > 2010-12-13 11:58:21.856029 [DEBUG] h323.cxx:356 Sending 
> > > release complete
> > > PDU: callRef=22716
> > > 2010-12-13 11:58:21.858228 [DEBUG] h323pdu.cxx:80 Sending PDU:
> > >   command endSessionCommand disconnect <<null>>
> > > 2010-12-13 11:58:21.858228 [DEBUG] h323pdu.cxx:80 Sending PDU:
> > >   {
> > >     q931pdu = {
> > >       protocolDiscriminator = 8
> > >       callReference = 22716
> > >       from = destination
> > >       messageType = ReleaseComplete
> > >       IE: Cause - Normal call clearing = {
> > >         80 90           
> > >                                    ..
> > >       }
> > >       IE: Display = {
> > >         72 6f 6f 74 00                                     root.
> > >       }
> > >       IE: User-User = {
> > >         25 80 06 00 08 91 4a 00  06 01 11 00 3a 76 b2 f2
> > > %.....J.....:v..
> > >         
> > > Log-Func: 
> > > Log-Line: 0
> > > User-Data: 
> > > 
> > > 1d 05 e0 11 9b b7 00 1e  64 1f 33 fe 04 c0 01 80   ........d.3.....
> > >         ...
> > >       }
> > >     }
> > >     h225pdu = {
> > >       h323_uu_pdu = {
> > >         h323_message_body = releaseComplete {
> > >           protocolIdentifier = 0.0.8= {
> > >             guid =  16 octets {
> > >               3a 76 b2 f2 1d 05 e0 11  9b b7 00 1e 64 1f 33
> > > fe   :v..........d.3.
> > >             }
> > >           }
> > >         }
> > >         h245Tunneling = true
> > >         h245Control = 1 entries {
> > >           [0]= 2 octets {
> > >       
> > >       4a 40                                              J@
> > >           }
> > >         }
> > >       }
> > >     }
> > >   }
> > > 2010-12-13 11:58:21.861189 [DEBUG] h323.cxx:412 Awaiting end session
> > > from remote for 9.997 seconds
> > > 2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:53
> > > opal/in:231543514280633 Standard REPORTING, cause: NORMAL_CLEARING
> > > 2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:617
> > > (opal/in:231543514280633) State REPORTING going to sleep
> > > 2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:345
> > > (opal/in:231543514280633) State Change CS_REPORTING -> CS_DESTROY
> > > 2010-12-13 11:58:21.861189 [DEBUG] switch_core_session.c:1083 Send
> > > signal opal/in:231543514280633 [BREAK]
> > > 2010-12-13 11:58:21.861189 [DEBUG] switch_core_session.c:1250 
> > > Session 1
> > > (opal/in:231543514280633) Locked, Waiting on external entities
> > > 2010-12-13 11:58:21.861189 [NOTICE] 
> > > switch_core_session.c:1268 Session 1
> > > (opal/in:231543514280633) Ended
> > > 2010-12-13 11:58:21.861189 [NOTICE] switch_core_session.c:1270 Close
> > > Channel opal/in:231543514280633 [CS_DESTROY]
> > > 2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:449
> > > (opal/in:231543514280633) Callstate Change HANGUP -> DOWN
> > > 2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:452
> > > (opal/in:231543514280633) Running State Change CS_DESTROY
> > > 2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:462
> > > (opal/in:231543514280633) State DESTROY
> > > 2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:60
> > > opal/in:231543514280633 Standard DESTROY
> > > 2010-12-13 11:58:21.861189 [DEBUG] switch_core_state_machine.c:462
> > > (opal/in:231543514280633) State DESTROY going to sleep
> > > 2010-12-13 11:58:22.087167 [DEBUG] h323pdu.cxx:80 Receiving PDU:
> > >   {
> > >     q931pdu = {
> > >       protocolDiscriminator = 8
> > >       callReference = 22716
> > >       from = originator
> > >       messageType = Facility
> > >       IE: Facility = {
> > > 
> > >       }
> > >       IE: Display = {
> > >         67 75 73 74
> > > (..........!... 
> > >         80                                                 .
> > >       }
> > >     }
> > >     h225pdu = {
> > >  
> > >      h323_uu_pdu = {
> > >         h323_message_body = empty <<null>>
> > >         h245Tunneling = true
> > >         h245Control = 2 entries {
> > >           [0]= 3 octets {
> > >             21 80 01                                           !..
> > >           }
> > >           [1]= 2 
> > > octets {
> > >             20 80                                               .
> > >           }
> > >         }
> > >       }
> > >     }
> > >   }
> > > 2010-12-13 11:58:22.087167 [DEBUG] h323.cxx:559 Handling PDU: Facility
> > > callRef=22716
> > > 2010-12-13 11:58:22.087167 [DEBUG] h323.cxx:2709 Checking for end
> > > session on PDU: response terminalCapabilitySetAck
> > > 2010-12-13 11:58:22.087167 [DEBUG] h323.cxx:2709 Checking for end
> > > session on PDU: response masterSlaveDeterminationAck
> > > 2010-12-13 11:58:22.087167 [DEBUG] connection.cxx:436 Already released
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:22.087167 [DEBUG] h323.cxx:549 Signal channel closed.
> > > 2010-12-13 11:58:22.087167 [DEBUG] transports.cxx:900 
> > > Transport clean up
> > > on termination
> > > 2010-12-13 11:58:22.087167 [DEBUG] transports.cxx:885 Transport Close
> > > 2010-12-13 11:58:22.090169 [DEBUG] osutil.cxx:196 File handle 
> > > low water
> > > mark set: 65 PTextFile
> > > 2010-12-13 11:58:22.090169 [DEBUG] tlibthrd.cxx:1020 Could not parse
> > > thread stat file /proc/21145/task/21190/stat
> > > 2010-12-13 11:58:22.306147 [DEBUG] connection.cxx:489 OnReleased
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:22.306147 [DEBUG] connection.cxx:808 Media streams
> > > closed.
> > > 2010-12-13 11:58:22.306147 [DEBUG] endpoint.cxx:450 OnReleased
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:22.306147 [DEBUG] manager.cxx:709 OnReleased
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:22.306147 [DEBUG] call.cxx:716 OnReleased
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:22.306147 [DEBUG] manager.cxx:535 OnClearedCall
> > > Call[Cc70e3cf91] from "h323:200.117.192.17" to "h323:231543514280633"
> > > 2010-12-13 11:58:22.306147 [DEBUG] connection.cxx:1516 SetPhase from
> > > ReleasingPhase to ReleasedPhase for Call[Cc70e3cf91]-EP<h323>[tcp
> > > $200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:22.306147 [DEBUG] connection.cxx:480 OnRelease thread
> > > completed for 
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716]
> > > 2010-12-13 11:58:22.308152 [DEBUG] tlibthrd.cxx:1020 Could not parse
> > > thread stat file /proc/21145/task/21194/stat
> > > 2010-12-13 11:58:22.588122 [DEBUG] transports.cxx:900 
> > > Transport clean up
> > > on termination
> > > 2010-12-13 11:58:22.588122 [DEBUG] transports.cxx:885 Transport Close
> > > 2010-12-13 11:58:22.588122 [DEBUG] transports.cxx:1045 
> > > Deleted transport
> > > tcp$200.117.192.17:19270<if=tcp$10.0.0.1:1720>
> > > 2010-12-13 11:58:22.590122 [DEBUG] h323.cxx:330 Connection tcp
> > > $200.117.192.17:19270/22716 deleted.
> > > 2010-12-13 11:58:22.590122 [DEBUG] connection.cxx:343 Connection
> > > Call[Cc70e3cf91]-EP<h323>[tcp$200.117.192.17:19270/22716] destroyed.
> > > 2010-12-13 11:58:22.590122 [DEBUG] localep.cxx:211 Deleted connection.
> > > 2010-12-13 11:58:22.590122 [DEBUG] connection.cxx:343 Connection
> > > Call[Cc70e3cf91]-EP<local>[Ldc8d6c462] destroyed.
> > > 2010-12-13 11:58:23.592039 [DEBUG] call.cxx:86 Call[Cc70e3cf91]
> > > destroyed.
> > > 
> > > --------------------------------------------------------------
> > > --------------------------------------------------------------
> > > -----------------------------------------
> > > Gustavo Espeche
> > > www.easyipcall.com
> > > 
> > > 
> > > 
> > > 
> > > 
> > > 
> > > 
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
> > > itch-users
> > > http://www.freeswitch.org
> > 
> > 
> > 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From gustavo.espeche at upper-soft.com  Mon Dec 13 20:42:11 2010
From: gustavo.espeche at upper-soft.com (Gustavo Espeche)
Date: Mon, 13 Dec 2010 14:42:11 -0300
Subject: [Freeswitch-users] opal to sip problem
In-Reply-To: <mailman.470.1292251727.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.470.1292251727.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <1292262131.2202.6.camel@gustavo-laptop>

Hi i try to start mod_h323 but i have follow issue.
        
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
        on socket
        accept on ip$10.0.0.1:1720
        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
        open
        transport, connection not started.
On Mon, 2010-12-13 at 17:48 +0300,
freeswitch-users-request at lists.freeswitch.org wrote:
> Re: opal to sip problem



From tculjaga at gmail.com  Mon Dec 13 20:50:16 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 13 Dec 2010 18:50:16 +0100
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <COL109-W556C528971964A0ED8308BF4100@phx.gbl>
References: <COL109-W556C528971964A0ED8308BF4100@phx.gbl>
Message-ID: <AANLkTiny4MUv4aBbNONPqqMNcD-_Rsey=EU3sCGooC+A@mail.gmail.com>

On Sat, Dec 11, 2010 at 2:28 PM, Abid Saleem <abid_freeswitch at live.com>wrote:

>  Hi,
>
> Sorry I was away from work. Just recalling the radius AAA and credit-time
> stuff, I am wondering if there is a way to announce the remaining time of 1
> minute or 30 seconds before the call has to disconnect.
>
> Please help if you FS supports this feature.
>
> Regards
> -----------
> Abid Saleem
>
> ------------------------------
> Date: Mon, 15 Nov 2010 08:53:23 +0100
> From: tculjaga at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Radius AAA
>
>
>
>     <action application="log" data="INFO  credit_time=${credit_time}"/>
>      <action application="sched_hangup" data="+${credit_time:17:-1}
> ${Core-UUID}"/>
>
> In this scenario you don't have to check for return code, because you will
> get hangup
> if authentication fails.
>
>

Hi,

i do it like this:


put this somewhere in the DP before you bridge the call ...

     <action inline="true" application="export"
data="nolocal:CTIME=${regex(${credit_time}|.*=(.*)|%1)}"/>
     <action application="export"
data="nolocal:execute_on_answer=execute_extension ANSWERED_CALL XML NXIVR
bleg"/>

WTIME is your delay time you want the prompt to be played before the hangup.



  <extension name="ANSWERED_CALL">
    <condition field="destination_number" expression="^ANSWERED_CALL$">
      <action application="export" data="send_accounting_stop_enable=true"/>
      <action inline="true" application="export"
data="ANSWERED_STATE=ANSWERED"/>
      <action application="set" data="playback_delimiter=!"/>
      <action application="sched_broadcast" data="+${CTIME}
playback::${KILL_CALL_PR} bleg"/>
      <action application="sched_broadcast"
data="+${expr(${CTIME}-${WTIME})} playback::${ACC_EXP_PR} bleg"/>
      <action application="sched_hangup" data="+${CTIME} alloted_timeout
aleg"/>
      <action application="app_init"/>
    </condition>
  </extension>
-------------- next part --------------
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From steveayre at gmail.com  Mon Dec 13 21:20:43 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 13 Dec 2010 18:20:43 +0000
Subject: [Freeswitch-users] opal to sip problem
In-Reply-To: <1292262131.2202.6.camel@gustavo-laptop>
References: <mailman.470.1292251727.24555.freeswitch-users@lists.freeswitch.org>
	<1292262131.2202.6.camel@gustavo-laptop>
Message-ID: <ADD33631-94C5-4189-8629-3844DCD64D55@gmail.com>

Please do not reply to digest messages. Start a new conversation thread by composing a new message and sending it to the list.

If you do not receive a reply, Don't just send it again. Have patience, someone will eventually reply when someone who can answer your message sees your message. If you send the same message multiple times you are likely to irritate people on the list and are less likely to receive a reply.

Your error looks like either the server doesn't have the 

Steve on iPhone

On 13 Dec 2010, at 17:42, Gustavo Espeche <gustavo.espeche at upper-soft.com> wrote:

> Hi i try to start mod_h323 but i have follow issue.
> 
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
> On Mon, 2010-12-13 at 17:48 +0300,
> freeswitch-users-request at lists.freeswitch.org wrote:
>> Re: opal to sip problem
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From steveayre at gmail.com  Mon Dec 13 21:22:47 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 13 Dec 2010 18:22:47 +0000
Subject: [Freeswitch-users] opal to sip problem
In-Reply-To: <1292262131.2202.6.camel@gustavo-laptop>
References: <mailman.470.1292251727.24555.freeswitch-users@lists.freeswitch.org>
	<1292262131.2202.6.camel@gustavo-laptop>
Message-ID: <BA2E64B5-0EBD-4601-B0B5-98C689F8038F@gmail.com>

Please do not reply to digest messages. Start a new conversation thread by composing a new message and sending it to the list.

If you do not receive a reply, Don't just send it again. Have patience, someone will eventually reply when someone who can answer your message sees your message. If you send the same message multiple times you are likely to irritate people on the list and are less likely to receive a reply.

Your error looks like either the server doesn't have the 10.0.0.1 IP assigned, or something else is already listening on port 1720. Check that you have not loaded both mod_h323 and mod_opal at the same time.

Sorry, I knocked send too early on my previous reply.

Steve on iPhone

On 13 Dec 2010, at 17:42, Gustavo Espeche <gustavo.espeche at upper-soft.com> wrote:

> Hi i try to start mod_h323 but i have follow issue.
> 
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.430472 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.430472 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
>        2010-12-13 17:29:22.432465 [DEBUG] transports.cxx:1475 Waiting
>        on socket
>        accept on ip$10.0.0.1:1720
>        2010-12-13 17:29:22.432465 [INFO] transports.cxx:1482 Failed to
>        open
>        transport, connection not started.
> On Mon, 2010-12-13 at 17:48 +0300,
> freeswitch-users-request at lists.freeswitch.org wrote:
>> Re: opal to sip problem
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From infos at madovsky.org  Mon Dec 13 22:14:48 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 13 Dec 2010 14:14:48 -0500
Subject: [Freeswitch-users] DTMF and dynamic conference
Message-ID: <4A6B12C1D6794B4DA9FD15827BD6812C@e1705>

I'm trying the default conference in autoload_configs/conference.conf.xml,
commented out the pin number line,

but now when the ivr asks the pin number, for testing I tried
to do this :

        <extension name="nb_conferences">
                <condition field="${sip_to_uri}" expression="^(999)@$${domain}$">
                        <action application="answer"/>
                        <action application="info"/>
                        <!-- <action application="conference" data="confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin number]"> -->
                        <action application="conference" data="$1-$${domain}@default"/>
                        <action application="send_dtmf" data="12345 at 120"/>
                </condition>
        </extension>

and this on CLI:

expand uuid_send_dtmf ${uuid} 12345


without success.
How the default conference example accept DTMF ?

Thanks

F
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From mel0torme at gmail.com  Mon Dec 13 22:26:27 2010
From: mel0torme at gmail.com (Tom C)
Date: Mon, 13 Dec 2010 11:26:27 -0800
Subject: [Freeswitch-users] DockStar compile failure, with possible fix
In-Reply-To: <AANLkTikiJ1hoGWSVcL7-dg4At=7SKRy+2hrPSu763StC@mail.gmail.com>
References: <AANLkTi=-1xKh21fR5PADAQs2YPRKHjssCLQMRWNikSu6@mail.gmail.com>
	<AANLkTikiJ1hoGWSVcL7-dg4At=7SKRy+2hrPSu763StC@mail.gmail.com>
Message-ID: <AANLkTi=7O6NAc1HnGF8dp9dDCMsOv0X0YKAZusdBMtPy@mail.gmail.com>

I thought the whole point of the va_start, va_end, va_arg and va_copy macros
was to make variadic functions portable.  (At least that's what it done said
on that there inter-web thingy.)  It's really sad if it doesn't actually
work that way.  :-)

I ran it for a little while with if(1), and it seemed to work fine, but that
was with a very mundane scenario.


On Mon, Dec 13, 2010 at 9:30 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> va_copy is not portable either =D welcome to C
>
> I think this code has evolved to the point at which i don't think its
> even possible for this to happen anyway.
> I am going to try running it without the test at all.
>
> commit dfecc914876b164ce64c53c4f048aa38ed65d9c5
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Mon Dec 13 11:20:12 2010 -0600
>
>
> On Mon, Dec 13, 2010 at 1:10 AM, Tom C <mel0torme at gmail.com> wrote:
> > I was attempting to build FreeSwitch on a Dockstar running Debian
> Squeeze,
> > and ran into the sofia.c logger() compile problem as described here:
> > http://jira.freeswitch.org/browse/FS-802
> >
> > The offered solution was to downgrade GCC, but being a Linux newbie, I
> > didn't know how to do that. So I decided to figure out what the problem
> > really is.
> >
> > After learning waaaay too much about va_list, I see that the logger()
> > function in sofia.c assumes that va_list will always be a pointer.  But
> on
> > different platforms, va_list can be an array or a struct.  On the
> Dockstar
> > (with Debian Squeeze and GCC 4.4), va_list is apparently a struct.
> >
> > I made the following modifications to logger() to make it truly portable,
> > and it compiles and runs fine on my DockStar, my Debian Lenny x86.  (My
> > Windows build is having bigger problems.)  But I'm no C guru, so someone
> > needs to thoroughly examine my code before anyone else takes it as
> gospel.
> > Am I actually checking what needs to be checked with the va_arg() macro?
> > Changing it to if(1) worked too, after all.
> >
> >
> > ORIGINAL logger() in sofia.c, to make it easy to see what I changed.
> > static void logger(void *logarg, char const *fmt, va_list ap)
> > {
> > /* gcc 4.4 gets mad at us for testing if (ap) so let's try to work around
> > it....*/
> > void *ap_ptr = (void *) (intptr_t) ap;  //Error now occurs here, because
> ap
> > is a struct.
> > if (!fmt) return;
> > if (ap_ptr) {  //Error used to occur here, before attempted fix above.
> >    switch_log_vprintf(SWITCH_CHANNEL_LOG_CLEAN,
> > mod_sofia_globals.tracelevel, fmt, ap);
> > } else {
> >    switch_log_printf(SWITCH_CHANNEL_LOG_CLEAN,
> mod_sofia_globals.tracelevel,
> > "%s", fmt);
> > }
> > }
> >
> >
> > My MODIFIED logger() function, additions and changes labelled with
> DOCKSTAR:
> >
> > static
> >
> > void logger(void *logarg, char const *fmt, va_list ap)
> > {
> > va_list temp_ap;  //DOCKSTAR: Added line (replacing previous ap_ptr
> cast).
> >
> > if (!fmt) return;
> >
> > va_copy(temp_ap, ap); //DOCKSTAR: Added Line. Make copy of "ap" so
> va_arg()
> > macro doesn't move pointer.
> >
> > if (va_arg(temp_ap, int)) {  //DOCKSTAR: Modified, get first argument
> from
> > ap, check non-null.
> >      switch_log_vprintf(SWITCH_CHANNEL_LOG_CLEAN,
> > mod_sofia_globals.tracelevel, fmt, ap);
> > } else {
> >      switch_log_printf(SWITCH_CHANNEL_LOG_CLEAN,
> > mod_sofia_globals.tracelevel, "%s", fmt);
> > }
> >
> > va_end(temp_ap);  //DOCKSTAR: Added line.  Release our copy of "ap".
> > }
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Mon Dec 13 22:35:08 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 13 Dec 2010 13:35:08 -0600
Subject: [Freeswitch-users] DockStar compile failure, with possible fix
In-Reply-To: <AANLkTi=7O6NAc1HnGF8dp9dDCMsOv0X0YKAZusdBMtPy@mail.gmail.com>
References: <AANLkTi=-1xKh21fR5PADAQs2YPRKHjssCLQMRWNikSu6@mail.gmail.com>
	<AANLkTikiJ1hoGWSVcL7-dg4At=7SKRy+2hrPSu763StC@mail.gmail.com>
	<AANLkTi=7O6NAc1HnGF8dp9dDCMsOv0X0YKAZusdBMtPy@mail.gmail.com>
Message-ID: <AANLkTinxUDksS=-BvFVd4J7r8E=aZagaWgUUdeJUsUO_@mail.gmail.com>

indeed,

http://www.delorie.com/gnu/docs/glibc/libc_675.html

In either case I think we now can function without the code at all.


On Mon, Dec 13, 2010 at 1:26 PM, Tom C <mel0torme at gmail.com> wrote:
> I thought the whole point of the va_start, va_end, va_arg and va_copy macros
> was to make variadic functions portable.? (At least that's what it done said
> on that there inter-web thingy.)? It's really sad if it doesn't actually
> work that way.? :-)
>
> I ran it for a little while with if(1), and it seemed to work fine, but that
> was with a very mundane scenario.
>
>
> On Mon, Dec 13, 2010 at 9:30 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> va_copy is not portable either =D welcome to C
>>
>> I think this code has evolved to the point at which i don't think its
>> even possible for this to happen anyway.
>> I am going to try running it without the test at all.
>>
>> commit dfecc914876b164ce64c53c4f048aa38ed65d9c5
>> Author: Anthony Minessale <anthm at freeswitch.org>
>> Date: ? Mon Dec 13 11:20:12 2010 -0600
>>
>>
>> On Mon, Dec 13, 2010 at 1:10 AM, Tom C <mel0torme at gmail.com> wrote:
>> > I was attempting to build FreeSwitch on a Dockstar running Debian
>> > Squeeze,
>> > and ran into?the sofia.c logger()?compile problem as described here:
>> > http://jira.freeswitch.org/browse/FS-802
>> >
>> > The offered solution was to downgrade GCC, but being a Linux newbie, I
>> > didn't know how to do that. So I decided to figure out what the problem
>> > really is.
>> >
>> > After learning waaaay too much about va_list, I see that the logger()
>> > function in sofia.c assumes that va_list will always be a pointer.? But
>> > on
>> > different platforms,?va_list can be an array or a struct.? On the
>> > Dockstar
>> > (with Debian Squeeze and GCC 4.4), va_list is apparently a struct.
>> >
>> > I made the following modifications to logger() to make it truly
>> > portable,
>> > and it compiles and runs fine on my DockStar, my Debian Lenny x86.? (My
>> > Windows build is having bigger problems.)? But I'm no C guru, so someone
>> > needs to thoroughly examine my code before anyone?else takes it as
>> > gospel.
>> > Am I actually checking what needs to be checked with the va_arg() macro?
>> > Changing it to if(1) worked too, after all.
>> >
>> >
>> > ORIGINAL logger() in sofia.c, to make it easy to see what I changed.
>> > static void logger(void *logarg, char const *fmt, va_list ap)
>> > {
>> > /* gcc 4.4 gets mad at us for testing if (ap) so let's try to work
>> > around
>> > it....*/
>> > void *ap_ptr = (void *) (intptr_t) ap;? //Error now occurs here, because
>> > ap
>> > is a struct.
>> > if (!fmt) return;
>> > if (ap_ptr) {??//Error used to occur here, before attempted fix above.
>> > ?? switch_log_vprintf(SWITCH_CHANNEL_LOG_CLEAN,
>> > mod_sofia_globals.tracelevel, fmt, ap);
>> > } else {
>> > ???switch_log_printf(SWITCH_CHANNEL_LOG_CLEAN,
>> > mod_sofia_globals.tracelevel,
>> > "%s", fmt);
>> > }
>> > }
>> >
>> >
>> > My MODIFIED logger() function, additions and changes labelled with
>> > DOCKSTAR:
>> >
>> > static
>> >
>> > void logger(void *logarg, char const *fmt, va_list ap)
>> > {
>> > va_list temp_ap;? //DOCKSTAR: Added line (replacing previous ap_ptr
>> > cast).
>> >
>> > if (!fmt) return;
>> >
>> > va_copy(temp_ap, ap); //DOCKSTAR: Added Line. Make copy of "ap" so
>> > va_arg()
>> > macro?doesn't move pointer.
>> >
>> > if (va_arg(temp_ap, int)) {? //DOCKSTAR: Modified, get first argument
>> > from
>> > ap, check non-null.
>> > ???? switch_log_vprintf(SWITCH_CHANNEL_LOG_CLEAN,
>> > mod_sofia_globals.tracelevel, fmt, ap);
>> > } else {
>> > ???? switch_log_printf(SWITCH_CHANNEL_LOG_CLEAN,
>> > mod_sofia_globals.tracelevel, "%s", fmt);
>> > }
>> >
>> > va_end(temp_ap);? //DOCKSTAR: Added line.? Release our copy of "ap".
>> > }
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From jerre at j-cope.com  Mon Dec 13 22:22:26 2010
From: jerre at j-cope.com (Jerre Cope)
Date: Mon, 13 Dec 2010 13:22:26 -0600
Subject: [Freeswitch-users] Incomplete offer/answer debugging
Message-ID: <4D067272.4060602@j-cope.com>

Hello!  Newbie question.

My wifi SIP phone has stopped answering lately and I don't know if I've 
induced a configuration problem by making freeswitch current, or whether 
I've staticed the phone firmware.  I have the wifi handset set up as a 
typical extension 1000.  I've got a Twinkle softphone on ext 1001 and I 
can call that extension from 1000, but when I call from 1001 to ext 
1000, the handset  returns the nua_media_error.  Both extensions are on 
the same network.  I've posted a pastebin of log level 9 sip trace on 
pastebin 14768

I could use some help understanding the log so I can correct my 
nua_media_error negotiation problem.  I started the log early in the 
conversation, where it negotiates past the IP to the user registration, 
but I don't see a clear reason for the "BYE" on line 526.

Thanks for FreeSWITCH!




From msc at freeswitch.org  Mon Dec 13 22:50:56 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Dec 2010 11:50:56 -0800
Subject: [Freeswitch-users] send dtmf on one leg bridge
In-Reply-To: <D10B61781EFA4CFB913F2FEAC7579B27@e1705>
References: <D10B61781EFA4CFB913F2FEAC7579B27@e1705>
Message-ID: <AANLkTikhpiqsFbQHeWuxNE2EojeuFVgrdseBC5CU=308@mail.gmail.com>

Have you tried uuid_send_dtmf?
-MC

On Sun, Dec 12, 2010 at 12:30 PM, Madovsky <infos at madovsky.org> wrote:

>  Is it possible to send dtfmf via
> send_dtmf from CLI on a one leg bridge ?
> it's for enter a pin conference and I can't use
> any RFC on my sip phone.
>
> Thanks
>
> Franck
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Dec 13 22:58:48 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Dec 2010 11:58:48 -0800
Subject: [Freeswitch-users] cepstral problem
In-Reply-To: <4D03307D.9000104@gmail.com>
References: <AANLkTi=tp_JK2H4cHtK8129TnWyYhArxaS2R1jmyPMPr@mail.gmail.com>
	<4CFEA333.1090706@gmail.com>
	<AANLkTinwWwfSY2uPhs5c8SuyxnwOn9myWv2u2WiKqGj8@mail.gmail.com>
	<4D03307D.9000104@gmail.com>
Message-ID: <AANLkTi=UvoZ1N11+VodLeVJW4LTx19L4kTUd6VgBxjkT@mail.gmail.com>

Looking at the source code it seems that this is not currently possible. One
might be able to modify the code to do this but it would take a little bit
of time. You might want to ask Mathieu Parent (tts_commandline author) if he
has any suggestions.

Another option, depending upon what the circumstances are, is you could use
the "system" command to launch swift -f and record it to a wav, and then use
"playback" to play the wav file.

-MC

On Sat, Dec 11, 2010 at 12:04 AM, phone.bytes <phone.bytes at gmail.com> wrote:

>  okay MC, mod_tts_commandline has run for a several days without a
> problem.  Good.
>
> However, now we are trying to get this to speak from a text file instead of
> putting text on the command line.
> Cepstral Swift has a -f option to do this, but it does not work here.
>
> I have looked through the source, and am wondering if this is not supported
> in this mod.
>
> Here is a partial trace that I think supports my conclusions below.
>
> EXECUTE FreeTDM/1:1/7878030 speak(tts_commandline|Callie|-f/tmp/test.txt)
> 2010-12-11 00:26:58.683522 [DEBUG] switch_ivr_play_say.c:2237 OPEN TTS
> tts_commandline
> 2010-12-11 00:26:58.683522 [DEBUG] switch_ivr_play_say.c:2246 Raw Codec
> Activated
> 2010-12-11 00:26:58.683522 [DEBUG] mod_tts_commandline.c:147 Executing:
> swift -p audio/sampling-rate=8000 -n 'Callie' '-f/tmp/test.txt'  -o
> '/tmp/bd5bb8db-ffc2-4830-a971-24a1454c725c.tmp.wav'
> 2010-12-11 00:26:58.696525 [ERR] mod_sndfile.c:194 Error Opening File
> [/tmp/bd5bb8db-ffc2-4830-a971-24a1454c725c.tmp.wav] [System error : No such
> file or directory.]
> 2010-12-11 00:26:58.696525 [ERR] mod_tts_commandline.c:157 Failed to open
> file: /tmp/bd5bb8db-ffc2-4830-a971-24a1454c725c.tmp.wav
> 2010-12-11 00:26:58.696525 [DEBUG] switch_ivr_play_say.c:1935 Speaking
> text: -f/tmp/test.txt
> 2010-12-11 00:26:59.097509 [DEBUG] switch_ivr_play_say.c:2127 done speaking
> text
>
> It seems to almost build the swift command correctly.  However, the -f
> options is inside the single quotes, instead of outside of them like the -o
> (output file) is.
>
> I don't think it is finding the .txt file, as the log later shows that it
> is trying to speak the filename including the -f option, instead of the
> contents of the file.
>
> What do you think?
>
> Thanks
>
>
>
>
> On 12/7/2010 2:17 PM, Michael Collins wrote:
>
> Can you try the mod_tts_commandline way of doing it? The wiki talks about
> using Cepstral IIRC.
> -MC
>
> On Tue, Dec 7, 2010 at 1:12 PM, Phone <phone.bytes at gmail.com> wrote:
>
>> Just wondering if you have had any luck in resolving this issue.
>>
>> We are running Cent 5.5 on FreeSWITCH Version 1.0.head (git-8825b6e
>> 2010-11-28 17-15-39 -0500) with Cepstral 5.1 with the Callie voice.
>>
>> We have good audio for a short period of time, then suddenly the audio
>> is gone.  Watching the logs, it show that it is trying to play the
>> correct TTS, but it is silent.
>>
>> We have tried to unload and reload mod_cepstral, but the only way we
>> have been able to restore the audio is with a FreeSWITCH restart.
>>
>> These voices sound good, but with these issues, it is really not useable.
>>
>> Any Ideas?  Thanks
>>
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Mon Dec 13 23:04:41 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 13 Dec 2010 15:04:41 -0500
Subject: [Freeswitch-users] send dtmf on one leg bridge
References: <D10B61781EFA4CFB913F2FEAC7579B27@e1705>
	<AANLkTikhpiqsFbQHeWuxNE2EojeuFVgrdseBC5CU=308@mail.gmail.com>
Message-ID: <7555EA3AD7864DA9B52A6899DD7FF2B3@e1705>

yes,

I tried expand uuid_send_dtmf [uuid] 12345 at 120 once
answer done and ivr asks pin number but the CLI
says "ERR- no reply".
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, December 13, 2010 2:50 PM
  Subject: Re: [Freeswitch-users] send dtmf on one leg bridge


  Have you tried uuid_send_dtmf?
  -MC


  On Sun, Dec 12, 2010 at 12:30 PM, Madovsky <infos at madovsky.org> wrote:

    Is it possible to send dtfmf via
    send_dtmf from CLI on a one leg bridge ?
    it's for enter a pin conference and I can't use
    any RFC on my sip phone.

    Thanks

    Franck

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






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


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From david.ponzone at ipeva.fr  Mon Dec 13 23:38:26 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 13 Dec 2010 21:38:26 +0100
Subject: [Freeswitch-users] Incomplete offer/answer debugging
In-Reply-To: <4D067272.4060602@j-cope.com>
References: <4D067272.4060602@j-cope.com>
Message-ID: <261FAA44-CAD2-4460-A5E1-BFF2ACFDB03B@ipeva.fr>

The brand/modem of your Wifi phone may be useful.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 13/12/2010 ? 20:22, Jerre Cope a ?crit :

> Hello!  Newbie question.
> 
> My wifi SIP phone has stopped answering lately and I don't know if I've 
> induced a configuration problem by making freeswitch current, or whether 
> I've staticed the phone firmware.  I have the wifi handset set up as a 
> typical extension 1000.  I've got a Twinkle softphone on ext 1001 and I 
> can call that extension from 1000, but when I call from 1001 to ext 
> 1000, the handset  returns the nua_media_error.  Both extensions are on 
> the same network.  I've posted a pastebin of log level 9 sip trace on 
> pastebin 14768
> 
> I could use some help understanding the log so I can correct my 
> nua_media_error negotiation problem.  I started the log early in the 
> conversation, where it negotiates past the IP to the user registration, 
> but I don't see a clear reason for the "BYE" on line 526.
> 
> Thanks for FreeSWITCH!
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jerre at j-cope.com  Mon Dec 13 23:48:58 2010
From: jerre at j-cope.com (Jerre Cope)
Date: Mon, 13 Dec 2010 14:48:58 -0600
Subject: [Freeswitch-users] Incomplete offer/answer debugging
Message-ID: <4D0686BA.3040606@j-cope.com>

Sorry.
The wifi phone is a Loctec WP04.   Not Cisco quality, but I needed a 
phone the technician could drop and kick across the parking lot.  It's 
been working fine, so I bought 6 more of them.  Picture and manual here 
if you're curious.  I doubt it will be very helpful: 
http://www.yippz.com/wp04-wifi-phone.html



From johnrose at comtex.net  Mon Dec 13 23:48:59 2010
From: johnrose at comtex.net (John Rose)
Date: Mon, 13 Dec 2010 13:48:59 -0700
Subject: [Freeswitch-users] SIP MESSAGE requests
In-Reply-To: <AANLkTikP1qHa4-PtMjseJHoDxvJ5Fo7a5mfWa2aTA-Kc@mail.gmail.com>
References: <004201cb9662$d770dad0$86529070$@comtex.net>	<4CFF3EC7.5030605@gmail.com>	<AANLkTin3gep6cpQoV0b4NEV2Tqgxe6LGyVJE5ymu46NA@mail.gmail.com>	<B6823144-18AC-42BA-BF9E-9027255A45D0@avgs.ca>
	<AANLkTikP1qHa4-PtMjseJHoDxvJ5Fo7a5mfWa2aTA-Kc@mail.gmail.com>
Message-ID: <004501cb9b07$2ba2d880$82e88980$@comtex.net>

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> Subject: Re: [Freeswitch-users] SIP MESSAGE requests
> 
> While FS does handle MESSAGE the interface has always struck me as a bit
> odd...
> 
> For example, why don't FS messages hit the dialplan to provide for some
> ability to route them if I want to?
> 
> In my case I typically use FS as an SBC facing carriers...  If a carrier
sends me a
> MESSAGE FS will generate an event.  While I *could* do something with
that,
> I suppose, I'd much rather have it hit the dialplan where I can use some
logic
> to handle it.  I could see at least the following:
> 
> - Post to an HTTP server
> - Route to another SIP endpoint
> - LUA!
> - Generate an event (as it does now)
> 
> Smells like a bounty...
> 

Just curious - I'm sending SIP MESSAGE requests from one FS box to another.
Works well. But the receiving box isn't generating an ESL event but does
reply 200 OK. Also, the receiver box shows an "Invalid profile x.x.x.x".  If
the MESSAGE doesn't propagate through the dialplan how do I get it to
generate *some* ESL event when I can grab it and do something like i.e.
strip out the MESSAGE text?

John



From david.varnes at gmail.com  Tue Dec 14 00:47:03 2010
From: david.varnes at gmail.com (david varnes)
Date: Tue, 14 Dec 2010 08:47:03 +1100
Subject: [Freeswitch-users] Fwd: Error Loading mod_java.so
In-Reply-To: <AANLkTimYd+7vrObud9iJ5t81u0pdO9+LtT2GC5Nar2KE@mail.gmail.com>
References: <AANLkTikz9gbLdKKx=aGJHxUZgwt4MqnLvVfad86PdyWV@mail.gmail.com>
	<AANLkTimYd+7vrObud9iJ5t81u0pdO9+LtT2GC5Nar2KE@mail.gmail.com>
Message-ID: <AANLkTim0G_FCdrmxx6CFBH3Uu9URhr6EG_rpF_zmP2e=@mail.gmail.com>

How are you configuring/managing your classpath ?

davidv

On 13 December 2010 22:39, srinivasula reddy
<srinivas.ksvreddy at gmail.com> wrote:
>
> Hi All,
>
> Currently i am using JDK1.06.0_23 and jre also same version for using
> mod_java module in freeswith,
>
> i can able to generate mod_java.so and freeswitch.jar. and building and
> installation had succussful without any proble.
>
> when i am trying to run the freeswitch, i am gettig the following error.
>
>
> Exception in thread "Thread-0" java.lang.NoClassDefFoundError:
> net/cog/fs/system/Control
> Caused by: java.lang.ClassNotFoundException: net.cog.fs.system.Control
> ??????? at java.net.URLClassLoader$1.run(URLClassLoader.java:217)
> ??????? at java.security.AccessController.doPrivileged(Native Method)
> ??????? at java.net.URLClassLoader.findClass(URLClassLoader.java:205)
> ??????? at java.lang.ClassLoader.loadClass(ClassLoader.java:319)
> ??????? at sun.misc.Launcher$AppClassLoader.loadClass(Launcher.java:294)
> ??????? at java.lang.ClassLoader.loadClass(ClassLoader.java:264)
> ??????? at java.lang.ClassLoader.loadClassInternal(ClassLoader.java:332)
> 2010-12-13 17:07:58.193373 [CRIT] switch_loadable_module.c:928 Error Loading
> module /usr/local/freeswitch/mod/mod_java.so
> **Module load routine returned an error**
>
>
>
> and please find the following java.conf.xml file.
>
> <configuration name="java.conf" description="Java Plug-Ins">
> ? <!-- Path to the Java 1.6 virtual machine to use -->
> ?<!--? <javavm
> path="/usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre/lib/x86_64/client/libjvm.so"/>
> -->
> ? <javavm
> path="/usr/lib/jvm/java-1.6.0-openjdk-1.6.0.0.x86_64/jre/lib/amd64/server/libjvm.so"/>
> ? <!-- Options to pass to Java -->
> ? <options>
> ??? <!-- Your class path (make sure freeswitch.jar is on it) -->
> ??? <option
> value="-Djava.class.path=/usr/local/freeswitch/scripts/freeswitch.jar
> "/>
> ??? <!-- Enable remote debugging -->
> ??? <!-- <option
> value="-agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address=127.0.0.1:8001"/>
> -->
> ? </options>
> ? <startup class="net/cog/fs/system/Control" method="startup" arg="start up
> arg"/>
> ? <shutdown class="net/cog/fs/system/Control" method="shutdown"
> arg="shutdown arg"/>
> </configuration>
>
>
> Any Idea why i am getting the error?
>
> Thanks
> Srinivasula Reddy K
>
>
>
> --
> Srinivasula Reddy K
>


-- 
david varnes


From davidjbrazier at gmail.com  Tue Dec 14 02:43:25 2010
From: davidjbrazier at gmail.com (David Brazier)
Date: Mon, 13 Dec 2010 23:43:25 +0000
Subject: [Freeswitch-users] Cepstral + FS question
In-Reply-To: <AANLkTi=agVz=fgCv+EN5PT0k5QRq8Cn=7AV2J3ZE70pT@mail.gmail.com>
References: <AANLkTi=N4yG-z5FLUo_hn8YCqDZezSpTcmn74=Qv7c8u@mail.gmail.com>
	<AANLkTimt4N9jGQ9aNWGCGYMk3L0pxJWhSOSPzuJ4h=N4@mail.gmail.com>
	<AANLkTi=agVz=fgCv+EN5PT0k5QRq8Cn=7AV2J3ZE70pT@mail.gmail.com>
Message-ID: <AANLkTinHNQyuhUWUTrOWwVYeu+EwXnPfkFAkoNJNuewZ@mail.gmail.com>

I meant Cepstral ports.  You might be better asking Cepstral for guidance.

On Mon, Dec 13, 2010 at 5:32 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
> Thank you for your response.
> By licenses, do you mean CPU license or Cepstral ports?
> Say if I have 30 simultaneous calls (some part using direct WAV and some
> using session speak), what algorithm I use to determine # or ports?
> My server configuration:
> Windows 2008 server 32-bit
> 4 GB RAM
> Intel Xeon CPU X3220 @ 2.40GHz?2.40GHz
> Malay
>
> On Mon, Dec 13, 2010 at 8:55 AM, David Brazier <davidjbrazier at gmail.com>
> wrote:
>>
>> On Tue, Dec 7, 2010 at 12:38 AM, Malay Thakershi <mthakershi at gmail.com>
>> wrote:
>> > Hello, it would be great help if someone who has used Cepstral from FS
>> > can
>> > share their views.
>> > 1. I just have one Cepstral Allison voice license (1 port) on my FS
>> > server.
>> > I use swift command to convert text files to WAV which are then played
>> > by
>> > mod_managed in FS call process. First question is regarding limitations
>> > on
>> > simultaneous conversions (TXT to WAV) using swift command. If I have
>> > multiple threads doing this, will there be any degradation because of
>> > Cepstral? I tried running two BAT files with 3 commands each. But
>> > running
>> > them simultaneously or separate produced same outcome. Does anyone know
>> > when
>> > Cepstral licensing kicks in and starts degrading quality ( or worse
>> > inserting "not licensed") prompt?
>>
>> The synthesis via the swift command is much faster than real time
>> (i.e. the time it takes to play the WAV) and is only restricted by CPU
>> speed and licenses. ?You'd need to ask Cepstral the details, but my
>> simple tests have shown that it only uses one CPU per command though
>> some of the processing of multiple command is on multiple CPUs. ?But I
>> think the licensing mechanism prevents multiple commands running
>> completely simultaneously on multiple CPUs. ?I think the "not
>> licensed" speech is only inserted when you have no license at all -
>> all that happens if you try to run multiple commands is that they are
>> just delayed until ?there is a free license. ?I don't think quality
>> will ever be degraded - it's just a question of CPUs and licenses that
>> will determine your throughput.
>>
>> > 2. When I call session speak from mod_managed (or stream file) after
>> > selecting Allison / Cepstral as my voice, does Cepstral engine interfere
>> > with quality of the playback? If yes, when will I see it and how can I
>> > produce their effects?
>>
>> Not unless something in Cepstral or FS is going badly wrong!
>>
>> > 3. What is the sensible number of ports (from Cepstral) I should be
>> > prepared
>> > to buy if findings in the previous points imposes significant
>> > limitations?
>>
>> Depends on the length of your prompts and number and speed of CPUs and
>> anticipated simultaneous calls. ?Try testing the time for a typical
>> prompt and work it out from there.
>>
>> David
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Tue Dec 14 03:47:11 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 14 Dec 2010 00:47:11 +0000
Subject: [Freeswitch-users] send dtmf on one leg bridge
In-Reply-To: <7555EA3AD7864DA9B52A6899DD7FF2B3@e1705>
References: <D10B61781EFA4CFB913F2FEAC7579B27@e1705>
	<AANLkTikhpiqsFbQHeWuxNE2EojeuFVgrdseBC5CU=308@mail.gmail.com>
	<7555EA3AD7864DA9B52A6899DD7FF2B3@e1705>
Message-ID: <AANLkTikFXnH1TYEUe4D0uM6q+WMnvWqCQ+19n0bSZYM_@mail.gmail.com>

What about  uuid_send_dtmf directly? i.e. not through expand.

-Steve


On 13 December 2010 20:04, Madovsky <infos at madovsky.org> wrote:
> yes,
>
> I tried expand uuid_send_dtmf [uuid] 12345 at 120 once
> answer done and ivr asks pin number but the CLI
> says "ERR- no reply".
>
> ----- Original Message -----
> From: Michael Collins
> To: FreeSWITCH Users Help
> Sent: Monday, December 13, 2010 2:50 PM
> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
> Have you tried uuid_send_dtmf?
> -MC
>
> On Sun, Dec 12, 2010 at 12:30 PM, Madovsky <infos at madovsky.org> wrote:
>>
>> Is it possible to send dtfmf via
>> send_dtmf from CLI on a one leg bridge ?
>> it's for enter a pin conference and I can't use
>> any RFC on my sip phone.
>>
>> Thanks
>>
>> Franck
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> ________________________________
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Tue Dec 14 04:18:58 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 13 Dec 2010 20:18:58 -0500
Subject: [Freeswitch-users] send dtmf on one leg bridge
References: <D10B61781EFA4CFB913F2FEAC7579B27@e1705><AANLkTikhpiqsFbQHeWuxNE2EojeuFVgrdseBC5CU=308@mail.gmail.com><7555EA3AD7864DA9B52A6899DD7FF2B3@e1705>
	<AANLkTikFXnH1TYEUe4D0uM6q+WMnvWqCQ+19n0bSZYM_@mail.gmail.com>
Message-ID: <05F742BEE64A443C93952DD22DC7E84C@e1705>

because I need to use variables inside the api request
is it make a difference ?

----- Original Message ----- 
From: "Steven Ayre" <steveayre at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, December 13, 2010 7:47 PM
Subject: Re: [Freeswitch-users] send dtmf on one leg bridge


> What about  uuid_send_dtmf directly? i.e. not through expand.
>
> -Steve
>
>
> On 13 December 2010 20:04, Madovsky <infos at madovsky.org> wrote:
>> yes,
>>
>> I tried expand uuid_send_dtmf [uuid] 12345 at 120 once
>> answer done and ivr asks pin number but the CLI
>> says "ERR- no reply".
>>
>> ----- Original Message -----
>> From: Michael Collins
>> To: FreeSWITCH Users Help
>> Sent: Monday, December 13, 2010 2:50 PM
>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
>> Have you tried uuid_send_dtmf?
>> -MC
>>
>> On Sun, Dec 12, 2010 at 12:30 PM, Madovsky <infos at madovsky.org> wrote:
>>>
>>> Is it possible to send dtfmf via
>>> send_dtmf from CLI on a one leg bridge ?
>>> it's for enter a pin conference and I can't use
>>> any RFC on my sip phone.
>>>
>>> Thanks
>>>
>>> Franck
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> ________________________________
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From infos at madovsky.org  Tue Dec 14 06:49:22 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 13 Dec 2010 22:49:22 -0500
Subject: [Freeswitch-users] send dtmf to IVR from CLI
Message-ID: <FA5EF544C81B4BD4B98234103E0F6151@e1705>

is it possible to send dtmf to an IVR from CLI
if I know the uuid of leg A only ?

Thanks

F
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From brian at freeswitch.org  Tue Dec 14 07:01:38 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 13 Dec 2010 22:01:38 -0600
Subject: [Freeswitch-users] send dtmf to IVR from CLI
In-Reply-To: <FA5EF544C81B4BD4B98234103E0F6151@e1705>
References: <FA5EF544C81B4BD4B98234103E0F6151@e1705>
Message-ID: <5BA5768C-3F4C-462D-9027-25067431C21C@freeswitch.org>

show api

search for dtmf

you'll find uuid_send_dtmf

/b

On Dec 13, 2010, at 9:49 PM, Madovsky wrote:

> is it possible to send dtmf to an IVR from CLI
> if I know the uuid of leg A only ?
>  
> Thanks
>  
> F

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From msc at freeswitch.org  Tue Dec 14 08:04:33 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Dec 2010 21:04:33 -0800
Subject: [Freeswitch-users] send dtmf to IVR from CLI
In-Reply-To: <FA5EF544C81B4BD4B98234103E0F6151@e1705>
References: <FA5EF544C81B4BD4B98234103E0F6151@e1705>
Message-ID: <AANLkTikUYFOhyoaobDM75S3ana3-gjs-J4=sQ=EGZVD6@mail.gmail.com>

Why do you know only the A leg? Furthermore, if there is a bridge then the
other leg uuid should be accessible. Just do "uuid_dump <uuid_of_a_leg>" and
see if you see the other leg's uuid in there somewhere. (I think it's
"other_leg_uuid") Then you can grab that uuid thusly:

uuid_getvar <uuid_a_leg> other_leg_uuid

-MC

On Mon, Dec 13, 2010 at 7:49 PM, Madovsky <infos at madovsky.org> wrote:

>  is it possible to send dtmf to an IVR from CLI
> if I know the uuid of leg A only ?
>
> Thanks
>
> F
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Dec 14 10:07:02 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Dec 2010 23:07:02 -0800
Subject: [Freeswitch-users] a2billing and freeswitch
In-Reply-To: <COL122-W4030C97574705F9B9DC69FB5120@phx.gbl>
References: <COL122-W4030C97574705F9B9DC69FB5120@phx.gbl>
Message-ID: <AANLkTimNVpoxuXMT8iZLheh9WL8yLZEhodEKahaJr2dV@mail.gmail.com>

FWIW you can use AstPP with FreeSWITCH:
http://www.astpp.org/?q=node/270

A2billing is probably doable with FreeSWITCH but it would require an
understanding both of Asterisk and FreeSWITCH. My guess is a bounty of a few
thousand dollars might motivate a talented programmer to do the work.

-MC

On Mon, Dec 13, 2010 at 3:16 AM, Rafqat . <rafonline at hotmail.com> wrote:

>
> Hi
>
> I was wondering if a2billing can be setup to work with freeswitch?
>
> If so, can anyone please point me to some documentation on how this can be
> done?
>
> Cheers
>
> Raf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jerre at j-cope.com  Tue Dec 14 10:04:04 2010
From: jerre at j-cope.com (Jerre Cope)
Date: Tue, 14 Dec 2010 01:04:04 -0600
Subject: [Freeswitch-users] Incomplete offer/answer debugging
Message-ID: <4D0716E4.6040501@j-cope.com>

OK, I have a more concise pastebin up (14773) where I gather I must have 
an sdb problem with my loctec phone (line 36).  I'm hoping there is a 
NDLB-something I can use to fix it.


From msc at freeswitch.org  Tue Dec 14 10:09:08 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Dec 2010 23:09:08 -0800
Subject: [Freeswitch-users] DTMF and dynamic conference
In-Reply-To: <4A6B12C1D6794B4DA9FD15827BD6812C@e1705>
References: <4A6B12C1D6794B4DA9FD15827BD6812C@e1705>
Message-ID: <AANLkTi=RNUT8ppOzmWo0TqxmyDhVZEaabtz1SkHx8fcy@mail.gmail.com>

Do you actually try calling this with a real telephone to make sure that the
extension is doing what you think it is doing?
-MC

On Mon, Dec 13, 2010 at 11:14 AM, Madovsky <infos at madovsky.org> wrote:

>  I'm trying the default conference in
> autoload_configs/conference.conf.xml,
> commented out the pin number line,
>
> but now when the ivr asks the pin number, for testing I tried
> to do this :
>
>         <extension name="nb_conferences">
>                 <condition field="${sip_to_uri}"
> expression="^(999)@$${domain}$">
>                         <action application="answer"/>
>                         <action application="info"/>
>                         <!-- <action application="conference" data="
> confname at profilename+flags{mute|deaf|waste|moderator}+[conference<confname at profilename+flags%7Bmute%7Cdeaf%7Cwaste%7Cmoderator%7D+%5Bconference>pin number]"> -->
>                         <action application="conference" data="
> $1-$${domain}@default"/ <$1-$$%7Bdomain%7D at default%22/>>
>                         <action application="send_dtmf" data="12345 at 120"/<12345 at 120%22/>
> >
>                 </condition>
>         </extension>
> and this on CLI:
>
> expand uuid_send_dtmf ${uuid} 12345
>
>
> without success.
> How the default conference example accept DTMF ?
>
> Thanks
>
> F
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Tue Dec 14 10:12:38 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 14 Dec 2010 07:12:38 +0000
Subject: [Freeswitch-users] send dtmf on one leg bridge
In-Reply-To: <05F742BEE64A443C93952DD22DC7E84C@e1705>
References: <D10B61781EFA4CFB913F2FEAC7579B27@e1705>
	<AANLkTikhpiqsFbQHeWuxNE2EojeuFVgrdseBC5CU=308@mail.gmail.com>
	<7555EA3AD7864DA9B52A6899DD7FF2B3@e1705>
	<AANLkTikFXnH1TYEUe4D0uM6q+WMnvWqCQ+19n0bSZYM_@mail.gmail.com>
	<05F742BEE64A443C93952DD22DC7E84C@e1705>
Message-ID: <AANLkTikL-ZWCyXhfHmW4daTdT1q-zARmEJZ3s7kFXiv+@mail.gmail.com>

Perhaps the UUID variable doesn't contain the correct value though...

Try uuid_send_dtmf using the known UUID yourself. That'll show you
whether uuid_send_dtmf works and the problem is in the expand api, or
whether it's uuid_send_dtmf that's having a problem.

-Steve


On 14 December 2010 01:18, Madovsky <infos at madovsky.org> wrote:
> because I need to use variables inside the api request
> is it make a difference ?
>
> ----- Original Message -----
> From: "Steven Ayre" <steveayre at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, December 13, 2010 7:47 PM
> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
>
>
>> What about ?uuid_send_dtmf directly? i.e. not through expand.
>>
>> -Steve
>>
>>
>> On 13 December 2010 20:04, Madovsky <infos at madovsky.org> wrote:
>>> yes,
>>>
>>> I tried expand uuid_send_dtmf [uuid] 12345 at 120 once
>>> answer done and ivr asks pin number but the CLI
>>> says "ERR- no reply".
>>>
>>> ----- Original Message -----
>>> From: Michael Collins
>>> To: FreeSWITCH Users Help
>>> Sent: Monday, December 13, 2010 2:50 PM
>>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
>>> Have you tried uuid_send_dtmf?
>>> -MC
>>>
>>> On Sun, Dec 12, 2010 at 12:30 PM, Madovsky <infos at madovsky.org> wrote:
>>>>
>>>> Is it possible to send dtfmf via
>>>> send_dtmf from CLI on a one leg bridge ?
>>>> it's for enter a pin conference and I can't use
>>>> any RFC on my sip phone.
>>>>
>>>> Thanks
>>>>
>>>> Franck
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> ________________________________
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From steveayre at gmail.com  Tue Dec 14 10:14:16 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 14 Dec 2010 07:14:16 +0000
Subject: [Freeswitch-users] DTMF and dynamic conference
In-Reply-To: <4A6B12C1D6794B4DA9FD15827BD6812C@e1705>
References: <4A6B12C1D6794B4DA9FD15827BD6812C@e1705>
Message-ID: <AANLkTim8EhLdHTLWNh4LPVFpTary=gsQ9LPHzTvauM0X@mail.gmail.com>

and this on CLI:

expand uuid_send_dtmf ${uuid} 12345

Where is this ${uuid} variable set?

-Steve


On 13 December 2010 19:14, Madovsky <infos at madovsky.org> wrote:
> I'm trying the default conference in autoload_configs/conference.conf.xml,
> commented out the pin number line,
>
> but now when the ivr asks the pin number, for testing I tried
> to do this :
>
> ??????? <extension name="nb_conferences">
> ??????????????? <condition field="${sip_to_uri}"
> expression="^(999)@$${domain}$">
> ??????????????????????? <action application="answer"/>
> ??????????????????????? <action application="info"/>
> ??????????????????????? <!-- <action application="conference"
> data="confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin
> number]"> -->
> ??????????????????????? <action application="conference"
> data="$1-$${domain}@default"/>
> ??????????????????????? <action application="send_dtmf" data="12345 at 120"/>
> ??????????????? </condition>
> ??????? </extension>
> and this on CLI:
>
> expand uuid_send_dtmf ${uuid} 12345
>
>
> without success.
> How the default conference example accept DTMF ?
>
> Thanks
> F
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Tue Dec 14 10:15:57 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 14 Dec 2010 07:15:57 +0000
Subject: [Freeswitch-users] DTMF and dynamic conference
In-Reply-To: <4A6B12C1D6794B4DA9FD15827BD6812C@e1705>
References: <4A6B12C1D6794B4DA9FD15827BD6812C@e1705>
Message-ID: <AANLkTimL1WxWAy9ZmucxH2XRpvHtXT33U9rwpV7=sVXP@mail.gmail.com>

AFAIK
   <action application="send_dtmf" data="12345 at 120"/>
won't occur until
   <action application="conference"> data="$1-$${domain}@default"/>
finishes, i.e. when the user leaves the conference.

Is that what you are expecting to occur?

-Steve


On 13 December 2010 19:14, Madovsky <infos at madovsky.org> wrote:
> I'm trying the default conference in autoload_configs/conference.conf.xml,
> commented out the pin number line,
>
> but now when the ivr asks the pin number, for testing I tried
> to do this :
>
> ??????? <extension name="nb_conferences">
> ??????????????? <condition field="${sip_to_uri}"
> expression="^(999)@$${domain}$">
> ??????????????????????? <action application="answer"/>
> ??????????????????????? <action application="info"/>
> ??????????????????????? <!-- <action application="conference"
> data="confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin
> number]"> -->
> ??????????????????????? <action application="conference"
> data="$1-$${domain}@default"/>
> ??????????????????????? <action application="send_dtmf" data="12345 at 120"/>
> ??????????????? </condition>
> ??????? </extension>
> and this on CLI:
>
> expand uuid_send_dtmf ${uuid} 12345
>
>
> without success.
> How the default conference example accept DTMF ?
>
> Thanks
> F
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Tue Dec 14 10:41:32 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 14 Dec 2010 02:41:32 -0500
Subject: [Freeswitch-users] DTMF and dynamic conference
References: <4A6B12C1D6794B4DA9FD15827BD6812C@e1705>
	<AANLkTimL1WxWAy9ZmucxH2XRpvHtXT33U9rwpV7=sVXP@mail.gmail.com>
Message-ID: <23780DB27B0B47149145C80C0D20AA59@e1705>

no, I was wrong.
my problem is I try to send dtmf to conference IVR
so it's one leg only.

----- Original Message ----- 
From: "Steven Ayre" <steveayre at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, December 14, 2010 2:15 AM
Subject: Re: [Freeswitch-users] DTMF and dynamic conference


AFAIK
   <action application="send_dtmf" data="12345 at 120"/>
won't occur until
   <action application="conference"> data="$1-$${domain}@default"/>
finishes, i.e. when the user leaves the conference.

Is that what you are expecting to occur?

-Steve


On 13 December 2010 19:14, Madovsky <infos at madovsky.org> wrote:
> I'm trying the default conference in autoload_configs/conference.conf.xml,
> commented out the pin number line,
>
> but now when the ivr asks the pin number, for testing I tried
> to do this :
>
> <extension name="nb_conferences">
> <condition field="${sip_to_uri}"
> expression="^(999)@$${domain}$">
> <action application="answer"/>
> <action application="info"/>
> <!-- <action application="conference"
> data="confname at profilename+flags{mute|deaf|waste|moderator}+[conference 
> pin
> number]"> -->
> <action application="conference"
> data="$1-$${domain}@default"/>
> <action application="send_dtmf" data="12345 at 120"/>
> </condition>
> </extension>
> and this on CLI:
>
> expand uuid_send_dtmf ${uuid} 12345
>
>
> without success.
> How the default conference example accept DTMF ?
>
> Thanks
> F
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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



From infos at madovsky.org  Tue Dec 14 10:42:42 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 14 Dec 2010 02:42:42 -0500
Subject: [Freeswitch-users] DTMF and dynamic conference
References: <4A6B12C1D6794B4DA9FD15827BD6812C@e1705>
	<AANLkTi=RNUT8ppOzmWo0TqxmyDhVZEaabtz1SkHx8fcy@mail.gmail.com>
Message-ID: <40AE384643F0467C8DAE944291A233FC@e1705>

I tried the default conference dialplan with a real phone and works well.

  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, December 14, 2010 2:09 AM
  Subject: Re: [Freeswitch-users] DTMF and dynamic conference


  Do you actually try calling this with a real telephone to make sure that the extension is doing what you think it is doing?
  -MC


  On Mon, Dec 13, 2010 at 11:14 AM, Madovsky <infos at madovsky.org> wrote:

    I'm trying the default conference in autoload_configs/conference.conf.xml,
    commented out the pin number line,

    but now when the ivr asks the pin number, for testing I tried
    to do this :

            <extension name="nb_conferences">
                    <condition field="${sip_to_uri}" expression="^(999)@$${domain}$">
                            <action application="answer"/>
                            <action application="info"/>
                            <!-- <action application="conference" data="confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin number]"> -->
                            <action application="conference" data="$1-$${domain}@default"/>
                            <action application="send_dtmf" data="12345 at 120"/>
                    </condition>
            </extension>

    and this on CLI:

    expand uuid_send_dtmf ${uuid} 12345


    without success.
    How the default conference example accept DTMF ?

    Thanks

    F

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From msc at freeswitch.org  Tue Dec 14 10:47:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 13 Dec 2010 23:47:26 -0800
Subject: [Freeswitch-users] Cepstral + FS question
In-Reply-To: <AANLkTinHNQyuhUWUTrOWwVYeu+EwXnPfkFAkoNJNuewZ@mail.gmail.com>
References: <AANLkTi=N4yG-z5FLUo_hn8YCqDZezSpTcmn74=Qv7c8u@mail.gmail.com>
	<AANLkTimt4N9jGQ9aNWGCGYMk3L0pxJWhSOSPzuJ4h=N4@mail.gmail.com>
	<AANLkTi=agVz=fgCv+EN5PT0k5QRq8Cn=7AV2J3ZE70pT@mail.gmail.com>
	<AANLkTinHNQyuhUWUTrOWwVYeu+EwXnPfkFAkoNJNuewZ@mail.gmail.com>
Message-ID: <AANLkTim2agpbEGqy=UyHg_Jzcd0kX9nT0T4R+=+0byu0@mail.gmail.com>

Cepstral can help with this. They have a licensing server that runs and
whenever there's a swift instance then a "port" is in use. I don't know what
happens if you try to go over, but I do believe that there is a command line
option that lets you poll the license server so you can see how many ports
are in use.

-MC

On Mon, Dec 13, 2010 at 3:43 PM, David Brazier <davidjbrazier at gmail.com>wrote:

> I meant Cepstral ports.  You might be better asking Cepstral for guidance.
>
> On Mon, Dec 13, 2010 at 5:32 PM, Malay Thakershi <mthakershi at gmail.com>
> wrote:
> > Thank you for your response.
> > By licenses, do you mean CPU license or Cepstral ports?
> > Say if I have 30 simultaneous calls (some part using direct WAV and some
> > using session speak), what algorithm I use to determine # or ports?
> > My server configuration:
> > Windows 2008 server 32-bit
> > 4 GB RAM
> > Intel Xeon CPU X3220 @ 2.40GHz 2.40GHz
> > Malay
> >
>
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From shamun.toha at gmail.com  Tue Dec 14 10:53:40 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Tue, 14 Dec 2010 08:53:40 +0100
Subject: [Freeswitch-users] Timer for mod_spidermonkey impossible ?
Message-ID: <AANLkTinVAOi9qrx0C8upJ+f1X-Jfjxs=xoiyw3Lvu5=_@mail.gmail.com>

Hello,

I have a test script like this. But is it possible to make a
setTimeout(function(){ console_log("Hello world") }, 1000); ??  Like a timer
which keep running with interval of 1 second ? Where i can do some Ajax ???

helloworld.js
=========
var argv;

function on_hangup(hup_session, how)
{
   console_log(
           "err", how + " HOOK" + "\n" +
           "hello world" + session.getVariable("test") + "\n"
    );

    exit();
  );

session.setHangupHook(on_hangup);



Thanks & Regards
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From msc at freeswitch.org  Tue Dec 14 11:12:27 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 14 Dec 2010 00:12:27 -0800
Subject: [Freeswitch-users] DTMF and dynamic conference
In-Reply-To: <40AE384643F0467C8DAE944291A233FC@e1705>
References: <4A6B12C1D6794B4DA9FD15827BD6812C@e1705>
	<AANLkTi=RNUT8ppOzmWo0TqxmyDhVZEaabtz1SkHx8fcy@mail.gmail.com>
	<40AE384643F0467C8DAE944291A233FC@e1705>
Message-ID: <AANLkTinriGsa6rLKtOk0_NusmZ1Wto8KTTOdTe4sKf=s@mail.gmail.com>

Madovsky,

I thing maybe don't understand what you are trying to do. Do you simply want
to have a conference that requires the caller to enter a PIN? Or are you
trying to send a caller to a conference that has a PIN but you want to send
the PIN externally?

In case it's the latter you can do the following...
Create a conference locked by a PIN. In the logs below I used conf
3300 at default and 1234 as the PIN. Then I called 3300 from user 1002:

2010-12-14 00:00:21.518045 [NOTICE] switch_channel.c:784 New Channel
sofia/internal/1002 at 10.15.0.94 [09cad98f-6e12-45fa-9f01-e7d90b9ea771]
2010-12-14 00:00:21.520627 [INFO] mod_dialplan_xml.c:331 Processing Michael
<1002>->3000 in context default
2010-12-14 00:00:21.530579 [NOTICE] mod_dptools.c:920 Channel
[sofia/internal/1002 at 10.15.0.94] has been answered

Then I did "show channels" to get the uuid:

show channels
uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid
09cad98f-6e12-45fa-9f01-e7d90b9ea771,inbound,2010-12-14
00:00:21,1292313621,sofia/internal/1002 at 10.15.0.94
,CS_EXECUTE,Michael,1002,10.15.30.10,3000,conference,3000-10.15.0.94 at default
+1234,XML,default,PCMU,8000,64000,PCMU,8000,64000,,ElToro2.FreePBXV3,,,ACTIVE,,,,

1 total.

Then I did "uuid_recv_dtmf <uuid> 1234" like this:
freeswitch at internal> uuid_recv_dtmf 09cad98f-6e12-45fa-9f01-e7d90b9ea771
1234
-ERR no reply

freeswitch at internal> 2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485
sofia/internal/1002 at 10.15.0.94 binding '0' to 'mute'
2010-12-14 00:00:41.505775 [INFO] switch_ivr_async.c:162 Digit parser
mod_conference: Setting realm to conf
2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/
1002 at 10.15.0.94 binding '*' to 'deaf mute'
2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/
1002 at 10.15.0.94 binding '9' to 'energy up'
2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/
1002 at 10.15.0.94 binding '8' to 'energy equ'
2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/
1002 at 10.15.0.94 binding '7' to 'energy dn'
2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/
1002 at 10.15.0.94 binding '3' to 'vol talk up'
2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/
1002 at 10.15.0.94 binding '2' to 'vol talk zero'
2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/
1002 at 10.15.0.94 binding '1' to 'vol talk dn'
2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/
1002 at 10.15.0.94 binding '6' to 'vol listen up'
2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/
1002 at 10.15.0.94 binding '5' to 'vol listen zero'
2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/
1002 at 10.15.0.94 binding '4' to 'vol listen dn'
2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/
1002 at 10.15.0.94 binding '#' to 'hangup'

As you can see you do get "-ERR No Reply" but the operation itself works
just fine.
-MC

On Mon, Dec 13, 2010 at 11:42 PM, Madovsky <infos at madovsky.org> wrote:

>  I tried the default conference dialplan with a real phone and works well.
>
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, December 14, 2010 2:09 AM
> *Subject:* Re: [Freeswitch-users] DTMF and dynamic conference
>
> Do you actually try calling this with a real telephone to make sure that
> the extension is doing what you think it is doing?
> -MC
>
> On Mon, Dec 13, 2010 at 11:14 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  I'm trying the default conference in
>> autoload_configs/conference.conf.xml,
>> commented out the pin number line,
>>
>> but now when the ivr asks the pin number, for testing I tried
>> to do this :
>>
>>         <extension name="nb_conferences">
>>                 <condition field="${sip_to_uri}"
>> expression="^(999)@$${domain}$">
>>                         <action application="answer"/>
>>                         <action application="info"/>
>>                         <!-- <action application="conference" data="
>> confname at profilename+flags{mute|deaf|waste|moderator}+[conference<confname at profilename+flags%7Bmute%7Cdeaf%7Cwaste%7Cmoderator%7D+%5Bconference>pin number]"> -->
>>                         <action application="conference" data="
>> $1-$${domain}@default"/ <$1-$$%7Bdomain%7D at default%22/>>
>>                         <action application="send_dtmf" data="12345 at 120"/<12345 at 120%22/>
>> >
>>                 </condition>
>>         </extension>
>> and this on CLI:
>>
>> expand uuid_send_dtmf ${uuid} 12345
>>
>>
>> without success.
>> How the default conference example accept DTMF ?
>>
>> Thanks
>>
>> F
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From shamun.toha at gmail.com  Tue Dec 14 11:30:08 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Tue, 14 Dec 2010 09:30:08 +0100
Subject: [Freeswitch-users] Fwd: Timer for mod_spidermonkey impossible ?
In-Reply-To: <AANLkTinVAOi9qrx0C8upJ+f1X-Jfjxs=xoiyw3Lvu5=_@mail.gmail.com>
References: <AANLkTinVAOi9qrx0C8upJ+f1X-Jfjxs=xoiyw3Lvu5=_@mail.gmail.com>
Message-ID: <AANLkTi=pJw4d1jcL=m_6d4DV_Jc9wyDenMQhXH8FAKcQ@mail.gmail.com>

BUG???? i tried this but i get this.

2010-12-14 08:27:55.142532 [ERR] helloworld.js:3 ReferenceError: setTimeout
is not defined


function timers()
{
    setTimeout(function() {
       console_log("err", '------------------ Hello world --------- Time
Time Timer...:) ');
    }, 1000);
}
timers();


---------- Forwarded message ----------
From: Shamun toha md <shamun.toha at gmail.com>
Date: Tue, Dec 14, 2010 at 8:53 AM
Subject: Timer for mod_spidermonkey impossible ?
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>


Hello,

I have a test script like this. But is it possible to make a
setTimeout(function(){ console_log("Hello world") }, 1000); ??  Like a timer
which keep running with interval of 1 second ? Where i can do some Ajax ???

helloworld.js
=========
var argv;

function on_hangup(hup_session, how)
{
   console_log(
           "err", how + " HOOK" + "\n" +
           "hello world" + session.getVariable("test") + "\n"
    );

    exit();
  );

session.setHangupHook(on_hangup);



Thanks & Regards
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From msc at freeswitch.org  Tue Dec 14 11:31:38 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 14 Dec 2010 00:31:38 -0800
Subject: [Freeswitch-users] Incomplete offer/answer debugging
In-Reply-To: <4D0716E4.6040501@j-cope.com>
References: <4D0716E4.6040501@j-cope.com>
Message-ID: <AANLkTikNx8k=Vz=4yhCwHAfrVNT3he_wD2iYdan3-i8K@mail.gmail.com>

Not knowing the phones involved too well I can only guess, however I think
line #531 on the first pastebin is the key:

Reason: SIP;cause=488;text="Incomplete offer/answer"

Have you been messing with the codec settings or the codec prefs? Also, if
you did a make current recently then you might want to use the latest
default configs and re-integrate your custom changes into them. I find that
starting over with clean default configs is usually easier than trying
figure out an existing config. Just backup your existing conf directory and
then do "make samples" from the source directory. Then slowly add in your
customizations, reloadxml, and test. Repeat if necessary.

-MC


On Mon, Dec 13, 2010 at 11:04 PM, Jerre Cope <jerre at j-cope.com> wrote:

> OK, I have a more concise pastebin up (14773) where I gather I must have
> an sdb problem with my loctec phone (line 36).  I'm hoping there is a
> NDLB-something I can use to fix it.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Dec 14 11:32:27 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 14 Dec 2010 00:32:27 -0800
Subject: [Freeswitch-users] send dtmf on one leg bridge
In-Reply-To: <AANLkTikL-ZWCyXhfHmW4daTdT1q-zARmEJZ3s7kFXiv+@mail.gmail.com>
References: <D10B61781EFA4CFB913F2FEAC7579B27@e1705>
	<AANLkTikhpiqsFbQHeWuxNE2EojeuFVgrdseBC5CU=308@mail.gmail.com>
	<7555EA3AD7864DA9B52A6899DD7FF2B3@e1705>
	<AANLkTikFXnH1TYEUe4D0uM6q+WMnvWqCQ+19n0bSZYM_@mail.gmail.com>
	<05F742BEE64A443C93952DD22DC7E84C@e1705>
	<AANLkTikL-ZWCyXhfHmW4daTdT1q-zARmEJZ3s7kFXiv+@mail.gmail.com>
Message-ID: <AANLkTin5ejmu-QAMSa_YafXUfp-NdCo-KfVSRi2gxT5E@mail.gmail.com>

Per my other message you may also just need to use uuid_recv_dtmf. Try it
and see.
-MC

On Mon, Dec 13, 2010 at 11:12 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Perhaps the UUID variable doesn't contain the correct value though...
>
> Try uuid_send_dtmf using the known UUID yourself. That'll show you
> whether uuid_send_dtmf works and the problem is in the expand api, or
> whether it's uuid_send_dtmf that's having a problem.
>
> -Steve
>
>
> On 14 December 2010 01:18, Madovsky <infos at madovsky.org> wrote:
> > because I need to use variables inside the api request
> > is it make a difference ?
> >
> > ----- Original Message -----
> > From: "Steven Ayre" <steveayre at gmail.com>
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Sent: Monday, December 13, 2010 7:47 PM
> > Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
> >
> >
> >> What about  uuid_send_dtmf directly? i.e. not through expand.
> >>
> >> -Steve
> >>
> >>
> >> On 13 December 2010 20:04, Madovsky <infos at madovsky.org> wrote:
> >>> yes,
> >>>
> >>> I tried expand uuid_send_dtmf [uuid] 12345 at 120 once
> >>> answer done and ivr asks pin number but the CLI
> >>> says "ERR- no reply".
> >>>
> >>> ----- Original Message -----
> >>> From: Michael Collins
> >>> To: FreeSWITCH Users Help
> >>> Sent: Monday, December 13, 2010 2:50 PM
> >>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
> >>> Have you tried uuid_send_dtmf?
> >>> -MC
> >>>
> >>> On Sun, Dec 12, 2010 at 12:30 PM, Madovsky <infos at madovsky.org> wrote:
> >>>>
> >>>> Is it possible to send dtfmf via
> >>>> send_dtmf from CLI on a one leg bridge ?
> >>>> it's for enter a pin conference and I can't use
> >>>> any RFC on my sip phone.
> >>>>
> >>>> Thanks
> >>>>
> >>>> Franck
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>> ________________________________
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sfrippiat at dti-be.com  Tue Dec 14 11:33:35 2010
From: sfrippiat at dti-be.com (=?ISO-8859-1?Q?S=E9bastien_Frippiat?=)
Date: Tue, 14 Dec 2010 09:33:35 +0100
Subject: [Freeswitch-users] originate call generates weird noise and
 freezes phone
In-Reply-To: <AANLkTi=2oSETkRnT88S4kVOELfXNKMjMB0-MzBPOkfCf@mail.gmail.com>
References: <4D063A56.9000801@dti-be.com>
	<AANLkTi=2oSETkRnT88S4kVOELfXNKMjMB0-MzBPOkfCf@mail.gmail.com>
Message-ID: <4D072BDF.9090103@dti-be.com>

Setting verbose_sdp to true didn't seem to change anything.

I enabled logs by pressing F8 (console loglevel 7) + F10 (sofia profile 
internal siptrace on). You can find them here: 
http://pastebin.freeswitch.org/14774

Sorry there are several unrelated SIP messages but I prefered not to 
delete anything from the logs.

Thank you for your help,
Sebastien Frippiat

Le 13/12/2010 17:10, Anthony Minessale a ?crit :
> try setting the global variable verbose_sdp=true in vars.xml
> if that doesn't work, post a trace at debug log level + sip trace with
> sofia global siptrace on
>
>
>
> On Mon, Dec 13, 2010 at 9:23 AM, S?bastien Frippiat
> <sfrippiat at dti-be.com>  wrote:
>> Hello.
>>
>> I was using FreeSwitch from 2010/09/09 (don't have the exact git version
>> sorry) but I recently updated it with version from 2010/12/09 (git-e680c82
>> 2010-12-09 08-59-06 -0600). I run an application that handles web requests
>> and convert them to originate calls. Everything was working fine but now,
>> anytime I try an originate call, everything goes wrong.
>>
>> Here is what I do (to originate a call from 26 to 27):
>> - bgapi originate {{parameters }}sofia/internal/26% 27
>>    with the following parameters:
>>      origination_caller_id_name='{0}',
>>      origination_caller_id_number='{1}',
>>      originate_timeout=20,
>>      effective_caller_id_name='{2}',
>>      effective_caller_id_number='{3}',
>>      allow_outside_calls={0},
>>      outside_calls_gateway={1}
>>    (the last two parameters are user parameters I use in my dialplan)
>> - phone 26 starts ringing and I pick it up (silence on the line)
>> - phone 27 starts ringing and I pick it up (lots of deafening noise + phone
>> freezes after a few seconds)
>>
>> As I said, it was working fine with version from early september. I did not
>> find anything useful in the logs or in the config. Has anything changed ?
>> What can I do to solve this problem ?
>>
>> I tested with a Grandstream GXP2000 and a SNOM M3 and only the Grandstream
>> has the problem. I suspect a codec issue but what puzzles me is that it was
>> previously working and that it completely freezes the phone.
>>
>> Thanks,
>> Sebastien
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>



From tayeb.meftah at gmail.com  Tue Dec 14 12:38:41 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 14 Dec 2010 10:38:41 +0100
Subject: [Freeswitch-users] Fwd: Timer for mod_spidermonkey impossible ?
In-Reply-To: <AANLkTi=pJw4d1jcL=m_6d4DV_Jc9wyDenMQhXH8FAKcQ@mail.gmail.com>
References: <AANLkTinVAOi9qrx0C8upJ+f1X-Jfjxs=xoiyw3Lvu5=_@mail.gmail.com>
	<AANLkTi=pJw4d1jcL=m_6d4DV_Jc9wyDenMQhXH8FAKcQ@mail.gmail.com>
Message-ID: <4D073B21.5040601@gmail.com>

DON'T Double Post
Le 14/12/2010 09:30, Shamun toha md a ?crit :
> BUG???? i tried this but i get this.
>
> 2010-12-14 08:27:55.142532 [ERR] helloworld.js:3 ReferenceError: 
> setTimeout is not defined
>
>
> function timers()
> {
>     setTimeout(function() {
>        console_log("err", '------------------ Hello world --------- 
> Time Time Timer...:) ');
>     }, 1000);
> }
> timers();
>
>
> ---------- Forwarded message ----------
> From: *Shamun toha md* <shamun.toha at gmail.com 
> <mailto:shamun.toha at gmail.com>>
> Date: Tue, Dec 14, 2010 at 8:53 AM
> Subject: Timer for mod_spidermonkey impossible ?
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org 
> <mailto:freeswitch-users at lists.freeswitch.org>>
>
>
> Hello,
>
> I have a test script like this. But is it possible to make a 
> setTimeout(function(){ console_log("Hello world") }, 1000); ??  Like a 
> timer which keep running with interval of 1 second ? Where i can do 
> some Ajax ???
>
> helloworld.js
> =========
> var argv;
>
> function on_hangup(hup_session, how)
> {
>    console_log(
>            "err", how + " HOOK" + "\n" +
>            "hello world" + session.getVariable("test") + "\n"
>     );
>
>     exit();
>   );
>
> session.setHangupHook(on_hangup);
>
>
>
> Thanks & Regards
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313

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From neilp at cs.stanford.edu  Tue Dec 14 13:23:55 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Tue, 14 Dec 2010 15:53:55 +0530
Subject: [Freeswitch-users] freetdm and ring_ready
Message-ID: <AANLkTikfyzKUZhb=2KS9=2v_Fmk=mj0=ruz328aTceM3@mail.gmail.com>

I am using ring_ready to let incoming calls to my IVR app ring couple times
before the app engages:

session:execute("ring_ready");
session:sleep(8000);
...
session:answer();

On one of my servers I am using freeTDM to interface with Sangoma hardware,
and another uses OpenZap (this is really the only difference between the two
FS instances). Ringing/early media is playing on the openzap server, but not
with freeTDM. Any idea why not?

Thanks,
Neil
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From riedinger at sns.eu  Tue Dec 14 15:11:21 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Tue, 14 Dec 2010 13:11:21 +0100
Subject: [Freeswitch-users] FS crash / Monitoring Script to Restart FS
Message-ID: <4D075EE9.1030901@sns.eu>

We are running FreeSWITCH Version 1.0.head (git-51cc00a 2010-10-06 
11-07-41 -0500).

On the 7th December FS crashed. The last lines of the log file with full 
debug output are:
22c143da-728d-4477-aa06-c8383fa604f2 2010-12-07 22:52:55.339619 [DEBUG] 
sofia_glue.c:4250 Audio Codec Compare 
[G729:18:8000:20:8000]/[G729:18:8000:20:8000]
22c143da-728d-4477-aa06-c8383fa604f2 2010-12-07 22:52:55.339619 [DEBUG] 
sofia_glue.c:2554 Changing Codec from G729 at 30ms to G729 at 20ms
2010-12-07 22:52:55.339619 [INFO] mod_com_g729.c:76 ENCODER DESTROYX - 
0x2aaacc1530b0 (nil)
2010-12-07 22:52:55.339619 [INFO] mod_com_g729.c:77 DECODER DESTROYX - 
0x2aaacc1530b0 (nil)
2010-12-07 22:52:55.339619 [INFO] mod_com_g729.c:76 ENCODER DESTROYX - 
0x2aaacc153130 (nil)

There were some dozen concurrent calls on the system at this moment, 
thus it is possible that the crash was caused by something else than 
mod_com_g729.c.

Furthermore, a 209 MB core dump file was written. Please let me know, if 
anyone wants to examine the reason for the crash in detail.

Does anyone have written a script, which monitors and restart FS in such 
cases?

Thank you in advance
          Jan



-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973



From vetali100 at gmail.com  Tue Dec 14 15:38:00 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Tue, 14 Dec 2010 14:38:00 +0200
Subject: [Freeswitch-users] FS crash / Monitoring Script to Restart FS
In-Reply-To: <4D075EE9.1030901@sns.eu>
References: <4D075EE9.1030901@sns.eu>
Message-ID: <AANLkTi=O8488u-Y+HOrpL+yJxakf6EaZ4wwT_mRcnVOa@mail.gmail.com>

Hi Jan,
Regarding the monitoring and restarting script - you can use monit:

<http://mmonit.com/monit/>Take one of the pre-compiled binaries from
http://mmonit.com/monit/download/

<http://mmonit.com/monit/download/>Unpack it on your system, create file:
/etc/monitrc

With the following content:

      set daemon 60
      set logfile syslog facility log_daemon

      check process freeswitch with pidfile
/usr/local/freeswitch/run/freeswitch.pid
          start program = "/usr/local/freeswitch/bin/freeswitch -nc"
          stop program = "/usr/local/freeswitch/bin/freeswitch -stop"


Start monit
/your/dir/monit/bin/monit
and enjoy :)

Also you can monitor the status from web, if you will add the following
lines to the monitrc:
set httpd port 1234
         use address your.domain.com
     allow your.ip.add.ress
     allow yourUser:yourPassword

Don't forget to add start monit to the system auto start list.
Refer to your linux doc how to do it.

Regards,
Vitalie





2010/12/14 Jan Riedinger <riedinger at sns.eu>

> We are running FreeSWITCH Version 1.0.head (git-51cc00a 2010-10-06
> 11-07-41 -0500).
>
> On the 7th December FS crashed. The last lines of the log file with full
> debug output are:
> 22c143da-728d-4477-aa06-c8383fa604f2 2010-12-07 22:52:55.339619 [DEBUG]
> sofia_glue.c:4250 Audio Codec Compare
> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
> 22c143da-728d-4477-aa06-c8383fa604f2 2010-12-07 22:52:55.339619 [DEBUG]
> sofia_glue.c:2554 Changing Codec from G729 at 30ms to G729 at 20ms
> 2010-12-07 22:52:55.339619 [INFO] mod_com_g729.c:76 ENCODER DESTROYX -
> 0x2aaacc1530b0 (nil)
> 2010-12-07 22:52:55.339619 [INFO] mod_com_g729.c:77 DECODER DESTROYX -
> 0x2aaacc1530b0 (nil)
> 2010-12-07 22:52:55.339619 [INFO] mod_com_g729.c:76 ENCODER DESTROYX -
> 0x2aaacc153130 (nil)
>
> There were some dozen concurrent calls on the system at this moment,
> thus it is possible that the crash was caused by something else than
> mod_com_g729.c.
>
> Furthermore, a 209 MB core dump file was written. Please let me know, if
> anyone wants to examine the reason for the crash in detail.
>
> Does anyone have written a script, which monitors and restart FS in such
> cases?
>
> Thank you in advance
>          Jan
>
>
>
> --
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gustavo.espeche at upper-soft.com  Tue Dec 14 16:13:55 2010
From: gustavo.espeche at upper-soft.com (Gustavo Espeche)
Date: Tue, 14 Dec 2010 10:13:55 -0300
Subject: [Freeswitch-users] mod_h323 issue
Message-ID: <1292332435.2134.10.camel@gustavo-laptop>

Hello, i'll try to interworker h323 to sip with fs but fs don't send the
call to sip gw, i test with some h323 endpoint with the same result,
follow is the debug log of FS, i'm apreciate a lot if someone can give
me some tips for that it work.
best Regards

Gustavo Espeche
www.easyipcall.com



2010-12-14 12:38:57.395263 [DEBUG] transports.cxx:1756 Started
connection:  host=200.117.192.17:20570, if=72.51.47.100:1720, handle=5
2010-12-14 12:38:57.395263 [DEBUG] transports.cxx:1550 Waiting on socket
accept on ip$72.51.47.100:1720
2010-12-14 12:38:57.395263 [DEBUG] transports.cxx:681 Started incoming
call thread
2010-12-14 12:38:57.395263 [DEBUG] transports.cxx:1334 Awaiting first
PDU
2010-12-14 12:38:57.635261 [DEBUG] h323pdu.cxx:618 Receiving PDU [ip
$72.51.47.100:1720/ip$200.117.192.17:20570] :
  {
    q931pdu = {
      protocolDiscriminator = 8
      callReference = 4293
      from = originator
      messageType = Setup
      IE: Bearer-Capability 
= {
        88 93 a5                                           ...
      }
      IE: Display = {
        67 75 73 74 61 76 6f 00                            gustavo.
      }
      IE: Called-Party-Number = {
        81 32 33 31 35 34 33 35  31 34 32 
38 30 36 33 33   .231543514280633
      }
      IE: User-User = {
        20 b8 06 00 08 91 4a 00  06 01 40 06 00 67 00
75    .....J... at ..g.u
        00 73 00 74 00 61 00 76  00 6f 22 c0 09 00 00
3d   .s.t.a.v.o"....=
        ...
      }
    }
    h
225pdu = {
      h323_uu_pdu = {
        h323_message_body = setup {
          protocolIdentifier = 0.0.8.2250.0.6
          sourceAddress = 1 entries {
            [0]=h323_ID  7 characters {
              0067 0075 0073 0074 0061 0076 006f        
gustavo
            }
          }
          sourceInfo = {
            vendor = {
              vendor = {
                t35CountryCode = 9
                t35Extension = 0
                manufacturerCode = 61
              }
              produc
(OPAL v3.6
                2e 36
 29 00 00                                     .6)..
              }
            }
            terminal = {
            }
            mc = false
            undefinedNode = false
          }
          destinationAddress = 1 entries {
            [0]=
dialedDigits "231543514280633"
          }
          destCallSignalAddress = ipAddress {
            ip =  4 octets {
              48 33 2f 64                                        H3/d
            }
            port = 1720
          }
          a
ctiveMC = false
          conferenceID =  16 octets {
            b6 4e 58 c9 ec 05 e0 11  9e 19 00 1e 64 1f 33
fe   .NX.........d.3.
          }
          conferenceGoal = create <<null>>
          callType = pointToPoint <<null>>
          sourceC
allSignalAddress = ipAddress {
            ip =  4 octets {
              c8 75 c0 11                                        .u..
            }
            port = 20570
          }
          callIdentifier = {
            guid =  16 octets {
       
       a2 4e 58 c9 ec 05 e0 11  9e 19 00 1e 64 1f 33
fe   .NX.........d.3.
            }
          }
          fastStart = 12 entries {
            [0]= 29 octets {
              40 00 00 06 04 01 00 4c  20 13 80 11 1c 00 01 00
@......L .......
  
Log-Func: 
Log-Line: 0
User-Data: 

            c8 75 c0 11 13 c6 00 c8  75 c0 11 13
c7            .u......u....
            }
            [1]= 19 octets {
              00 00 64 0c 20 13 80 0b  0d 00 01 00 c8 75 c0
11   ..d. ........u..
                              ...
            }
            [2]= 32 octets {
              40 00 00 06 04 01 00 4e  0c 03 00 20 00 80 11 1c
@......N... ....
              00 01 00 c8 75 c0 11 13  c6 00 c8 75 c0 11 13
c7   ....u......u....
            }
      
      [3]= 22 octets {
              00 00 65 0e 0c 03 00 20  00 80 0b 0d 00 01 00
c8   ..e.... ........
              75 c0 11 13 c7 00                                  u.....
            }
            [4]= 43 octets {
              40 00 00 06 04 
01 00 4c  10 09 00 00 3d 09 47 2e   @......L....=.G.
              37 32 36 2d 31 36 6b 80  12 1c 40 01 00 c8 75 c0
726-16k... at ...u.
              ...
            }
            [5]= 32 octets {
              00 00 66 0c 10 09 00 00  3d 09 47 2e 37
 32 36 2d   ..f.....=.G.726-
              31 36 6b 80 0b 0d 40 01  00 c8 75 c0 11 13 c7 54
16k... at ...u....T
            }
            [6]= 35 octets {
              40 00 00 06 04 01 00 48  78 00 4a ff 00 80 01 00
@......Hx.J.....
             
 80 11 1c 00 02 00 c8 75  c0 11 13 c8 00 c8 75 c0   .......u......u.
              ...
            }
            [7]= 25 octets {
              00 00 67 08 78 00 4a ff  00 80 01 00 80 0b 0d
00   ..g.x.J.........
              02 00 c8 75 c0 11 13 c9
  00                        ...u.....
            }
            [8]= 34 octets {
              40 00 00 06 04 01 00 48  68 4a ff 00 80 01 00 80
@......HhJ......
              11 1c 00 02 00 c8 75 c0  11 13 c8 00 c8 75 c0
11   ......u......u..
    
          ...
            }
            [9]= 24 octets {
              00 00 68 08 68 4a ff 00  80 01 00 80 0b 0d 00
02   ..h.hJ..........
              00 c8 75 c0 11 13 c9
00                            ..u.....
            }
            [10]= 34 o
ctets {
              40 00 00 06 04 01 00 48  70 4a ff 00 80 01 00 80
@......HpJ......
              11 1c 00 02 00 c8 75 c0  11 13 c8 00 c8 75 c0
11   ......u......u..
              ...
            }
            [11]= 24 octets {
              0
0 00 69 08 70 4a ff 00  80 01 00 80 0b 0d 00 02   ..i.pJ..........
              00 c8 75 c0 11 13 c9
00                            ..u.....
            }
          }
          mediaWaitForConnect = false
          canOverlapSend = false
          m
ultipleCalls = false
          maintainConnection = false
          parallelH245Control = 2 entries {
            [0]= 157 octets {
              02 70 01 06 00 08 81 75  00 0d 80 13 80 01 f4
00   .p.....u........
              01 00 00 01 00 00 01 
00  00 0c c0 01 00 01 80 09   ................
              ...
            }
            [1]= 7 octets {
              01 00 32 80 1e 40 29                               ..2..@)
            }
          }
        }
        h245Tunneling = true
    
Log-Func: 
LogINCOMING DATA [(null)]

RECV EVENT
Event-Name: SOCKET_DATA
-Line: 0
User-Data: _undef_


2010-12-14 12:38:57.635261 [DEBUG] transports.cxx:1344 Incoming call,
first PDU: callReference=4293
2010-12-14 12:38:57.635261 [DEBUG] mod_h323.cpp:608
======>FSH323EndPoint::CreateConnection callReference = 4293  userDate =
(nil) [0x9a76490]
2010-12-14 12:38:57.635261 [DEBUG] mod_h323.cpp:613 ------>
SWITCH_CALL_DIRECTION_INBOUND
2010-12-14 12:38:57.638241 [DEBUG] mod_h323.cpp:621 ------> fsSession =
0x9bfe898
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3598 Added capability:
G.711-ALaw-64k <1>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3598 Added capability:
G.711-uLaw-64k <2>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3598 Added capability:
GSM-06.10{sw} <3>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3598 Added capability:
G.729A/B{sw} <4>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3598 Added capability:
G.729A{sw} <5>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3598 Added capability:
G.729B{sw} <6>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3598 Added capability:
G.729{sw} <7>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3598 Added capability:
T.38-IFP-PRE <8>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/hookflash <9>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/basicString <10>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/dtmf <11>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/RFC2833 <12>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/Navigation <13>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/Softkey <14>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/PointDevice <15>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/Modal <16>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3660 FindCapability: 1
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3664 Found capability:
G.711-ALaw-64k <1>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3660 FindCapability: 2
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3664 Found capability:
G.711-uLaw-64k <2>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3660 FindCapability: 3
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3664 Found capability:
GSM-06.10{sw} <3>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3660 FindCapability: 4
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3664 Found capability:
G.729A/B{sw} <4>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3660 FindCapability: 5
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3664 Found capability:
G.729A{sw} <5>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3660 FindCapability: 6
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3664 Found capability:
G.729B{sw} <6>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3660 FindCapability: 7
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3664 Found capability:
G.729{sw} <7>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3660 FindCapability: 8
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3664 Found capability:
T.38-IFP-PRE <8>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3660 FindCapability: 9
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3664 Found capability:
UserInput/hookflash <9>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3660 FindCapability: 10
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3664 Found capability:
UserInput/basicString <10>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3660 FindCapability: 11
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3664 Found capability:
UserInput/dtmf <11>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3660 FindCapability: 12
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3664 Found capability:
UserInput/RFC2833 <12>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3660 FindCapability: 13
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3664 Found capability:
UserInput/Navigation <13>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3660 FindCapability: 14
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3664 Found capability:
UserInput/Softkey <14>
2010-12-14 12:38:57.638241 [DEBUG] h323caps.cxx:3660 FindCapability: 15
2010-12-14 12:38:57.641242 [DEBUG] h323caps.cxx:3664 Found capability:
UserInput/PointDevice <15>
2010-12-14 12:38:57.641244 [DEBUG] h323caps.cxx:3660 FindCapability: 16
2010-12-14 12:38:57.641244 [DEBUG] h323caps.cxx:3664 Found capability:
UserInput/Modal <16>
2010-12-14 12:38:57.641244 [DEBUG] h4601.cxx:1792 Endpoint Attached
2010-12-14 12:38:57.641244 [DEBUG] rfc2833.cxx:87 Handler created
2010-12-14 12:38:57.641244 [DEBUG] h4601.cxx:1287 Loaded Std 24
2010-12-14 12:38:57.641244 [DEBUG] h4601.cxx:1260 Loaded Feature Std24
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:707
======>FSH323Connection::FSH323Connection [0x9a7d140]
2010-12-14 12:38:57.641244 [DEBUG] h323ep.cxx:3160 Created new
connection: ip$200.117.192.17:20570/4293
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:764 ---------->token =
ip$200.117.192.17:20570/4293 [0x9a7d140]
2010-12-14 12:38:57.641244 [DEBUG] h323.cxx:1426 Handling PDU: Setup
callRef=4293
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:1013
======>FSH323Connection::OnReceivedSignalSetup [0x9a7d140]
2010-12-14 12:38:57.641244 [DEBUG] h323.cxx:1707 Set protocol version to
6 and implying H.245 version 13
2010-12-14 12:38:57.641244 [DEBUG] h323.cxx:1993 Set remote application
name: " 3.2.6 (OPAL v3.6.6) 9/61"
2010-12-14 12:38:57.641244 [DEBUG] h4601.cxx:1775 Removing all Features
are remote/Gk does not appear to support H.460
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:771
======>FSH323Connection::OnSetLocalCapabilities() [0x9a7d140]
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:778
======>FSH323Connection::SetLocalCapabilities() Size local capability =
16 [0x9a7d140]
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:819
======>FSH323Connection::decodeCapability [0x9a7d140]
2010-12-14 12:38:57.641244 [NOTICE] mod_h323.cpp:841 capability
'G.711-ALaw-64k' format 'PCMA' 8
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:819
======>FSH323Connection::decodeCapability [0x9a7d140]
2010-12-14 12:38:57.641244 [NOTICE] mod_h323.cpp:841 capability
'G.711-uLaw-64k' format 'PCMU' 0
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:819
======>FSH323Connection::decodeCapability [0x9a7d140]
2010-12-14 12:38:57.641244 [NOTICE] mod_h323.cpp:841 capability
'GSM-06.10' format 'GSM' 3
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:819
======>FSH323Connection::decodeCapability [0x9a7d140]
2010-12-14 12:38:57.641244 [NOTICE] mod_h323.cpp:841 capability
'G.729A/B' format 'G729' 18
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:819
======>FSH323Connection::decodeCapability [0x9a7d140]
2010-12-14 12:38:57.641244 [NOTICE] mod_h323.cpp:841 capability 'G.729A'
format 'G729' 18
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:819
======>FSH323Connection::decodeCapability [0x9a7d140]
2010-12-14 12:38:57.641244 [NOTICE] mod_h323.cpp:841 capability 'G.729B'
format 'G729b' 18
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:819
======>FSH323Connection::decodeCapability [0x9a7d140]
2010-12-14 12:38:57.641244 [NOTICE] mod_h323.cpp:841 capability 'G.729'
format 'G729' 18
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:819
======>FSH323Connection::decodeCapability [0x9a7d140]
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:819
======>FSH323Connection::decodeCapability [0x9a7d140]
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:819
======>FSH323Connection::decodeCapability [0x9a7d140]
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:819
======>FSH323Connection::decodeCapability [0x9a7d140]
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:819
======>FSH323Connection::decodeCapability [0x9a7d140]
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:819
======>FSH323Connection::decodeCapability [0x9a7d140]
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:819
======>FSH323Connection::decodeCapability [0x9a7d140]
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:819
======>FSH323Connection::decodeCapability [0x9a7d140]
2010-12-14 12:38:57.641244 [DEBUG] mod_h323.cpp:819
======>FSH323Connection::decodeCapability [0x9a7d140]
2010-12-14 12:38:57.644242 [DEBUG] h323.cxx:1830 Sending call proceeding
PDU
2010-12-14 12:38:57.644242 [DEBUG] mod_h323.cpp:634
======>FSH323EndPoint::OnSetGatewayPrefixes [0x9a76490]
2010-12-14 12:38:57.644242 [DEBUG] mod_h323.cpp:1077
======>FSH323Connection::OnSendCallProceeding fastStartState =
FastStartInitiate [0x9a7d140]
2010-12-14 12:38:57.644242 [DEBUG] mod_h323.cpp:634
======>FSH323EndPoint::OnSetGatewayPrefixes [0x9a76490]
2010-12-14 12:38:57.644242 [DEBUG] h323.cxx:1873 Incoming call accepted
2010-12-14 12:38:57.644242 [DEBUG] h323.cxx:1715 Fast start detected
2010-12-14 12:38:57.644242 [DEBUG] h323.cxx:1722 Fast start open:
  {
    forwardLogicalChannelNumber = 1
    forwardLogicalChannelParameters = {
      dataType = nullData <<null>>
      multiplexParameters = none <<null>>
    }
    reverseLogicalChannelParameters = {

      dataType = audioData g711Alaw64k 20
      multiplexParameters = h2250LogicalChannelParameters {
        sessionID = 1
        mediaChannel = unicastAddress iPAddress {
          network =  4 octets {
            c8 75 c0 11                   
                     .u..
          }
          tsapIdentifier = 5062
        }
        mediaGuaranteedDelivery = false
        mediaControlChannel = unicastAddress iPAddress {
          network =  4 octets {
            c8 75 c0 11                 
Log-Func: 
Log-Line: 0
User-Data: 

                       .u..
          }
          tsapIdentifier2010-12-14 12:38:57.644242 [DEBUG] h323.cxx:4747
CreateLogicalChannel - reverse channel
2010-12-14 12:38:57.644242 [DEBUG] h323caps.cxx:3792 FindCapability:
audioData
2010-12-14 12:38:57.644242 [DEBUG] h323caps.cxx:3833 Found capability:
G.711-ALaw-64k <1>
2010-12-14 12:38:57.644242 [DEBUG] h323caps.cxx:3712 FindCapability:
G.711-ALaw-64k <1>
2010-12-14 12:38:57.644242 [DEBUG] h323caps.cxx:3598 Added capability:
G.711-ALaw-64k <1>
2010-12-14 12:38:57.644242 [DEBUG] mod_h323.cpp:978
======>FSH323Connection::OnCreateLogicalChannel
('G.711-ALaw-64k',IsTransmitter) [0x9a7d140]
2010-12-14 12:38:57.644242 [DEBUG] mod_h323.cpp:961
======>FSH323Connection::CreateRealTimeLogicalChannel [0x9a7d140]
2010-12-14 12:38:57.644242 [DEBUG] mod_h323.cpp:1880
======>FSH323_ExternalRTPChannel::FSH323_ExternalRTPChannel sessionID =
1 :IsTransmitter addr = 72.51.47.100:56790 [0xb773dc88]
2010-12-14 12:38:57.644242 [DEBUG] mod_h323.cpp:1884
------->capability.GetPayloadType() return = [pt=128]
2010-12-14 12:38:57.644242 [DEBUG] mod_h323.cpp:1885
------->capability.GetFormatName() return = G.711-ALaw-64k
2010-12-14 12:38:57.644242 [DEBUG] mod_h323.cpp:1897 ------->payloadCode
= 8
2010-12-14 12:38:57.644242 [DEBUG] codecs.cxx:1534 G711 ALaw encoder
created for at 64k, 160 samples
2010-12-14 12:38:57.644242 [DEBUG] channels.cxx:866 Bandwidth
requested/used = 64.0/0.0 kb/s
2010-12-14 12:38:57.644242 [DEBUG] h323.cxx:5105 Bandwidth request:
-0.0kb/s, available: 10000.0kb/s
2010-12-14 12:38:57.644242 [DEBUG] h323.cxx:5105 Bandwidth request:
+64.0kb/s, available: 10000.0kb/s
2010-12-14 12:38:57.644242 [DEBUG] channels.cxx:1056 OnReceivedPDU for
channel: T-0
2010-12-14 12:38:57.644242 [DEBUG] mod_h323.cpp:2223
======>FSH323_ExternalRTPChannel::OnReceivedPDU [0xb773dc88]
2010-12-14 12:38:57.644242 [DEBUG] mod_h323.cpp:2230 Remote RTP address
200.117.192.17:5062
2010-12-14 12:38:57.644242 [DEBUG] mod_h323.cpp:1357
======>PFSH323Connection::setRemoteAddress remoteIP = 200.117.192.17 ,
remotePort = 5062 [0x9a7d140]
2010-12-14 12:38:57.644242 [DEBUG] mod_h323.cpp:1360 Got remote RTP
address 200.117.192.17:5062 [0x9a7d140]
2010-12-14 12:38:57.647243 [DEBUG] h323.cxx:1722 Fast start open:
  {
    forwardLogicalChannelNumber = 101
    forwardLogicalChannelParameters = {
      dataType = audioData g711Alaw64k 20
      multiplexParameters = h2250LogicalChannelParameters {
        sessionID 
= 1
        mediaGuaranteedDelivery = false
        mediaControlChannel = unicastAddress iPAddress {
          network =  4 octets {
            c8 75 c0 11                                        .u..
          }
          tsapIdentifier = 5063
    
Log-Func: 
Log-Line: 0
User-Data: 

    }
        silenceSu2010-12-14 12:38:57.647243 [DEBUG] h323.cxx:4762
CreateLogicalChannel - forward channel
2010-12-14 12:38:57.647243 [DEBUG] h323caps.cxx:3792 FindCapability:
audioData
2010-12-14 12:38:57.647243 [DEBUG] h323caps.cxx:3833 Found capability:
G.711-ALaw-64k <1>
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:978
======>FSH323Connection::OnCreateLogicalChannel
('G.711-ALaw-64k',IsReceiver) [0x9a7d140]
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:961
======>FSH323Connection::CreateRealTimeLogicalChannel [0x9a7d140]
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:1880
======>FSH323_ExternalRTPChannel::FSH323_ExternalRTPChannel sessionID =
1 :IsReceiver addr = 72.51.47.100:56790 [0xb773b9a0]
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:1884
------->capability.GetPayloadType() return = [pt=128]
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:1885
------->capability.GetFormatName() return = G.711-ALaw-64k
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:1897 ------->payloadCode
= 8
2010-12-14 12:38:57.647243 [DEBUG] codecs.cxx:1534 G711 ALaw decoder
created for at 64k, 160 samples
2010-12-14 12:38:57.647243 [DEBUG] channels.cxx:866 Bandwidth
requested/used = 64.0/0.0 kb/s
2010-12-14 12:38:57.647243 [DEBUG] h323.cxx:5105 Bandwidth request:
-0.0kb/s, available: 9936.0kb/s
2010-12-14 12:38:57.647243 [DEBUG] h323.cxx:5105 Bandwidth request:
+64.0kb/s, available: 9936.0kb/s
2010-12-14 12:38:57.647243 [DEBUG] channels.cxx:1056 OnReceivedPDU for
channel: R-101
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:2223
======>FSH323_ExternalRTPChannel::OnReceivedPDU [0xb773b9a0]
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:2230 Remote RTP address
200.117.192.17:5062
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:1357
======>PFSH323Connection::setRemoteAddress remoteIP = 200.117.192.17 ,
remotePort = 5062 [0x9a7d140]
2010-12-14 12:38:57.647243 [DEBUG] h323.cxx:1722 Fast start open:
  {
    forwardLogicalChannelNumber = 1
    forwardLogicalChannelParameters = {
      dataType = nullData <<null>>
      multiplexParameters = none <<null>>
    }
    reverseLogicalChannelParameters = {

      dataType = audioData gsmFullRate {
        audioUnitSize = 33
        comfortNoise = false
        scrambled = false
      }
      multiplexParameters = h2250LogicalChannelParameters {
        sessionID = 1
        mediaChannel = unicastAddre
ss iPAddress {
          network =  4 octets {
            c8 75 c0 11                                        .u..
          }
          tsapIdentifier = 5062
        }
        mediaGuaranteedDelivery = false
        mediaControlChannel = unicastAdd
ress iPAddress {
          network =  4 octets {
            c8 75 c0 11                                        .u..
          }
          tsapIdentifier = 5063
        }
      }
    }
  }
2010-12-14 12:38:57.647243 [DEBUG] h323.cxx:4747 CreateLogicalChannel -
reverse channel
2010-12-14 12:38:57.647243 [DEBUG] h323caps.cxx:3792 FindCapability:
audioData
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.h:529
==============>BaseGSM0610Cap::OnReceivedPDU [0xb790e360]
2010-12-14 12:38:57.647243 [DEBUG] h323caps.cxx:1543 Capability tx
frames reduced from 2 to 1
2010-12-14 12:38:57.647243 [DEBUG] h323caps.cxx:3833 Found capability:
GSM-06.10{sw} <3>
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.h:529
==============>BaseGSM0610Cap::OnReceivedPDU [0x9b1c240]
2010-12-14 12:38:57.647243 [DEBUG] h323caps.cxx:1543 Capability tx
frames reduced from 2 to 1
2010-12-14 12:38:57.647243 [DEBUG] h323caps.cxx:3712 FindCapability:
GSM-06.10{sw} <3>
2010-12-14 12:38:57.647243 [DEBUG] h323caps.cxx:3598 Added capability:
GSM-06.10{sw} <3>
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:978
======>FSH323Connection::OnCreateLogicalChannel
('GSM-06.10{sw}',IsTransmitter) [0x9a7d140]
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:961
======>FSH323Connection::CreateRealTimeLogicalChannel [0x9a7d140]
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:1880
======>FSH323_ExternalRTPChannel::FSH323_ExternalRTPChannel sessionID =
1 :IsTransmitter addr = 72.51.47.100:56790 [0xb7922880]
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:1884
------->capability.GetPayloadType() return = [pt=128]
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:1885
------->capability.GetFormatName() return = GSM-06.10{sw}
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:1897 ------->payloadCode
= 3
2010-12-14 12:38:57.647243 [DEBUG] channels.cxx:1056 OnReceivedPDU for
channel: T-0
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.h:529
==============>BaseGSM0610Cap::OnReceivedPDU [0xb79211f0]
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:2223
======>FSH323_ExternalRTPChannel::OnReceivedPDU [0xb7922880]
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:2230 Remote RTP address
200.117.192.17:5062
2010-12-14 12:38:57.647243 [DEBUG] mod_h323.cpp:1357
======>PFSH323Connection::setRemoteAddress remoteIP = 200.117.192.17 ,
remotePort = 5062 [0x9a7d140]
2010-12-14 12:38:57.650240 [DEBUG] h323.cxx:1722 Fast start open:
  {
    forwardLogicalChannelNumber = 102
    forwardLogicalChannelParameters = {
      dataType = audioData gsmFullRate {
        audioUnitSize = 33
        comfortNoise = false
        scrambled = fal
se
      }
      multiplexParameters = h2250LogicalChannelParameters {
        sessionID = 1
        mediaGuaranteedDelivery = false
        mediaControlChannel = unicastAddress iPAddress {
          network =  4 octets {
            c8 75 c0 11    
                                    .u..
          }
          tsapIdentifier = 5063
        }
        silenceSuppression = false
      }
    }
  }
2010-12-14 12:38:57.650240 [DEBUG] h323.cxx:4762 CreateLogicalChannel -
forward channel
2010-12-14 12:38:57.650240 [DEBUG] h323caps.cxx:3792 FindCapability:
audioData
2010-12-14 12:38:57.650240 [DEBUG] mod_h323.h:529
==============>BaseGSM0610Cap::OnReceivedPDU [0xb79769f8]
2010-12-14 12:38:57.650240 [DEBUG] h323caps.cxx:3833 Found capability:
GSM-06.10{sw} <3>
2010-12-14 12:38:57.650240 [DEBUG] mod_h323.h:529
==============>BaseGSM0610Cap::OnReceivedPDU [0x9b1c240]
2010-12-14 12:38:57.650240 [DEBUG] h323caps.cxx:1543 Capability rx
frames reduced from 24 to 1
2010-12-14 12:38:57.650240 [DEBUG] mod_h323.cpp:978
======>FSH323Connection::OnCreateLogicalChannel
('GSM-06.10{sw}',IsReceiver) [0x9a7d140]
2010-12-14 12:38:57.650240 [DEBUG] mod_h323.cpp:961
======>FSH323Connection::CreateRealTimeLogicalChannel [0x9a7d140]
2010-12-14 12:38:57.650240 [DEBUG] mod_h323.cpp:1880
======>FSH323_ExternalRTPChannel::FSH323_ExternalRTPChannel sessionID =
1 :IsReceiver addr = 72.51.47.100:56790 [0xb791e1d8]
2010-12-14 12:38:57.650240 [DEBUG] mod_h323.cpp:1884
------->capability.GetPayloadType() return = [pt=128]
2010-12-14 12:38:57.650240 [DEBUG] mod_h323.cpp:1885
------->capability.GetFormatName() return = GSM-06.10{sw}
2010-12-14 12:38:57.650240 [DEBUG] mod_h323.cpp:1897 ------->payloadCode
= 3
2010-12-14 12:38:57.650240 [DEBUG] channels.cxx:1056 OnReceivedPDU for
channel: R-102
2010-12-14 12:38:57.650240 [DEBUG] mod_h323.h:529
==============>BaseGSM0610Cap::OnReceivedPDU [0xb79769f8]
2010-12-14 12:38:57.650240 [DEBUG] mod_h323.cpp:2223
======>FSH323_ExternalRTPChannel::OnReceivedPDU [0xb791e1d8]
2010-12-14 12:38:57.650240 [DEBUG] mod_h323.cpp:2230 Remote RTP address
200.117.192.17:5062
2010-12-14 12:38:57.650240 [DEBUG] mod_h323.cpp:1357
======>PFSH323Connection::setRemoteAddress remoteIP = 200.117.192.17 ,
remotePort = 5062 [0x9a7d140]
2010-12-14 12:38:57.650240 [DEBUG] h323.cxx:1722 Fast start open:
  {
    forwardLogicalChannelNumber = 1
    forwardLogicalChannelParameters = {
      dataType = nullData <<null>>
      multiplexParameters = none <<null>>
    }
    reverseLogicalChannelParameters = {

      dataType = audioData nonStandard [Equivalence G.726-16k] {
        nonStandardIdentifier = h221NonStandard {
          t35CountryCode = 9
          t35Extension = 0
          manufacturerCode = 61
        }
        data =  9 octets {
        
  47 2e 37 32 36 2d 31 36  6b                        G.726-16k
        }
      }
      multiplexParameters = h2250LogicalChannelParameters {
        sessionID = 1
        mediaChannel = unicastAddress iPAddress {
          network =  4 octets {
    
        c8 75 c0 11                                        .u..
          }
          tsapIdentifier = 5062
        }
        mediaGuaranteedDelivery = false
        mediaControlChannel = unicastAddress iPAddress {
          network =  4 octets {
  
          c8 75 c0 11                                        .u..
          }
          tsapIdentifier = 5063
        }
        dynamicRTPPayloadType = 117
      }
    }
  }
2010-12-14 12:38:57.650240 [DEBUG] h323.cxx:4747 CreateLogicalChannel -
reverse channel
2010-12-14 12:38:57.650240 [DEBUG] h323caps.cxx:3792 FindCapability:
audioData
2010-12-14 12:38:57.650240 [DEBUG] h323.cxx:4806 CreateLogicalChannel -
unknown data type
2010-12-14 12:38:57.650240 [DEBUG] h323.cxx:1722 Fast start open:
  {
    forwardLogicalChannelNumber = 103
    forwardLogicalChannelParameters = {
      dataType = audioData nonStandard [Equivalence G.726-16k] {
        nonStandardIdentifier = h221NonStandard {
     
     t35CountryCode = 9
          t35Extension = 0
          manufacturerCode = 61
        }
        data =  9 octets {
          47 2e 37 32 36 2d 31 36  6b                        G.726-16k
        }
      }
      multiplexParameters = h2250Logical
ChannelParameters {
        sessionID = 1
        mediaGuaranteedDelivery = false
        mediaControlChannel = unicastAddress iPAddress {
          network =  4 octets {
            c8 75 c0 11                                        .u..
          
Log-Func: 
Log-Line: 0
User-Data: 

}
          tsapIdentifier = 5063
        }
        silenceSuppression = false
        dynamicRTP2010-12-14 12:38:57.650240 [DEBUG] h323.cxx:4762
CreateLogicalChannel - forward channel
2010-12-14 12:38:57.650240 [DEBUG] h323caps.cxx:3792 FindCapability:
audioData
2010-12-14 12:38:57.650240 [DEBUG] h323.cxx:4806 CreateLogicalChannel -
unknown data type
2010-12-14 12:38:57.653258 [DEBUG] h323.cxx:1722 Fast start open:
  {
    forwardLogicalChannelNumber = 1
    forwardLogicalChannelParameters = {
      dataType = nullData <<null>>
      multiplexParameters = none <<null>>
    }
    reverseLogicalChannelParameters = {

      dataType = videoData h261VideoCapability {
        qcifMPI = 1
        cifMPI = 1
        temporalSpatialTradeOffCapability = false
        maxBitRate = 19200
        stillImageTransmission = false
        videoBadMBsCap = false
      }
     
 multiplexParameters = h2250LogicalChannelParameters {
        sessionID = 2
        mediaChannel = unicastAddress iPAddress {
          network =  4 octets {
            c8 75 c0 11                                        .u..
          }
          
tsapIdentifier = 5064
        }
        mediaGuaranteedDelivery = false
        mediaControlChannel = unicastAddress iPAddress {
          network =  4 octets {
            c8 75 c0 11                                        .u..
          }
        
Log-Func: 
Log-Line: 0
User-Data: 

  tsapIdentifier2010-12-14 12:38:57.653258 [DEBUG] h323.cxx:4747
CreateLogicalChannel - reverse channel
2010-12-14 12:38:57.653258 [DEBUG] h323caps.cxx:3792 FindCapability:
videoData
2010-12-14 12:38:57.653258 [DEBUG] h323.cxx:4806 CreateLogicalChannel -
unknown data type
2010-12-14 12:38:57.653258 [DEBUG] h323.cxx:1722 Fast start open:
  {
    forwardLogicalChannelNumber = 104
    forwardLogicalChannelParameters = {
      dataType = videoData h261VideoCapability {
        qcifMPI = 1
        cifMPI = 1
        temporalSpatialTradeOffC
apability = false
        maxBitRate = 19200
        stillImageTransmission = false
        videoBadMBsCap = false
      }
      multiplexParameters = h2250LogicalChannelParameters {
        sessionID = 2
        mediaGuaranteedDelivery = false
    
    mediaControlChannel = unicastAddress iPAddress {
          network =  4 octets {
            c8 75 c0 11                                        .u..
          }
          tsapIdentifier = 5065
        }
        silenceSuppression = false
      }
Log-Func: 
INCOMING DATA [(null)]

RECV EVENT
Event-Name: SOCKET_DATA
Log-Line: 0
User-Data: _undef_


2010-12-14 12:38:57.653258 [DEBUG] h323.cxx:4762 CreateLogicalChannel -
forward channel
2010-12-14 12:38:57.653258 [DEBUG] h323caps.cxx:3792 FindCapability:
videoData
2010-12-14 12:38:57.653258 [DEBUG] h323.cxx:4806 CreateLogicalChannel -
unknown data type
2010-12-14 12:38:57.653258 [DEBUG] h323.cxx:1722 Fast start open:
  {
    forwardLogicalChannelNumber = 1
    forwardLogicalChannelParameters = {
      dataType = nullData <<null>>
      multiplexParameters = none <<null>>
    }
    reverseLogicalChannelParameters = {

      dataType = videoData h261VideoCapability {
        cifMPI = 1
        temporalSpatialTradeOffCapability = false
        maxBitRate = 19200
        stillImageTransmission = false
        videoBadMBsCap = false
      }
      multiplexParameters
 = h2250LogicalChannelParameters {
        sessionID = 2
        mediaChannel = unicastAddress iPAddress {
          network =  4 octets {
            c8 75 c0 11                                        .u..
          }
          tsapIdentifier = 506
4
        }
        mediaGuaranteedDelivery = false
        mediaControlChannel = unicastAddress iPAddress {
          network =  4 octets {
            c8 75 c0 11                                        .u..
          }
          tsapIdentifier = 5
Log-Func: 
Log-Line: 0
User-DataINCOMING DATA [(null)]

RECV EVENT
Event-Name: SOCKET_DATA
: _undef_


2010-12-14 12:38:57.653258 [DEBUG] h323.cxx:4747 CreateLogicalChannel -
reverse channel
2010-12-14 12:38:57.653258 [DEBUG] h323caps.cxx:3792 FindCapability:
videoData
2010-12-14 12:38:57.653258 [DEBUG] h323.cxx:4806 CreateLogicalChannel -
unknown data type
2010-12-14 12:38:57.653258 [DEBUG] h323.cxx:1722 Fast start open:
  {
    forwardLogicalChannelNumber = 105
    forwardLogicalChannelParameters = {
      dataType = videoData h261VideoCapability {
        cifMPI = 1
        temporalSpatialTradeOffCapability = false
  
      maxBitRate = 19200
        stillImageTransmission = false
        videoBadMBsCap = false
      }
      multiplexParameters = h2250LogicalChannelParameters {
        sessionID = 2
        mediaGuaranteedDelivery = false
        mediaControlChan
nel = unicastAddress iPAddress {
          network =  4 octets {
            c8 75 c0 11                                        .u..
          }
          tsapIdentifier = 5065
        }
        silenceSuppression = false
      }
    }
  }
2010-12-14 12:38:57.653258 [DEBUG] h323.cxx:4762 CreateLogicalChannel -
forward channel
2010-12-14 12:38:57.653258 [DEBUG] h323caps.cxx:3792 FindCapability:
videoData
2010-12-14 12:38:57.653258 [DEBUG] h323.cxx:4806 CreateLogicalChannel -
unknown data type
2010-12-14 12:38:57.653258 [DEBUG] h323.cxx:1722 Fast start open:
  {
    forwardLogicalChannelNumber = 1
    forwardLogicalChannelParameters = {
      dataType = nullData <<null>>
      multiplexParameters = none <<null>>
    }
    reverseLogicalChannelParameters = {

      dataType = videoData h261VideoCapability {
        qcifMPI = 1
        temporalSpatialTradeOffCapability = false
        maxBitRate = 19200
        stillImageTransmission = false
        videoBadMBsCap = false
      }
      multiplexParameter
s = h2250LogicalChannelParameters {
        sessionID = 2
        mediaChannel = unicastAddress iPAddress {
          network =  4 octets {
            c8 75 c0 11                                        .u..
          }
          tsapIdentifier = 50
64
        }
        mediaGuaranteedDelivery = false
        mediaControlChannel = unicastAddress iPAddress {
          network =  4 octets {
            c8 75 c0 11                                        .u..
          }
          tsapIdentifier = 
Log-Func: 
Log-Line: 0
User-Data:INCOMING DATA [(null)]

RECV EVENT
Event-Name: SOCKET_DATA


2010-12-14 12:38:57.653258 [DEBUG] h323.cxx:4747 CreateLogicalChannel -
reverse channel
2010-12-14 12:38:57.653258 [DEBUG] h323caps.cxx:3792 FindCapability:
videoData
2010-12-14 12:38:57.653258 [DEBUG] h323.cxx:4806 CreateLogicalChannel -
unknown data type
2010-12-14 12:38:57.656259 [DEBUG] h323.cxx:1722 Fast start open:
  {
    forwardLogicalChannelNumber = 106
    forwardLogicalChannelParameters = {
      dataType = videoData h261VideoCapability {
        qcifMPI = 1
        temporalSpatialTradeOffCapability = false
 
       maxBitRate = 19200
        stillImageTransmission = false
        videoBadMBsCap = false
      }
      multiplexParameters = h2250LogicalChannelParameters {
        sessionID = 2
        mediaGuaranteedDelivery = false
        mediaControlCha
nnel = unicastAddress iPAddress {
          network =  4 octets {
            c8 75 c0 11                                        .u..
          }
          tsapIdentifier = 5065
        }
        silenceSuppression = false
      }
    }
  }
2010-12-14 12:38:57.656259 [DEBUG] h323.cxx:4762 CreateLogicalChannel -
forward channel
2010-12-14 12:38:57.656259 [DEBUG] h323caps.cxx:3792 FindCapability:
videoData
2010-12-14 12:38:57.656259 [DEBUG] h323.cxx:4806 CreateLogicalChannel -
unknown data type
2010-12-14 12:38:57.656259 [DEBUG] h323.cxx:1736 Opened 4 fast start
channels
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:634
======>FSH323EndPoint::OnSetGatewayPrefixes [0x9a76490]
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:857
======>FSH323Connection::OnAnswerCall caller = gustavo [200.117.192.17]
[0x9a7d140]
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:877 address index = 0
value = 2315435142806332010-12-14 12:38:57.656259 [NOTICE]
mod_h323.cpp:884 Called number or alias = 231543514280633
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:925 Created switch
caller profile:
  username                = 
  dialplan           = XML
  caller_id_name     = gustavo [200.117.192.17]
  caller_id_number   = 0000000000
  network_addr       = 200.117.192.17
  source             = h323
  context            = default
  destination_number = 231543514280633
2010-12-14 12:38:57.656259 [NOTICE] switch_channel.c:784 New Channel
h323/231543514280633 [c79e219b-ecd5-4481-b08d-7de04a99be6c]
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:949
(h323/231543514280633) State Change CS_NEW -> CS_INIT
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:1584
======>FSH323Connection::state_change [0x9a7d140]
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:1585 State changed on
connection [0x9a7d140]
2010-12-14 12:38:57.656259 [DEBUG] switch_core_session.c:1083 Send
signal h323/231543514280633 [BREAK]
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:1382
======>FSH323Connection::kill_channel sig = 3 [0x9a7d140]
2010-12-14 12:38:57.656259 [DEBUG] h323.cxx:2505 Answering call:
AnswerCallDeferred
2010-12-14 12:38:57.656259 [DEBUG] h323.cxx:4369
InternalEstablishedConnectionCheck: connectionState=AwaitingLocalAnswer
fastStartState=FastStartResponse
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:1018 ---------> after
FSH323Connection::OnReceivedSignalSetup connectionState =
AwaitingLocalAnswer [0x9a7d140]
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:634
======>FSH323EndPoint::OnSetGatewayPrefixes [0x9a76490]
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:1022 ---------> after
callProceedingPDU.BuildCallProceeding connectionState =
AwaitingLocalAnswer [0x9a7d140]
2010-12-14 12:38:57.656259 [DEBUG] h323.cxx:4451 Default
OnSelectLogicalChannels, FastStartResponse
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:1915
------------->h323_mutex_lock
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:1917
======>FSH323_ExternalRTPChannel::Start() [0xb773dc88]
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:1924
------------->m_sessionID = 1  m_active_sessionID = 0
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:971
======>FSH323Connection::OnStartLogicalChannel chennel = 0xb773dc88
[0x9a7d140]
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:972
======>FSH323Connection::OnStartLogicalChannel connectionState =
AwaitingLocalAnswer [0x9a7d140]
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:1973
------------------------->H323Capability::e_Audio
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:1992
------------------->GetFrameSize() return = 634
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:1993
------------------->GetFrameTime() return = 3077839696
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:1994
------------------->payloadCode = 8
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:1995
------------------->m_codec_ms return =  20
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:1996
------------------->m_capability->GetFormatName() return =
G.711-ALaw-64k
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:1997
------------------->GetH245CodecName() return =  PCMA
2010-12-14 12:38:57.656259 [DEBUG] mod_h323.cpp:2036
h323/231543514280633 initialise  write codec Audio for connection
[0x9a9d4a0]
2010-12-14 12:38:57.659246 [DEBUG] mod_h323.cpp:2091 Set  write Audio
codec to G.711-ALaw-64k  for connection [0xb773dc88]
2010-12-14 12:38:57.659246 [DEBUG] mod_h323.cpp:2097
------------------->tech_pvt->rtp_session = [(nil)]
2010-12-14 12:38:57.659246 [DEBUG] mod_h323.cpp:2098
------------------->samples_per_packet = 160
2010-12-14 12:38:57.659246 [DEBUG] mod_h323.cpp:2099
------------------->actual_samples_per_second = 8000
2010-12-14 12:38:57.659246 [DEBUG] mod_h323.cpp:2134
------------------->timer_name = soft
2010-12-14 12:38:57.659246 [DEBUG] switch_rtp.c:1423 Starting timer
[soft] 160 bytes per 20ms
2010-12-14 12:38:57.659246 [DEBUG] switch_core_state_machine.c:320
(h323/231543514280633) Running State Change CS_INIT
2010-12-14 12:38:57.659246 [DEBUG] switch_core_state_machine.c:356
(h323/231543514280633) State INIT
2010-12-14 12:38:57.659246 [DEBUG] mod_h323.cpp:1592
======>FSH323Connection::on_init [0x9a7d140]
2010-12-14 12:38:57.659246 [DEBUG] mod_h323.cpp:1599 Started routing for
connection [0x9a7d140]
2010-12-14 12:38:57.659246 [DEBUG] mod_h323.cpp:1600
(h323/231543514280633) State Change CS_INIT -> CS_ROUTING
2010-12-14 12:38:57.659246 [DEBUG] mod_h323.cpp:1584
======>FSH323Connection::state_change [0x9a7d140]
2010-12-14 12:38:57.659246 [DEBUG] mod_h323.cpp:1585 State changed on
connection [0x9a7d140]
2010-12-14 12:38:57.659246 [DEBUG] switch_core_session.c:1083 Send
signal h323/231543514280633 [BREAK]
2010-12-14 12:38:57.659246 [DEBUG] mod_h323.cpp:1382
======>FSH323Connection::kill_channel sig = 3 [0x9a7d140]
2010-12-14 12:38:57.659246 [DEBUG] switch_core_state_machine.c:356
(h323/231543514280633) State INIT going to sleep
2010-12-14 12:38:57.659246 [DEBUG] switch_core_state_machine.c:320
(h323/231543514280633) Running State Change CS_ROUTING
2010-12-14 12:38:57.659246 [DEBUG] switch_channel.c:1615
(h323/231543514280633) Callstate Change DOWN -> RINGING
2010-12-14 12:38:57.659246 [DEBUG] switch_core_state_machine.c:359
(h323/231543514280633) State ROUTING
2010-12-14 12:38:57.659246 [DEBUG] mod_h323.cpp:1375
======>FSH323Connection::on_routing [0x9a7d140]
2010-12-14 12:38:57.659246 [DEBUG] switch_core_state_machine.c:77
h323/231543514280633 Standard ROUTING
2010-12-14 12:38:57.659246 [INFO] mod_dialplan_xml.c:331 Processing
gustavo [200.117.192.17] <0000000000>->231543514280633 in context
default
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:2146
------------------------->tech_pvt->rtp_session = 0xb794bf18
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:2169
------------->External RTP address 200.117.192.17:5062
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:2215
------------->h323_mutex_unlock
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:1915
------------->h323_mutex_lock
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:1917
======>FSH323_ExternalRTPChannel::Start() [0xb773b9a0]
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:1924
------------->m_sessionID = 1  m_active_sessionID = 1
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:971
======>FSH323Connection::OnStartLogicalChannel chennel = 0xb773b9a0
[0x9a7d140]
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:972
======>FSH323Connection::OnStartLogicalChannel connectionState =
AwaitingLocalAnswer [0x9a7d140]
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:1973
------------------------->H323Capability::e_Audio
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:1992
------------------->GetFrameSize() return = 634
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:1993
------------------->GetFrameTime() return = 3077839696
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:1994
------------------->payloadCode = 8
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:1995
------------------->m_codec_ms return =  20
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:1996
------------------->m_capability->GetFormatName() return =
G.711-ALaw-64k
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:1997
------------------->GetH245CodecName() return =  PCMA
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:2036
h323/231543514280633 initialise  read codec Audio for connection
[0x9a9d488]
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:2091 Set  read Audio
codec to G.711-ALaw-64k  for connection [0xb773b9a0]
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:2097
------------------->tech_pvt->rtp_session = [0xb794bf18]
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:2098
------------------->samples_per_packet = 160
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:2099
------------------->actual_samples_per_second = 8000
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:2103
------------------->old remot port = 5062 new remote port = 5062
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:2169
------------->External RTP address 200.117.192.17:5062
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:2172
------------->h323_io_mutex_lock
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:2176
------------->h323_io_mutex_unlock
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:2215
------------->h323_mutex_unlock
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:1903
======>FSH323_ExternalRTPChannel::~FSH323_ExternalRTPChannel
IsTransmitter [0xb7922880]
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:1906
------------->switch_core_session_unlock_codec_read [0x9bfe898]
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:1908
------------->switch_core_session_unlock_codec_write [0x9bfe898]
2010-12-14 12:38:57.662268 [DEBUG] h323.cxx:5105 Bandwidth request:
-0.0kb/s, available: 9872.0kb/s
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:1903
======>FSH323_ExternalRTPChannel::~FSH323_ExternalRTPChannel  IsReceiver
[0xb791e1d8]
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:1906
------------->switch_core_session_unlock_codec_read [0x9bfe898]
2010-12-14 12:38:57.662268 [DEBUG] mod_h323.cpp:1908
------------->switch_core_session_unlock_codec_write [0x9bfe898]
2010-12-14 12:38:57.662268 [DEBUG] h323.cxx:5105 Bandwidth request:
-0.0kb/s, available: 9872.0kb/s
2010-12-14 12:38:57.662268 [DEBUG] h323.cxx:3094 Accepting fastStart for
2 channels
2010-12-14 12:38:57.665259 [DEBUG] channels.cxx:989 OnSendingPDU
2010-12-14 12:38:57.665259 [DEBUG] h323.cxx:1652 Build fastStart:
  {
    forwardLogicalChannelNumber = 101
    forwardLogicalChannelParameters = {
      dataType = nullData <<null>>
      multiplexParameters = none <<null>>
    }
    reverseLogicalChannelParameters =
 {
      dataType = audioData g711Alaw64k 20
      multiplexParameters = h2250LogicalChannelParameters {
        sessionID = 1
        mediaGuaranteedDelivery = false
        mediaControlChannel = unicastAddress iPAddress {
          network =  4 oc
tets {
            48 33 2f 64                                        H3/d
          }
          tsapIdentifier = 56791
        }
        silenceSuppression = false
      }
    }
  }
2010-12-14 12:38:57.665259 [DEBUG] h323.cxx:1657 Built fastStart for
G.711-ALaw-64k <1>
2010-12-14 12:38:57.665259 [DEBUG] channels.cxx:989 OnSendingPDU
2010-12-14 12:38:57.665259 [DEBUG] h323.cxx:1652 Build fastStart:
  {
    forwardLogicalChannelNumber = 101
    forwardLogicalChannelParameters = {
      dataType = audioData g711Alaw64k 20
      multiplexParameters = h2250LogicalChannelParameters {
        sessionID 
= 1
        mediaChannel = unicastAddress iPAddress {
          network =  4 octets {
            48 33 2f 64                                        H3/d
          }
          tsapIdentifier = 56790
        }
        mediaGuaranteedDelivery = false
        mediaControlChannel = unicastAddress iPAddress {
          network =  4 octets {
            48 33 2f 64                                        H3/d
          }
          tsapIdentifier = 56791
        }
        silenceSuppression = false
  
Log-Func: 
Log-LINCOMING DATA [(null)]

RECV EVENT
Event-Name: SOCKET_DATA
ine: 0
User-Data: _undef_


2010-12-14 12:38:57.665259 [DEBUG] h323.cxx:1657 Built fastStart for
G.711-ALaw-64k <1>
2010-12-14 12:38:57.668246 [DEBUG] h323pdu.cxx:618 Sending PDU [ip
$72.51.47.100:1720/ip$200.117.192.17:20570] :
  {
    q931pdu = {
      protocolDiscriminator = 8
      callReference = 4293
      from = destination
      messageType = CallProceeding
      IE: Display = 
(....=.
        46 72 65 65 53 57 49 54  43 48 20 6d 6f 64 5f 68   FreeSWITCH m
od_h
        ...
      }
    }
    h225pdu = {
      h323_uu_pdu = {
        h323_message_body = callProceeding {
          protocolIdentifier = 0.0.8.2250.0.6
          destinationInfo = {
            vendor = {
              vendor = {
           
     t35CountryCode = 9
                t35Extension = 0
                manufacturerCode = 61
              }
              productId =  21 octets {
                46 72 65 65 53 57 49 54  43 48 20 6d 6f 64 5f 68
FreeSWITCH mod_h
               
(H323p
                6c 75 73 20 76 31 2e 32  32 2e 30 29
 00 00         lus v1.22.0)..
              }
            }
            gateway = {
            }
            mc = false
            undefinedNode = false
          }
          callIdentifier = {
            guid =  16 octets {
              a2 4e 5
8 c9 ec 05 e0 11  9e 19 00 1e 64 1f 33 fe   .NX.........d.3.
            }
          }
          fastStart = 2 entries {
            [0]= 23 octets {
              40 00 64 06 04 01 00 4c  20 13 80 0b 0d 00 01 00
@.d....L .......
              48 
33 2f 64 dd d7 00                               H3/d...
            }
            [1]= 26 octets {
              00 00 64 0c 20 13 80 12  1d 00 01 00 48 33 2f
64   ..d. .......H3/d
              dd d6 00 48 33 2f 64 dd  d7 00                     ...
H3/d...
            }
          }
          multipleCalls = false
          maintainConnection = false
        }
        h245Tunneling = true
      }
    }
  }
2010-12-14 12:38:57.668246 [DEBUG] h323pdu.cxx:618 Receiving PDU
[(noaddr)/(noaddr)] :
  request terminalCapabilitySet {
    sequenceNumber = 1
    protocolIdentifier = 0.0.8.245.0.13
    multiplexCapability = h2250Capability {
      maximumAudioDelayJitter = 500
      r
eceiveMultipointCapability = {
        multicastCapability = false
        multiUniCastConference = false
        mediaDistributionCapability = 1 entries {
          [0]={
            centralizedControl = false
            distributedControl = false

            centralizedAudio = false
            distributedAudio = false
            centralizedVideo = false
            distributedVideo = false
          }
        }
      }
      transmitMultipointCapability = {
        multicastCapability = f
alse
        multiUniCastConference = false
        mediaDistributionCapability = 1 entries {
          [0]={
            centralizedControl = false
            distributedControl = false
            centralizedAudio = false
            distributedA
udio = false
            centralizedVideo = false
            distributedVideo = false
          }
        }
      }
      receiveAndTransmitMultipointCapability = {
        multicastCapability = false
        multiUniCastConference = false
        
mediaDistributionCapability = 1 entries {
          [0]={
            centralizedControl = false
            distributedControl = false
            centralizedAudio = false
            distributedAudio = false
            centralizedVideo = false
  
          distributedVideo = false
          }
        }
      }
      mcCapability = {
        centralizedConferenceMC = false
        decentralizedConferenceMC = false
      }
      rtcpVideoControlCapability = false
      mediaPacketizationCapabi
lity = {
        h261aVideoPacketization = false
      }
      logicalChannelSwitchingCapability = false
      t120DynamicPortCapability = true
    }
    capabilityTable = 10 entries {
      [0]={
        capabilityTableEntryNumber = 1
        capab
ility = receiveAudioCapability g711Alaw64k 240
      }
      [1]={
        capabilityTableEntryNumber = 2
        capability = receiveAudioCapability gsmFullRate {
          audioUnitSize = 231
          comfortNoise = false
          scrambled = fa
lse
        }
      }
      [2]={
        capabilityTableEntryNumber = 3
        capability = receiveAudioCapability nonStandard [Equivalence
G.726-16k] {
          nonStandardIdentifier = h221NonStandard {
            t35CountryCode = 9
           
 t35Extension = 0
            manufacturerCode = 61
          }
          data =  9 octets {
            47 2e 37 32 36 2d 31 36  6b                        G.726-16k
          }
        }
      }
      [3]={
        capabilityTableEntryNumber = 4
  
      capability = receiveVideoCapability h261VideoCapability {
          qcifMPI = 1
          cifMPI = 1
          temporalSpatialTradeOffCapability = false
          maxBitRate = 19200
          stillImageTransmission = false
          videoBadMB
sCap = false
        }
      }
      [4]={
        capabilityTableEntryNumber = 5
        capability = receiveVideoCapability h261VideoCapability {
          cifMPI = 1
          temporalSpatialTradeOffCapability = false
          maxBitRate = 19200

          stillImageTransmission = false
          videoBadMBsCap = false
        }
      }
      [5]={
        capabilityTableEntryNumber = 6
        capability = receiveVideoCapability h261VideoCapability {
          qcifMPI = 1
          tempora
lSpatialTradeOffCapability = false
          maxBitRate = 19200
          stillImageTransmission = false
          videoBadMBsCap = false
        }
      }
      [6]={
        capabilityTableEntryNumber = 7
        capability = receiveUserInputCapab
ility hookflash <<null>>
      }
      [7]={
        capabilityTableEntryNumber = 8
        capability = receiveUserInputCapability basicString <<null>>
      }
      [8]={
        capabilityTableEntryNumber = 9
        capability = receiveUserInput
Capability dtmf <<null>>
      }
      [9]={
        capabilityTableEntryNumber = 10
        capability = receiveRTPAudioTelephonyEventCapability {
          dynamicRTPPayloadType = 101
          audioTelephoneEvent = "0-16"
        }
      }
    }
    capabilityDescriptors = 1 entries {
      [0]={
        capabilityDescriptorNumber = 1
        simultaneousCapabilities = 4 entries {
          [0]=3 entries {
            [0]=1
            [1]=2
            [2]=3
          }
          [1]=3 ent
ries {
            [0]=4
            [1]=5
            [2]=6
          }
          [2]=1 entries {
            [0]=7
          }
          [3]=3 entries {
            [0]=8
            [1]=9
            [2]=10
          }
        }
      }
    }
  }

2010-12-14 12:38:57.668246 [DEBUG] h323.cxx:3611 Set protocol version to
13
2010-12-14 12:38:57.668246 [DEBUG] h323neg.cxx:631 Received
TerminalCapabilitySet: state=Idle pduSeq=1 inSeq=4294967295
Dialplan: h323/231543514280633 parsing [default->external]
continue=false
Dialplan: h323/231543514280633 Regex (PASS) [external]
destination_number(231543514280633) =~ /^231543514280633/ break=on-false
Dialplan: h323/231543514280633 Action set(continue_on_fail=true) 
Dialplan: h323/231543514280633 Action set(hangup_after_bridge=true) 
Dialplan: h323/231543514280633 Action set(progress_timeout=15) 
Dialplan: h323/231543514280633 Action set(proxy_media=true) 
Dialplan: h323/231543514280633 Action set(bypass_media=false) 
Dialplan: h323/231543514280633 Action
set(absolute_codec_string=GSM at 40i,G726-16,PCMA) 
Dialplan: h323/231543514280633 Action
bridge(sofia/external/21543514280633 at 200.35.145.149) 
2010-12-14 12:38:57.671243 [DEBUG] switch_core_state_machine.c:119
(h323/231543514280633) State Change CS_ROUTING -> CS_EXECUTE
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1584
======>FSH323Connection::state_change [0x9a7d140]
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3598 Added capability:
G.711-ALaw-64k <1>
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1585 State changed on
connection [0x9a7d140]
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3598 Added capability:
G.711-uLaw-64k <2>
2010-12-14 12:38:57.671243 [DEBUG] switch_core_session.c:1083 Send
signal h323/231543514280633 [BREAK]
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1382
======>FSH323Connection::kill_channel sig = 3 [0x9a7d140]
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3598 Added capability:
GSM-06.10{sw} <3>
2010-12-14 12:38:57.671243 [DEBUG] switch_core_state_machine.c:359
(h323/231543514280633) State ROUTING going to sleep
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3598 Added capability:
G.729A/B{sw} <4>
2010-12-14 12:38:57.671243 [DEBUG] switch_core_state_machine.c:320
(h323/231543514280633) Running State Change CS_EXECUTE
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3598 Added capability:
G.729A{sw} <5>
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3598 Added capability:
G.729B{sw} <6>
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3598 Added capability:
G.729{sw} <7>
2010-12-14 12:38:57.671243 [DEBUG] switch_core_state_machine.c:366
(h323/231543514280633) State EXECUTE
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1368
======>FSH323Connection::on_execute [0x9a7d140]
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3598 Added capability:
T.38-IFP-PRE <8>
2010-12-14 12:38:57.671243 [DEBUG] switch_core_state_machine.c:157
h323/231543514280633 Standard EXECUTE
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/hookflash <9>
EXECUTE h323/231543514280633 set(continue_on_fail=true)
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1460
======>FSH323Connection::receive_message MSG = 26
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1567 Received message id
= 26 [0x9a7d140]
2010-12-14 12:38:57.671243 [DEBUG] mod_dptools.c:1028
h323/231543514280633 SET [continue_on_fail]=[true]
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1460
======>FSH323Connection::receive_message MSG = 27
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1567 Received message id
= 27 [0x9a7d140]
EXECUTE h323/231543514280633 set(hangup_after_bridge=true)
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1460
======>FSH323Connection::receive_message MSG = 26
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1567 Received message id
= 26 [0x9a7d140]
2010-12-14 12:38:57.671243 [DEBUG] mod_dptools.c:1028
h323/231543514280633 SET [hangup_after_bridge]=[true]
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1460
======>FSH323Connection::receive_message MSG = 27
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1567 Received message id
= 27 [0x9a7d140]
EXECUTE h323/231543514280633 set(progress_timeout=15)
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/basicString <10>
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1460
======>FSH323Connection::receive_message MSG = 26
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1567 Received message id
= 26 [0x9a7d140]
2010-12-14 12:38:57.671243 [DEBUG] mod_dptools.c:1028
h323/231543514280633 SET [progress_timeout]=[15]
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/dtmf <11>
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/RFC2833 <12>
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/Navigation <13>
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/Softkey <14>
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1460
======>FSH323Connection::receive_message MSG = 27
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1567 Received message id
= 27 [0x9a7d140]
EXECUTE h323/231543514280633 set(proxy_media=true)
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/PointDevice <15>
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/Modal <16>
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3727 FindCapability:
receiveAudioCapability
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:4002 FindCapability:
Audio subtype=1
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1460
======>FSH323Connection::receive_message MSG = 26
2010-12-14 12:38:57.671243 [DEBUG] mod_h323.cpp:1567 Received message id
= 26 [0x9a7d140]
2010-12-14 12:38:57.671243 [DEBUG] mod_dptools.c:1028
h323/231543514280633 SET [proxy_media]=[true]
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:4009 Found capability:
G.711-ALaw-64k <1>
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:1523 Capability tx
frames left at 20 as remote allows 240
2010-12-14 12:38:57.671243 [DEBUG] h323caps.cxx:3727 FindCapability:
receiveAudioCapability
2010-12-14 12:38:57.674242 [DEBUG] mod_h323.cpp:1460
======>FSH323Connection::receive_message MSG = 27
2010-12-14 12:38:57.674242 [DEBUG] mod_h323.cpp:1567 Received message id
= 27 [0x9a7d140]
EXECUTE h323/231543514280633 set(bypass_media=false)
2010-12-14 12:38:57.674242 [DEBUG] mod_h323.cpp:1460
======>FSH323Connection::receive_message MSG = 26
2010-12-14 12:38:57.674242 [DEBUG] mod_h323.cpp:1567 Received message id
= 26 [0x9a7d140]
2010-12-14 12:38:57.674242 [DEBUG] mod_dptools.c:1028
h323/231543514280633 SET [bypass_media]=[false]
2010-12-14 12:38:57.674242 [DEBUG] mod_h323.cpp:1460
======>FSH323Connection::receive_message MSG = 27
2010-12-14 12:38:57.674242 [DEBUG] mod_h323.cpp:1567 Received message id
= 27 [0x9a7d140]
EXECUTE h323/231543514280633
set(absolute_codec_string=GSM at 40i,G726-16,PCMA)
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:4002 FindCapability:
Audio subtype=17
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:4009 Found capability:
GSM-06.10{sw} <3>
2010-12-14 12:38:57.674242 [DEBUG] mod_h323.h:529
==============>BaseGSM0610Cap::OnReceivedPDU [0xb7948a88]
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:1523 Capability tx
frames left at 1 as remote allows 7
2010-12-14 12:38:57.674242 [DEBUG] mod_h323.cpp:1460
======>FSH323Connection::receive_message MSG = 26
2010-12-14 12:38:57.674242 [DEBUG] mod_h323.cpp:1567 Received message id
= 26 [0x9a7d140]
2010-12-14 12:38:57.674242 [DEBUG] mod_dptools.c:1028
h323/231543514280633 SET [absolute_codec_string]=[GSM at 40i,G726-16,PCMA]
2010-12-14 12:38:57.674242 [DEBUG] mod_h323.cpp:1460
======>FSH323Connection::receive_message MSG = 27
2010-12-14 12:38:57.674242 [DEBUG] mod_h323.cpp:1567 Received message id
= 27 [0x9a7d140]
EXECUTE h323/231543514280633
bridge(sofia/external/21543514280633 at 200.35.145.149)
2010-12-14 12:38:57.674242 [DEBUG] mod_h323.cpp:1460
======>FSH323Connection::receive_message MSG = 26
2010-12-14 12:38:57.674242 [DEBUG] mod_h323.cpp:1567 Received message id
= 26 [0x9a7d140]
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:3727 FindCapability:
receiveAudioCapability
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:3984 FindCapability:
Audio nonStandard
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:3727 FindCapability:
receiveVideoCapability
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:4002 FindCapability:
Video subtype=1
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:3727 FindCapability:
receiveVideoCapability
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:4002 FindCapability:
Video subtype=1
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:3727 FindCapability:
receiveVideoCapability
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:4002 FindCapability:
Video subtype=1
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:3727 FindCapability:
receiveUserInputCapability
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:4002 FindCapability:
UserInput subtype=5
2010-12-14 12:38:57.674242 [NOTICE] switch_channel.c:784 New Channel
sofia/external/21543514280633 at 200.35.145.149
[de3adda4-e34b-4ec0-99ca-b3ec50a2a267]
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:4009 Found capability:
UserInput/hookflash <9>
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:3727 FindCapability:
receiveUserInputCapability
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:4002 FindCapability:
UserInput subtype=1
2010-12-14 12:38:57.674242 [DEBUG] mod_sofia.c:3995
(sofia/external/21543514280633 at 200.35.145.149) State Change CS_NEW ->
CS_INIT
2010-12-14 12:38:57.674242 [DEBUG] switch_core_session.c:1083 Send
signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:4009 Found capability:
UserInput/basicString <10>
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:3727 FindCapability:
receiveUserInputCapability
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:4002 FindCapability:
UserInput subtype=4
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:4009 Found capability:
UserInput/dtmf <11>
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:3727 FindCapability:
receiveRTPAudioTelephonyEventCapability
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:4002 FindCapability:
UserInput subtype=10000
2010-12-14 12:38:57.674242 [DEBUG] h323caps.cxx:4009 Found capability:
UserInput/RFC2833 <12>
2010-12-14 12:38:57.677302 [DEBUG] switch_core_state_machine.c:320
(sofia/external/21543514280633 at 200.35.145.149) Running State Change
CS_INIT
2010-12-14 12:38:57.677302 [DEBUG] switch_core_state_machine.c:356
(sofia/external/21543514280633 at 200.35.145.149) State INIT
2010-12-14 12:38:57.677302 [DEBUG] mod_sofia.c:86
sofia/external/21543514280633 at 200.35.145.149 SOFIA INIT
2010-12-14 12:38:57.677302 [DEBUG] mod_h323.cpp:1103
======>FSH323Connection::OnReceivedCapabilitySet [0x9a7d140]
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3712 FindCapability:
G.711-ALaw-64k <1>
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3598 Added capability:
G.711-ALaw-64k <1>
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3712 FindCapability:
GSM-06.10{sw} <2>
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3598 Added capability:
GSM-06.10{sw} <2>
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3712 FindCapability:
UserInput/hookflash <7>
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/hookflash <7>
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3712 FindCapability:
UserInput/basicString <8>
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/basicString <8>
2010-12-14 12:38:57.677302 [DEBUG] mod_sofia.c:126
(sofia/external/21543514280633 at 200.35.145.149) State Change CS_INIT ->
CS_ROUTING
2010-12-14 12:38:57.677302 [DEBUG] switch_core_session.c:1083 Send
signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3712 FindCapability:
UserInput/dtmf <9>
2010-12-14 12:38:57.677302 [DEBUG] switch_core_state_machine.c:356
(sofia/external/21543514280633 at 200.35.145.149) State INIT going to sleep
2010-12-14 12:38:57.677302 [DEBUG] switch_core_state_machine.c:320
(sofia/external/21543514280633 at 200.35.145.149) Running State Change
CS_ROUTING
2010-12-14 12:38:57.677302 [DEBUG] sofia.c:4604 Channel
sofia/external/21543514280633 at 200.35.145.149 entering state
[terminated][900]
2010-12-14 12:38:57.677302 [DEBUG] switch_channel.c:1615
(sofia/external/21543514280633 at 200.35.145.149) Callstate Change DOWN ->
RINGING
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/dtmf <9>
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3712 FindCapability:
UserInput/RFC2833 <10>
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3598 Added capability:
UserInput/RFC2833 <10>
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3660 FindCapability: 1
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3664 Found capability:
G.711-ALaw-64k <1>
2010-12-14 12:38:57.677302 [DEBUG] switch_core_state_machine.c:359
(sofia/external/21543514280633 at 200.35.145.149) State ROUTING
2010-12-14 12:38:57.677302 [DEBUG] mod_sofia.c:149
sofia/external/21543514280633 at 200.35.145.149 SOFIA ROUTING
2010-12-14 12:38:57.677302 [DEBUG] switch_ivr_originate.c:66
(sofia/external/21543514280633 at 200.35.145.149) State Change CS_ROUTING
-> CS_CONSUME_MEDIA
2010-12-14 12:38:57.677302 [DEBUG] switch_core_session.c:1083 Send
signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
2010-12-14 12:38:57.677302 [DEBUG] switch_core_state_machine.c:359
(sofia/external/21543514280633 at 200.35.145.149) State ROUTING going to
sleep
2010-12-14 12:38:57.677302 [DEBUG] switch_channel.c:2455
(sofia/external/21543514280633 at 200.35.145.149) Callstate Change RINGING
-> HANGUP
2010-12-14 12:38:57.677302 [DEBUG] switch_core_state_machine.c:320
(sofia/external/21543514280633 at 200.35.145.149) Running State Change
CS_HANGUP
2010-12-14 12:38:57.677302 [NOTICE] sofia.c:5244 Hangup
sofia/external/21543514280633 at 200.35.145.149 [CS_CONSUME_MEDIA]
[NORMAL_UNSPECIFIED]
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3660 FindCapability: 2
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3664 Found capability:
GSM-06.10{sw} <2>
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3660 FindCapability: 7
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3664 Found capability:
UserInput/hookflash <7>
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3660 FindCapability: 8
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3664 Found capability:
UserInput/basicString <8>
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3660 FindCapability: 9
2010-12-14 12:38:57.677302 [DEBUG] h323caps.cxx:3664 Found capability:
UserInput/dtmf <9>
2010-12-14 12:38:57.680240 [DEBUG] switch_ivr_originate.c:3448 Originate
Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED]
2010-12-14 12:38:57.680240 [INFO] mod_dptools.c:2579 Originate Failed.
Cause: NORMAL_UNSPECIFIED
2010-12-14 12:38:57.680240 [DEBUG] mod_h323.cpp:1460
======>FSH323Connection::receive_message MSG = 27
2010-12-14 12:38:57.680240 [DEBUG] mod_h323.cpp:1567 Received message id
= 27 [0x9a7d140]
2010-12-14 12:38:57.680240 [NOTICE] switch_core_state_machine.c:189
h323/231543514280633 has executed the last dialplan instruction, hanging
up.
2010-12-14 12:38:57.680240 [DEBUG] switch_channel.c:2455
(h323/231543514280633) Callstate Change RINGING -> HANGUP
2010-12-14 12:38:57.680240 [DEBUG] h323caps.cxx:3660 FindCapability: 10
2010-12-14 12:38:57.680240 [DEBUG] h323caps.cxx:3664 Found capability:
UserInput/RFC2833 <10>
2010-12-14 12:38:57.680240 [NOTICE] switch_core_state_machine.c:191
Hangup h323/231543514280633 [CS_EXECUTE] [NORMAL_CLEARING]
2010-12-14 12:38:57.680240 [DEBUG] h323caps.cxx:4197 Capability merge
result:
     Table:
       G.711-ALaw-64k <1>
       GSM-06.10{sw} <2>
       UserInput/hookflash <7>
       UserInput/basicString <8>
       UserInput/dtmf <9>
       UserInput/RFC2833 <10>
     Set:
 
      0:
         0:
           G.711-ALaw-64k <1>
           GSM-06.10{sw} <2>
         1:
         2:
           UserInput/hookflash <7>
         3:
           UserInput/basicString <8>
           UserInput/dtmf <9>
           UserInput/RFC2833 <1

LogINCOMING DATA [(null)]

RECV EVENT
Event-Name: SOCKET_DATA
-Func: _undef_
Log-Line: 0
User-Data: _undef_


INCOMING DATA [(null)]

RECV EVENT
Event-Name: SOCKET_DATA


2010-12-14 12:38:57.680240 [DEBUG] h323caps.cxx:4199 Received capability
set, is accepted
2010-12-14 12:38:57.680240 [DEBUG] switch_channel.c:2471 Send signal
h323/231543514280633 [KILL]
2010-12-14 12:38:57.680240 [DEBUG] mod_h323.cpp:1382
======>FSH323Connection::kill_channel sig = 1 [0x9a7d140]
2010-12-14 12:38:57.680240 [DEBUG] mod_h323.cpp:1407 --->Kill soket
[0x9a7d140]
2010-12-14 12:38:57.680240 [DEBUG] mod_h323.cpp:1584
======>FSH323Connection::state_change [0x9a7d140]
2010-12-14 12:38:57.680240 [DEBUG] mod_h323.cpp:1585 State changed on
connection [0x9a7d140]
2010-12-14 12:38:57.680240 [DEBUG] switch_core_session.c:1083 Send
signal h323/231543514280633 [BREAK]
2010-12-14 12:38:57.680240 [DEBUG] mod_h323.cpp:1382
======>FSH323Connection::kill_channel sig = 3 [0x9a7d140]
2010-12-14 12:38:57.680240 [DEBUG] switch_core_state_machine.c:366
(h323/231543514280633) State EXECUTE going to sleep
2010-12-14 12:38:57.680240 [DEBUG] switch_core_state_machine.c:320
(h323/231543514280633) Running State Change CS_HANGUP
2010-12-14 12:38:57.680240 [DEBUG] h323neg.cxx:600 Sending
TerminalCapabilitySet: outSeq=1
2010-12-14 12:38:57.680240 [DEBUG] switch_core_state_machine.c:557
(h323/231543514280633) State HANGUP
2010-12-14 12:38:57.680240 [DEBUG] mod_h323.cpp:2338
======>switch_status_t on_hangup [0x9bfe898]
2010-12-14 12:38:57.680240 [DEBUG] mod_h323.cpp:2353
------------->h323_mutex_lock
2010-12-14 12:38:57.680240 [DEBUG] mod_h323.cpp:2356
------------->h323_mutex_unlock
2010-12-14 12:38:57.680240 [DEBUG] mod_h323.cpp:2363 ----->ip
$200.117.192.17:20570/4293
2010-12-14 12:38:57.680240 [DEBUG] mod_h323.cpp:2366 -----> () = -1
2010-12-14 12:38:57.680240 [DEBUG] h323pdu.cxx:1281 Call End Reason
Normal call clearing
2010-12-14 12:38:57.680240 [DEBUG] h323ep.cxx:2802 Clearing connection
ip$200.117.192.17:20570/4293 reason=EndedByRemoteUser
2010-12-14 12:38:57.680240 [DEBUG] h323.cxx:1139 Call end reason for ip
$200.117.192.17:20570/4293 set to EndedByRemoteUser
2010-12-14 12:38:57.680240 [DEBUG] h323.cxx:1157 Sending release
complete PDU: callRef=4293
2010-12-14 12:38:57.680240 [DEBUG] h323t38.cxx:216 OnSendingPDU for
capability
2010-12-14 12:38:57.683262 [DEBUG] mod_h323.cpp:1085
======>FSH323Connection::OnSendReleaseComplete cause = 16
2010-12-14 12:38:57.683262 [DEBUG] h323pdu.cxx:618 Sending PDU [ip
$72.51.47.100:1720/ip$200.117.192.17:20570] :
  {
    q931pdu = {
      protocolDiscriminator = 8
      callReference = 4293
      from = destination
      messageType = ReleaseComplete
      IE: Cause - N
ormal call clearing = {
        80 90                                              ..
      }
      IE: User-User = {
        25 80 06 00 08 91 4a 00  06 01 11 00 a2 4e 58 c9
%.....J......NX.
        ec 05 e0 11 9e 19 00 1e  64 1f 33 fe 02 80 01 8
0   ........d.3.....
      }
    }
    h225pdu = {
      h323_uu_pdu = {
        h323_message_body = releaseComplete {
          protocolIdentifier = 0.0.8.2250.0.6
          callIdentifier = {
            guid =  16 octets {
              a2 4e 58 
c9 ec 05 e0 11  9e 19 00 1e 64 1f 33 fe   .NX.........d.3.
            }
          }
        }
        h245Tunneling = true
      }
    }
  }
2010-12-14 12:38:57.683262 [DEBUG] h323ep.cxx:2861 Cleaning up
connections
2010-12-14 12:38:57.683262 [DEBUG] h323.cxx:1199 Connection ip
$200.117.192.17:20570/4293 closing: connectionState=AwaitingLocalAnswer
2010-12-14 12:38:57.683262 [DEBUG] mod_h323.cpp:2387
------------->h323_mutex_lock
2010-12-14 12:38:57.683262 [DEBUG] mod_h323.cpp:2389
------------->h323_mutex_unlock
2010-12-14 12:38:57.683262 [DEBUG] switch_core_state_machine.c:46
h323/231543514280633 Standard HANGUP, cause: NORMAL_CLEARING
2010-12-14 12:38:57.683262 [DEBUG] switch_core_state_machine.c:557
(h323/231543514280633) State HANGUP going to sleep
2010-12-14 12:38:57.683262 [DEBUG] switch_core_state_machine.c:351
(h323/231543514280633) State Change CS_HANGUP -> CS_REPORTING
2010-12-14 12:38:57.683262 [DEBUG] switch_core_session.c:1083 Send
signal h323/231543514280633 [BREAK]
2010-12-14 12:38:57.683262 [DEBUG] switch_core_state_machine.c:320
(h323/231543514280633) Running State Change CS_REPORTING
2010-12-14 12:38:57.683262 [DEBUG] switch_core_state_machine.c:617
(h323/231543514280633) State REPORTING
2010-12-14 12:38:57.683262 [DEBUG] h323pdu.cxx:618 Sending PDU
[(noaddr)/(noaddr)] :
  request terminalCapabilitySet {
    sequenceNumber = 1
    protocolIdentifier = 0.0.8.245.0.13
    multiplexCapability = h2250Capability {
      maximumAudioDelayJitter = 60
      rece
iveMultipointCapability = {
        multicastCapability = false
        multiUniCastConference = false
        mediaDistributionCapability = 1 entries {
          [0]={
            centralizedControl = false
            distributedControl = false
  
2010-12-14 12:38:57.683262 [DEBUG] switch_core_state_machine.c:557
(sofia/external/21543514280633 at 200.35.145.149) State HANGUP
2010-12-14 12:38:57.686742 [DEBUG] switch_channel.c:2471 Send signal
sofia/external/21543514280633 at 200.35.145.149 [KILL]
2010-12-14 12:38:57.686742 [DEBUG] switch_core_session.c:1083 Send
signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
2010-12-14 12:38:57.686742 [DEBUG] mod_sofia.c:453
sofia/external/21543514280633 at 200.35.145.149 Overriding SIP cause 480
with 900 from the other leg
2010-12-14 12:38:57.686742 [DEBUG] mod_sofia.c:459 Channel
sofia/external/21543514280633 at 200.35.145.149 hanging up, cause:
NORMAL_UNSPECIFIED
2010-12-14 12:38:57.686742 [DEBUG] switch_core_state_machine.c:46
sofia/external/21543514280633 at 200.35.145.149 Standard HANGUP, cause:
NORMAL_UNSPECIFIED
2010-12-14 12:38:57.686742 [DEBUG] switch_core_state_machine.c:557
(sofia/external/21543514280633 at 200.35.145.149) State HANGUP going to
sleep
2010-12-14 12:38:57.686742 [DEBUG] switch_core_state_machine.c:351
(sofia/external/21543514280633 at 200.35.145.149) State Change CS_HANGUP ->
CS_REPORTING
2010-12-14 12:38:57.686742 [DEBUG] switch_core_session.c:1083 Send
signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
2010-12-14 12:38:57.686742 [DEBUG] switch_core_state_machine.c:320
(sofia/external/21543514280633 at 200.35.145.149) Running State Change
CS_REPORTING
          centralizedAudio = false
            distributedAudio = false
            centralizedVideo = false
            distributedVideo = false
          }
        }
      }
      transmitMultipointCapability = {
        multicastCapability = fals
e
        multiUniCastConference = false
        mediaDistributionCapability = 1 entries {
          [0]={
            centralizedControl = false
            distributedControl = false
            centralizedAudio = false
            distributedAudi
o = false
            centralizedVideo = false
            distributedVideo = false
          }
        }
      }
      receiveAndTransmitMultipointCapability = {
        multicastCapability = false
        multiUniCastConference = false
        med
iaDistributionCapability = 1 entries {
          [0]={
            centralizedControl = false
            distributedControl = false
            centralizedAudio = false
            distributedAudio = false
            centralizedVideo = false
     
       distributedVideo = false
          }
        }
      }
      mcCapability = {
        centralizedConferenceMC = false
        decentralizedConferenceMC = false
      }
      rtcpVideoControlCapability = false
      mediaPacketizationCapabilit
y = {
        h261aVideoPacketization = false
      }
      logicalChannelSwitchingCapability = false
      t120DynamicPortCapability = true
    }
    capabilityTable = 16 entries {
      [0]={
        capabilityTableEntryNumber = 1
        capabili
ty = receiveAudioCapability g711Alaw64k 20
      }
      [1]={
        capabilityTableEntryNumber = 2
        capability = receiveAudioCapability g711Ulaw64k 20
      }
      [2]={
        capabilityTableEntryNumber = 3
        capability = receiveA
udioCapability gsmFullRate {
          audioUnitSize = 33
          comfortNoise = false
          scrambled = false
        }
      }
      [3]={
        capabilityTableEntryNumber = 4
        capability = receiveAudioCapability g729AnnexAwAnnexB 2
4
      }
      [4]={
        capabilityTableEntryNumber = 5
        capability = receiveAudioCapability g729AnnexA 24
      }
      [5]={
        capabilityTableEntryNumber = 6
        capability = receiveAudioCapability g729wAnnexB 24
      }
    
  [6]={
        capabilityTableEntryNumber = 7
        capability = receiveAudioCapability g729 24
      }
      [7]={
        capabilityTableEntryNumber = 8
        capability = receiveAndTransmitDataApplicationCapability {
          application = 
t38fax {
            t38FaxProtocol = udp <<null>>
            t38FaxProfile = {
              fillBitRemoval = false
              transcodingJBIG = false
              transcodingMMR = false
              version = 0
              t38FaxRateManage
ment = transferredTCF <<null>>
              t38FaxUdpOptions = {
                t38FaxMaxBuffer = 200
                t38FaxMaxDatagram = 72
                t38FaxUdpEC = t38UDPRedundancy <<null>>
              }
            }
          }
        
  maxBitRate = 144
        }
      }
      [8]={
        capabilityTableEntryNumber = 9
        capability = receiveUserInputCapability hookflash <<null>>
      }
      [9]={
        capabilityTableEntryNumber = 10
        capability = receiveUserIn
putCapability basicString <<null>>
      }
      [10]={
        capabilityTableEntryNumber = 11
        capability = receiveUserInputCapability dtmf <<null>>
      }
      [11]={
        capabilityTableEntryNumber = 12
        capability = receiveRT
PAudioTelephonyEventCapability {
          dynamicRTPPayloadType = 101
          audioTelephoneEvent = "0-16"
        }
      }
      [12]={
        capabilityTableEntryNumber = 13
        capability = receiveUserInputCapability genericUserInputCapa
bility {
          capabilityIdentifier = standard 0.0.8.249.1
        }
      }
      [13]={
        capabilityTableEntryNumber = 14
        capability = receiveUserInputCapability
genericUserInputCapability {
          capabilityIdentifier = stand
ard 0.0.8.249.2
        }
      }
      [14]={
        capabilityTableEntryNumber = 15
        capability = receiveUserInputCapability
genericUserInputCapability {
          capabilityIdentifier = standard 0.0.8.249.3
        }
      }
      [15]={
        capabilityTableEntryNumber = 16
        capability = receiveUserInputCapability
genericUserInputCapability {
          capabilityIdentifier = standard 0.0.8.249.4
        }
      }
    }
    capabilityDescriptors = 1 entries {
      [0]={
  
      capabilityDescriptorNumber = 1
        simultaneousCapabilities = 2 entries {
          [0]=8 entries {
            [0]=1
            [1]=2
            [2]=3
            [3]=4
            [4]=5
            [5]=6
            [6]=7
            [
Log-Func: 
Log-Line: 0
User-Data: 

7]=8
          }
          [1]=8 entries {
            [0]=9
            [1]=10
            [2]=11
            [3]=12
            [4]=13
            [5]=14
            [6]=15
            [7]=16
    2010-12-14 12:38:57.689292 [DEBUG] h323pdu.cxx:618 Sending PDU [ip
$72.51.47.100:1720/ip$200.117.192.17:20570] :
  {
    q931pdu = {
      protocolDiscriminator = 8
      callReference = 4293
      from = destination
      messageType = Facility
      IE: Facility = {

  
(.............p.
        06 00 08 81 75 00 0d 80  13 80 00 3c 00 01 00
00   ....u......<....
        ...
      }
    }
    h225pdu = {
      h323_uu_pdu = {
   
     h323_message_body = empty <<null>>
        h245Tunneling = true
        h245Control = 1 entries {
          [0]= 231 octets {
            02 70 01 06 00 08 81 75  00 0d 80 13 80 00 3c
00   .p.....u......<.
            01 00 00 01 00 00 01 00  0
0 0c c0 01 00 01 80 0f   ................
            ...
          }
        }
      }
    }
  }
2010-12-14 12:38:57.689292 [DEBUG] switch_core_state_machine.c:53
h323/231543514280633 Standard REPORTING, cause: NORMAL_CLEARING
2010-12-14 12:38:57.689292 [DEBUG] switch_core_state_machine.c:617
(h323/231543514280633) State REPORTING going to sleep
2010-12-14 12:38:57.689292 [DEBUG] switch_core_state_machine.c:345
(h323/231543514280633) State Change CS_REPORTING -> CS_DESTROY
2010-12-14 12:38:57.689292 [DEBUG] switch_core_session.c:1083 Send
signal h323/231543514280633 [BREAK]
2010-12-14 12:38:57.689292 [DEBUG] switch_core_session.c:1250 Session 9
(h323/231543514280633) Locked, Waiting on external entities
2010-12-14 12:38:57.689292 [NOTICE] switch_core_session.c:1268 Session 9
(h323/231543514280633) Ended
2010-12-14 12:38:57.689292 [NOTICE] switch_core_session.c:1270 Close
Channel h323/231543514280633 [CS_DESTROY]
2010-12-14 12:38:57.689292 [DEBUG] switch_core_state_machine.c:449
(h323/231543514280633) Callstate Change HANGUP -> DOWN
2010-12-14 12:38:57.689292 [DEBUG] switch_core_state_machine.c:452
(h323/231543514280633) Running State Change CS_DESTROY
2010-12-14 12:38:57.689292 [DEBUG] switch_core_state_machine.c:462
(h323/231543514280633) State DESTROY
2010-12-14 12:38:57.689292 [DEBUG] mod_h323.cpp:2307 ======>on_destroy
2010-12-14 12:38:57.689292 [DEBUG] switch_core_state_machine.c:60
h323/231543514280633 Standard DESTROY
2010-12-14 12:38:57.689292 [DEBUG] switch_core_state_machine.c:462
(h323/231543514280633) State DESTROY going to sleep
2010-12-14 12:38:57.686742 [DEBUG] switch_core_state_machine.c:617
(sofia/external/21543514280633 at 200.35.145.149) State REPORTING
2010-12-14 12:38:57.689292 [DEBUG] h323caps.cxx:3676 FindCapability:
"UserInput/RFC2833"
2010-12-14 12:38:57.689292 [DEBUG] h323caps.cxx:3685 Found capability:
UserInput/RFC2833 <10>
2010-12-14 12:38:57.689292 [DEBUG] h323.cxx:4228 User Input RFC2833
payload type set to [pt=101]
2010-12-14 12:38:57.689292 [DEBUG] mod_h323.cpp:1109 ======>END
H323Connection::OnReceivedCapabilitySet [0x9a7d140]
2010-12-14 12:38:57.689292 [DEBUG] mod_h323.cpp:1112 ----> Capabilities
= G.711-ALaw-64k
2010-12-14 12:38:57.689292 [DEBUG] mod_h323.cpp:1112 ----> Capabilities
= GSM-06.10{sw}
2010-12-14 12:38:57.689292 [DEBUG] mod_h323.cpp:1112 ----> Capabilities
= UserInput/hookflash
2010-12-14 12:38:57.689292 [DEBUG] mod_h323.cpp:1112 ----> Capabilities
= UserInput/basicString
2010-12-14 12:38:57.689292 [DEBUG] mod_h323.cpp:1112 ----> Capabilities
= UserInput/dtmf
2010-12-14 12:38:57.689292 [DEBUG] mod_h323.cpp:1112 ----> Capabilities
= UserInput/RFC2833
2010-12-14 12:38:57.689292 [DEBUG] h323caps.cxx:4009 Found capability:
G.711-ALaw-64k <1>
2010-12-14 12:38:57.689292 [DEBUG] mod_h323.cpp:1120 ----> Capabilities
not NULL 
2010-12-14 12:38:57.689292 [DEBUG] h323pdu.cxx:618 Sending PDU
[(noaddr)/(noaddr)] :
  response terminalCapabilitySetAck {
    sequenceNumber = 1
  }
2010-12-14 12:38:57.692254 [DEBUG] h323pdu.cxx:618 Sending PDU [ip
$72.51.47.100:1720/ip$200.117.192.17:20570] :
  {
    q931pdu = {
      protocolDiscriminator = 8
      callReference = 4293
      from = destination
      messageType = Facility
      IE: Facility = {

  
(..........!..
      }
    }
    h225pdu = {
      h323_uu_pdu = {
        h323_message_body = empty <<null>>
        h245Tunneling = true
        h245Control =
 1 entries {
          [0]= 3 octets {
            21 80 01                                           !..
          }
        }
      }
    }
  }
2010-12-14 12:38:57.692254 [DEBUG] h323.cxx:4369
InternalEstablishedConnectionCheck: connectionState=AwaitingLocalAnswer
fastStartState=FastStartAcknowledged
2010-12-14 12:38:57.692254 [DEBUG] h323pdu.cxx:618 Receiving PDU
[(noaddr)/(noaddr)] :
  request masterSlaveDetermination {
    terminalType = 50
    statusDeterminationNumber = 1982505
  }
2010-12-14 12:38:57.692254 [DEBUG] h323neg.cxx:395 Received
MasterSlaveDetermination: state=Idle
2010-12-14 12:38:57.692254 [DEBUG] h323neg.cxx:426
MasterSlaveDetermination: local is master
2010-12-14 12:38:57.692254 [DEBUG] h323pdu.cxx:618 Sending PDU
[(noaddr)/(noaddr)] :
  response masterSlaveDeterminationAck {
    decision = slave <<null>>
  }
2010-12-14 12:38:57.695276 [DEBUG] switch_core_state_machine.c:53
sofia/external/21543514280633 at 200.35.145.149 Standard REPORTING, cause:
NORMAL_UNSPECIFIED
2010-12-14 12:38:57.695276 [DEBUG] switch_core_state_machine.c:617
(sofia/external/21543514280633 at 200.35.145.149) State REPORTING going to
sleep
2010-12-14 12:38:57.695276 [DEBUG] h323pdu.cxx:618 Sending PDU [ip
$72.51.47.100:1720/ip$200.117.192.17:20570] :
  {
    q931pdu = {
      protocolDiscriminator = 8
      callReference = 4293
      from = destination
      messageType = Facility
      IE: Facility = {

  
(.......... .
      }
    }
    h225pdu = {
      h323_uu_pdu = {
        h323_message_body = empty <<null>>
        h245Tunneling = true
        h245Control = 
1 entries {
          [0]= 2 octets {
            20 a0                                               .
          }
        }
      }
    }
  }
2010-12-14 12:38:57.695276 [DEBUG] switch_core_state_machine.c:345
(sofia/external/21543514280633 at 200.35.145.149) State Change CS_REPORTING
-> CS_DESTROY
2010-12-14 12:38:57.695276 [DEBUG] switch_core_session.c:1083 Send
signal sofia/external/21543514280633 at 200.35.145.149 [BREAK]
2010-12-14 12:38:57.695276 [DEBUG] switch_core_session.c:1250 Session 10
(sofia/external/21543514280633 at 200.35.145.149) Locked, Waiting on
external entities
2010-12-14 12:38:57.695276 [NOTICE] switch_core_session.c:1268 Session
10 (sofia/external/21543514280633 at 200.35.145.149) Ended
2010-12-14 12:38:57.695276 [NOTICE] switch_core_session.c:1270 Close
Channel sofia/external/21543514280633 at 200.35.145.149 [CS_DESTROY]
2010-12-14 12:38:57.695276 [DEBUG] switch_core_state_machine.c:449
(sofia/external/21543514280633 at 200.35.145.149) Callstate Change HANGUP
-> DOWN
2010-12-14 12:38:57.695276 [DEBUG] switch_core_state_machine.c:452
(sofia/external/21543514280633 at 200.35.145.149) Running State Change
CS_DESTROY
2010-12-14 12:38:57.695276 [DEBUG] switch_core_state_machine.c:462
(sofia/external/21543514280633 at 200.35.145.149) State DESTROY
2010-12-14 12:38:57.695276 [DEBUG] mod_sofia.c:364
sofia/external/21543514280633 at 200.35.145.149 SOFIA DESTROY
2010-12-14 12:38:57.695276 [DEBUG] switch_core_state_machine.c:60
sofia/external/21543514280633 at 200.35.145.149 Standard DESTROY
2010-12-14 12:38:57.695276 [DEBUG] switch_core_state_machine.c:462
(sofia/external/21543514280633 at 200.35.145.149) State DESTROY going to
sleep
2010-12-14 12:38:57.695276 [DEBUG] h323.cxx:4369
InternalEstablishedConnectionCheck: connectionState=AwaitingLocalAnswer
fastStartState=FastStartAcknowledged
2010-12-14 12:38:57.695276 [DEBUG] h323.cxx:4369
InternalEstablishedConnectionCheck: connectionState=AwaitingLocalAnswer
fastStartState=FastStartAcknowledged
2010-12-14 12:38:57.695276 [DEBUG] h323.cxx:1353 Reading PDUs:
callRef=4293
2010-12-14 12:38:57.695276 [DEBUG] h323neg.cxx:380 Stopping
MasterSlaveDetermination: state=Incoming
2010-12-14 12:38:57.695276 [DEBUG] h323neg.cxx:612 Stopping
TerminalCapabilitySet: state=InProgress
2010-12-14 12:38:57.695276 [DEBUG] channels.cxx:743 Cleaning up T-101
2010-12-14 12:38:57.695276 [DEBUG] h323ep.cxx:3213 Stopped sending
logical channel: G.711-ALaw-64k <1>
2010-12-14 12:38:57.695276 [DEBUG] channels.cxx:771 Cleaned up T-101
2010-12-14 12:38:57.695276 [DEBUG] channels.cxx:743 Cleaning up R-101
2010-12-14 12:38:57.695276 [DEBUG] h323ep.cxx:3213 Stopped receiving
logical channel: G.711-ALaw-64k <1>
2010-12-14 12:38:57.695276 [DEBUG] channels.cxx:771 Cleaned up R-101
2010-12-14 12:38:57.695276 [DEBUG] mod_h323.cpp:1903
======>FSH323_ExternalRTPChannel::~FSH323_ExternalRTPChannel
IsTransmitter [0xb773dc88]
2010-12-14 12:38:57.695276 [DEBUG] mod_h323.cpp:1906
------------->switch_core_session_unlock_codec_read [0x9bfe898]
2010-12-14 12:38:57.695276 [DEBUG] mod_h323.cpp:1908
------------->switch_core_session_unlock_codec_write [0x9bfe898]
2010-12-14 12:38:57.695276 [DEBUG] h323.cxx:5105 Bandwidth request:
-64.0kb/s, available: 9872.0kb/s
2010-12-14 12:38:57.695276 [DEBUG] mod_h323.cpp:1903
======>FSH323_ExternalRTPChannel::~FSH323_ExternalRTPChannel  IsReceiver
[0xb773b9a0]
2010-12-14 12:38:57.695276 [DEBUG] mod_h323.cpp:1906
------------->switch_core_session_unlock_codec_read [0x9bfe898]
2010-12-14 12:38:57.695276 [DEBUG] mod_h323.cpp:1908
------------->switch_core_session_unlock_codec_write [0x9bfe898]
2010-12-14 12:38:57.695276 [DEBUG] h323.cxx:5105 Bandwidth request:
-64.0kb/s, available: 9936.0kb/s
2010-12-14 12:38:57.698292 [DEBUG] transports.cxx:1295
H323Transport::Close
2010-12-14 12:38:57.698292 [INFO] h323pdu.cxx:1576 Read error (4):
Interrupted system call
2010-12-14 12:38:57.698292 [DEBUG] h323ep.cxx:2802 Clearing connection
ip$200.117.192.17:20570/4293 reason=EndedByTransportFail
2010-12-14 12:38:57.698292 [DEBUG] transports.cxx:1295
H323Transport::Close
2010-12-14 12:38:57.698292 [DEBUG] h323.cxx:1367 Signal channel closed.
2010-12-14 12:38:57.701253 [DEBUG] tlibthrd.cxx:1020 Could not parse
thread stat file /proc/23814/task/23999/stat
2010-12-14 12:38:57.966214 [DEBUG] transports.cxx:1432
H323Transport::CleanUpOnTermination for H225 Answer:b73c5b90
2010-12-14 12:38:57.966214 [INFO] h323.cxx:1273 Connection ip
$200.117.192.17:20570/4293 terminated.
2010-12-14 12:38:57.966214 [DEBUG] mod_h323.cpp:737
======>FSH323Connection::~FSH323Connection  [0x9a7d140]
2010-12-14 12:38:57.966214 [DEBUG] h323.cxx:1089 Connection ip
$200.117.192.17:20570/4293 deleted.
2010-12-14 12:38:57.966214 [DEBUG] h323ep.cxx:2861 Cleaning up
connections





From steveayre at gmail.com  Tue Dec 14 16:14:16 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 14 Dec 2010 13:14:16 +0000
Subject: [Freeswitch-users] FS crash / Monitoring Script to Restart FS
In-Reply-To: <AANLkTi=O8488u-Y+HOrpL+yJxakf6EaZ4wwT_mRcnVOa@mail.gmail.com>
References: <4D075EE9.1030901@sns.eu>
	<AANLkTi=O8488u-Y+HOrpL+yJxakf6EaZ4wwT_mRcnVOa@mail.gmail.com>
Message-ID: <4E127671-6A15-4A2C-9273-4D6659ABB02C@gmail.com>

Also see freeswitch.monitrc in git (I forget which directory) which also has an example config.

Steve on iPhone

On 14 Dec 2010, at 12:38, Vitalii Colosov <vetali100 at gmail.com> wrote:

> Hi Jan,
> Regarding the monitoring and restarting script - you can use monit:
> 
> Take one of the pre-compiled binaries from
> http://mmonit.com/monit/download/
> 
> Unpack it on your system, create file:
> /etc/monitrc
> 
> With the following content:
> 
>       set daemon 60
>       set logfile syslog facility log_daemon
> 
>       check process freeswitch with pidfile /usr/local/freeswitch/run/freeswitch.pid
>           start program = "/usr/local/freeswitch/bin/freeswitch -nc"
>           stop program = "/usr/local/freeswitch/bin/freeswitch -stop"
> 
> 
> Start monit 
> /your/dir/monit/bin/monit
> and enjoy :)
> 
> Also you can monitor the status from web, if you will add the following lines to the monitrc:
> set httpd port 1234
>          use address your.domain.com
>      allow your.ip.add.ress
>      allow yourUser:yourPassword
> 
> Don't forget to add start monit to the system auto start list.
> Refer to your linux doc how to do it.
> 
> Regards,
> Vitalie
> 
> 
> 
> 
> 
> 2010/12/14 Jan Riedinger <riedinger at sns.eu>
> We are running FreeSWITCH Version 1.0.head (git-51cc00a 2010-10-06
> 11-07-41 -0500).
> 
> On the 7th December FS crashed. The last lines of the log file with full
> debug output are:
> 22c143da-728d-4477-aa06-c8383fa604f2 2010-12-07 22:52:55.339619 [DEBUG]
> sofia_glue.c:4250 Audio Codec Compare
> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
> 22c143da-728d-4477-aa06-c8383fa604f2 2010-12-07 22:52:55.339619 [DEBUG]
> sofia_glue.c:2554 Changing Codec from G729 at 30ms to G729 at 20ms
> 2010-12-07 22:52:55.339619 [INFO] mod_com_g729.c:76 ENCODER DESTROYX -
> 0x2aaacc1530b0 (nil)
> 2010-12-07 22:52:55.339619 [INFO] mod_com_g729.c:77 DECODER DESTROYX -
> 0x2aaacc1530b0 (nil)
> 2010-12-07 22:52:55.339619 [INFO] mod_com_g729.c:76 ENCODER DESTROYX -
> 0x2aaacc153130 (nil)
> 
> There were some dozen concurrent calls on the system at this moment,
> thus it is possible that the crash was caused by something else than
> mod_com_g729.c.
> 
> Furthermore, a 209 MB core dump file was written. Please let me know, if
> anyone wants to examine the reason for the crash in detail.
> 
> Does anyone have written a script, which monitors and restart FS in such
> cases?
> 
> Thank you in advance
>          Jan
> 
> 
> 
> --
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
> 
> AG Charlottenburg - HRB 71973
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From melkybes at mail.ru  Tue Dec 14 16:29:56 2010
From: melkybes at mail.ru (=?koi8-r?Q?=ED=C9=C8=C1=C9=CC_=F3=C1=CC=D4=C1=CE=CF=D7?=)
Date: Tue, 14 Dec 2010 16:29:56 +0300
Subject: [Freeswitch-users] import vars from other context
In-Reply-To: <1292332435.2134.10.camel@gustavo-laptop>
References: <1292332435.2134.10.camel@gustavo-laptop>
Message-ID: <E1PSUx2-0002OC-00.melkybes-mail-ru@f253.mail.ru>


Hello

How I can  import variables from other context in FS ?

? ?????????
?????? ????????





From steveayre at gmail.com  Tue Dec 14 17:01:04 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 14 Dec 2010 14:01:04 +0000
Subject: [Freeswitch-users] import vars from other context
In-Reply-To: <E1PSUx2-0002OC-00.melkybes-mail-ru@f253.mail.ru>
References: <1292332435.2134.10.camel@gustavo-laptop>
	<E1PSUx2-0002OC-00.melkybes-mail-ru@f253.mail.ru>
Message-ID: <AANLkTinqx2uyi_GEDmU127TiAmRf=KWS0QQGtXQnOrc4@mail.gmail.com>

I'm not sure what you mean?

If you mean a dialplan context, then a variable is set on a channel
(call). It will still be set on that channel if it is transferred to
another dialplan context.

If you mean during a bridged call exporting a A-Leg (incoming channel)
variable onto the B-Leg (outgoing channel) look at the export app:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_export

Regards
-Steve


2010/12/14 ?????? ???????? <melkybes at mail.ru>:
>
> Hello
>
> How I can ?import variables from other context in FS ?
>
> ? ?????????
> ?????? ????????
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From math.parent at gmail.com  Tue Dec 14 17:56:17 2010
From: math.parent at gmail.com (Mathieu Parent)
Date: Tue, 14 Dec 2010 15:56:17 +0100
Subject: [Freeswitch-users] tts_commandline question
In-Reply-To: <4D06B88A.10206@gmail.com>
References: <4D06B88A.10206@gmail.com>
Message-ID: <AANLkTi=jRWPRu4jA_Ej3c0jU_Go7bYCGKdh1qKoDH66z@mail.gmail.com>

2010/12/14 Phone <phone.bytes at gmail.com>:
> Hi Mathieu,
>
> I have been trying to use mod_cepstral. ?I had trouble with it hanging and
> needing to restart freeswitch after funning only a few hours.
>
> MC suggested that I try tts_commandline to see if that worked better. ?I did
> and found I did not have any issues doing cepstral this way.
>
> However, my need is to be able to feed a text file to Swift instead of
> sending the text on the command line. ?I have tried using a -f switch as
> Swift allows, but it seems that tts_commandline does not support this
> option.

mod_tts_commandline only work with text as a ARG. ${text} is replaced
by the actual string, quoted (for security reasons).

>
> Am I missing something here or is there another way of doing this?

You probably want to use cepstral as a system command
(http://wiki.freeswitch.org/wiki/Mod_commands#system) and the playback
(http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_playback).

> Following is a piece of a trace from the log.
>
> It seems to almost build the swift command correctly. ?However, the -f
> option is inside the single quotes, instead of outside of them like the -o
> (output file) is.
>
> I don't think it is finding the .txt file, as the log later shows that it is
> trying to speak the filename including the -f option, instead of the
> contents of the file.
>
> Thanks
>
> EXECUTE FreeTDM/1:1/7878030 speak(tts_commandline|Callie|-f/tmp/test.txt)
> 2010-12-11 00:26:58.683522 [DEBUG] switch_ivr_play_say.c:2237 OPEN TTS
> tts_commandline
> 2010-12-11 00:26:58.683522 [DEBUG] switch_ivr_play_say.c:2246 Raw Codec
> Activated
> 2010-12-11 00:26:58.683522 [DEBUG] mod_tts_commandline.c:147 Executing:
> swift -p audio/sampling-rate=8000 -n 'Callie' '-f/tmp/test.txt' ?-o
> '/tmp/bd5bb8db-ffc2-4830-a971-24a1454c725c.tmp.wav'
> 2010-12-11 00:26:58.696525 [ERR] mod_sndfile.c:194 Error Opening File
> [/tmp/bd5bb8db-ffc2-4830-a971-24a1454c725c.tmp.wav] [System error : No such
> file or directory.]
> 2010-12-11 00:26:58.696525 [ERR] mod_tts_commandline.c:157 Failed to open
> file: /tmp/bd5bb8db-ffc2-4830-a971-24a1454c725c.tmp.wav
> 2010-12-11 00:26:58.696525 [DEBUG] switch_ivr_play_say.c:1935 Speaking text:
> -f/tmp/test.txt
> 2010-12-11 00:26:59.097509 [DEBUG] switch_ivr_play_say.c:2127 done speaking
> text
>
>
>
>



-- 
Mathieu


From anthony.minessale at gmail.com  Tue Dec 14 18:40:21 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 14 Dec 2010 09:40:21 -0600
Subject: [Freeswitch-users] FS crash / Monitoring Script to Restart FS
In-Reply-To: <4D075EE9.1030901@sns.eu>
References: <4D075EE9.1030901@sns.eu>
Message-ID: <AANLkTikw2E2PbdeEJ89FDEzZnj+6CbPsuDdpYgNxVfGu@mail.gmail.com>

I think this issue is fixed.
issue "make current" to update to latest


On Tue, Dec 14, 2010 at 6:11 AM, Jan Riedinger <riedinger at sns.eu> wrote:
> We are running FreeSWITCH Version 1.0.head (git-51cc00a 2010-10-06
> 11-07-41 -0500).
>
> On the 7th December FS crashed. The last lines of the log file with full
> debug output are:
> 22c143da-728d-4477-aa06-c8383fa604f2 2010-12-07 22:52:55.339619 [DEBUG]
> sofia_glue.c:4250 Audio Codec Compare
> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
> 22c143da-728d-4477-aa06-c8383fa604f2 2010-12-07 22:52:55.339619 [DEBUG]
> sofia_glue.c:2554 Changing Codec from G729 at 30ms to G729 at 20ms
> 2010-12-07 22:52:55.339619 [INFO] mod_com_g729.c:76 ENCODER DESTROYX -
> 0x2aaacc1530b0 (nil)
> 2010-12-07 22:52:55.339619 [INFO] mod_com_g729.c:77 DECODER DESTROYX -
> 0x2aaacc1530b0 (nil)
> 2010-12-07 22:52:55.339619 [INFO] mod_com_g729.c:76 ENCODER DESTROYX -
> 0x2aaacc153130 (nil)
>
> There were some dozen concurrent calls on the system at this moment,
> thus it is possible that the crash was caused by something else than
> mod_com_g729.c.
>
> Furthermore, a 209 MB core dump file was written. Please let me know, if
> anyone wants to examine the reason for the crash in detail.
>
> Does anyone have written a script, which monitors and restart FS in such
> cases?
>
> Thank you in advance
> ? ? ? ? ?Jan
>
>
>
> --
> Jan Riedinger ? ? ? ? ? ? ? ? ? ? ? ? ? Phone : ?+49-30-39 73 19 66
> Dipl.-Inf. | Managing Director ? ? ? ? ?Fax ? : ?+49-30-39 73 19 64
> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? E-Mail: ?riedinger at sns.eu
> SNS Consult GmbH ? ? ? ? ? ? ? ? ? ? ? ?ICQ ? : ?163-237-041
> S?dwestkorso 49a ? ? ? ? ? ? ? ? ? ? ? ?MSN ? : ?jan at sns-consult.de
> 14197 Berlin GERMANY ? ? ? ? ? ? ? ? ? ?Skype : ?Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From steveayre at gmail.com  Tue Dec 14 18:56:52 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 14 Dec 2010 15:56:52 +0000
Subject: [Freeswitch-users] FS crash / Monitoring Script to Restart FS
In-Reply-To: <AANLkTikw2E2PbdeEJ89FDEzZnj+6CbPsuDdpYgNxVfGu@mail.gmail.com>
References: <4D075EE9.1030901@sns.eu>
	<AANLkTikw2E2PbdeEJ89FDEzZnj+6CbPsuDdpYgNxVfGu@mail.gmail.com>
Message-ID: <AANLkTi=+ugjfWdK16MYfCYnEdndwy7ncVK9B1DCum-ND@mail.gmail.com>

Jan,

>From your version number your current version predates the latest
version of mod_com_g729, you'll probably also need to upgrade to
fsg729-167-installer from http://files.freeswitch.org/g729/ to get the
module to load.

-Steve



On 14 December 2010 15:40, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I think this issue is fixed.
> issue "make current" to update to latest
>
>
> On Tue, Dec 14, 2010 at 6:11 AM, Jan Riedinger <riedinger at sns.eu> wrote:
>> We are running FreeSWITCH Version 1.0.head (git-51cc00a 2010-10-06
>> 11-07-41 -0500).
>>
>> On the 7th December FS crashed. The last lines of the log file with full
>> debug output are:
>> 22c143da-728d-4477-aa06-c8383fa604f2 2010-12-07 22:52:55.339619 [DEBUG]
>> sofia_glue.c:4250 Audio Codec Compare
>> [G729:18:8000:20:8000]/[G729:18:8000:20:8000]
>> 22c143da-728d-4477-aa06-c8383fa604f2 2010-12-07 22:52:55.339619 [DEBUG]
>> sofia_glue.c:2554 Changing Codec from G729 at 30ms to G729 at 20ms
>> 2010-12-07 22:52:55.339619 [INFO] mod_com_g729.c:76 ENCODER DESTROYX -
>> 0x2aaacc1530b0 (nil)
>> 2010-12-07 22:52:55.339619 [INFO] mod_com_g729.c:77 DECODER DESTROYX -
>> 0x2aaacc1530b0 (nil)
>> 2010-12-07 22:52:55.339619 [INFO] mod_com_g729.c:76 ENCODER DESTROYX -
>> 0x2aaacc153130 (nil)
>>
>> There were some dozen concurrent calls on the system at this moment,
>> thus it is possible that the crash was caused by something else than
>> mod_com_g729.c.
>>
>> Furthermore, a 209 MB core dump file was written. Please let me know, if
>> anyone wants to examine the reason for the crash in detail.
>>
>> Does anyone have written a script, which monitors and restart FS in such
>> cases?
>>
>> Thank you in advance
>> ? ? ? ? ?Jan
>>
>>
>>
>> --
>> Jan Riedinger ? ? ? ? ? ? ? ? ? ? ? ? ? Phone : ?+49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director ? ? ? ? ?Fax ? : ?+49-30-39 73 19 64
>> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? E-Mail: ?riedinger at sns.eu
>> SNS Consult GmbH ? ? ? ? ? ? ? ? ? ? ? ?ICQ ? : ?163-237-041
>> S?dwestkorso 49a ? ? ? ? ? ? ? ? ? ? ? ?MSN ? : ?jan at sns-consult.de
>> 14197 Berlin GERMANY ? ? ? ? ? ? ? ? ? ?Skype : ?Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From bcxml at hotmail.com  Tue Dec 14 22:20:40 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Tue, 14 Dec 2010 14:20:40 -0500
Subject: [Freeswitch-users] Custom Channel Variables via C#
In-Reply-To: <9C89CC76-01FC-4F8D-B00B-89EE3A74EB05@avgs.ca>
References: <SNT135-w336CAB94DFC78D328AE6E0B7260@phx.gbl>, ,
	<0E462B4C-C436-462E-9DA7-9EC1241FB380@avgs.ca>,
	<SNT135-w4BF48F8FEB9054D804759B7270@phx.gbl>,
	<9C89CC76-01FC-4F8D-B00B-89EE3A74EB05@avgs.ca>
Message-ID: <SNT135-w6446098BFC470AA8CA2E3B7130@phx.gbl>


Thanks Mathieu 
 
I have up dated my code as per your instructions.
 
FreeSwitch answers the call and then does a bridge to my Speech Server application. If there are 10 simultaneous calls going on there would be 20 channels. When I send 'api show channels', I get info back for all 20 channels. How do I go about deciding which one contains the correct uuid for the session that I am trying to set a custom channel variable for ?
 
Here is my code
 
     private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
    {
      try
      {
        TcpClient newClient = new TcpClient();
        // Connect to the port
        newClient.Connect(_host, _port);
        NetworkStream tcpStream = newClient.GetStream();
        byte[] receivedBytes = new byte[newClient.ReceiveBufferSize];
        int bytesRead = tcpStream.Read(receivedBytes, 0, newClient.ReceiveBufferSize);
        string returnData = Encoding.ASCII.GetString(receivedBytes);
        // Authenticate
        byte[] sendBytes = Encoding.ASCII.GetBytes("auth ClueCon\n\n");
        tcpStream.Write(sendBytes, 0, sendBytes.Length);
        receivedBytes = new byte[newClient.ReceiveBufferSize];
        bytesRead = tcpStream.Read(receivedBytes, 0, newClient.ReceiveBufferSize);
        returnData = Encoding.ASCII.GetString(receivedBytes);
        // Get channel info
        sendBytes = Encoding.ASCII.GetBytes("api show channels\n\n");
        tcpStream.Write(sendBytes, 0, sendBytes.Length);
        receivedBytes = new byte[newClient.ReceiveBufferSize];
        bytesRead = tcpStream.Read(receivedBytes, 0, newClient.ReceiveBufferSize);
        returnData = Encoding.ASCII.GetString(receivedBytes);

        // How do I get the proper UUID from the info provided by 'api show channels'
        
        string uuid = '????????'
        
        
        
        // Set the Custom Channel Variable
        sendBytes = Encoding.ASCII.GetBytes("api uuid_setvar " + uuid + " qqOneReach_ApplicationId BRIAN" + "\n\n");
                                                                          
        tcpStream.Write(sendBytes, 0, sendBytes.Length);
        receivedBytes = new byte[newClient.ReceiveBufferSize];
        bytesRead = tcpStream.Read(receivedBytes, 0, newClient.ReceiveBufferSize);
        returnData = Encoding.ASCII.GetString(receivedBytes);
        // Clean everything up
        
        tcpStream.Flush();
        tcpStream.Close();
        newClient.Close();
      }
      catch (Exception ex)
      {
        throw ex;
      }
    }

 
Thanks
 
Brian Campbell
 


From: mrene_lists at avgs.ca
Date: Thu, 2 Dec 2010 12:05:42 -0500
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Custom Channel Variables via C#


You need to authenticate first.



Trying 127.0.0.1...
Connected to localhost
Escape character is '^]'.
Content-Type: auth/request


auth ClueCon


Content-Type: command/reply
Reply-Text: +OK accepted


api show channels


Content-Type: api/response
Content-Length: 748


uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid
aa6e6bee-1c3b-4493-b824-765c5a15f02c,outbound,2010-12-02 12:03:58,1291309438,loopback/park-a,CS_CONSUME_MEDIA,,0000000000,,park,,,inline,inline,L16,8000,128000,L16,8000,128000,,mrene.local,,,RINGING,,,,c9781e01-667e-446b-93e8-a081f0287d15
78ff4bb6-573e-4a60-a920-bfdb67f3ed6b,inbound,2010-12-02 12:03:58,1291309438,loopback/park-b,CS_EXECUTE,,0000000000,,park,park,,inline,inline,L16,8000,128000,L16,8000,128000,,mrene.local,,,RINGING,,,,


2 total.


Then you can use the uuid_* commands.





Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca






On 2010-12-02, at 10:03 AM, Brian Campbell wrote:

Thanks, the link was very helpful
 
I have taken a look at the link that you provided
 
I assume now that I can use uuid_setvar to accomplish what I want
 
Assuming that the uuid is 53d37581-1f90-44bf-860a-addbc8430e3a, it seems that my code would need to change to
 
 
private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
{
  try
  {
    TcpClient newClient = new TcpClient();
 
    newClient.Connect("127.0.0.1", 8021);
 
    NetworkStream tcpStream = newClient.GetStream();
 
    byte[] sendBytes = Encoding.ASCII.GetBytes("api uuid_setvar 53d37581-1f90-44bf-860a-addbc8430e3a myChannelVariable 12345");
 
    tcpStream.Write(sendBytes, 0, sendBytes.Length);
 
    tcpStream.Close();
 
    newClient.Close();
  }
  catch(Exception ex)
  {
    throw ex;
  }
}

My question now becomes, how do I go about finding out the uuid for the session ? Is that obtainable via mod_event_socket or is there another way ?
 
Thanks
 
Brian 


From: mrene_lists at avgs.ca
Date: Wed, 1 Dec 2010 16:53:53 -0500
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Custom Channel Variables via C#


Hi,


You can't send XML dialplan actions on the socket and expect FreeSWITCH to understand it, you must follow the event socket protocol.


See http://wiki.freeswitch.org/wiki/Event_socket for more information.




Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca






On 2010-12-01, at 4:46 PM, Brian Campbell wrote:

 
I can set a custom channel variable in my incomming dial plan like this...
 
<action application='set' data='MyChannelVariable=12345' />
 
And it shows up fine in the CDR
 
I am now attempting to use mod_event_socket to set a custom channel variable and have it appear in the CDR
 
Here is the C# code so far, the code is part of a Microsoft Speech Server application that is answering the call
 
private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
{
  try
  {
    TcpClient newClient = new TcpClient();
 
    newClient.Connect("127.0.0.1", 8021);
 
    NetworkStream tcpStream = newClient.GetStream();
 
    byte[] sendBytes = Encoding.ASCII.GetBytes("<action application='set' data='MyChannelVariable=12345' />");
 
    tcpStream.Write(sendBytes, 0, sendBytes.Length);
 
    tcpStream.Close();
 
    newClient.Close();
  }
  catch(Exception ex)
  {
    throw ex;
  }
}
 
After the call is answered, the C# code seems to run fine, but I dont see the custom channel variable in the resulting CDR
 
I figure I am not setting it correctly
 
Can anyone advise on what I am doing wrong ?
 
Thanks
 
 
Brian
 
 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


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


_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 		 	   		  
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From shamun.toha at gmail.com  Tue Dec 14 22:45:02 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Tue, 14 Dec 2010 20:45:02 +0100
Subject: [Freeswitch-users] Pre-paid impossible with freeSwitch ?
Message-ID: <AANLkTim3mXSPXcO3AjOt9-C6zj82MmONH+2Anc3n_S5k@mail.gmail.com>

Hello,

I am trying to make a Pre-Paid or Advanced billing method for test. But i
applied Javascript with FS.

- In my test i am running calls for 3 or 4 hours
- With a billing per minute of 10 US Dollar

Now those calls balance is -200 or -300, how can i tell FS via javascript
please kindly stop it because its crossing the limit.

Is there any way to manage it ?

Thanks & Regards
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From msc at freeswitch.org  Tue Dec 14 23:02:03 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 14 Dec 2010 12:02:03 -0800
Subject: [Freeswitch-users] Pre-paid impossible with freeSwitch ?
In-Reply-To: <AANLkTim3mXSPXcO3AjOt9-C6zj82MmONH+2Anc3n_S5k@mail.gmail.com>
References: <AANLkTim3mXSPXcO3AjOt9-C6zj82MmONH+2Anc3n_S5k@mail.gmail.com>
Message-ID: <AANLkTikKHg5NYCcjJ+vbnsEZYVKKLCMjZw4U0eRm3kzK@mail.gmail.com>

Did mod_nibblebill not meet your needs for some reason?
-MC

On Tue, Dec 14, 2010 at 11:45 AM, Shamun toha md <shamun.toha at gmail.com>wrote:

> Hello,
>
> I am trying to make a Pre-Paid or Advanced billing method for test. But i
> applied Javascript with FS.
>
> - In my test i am running calls for 3 or 4 hours
> - With a billing per minute of 10 US Dollar
>
> Now those calls balance is -200 or -300, how can i tell FS via javascript
> please kindly stop it because its crossing the limit.
>
> Is there any way to manage it ?
>
> Thanks & Regards
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Avi at aMarcus.com  Tue Dec 14 23:10:21 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Tue, 14 Dec 2010 22:10:21 +0200
Subject: [Freeswitch-users] Pre-paid impossible with freeSwitch ?
In-Reply-To: <AANLkTikKHg5NYCcjJ+vbnsEZYVKKLCMjZw4U0eRm3kzK@mail.gmail.com>
References: <AANLkTim3mXSPXcO3AjOt9-C6zj82MmONH+2Anc3n_S5k@mail.gmail.com>
	<AANLkTikKHg5NYCcjJ+vbnsEZYVKKLCMjZw4U0eRm3kzK@mail.gmail.com>
Message-ID: <AANLkTimBGyzFHON5xUNTUthWwPX=N7umNH0S1b-34oHM@mail.gmail.com>

Mod_nibblebill has this functionality. Give it a try.

There's also schedule hangup, but if you have more than one channel open on
the same account that will skew the numbers.
The alternative is some sort of ESL application to handle that the
multi-channel calculations.
-Avi

On Tue, Dec 14, 2010 at 10:02 PM, Michael Collins <msc at freeswitch.org>wrote:

> Did mod_nibblebill not meet your needs for some reason?
> -MC
>
> On Tue, Dec 14, 2010 at 11:45 AM, Shamun toha md <shamun.toha at gmail.com>wrote:
>
>> Hello,
>>
>> I am trying to make a Pre-Paid or Advanced billing method for test. But i
>> applied Javascript with FS.
>>
>> - In my test i am running calls for 3 or 4 hours
>> - With a billing per minute of 10 US Dollar
>>
>> Now those calls balance is -200 or -300, how can i tell FS via javascript
>> please kindly stop it because its crossing the limit.
>>
>> Is there any way to manage it ?
>>
>> Thanks & Regards
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Tue Dec 14 23:20:12 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 14 Dec 2010 15:20:12 -0500
Subject: [Freeswitch-users] Custom Channel Variables via C#
In-Reply-To: <SNT135-w6446098BFC470AA8CA2E3B7130@phx.gbl>
References: <SNT135-w336CAB94DFC78D328AE6E0B7260@phx.gbl>, ,
	<0E462B4C-C436-462E-9DA7-9EC1241FB380@avgs.ca>,
	<SNT135-w4BF48F8FEB9054D804759B7270@phx.gbl>,
	<9C89CC76-01FC-4F8D-B00B-89EE3A74EB05@avgs.ca>
	<SNT135-w6446098BFC470AA8CA2E3B7130@phx.gbl>
Message-ID: <904C3740-32D5-4B78-B444-6FA21E049BDE@avgs.ca>

Hi,

You have to somehow pass that information to your script. If you wish to control the channel, you could use the "socket" application which establishes an outbound tcp connection to your application.

Otherwise, you could always send an event from the dialplan containing the call's uuid. The event socket connection can control of any calls on the system, how exactly are you bridging the call to your speech server?

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-12-14, at 2:20 PM, Brian Campbell wrote:

> Thanks Mathieu 
>  
> I have up dated my code as per your instructions.
>  
> FreeSwitch answers the call and then does a bridge to my Speech Server application. If there are 10 simultaneous calls going on there would be 20 channels. When I send 'api show channels', I get info back for all 20 channels. How do I go about deciding which one contains the correct uuid for the session that I am trying to set a custom channel variable for ?
>  
> Here is my code
>  
>      private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
>     {
>       try
>       {
>         TcpClient newClient = new TcpClient();
>         // Connect to the port
>         newClient.Connect(_host, _port);
>         NetworkStream tcpStream = newClient.GetStream();
>         byte[] receivedBytes = new byte[newClient.ReceiveBufferSize];
>         int bytesRead = tcpStream.Read(receivedBytes, 0, newClient.ReceiveBufferSize);
>         string returnData = Encoding.ASCII.GetString(receivedBytes);
>         // Authenticate
>         byte[] sendBytes = Encoding.ASCII.GetBytes("auth ClueCon\n\n");
>         tcpStream.Write(sendBytes, 0, sendBytes.Length);
>         receivedBytes = new byte[newClient.ReceiveBufferSize];
>         bytesRead = tcpStream.Read(receivedBytes, 0, newClient.ReceiveBufferSize);
>         returnData = Encoding.ASCII.GetString(receivedBytes);
>         // Get channel info
>         sendBytes = Encoding.ASCII.GetBytes("api show channels\n\n");
>         tcpStream.Write(sendBytes, 0, sendBytes.Length);
>         receivedBytes = new byte[newClient.ReceiveBufferSize];
>         bytesRead = tcpStream.Read(receivedBytes, 0, newClient.ReceiveBufferSize);
>         returnData = Encoding.ASCII.GetString(receivedBytes);
> 
>         // How do I get the proper UUID from the info provided by 'api show channels'
>         
>         string uuid = '????????'
>         
>         
>         
>         // Set the Custom Channel Variable
>         sendBytes = Encoding.ASCII.GetBytes("api uuid_setvar " + uuid + " qqOneReach_ApplicationId BRIAN" + "\n\n");
>                                                                           
>         tcpStream.Write(sendBytes, 0, sendBytes.Length);
>         receivedBytes = new byte[newClient.ReceiveBufferSize];
>         bytesRead = tcpStream.Read(receivedBytes, 0, newClient.ReceiveBufferSize);
>         returnData = Encoding.ASCII.GetString(receivedBytes);
>         // Clean everything up
>         
>         tcpStream.Flush();
>         tcpStream.Close();
>         newClient.Close();
>       }
>       catch (Exception ex)
>       {
>         throw ex;
>       }
>     }
> 
>  
> Thanks
>  
> Brian Campbell
>  
> From: mrene_lists at avgs.ca
> Date: Thu, 2 Dec 2010 12:05:42 -0500
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Custom Channel Variables via C#
> 
> You need to authenticate first.
> 
> Trying 127.0.0.1...
> Connected to localhost
> Escape character is '^]'.
> Content-Type: auth/request
> 
> auth ClueCon
> 
> Content-Type: command/reply
> Reply-Text: +OK accepted
> 
> api show channels
> 
> Content-Type: api/response
> Content-Length: 748
> 
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid
> aa6e6bee-1c3b-4493-b824-765c5a15f02c,outbound,2010-12-02 12:03:58,1291309438,loopback/park-a,CS_CONSUME_MEDIA,,0000000000,,park,,,inline,inline,L16,8000,128000,L16,8000,128000,,mrene.local,,,RINGING,,,,c9781e01-667e-446b-93e8-a081f0287d15
> 78ff4bb6-573e-4a60-a920-bfdb67f3ed6b,inbound,2010-12-02 12:03:58,1291309438,loopback/park-b,CS_EXECUTE,,0000000000,,park,park,,inline,inline,L16,8000,128000,L16,8000,128000,,mrene.local,,,RINGING,,,,
> 
> 2 total.
> 
> Then you can use the uuid_* commands.
> 
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
> 
> 
> 
> 
> On 2010-12-02, at 10:03 AM, Brian Campbell wrote:
> 
> Thanks, the link was very helpful
>  
> I have taken a look at the link that you provided
>  
> I assume now that I can use uuid_setvar to accomplish what I want
>  
> Assuming that the uuid is 53d37581-1f90-44bf-860a-addbc8430e3a, it seems that my code would need to change to
>  
>  
> private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
> {
>   try
>   {
>     TcpClient newClient = new TcpClient();
>  
>     newClient.Connect("127.0.0.1", 8021);
>  
>     NetworkStream tcpStream = newClient.GetStream();
>  
>     byte[] sendBytes = Encoding.ASCII.GetBytes("api uuid_setvar 53d37581-1f90-44bf-860a-addbc8430e3a myChannelVariable 12345");
>  
>     tcpStream.Write(sendBytes, 0, sendBytes.Length);
>  
>     tcpStream.Close();
>  
>     newClient.Close();
>   }
>   catch(Exception ex)
>   {
>     throw ex;
>   }
> }
> 
> My question now becomes, how do I go about finding out the uuid for the session ? Is that obtainable via mod_event_socket or is there another way ?
>  
> Thanks
>  
> Brian 
> From: mrene_lists at avgs.ca
> Date: Wed, 1 Dec 2010 16:53:53 -0500
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Custom Channel Variables via C#
> 
> Hi,
> 
> You can't send XML dialplan actions on the socket and expect FreeSWITCH to understand it, you must follow the event socket protocol.
> 
> See http://wiki.freeswitch.org/wiki/Event_socket for more information.
> 
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
> 
> 
> 
> 
> On 2010-12-01, at 4:46 PM, Brian Campbell wrote:
> 
>  
> I can set a custom channel variable in my incomming dial plan like this...
>  
> <action application='set' data='MyChannelVariable=12345' />
>  
> And it shows up fine in the CDR
>  
> I am now attempting to use mod_event_socket to set a custom channel variable and have it appear in the CDR
>  
> Here is the C# code so far, the code is part of a Microsoft Speech Server application that is answering the call
>  
> private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
> {
>   try
>   {
>     TcpClient newClient = new TcpClient();
>  
>     newClient.Connect("127.0.0.1", 8021);
>  
>     NetworkStream tcpStream = newClient.GetStream();
>  
>     byte[] sendBytes = Encoding.ASCII.GetBytes("<action application='set' data='MyChannelVariable=12345' />");
>  
>     tcpStream.Write(sendBytes, 0, sendBytes.Length);
>  
>     tcpStream.Close();
>  
>     newClient.Close();
>   }
>   catch(Exception ex)
>   {
>     throw ex;
>   }
> }
>  
> After the call is answered, the C# code seems to run fine, but I dont see the custom channel variable in the resulting CDR
>  
> I figure I am not setting it correctly
>  
> Can anyone advise on what I am doing wrong ?
>  
> Thanks
>  
>  
> Brian
>  
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From bcxml at hotmail.com  Tue Dec 14 23:38:34 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Tue, 14 Dec 2010 15:38:34 -0500
Subject: [Freeswitch-users] Custom Channel Variables via C#
In-Reply-To: <904C3740-32D5-4B78-B444-6FA21E049BDE@avgs.ca>
References: <SNT135-w336CAB94DFC78D328AE6E0B7260@phx.gbl>,
	, , <0E462B4C-C436-462E-9DA7-9EC1241FB380@avgs.ca>, ,
	<SNT135-w4BF48F8FEB9054D804759B7270@phx.gbl>, ,
	<9C89CC76-01FC-4F8D-B00B-89EE3A74EB05@avgs.ca>,
	<SNT135-w6446098BFC470AA8CA2E3B7130@phx.gbl>,
	<904C3740-32D5-4B78-B444-6FA21E049BDE@avgs.ca>
Message-ID: <SNT135-w19C390D1A8FCFD0A46CE4AB7130@phx.gbl>


The bridging is done in the incomming dial plan
 
<extension name="Inbound_3149876543"> 
  <condition field='destination_number' expression='3149876543'> 
    <action application="bridge" data="sofia/internal/3149876543 at 127.0.0.1:5060;transport=tcp"/> 
  </condition> 
</extension>
 
 
Brian
 


From: mrene_lists at avgs.ca
Date: Tue, 14 Dec 2010 15:20:12 -0500
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Custom Channel Variables via C#


Hi,


You have to somehow pass that information to your script. If you wish to control the channel, you could use the "socket" application which establishes an outbound tcp connection to your application.


Otherwise, you could always send an event from the dialplan containing the call's uuid. The event socket connection can control of any calls on the system, how exactly are you bridging the call to your speech server?




Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca






On 2010-12-14, at 2:20 PM, Brian Campbell wrote:

Thanks Mathieu 
 
I have up dated my code as per your instructions.
 
FreeSwitch answers the call and then does a bridge to my Speech Server application. If there are 10 simultaneous calls going on there would be 20 channels. When I send 'api show channels', I get info back for all 20 channels. How do I go about deciding which one contains the correct uuid for the session that I am trying to set a custom channel variable for ?
 
Here is my code
 
     private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
    {
      try
      {
        TcpClient newClient = new TcpClient();
        // Connect to the port
        newClient.Connect(_host, _port);
        NetworkStream tcpStream = newClient.GetStream();
        byte[] receivedBytes = new byte[newClient.ReceiveBufferSize];
        int bytesRead = tcpStream.Read(receivedBytes, 0, newClient.ReceiveBufferSize);
        string returnData = Encoding.ASCII.GetString(receivedBytes);
        // Authenticate
        byte[] sendBytes = Encoding.ASCII.GetBytes("auth ClueCon\n\n");
        tcpStream.Write(sendBytes, 0, sendBytes.Length);
        receivedBytes = new byte[newClient.ReceiveBufferSize];
        bytesRead = tcpStream.Read(receivedBytes, 0, newClient.ReceiveBufferSize);
        returnData = Encoding.ASCII.GetString(receivedBytes);
        // Get channel info
        sendBytes = Encoding.ASCII.GetBytes("api show channels\n\n");
        tcpStream.Write(sendBytes, 0, sendBytes.Length);
        receivedBytes = new byte[newClient.ReceiveBufferSize];
        bytesRead = tcpStream.Read(receivedBytes, 0, newClient.ReceiveBufferSize);
        returnData = Encoding.ASCII.GetString(receivedBytes);

        // How do I get the proper UUID from the info provided by 'api show channels'
        
        string uuid = '????????'
        
        
        
        // Set the Custom Channel Variable
        sendBytes = Encoding.ASCII.GetBytes("api uuid_setvar " + uuid + " qqOneReach_ApplicationId BRIAN" + "\n\n");
                                                                          
        tcpStream.Write(sendBytes, 0, sendBytes.Length);
        receivedBytes = new byte[newClient.ReceiveBufferSize];
        bytesRead = tcpStream.Read(receivedBytes, 0, newClient.ReceiveBufferSize);
        returnData = Encoding.ASCII.GetString(receivedBytes);
        // Clean everything up
        
        tcpStream.Flush();
        tcpStream.Close();
        newClient.Close();
      }
      catch (Exception ex)
      {
        throw ex;
      }
    }

 
Thanks
 
Brian Campbell
 


From: mrene_lists at avgs.ca
Date: Thu, 2 Dec 2010 12:05:42 -0500
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Custom Channel Variables via C#

You need to authenticate first.



Trying 127.0.0.1...
Connected to localhost
Escape character is '^]'.
Content-Type: auth/request


auth ClueCon


Content-Type: command/reply
Reply-Text: +OK accepted


api show channels


Content-Type: api/response
Content-Length: 748


uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid
aa6e6bee-1c3b-4493-b824-765c5a15f02c,outbound,2010-12-02 12:03:58,1291309438,loopback/park-a,CS_CONSUME_MEDIA,,0000000000,,park,,,inline,inline,L16,8000,128000,L16,8000,128000,,mrene.local,,,RINGING,,,,c9781e01-667e-446b-93e8-a081f0287d15
78ff4bb6-573e-4a60-a920-bfdb67f3ed6b,inbound,2010-12-02 12:03:58,1291309438,loopback/park-b,CS_EXECUTE,,0000000000,,park,park,,inline,inline,L16,8000,128000,L16,8000,128000,,mrene.local,,,RINGING,,,,


2 total.


Then you can use the uuid_* commands.





Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca






On 2010-12-02, at 10:03 AM, Brian Campbell wrote:

Thanks, the link was very helpful
 
I have taken a look at the link that you provided
 
I assume now that I can use uuid_setvar to accomplish what I want
 
Assuming that the uuid is 53d37581-1f90-44bf-860a-addbc8430e3a, it seems that my code would need to change to
 
 
private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
{
  try
  {
    TcpClient newClient = new TcpClient();
 
    newClient.Connect("127.0.0.1", 8021);
 
    NetworkStream tcpStream = newClient.GetStream();
 
    byte[] sendBytes = Encoding.ASCII.GetBytes("api uuid_setvar 53d37581-1f90-44bf-860a-addbc8430e3a myChannelVariable 12345");
 
    tcpStream.Write(sendBytes, 0, sendBytes.Length);
 
    tcpStream.Close();
 
    newClient.Close();
  }
  catch(Exception ex)
  {
    throw ex;
  }
}

My question now becomes, how do I go about finding out the uuid for the session ? Is that obtainable via mod_event_socket or is there another way ?
 
Thanks
 
Brian 


From: mrene_lists at avgs.ca
Date: Wed, 1 Dec 2010 16:53:53 -0500
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Custom Channel Variables via C#


Hi,


You can't send XML dialplan actions on the socket and expect FreeSWITCH to understand it, you must follow the event socket protocol.


See http://wiki.freeswitch.org/wiki/Event_socket for more information.




Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca






On 2010-12-01, at 4:46 PM, Brian Campbell wrote:

 
I can set a custom channel variable in my incomming dial plan like this...
 
<action application='set' data='MyChannelVariable=12345' />
 
And it shows up fine in the CDR
 
I am now attempting to use mod_event_socket to set a custom channel variable and have it appear in the CDR
 
Here is the C# code so far, the code is part of a Microsoft Speech Server application that is answering the call
 
private void injectCdrCode_ExecuteCode(object sender, EventArgs e)
{
  try
  {
    TcpClient newClient = new TcpClient();
 
    newClient.Connect("127.0.0.1", 8021);
 
    NetworkStream tcpStream = newClient.GetStream();
 
    byte[] sendBytes = Encoding.ASCII.GetBytes("<action application='set' data='MyChannelVariable=12345' />");
 
    tcpStream.Write(sendBytes, 0, sendBytes.Length);
 
    tcpStream.Close();
 
    newClient.Close();
  }
  catch(Exception ex)
  {
    throw ex;
  }
}
 
After the call is answered, the C# code seems to run fine, but I dont see the custom channel variable in the resulting CDR
 
I figure I am not setting it correctly
 
Can anyone advise on what I am doing wrong ?
 
Thanks
 
 
Brian
 
 
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From moises.silva at gmail.com  Wed Dec 15 00:03:46 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 14 Dec 2010 16:03:46 -0500
Subject: [Freeswitch-users] freetdm and ring_ready
In-Reply-To: <AANLkTikfyzKUZhb=2KS9=2v_Fmk=mj0=ruz328aTceM3@mail.gmail.com>
References: <AANLkTikfyzKUZhb=2KS9=2v_Fmk=mj0=ruz328aTceM3@mail.gmail.com>
Message-ID: <AANLkTimO_0JV30SoC62we=aX3Lh2-nV5wQ+UsL7oN5Gm@mail.gmail.com>

On Tue, Dec 14, 2010 at 5:23 AM, Neil Patel <neilp at cs.stanford.edu> wrote:

> I am using ring_ready to let incoming calls to my IVR app ring couple times
> before the app engages:
>
> session:execute("ring_ready");
> session:sleep(8000);
> ...
> session:answer();
>
> On one of my servers I am using freeTDM to interface with Sangoma hardware,
> and another uses OpenZap (this is really the only difference between the two
> FS instances). Ringing/early media is playing on the openzap server, but not
> with freeTDM. Any idea why not?
>
>
Which signaling is this?

Can you post the debug output from FreeSWITCH?

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com
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From infos at madovsky.org  Wed Dec 15 00:28:10 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 14 Dec 2010 16:28:10 -0500
Subject: [Freeswitch-users] send dtmf on one leg bridge
References: <D10B61781EFA4CFB913F2FEAC7579B27@e1705><AANLkTikhpiqsFbQHeWuxNE2EojeuFVgrdseBC5CU=308@mail.gmail.com><7555EA3AD7864DA9B52A6899DD7FF2B3@e1705><AANLkTikFXnH1TYEUe4D0uM6q+WMnvWqCQ+19n0bSZYM_@mail.gmail.com><05F742BEE64A443C93952DD22DC7E84C@e1705><AANLkTikL-ZWCyXhfHmW4daTdT1q-zARmEJZ3s7kFXiv+@mail.gmail.com>
	<AANLkTin5ejmu-QAMSa_YafXUfp-NdCo-KfVSRi2gxT5E@mail.gmail.com>
Message-ID: <679F7530A0B449FB9DF4E14C6CCCECEE@e1705>

Hi Mike,

I finally decided to finish to implement RFC2833
in my sip phone, I think it's the clearest way
thanks for your help
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, December 14, 2010 3:32 AM
  Subject: Re: [Freeswitch-users] send dtmf on one leg bridge


  Per my other message you may also just need to use uuid_recv_dtmf. Try it and see.
  -MC


  On Mon, Dec 13, 2010 at 11:12 PM, Steven Ayre <steveayre at gmail.com> wrote:

    Perhaps the UUID variable doesn't contain the correct value though...

    Try uuid_send_dtmf using the known UUID yourself. That'll show you
    whether uuid_send_dtmf works and the problem is in the expand api, or
    whether it's uuid_send_dtmf that's having a problem.

    -Steve


    On 14 December 2010 01:18, Madovsky <infos at madovsky.org> wrote:
    > because I need to use variables inside the api request
    > is it make a difference ?
    >
    > ----- Original Message -----
    > From: "Steven Ayre" <steveayre at gmail.com>
    > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
    > Sent: Monday, December 13, 2010 7:47 PM
    > Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
    >
    >
    >> What about  uuid_send_dtmf directly? i.e. not through expand.
    >>
    >> -Steve
    >>
    >>
    >> On 13 December 2010 20:04, Madovsky <infos at madovsky.org> wrote:
    >>> yes,
    >>>
    >>> I tried expand uuid_send_dtmf [uuid] 12345 at 120 once
    >>> answer done and ivr asks pin number but the CLI
    >>> says "ERR- no reply".
    >>>
    >>> ----- Original Message -----
    >>> From: Michael Collins
    >>> To: FreeSWITCH Users Help
    >>> Sent: Monday, December 13, 2010 2:50 PM
    >>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
    >>> Have you tried uuid_send_dtmf?
    >>> -MC
    >>>
    >>> On Sun, Dec 12, 2010 at 12:30 PM, Madovsky <infos at madovsky.org> wrote:
    >>>>
    >>>> Is it possible to send dtfmf via
    >>>> send_dtmf from CLI on a one leg bridge ?
    >>>> it's for enter a pin conference and I can't use
    >>>> any RFC on my sip phone.
    >>>>
    >>>> Thanks
    >>>>
    >>>> Franck
    >>>> _______________________________________________
    >>>> FreeSWITCH-users mailing list
    >>>> FreeSWITCH-users at lists.freeswitch.org
    >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >>>> http://www.freeswitch.org
    >>>>
    >>>
    >>> ________________________________
    >>>
    >>> _______________________________________________
    >>> FreeSWITCH-users mailing list
    >>> FreeSWITCH-users at lists.freeswitch.org
    >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >>> http://www.freeswitch.org
    >>>
    >>> _______________________________________________
    >>> FreeSWITCH-users mailing list
    >>> FreeSWITCH-users at lists.freeswitch.org
    >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >>> http://www.freeswitch.org
    >>>
    >>>
    >>
    >> _______________________________________________
    >> FreeSWITCH-users mailing list
    >> FreeSWITCH-users at lists.freeswitch.org
    >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >> http://www.freeswitch.org
    >
    >
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >

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





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


  _______________________________________________
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From shamun.toha at gmail.com  Wed Dec 15 00:32:01 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Tue, 14 Dec 2010 22:32:01 +0100
Subject: [Freeswitch-users] Pre-paid impossible with freeSwitch ?
In-Reply-To: <AANLkTikKHg5NYCcjJ+vbnsEZYVKKLCMjZw4U0eRm3kzK@mail.gmail.com>
References: <AANLkTim3mXSPXcO3AjOt9-C6zj82MmONH+2Anc3n_S5k@mail.gmail.com>
	<AANLkTikKHg5NYCcjJ+vbnsEZYVKKLCMjZw4U0eRm3kzK@mail.gmail.com>
Message-ID: <AANLkTik2jVWMOjpVZTzkePVw5jO65cFx+8LYjC_mj1XW@mail.gmail.com>

Hello There,

No the existing billing not solving my problem. Because of custom project,
it for many channels when gets connected it becomes 1 call and then it needs
to check the balance and also deduct the balance when the call is connected.

If the balance is 0 or -0 FS trigger the call drop. I build the whole
billing using Javascript but i cant find any resource where i can say while
call running, allow me to negotiate with my alternative pipe line.

Code example:
... connected call .. voice will now get freezed.....
    default:
         module = EYE_HELPER + "________FAIL__________";
     break;
}
while(session.state == "CS_EXECUTE") { i++; if (i > 1000) {
console_log('err', '.'); i=0;  }  // here i wanted to go to my BALANCE
module, But whole fs_cli gets FREEZED. }


Already tried those, but FS gets freezed:
========================
CS_NEW Channel is newly created.
CS_INIT Channel has been initialized.
CS_ROUTING Channel is looking for an extension to execute.
CS_SOFT_EXECUTE Channel is ready to execute from 3rd party control.
CS_EXECUTE Channel is executing its dialplan.
CS_EXCHANGE_MEDIA Channel is exchanging media with another channel.
CS_PARK Channel is accepting media awaiting commands.
CS_CONSUME_MEDIA Channel is consuming all media and dropping it.
CS_HIBERNATE Channel is in a sleep state.
CS_RESET Channel is in a reset state.
CS_HANGUP Channel is flagged for hangup and ready to end. Media will now
end, and no further call routing will occur.
CS_REPORTING The channel is already hung up, media is already down, and now
it's time to do any sort of reporting processes such as CDR logging.
CS_DESTROY Channel is ready to be destroyed and out of the state machine.
Memory pools are returned to the core and utilized memory from the channel
is freed.


Please guide kindly....

Thanks & Regards
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From brian at freeswitch.org  Wed Dec 15 00:32:35 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 14 Dec 2010 15:32:35 -0600
Subject: [Freeswitch-users] send dtmf on one leg bridge
In-Reply-To: <679F7530A0B449FB9DF4E14C6CCCECEE@e1705>
References: <D10B61781EFA4CFB913F2FEAC7579B27@e1705><AANLkTikhpiqsFbQHeWuxNE2EojeuFVgrdseBC5CU=308@mail.gmail.com><7555EA3AD7864DA9B52A6899DD7FF2B3@e1705><AANLkTikFXnH1TYEUe4D0uM6q+WMnvWqCQ+19n0bSZYM_@mail.gmail.com><05F742BEE64A443C93952DD22DC7E84C@e1705><AANLkTikL-ZWCyXhfHmW4daTdT1q-zARmEJZ3s7kFXiv+@mail.gmail.com>
	<AANLkTin5ejmu-QAMSa_YafXUfp-NdCo-KfVSRi2gxT5E@mail.gmail.com>
	<679F7530A0B449FB9DF4E14C6CCCECEE@e1705>
Message-ID: <46DD9C1B-B920-4793-98AB-BB7D9AE359B4@freeswitch.org>

be careful... you only THINK you have done an implementation of 2833 to find out that your version doesn't work with 100's of things out there :)

/b

On Dec 14, 2010, at 3:28 PM, Madovsky wrote:

> Hi Mike,
>  
> I finally decided to finish to implement RFC2833
> in my sip phone, I think it's the clearest way
> thanks for your help
> 

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From brian at freeswitch.org  Wed Dec 15 00:34:32 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 14 Dec 2010 15:34:32 -0600
Subject: [Freeswitch-users] Pre-paid impossible with freeSwitch ?
In-Reply-To: <AANLkTik2jVWMOjpVZTzkePVw5jO65cFx+8LYjC_mj1XW@mail.gmail.com>
References: <AANLkTim3mXSPXcO3AjOt9-C6zj82MmONH+2Anc3n_S5k@mail.gmail.com>
	<AANLkTikKHg5NYCcjJ+vbnsEZYVKKLCMjZw4U0eRm3kzK@mail.gmail.com>
	<AANLkTik2jVWMOjpVZTzkePVw5jO65cFx+8LYjC_mj1XW@mail.gmail.com>
Message-ID: <AAF173CF-95E5-4A91-925A-4BE5B152D931@freeswitch.org>

I have to say that statement right there is WRONG.  You should never ever be doing billing inline like this and if you are you're doing it wrong.

Just my advice here... you don't have to follow it you can take it with a grain of salt but if you really want to do this correctly then you had better learn C.

/b

On Dec 14, 2010, at 3:32 PM, Shamun toha md wrote:

> I build the whole billing using Javascript 



From shamun.toha at gmail.com  Wed Dec 15 00:45:44 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Tue, 14 Dec 2010 22:45:44 +0100
Subject: [Freeswitch-users] Pre-paid impossible with freeSwitch ?
In-Reply-To: <AAF173CF-95E5-4A91-925A-4BE5B152D931@freeswitch.org>
References: <AANLkTim3mXSPXcO3AjOt9-C6zj82MmONH+2Anc3n_S5k@mail.gmail.com>
	<AANLkTikKHg5NYCcjJ+vbnsEZYVKKLCMjZw4U0eRm3kzK@mail.gmail.com>
	<AANLkTik2jVWMOjpVZTzkePVw5jO65cFx+8LYjC_mj1XW@mail.gmail.com>
	<AAF173CF-95E5-4A91-925A-4BE5B152D931@freeswitch.org>
Message-ID: <AANLkTinFW451A5M5KVB+bZ7qysn9yODhHehTXv4cp5+Z@mail.gmail.com>

Line 577 ex: http://pastebin.freeswitch.org/14782

I need to have a background thread, to lookup my billing via CURL and deduct
that live real time. How can i do that any suggestion plz.

** There are almost relative more 5,000 lines of codes and modules i wrote,
when this module get passed.

Looking forward for any tips or guides....

Thanks & Regards



On Tue, Dec 14, 2010 at 10:34 PM, Brian West <brian at freeswitch.org> wrote:

> I have to say that statement right there is WRONG.  You should never ever
> be doing billing inline like this and if you are you're doing it wrong.
>
> Just my advice here... you don't have to follow it you can take it with a
> grain of salt but if you really want to do this correctly then you had
> better learn C.
>
> /b
>
> On Dec 14, 2010, at 3:32 PM, Shamun toha md wrote:
>
> > I build the whole billing using Javascript
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From shamun.toha at gmail.com  Wed Dec 15 01:27:21 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Tue, 14 Dec 2010 23:27:21 +0100
Subject: [Freeswitch-users] Pre-paid impossible with freeSwitch ?
In-Reply-To: <AAF173CF-95E5-4A91-925A-4BE5B152D931@freeswitch.org>
References: <AANLkTim3mXSPXcO3AjOt9-C6zj82MmONH+2Anc3n_S5k@mail.gmail.com>
	<AANLkTikKHg5NYCcjJ+vbnsEZYVKKLCMjZw4U0eRm3kzK@mail.gmail.com>
	<AANLkTik2jVWMOjpVZTzkePVw5jO65cFx+8LYjC_mj1XW@mail.gmail.com>
	<AAF173CF-95E5-4A91-925A-4BE5B152D931@freeswitch.org>
Message-ID: <AANLkTikQyHarLqvP=nE9wvDvoQ6qFW0RE6NjTJh9toQy@mail.gmail.com>

Is there no scanner using crontab to connnect FS and trigger command
uuid_drop uuid that i want to drop ?

- How can i solve this using ? "SpiderMonkey" is there any patch to have the
setTimeout() function alteast, which will be a scanner for my session.uuid's
???





On Tue, Dec 14, 2010 at 10:34 PM, Brian West <brian at freeswitch.org> wrote:

> I have to say that statement right there is WRONG.  You should never ever
> be doing billing inline like this and if you are you're doing it wrong.
>
> Just my advice here... you don't have to follow it you can take it with a
> grain of salt but if you really want to do this correctly then you had
> better learn C.
>
> /b
>
> On Dec 14, 2010, at 3:32 PM, Shamun toha md wrote:
>
> > I build the whole billing using Javascript
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tayeb.meftah at gmail.com  Wed Dec 15 02:24:35 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Wed, 15 Dec 2010 00:24:35 +0100
Subject: [Freeswitch-users] Pre-paid impossible with freeSwitch ?
In-Reply-To: <AANLkTikQyHarLqvP=nE9wvDvoQ6qFW0RE6NjTJh9toQy@mail.gmail.com>
References: <AANLkTim3mXSPXcO3AjOt9-C6zj82MmONH+2Anc3n_S5k@mail.gmail.com>	<AANLkTikKHg5NYCcjJ+vbnsEZYVKKLCMjZw4U0eRm3kzK@mail.gmail.com>	<AANLkTik2jVWMOjpVZTzkePVw5jO65cFx+8LYjC_mj1XW@mail.gmail.com>	<AAF173CF-95E5-4A91-925A-4BE5B152D931@freeswitch.org>
	<AANLkTikQyHarLqvP=nE9wvDvoQ6qFW0RE6NjTJh9toQy@mail.gmail.com>
Message-ID: <4D07FCB3.8040203@gmail.com>

USE ESL
http://wiki.freeswitch.org/wiki/ESL
Le 14/12/2010 23:27, Shamun toha md a ?crit :
> Is there no scanner using crontab to connnect FS and trigger command 
> uuid_drop uuid that i want to drop ?
>
> - How can i solve this using ? "SpiderMonkey" is there any patch to 
> have the setTimeout() function alteast, which will be a scanner for my 
> session.uuid's ???
>
>
>
>
>
> On Tue, Dec 14, 2010 at 10:34 PM, Brian West <brian at freeswitch.org 
> <mailto:brian at freeswitch.org>> wrote:
>
>     I have to say that statement right there is WRONG.  You should
>     never ever be doing billing inline like this and if you are you're
>     doing it wrong.
>
>     Just my advice here... you don't have to follow it you can take it
>     with a grain of salt but if you really want to do this correctly
>     then you had better learn C.
>
>     /b
>
>     On Dec 14, 2010, at 3:32 PM, Shamun toha md wrote:
>
>     > I build the whole billing using Javascript
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313

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From msc at freeswitch.org  Wed Dec 15 03:20:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 14 Dec 2010 16:20:16 -0800
Subject: [Freeswitch-users] Pre-paid impossible with freeSwitch ?
In-Reply-To: <AANLkTikQyHarLqvP=nE9wvDvoQ6qFW0RE6NjTJh9toQy@mail.gmail.com>
References: <AANLkTim3mXSPXcO3AjOt9-C6zj82MmONH+2Anc3n_S5k@mail.gmail.com>
	<AANLkTikKHg5NYCcjJ+vbnsEZYVKKLCMjZw4U0eRm3kzK@mail.gmail.com>
	<AANLkTik2jVWMOjpVZTzkePVw5jO65cFx+8LYjC_mj1XW@mail.gmail.com>
	<AAF173CF-95E5-4A91-925A-4BE5B152D931@freeswitch.org>
	<AANLkTikQyHarLqvP=nE9wvDvoQ6qFW0RE6NjTJh9toQy@mail.gmail.com>
Message-ID: <AANLkTi=GrckXqUqpG2Tzc0E2okRHvey1pXmDUabq+3Fh@mail.gmail.com>

If you can execute a system command then you could do something as simple as
this:
system('fs_cli -x "uuid_kill xxxxxxxx"')

Where xxx is the uuid you want to hang up.

Just so you know, the method you've chosen may have been somewhat easy to
implement, it probably won't scale very well because you'll have a JS
instance for every call that goes through this billing system. A few dozen
calls won't be a problem, but if you intend to scale then I would recommend
you look at creating a call control system written in C or use ESL (event
socket library) to have a program connect to FS via TCP where you can
control many hundreds of calls simultaneously and you can have your program
connect to an external database, checking balances, etc.

Just my two cents.

-MC

On Tue, Dec 14, 2010 at 2:27 PM, Shamun toha md <shamun.toha at gmail.com>wrote:

> Is there no scanner using crontab to connnect FS and trigger command
> uuid_drop uuid that i want to drop ?
>
> - How can i solve this using ? "SpiderMonkey" is there any patch to have
> the setTimeout() function alteast, which will be a scanner for my
> session.uuid's ???
>
>
>
>
>
> On Tue, Dec 14, 2010 at 10:34 PM, Brian West <brian at freeswitch.org> wrote:
>
>> I have to say that statement right there is WRONG.  You should never ever
>> be doing billing inline like this and if you are you're doing it wrong.
>>
>> Just my advice here... you don't have to follow it you can take it with a
>> grain of salt but if you really want to do this correctly then you had
>> better learn C.
>>
>> /b
>>
>> On Dec 14, 2010, at 3:32 PM, Shamun toha md wrote:
>>
>> > I build the whole billing using Javascript
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From edpimentl at gmail.com  Wed Dec 15 03:32:25 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Tue, 14 Dec 2010 19:32:25 -0500
Subject: [Freeswitch-users] Pre-paid impossible with freeSwitch ?
In-Reply-To: <AANLkTi=GrckXqUqpG2Tzc0E2okRHvey1pXmDUabq+3Fh@mail.gmail.com>
References: <AANLkTim3mXSPXcO3AjOt9-C6zj82MmONH+2Anc3n_S5k@mail.gmail.com>
	<AANLkTikKHg5NYCcjJ+vbnsEZYVKKLCMjZw4U0eRm3kzK@mail.gmail.com>
	<AANLkTik2jVWMOjpVZTzkePVw5jO65cFx+8LYjC_mj1XW@mail.gmail.com>
	<AAF173CF-95E5-4A91-925A-4BE5B152D931@freeswitch.org>
	<AANLkTikQyHarLqvP=nE9wvDvoQ6qFW0RE6NjTJh9toQy@mail.gmail.com>
	<AANLkTi=GrckXqUqpG2Tzc0E2okRHvey1pXmDUabq+3Fh@mail.gmail.com>
Message-ID: <AANLkTikq2Y_7DrGdy4iKUJSVosOFydYBb=DpZg1UiLqX@mail.gmail.com>

Today JS, does scale very well, when you use the techniques and knowledge
below:

How to scale JS ... Node.JS , MongoDB/CouchDB
http://howtonode.org/node-and-mongo
http://developer.yahoo.com/yui/theater/video.php?v=zakas-architecture
JavaScript The Good Parts  (D Crockford)
http://oreilly.com/catalog/9780596517748
Event Faster WebSites (Steve Souder)
http://oreilly.com/catalog/9780596522308/



-E
Gpro.ws
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From msc at freeswitch.org  Wed Dec 15 03:57:17 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 14 Dec 2010 16:57:17 -0800
Subject: [Freeswitch-users] Pre-paid impossible with freeSwitch ?
In-Reply-To: <AANLkTikq2Y_7DrGdy4iKUJSVosOFydYBb=DpZg1UiLqX@mail.gmail.com>
References: <AANLkTim3mXSPXcO3AjOt9-C6zj82MmONH+2Anc3n_S5k@mail.gmail.com>
	<AANLkTikKHg5NYCcjJ+vbnsEZYVKKLCMjZw4U0eRm3kzK@mail.gmail.com>
	<AANLkTik2jVWMOjpVZTzkePVw5jO65cFx+8LYjC_mj1XW@mail.gmail.com>
	<AAF173CF-95E5-4A91-925A-4BE5B152D931@freeswitch.org>
	<AANLkTikQyHarLqvP=nE9wvDvoQ6qFW0RE6NjTJh9toQy@mail.gmail.com>
	<AANLkTi=GrckXqUqpG2Tzc0E2okRHvey1pXmDUabq+3Fh@mail.gmail.com>
	<AANLkTikq2Y_7DrGdy4iKUJSVosOFydYBb=DpZg1UiLqX@mail.gmail.com>
Message-ID: <AANLkTinN8ZQH+sPaWhJL4isi9Adnsdwj_DKSEzQ4Smnc@mail.gmail.com>

Let me know when you have all that integrated into mod_spidermonkey. :)
-MC

On Tue, Dec 14, 2010 at 4:32 PM, EdPimentl <edpimentl at gmail.com> wrote:

> Today JS, does scale very well, when you use the techniques and knowledge
> below:
>
> How to scale JS ... Node.JS , MongoDB/CouchDB
> http://howtonode.org/node-and-mongo
> http://developer.yahoo.com/yui/theater/video.php?v=zakas-architecture
> JavaScript The Good Parts  (D Crockford)
> http://oreilly.com/catalog/9780596517748
> Event Faster WebSites (Steve Souder)
> http://oreilly.com/catalog/9780596522308/
>
>
>
> -E
> Gpro.ws
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Wed Dec 15 04:09:08 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 14 Dec 2010 20:09:08 -0500
Subject: [Freeswitch-users] send dtmf on one leg bridge
References: <D10B61781EFA4CFB913F2FEAC7579B27@e1705><AANLkTikhpiqsFbQHeWuxNE2EojeuFVgrdseBC5CU=308@mail.gmail.com><7555EA3AD7864DA9B52A6899DD7FF2B3@e1705><AANLkTikFXnH1TYEUe4D0uM6q+WMnvWqCQ+19n0bSZYM_@mail.gmail.com><05F742BEE64A443C93952DD22DC7E84C@e1705><AANLkTikL-ZWCyXhfHmW4daTdT1q-zARmEJZ3s7kFXiv+@mail.gmail.com>
	<AANLkTin5ejmu-QAMSa_YafXUfp-NdCo-KfVSRi2gxT5E@mail.gmail.com>
Message-ID: <431EA9FFDE8A496AA860C83C6F692A33@e1705>

Ok I will try. Bingo Brian, 
the rfc2833 RTP payload is not interpreted by FS
I have to debug it.
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, December 14, 2010 3:32 AM
  Subject: Re: [Freeswitch-users] send dtmf on one leg bridge


  Per my other message you may also just need to use uuid_recv_dtmf. Try it and see.
  -MC


  On Mon, Dec 13, 2010 at 11:12 PM, Steven Ayre <steveayre at gmail.com> wrote:

    Perhaps the UUID variable doesn't contain the correct value though...

    Try uuid_send_dtmf using the known UUID yourself. That'll show you
    whether uuid_send_dtmf works and the problem is in the expand api, or
    whether it's uuid_send_dtmf that's having a problem.

    -Steve


    On 14 December 2010 01:18, Madovsky <infos at madovsky.org> wrote:
    > because I need to use variables inside the api request
    > is it make a difference ?
    >
    > ----- Original Message -----
    > From: "Steven Ayre" <steveayre at gmail.com>
    > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
    > Sent: Monday, December 13, 2010 7:47 PM
    > Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
    >
    >
    >> What about  uuid_send_dtmf directly? i.e. not through expand.
    >>
    >> -Steve
    >>
    >>
    >> On 13 December 2010 20:04, Madovsky <infos at madovsky.org> wrote:
    >>> yes,
    >>>
    >>> I tried expand uuid_send_dtmf [uuid] 12345 at 120 once
    >>> answer done and ivr asks pin number but the CLI
    >>> says "ERR- no reply".
    >>>
    >>> ----- Original Message -----
    >>> From: Michael Collins
    >>> To: FreeSWITCH Users Help
    >>> Sent: Monday, December 13, 2010 2:50 PM
    >>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
    >>> Have you tried uuid_send_dtmf?
    >>> -MC
    >>>
    >>> On Sun, Dec 12, 2010 at 12:30 PM, Madovsky <infos at madovsky.org> wrote:
    >>>>
    >>>> Is it possible to send dtfmf via
    >>>> send_dtmf from CLI on a one leg bridge ?
    >>>> it's for enter a pin conference and I can't use
    >>>> any RFC on my sip phone.
    >>>>
    >>>> Thanks
    >>>>
    >>>> Franck
    >>>> _______________________________________________
    >>>> FreeSWITCH-users mailing list
    >>>> FreeSWITCH-users at lists.freeswitch.org
    >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >>>> http://www.freeswitch.org
    >>>>
    >>>
    >>> ________________________________
    >>>
    >>> _______________________________________________
    >>> FreeSWITCH-users mailing list
    >>> FreeSWITCH-users at lists.freeswitch.org
    >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >>> http://www.freeswitch.org
    >>>
    >>> _______________________________________________
    >>> FreeSWITCH-users mailing list
    >>> FreeSWITCH-users at lists.freeswitch.org
    >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >>> http://www.freeswitch.org
    >>>
    >>>
    >>
    >> _______________________________________________
    >> FreeSWITCH-users mailing list
    >> FreeSWITCH-users at lists.freeswitch.org
    >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >> http://www.freeswitch.org
    >
    >
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >

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





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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Wed Dec 15 05:23:11 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 14 Dec 2010 20:23:11 -0600
Subject: [Freeswitch-users] CBeyond interop
Message-ID: <BFA72E80-0911-4FC5-8198-58B95C64F15A@freeswitch.org>

Has anyone gotten FreeSWITCH working with CBeyond?  I'm stumped as to what they expect and they don't wanna tell me or share any working traces.

Thanks,
Brian



From infos at madovsky.org  Wed Dec 15 05:27:47 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 14 Dec 2010 21:27:47 -0500
Subject: [Freeswitch-users] send dtmf on one leg bridge
Message-ID: <EB50CB0701AE4FF5B27938870B853740@e1705>

is it need any specific sdp for rfc2833 ?

variable_switch_r_sdp: [v=0
o=- 1292379777 1292379777 IN IP4 67.205.XX.XX
s=-
c=IN IP4 67.205.XX.XX
t=0 0
m=audio 46566 RTP/AVP 96
a=rtpmap:96 speex/16000
m=video 53492 RTP/AVP 97
]

seems missing
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

does dtmf need it ?

Thanks





  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, December 14, 2010 8:09 PM
  Subject: Re: [Freeswitch-users] send dtmf on one leg bridge


  Ok I will try. Bingo Brian, 
  the rfc2833 RTP payload is not interpreted by FS
  I have to debug it.
    ----- Original Message ----- 
    From: Michael Collins 
    To: FreeSWITCH Users Help 
    Sent: Tuesday, December 14, 2010 3:32 AM
    Subject: Re: [Freeswitch-users] send dtmf on one leg bridge


    Per my other message you may also just need to use uuid_recv_dtmf. Try it and see. 
    -MC


    On Mon, Dec 13, 2010 at 11:12 PM, Steven Ayre <steveayre at gmail.com> wrote:

      Perhaps the UUID variable doesn't contain the correct value though...

      Try uuid_send_dtmf using the known UUID yourself. That'll show you
      whether uuid_send_dtmf works and the problem is in the expand api, or
      whether it's uuid_send_dtmf that's having a problem.

      -Steve


      On 14 December 2010 01:18, Madovsky <infos at madovsky.org> wrote:
      > because I need to use variables inside the api request
      > is it make a difference ?
      >
      > ----- Original Message -----
      > From: "Steven Ayre" <steveayre at gmail.com>
      > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
      > Sent: Monday, December 13, 2010 7:47 PM
      > Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
      >
      >
      >> What about  uuid_send_dtmf directly? i.e. not through expand.
      >>
      >> -Steve
      >>
      >>
      >> On 13 December 2010 20:04, Madovsky <infos at madovsky.org> wrote:
      >>> yes,
      >>>
      >>> I tried expand uuid_send_dtmf [uuid] 12345 at 120 once
      >>> answer done and ivr asks pin number but the CLI
      >>> says "ERR- no reply".
      >>>
      >>> ----- Original Message -----
      >>> From: Michael Collins
      >>> To: FreeSWITCH Users Help
      >>> Sent: Monday, December 13, 2010 2:50 PM
      >>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
      >>> Have you tried uuid_send_dtmf?
      >>> -MC
      >>>
      >>> On Sun, Dec 12, 2010 at 12:30 PM, Madovsky <infos at madovsky.org> wrote:
      >>>>
      >>>> Is it possible to send dtfmf via
      >>>> send_dtmf from CLI on a one leg bridge ?
      >>>> it's for enter a pin conference and I can't use
      >>>> any RFC on my sip phone.
      >>>>
      >>>> Thanks
      >>>>
      >>>> Franck
      >>>> _______________________________________________
      >>>> FreeSWITCH-users mailing list
      >>>> FreeSWITCH-users at lists.freeswitch.org
      >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      >>>> http://www.freeswitch.org
      >>>>
      >>>
      >>> ________________________________
      >>>
      >>> _______________________________________________
      >>> FreeSWITCH-users mailing list
      >>> FreeSWITCH-users at lists.freeswitch.org
      >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      >>> http://www.freeswitch.org
      >>>
      >>> _______________________________________________
      >>> FreeSWITCH-users mailing list
      >>> FreeSWITCH-users at lists.freeswitch.org
      >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      >>> http://www.freeswitch.org
      >>>
      >>>
      >>
      >> _______________________________________________
      >> FreeSWITCH-users mailing list
      >> FreeSWITCH-users at lists.freeswitch.org
      >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      >> http://www.freeswitch.org
      >
      >
      > _______________________________________________
      > FreeSWITCH-users mailing list
      > FreeSWITCH-users at lists.freeswitch.org
      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      > http://www.freeswitch.org
      >

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





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


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From mrene_lists at avgs.ca  Wed Dec 15 05:29:55 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 14 Dec 2010 21:29:55 -0500
Subject: [Freeswitch-users] send dtmf on one leg bridge
In-Reply-To: <EB50CB0701AE4FF5B27938870B853740@e1705>
References: <EB50CB0701AE4FF5B27938870B853740@e1705>
Message-ID: <C3401235-C0CD-4589-8CAC-B1DE99D0A1A6@avgs.ca>

Indeed it is.

http://www.faqs.org/rfcs/rfc2833.html

  The RTP payload format is designated as "telephone-event", the MIME
   type as "audio/telephone-event". The default timestamp rate is 8000
   Hz, but other rates may be defined. In accordance with current
   practice, this payload format does not have a static payload type
   number, but uses a RTP payload type number established dynamically
   and out-of-band.


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-12-14, at 9:27 PM, Madovsky wrote:

> is it need any specific sdp for rfc2833 ?
>  
> variable_switch_r_sdp: [v=0
> o=- 1292379777 1292379777 IN IP4 67.205.XX.XX
> s=-
> c=IN IP4 67.205.XX.XX
> t=0 0
> m=audio 46566 RTP/AVP 96
> a=rtpmap:96 speex/16000
> m=video 53492 RTP/AVP 97
> ]
>  
> seems missing
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
>  
> does dtmf need it ?
>  
> Thanks
>  
> 
> 
> ----- Original Message -----
> From: Madovsky
> To: FreeSWITCH Users Help
> Sent: Tuesday, December 14, 2010 8:09 PM
> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
> 
> Ok I will try. Bingo Brian,
> the rfc2833 RTP payload is not interpreted by FS
> I have to debug it.
> ----- Original Message -----
> From: Michael Collins
> To: FreeSWITCH Users Help
> Sent: Tuesday, December 14, 2010 3:32 AM
> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
> 
> Per my other message you may also just need to use uuid_recv_dtmf. Try it and see.
> -MC
> 
> On Mon, Dec 13, 2010 at 11:12 PM, Steven Ayre <steveayre at gmail.com> wrote:
> Perhaps the UUID variable doesn't contain the correct value though...
> 
> Try uuid_send_dtmf using the known UUID yourself. That'll show you
> whether uuid_send_dtmf works and the problem is in the expand api, or
> whether it's uuid_send_dtmf that's having a problem.
> 
> -Steve
> 
> 
> On 14 December 2010 01:18, Madovsky <infos at madovsky.org> wrote:
> > because I need to use variables inside the api request
> > is it make a difference ?
> >
> > ----- Original Message -----
> > From: "Steven Ayre" <steveayre at gmail.com>
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Sent: Monday, December 13, 2010 7:47 PM
> > Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
> >
> >
> >> What about  uuid_send_dtmf directly? i.e. not through expand.
> >>
> >> -Steve
> >>
> >>
> >> On 13 December 2010 20:04, Madovsky <infos at madovsky.org> wrote:
> >>> yes,
> >>>
> >>> I tried expand uuid_send_dtmf [uuid] 12345 at 120 once
> >>> answer done and ivr asks pin number but the CLI
> >>> says "ERR- no reply".
> >>>
> >>> ----- Original Message -----
> >>> From: Michael Collins
> >>> To: FreeSWITCH Users Help
> >>> Sent: Monday, December 13, 2010 2:50 PM
> >>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
> >>> Have you tried uuid_send_dtmf?
> >>> -MC
> >>>
> >>> On Sun, Dec 12, 2010 at 12:30 PM, Madovsky <infos at madovsky.org> wrote:
> >>>>
> >>>> Is it possible to send dtfmf via
> >>>> send_dtmf from CLI on a one leg bridge ?
> >>>> it's for enter a pin conference and I can't use
> >>>> any RFC on my sip phone.
> >>>>
> >>>> Thanks
> >>>>
> >>>> Franck
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>> ________________________________
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From infos at madovsky.org  Wed Dec 15 05:42:07 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 14 Dec 2010 21:42:07 -0500
Subject: [Freeswitch-users] send dtmf on one leg bridge
References: <EB50CB0701AE4FF5B27938870B853740@e1705>
	<C3401235-C0CD-4589-8CAC-B1DE99D0A1A6@avgs.ca>
Message-ID: <28B5925632F347C89AEA994DB017359B@e1705>

if I send these sdp value at every dtmf send
is FS can interpret it or do I need to set it when session is done ?


----- Original Message ----- 
From: "Mathieu Rene" <mrene_lists at avgs.ca>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, December 14, 2010 9:29 PM
Subject: Re: [Freeswitch-users] send dtmf on one leg bridge


> Indeed it is.
>
> http://www.faqs.org/rfcs/rfc2833.html
>
>  The RTP payload format is designated as "telephone-event", the MIME
>   type as "audio/telephone-event". The default timestamp rate is 8000
>   Hz, but other rates may be defined. In accordance with current
>   practice, this payload format does not have a static payload type
>   number, but uses a RTP payload type number established dynamically
>   and out-of-band.
>
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-12-14, at 9:27 PM, Madovsky wrote:
>
>> is it need any specific sdp for rfc2833 ?
>>
>> variable_switch_r_sdp: [v=0
>> o=- 1292379777 1292379777 IN IP4 67.205.XX.XX
>> s=-
>> c=IN IP4 67.205.XX.XX
>> t=0 0
>> m=audio 46566 RTP/AVP 96
>> a=rtpmap:96 speex/16000
>> m=video 53492 RTP/AVP 97
>> ]
>>
>> seems missing
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>>
>> does dtmf need it ?
>>
>> Thanks
>>
>>
>>
>> ----- Original Message -----
>> From: Madovsky
>> To: FreeSWITCH Users Help
>> Sent: Tuesday, December 14, 2010 8:09 PM
>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
>>
>> Ok I will try. Bingo Brian,
>> the rfc2833 RTP payload is not interpreted by FS
>> I have to debug it.
>> ----- Original Message -----
>> From: Michael Collins
>> To: FreeSWITCH Users Help
>> Sent: Tuesday, December 14, 2010 3:32 AM
>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
>>
>> Per my other message you may also just need to use uuid_recv_dtmf. Try it 
>> and see.
>> -MC
>>
>> On Mon, Dec 13, 2010 at 11:12 PM, Steven Ayre <steveayre at gmail.com> 
>> wrote:
>> Perhaps the UUID variable doesn't contain the correct value though...
>>
>> Try uuid_send_dtmf using the known UUID yourself. That'll show you
>> whether uuid_send_dtmf works and the problem is in the expand api, or
>> whether it's uuid_send_dtmf that's having a problem.
>>
>> -Steve
>>
>>
>> On 14 December 2010 01:18, Madovsky <infos at madovsky.org> wrote:
>> > because I need to use variables inside the api request
>> > is it make a difference ?
>> >
>> > ----- Original Message -----
>> > From: "Steven Ayre" <steveayre at gmail.com>
>> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> > Sent: Monday, December 13, 2010 7:47 PM
>> > Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
>> >
>> >
>> >> What about  uuid_send_dtmf directly? i.e. not through expand.
>> >>
>> >> -Steve
>> >>
>> >>
>> >> On 13 December 2010 20:04, Madovsky <infos at madovsky.org> wrote:
>> >>> yes,
>> >>>
>> >>> I tried expand uuid_send_dtmf [uuid] 12345 at 120 once
>> >>> answer done and ivr asks pin number but the CLI
>> >>> says "ERR- no reply".
>> >>>
>> >>> ----- Original Message -----
>> >>> From: Michael Collins
>> >>> To: FreeSWITCH Users Help
>> >>> Sent: Monday, December 13, 2010 2:50 PM
>> >>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
>> >>> Have you tried uuid_send_dtmf?
>> >>> -MC
>> >>>
>> >>> On Sun, Dec 12, 2010 at 12:30 PM, Madovsky <infos at madovsky.org> 
>> >>> wrote:
>> >>>>
>> >>>> Is it possible to send dtfmf via
>> >>>> send_dtmf from CLI on a one leg bridge ?
>> >>>> it's for enter a pin conference and I can't use
>> >>>> any RFC on my sip phone.
>> >>>>
>> >>>> Thanks
>> >>>>
>> >>>> Franck
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>
>> >>> ________________________________
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From mrene_lists at avgs.ca  Wed Dec 15 07:04:15 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 14 Dec 2010 23:04:15 -0500
Subject: [Freeswitch-users] send dtmf on one leg bridge
In-Reply-To: <28B5925632F347C89AEA994DB017359B@e1705>
References: <EB50CB0701AE4FF5B27938870B853740@e1705>
	<C3401235-C0CD-4589-8CAC-B1DE99D0A1A6@avgs.ca>
	<28B5925632F347C89AEA994DB017359B@e1705>
Message-ID: <0A60C632-A7E4-45AA-B8FB-3C8663421B86@avgs.ca>

I suggest you read the SIP RFC too. The SDP isn't sent at every DTMF digit, it is only exchanged in an offer-answer scenario (e.g.: INVITE/18x/200).

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-12-14, at 9:42 PM, Madovsky wrote:

> if I send these sdp value at every dtmf send
> is FS can interpret it or do I need to set it when session is done ?
> 
> 
> ----- Original Message ----- 
> From: "Mathieu Rene" <mrene_lists at avgs.ca>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, December 14, 2010 9:29 PM
> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
> 
> 
>> Indeed it is.
>> 
>> http://www.faqs.org/rfcs/rfc2833.html
>> 
>> The RTP payload format is designated as "telephone-event", the MIME
>>  type as "audio/telephone-event". The default timestamp rate is 8000
>>  Hz, but other rates may be defined. In accordance with current
>>  practice, this payload format does not have a static payload type
>>  number, but uses a RTP payload type number established dynamically
>>  and out-of-band.
>> 
>> 
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>> 
>> 
>> 
>> 
>> On 2010-12-14, at 9:27 PM, Madovsky wrote:
>> 
>>> is it need any specific sdp for rfc2833 ?
>>> 
>>> variable_switch_r_sdp: [v=0
>>> o=- 1292379777 1292379777 IN IP4 67.205.XX.XX
>>> s=-
>>> c=IN IP4 67.205.XX.XX
>>> t=0 0
>>> m=audio 46566 RTP/AVP 96
>>> a=rtpmap:96 speex/16000
>>> m=video 53492 RTP/AVP 97
>>> ]
>>> 
>>> seems missing
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> 
>>> does dtmf need it ?
>>> 
>>> Thanks
>>> 
>>> 
>>> 
>>> ----- Original Message -----
>>> From: Madovsky
>>> To: FreeSWITCH Users Help
>>> Sent: Tuesday, December 14, 2010 8:09 PM
>>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
>>> 
>>> Ok I will try. Bingo Brian,
>>> the rfc2833 RTP payload is not interpreted by FS
>>> I have to debug it.
>>> ----- Original Message -----
>>> From: Michael Collins
>>> To: FreeSWITCH Users Help
>>> Sent: Tuesday, December 14, 2010 3:32 AM
>>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
>>> 
>>> Per my other message you may also just need to use uuid_recv_dtmf. Try it 
>>> and see.
>>> -MC
>>> 
>>> On Mon, Dec 13, 2010 at 11:12 PM, Steven Ayre <steveayre at gmail.com> 
>>> wrote:
>>> Perhaps the UUID variable doesn't contain the correct value though...
>>> 
>>> Try uuid_send_dtmf using the known UUID yourself. That'll show you
>>> whether uuid_send_dtmf works and the problem is in the expand api, or
>>> whether it's uuid_send_dtmf that's having a problem.
>>> 
>>> -Steve
>>> 
>>> 
>>> On 14 December 2010 01:18, Madovsky <infos at madovsky.org> wrote:
>>>> because I need to use variables inside the api request
>>>> is it make a difference ?
>>>> 
>>>> ----- Original Message -----
>>>> From: "Steven Ayre" <steveayre at gmail.com>
>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, December 13, 2010 7:47 PM
>>>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
>>>> 
>>>> 
>>>>> What about  uuid_send_dtmf directly? i.e. not through expand.
>>>>> 
>>>>> -Steve
>>>>> 
>>>>> 
>>>>> On 13 December 2010 20:04, Madovsky <infos at madovsky.org> wrote:
>>>>>> yes,
>>>>>> 
>>>>>> I tried expand uuid_send_dtmf [uuid] 12345 at 120 once
>>>>>> answer done and ivr asks pin number but the CLI
>>>>>> says "ERR- no reply".
>>>>>> 
>>>>>> ----- Original Message -----
>>>>>> From: Michael Collins
>>>>>> To: FreeSWITCH Users Help
>>>>>> Sent: Monday, December 13, 2010 2:50 PM
>>>>>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
>>>>>> Have you tried uuid_send_dtmf?
>>>>>> -MC
>>>>>> 
>>>>>> On Sun, Dec 12, 2010 at 12:30 PM, Madovsky <infos at madovsky.org> 
>>>>>> wrote:
>>>>>>> 
>>>>>>> Is it possible to send dtfmf via
>>>>>>> send_dtmf from CLI on a one leg bridge ?
>>>>>>> it's for enter a pin conference and I can't use
>>>>>>> any RFC on my sip phone.
>>>>>>> 
>>>>>>> Thanks
>>>>>>> 
>>>>>>> Franck
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>> 
>>>>>> ________________________________
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From infos at madovsky.org  Wed Dec 15 07:16:39 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 14 Dec 2010 23:16:39 -0500
Subject: [Freeswitch-users] send dtmf on one leg bridge
References: <EB50CB0701AE4FF5B27938870B853740@e1705><C3401235-C0CD-4589-8CAC-B1DE99D0A1A6@avgs.ca><28B5925632F347C89AEA994DB017359B@e1705>
	<0A60C632-A7E4-45AA-B8FB-3C8663421B86@avgs.ca>
Message-ID: <3E3D46F1D8534D9185641CA8AE0B442C@e1705>

ok understood, the DTMF seems to be ok in RTP flow,
but FS doesn't react. maybe the RTP packet is not clean...

----- Original Message ----- 
From: "Mathieu Rene" <mrene_lists at avgs.ca>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, December 14, 2010 11:04 PM
Subject: Re: [Freeswitch-users] send dtmf on one leg bridge


>I suggest you read the SIP RFC too. The SDP isn't sent at every DTMF digit, 
>it is only exchanged in an offer-answer scenario (e.g.: INVITE/18x/200).
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-12-14, at 9:42 PM, Madovsky wrote:
>
>> if I send these sdp value at every dtmf send
>> is FS can interpret it or do I need to set it when session is done ?
>>
>>
>> ----- Original Message ----- 
>> From: "Mathieu Rene" <mrene_lists at avgs.ca>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, December 14, 2010 9:29 PM
>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
>>
>>
>>> Indeed it is.
>>>
>>> http://www.faqs.org/rfcs/rfc2833.html
>>>
>>> The RTP payload format is designated as "telephone-event", the MIME
>>>  type as "audio/telephone-event". The default timestamp rate is 8000
>>>  Hz, but other rates may be defined. In accordance with current
>>>  practice, this payload format does not have a static payload type
>>>  number, but uses a RTP payload type number established dynamically
>>>  and out-of-band.
>>>
>>>
>>> Mathieu Rene
>>> Avant-Garde Solutions Inc
>>> Office: + 1 (514) 664-1044 x100
>>> Cell: +1 (514) 664-1044 x200
>>> mrene at avgs.ca
>>>
>>>
>>>
>>>
>>> On 2010-12-14, at 9:27 PM, Madovsky wrote:
>>>
>>>> is it need any specific sdp for rfc2833 ?
>>>>
>>>> variable_switch_r_sdp: [v=0
>>>> o=- 1292379777 1292379777 IN IP4 67.205.XX.XX
>>>> s=-
>>>> c=IN IP4 67.205.XX.XX
>>>> t=0 0
>>>> m=audio 46566 RTP/AVP 96
>>>> a=rtpmap:96 speex/16000
>>>> m=video 53492 RTP/AVP 97
>>>> ]
>>>>
>>>> seems missing
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>>
>>>> does dtmf need it ?
>>>>
>>>> Thanks
>>>>
>>>>
>>>>
>>>> ----- Original Message -----
>>>> From: Madovsky
>>>> To: FreeSWITCH Users Help
>>>> Sent: Tuesday, December 14, 2010 8:09 PM
>>>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
>>>>
>>>> Ok I will try. Bingo Brian,
>>>> the rfc2833 RTP payload is not interpreted by FS
>>>> I have to debug it.
>>>> ----- Original Message -----
>>>> From: Michael Collins
>>>> To: FreeSWITCH Users Help
>>>> Sent: Tuesday, December 14, 2010 3:32 AM
>>>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
>>>>
>>>> Per my other message you may also just need to use uuid_recv_dtmf. Try 
>>>> it
>>>> and see.
>>>> -MC
>>>>
>>>> On Mon, Dec 13, 2010 at 11:12 PM, Steven Ayre <steveayre at gmail.com>
>>>> wrote:
>>>> Perhaps the UUID variable doesn't contain the correct value though...
>>>>
>>>> Try uuid_send_dtmf using the known UUID yourself. That'll show you
>>>> whether uuid_send_dtmf works and the problem is in the expand api, or
>>>> whether it's uuid_send_dtmf that's having a problem.
>>>>
>>>> -Steve
>>>>
>>>>
>>>> On 14 December 2010 01:18, Madovsky <infos at madovsky.org> wrote:
>>>>> because I need to use variables inside the api request
>>>>> is it make a difference ?
>>>>>
>>>>> ----- Original Message -----
>>>>> From: "Steven Ayre" <steveayre at gmail.com>
>>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>>> Sent: Monday, December 13, 2010 7:47 PM
>>>>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
>>>>>
>>>>>
>>>>>> What about  uuid_send_dtmf directly? i.e. not through expand.
>>>>>>
>>>>>> -Steve
>>>>>>
>>>>>>
>>>>>> On 13 December 2010 20:04, Madovsky <infos at madovsky.org> wrote:
>>>>>>> yes,
>>>>>>>
>>>>>>> I tried expand uuid_send_dtmf [uuid] 12345 at 120 once
>>>>>>> answer done and ivr asks pin number but the CLI
>>>>>>> says "ERR- no reply".
>>>>>>>
>>>>>>> ----- Original Message -----
>>>>>>> From: Michael Collins
>>>>>>> To: FreeSWITCH Users Help
>>>>>>> Sent: Monday, December 13, 2010 2:50 PM
>>>>>>> Subject: Re: [Freeswitch-users] send dtmf on one leg bridge
>>>>>>> Have you tried uuid_send_dtmf?
>>>>>>> -MC
>>>>>>>
>>>>>>> On Sun, Dec 12, 2010 at 12:30 PM, Madovsky <infos at madovsky.org>
>>>>>>> wrote:
>>>>>>>>
>>>>>>>> Is it possible to send dtfmf via
>>>>>>>> send_dtmf from CLI on a one leg bridge ?
>>>>>>>> it's for enter a pin conference and I can't use
>>>>>>>> any RFC on my sip phone.
>>>>>>>>
>>>>>>>> Thanks
>>>>>>>>
>>>>>>>> Franck
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>> ________________________________
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From rajkumar.kmry at gmail.com  Tue Dec 14 12:19:48 2010
From: rajkumar.kmry at gmail.com (Rajkumar K)
Date: Tue, 14 Dec 2010 14:49:48 +0530
Subject: [Freeswitch-users] priority call queue using mod_fifo
Message-ID: <AANLkTik-Q5e4vGmav0o00A8D9C3iNbbBzv47HGTAc3gv@mail.gmail.com>

Hi,

I would like to implement the priority queue using mod_fifo. Please provide
the following informations.
1. How to set the priority for the caller or consumer (who should be served
first)?
2. Is it possible to choose the agent with different methods such as round
robin (before bridging the call)?
3. How can we control the call flow after bridging the call with agent (for
call recording and other options)?
4. Is there any detailed documentation available for mod_fifo module.

regards
rajkumar k
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From jerre at j-cope.com  Wed Dec 15 08:31:08 2010
From: jerre at j-cope.com (Jerre Cope)
Date: Tue, 14 Dec 2010 23:31:08 -0600
Subject: [Freeswitch-users] Incomplete offer/answer debugging
References: 4D0716E4.6040501@j-cope.com
Message-ID: <4D08529C.1040304@j-cope.com>

Thank MC!

I was looking hard at that line, but didn't know what to do about it.  I 
will start from scratch to get a working extension, then check the codec 
lines of the previous version(s) of my conf to see where I fell off the 
wagon.  I was at one point, attempting to limit the codec choices for 
the little wifi phone.  Since I only know how to spell codec, I should 
probably just let Freeswitch decide.

I'm sure you're right because when I started down the FS experiment, FS 
made the phone reliably ring every time, as opposed making the little 
phone negotiate the nat and the net by itself.

Mighty fine book by the way.  It really helped with perspective to know 
where to start.

Thanks again.


From marcdecorny at gmail.com  Wed Dec 15 09:59:51 2010
From: marcdecorny at gmail.com (Marc de Corny)
Date: Wed, 15 Dec 2010 06:59:51 +0000
Subject: [Freeswitch-users] Lua not playing any wav files
Message-ID: <AANLkTimHrCzzzg8S=PVTVDmH1sd4iv4jQiMFbFHR7teo@mail.gmail.com>

Hi all,

I have run into an issue on something so basic that I must be as simple as
enabling a feature somewhere.

I have been trying to get lua to play a message from a WAV file. I have
tried session:execute("playback", main_msg) and
session:streamFile(ivr_invalid_msg) but neither of them play any music to
the caller. I tried both to answer and preAnswer the call first but it made
no difference. However if I put the same file into the XML dialplan and play
it with the commands below I hear the music fine.
<action application="set" data="playback_terminators=#"/>
<action application="playback" data="${401_sound_file_path}"/>

The issue only seems to be from lua when playing any type of wav file and
those files are definitelly there as can be read by the XML

The error message is below for the execute(playback) command, but nothing
can be seen for the
2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application
playback Requires media! pre_answering channel
sofia/external/2031701665 at 194.0.147.16:5060 EXECUTE
sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswitch/sounds/svc_sound_files/default_autoattendant.wav)
But there is no mention of the streamFile command. I have had similar issue
with the PlayAndGetDigits command.
Is there something that I need to enable in lua so that is can playback
messages to the caller.

Many thanks to anyone who can help.
Marc


below is the XML dialplan and lua script as well as the log at the very end.

XML DIALPLAN:
<include>
<extension name="IVR_FROM_MYSQL">
<condition field="caller_id_number" expression="Anonymous" break="never">
<action application="set" data="effective_caller_id_number=0000000000"/>
</condition>
<condition field="destination_number" expression="^(404)">
<action inline="true" application="lua" data="ivr_mysql.lua
${destination_number:3}"/>
<action application="set" data="effective_caller_id_name=${404_tag}"/>
<action application="bridge" data="${404_dial}"/>
</condition>
</extension>
</include>

The LUA script ivr_mysql.lua is callsed and this is it.
-- IVR : PLAY IVR WAV FILES
-- Global Variables:
local dialstr_prefix = "sofia/gateway/CS2k/"
local dialstr_main = ""
local breakoutcode = "184"
local sound_file_folder = "/usr/local/freeswitch/sounds/svc_sound_files/"
local ddi = argv[1]
-- answer the call
session:preAnswer();
freeswitch.consoleLog("info", "All Answered\n");
ivr_invalid_msg = sound_file_folder .. "invalid_msg.wav"
main_msg = sound_file_folder .. "default_autoattendant.wav"
-- Play with Execute
session:execute("playback", main_msg)
-- Play with StreamFile
session:streamFile(ivr_invalid_msg);
dialstr_main  = dialstr_main .. dialstr_prefix .. breakoutcode ..
"02031701665"
session:setVariable("404_dial",dialstr_main)
session:setVariable("404_tag","IVR")


RELEVANT LOGS :
Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Regex (PASS)
[IVR_FROM_MYS                          QL] destination_number(4042031956241)
=~ /^(404)/ break=on-false
Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
lua(ivr_mysql.lua $                          {destination_number:3}) INLINE
EXECUTE sofia/external/2031701665 at 194.0.147.16:5060 lua(ivr_mysql.lua
2031956241                          )
2010-12-01 11:56:01.525426 [INFO] switch_cpp.cpp:584 Sending early media
2010-12-01 11:56:01.525426 [DEBUG] sofia_glue.c:2972 AUDIO RTP
[sofia/external/2                          031701665 at 194.0.147.16:5060]
10.5.2.105 port 29900 -> 194.0.147.164 port 50202 c
odec: 8 ms: 20
2010-12-01 11:56:01.525426 [DEBUG] switch_rtp.c:1418 Starting timer [soft]
160 b                          ytes per 20ms
2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3190 Set 2833 dtmf send
payload                           to 101
2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3195 Set 2833 dtmf receive
paylo                          ad to 101
2010-12-01 11:56:01.532280 [DEBUG] mod_sofia.c:2172 Ring SDP:
v=0
o=FreeSWITCH 1291174661 1291174662 IN IP4 10.5.2.105
s=FreeSWITCH
c=IN IP4 10.5.2.105
t=0 0
m=audio 29900 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
2010-12-01 11:56:01.532280 [NOTICE] mod_sofia.c:2175 Pre-Answer
sofia/external/2                          031701665 at 194.0.147.16:5060!
2010-12-01 11:56:01.532280 [DEBUG] switch_channel.c:2544
(sofia/external/2031701                          665 at 194.0.147.16:5060)
Callstate Change RINGING -> EARLY
2010-12-01 11:56:01.534727 [DEBUG] sofia.c:4576 Channel
sofia/external/203170166
5 at 194.0.147.16:5060skipping state [early][183]
2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:676 Send signal
sofia/e                          xternal/2031701665 at 194.0.147.16:5060[BREAK]
2010-12-01 11:56:01.534727 [INFO] switch_cpp.cpp:1181 All Answered
2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application
playba                          ck Requires media! pre_answering channel
sofia/external/2031701665 at 194.0.147.16:                          5060
EXECUTE sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswit
ch/sounds/svc_sound_files/default_autoattendant.wav)
2010-12-01 11:56:01.537644 [DEBUG] switch_cpp.cpp:972
sofia/external/2031701665@
194.0.147.16:5060destroy/unlink session from object
Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
set(effective_calle                          r_id_name=${404_tag})
Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
bridge(${404_dial})
2010-12-01 11:56:01.537644 [DEBUG] switch_core_state_machine.c:119
(sofia/extern                          al/2031701665 at 194.0.147.16:5060)
State Change CS_ROUTING -> CS_EXECUTE
2010-12-01 11:56:01.537644 [DEBUG] switch_core_session.c:1057 Send signal
sofia/                          external/2031701665 at 194.0.147.16:5060[BREAK]
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From u2nsam at gmail.com  Tue Dec 14 12:43:51 2010
From: u2nsam at gmail.com (samir)
Date: Tue, 14 Dec 2010 01:43:51 -0800 (PST)
Subject: [Freeswitch-users] collecting dtmf digits
Message-ID: <1292319831757-5833942.post@n2.nabble.com>


hello,

Is there any method to collect digits by a variable clause ?

Suppose i have a ivr playing and user inputs digits , i want to collect the
dtmf digits and send it to a different application where that digits will be
used for routing purpose.

Any ideas !

Regards
Sam
-- 
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From u2nsam at gmail.com  Wed Dec 15 07:18:39 2010
From: u2nsam at gmail.com (samir)
Date: Tue, 14 Dec 2010 20:18:39 -0800 (PST)
Subject: [Freeswitch-users] routing via hangup_cause
Message-ID: <1292386719584-5837136.post@n2.nabble.com>


Hello friends,

was trying to create a routing rule to to route calls by accounting to the
hangup causes,

I have written below syntax but it fails to give the cause code to the
varriable for routing.

<extension name="refer">
               <condition field="destination_number" expression="^1234$">


       <action application="bridge" data="sofia/external/1234 at X.X.X.X"/>
       <action application="set" data="hangup_after_bridge=false"/>
       <action application="set" data="continue_on_fail=true"/>
       <action application="log" data="1 B-leg hangup cause:
${bridge_hangup_cause}"/>
       <action application="transfer" data="${bridge_hangup_cause} XML
conbridge"/>

               </condition>


   </extension>

       <extension name="no user">
               <condition field="destination_number"
expression="^(NO_USER_RESPONSE)$">
               <action application="bridge"
data="sofia/external/4567 at X.X.X.X"/>

               </condition>
       </extension>


here the ${bridge_hangup_cause} is not getting executed. Am I doing it right
or is there any other way to do it.

Regards
Sam
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From u2nsam at gmail.com  Wed Dec 15 07:21:47 2010
From: u2nsam at gmail.com (samir)
Date: Tue, 14 Dec 2010 20:21:47 -0800 (PST)
Subject: [Freeswitch-users] collecting DTMF digits
Message-ID: <1292386907862-5837142.post@n2.nabble.com>


hello,

Is there any method to collect digits by a variable clause ?

Suppose i have a ivr playing and user inputs digits , i want to collect the
dtmf digits and send it to a different application where that digits will be
used for routing purpose.

Any ideas !

Regards
Sam 
-- 
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Sent from the freeswitch-users mailing list archive at Nabble.com.


From u2nsam at gmail.com  Wed Dec 15 08:42:53 2010
From: u2nsam at gmail.com (samir)
Date: Tue, 14 Dec 2010 21:42:53 -0800 (PST)
Subject: [Freeswitch-users] call pickup
Message-ID: <1292391773589-5837233.post@n2.nabble.com>


hello,

I am trying call "pickup" for the incoming calls with intercept function but
it do not seems to work, am i missing something ?


		
         
	 
     	 
     
		
  

Regards
Sam
-- 
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From u2nsam at gmail.com  Wed Dec 15 08:51:43 2010
From: u2nsam at gmail.com (samir)
Date: Tue, 14 Dec 2010 21:51:43 -0800 (PST)
Subject: [Freeswitch-users] call pickup
Message-ID: <1292392303190-5837242.post@n2.nabble.com>


Hello folks,

I am trying call pickup for incoming calls by intercept function, its not
working , am i missing something ?


<extension name="group-intercept">
		<condition field="destination_number" expression="^08$">
     <action application="answer"/>
	 <action application="set" data="intercept_unbridged_only=true"/>
     
	 <action application="intercept"
data="${hash(select/${domain_name}-last_dial_ext/${callgroup})}"/>
     <action application="sleep" data="2000"/>
		</condition>
  </extension>


Regds
Sam
-- 
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From u2nsam at gmail.com  Wed Dec 15 08:57:11 2010
From: u2nsam at gmail.com (Sam)
Date: Wed, 15 Dec 2010 11:27:11 +0530
Subject: [Freeswitch-users] call pickup/ DTMF / routing
Message-ID: <AANLkTikat1KDNDxufU=6bo102hoM+74F32bCAFU7hkkB@mail.gmail.com>

Hello folks,

###
call pickup

I am trying call pickup for incoming calls by intercept function, its not
working , am i missing something ?


<extension name="group-intercept">
                <condition field="destination_number" expression="^08$">
     <action application="answer"/>
         <action application="set" data="intercept_unbridged_only=true"/>

         <action application="intercept"
data="${hash(select/${domain_name}-last_dial_ext/${callgroup})}"/>
     <action application="sleep" data="2000"/>
                </condition>
  </extension>




###
collecting DTMF digits

Is there any method to collect digits by a variable clause ?

Suppose i have a ivr playing and user inputs digits , i want to collect the
dtmf digits and send it to a different application where that digits will be
used for routing purpose.

Any ideas !





###
routing via hangup_cause
was trying to create a routing rule to to route calls by accounting to the
hangup causes,

I have written below syntax but it fails to give the cause code to the
varriable for routing.

<extension name="refer">
               <condition field="destination_number" expression="^1234$">


       <action application="bridge" data="sofia/external/1234 at X.X.X.X"/>
       <action application="set" data="hangup_after_bridge=false"/>
       <action application="set" data="continue_on_fail=true"/>
       <action application="log" data="1 B-leg hangup cause:
${bridge_hangup_cause}"/>
       <action application="transfer" data="${bridge_hangup_cause} XML
conbridge"/>

               </condition>


   </extension>

       <extension name="no user">
               <condition field="destination_number"
expression="^(NO_USER_RESPONSE)$">
               <action application="bridge"
data="sofia/external/4567 at X.X.X.X"/>

               </condition>
       </extension>


here the ${bridge_hangup_cause} is not getting executed. Am I doing it right

or is there any other way to do it.
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From david.ponzone at ipeva.fr  Wed Dec 15 13:17:23 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 15 Dec 2010 11:17:23 +0100
Subject: [Freeswitch-users] call pickup
In-Reply-To: <1292391773589-5837233.post@n2.nabble.com>
References: <1292391773589-5837233.post@n2.nabble.com>
Message-ID: <665DE8BC-65A7-46D2-8F8E-002649C59C21@ipeva.fr>

Samir,

Perhaps next time, you could think before sending your mail.
It will avoid you sending multiple emails for the same question.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 15/12/2010 ? 06:42, samir a ?crit :

> hello, I am trying call "pickup" for the incoming calls with intercept function but it do not seems to work, am i missing something ? Regards Sam 
> View this message in context: call pickup
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mustafa.pk at gmail.com  Wed Dec 15 13:18:12 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Wed, 15 Dec 2010 15:18:12 +0500
Subject: [Freeswitch-users] call pickup/ DTMF / routing
In-Reply-To: <AANLkTikat1KDNDxufU=6bo102hoM+74F32bCAFU7hkkB@mail.gmail.com>
References: <AANLkTikat1KDNDxufU=6bo102hoM+74F32bCAFU7hkkB@mail.gmail.com>
Message-ID: <AANLkTimw9em9mdA0McQsc1zaPt7crZeT4TedKaLURjNk@mail.gmail.com>

I would suggest you to read FreeSWITCH wiki docs before posting at this
mailing list.

All of your questions lack basic knowledge of FreeSWITCH and are already
addressed in the wiki.

Also when you post please be patient and post one question at a time instead
of publishing a digest on the mailing list.

1. A simple search on google for "freeswitch call pickup" takes you to the
following page.

http://wiki.freeswitch.org/wiki/User:Agx#Call_Pickup


2. Follow instructions on this page for collecting dtmf digits.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits

3. make sure context conbridge exists.


Regards,


On Wed, Dec 15, 2010 at 10:57 AM, Sam <u2nsam at gmail.com> wrote:

> Hello folks,
>
> ###
> call pickup
>
> I am trying call pickup for incoming calls by intercept function, its not
> working , am i missing something ?
>
>
> <extension name="group-intercept">
>                 <condition field="destination_number" expression="^08$">
>      <action application="answer"/>
>          <action application="set" data="intercept_unbridged_only=true"/>
>
>          <action application="intercept"
> data="${hash(select/${domain_name}-last_dial_ext/${callgroup})}"/>
>      <action application="sleep" data="2000"/>
>                 </condition>
>   </extension>
>
>
>
>
> ###
> collecting DTMF digits
>
> Is there any method to collect digits by a variable clause ?
>
> Suppose i have a ivr playing and user inputs digits , i want to collect the
> dtmf digits and send it to a different application where that digits will be
> used for routing purpose.
>
> Any ideas !
>
>
>
>
>
> ###
> routing via hangup_cause
> was trying to create a routing rule to to route calls by accounting to the
> hangup causes,
>
> I have written below syntax but it fails to give the cause code to the
> varriable for routing.
>
> <extension name="refer">
>                <condition field="destination_number" expression="^1234$">
>
>
>        <action application="bridge" data="sofia/external/1234 at X.X.X.X"/>
>        <action application="set" data="hangup_after_bridge=false"/>
>        <action application="set" data="continue_on_fail=true"/>
>        <action application="log" data="1 B-leg hangup cause:
> ${bridge_hangup_cause}"/>
>        <action application="transfer" data="${bridge_hangup_cause} XML
> conbridge"/>
>
>                </condition>
>
>
>    </extension>
>
>        <extension name="no user">
>                <condition field="destination_number"
> expression="^(NO_USER_RESPONSE)$">
>                <action application="bridge"
> data="sofia/external/4567 at X.X.X.X"/>
>
>                </condition>
>        </extension>
>
>
> here the ${bridge_hangup_cause} is not getting executed. Am I doing it
> right
> or is there any other way to do it.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com
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From fdelawarde at wirelessmundi.com  Wed Dec 15 13:55:19 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Wed, 15 Dec 2010 11:55:19 +0100
Subject: [Freeswitch-users] error compiling today's git
Message-ID: <1292410519.23335.33.camel@luna.tc.commsmundi.com>

Am I doing something wrong?


Compiling src/switch_rtp.c ...
cc1: warnings being treated as errors
src/switch_rtp.c: In function ?jb_logger?:
src/switch_rtp.c:1662: error: implicit declaration of function ?stfu_vasprintf?
src/switch_rtp.c: In function ?switch_rtp_debug_jitter_buffer?:
src/switch_rtp.c:1675: error: implicit declaration of function ?stfu_n_debug?
src/switch_rtp.c:1676: error: implicit declaration of function ?stfu_global_set_logger?
src/switch_rtp.c:2141:62: error: macro "stfu_n_eat" requires 6 arguments, but only 5 given
src/switch_rtp.c: In function ?do_flush?:
src/switch_rtp.c:2139: error: ?stfu_n_eat? undeclared (first use in this function)
src/switch_rtp.c:2139: error: (Each undeclared identifier is reported only once
src/switch_rtp.c:2139: error: for each function it appears in.)
src/switch_rtp.c:2207:59: error: macro "stfu_n_eat" requires 6 arguments, but only 5 given
src/switch_rtp.c: In function ?read_rtp_packet?:
src/switch_rtp.c:2205: error: ?stfu_n_eat? undeclared (first use in this function)
make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1


Fran?ois.



From peter.olsson at visionutveckling.se  Wed Dec 15 14:20:40 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 15 Dec 2010 12:20:40 +0100
Subject: [Freeswitch-users] error compiling today's git
In-Reply-To: <1292410519.23335.33.camel@luna.tc.commsmundi.com>
References: <1292410519.23335.33.camel@luna.tc.commsmundi.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57ECCA2836@cooper>

I think Anthony might have forgotten to push some changes to GIT yesterday - I just noticed the same thing. Please add a Jira for this, so it's reported correctly.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Fran?ois Delawarde
Skickat: den 15 december 2010 11:55
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] error compiling today's git

Am I doing something wrong?


Compiling src/switch_rtp.c ...
cc1: warnings being treated as errors
src/switch_rtp.c: In function ?jb_logger?:
src/switch_rtp.c:1662: error: implicit declaration of function ?stfu_vasprintf?
src/switch_rtp.c: In function ?switch_rtp_debug_jitter_buffer?:
src/switch_rtp.c:1675: error: implicit declaration of function ?stfu_n_debug?
src/switch_rtp.c:1676: error: implicit declaration of function ?stfu_global_set_logger?
src/switch_rtp.c:2141:62: error: macro "stfu_n_eat" requires 6 arguments, but only 5 given
src/switch_rtp.c: In function ?do_flush?:
src/switch_rtp.c:2139: error: ?stfu_n_eat? undeclared (first use in this function)
src/switch_rtp.c:2139: error: (Each undeclared identifier is reported only once
src/switch_rtp.c:2139: error: for each function it appears in.)
src/switch_rtp.c:2207:59: error: macro "stfu_n_eat" requires 6 arguments, but only 5 given
src/switch_rtp.c: In function ?read_rtp_packet?:
src/switch_rtp.c:2205: error: ?stfu_n_eat? undeclared (first use in this function)
make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1


Fran?ois.


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

!DSPAM:4d08a00932769772718559!


From steveayre at gmail.com  Wed Dec 15 14:25:35 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 15 Dec 2010 11:25:35 +0000
Subject: [Freeswitch-users] error compiling today's git
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57ECCA2836@cooper>
References: <1292410519.23335.33.camel@luna.tc.commsmundi.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57ECCA2836@cooper>
Message-ID: <AANLkTi=1mMK+cMBRa5HW_P3g2pNosMPn6Gg6mdC=9s3p@mail.gmail.com>

I've just tested on a fresh checkout and can confirm this too.

stfu_vasprintf is used in switch_rtp.c but isn't referenced anywhere
else in code, so yes looks like a file wasn't committed.

-Steve


On 15 December 2010 11:20, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> I think Anthony might have forgotten to push some changes to GIT yesterday - I just noticed the same thing. Please add a Jira for this, so it's reported correctly.
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Fran?ois Delawarde
> Skickat: den 15 december 2010 11:55
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] error compiling today's git
>
> Am I doing something wrong?
>
>
> Compiling src/switch_rtp.c ...
> cc1: warnings being treated as errors
> src/switch_rtp.c: In function ?jb_logger?:
> src/switch_rtp.c:1662: error: implicit declaration of function ?stfu_vasprintf?
> src/switch_rtp.c: In function ?switch_rtp_debug_jitter_buffer?:
> src/switch_rtp.c:1675: error: implicit declaration of function ?stfu_n_debug?
> src/switch_rtp.c:1676: error: implicit declaration of function ?stfu_global_set_logger?
> src/switch_rtp.c:2141:62: error: macro "stfu_n_eat" requires 6 arguments, but only 5 given
> src/switch_rtp.c: In function ?do_flush?:
> src/switch_rtp.c:2139: error: ?stfu_n_eat? undeclared (first use in this function)
> src/switch_rtp.c:2139: error: (Each undeclared identifier is reported only once
> src/switch_rtp.c:2139: error: for each function it appears in.)
> src/switch_rtp.c:2207:59: error: macro "stfu_n_eat" requires 6 arguments, but only 5 given
> src/switch_rtp.c: In function ?read_rtp_packet?:
> src/switch_rtp.c:2205: error: ?stfu_n_eat? undeclared (first use in this function)
> make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1
>
>
> Fran?ois.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4d08a00932769772718559!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From fdelawarde at wirelessmundi.com  Wed Dec 15 14:47:06 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Wed, 15 Dec 2010 12:47:06 +0100
Subject: [Freeswitch-users] error compiling today's git
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57ECCA2836@cooper>
References: <1292410519.23335.33.camel@luna.tc.commsmundi.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57ECCA2836@cooper>
Message-ID: <1292413626.23335.46.camel@luna.tc.commsmundi.com>

On Wed, 2010-12-15 at 12:20 +0100, Peter Olsson wrote:
> Please add a Jira for this, so it's reported correctly.

http://jira.freeswitch.org/browse/FS-2932

Fran?ois.



From erik.dekkers at wvds.nl  Wed Dec 15 14:52:32 2010
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Wed, 15 Dec 2010 12:52:32 +0100
Subject: [Freeswitch-users] error compiling today's git
In-Reply-To: <1292410519.23335.33.camel@luna.tc.commsmundi.com>
References: <1292410519.23335.33.camel@luna.tc.commsmundi.com>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE077704C783@FS1.wvds.local>

Here the same

-----Oorspronkelijk bericht-----
Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Fran?ois Delawarde
Verzonden: woensdag 15 december 2010 11:55
Aan: FreeSWITCH Users Help
Onderwerp: [Freeswitch-users] error compiling today's git

Am I doing something wrong?


Compiling src/switch_rtp.c ...
cc1: warnings being treated as errors
src/switch_rtp.c: In function ?jb_logger?:
src/switch_rtp.c:1662: error: implicit declaration of function ?stfu_vasprintf?
src/switch_rtp.c: In function ?switch_rtp_debug_jitter_buffer?:
src/switch_rtp.c:1675: error: implicit declaration of function ?stfu_n_debug?
src/switch_rtp.c:1676: error: implicit declaration of function ?stfu_global_set_logger?
src/switch_rtp.c:2141:62: error: macro "stfu_n_eat" requires 6 arguments, but only 5 given
src/switch_rtp.c: In function ?do_flush?:
src/switch_rtp.c:2139: error: ?stfu_n_eat? undeclared (first use in this function)
src/switch_rtp.c:2139: error: (Each undeclared identifier is reported only once
src/switch_rtp.c:2139: error: for each function it appears in.)
src/switch_rtp.c:2207:59: error: macro "stfu_n_eat" requires 6 arguments, but only 5 given
src/switch_rtp.c: In function ?read_rtp_packet?:
src/switch_rtp.c:2205: error: ?stfu_n_eat? undeclared (first use in this function)
make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1


Fran?ois.


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

From steveayre at gmail.com  Wed Dec 15 15:17:32 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 15 Dec 2010 12:17:32 +0000
Subject: [Freeswitch-users] error compiling today's git
In-Reply-To: <1292410519.23335.33.camel@luna.tc.commsmundi.com>
References: <1292410519.23335.33.camel@luna.tc.commsmundi.com>
Message-ID: <AANLkTinwQynAtpV5UdJX5Y_21QEANQ616yiahOGhMEPf@mail.gmail.com>

For the benefit of everyone now familiar with Git...

If you need to build FS you can check out the previous commit using:

$ git clone git://git.freeswitch.org/freeswitch.git
$ git checkout 2324c299177be42375610c4928a3b77e60a8bf10

You should see:
HEAD is now at 2324c29...  round two better code thanks mikej

You can ignore any messages above that about creating a new branch.

That'll roll the checkout back to the commit before the error was
introduced and you should then be able to build.

-Steve





On 15 December 2010 10:55, Fran?ois Delawarde
<fdelawarde at wirelessmundi.com> wrote:
> Am I doing something wrong?
>
>
> Compiling src/switch_rtp.c ...
> cc1: warnings being treated as errors
> src/switch_rtp.c: In function ?jb_logger?:
> src/switch_rtp.c:1662: error: implicit declaration of function ?stfu_vasprintf?
> src/switch_rtp.c: In function ?switch_rtp_debug_jitter_buffer?:
> src/switch_rtp.c:1675: error: implicit declaration of function ?stfu_n_debug?
> src/switch_rtp.c:1676: error: implicit declaration of function ?stfu_global_set_logger?
> src/switch_rtp.c:2141:62: error: macro "stfu_n_eat" requires 6 arguments, but only 5 given
> src/switch_rtp.c: In function ?do_flush?:
> src/switch_rtp.c:2139: error: ?stfu_n_eat? undeclared (first use in this function)
> src/switch_rtp.c:2139: error: (Each undeclared identifier is reported only once
> src/switch_rtp.c:2139: error: for each function it appears in.)
> src/switch_rtp.c:2207:59: error: macro "stfu_n_eat" requires 6 arguments, but only 5 given
> src/switch_rtp.c: In function ?read_rtp_packet?:
> src/switch_rtp.c:2205: error: ?stfu_n_eat? undeclared (first use in this function)
> make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1
>
>
> Fran?ois.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From acosgrov at gmail.com  Wed Dec 15 17:20:05 2010
From: acosgrov at gmail.com (Anthony Cosgrove)
Date: Wed, 15 Dec 2010 09:20:05 -0500
Subject: [Freeswitch-users] CBeyond interop
In-Reply-To: <BFA72E80-0911-4FC5-8198-58B95C64F15A@freeswitch.org>
References: <BFA72E80-0911-4FC5-8198-58B95C64F15A@freeswitch.org>
Message-ID: <A517EDC5-0656-4C70-A97F-BB0E8412EB5D@gmail.com>

I work for Cbeyond in their new cloud services division (formerly Aretta Communications)... I'll see if I can get a test number assigned so I can log into their SBC. Would you need anything specific besides traces?

Anthony C
acosgrove at aretta.com or anthony.cosgrove at cbeyond.net (in case you want to get me off-list)

On Dec 14, 2010, at 9:23 PM, Brian West wrote:

> Has anyone gotten FreeSWITCH working with CBeyond?  I'm stumped as to what they expect and they don't wanna tell me or share any working traces.
> 
> Thanks,
> Brian
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mranga at gmail.com  Wed Dec 15 17:55:23 2010
From: mranga at gmail.com (M. Ranganathan)
Date: Wed, 15 Dec 2010 09:55:23 -0500
Subject: [Freeswitch-users] CBeyond interop
In-Reply-To: <BFA72E80-0911-4FC5-8198-58B95C64F15A@freeswitch.org>
References: <BFA72E80-0911-4FC5-8198-58B95C64F15A@freeswitch.org>
Message-ID: <AANLkTi==TJnWmk90tGJBUxEoj67rXhtg0seFgzoMcbRz@mail.gmail.com>

On Tue, Dec 14, 2010 at 9:23 PM, Brian West <brian at freeswitch.org> wrote:
> Has anyone gotten FreeSWITCH working with CBeyond? ?I'm stumped as to what they expect and they don't wanna tell me or share any working traces.
>
> Thanks,
> Brian

Recalling from some time ago:

1. REGISTER your SBC (and periodically re-REGISTER )

2. CRLFCRLF NAT keepalive (if behind a NAT)

3. No media keepalive (if behind a NAT)

4. Use public addressing for call setup and SDP ( no server NAT compensation ).

5. No REFER support ( you have to convert to INVITE )

6. No INVITE(Replaces) support for consultative transfer ( you have to
handle in the SBC).

 re-INVITE is supported (with delayed media negotiation ACK(SDP) support )

Things could have changed a bit but that was working for the sipxbridge sbc.

Regards,

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



-- 
M. Ranganathan


From larclap at yahoo.com  Wed Dec 15 17:56:18 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 15 Dec 2010 06:56:18 -0800
Subject: [Freeswitch-users] Compilation error?
Message-ID: <009e01cb9c68$3aa42980$afec7c80$@yahoo.com>

I pulled from git this morning via 'make current' and received the
following:

Compiling src/switch_regex.c ...
Compiling src/switch_rtp.c ...
cc1: warnings being treated as errors
src/switch_rtp.c: In function 'jb_logger':
src/switch_rtp.c:1662: warning: implicit declaration of function
'stfu_vasprintf'
src/switch_rtp.c: In function 'switch_rtp_debug_jitter_buffer':
src/switch_rtp.c:1675: warning: implicit declaration of function
'stfu_n_debug'
src/switch_rtp.c:1676: warning: implicit declaration of function
'stfu_global_set_logger'
src/switch_rtp.c:2141:62: error: macro "stfu_n_eat" requires 6 arguments,
but only 5 given
src/switch_rtp.c: In function 'do_flush':
src/switch_rtp.c:2139: error: 'stfu_n_eat' undeclared (first use in this
function)
src/switch_rtp.c:2139: error: (Each undeclared identifier is reported only
once
src/switch_rtp.c:2139: error: for each function it appears in.)
src/switch_rtp.c:2207:59: error: macro "stfu_n_eat" requires 6 arguments,
but only 5 given
src/switch_rtp.c: In function 'read_rtp_packet':
src/switch_rtp.c:2205: error: 'stfu_n_eat' undeclared (first use in this
function)
make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1
make[2]: Leaving directory `/usr/src/freeswitch_git'
make[1]: *** [all] Error 2
make[1]: Leaving directory `/usr/src/freeswitch_git'
make: *** [current] Error 2

Have I done something wrong?

Thanks, Lars



From steveayre at gmail.com  Wed Dec 15 18:31:32 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 15 Dec 2010 15:31:32 +0000
Subject: [Freeswitch-users] Compilation error?
In-Reply-To: <009e01cb9c68$3aa42980$afec7c80$@yahoo.com>
References: <009e01cb9c68$3aa42980$afec7c80$@yahoo.com>
Message-ID: <AANLkTinAz3vVieTrPppx-Qt0s4XtJNr3+hwiCdQ3XM0w@mail.gmail.com>

This has already been discussed on the list today. Download the latest
Git and it should now be fixed.

-Steve


On 15 December 2010 14:56, Lars Zeb <larclap at yahoo.com> wrote:
> I pulled from git this morning via 'make current' and received the
> following:
>
> Compiling src/switch_regex.c ...
> Compiling src/switch_rtp.c ...
> cc1: warnings being treated as errors
> src/switch_rtp.c: In function 'jb_logger':
> src/switch_rtp.c:1662: warning: implicit declaration of function
> 'stfu_vasprintf'
> src/switch_rtp.c: In function 'switch_rtp_debug_jitter_buffer':
> src/switch_rtp.c:1675: warning: implicit declaration of function
> 'stfu_n_debug'
> src/switch_rtp.c:1676: warning: implicit declaration of function
> 'stfu_global_set_logger'
> src/switch_rtp.c:2141:62: error: macro "stfu_n_eat" requires 6 arguments,
> but only 5 given
> src/switch_rtp.c: In function 'do_flush':
> src/switch_rtp.c:2139: error: 'stfu_n_eat' undeclared (first use in this
> function)
> src/switch_rtp.c:2139: error: (Each undeclared identifier is reported only
> once
> src/switch_rtp.c:2139: error: for each function it appears in.)
> src/switch_rtp.c:2207:59: error: macro "stfu_n_eat" requires 6 arguments,
> but only 5 given
> src/switch_rtp.c: In function 'read_rtp_packet':
> src/switch_rtp.c:2205: error: 'stfu_n_eat' undeclared (first use in this
> function)
> make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1
> make[2]: Leaving directory `/usr/src/freeswitch_git'
> make[1]: *** [all] Error 2
> make[1]: Leaving directory `/usr/src/freeswitch_git'
> make: *** [current] Error 2
>
> Have I done something wrong?
>
> Thanks, Lars
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From jeff at jefflenk.com  Wed Dec 15 18:53:55 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 15 Dec 2010 07:53:55 -0800 (PST)
Subject: [Freeswitch-users] error compiling today's git
In-Reply-To: <1292426061456-5838024.post@n2.nabble.com>
References: <1292410519.23335.33.camel@luna.tc.commsmundi.com>
	<1292426061456-5838024.post@n2.nabble.com>
Message-ID: <1292428435182-5838070.post@n2.nabble.com>


This will be fixed very shortly
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/error-compiling-today-s-git-tp5837785p5838070.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Wed Dec 15 19:55:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 15 Dec 2010 08:55:50 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <AANLkTimqD-S26HNRMDjr5EFxXxfAR81jBvvOxxjGk0X_@mail.gmail.com>

Hello all,

Here's today's agenda:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_15

Chad Phillips is doing a follow up on Jester Mail and his Lua IVR toolkit.
Talk to you all in an hour!

-Michael
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From helmut.kuper at ewetel.de  Wed Dec 15 21:15:48 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 15 Dec 2010 19:15:48 +0100
Subject: [Freeswitch-users] intercepting a-leg from other context
Message-ID: <4D0905D4.3040800@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I try to intercept an incoming call from pstn to internal.

I have two contexts: default and "from_pstn".

Phone C (from_pstn) is calling phone A (default)
Phone B (default) is trying to pickup phone C

Three cases:

1)
In from_pstn I bridge the call via "user/<extension>@domain" to default
context.
When B tried to pickup phone C (in context default) all three phones
hung up. Display of Phone A shows "completed_elsewhere".

2)
When I transfer the call from phone C to default context, bridging it
there to "user/<phone_A>@domain" and intercepting the call with phone B
everything works fine.

3)
When I bridging it in context "from_pstn" to
"sofia/internal/<phone_A>@domain" and intercepting the call with phone B
it works as well. ("internal" is an alias of "default" context)

Why it is impossible to do an intercept the call as described in case 1?
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.10 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

iEYEARECAAYFAk0JBdQACgkQ4tZeNddg3dyR3gCgggf692CBwZVf7fLKRtsN1/H6
ecgAoK8THuDOS2TJv4aWZ3FEyzS8qhIE
=d+jo
-----END PGP SIGNATURE-----


From johnrose at comtex.net  Wed Dec 15 19:49:21 2010
From: johnrose at comtex.net (John Rose)
Date: Wed, 15 Dec 2010 09:49:21 -0700
Subject: [Freeswitch-users] SIP MESSAGE requests
In-Reply-To: <004501cb9b07$2ba2d880$82e88980$@comtex.net>
References: <004201cb9662$d770dad0$86529070$@comtex.net>	<4CFF3EC7.5030605@gmail.com>	<AANLkTin3gep6cpQoV0b4NEV2Tqgxe6LGyVJE5ymu46NA@mail.gmail.com>	<B6823144-18AC-42BA-BF9E-9027255A45D0@avgs.ca>	<AANLkTikP1qHa4-PtMjseJHoDxvJ5Fo7a5mfWa2aTA-Kc@mail.gmail.com>
	<004501cb9b07$2ba2d880$82e88980$@comtex.net>
Message-ID: <00d201cb9c78$065f5270$131df750$@comtex.net>

Hi All,

I've been banging my head on this  - 

I've been trying to get ESL events from incoming SIP MESSAGE requests to a
non-registered SIP user. From my research there have been others who had
been trying to do this but it seems like FS does not support it since it
would take some modifying of the Sofia stack to make this happen and
supposedly it's not trivial spitting out an event here since there is no
active SIP session. I was wondering if this modification had been done
recently?

This is what I get on the console of the FS receiving box:

2010-12-15 09:41:51.564568 [ERR] sofia_presence.c:130 Chat proto [sip]
from [dp+9999 at 64.27.3.63]
to [9898 at 67.162.136.150]
MESSAGE test from John
Invalid Profile 67.162.136.150

I've been trying to get rid of that "Invalid Profile" by changing various
settings....

Any suggestions? Does anyone know if it is possible to get ESL events from
SIP MESSAGEs or is this still a TODO for FS?

John




From massimiliano.ravelli at gmail.com  Wed Dec 15 18:16:43 2010
From: massimiliano.ravelli at gmail.com (Massimiliano Ravelli)
Date: Wed, 15 Dec 2010 16:16:43 +0100
Subject: [Freeswitch-users] Compilation error?
In-Reply-To: <009e01cb9c68$3aa42980$afec7c80$@yahoo.com>
References: <009e01cb9c68$3aa42980$afec7c80$@yahoo.com>
Message-ID: <AANLkTiktESyzR2ahWtSibWdjxcPqVFkjP6r2zaGV48AO@mail.gmail.com>

2010/12/15 Lars Zeb <larclap at yahoo.com>:
> I pulled from git this morning via 'make current' and received the
> following:

Me too...  it was already reported: have a look at "error compiling
today's git" message.

Massimiliano


From toru.tomita at gmail.com  Wed Dec 15 18:14:21 2010
From: toru.tomita at gmail.com (Toru Tomita)
Date: Wed, 15 Dec 2010 07:14:21 -0800 (PST)
Subject: [Freeswitch-users] error compiling today's git
In-Reply-To: <1292410519.23335.33.camel@luna.tc.commsmundi.com>
References: <1292410519.23335.33.camel@luna.tc.commsmundi.com>
Message-ID: <1292426061456-5838024.post@n2.nabble.com>




I got same error on Cent OS 5.5 64 bit , also from git repo source  


cc1: warnings being treated as errors
src/switch_rtp.c: In function 'jb_logger':
src/switch_rtp.c:1662: warning: implicit declaration of function
'stfu_vasprintf'
src/switch_rtp.c: In function 'switch_rtp_debug_jitter_buffer':
src/switch_rtp.c:1675: warning: implicit declaration of function
'stfu_n_debug'
src/switch_rtp.c:1676: warning: implicit declaration of function
'stfu_global_set_logger'
src/switch_rtp.c:2141:62: error: macro "stfu_n_eat" requires 6 arguments,
but only 5 given
src/switch_rtp.c: In function 'do_flush':
src/switch_rtp.c:2139: error: 'stfu_n_eat' undeclared (first use in this
function)
src/switch_rtp.c:2139: error: (Each undeclared identifier is reported only
once
src/switch_rtp.c:2139: error: for each function it appears in.)
src/switch_rtp.c:2207:59: error: macro "stfu_n_eat" requires 6 arguments,
but only 5 given
src/switch_rtp.c: In function 'read_rtp_packet':
src/switch_rtp.c:2205: error: 'stfu_n_eat' undeclared (first use in this
function)
make[1]: *** [libfreeswitch_la-switch_rtp.lo] Error 1
make: *** [all] Error 2

any help is appreciated

-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/error-compiling-today-s-git-tp5837785p5838024.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From toru.tomita at gmail.com  Wed Dec 15 19:06:51 2010
From: toru.tomita at gmail.com (Toru Tomita)
Date: Wed, 15 Dec 2010 08:06:51 -0800 (PST)
Subject: [Freeswitch-users] error compiling today's git
In-Reply-To: <1292428435182-5838070.post@n2.nabble.com>
References: <1292410519.23335.33.camel@luna.tc.commsmundi.com>
	<1292426061456-5838024.post@n2.nabble.com>
	<1292428435182-5838070.post@n2.nabble.com>
Message-ID: <AANLkTi=sjzP7JvOPqRshvoA=ngx0GxKwvfGJn_acRdi6@mail.gmail.com>


Hi Jeff

Thank you for your reply rapidly

I am looking forward to get fixed source code,

Toru Tomita

On Thu, Dec 16, 2010 at 12:53 AM, Jeff Lenk [via freeswitch-users]
<ml-node+5838070-96506956-308208 at n2.nabble.com> wrote:
> This will be fixed very shortly
>
> ________________________________
> View message @
> http://freeswitch-users.2379917.n2.nabble.com/error-compiling-today-s-git-tp5837785p5838070.html
> To unsubscribe from error compiling today's git, click here.

-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/error-compiling-today-s-git-tp5837785p5838083.html
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From u2nsam at gmail.com  Wed Dec 15 18:31:40 2010
From: u2nsam at gmail.com (Sam)
Date: Wed, 15 Dec 2010 21:01:40 +0530
Subject: [Freeswitch-users] call pickup/ DTMF / routing
In-Reply-To: <AANLkTimw9em9mdA0McQsc1zaPt7crZeT4TedKaLURjNk@mail.gmail.com>
References: <AANLkTikat1KDNDxufU=6bo102hoM+74F32bCAFU7hkkB@mail.gmail.com>
	<AANLkTimw9em9mdA0McQsc1zaPt7crZeT4TedKaLURjNk@mail.gmail.com>
Message-ID: <AANLkTi=9HV_HJ1eenmETiH6BMueLQtHR+nSX4b_=hAAZ@mail.gmail.com>

Thanks Ghulam,

I recently had started work on freeswitch ,

1.For the call pickup I was planning to use it with callgroup such that a
group of extension would be picked up by a particular group only.
like support will pickup support extension and billing will pickup billing
extensions.

2. Is there any function just to get digits without playing ?
     the digits should get captured on $(digits) var so that i can use those
digits for further routing.

3.for the routing via hangup cause the context conbridge is all there , here
the log is not generated/captured for the hangup cause.
<action application="log" data="1 B-leg hangup cause:
${bridge_hangup_cause}"/>

so the next statement is not getting executed.

regards
Sam




On Wed, Dec 15, 2010 at 3:48 PM, Ghulam Mustafa <mustafa.pk at gmail.com>wrote:

> I would suggest you to read FreeSWITCH wiki docs before posting at this
> mailing list.
>
> All of your questions lack basic knowledge of FreeSWITCH and are already
> addressed in the wiki.
>
> Also when you post please be patient and post one question at a time
> instead of publishing a digest on the mailing list.
>
> 1. A simple search on google for "freeswitch call pickup" takes you to the
> following page.
>
> http://wiki.freeswitch.org/wiki/User:Agx#Call_Pickup
>
>
> 2. Follow instructions on this page for collecting dtmf digits.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
>
> 3. make sure context conbridge exists.
>
>
> Regards,
>
>
> On Wed, Dec 15, 2010 at 10:57 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Hello folks,
>>
>> ###
>> call pickup
>>
>> I am trying call pickup for incoming calls by intercept function, its not
>> working , am i missing something ?
>>
>>
>> <extension name="group-intercept">
>>                 <condition field="destination_number" expression="^08$">
>>      <action application="answer"/>
>>          <action application="set" data="intercept_unbridged_only=true"/>
>>
>>          <action application="intercept"
>> data="${hash(select/${domain_name}-last_dial_ext/${callgroup})}"/>
>>      <action application="sleep" data="2000"/>
>>                 </condition>
>>   </extension>
>>
>>
>>
>>
>> ###
>> collecting DTMF digits
>>
>> Is there any method to collect digits by a variable clause ?
>>
>> Suppose i have a ivr playing and user inputs digits , i want to collect
>> the dtmf digits and send it to a different application where that digits
>> will be used for routing purpose.
>>
>> Any ideas !
>>
>>
>>
>>
>>
>> ###
>> routing via hangup_cause
>> was trying to create a routing rule to to route calls by accounting to the
>>
>> hangup causes,
>>
>> I have written below syntax but it fails to give the cause code to the
>> varriable for routing.
>>
>> <extension name="refer">
>>                <condition field="destination_number" expression="^1234$">
>>
>>
>>        <action application="bridge" data="sofia/external/1234 at X.X.X.X"/>
>>        <action application="set" data="hangup_after_bridge=false"/>
>>        <action application="set" data="continue_on_fail=true"/>
>>        <action application="log" data="1 B-leg hangup cause:
>> ${bridge_hangup_cause}"/>
>>        <action application="transfer" data="${bridge_hangup_cause} XML
>> conbridge"/>
>>
>>                </condition>
>>
>>
>>    </extension>
>>
>>        <extension name="no user">
>>                <condition field="destination_number"
>> expression="^(NO_USER_RESPONSE)$">
>>                <action application="bridge"
>> data="sofia/external/4567 at X.X.X.X"/>
>>
>>                </condition>
>>        </extension>
>>
>>
>> here the ${bridge_hangup_cause} is not getting executed. Am I doing it
>> right
>> or is there any other way to do it.
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Thu Dec 16 00:58:47 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 15 Dec 2010 15:58:47 -0600
Subject: [Freeswitch-users] SIP MESSAGE requests
In-Reply-To: <00d201cb9c78$065f5270$131df750$@comtex.net>
References: <004201cb9662$d770dad0$86529070$@comtex.net>	<4CFF3EC7.5030605@gmail.com>	<AANLkTin3gep6cpQoV0b4NEV2Tqgxe6LGyVJE5ymu46NA@mail.gmail.com>	<B6823144-18AC-42BA-BF9E-9027255A45D0@avgs.ca>	<AANLkTikP1qHa4-PtMjseJHoDxvJ5Fo7a5mfWa2aTA-Kc@mail.gmail.com>
	<004501cb9b07$2ba2d880$82e88980$@comtex.net>
	<00d201cb9c78$065f5270$131df750$@comtex.net>
Message-ID: <F0C59AE6-D04A-4BAC-BC3A-D03453231D8B@freeswitch.org>

alias the ip to the profile then.

/b

On Dec 15, 2010, at 10:49 AM, John Rose wrote:

> Hi All,
> 
> I've been banging my head on this  - 
> 
> I've been trying to get ESL events from incoming SIP MESSAGE requests to a
> non-registered SIP user. From my research there have been others who had
> been trying to do this but it seems like FS does not support it since it
> would take some modifying of the Sofia stack to make this happen and
> supposedly it's not trivial spitting out an event here since there is no
> active SIP session. I was wondering if this modification had been done
> recently?
> 
> This is what I get on the console of the FS receiving box:
> 
> 2010-12-15 09:41:51.564568 [ERR] sofia_presence.c:130 Chat proto [sip]
> from [dp+9999 at 64.27.3.63]
> to [9898 at 67.162.136.150]
> MESSAGE test from John
> Invalid Profile 67.162.136.150
> 
> I've been trying to get rid of that "Invalid Profile" by changing various
> settings....
> 
> Any suggestions? Does anyone know if it is possible to get ESL events from
> SIP MESSAGEs or is this still a TODO for FS?
> 
> John



From msc at freeswitch.org  Thu Dec 16 02:05:59 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 15 Dec 2010 15:05:59 -0800
Subject: [Freeswitch-users] collecting dtmf digits
In-Reply-To: <1292319831757-5833942.post@n2.nabble.com>
References: <1292319831757-5833942.post@n2.nabble.com>
Message-ID: <AANLkTimLDbh1tv8uKbWDRTbGfV5usVmqW8eQ8zFjz=0t@mail.gmail.com>

Is this an XML IVR?
-MC

On Tue, Dec 14, 2010 at 1:43 AM, samir <u2nsam at gmail.com> wrote:

>
> hello,
>
> Is there any method to collect digits by a variable clause ?
>
> Suppose i have a ivr playing and user inputs digits , i want to collect the
> dtmf digits and send it to a different application where that digits will
> be
> used for routing purpose.
>
> Any ideas !
>
> Regards
> Sam
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5833942.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Dec 16 02:14:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 15 Dec 2010 15:14:00 -0800
Subject: [Freeswitch-users] intercepting a-leg from other context
In-Reply-To: <4D0905D4.3040800@ewetel.de>
References: <4D0905D4.3040800@ewetel.de>
Message-ID: <AANLkTi=8c7FmNUi3J6Qf=bdqoNUxFrSET+0Kkj-6Hg2o@mail.gmail.com>

In your "from_pstn" context do you have all the same stuff that happens in
the "global" extension  that's found in the default context?
-MC

On Wed, Dec 15, 2010 at 10:15 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I try to intercept an incoming call from pstn to internal.
>
> I have two contexts: default and "from_pstn".
>
> Phone C (from_pstn) is calling phone A (default)
> Phone B (default) is trying to pickup phone C
>
> Three cases:
>
> 1)
> In from_pstn I bridge the call via "user/<extension>@domain" to default
> context.
> When B tried to pickup phone C (in context default) all three phones
> hung up. Display of Phone A shows "completed_elsewhere".
>
> 2)
> When I transfer the call from phone C to default context, bridging it
> there to "user/<phone_A>@domain" and intercepting the call with phone B
> everything works fine.
>
> 3)
> When I bridging it in context "from_pstn" to
> "sofia/internal/<phone_A>@domain" and intercepting the call with phone B
> it works as well. ("internal" is an alias of "default" context)
>
> Why it is impossible to do an intercept the call as described in case 1?
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.10 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/
>
> iEYEARECAAYFAk0JBdQACgkQ4tZeNddg3dyR3gCgggf692CBwZVf7fLKRtsN1/H6
> ecgAoK8THuDOS2TJv4aWZ3FEyzS8qhIE
> =d+jo
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Thu Dec 16 06:37:05 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 15 Dec 2010 22:37:05 -0500
Subject: [Freeswitch-users] DTMF and dynamic conference
References: <4A6B12C1D6794B4DA9FD15827BD6812C@e1705><AANLkTi=RNUT8ppOzmWo0TqxmyDhVZEaabtz1SkHx8fcy@mail.gmail.com><40AE384643F0467C8DAE944291A233FC@e1705>
	<AANLkTinriGsa6rLKtOk0_NusmZ1Wto8KTTOdTe4sKf=s@mail.gmail.com>
Message-ID: <FBD401E7D64B458191270AA7245912F2@e1705>

Well Mike,

I think for now I will use your suggestion ;)
and will update step by step my rfc2833 implementation...

Thanks

F
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, December 14, 2010 3:12 AM
  Subject: Re: [Freeswitch-users] DTMF and dynamic conference


  Madovsky,


  I thing maybe don't understand what you are trying to do. Do you simply want to have a conference that requires the caller to enter a PIN? Or are you trying to send a caller to a conference that has a PIN but you want to send the PIN externally? 


  In case it's the latter you can do the following...
  Create a conference locked by a PIN. In the logs below I used conf 3300 at default and 1234 as the PIN. Then I called 3300 from user 1002:


  2010-12-14 00:00:21.518045 [NOTICE] switch_channel.c:784 New Channel sofia/internal/1002 at 10.15.0.94 [09cad98f-6e12-45fa-9f01-e7d90b9ea771]
  2010-12-14 00:00:21.520627 [INFO] mod_dialplan_xml.c:331 Processing Michael <1002>->3000 in context default
  2010-12-14 00:00:21.530579 [NOTICE] mod_dptools.c:920 Channel [sofia/internal/1002 at 10.15.0.94] has been answered


  Then I did "show channels" to get the uuid:


  show channels
  uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid
  09cad98f-6e12-45fa-9f01-e7d90b9ea771,inbound,2010-12-14 00:00:21,1292313621,sofia/internal/1002 at 10.15.0.94,CS_EXECUTE,Michael,1002,10.15.30.10,3000,conference,3000-10.15.0.94 at default+1234,XML,default,PCMU,8000,64000,PCMU,8000,64000,,ElToro2.FreePBXV3,,,ACTIVE,,,,


  1 total.


  Then I did "uuid_recv_dtmf <uuid> 1234" like this:
  freeswitch at internal> uuid_recv_dtmf 09cad98f-6e12-45fa-9f01-e7d90b9ea771 1234
  -ERR no reply


  freeswitch at internal> 2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/1002 at 10.15.0.94 binding '0' to 'mute'
  2010-12-14 00:00:41.505775 [INFO] switch_ivr_async.c:162 Digit parser mod_conference: Setting realm to conf
  2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/1002 at 10.15.0.94 binding '*' to 'deaf mute'
  2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/1002 at 10.15.0.94 binding '9' to 'energy up'
  2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/1002 at 10.15.0.94 binding '8' to 'energy equ'
  2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/1002 at 10.15.0.94 binding '7' to 'energy dn'
  2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/1002 at 10.15.0.94 binding '3' to 'vol talk up'
  2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/1002 at 10.15.0.94 binding '2' to 'vol talk zero'
  2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/1002 at 10.15.0.94 binding '1' to 'vol talk dn'
  2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/1002 at 10.15.0.94 binding '6' to 'vol listen up'
  2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/1002 at 10.15.0.94 binding '5' to 'vol listen zero'
  2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/1002 at 10.15.0.94 binding '4' to 'vol listen dn'
  2010-12-14 00:00:41.505775 [INFO] mod_conference.c:6485 sofia/internal/1002 at 10.15.0.94 binding '#' to 'hangup'


  As you can see you do get "-ERR No Reply" but the operation itself works just fine.
  -MC


  On Mon, Dec 13, 2010 at 11:42 PM, Madovsky <infos at madovsky.org> wrote:

    I tried the default conference dialplan with a real phone and works well.

      ----- Original Message ----- 
      From: Michael Collins 
      To: FreeSWITCH Users Help 
      Sent: Tuesday, December 14, 2010 2:09 AM
      Subject: Re: [Freeswitch-users] DTMF and dynamic conference


      Do you actually try calling this with a real telephone to make sure that the extension is doing what you think it is doing? 
      -MC


      On Mon, Dec 13, 2010 at 11:14 AM, Madovsky <infos at madovsky.org> wrote:

        I'm trying the default conference in autoload_configs/conference.conf.xml,
        commented out the pin number line,

        but now when the ivr asks the pin number, for testing I tried
        to do this :

                <extension name="nb_conferences">
                        <condition field="${sip_to_uri}" expression="^(999)@$${domain}$">
                                <action application="answer"/>
                                <action application="info"/>
                                <!-- <action application="conference" data="confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin number]"> -->
                                <action application="conference" data="$1-$${domain}@default"/>
                                <action application="send_dtmf" data="12345 at 120"/>
                        </condition>
                </extension>

        and this on CLI:

        expand uuid_send_dtmf ${uuid} 12345


        without success.
        How the default conference example accept DTMF ?

        Thanks

        F

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From curriegrad2004 at gmail.com  Thu Dec 16 06:50:59 2010
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 15 Dec 2010 19:50:59 -0800
Subject: [Freeswitch-users] error compiling today's git
In-Reply-To: <AANLkTi=sjzP7JvOPqRshvoA=ngx0GxKwvfGJn_acRdi6@mail.gmail.com>
References: <1292410519.23335.33.camel@luna.tc.commsmundi.com>
	<1292426061456-5838024.post@n2.nabble.com>
	<1292428435182-5838070.post@n2.nabble.com>
	<AANLkTi=sjzP7JvOPqRshvoA=ngx0GxKwvfGJn_acRdi6@mail.gmail.com>
Message-ID: <AANLkTinVRxEdyi+b9KkxfyBdLhkD60i4pBOMZ=N5X6Eg@mail.gmail.com>

Seems like somebody flipped on the -Wall switch on the makefiles...

On Wed, Dec 15, 2010 at 8:06 AM, Toru Tomita <toru.tomita at gmail.com> wrote:
> Hi Jeff
>
> Thank you for your reply rapidly
>
> I am looking forward to get fixed source code,
>
> Toru Tomita
>
> On Thu, Dec 16, 2010 at 12:53 AM, Jeff Lenk [via freeswitch-users]
> <[hidden email]> wrote:
>> This will be fixed very shortly
>>
>> ________________________________
>> View message @
>>
>> http://freeswitch-users.2379917.n2.nabble.com/error-compiling-today-s-git-tp5837785p5838070.html
>> To unsubscribe from error compiling today's git, click here.
>
> ________________________________
> View this message in context: Re: error compiling today's git
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Thu Dec 16 07:11:59 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 15 Dec 2010 23:11:59 -0500
Subject: [Freeswitch-users] respond var in dialplan
Message-ID: <C55E6A8D97ED44F59F0A87414C1C3439@e1705>

Is it possilbe to specify a description/enumeration
in respond variable ?
like
<action application="respond" data="404 OUTGOING_CALL_BARRED"/> ?

thanks
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From infos at madovsky.org  Thu Dec 16 07:38:23 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 15 Dec 2010 23:38:23 -0500
Subject: [Freeswitch-users] respond var in dialplan
Message-ID: <99C2437E4E8A471C8C6215C6D7201510@e1705>

sorry forget my question

thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, December 15, 2010 11:11 PM
  Subject: respond var in dialplan


  Is it possilbe to specify a description/enumeration
  in respond variable ?
  like
  <action application="respond" data="404 OUTGOING_CALL_BARRED"/> ?

  thanks
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From infos at madovsky.org  Thu Dec 16 09:31:48 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 16 Dec 2010 01:31:48 -0500
Subject: [Freeswitch-users] mod_conference wiki error
Message-ID: <3EA0ACF29C8543E28D75F564C3B67525@e1705>

<action application="conference" data="confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin number]">
should be<action application="conference" data=confname at profilename+[conference pin number]+flags{mute|deaf|waste|moderator}>
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From neilp at cs.stanford.edu  Thu Dec 16 09:58:05 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Thu, 16 Dec 2010 12:28:05 +0530
Subject: [Freeswitch-users] freetdm and ring_ready
In-Reply-To: <AANLkTimO_0JV30SoC62we=aX3Lh2-nV5wQ+UsL7oN5Gm@mail.gmail.com>
References: <AANLkTikfyzKUZhb=2KS9=2v_Fmk=mj0=ruz328aTceM3@mail.gmail.com>
	<AANLkTimO_0JV30SoC62we=aX3Lh2-nV5wQ+UsL7oN5Gm@mail.gmail.com>
Message-ID: <AANLkTin2FNYvy9_VXo_rp9FvZJ1a7avWGBwBk1VU-T5D@mail.gmail.com>

Hi Moises,

Here is console output during the period where openzap rings, but freetdm
doesn't:

http://pastebin.freeswitch.org/14800

Thanks,
Neil


On Wed, Dec 15, 2010 at 2:33 AM, Moises Silva <moises.silva at gmail.com>wrote:

> On Tue, Dec 14, 2010 at 5:23 AM, Neil Patel <neilp at cs.stanford.edu> wrote:
>
>> I am using ring_ready to let incoming calls to my IVR app ring couple
>> times before the app engages:
>>
>> session:execute("ring_ready");
>> session:sleep(8000);
>> ...
>> session:answer();
>>
>> On one of my servers I am using freeTDM to interface with Sangoma
>> hardware, and another uses OpenZap (this is really the only difference
>> between the two FS instances). Ringing/early media is playing on the openzap
>> server, but not with freeTDM. Any idea why not?
>>
>>
> Which signaling is this?
>
> Can you post the debug output from FreeSWITCH?
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R
> 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From helmut.kuper at ewetel.de  Thu Dec 16 11:26:14 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Thu, 16 Dec 2010 09:26:14 +0100
Subject: [Freeswitch-users] intercepting a-leg from other context
In-Reply-To: <AANLkTi=8c7FmNUi3J6Qf=bdqoNUxFrSET+0Kkj-6Hg2o@mail.gmail.com>
References: <4D0905D4.3040800@ewetel.de>
	<AANLkTi=8c7FmNUi3J6Qf=bdqoNUxFrSET+0Kkj-6Hg2o@mail.gmail.com>
Message-ID: <4D09CD26.2070503@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Michael,



Am 16.12.2010 00:14, schrieb Michael Collins:
> In your "from_pstn" context do you have all the same stuff that happens
> in the "global" extension  that's found in the default context?
> -MC

I found the reason:

When bridging to phone C to A I use this in from_pstn:


1 session:setVariable("continue_on_fail", "true")
2 session:setVariable("hangup_after_bridge", "false")
3 session:execute("export", "ignore_display_updates=true")
4 session:execute("hash",
"insert/last_caller/"..dialed_ext.."/"..session:get_uuid())
5 session:execute("bridge", "user/"..dialed_ext.."@"..sip_domain)
6 hc = session:getVariable("originate_disposition")

7 if (callee_hasVM == "1" and (hc == "USER_BUSY"  or hc == "NO_ANSWER"
or hc == "NO_USER_RESPONSE")) then
8 ...voicemail...
9 end
10 freeswitch.consoleLog("NOTICE", "BRIDGE-HANGUP-REASON: "..hc.."\n")
11 session:execute("info");
12 session:hangup(hc)


I thought pickup handling resides in bridge and the call stays there
until one leg hung up. But it seems, that the dialplan-process (ROUTING)
from phone C ends as soon as Phone A got the cancel from intercept.
Bridge ends and session:hangup() is called, which terminates the call
from phone C.

Solving this seems to be easy: Just remove the hangup for originator_cancel.

Well, I think I have learned again a bit about FS call handling.

Works now! Great project :)

Thanks for reading
Helmut



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iEYEARECAAYFAk0JzSYACgkQ4tZeNddg3dw6HwCgklAlOxSugcoB+W2lXBYRDkhh
STQAnjGpkZ6J4fbAhyRrMt7dOkQmNH6N
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From trob at freemail.hu  Thu Dec 16 12:39:02 2010
From: trob at freemail.hu (=?ISO-8859-2?Q?T=F3th_R=F3bert?=)
Date: Thu, 16 Dec 2010 10:39:02 +0100
Subject: [Freeswitch-users] LUA session prevents channel's DOWN state
Message-ID: <4D09DE36.3070009@freemail.hu>

Hello

I have a LUA script, what runs permamently and listen to callstate-change events.

When i attached to a channel by
         session = freeswitch.session(uuid),
it prevents the finish of the channel. (I get some variables from the channel.)
It hangs up, the call finishes, but the channel exists until i shutdown FS.
state=CS_REPORING and callstate=HANGUP until the end of times...

How can i detach from the channel?

thx
rob



From freeswitch-list at puzzled.xs4all.nl  Thu Dec 16 13:23:23 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Thu, 16 Dec 2010 11:23:23 +0100
Subject: [Freeswitch-users] LUA session prevents channel's DOWN state
In-Reply-To: <4D09DE36.3070009@freemail.hu>
References: <4D09DE36.3070009@freemail.hu>
Message-ID: <4D09E89B.7020501@puzzled.xs4all.nl>

On 12/16/2010 10:39 AM, T?th R?bert wrote:
> Hello
> 
> I have a LUA script, what runs permamently and listen to callstate-change events.
> 
> When i attached to a channel by
>          session = freeswitch.session(uuid),
> it prevents the finish of the channel. (I get some variables from the channel.)
> It hangs up, the call finishes, but the channel exists until i shutdown FS.
> state=CS_REPORING and callstate=HANGUP until the end of times...
> 
> How can i detach from the channel?

Iirc you need to have all your sessions enclosed in
while (session:ready() == true) do

Regards,
Patrick


From trob at freemail.hu  Thu Dec 16 13:34:20 2010
From: trob at freemail.hu (=?ISO-8859-2?Q?T=F3th_R=F3bert?=)
Date: Thu, 16 Dec 2010 11:34:20 +0100
Subject: [Freeswitch-users]  LUA session prevents channel's DOWN state
Message-ID: <4D09EB2C.1060907@freemail.hu>

>  On 12/16/2010 10:39 AM, T?th R?bert wrote:
>>/  Hello
/>>/
/>>/  I have a LUA script, what runs permamently and listen to callstate-change events.
/>>/
/>/>  When i attached to a channel by
/>/>           session = freeswitch.session(uuid),
/>/>  it prevents the finish of the channel. (I get some variables from the channel.)
/>/>  It hangs up, the call finishes, but the channel exists until i shutdown FS.
/>/>  state=CS_REPORING and callstate=HANGUP until the end of times...
/>/>
/>/>  How can i detach from the channel?
/>
>  Iirc you need to have all your sessions enclosed in
>  while (session:ready() == true) do
>
>  Regards,
>  Patrick

Sorry, i don't understand what you think.

When the script receives an event, uses the event's headers, but some variables
have to be asked from the channel directly. After these datas are collected, they
are saved in the db, and wait for the next the callstate-event.

So i dont know where could i put this line where it would be useful.

(But this does not mean i'm thinking right...)




From freeswitch-list at puzzled.xs4all.nl  Thu Dec 16 15:06:59 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Thu, 16 Dec 2010 13:06:59 +0100
Subject: [Freeswitch-users] LUA session prevents channel's DOWN state
In-Reply-To: <4D09EB2C.1060907@freemail.hu>
References: <4D09EB2C.1060907@freemail.hu>
Message-ID: <4D0A00E3.3080705@puzzled.xs4all.nl>

On 12/16/2010 11:34 AM, T?th R?bert wrote:
>>  On 12/16/2010 10:39 AM, T?th R?bert wrote:
>>> /  Hello
> />>/
> />>/  I have a LUA script, what runs permamently and listen to callstate-change events.
> />>/
> />/>  When i attached to a channel by
> />/>           session = freeswitch.session(uuid),
> />/>  it prevents the finish of the channel. (I get some variables from the channel.)
> />/>  It hangs up, the call finishes, but the channel exists until i shutdown FS.
> />/>  state=CS_REPORING and callstate=HANGUP until the end of times...
> />/>
> />/>  How can i detach from the channel?
> />
>>  Iirc you need to have all your sessions enclosed in
>>  while (session:ready() == true) do
>>
>>  Regards,
>>  Patrick
> 
> Sorry, i don't understand what you think.
> 
> When the script receives an event, uses the event's headers, but some variables
> have to be asked from the channel directly. After these datas are collected, they
> are saved in the db, and wait for the next the callstate-event.
> 
> So i dont know where could i put this line where it would be useful.
> 
> (But this does not mean i'm thinking right...)

Look at the Lua examples on the wiki. There you can see how this is used.

Regards,
Patrick


From steveayre at gmail.com  Thu Dec 16 16:51:55 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 16 Dec 2010 13:51:55 +0000
Subject: [Freeswitch-users] LUA session prevents channel's DOWN state
In-Reply-To: <4D09EB2C.1060907@freemail.hu>
References: <4D09EB2C.1060907@freemail.hu>
Message-ID: <23AE1CCE-37F4-4664-B782-19988AF2412F@gmail.com>

Anywhere you wait for a while you should check session:ready(), and before any long function calls.

While your lua script has the session it cannot hangup, only when you return the session to the core during the ready() call.

Steve on iPhone

On 16 Dec 2010, at 10:34, T?th R?bert <trob at freemail.hu> wrote:

>> On 12/16/2010 10:39 AM, T?th R?bert wrote:
>>> /  Hello
> />>/
> />>/  I have a LUA script, what runs permamently and listen to callstate-change events.
> />>/
> />/>  When i attached to a channel by
> />/>           session = freeswitch.session(uuid),
> />/>  it prevents the finish of the channel. (I get some variables from the channel.)
> />/>  It hangs up, the call finishes, but the channel exists until i shutdown FS.
> />/>  state=CS_REPORING and callstate=HANGUP until the end of times...
> />/>
> />/>  How can i detach from the channel?
> />
>> Iirc you need to have all your sessions enclosed in
>> while (session:ready() == true) do
>> 
>> Regards,
>> Patrick
> 
> Sorry, i don't understand what you think.
> 
> When the script receives an event, uses the event's headers, but some variables
> have to be asked from the channel directly. After these datas are collected, they
> are saved in the db, and wait for the next the callstate-event.
> 
> So i dont know where could i put this line where it would be useful.
> 
> (But this does not mean i'm thinking right...)
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From brian at freeswitch.org  Thu Dec 16 17:23:42 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 16 Dec 2010 08:23:42 -0600
Subject: [Freeswitch-users] mod_conference wiki error
In-Reply-To: <3EA0ACF29C8543E28D75F564C3B67525@e1705>
References: <3EA0ACF29C8543E28D75F564C3B67525@e1705>
Message-ID: <F90CC48D-FBB0-4DB4-9E3D-ACBD5DB8CCBC@freeswitch.org>

The beauty of the wiki is you can update it yourself... have you done so already?

/b

On Dec 16, 2010, at 12:31 AM, Madovsky wrote:

> <action application="conference" data="confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin number]">
> should be
> <action application="conference" data=confname at profilename+[conference pin number]+flags{mute|deaf|waste|moderator}>
> _

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From infos at madovsky.org  Thu Dec 16 17:43:57 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 16 Dec 2010 09:43:57 -0500
Subject: [Freeswitch-users] mod_conference wiki error
References: <3EA0ACF29C8543E28D75F564C3B67525@e1705>
	<F90CC48D-FBB0-4DB4-9E3D-ACBD5DB8CCBC@freeswitch.org>
Message-ID: <199F2B287ABE4DF7AA42C9E6E64550A6@e1705>

I didn't want without your authorization...
  ----- Original Message ----- 
  From: Brian West 
  To: FreeSWITCH Users Help 
  Sent: Thursday, December 16, 2010 9:23 AM
  Subject: Re: [Freeswitch-users] mod_conference wiki error


  The beauty of the wiki is you can update it yourself... have you done so already?


  /b


  On Dec 16, 2010, at 12:31 AM, Madovsky wrote:


<action application="conference" data="confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin number]">
should be<action application="conference" data=confname at profilename+[conference pin number]+flags{mute|deaf|waste|moderator}>
_




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From steveayre at gmail.com  Thu Dec 16 18:46:55 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 16 Dec 2010 15:46:55 +0000
Subject: [Freeswitch-users] mod_conference wiki error
In-Reply-To: <199F2B287ABE4DF7AA42C9E6E64550A6@e1705>
References: <3EA0ACF29C8543E28D75F564C3B67525@e1705>
	<F90CC48D-FBB0-4DB4-9E3D-ACBD5DB8CCBC@freeswitch.org>
	<199F2B287ABE4DF7AA42C9E6E64550A6@e1705>
Message-ID: <543C3817-CD01-4F53-BB0B-3A05024BC6F7@gmail.com>

Don't wait for authorisation.

Anything that's found to be wrong can always be undone or corrected later.

Steve on iPhone

On 16 Dec 2010, at 14:43, "Madovsky" <infos at madovsky.org> wrote:

> I didn't want without your authorization...
> ----- Original Message -----
> From: Brian West
> To: FreeSWITCH Users Help
> Sent: Thursday, December 16, 2010 9:23 AM
> Subject: Re: [Freeswitch-users] mod_conference wiki error
> 
> The beauty of the wiki is you can update it yourself... have you done so already?
> 
> /b
> 
> On Dec 16, 2010, at 12:31 AM, Madovsky wrote:
> 
>>  <action application="conference" data="confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin number]">
>> should be
>> <action application="conference" data=confname at profilename+[conference pin number]+flags{mute|deaf|waste|moderator}>
>> _
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From infos at madovsky.org  Thu Dec 16 19:19:55 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 16 Dec 2010 11:19:55 -0500
Subject: [Freeswitch-users] mod_conference wiki error
Message-ID: <4BC020FE9B4544968F47E6FE8D16D579@e1705>

Done...

  Don't wait for authorisation.

  Anything that's found to be wrong can always be undone or corrected later.

  Steve on iPhone

  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Thursday, December 16, 2010 9:43 AM
  Subject: Re: [Freeswitch-users] mod_conference wiki error


  I didn't want without your authorization...
    ----- Original Message ----- 
    From: Brian West 
    To: FreeSWITCH Users Help 
    Sent: Thursday, December 16, 2010 9:23 AM
    Subject: Re: [Freeswitch-users] mod_conference wiki error


    The beauty of the wiki is you can update it yourself... have you done so already? 


    /b


    On Dec 16, 2010, at 12:31 AM, Madovsky wrote:


<action application="conference" data="confname at profilename+flags{mute|deaf|waste|moderator}+[conference pin number]">
should be<action application="conference" data=confname at profilename+[conference pin number]+flags{mute|deaf|waste|moderator}>
_




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


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From william.suffill at gmail.com  Thu Dec 16 19:20:32 2010
From: william.suffill at gmail.com (William Suffill)
Date: Thu, 16 Dec 2010 11:20:32 -0500
Subject: [Freeswitch-users] mod_conference wiki error
In-Reply-To: <543C3817-CD01-4F53-BB0B-3A05024BC6F7@gmail.com>
References: <3EA0ACF29C8543E28D75F564C3B67525@e1705>
	<F90CC48D-FBB0-4DB4-9E3D-ACBD5DB8CCBC@freeswitch.org>
	<199F2B287ABE4DF7AA42C9E6E64550A6@e1705>
	<543C3817-CD01-4F53-BB0B-3A05024BC6F7@gmail.com>
Message-ID: <AANLkTi=8bBbwKGMMz7vucCPRLE_+VyOyFDCxDdq7Wo7o@mail.gmail.com>

It be just as quick to change it and let the community know that you updated
it so someone can give it a read over to see if anything should be modified.
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From srinivas.ksvreddy at gmail.com  Thu Dec 16 19:32:55 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Thu, 16 Dec 2010 22:02:55 +0530
Subject: [Freeswitch-users] SqlCommand could not be found: mod_managed
	compile error in CentOS
Message-ID: <AANLkTikLv62+oNdk3UhGDcMQzv-g2CyxECp2wasvRT-G@mail.gmail.com>

Hi All,

When i am trying to build mod_managed in Freesiwtch Latest version, i am
getting this error, this error i got in my own class created for updating
SqlServer Database,

any idea?

-- 
Srinivasula Reddy K
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From moises.silva at gmail.com  Thu Dec 16 19:42:46 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 16 Dec 2010 11:42:46 -0500
Subject: [Freeswitch-users] freetdm and ring_ready
In-Reply-To: <AANLkTin2FNYvy9_VXo_rp9FvZJ1a7avWGBwBk1VU-T5D@mail.gmail.com>
References: <AANLkTikfyzKUZhb=2KS9=2v_Fmk=mj0=ruz328aTceM3@mail.gmail.com>
	<AANLkTimO_0JV30SoC62we=aX3Lh2-nV5wQ+UsL7oN5Gm@mail.gmail.com>
	<AANLkTin2FNYvy9_VXo_rp9FvZJ1a7avWGBwBk1VU-T5D@mail.gmail.com>
Message-ID: <AANLkTiniZ_6utTHyWQQJx2V8QnKur27JaaP2NBsmu+rS@mail.gmail.com>

On Thu, Dec 16, 2010 at 1:58 AM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Hi Moises,
>
> Here is console output during the period where openzap rings, but freetdm
> doesn't:
>
> http://pastebin.freeswitch.org/14800
>
> Thanks,
> Neil
>
>
Hi Neil,

Who is generating the ringing tone? are you doing it? from your post I
understand you want early media.

The reason OpenZAP works for you is that  the ring_ready application was
also enabling early media in OpenZAP, which is not 100% correct. In FreeTDM,
ring_ready should only send an alerting (no inband indicator). If you want
inband, please use the "pre_answer" application.

Let me know if that works for you,

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com
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From davidjbrazier at gmail.com  Thu Dec 16 20:39:48 2010
From: davidjbrazier at gmail.com (David Brazier)
Date: Thu, 16 Dec 2010 17:39:48 +0000
Subject: [Freeswitch-users] SqlCommand could not be found: mod_managed
 compile error in CentOS
In-Reply-To: <AANLkTikLv62+oNdk3UhGDcMQzv-g2CyxECp2wasvRT-G@mail.gmail.com>
References: <AANLkTikLv62+oNdk3UhGDcMQzv-g2CyxECp2wasvRT-G@mail.gmail.com>
Message-ID: <AANLkTi=FO4L6_+Pk5fhGwcq1c4mSsaWs7gwinv743-1k@mail.gmail.com>

You shouldn't add code to mod_managed itself.  Compile your code in a
separate dll - see the Demo sample
(src/mod/languages/mod_managed/managed/Demo.csx).  If you want to run
it on Linux you might find that Microsoft stuff like SqlCommand
doesn't work on Mono though.

David

On Thu, Dec 16, 2010 at 4:32 PM, srinivasula reddy
<srinivas.ksvreddy at gmail.com> wrote:
>
> When i am trying to build mod_managed in Freesiwtch Latest version, i am
> getting this error, this error i got in my own class created for updating
> SqlServer Database,
>
> any idea?
>


From msc at freeswitch.org  Thu Dec 16 21:14:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Dec 2010 10:14:32 -0800
Subject: [Freeswitch-users] freetdm and ring_ready
In-Reply-To: <AANLkTiniZ_6utTHyWQQJx2V8QnKur27JaaP2NBsmu+rS@mail.gmail.com>
References: <AANLkTikfyzKUZhb=2KS9=2v_Fmk=mj0=ruz328aTceM3@mail.gmail.com>
	<AANLkTimO_0JV30SoC62we=aX3Lh2-nV5wQ+UsL7oN5Gm@mail.gmail.com>
	<AANLkTin2FNYvy9_VXo_rp9FvZJ1a7avWGBwBk1VU-T5D@mail.gmail.com>
	<AANLkTiniZ_6utTHyWQQJx2V8QnKur27JaaP2NBsmu+rS@mail.gmail.com>
Message-ID: <AANLkTim3DVdUnYVsGprgXggQeGU=Pq6Lyp9WNeQ_mB8L@mail.gmail.com>

<snip>

>
> The reason OpenZAP works for you is that  the ring_ready application was
> also enabling early media in OpenZAP, which is not 100% correct. In FreeTDM,
> ring_ready should only send an alerting (no inband indicator). If you want
> inband, please use the "pre_answer" application.
>
> <snip>

Moy,
So is this a case of relying on incorrect behavior as a "feature"? Just
confirming. I'll make sure that the pre_answer app is mentioned in the
FreeTDM wiki pages where appropriate.

Neil,
Hope your project is going well!

-MC
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From marty at maui-systems.co.uk  Thu Dec 16 14:02:31 2010
From: marty at maui-systems.co.uk (Marty Lee)
Date: Thu, 16 Dec 2010 11:02:31 +0000
Subject: [Freeswitch-users] Compile Error on Solaris 10
Message-ID: <D4C4C7D8-131D-4A37-8382-9CFA0FC8C04C@maui-systems.co.uk>


Hi,

just playing with building FreeSwitch on a Solaris 10 box and am hitting a link
error when building the target '.libs/freeswitch'

Undefined                       first referenced
symbol                             in file
make_mask32                         freeswitch-switch.o
ld: fatal: Symbol referencing errors. No output written to .libs/freeswitch

Looking at the source, make_mask32 seems to be defined in the spandsp/src/bit_operations.c file, which isn't involved in the linking at
this point.

Building with Sun Studio 12, just in case it's of any interest.

I'll try and move the make_mask32 function in to the library and see if that cures
the problem. Thought I'd ask here, as this kind of thing should appear on most
platforms if it really is a problem - meaning it might be something at my end.

m

(Tried posting this to the -dev alias, but I'm guessing it's held by the moderators;
 I've put the 'make_mask32' function into the freeswitch library and now the compile
 works ok)

-----
Marty Lee                         e: marty at maui-systems.co.uk
Technical Director                v: +44 845 869 2661
Maui Systems Ltd                  f: +44 871 433 8922
Scotland, UK                      w: http://www.maui-systems.co.uk



From marty at maui-systems.co.uk  Thu Dec 16 14:33:52 2010
From: marty at maui-systems.co.uk (Marty Lee)
Date: Thu, 16 Dec 2010 11:33:52 +0000
Subject: [Freeswitch-users] FreeSwitch / Solaris 10 / Sun Studio 12
Message-ID: <43979AAE-A757-437A-92CA-3BEE1A0D1466@maui-systems.co.uk>


Current GIT repository doesn't compile on Solaris 10 with Sun Studio due to
unresolved link errors on the 'make_mask32' function.

I've isolated the function and added it to the libfreeswitch library - this
cures the link problems.

Patches attached if anyone is interested.

m

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-----
Marty Lee                         e: marty at maui-systems.co.uk
Technical Director                v: +44 845 869 2661
Maui Systems Ltd                  f: +44 871 433 8922
Scotland, UK                      w: http://www.maui-systems.co.uk


From taras at 1adm.ru  Thu Dec 16 18:06:34 2010
From: taras at 1adm.ru (=?gb2312?B?p7Sn0afip9Gn4yCns6fRp9On6aflp9w=?=)
Date: Thu, 16 Dec 2010 18:06:34 +0300
Subject: [Freeswitch-users] FS G729 Release: 147
In-Reply-To: <69A9A2CA-787F-4494-B750-86A296E388AB@freeswitch.org>
References: <6380BBD4-ACDF-43B1-8958-FEFD8A5BC886@freeswitch.org>
	<4C8BC51C.30804@puzzled.xs4all.nl>
	<AANLkTi=wUcO0REk_-a5hz4dH1cmoZes-EOkzebo5CqJ+@mail.gmail.com>
	<AANLkTinciH42m2GzY3Ycb4dPiOnrzuneLmNyUeEtwph6@mail.gmail.com>
	<005348A1-1945-4F38-8A58-78AB5DA836FE@ipeva.fr>
	<66B0274C-5240-4083-8942-7D0808938854@freeswitch.org>
	<B162BFD4-FC00-45C7-A4FB-AB3BD1726B1C@ipeva.fr>
	<69A9A2CA-787F-4494-B750-86A296E388AB@freeswitch.org>
Message-ID: <6ADDA2AC9C82F04AAFF4CD5C6884888F54800A0C69@SRV.o.1adm.ru>

Brian,

I can test g729 for FreeBSD.

--
Savchuk Taras


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Wednesday, September 15, 2010 3:45 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FS G729 Release: 147

Good I think I know what that is.. expect a 148 or 149 tonight or tomorrow morning to correct that.  I'm also working on FreeBSD for everyone interested but I'll have to have some testers.

/b

On Sep 14, 2010, at 6:35 PM, David Ponzone wrote:

> Brian,
> 
> the backtrace of the crash when loading mod_com_g729 is available there:
> http://pastebin.freeswitch.org/13902
> 
> Thanks a lot for your efforts to fix this.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 


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

From u2nsam at gmail.com  Thu Dec 16 06:29:40 2010
From: u2nsam at gmail.com (Sam)
Date: Thu, 16 Dec 2010 08:59:40 +0530
Subject: [Freeswitch-users] collecting dtmf digits
In-Reply-To: <AANLkTimLDbh1tv8uKbWDRTbGfV5usVmqW8eQ8zFjz=0t@mail.gmail.com>
References: <1292319831757-5833942.post@n2.nabble.com>
	<AANLkTimLDbh1tv8uKbWDRTbGfV5usVmqW8eQ8zFjz=0t@mail.gmail.com>
Message-ID: <AANLkTi=o-6h9q2UWwaYZ9hHuC7sMNTNYQs=nJm6yCaBH@mail.gmail.com>

hi,

Its not an XML IVR but trying to collect digits when the user punches in
some digits after a playback sound file,
so that the digits would get stored into $(digits) and i can use those
digits for further processing or transferring
to other dial-plan as per the digits punched in.

Suppose a digits punched are 4567, I collect them in $(digits) and by
transfer function i do the transfer.

<action application="transfer" data="$(digits) default XML"/>

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

Also i could see that in IVR.conf can do this by pressing 4567 and executing
a statement , but here also question
is how can i use it by storing those collected digits in an variable and the
using it further.

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

I was thinking how in both the cases it could be done.

Thnx & Regds
Sam



On Thu, Dec 16, 2010 at 4:35 AM, Michael Collins <msc at freeswitch.org> wrote:

> Is this an XML IVR?
> -MC
>
>
> On Tue, Dec 14, 2010 at 1:43 AM, samir <u2nsam at gmail.com> wrote:
>
>>
>> hello,
>>
>> Is there any method to collect digits by a variable clause ?
>>
>> Suppose i have a ivr playing and user inputs digits , i want to collect
>> the
>> dtmf digits and send it to a different application where that digits will
>> be
>> used for routing purpose.
>>
>> Any ideas !
>>
>> Regards
>> Sam
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5833942.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Thu Dec 16 06:34:49 2010
From: u2nsam at gmail.com (Sam)
Date: Thu, 16 Dec 2010 09:04:49 +0530
Subject: [Freeswitch-users] call pickup
In-Reply-To: <665DE8BC-65A7-46D2-8F8E-002649C59C21@ipeva.fr>
References: <1292391773589-5837233.post@n2.nabble.com>
	<665DE8BC-65A7-46D2-8F8E-002649C59C21@ipeva.fr>
Message-ID: <AANLkTimZeMcBbDkW-W=O8661Vx4ZXququv8qPtLFTDgJ@mail.gmail.com>

Yes David ,

You are right from yesterday i was trying to post it to the forum and it was
giving me message that your post is not posted in the forum.
And at last it did with multiple emails !! may be it could have happened as
i created the account day before yesterday and it takes time.
Sorry all for any inconvenience caused !

Regds

On Wed, Dec 15, 2010 at 3:47 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Samir,
>
> Perhaps next time, you could think before sending your mail.
> It will avoid you sending multiple emails for the same question.
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 15/12/2010 ? 06:42, samir a ?crit :
>
> hello, I am trying call "pickup" for the incoming calls with intercept
> function but it do not seems to work, am i missing something ? Regards Sam
> ------------------------------
> View this message in context: call pickup<http://freeswitch-users.2379917.n2.nabble.com/call-pickup-tp5837233p5837233.html>
> Sent from the freeswitch-users mailing list archive<http://freeswitch-users.2379917.n2.nabble.com/>at
> Nabble.com.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Thu Dec 16 13:28:20 2010
From: u2nsam at gmail.com (Sam)
Date: Thu, 16 Dec 2010 15:58:20 +0530
Subject: [Freeswitch-users] routing via hangup_cause
In-Reply-To: <1292386719584-5837136.post@n2.nabble.com>
References: <1292386719584-5837136.post@n2.nabble.com>
Message-ID: <AANLkTiko+9M0MNeKgZO2aprbZg8LevA_7pHMSsKqn=AE@mail.gmail.com>

I am not able to fetch the hangup cause in the dial-plan by using log
function .
I would like to use that hangup cause variable for further routing so that
the routing would be conditional to hangup cause.

Regards
Sam

On Wed, Dec 15, 2010 at 9:48 AM, samir <u2nsam at gmail.com> wrote:

>
> Hello friends,
>
> was trying to create a routing rule to to route calls by accounting to the
> hangup causes,
>
> I have written below syntax but it fails to give the cause code to the
> varriable for routing.
>
> <extension name="refer">
>               <condition field="destination_number" expression="^1234$">
>
>
>       <action application="bridge" data="sofia/external/1234 at X.X.X.X"/>
>       <action application="set" data="hangup_after_bridge=false"/>
>       <action application="set" data="continue_on_fail=true"/>
>       <action application="log" data="1 B-leg hangup cause:
> ${bridge_hangup_cause}"/>
>       <action application="transfer" data="${bridge_hangup_cause} XML
> conbridge"/>
>
>               </condition>
>
>
>   </extension>
>
>       <extension name="no user">
>               <condition field="destination_number"
> expression="^(NO_USER_RESPONSE)$">
>               <action application="bridge"
> data="sofia/external/4567 at X.X.X.X"/>
>
>               </condition>
>       </extension>
>
>
> here the ${bridge_hangup_cause} is not getting executed. Am I doing it
> right
> or is there any other way to do it.
>
> Regards
> Sam
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/routing-via-hangup-cause-tp5837136p5837136.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From marty at maui-systems.co.uk  Thu Dec 16 19:07:15 2010
From: marty at maui-systems.co.uk (Marty Lee)
Date: Thu, 16 Dec 2010 16:07:15 +0000
Subject: [Freeswitch-users] FreeSwitch / Solaris 10 Zone / Sun Studio
Message-ID: <EA71D56B-B3A3-4F15-ACD8-4BE0210D7846@maui-systems.co.uk>


ok, so now I have FreeSwitch up and running in a Solaris 10 Zone, compiled with Sun
Studio 12 and two SIP phones attached and passing calls - result!

For the developers, the libs/esl/Makefile is not autogenerated and has a load
of 'g++' specifics in it; once I adjusted those to the Sun Studio settings, the
'hash' module built fine.

Last one; know this isn't your issue, but the 'flite' module uses the csw_sts.h
file and defines a number of 'const' variables in structures. The code to do
a '#define const' to disable those seems to be being applied to the system
includes too, resulting in a number of function conflicts - the ones that preceed
the #define and any that are nested that follow.

Instead of trying to 'bodge' the code to disable 'const', it was much easier to
just remove the 'const' from the csw_sts.h file itself. Again, more ramblings
in case any other 'Solaris' people ever end up here.

BTW, this has just been tested within a Solaris Zone - something that I can't 
get to work on Asterisk, which was my main reason for looking for an alternative.

If I can get IAX and SIP working reliably, then this will become the next prod'
solution.

Right - onwards and inwards to understand the xml file etc.

:-)

m


-----
Marty Lee                         e: marty at maui-systems.co.uk
Technical Director                v: +44 845 869 2661
Maui Systems Ltd                  f: +44 871 433 8922
Scotland, UK                      w: http://www.maui-systems.co.uk



From eagle.antonio at gmail.com  Thu Dec 16 20:30:42 2010
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Thu, 16 Dec 2010 17:30:42 +0000
Subject: [Freeswitch-users] CallCenter Agent UUID-Standby
Message-ID: <AANLkTikNKi7Rzo8tKdMq+=iG7MaO304Q2CD15e6CJTs9@mail.gmail.com>

Good Afternoon.

I'm leaving Asterisk ( yayyyyyyyyyyyyy) and was curious in trying out the
Call Center feature,  uuid-standby , now i created a dialplan and I'm able
to connect a caller to an agent.
Now by my assumption as soon as the caller disconnects the agent should be
trow back ( transfer) into the loop (4099 ext.) but instead i get an hangup
and the sip client terminates the call.

Any ideas ?

Sample Dialplan

<extension>
  <condition field="destination_number" expression="^(4099)$">
    <action application="set" data="res=${callcenter_config(agent set
uuid ${caller_id_number}@${domain_name} '${uuid}')}" />
    <action application="set" data="res=${callcenter_config(agent set
type ${caller_id_number}@${domain_name} 'uuid-standby')}" />
    <action application="set" data="res=${callcenter_config(agent set
status ${caller_id_number}@${domain_name} 'Available (On Demand)')}"
/>
    <action application="set" data="res=${callcenter_config(agent set
state ${caller_id_number}@${domain_name} 'Waiting')}" />
    <action application="set"
data="cc_warning_tone=tone_stream://%(200,0,500,600,700)"/>
    <action application="answer" />
    <action application="playback" data="$${hold_music}"/>
    <action application="transfer" data="4099"/> <!-- Remove this if
you just want to get a single call -->
  </condition>
</extension>


Thanks For Your Time
Ant?nio Teixeira
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From johnrose at comtex.net  Thu Dec 16 22:52:05 2010
From: johnrose at comtex.net (John Rose)
Date: Thu, 16 Dec 2010 12:52:05 -0700
Subject: [Freeswitch-users] SIP MESSAGE requests
In-Reply-To: <F0C59AE6-D04A-4BAC-BC3A-D03453231D8B@freeswitch.org>
References: <004201cb9662$d770dad0$86529070$@comtex.net>	<4CFF3EC7.5030605@gmail.com>	<AANLkTin3gep6cpQoV0b4NEV2Tqgxe6LGyVJE5ymu46NA@mail.gmail.com>	<B6823144-18AC-42BA-BF9E-9027255A45D0@avgs.ca>	<AANLkTikP1qHa4-PtMjseJHoDxvJ5Fo7a5mfWa2aTA-Kc@mail.gmail.com>	<004501cb9b07$2ba2d880$82e88980$@comtex.net>	<00d201cb9c78$065f5270$131df750$@comtex.net>
	<F0C59AE6-D04A-4BAC-BC3A-D03453231D8B@freeswitch.org>
Message-ID: <003301cb9d5a$b7c77230$27565690$@comtex.net>

Brian,

OK, thanks for that it does get rid of "Invalid profile".

Next error once that is gone is:
freeswitch at JohnRose-Laptop> 2010-12-16 09:17:19.509892 [ERR]
sofia_presence.c:146 Can't find registered user 9898 at 67.162.136.150

Still no ESL event as I would want one even if no registered user. Unless
there is a way to send it somewhere else...  the user 9898 does exist.

Any ideas?

John


> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> 
> alias the ip to the profile then.
> 
> /b
> 
> On Dec 15, 2010, at 10:49 AM, John Rose wrote:
> 
> > Hi All,
> >
> > I've been banging my head on this  -
> >
> > I've been trying to get ESL events from incoming SIP MESSAGE requests
> > to a non-registered SIP user. From my research there have been others
> > who had been trying to do this but it seems like FS does not support
> > it since it would take some modifying of the Sofia stack to make this
> > happen and supposedly it's not trivial spitting out an event here
> > since there is no active SIP session. I was wondering if this
> > modification had been done recently?
> >
> > This is what I get on the console of the FS receiving box:
> >
> > 2010-12-15 09:41:51.564568 [ERR] sofia_presence.c:130 Chat proto [sip]
> > from [dp+9999 at 64.27.3.63] to [9898 at 67.162.136.150] MESSAGE test from
> > John Invalid Profile 67.162.136.150
> >
> > I've been trying to get rid of that "Invalid Profile" by changing
> > various settings....
> >
> > Any suggestions? Does anyone know if it is possible to get ESL events
> > from SIP MESSAGEs or is this still a TODO for FS?
> >
> > John
> 




From brian at freeswitch.org  Thu Dec 16 23:10:55 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 16 Dec 2010 14:10:55 -0600
Subject: [Freeswitch-users] FS G729 Release: 147
In-Reply-To: <6ADDA2AC9C82F04AAFF4CD5C6884888F54800A0C69@SRV.o.1adm.ru>
References: <6380BBD4-ACDF-43B1-8958-FEFD8A5BC886@freeswitch.org>
	<4C8BC51C.30804@puzzled.xs4all.nl>
	<AANLkTi=wUcO0REk_-a5hz4dH1cmoZes-EOkzebo5CqJ+@mail.gmail.com>
	<AANLkTinciH42m2GzY3Ycb4dPiOnrzuneLmNyUeEtwph6@mail.gmail.com>
	<005348A1-1945-4F38-8A58-78AB5DA836FE@ipeva.fr>
	<66B0274C-5240-4083-8942-7D0808938854@freeswitch.org>
	<B162BFD4-FC00-45C7-A4FB-AB3BD1726B1C@ipeva.fr>
	<69A9A2CA-787F-4494-B750-86A296E388AB@freeswitch.org>
	<6ADDA2AC9C82F04AAFF4CD5C6884888F54800A0C69@SRV.o.1adm.ru>
Message-ID: <B6C50173-AC9B-4580-849C-678131F12E67@freeswitch.org>

I currently do not have a build for FreeBSD but its on my roadmap at some point in the future... are you 64bit or 32bit?

/b

On Dec 16, 2010, at 9:06 AM, ????? ?????? wrote:

> Brian,
> 
> I can test g729 for FreeBSD.
> 
> --
> Savchuk Taras



From brian at freeswitch.org  Thu Dec 16 23:11:16 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 16 Dec 2010 14:11:16 -0600
Subject: [Freeswitch-users] FreeSwitch / Solaris 10 / Sun Studio 12
In-Reply-To: <43979AAE-A757-437A-92CA-3BEE1A0D1466@maui-systems.co.uk>
References: <43979AAE-A757-437A-92CA-3BEE1A0D1466@maui-systems.co.uk>
Message-ID: <FFEA2FC0-EB45-402D-8C52-1576938EF4C5@freeswitch.org>

Please post the diff's on http://jira.freeswitch.org

Thanks,
Brian

On Dec 16, 2010, at 5:33 AM, Marty Lee wrote:

> 
> Current GIT repository doesn't compile on Solaris 10 with Sun Studio due to
> unresolved link errors on the 'make_mask32' function.
> 
> I've isolated the function and added it to the libfreeswitch library - this
> cures the link problems.
> 
> Patches attached if anyone is interested.
> 
> m
> 
> <Makefile.am><switch_mask.c>
> -----
> Marty Lee                         e: marty at maui-systems.co.uk
> Technical Director                v: +44 845 869 2661
> Maui Systems Ltd                  f: +44 871 433 8922
> Scotland, UK                      w: http://www.maui-systems.co.uk
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From tgraziano at myitdepartment.net  Thu Dec 16 23:26:39 2010
From: tgraziano at myitdepartment.net (Tony Graziano)
Date: Thu, 16 Dec 2010 15:26:39 -0500
Subject: [Freeswitch-users] FS G729 Release: 147
In-Reply-To: <B6C50173-AC9B-4580-849C-678131F12E67@freeswitch.org>
References: <6380BBD4-ACDF-43B1-8958-FEFD8A5BC886@freeswitch.org>
	<4C8BC51C.30804@puzzled.xs4all.nl>
	<AANLkTi=wUcO0REk_-a5hz4dH1cmoZes-EOkzebo5CqJ+@mail.gmail.com>
	<AANLkTinciH42m2GzY3Ycb4dPiOnrzuneLmNyUeEtwph6@mail.gmail.com>
	<005348A1-1945-4F38-8A58-78AB5DA836FE@ipeva.fr>
	<66B0274C-5240-4083-8942-7D0808938854@freeswitch.org>
	<B162BFD4-FC00-45C7-A4FB-AB3BD1726B1C@ipeva.fr>
	<69A9A2CA-787F-4494-B750-86A296E388AB@freeswitch.org>
	<6ADDA2AC9C82F04AAFF4CD5C6884888F54800A0C69@SRV.o.1adm.ru>
	<B6C50173-AC9B-4580-849C-678131F12E67@freeswitch.org>
Message-ID: <AANLkTinfC8cO5iW7bzw41NRfwsVGS4g261Yx81a0ubLY@mail.gmail.com>

2010/12/16 Brian West <brian at freeswitch.org>:
> I currently do not have a build for FreeBSD but its on my roadmap at some point in the future... are you 64bit or 32bit?
>
> /b
I am also interested in seeing this available for freebsd .

Tony


From Avi at aMarcus.com  Fri Dec 17 00:28:04 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Thu, 16 Dec 2010 23:28:04 +0200
Subject: [Freeswitch-users] Calling Card System Setup - Suggestions for New
	Setup?
In-Reply-To: <AANLkTi=y+L26Ff+afY0+HRS9Ct=yip+U3DATukO8k8Cq@mail.gmail.com>
References: <AANLkTi=y+L26Ff+afY0+HRS9Ct=yip+U3DATukO8k8Cq@mail.gmail.com>
Message-ID: <AANLkTikikbjg3+P=QnK18hVpBfMM+TCQkeLCuWbHTS+4@mail.gmail.com>

Hi, I'm setting up a calling card system with FreeSWITCH. Figured I'd ask
before setting it all up-

Does anyone have any suggestions, or scripts, that they use for auto-login
or for logging in users to check their balance?

Any specific pitfalls to worry about? I figured I could bypass media once
the user places their call. Is that a bad idea?

And.. any suggestions for payment processors other than paypal - some people
signing up won't have internet, their CC will have to be manually keyed in.
I've heard one place suggest authorize.net

Thanks!
-Avi
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From cliff at develix.com  Fri Dec 17 01:19:10 2010
From: cliff at develix.com (Cliff Wells)
Date: Thu, 16 Dec 2010 14:19:10 -0800
Subject: [Freeswitch-users] FS ends call on DTMF **
Message-ID: <1292537950.5800.147.camel@portable-evil>

Hi,

I've got a Lua application that takes the caller id and generates
particular sequences of DTMF tones (for a testing system) based on the
CID.  If I call into the system (using my cellphone), it correctly plays
the sequence.   However, if I press ** on the keypad it stops the
generated DTMF tones and then apparently hangs up.    
This wouldn't be a huge issue (the app isn't supposed to receive any
DTMF anyway) except that one of the systems calling in apparently has
enough echo on the remote end that the DTMF I'm generating is being
echoed back to FS and the first two DTMF tones I generate are, of
course, "**", so the end result is the system calls into FS, gets two
tones and the call is terminated.

As an aside, it appears that ** is unique in this way.   If I dial "99"
or "77", the DTMF still pauses, but then resumes (or maybe restarts...
it's difficult to tell).

I was using an older version of FS, so I did "make current" a couple of
hours ago, but it didn't help.

My one and only dialplan is this:

  <extension name="responder">
    <condition field="destination_number" expression="^(.*)$">
      <action application="lua" data="responder.lua"/>
    </condition>
  </extension>

And responder.lua does this:

session:answer ()
session:sleep (3000)
session:execute ("playback", genstream (val))  -- genstream uses the CID to produce a tone_stream
session:hangup()

(clearly there's a bit more to the Lua app, but nothing relevant to this
issue).

I see this in the console when this happens:

2010-12-17 01:04:34.603404 [NOTICE] mod_dptools.c:920 Channel [sofia/internal/3232269108 at 99.99.99.99] has been answered
EXECUTE sofia/internal/3232269108 at 99.99.99.99 lua(responder.lua)
2010-12-17 01:04:34.673314 [DEBUG] sofia.c:4606 Channel sofia/internal/5551231234 at 99.99.99.99 entering state [ready][200]
2010-12-17 01:04:34.751303 [DEBUG] switch_rtp.c:2657 Correct ip/port confirmed.
EXECUTE sofia/internal/5551231234 at 99.99.99.99 playback(tone_stream://*(200,200);*(200,200); ...
2010-12-17 01:04:37.915169 [DEBUG] switch_ivr_play_say.c:1236 Codec Activated L16 at 8000hz 1 channels 20ms
2010-12-17 01:04:38.295137 [DEBUG] switch_rtp.c:3033 RTP RECV DTMF *:1440
2010-12-17 01:04:38.295137 [DEBUG] mod_dptools.c:1634 Digit *
2010-12-17 01:04:38.295137 [DEBUG] switch_ivr_play_say.c:1573 done playing file
2010-12-17 01:04:38.295137 [DEBUG] switch_cpp.cpp:602 CoreSession::hangup
2010-12-17 01:04:38.295137 [DEBUG] switch_channel.c:2455 (sofia/internal/5551231234 at 99.99.99.99) Callstate Change ACTIVE -> HANGUP
2010-12-17 01:04:38.295137 [NOTICE] switch_cpp.cpp:604 Hangup sofia/internal/5551231234 at 99.99.99.99 [CS_EXECUTE] [NORMAL_CLEARING]
2010-12-17 01:04:38.295137 [DEBUG] switch_channel.c:2471 Send signal sofia/internal/5551231234 at 99.99.99.99 [KILL]
2010-12-17 01:04:38.295137 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/5551231234 at 99.99.99.99 [BREAK]
2010-12-17 01:04:38.295137 [DEBUG] switch_cpp.cpp:972 sofia/internal/5551231234 at 99.99.99.99 destroy/unlink session from object
2010-12-17 01:04:38.295137 [DEBUG] switch_core_session.c:1998 sofia/internal/5551231234 at 99.99.99.99 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)

At this point I'd be content with a simple workaround (mute the inbound
audio, ignore DTMF, etc), since I do not need any feedback from the
other end.


Cliff Wells <cliff at develix.com>



From anthony.minessale at gmail.com  Fri Dec 17 01:32:35 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 16 Dec 2010 16:32:35 -0600
Subject: [Freeswitch-users] FS ends call on DTMF **
In-Reply-To: <1292537950.5800.147.camel@portable-evil>
References: <1292537950.5800.147.camel@portable-evil>
Message-ID: <AANLkTimWkpnSRvcc2dsPNtRNQA_DhKczFDBxpPDTfubx@mail.gmail.com>

the playback app has default terminate keys

use session:streamFile(file); or set the variable playback_terminators to "none"


On Thu, Dec 16, 2010 at 4:19 PM, Cliff Wells <cliff at develix.com> wrote:
> Hi,
>
> I've got a Lua application that takes the caller id and generates
> particular sequences of DTMF tones (for a testing system) based on the
> CID. ?If I call into the system (using my cellphone), it correctly plays
> the sequence. ? However, if I press ** on the keypad it stops the
> generated DTMF tones and then apparently hangs up.
> This wouldn't be a huge issue (the app isn't supposed to receive any
> DTMF anyway) except that one of the systems calling in apparently has
> enough echo on the remote end that the DTMF I'm generating is being
> echoed back to FS and the first two DTMF tones I generate are, of
> course, "**", so the end result is the system calls into FS, gets two
> tones and the call is terminated.
>
> As an aside, it appears that ** is unique in this way. ? If I dial "99"
> or "77", the DTMF still pauses, but then resumes (or maybe restarts...
> it's difficult to tell).
>
> I was using an older version of FS, so I did "make current" a couple of
> hours ago, but it didn't help.
>
> My one and only dialplan is this:
>
> ?<extension name="responder">
> ? ?<condition field="destination_number" expression="^(.*)$">
> ? ? ?<action application="lua" data="responder.lua"/>
> ? ?</condition>
> ?</extension>
>
> And responder.lua does this:
>
> session:answer ()
> session:sleep (3000)
> session:execute ("playback", genstream (val)) ?-- genstream uses the CID to produce a tone_stream
> session:hangup()
>
> (clearly there's a bit more to the Lua app, but nothing relevant to this
> issue).
>
> I see this in the console when this happens:
>
> 2010-12-17 01:04:34.603404 [NOTICE] mod_dptools.c:920 Channel [sofia/internal/3232269108 at 99.99.99.99] has been answered
> EXECUTE sofia/internal/3232269108 at 99.99.99.99 lua(responder.lua)
> 2010-12-17 01:04:34.673314 [DEBUG] sofia.c:4606 Channel sofia/internal/5551231234 at 99.99.99.99 entering state [ready][200]
> 2010-12-17 01:04:34.751303 [DEBUG] switch_rtp.c:2657 Correct ip/port confirmed.
> EXECUTE sofia/internal/5551231234 at 99.99.99.99 playback(tone_stream://*(200,200);*(200,200); ...
> 2010-12-17 01:04:37.915169 [DEBUG] switch_ivr_play_say.c:1236 Codec Activated L16 at 8000hz 1 channels 20ms
> 2010-12-17 01:04:38.295137 [DEBUG] switch_rtp.c:3033 RTP RECV DTMF *:1440
> 2010-12-17 01:04:38.295137 [DEBUG] mod_dptools.c:1634 Digit *
> 2010-12-17 01:04:38.295137 [DEBUG] switch_ivr_play_say.c:1573 done playing file
> 2010-12-17 01:04:38.295137 [DEBUG] switch_cpp.cpp:602 CoreSession::hangup
> 2010-12-17 01:04:38.295137 [DEBUG] switch_channel.c:2455 (sofia/internal/5551231234 at 99.99.99.99) Callstate Change ACTIVE -> HANGUP
> 2010-12-17 01:04:38.295137 [NOTICE] switch_cpp.cpp:604 Hangup sofia/internal/5551231234 at 99.99.99.99 [CS_EXECUTE] [NORMAL_CLEARING]
> 2010-12-17 01:04:38.295137 [DEBUG] switch_channel.c:2471 Send signal sofia/internal/5551231234 at 99.99.99.99 [KILL]
> 2010-12-17 01:04:38.295137 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/5551231234 at 99.99.99.99 [BREAK]
> 2010-12-17 01:04:38.295137 [DEBUG] switch_cpp.cpp:972 sofia/internal/5551231234 at 99.99.99.99 destroy/unlink session from object
> 2010-12-17 01:04:38.295137 [DEBUG] switch_core_session.c:1998 sofia/internal/5551231234 at 99.99.99.99 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>
> At this point I'd be content with a simple workaround (mute the inbound
> audio, ignore DTMF, etc), since I do not need any feedback from the
> other end.
>
>
> Cliff Wells <cliff at develix.com>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From nazim.aghabayov at gmail.com  Fri Dec 17 01:44:37 2010
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Fri, 17 Dec 2010 02:44:37 +0400
Subject: [Freeswitch-users] collecting dtmf digits
In-Reply-To: <AANLkTi=o-6h9q2UWwaYZ9hHuC7sMNTNYQs=nJm6yCaBH@mail.gmail.com>
References: <1292319831757-5833942.post@n2.nabble.com>	<AANLkTimLDbh1tv8uKbWDRTbGfV5usVmqW8eQ8zFjz=0t@mail.gmail.com>
	<AANLkTi=o-6h9q2UWwaYZ9hHuC7sMNTNYQs=nJm6yCaBH@mail.gmail.com>
Message-ID: <4D0A9655.7070007@gmail.com>

Why not using Lua script? Lua is quite powerful and you could benefit of
using the luasql for db connectivity.

g_caller_id = session:getVariable("caller_id_number")
g_caller_destination_number = session:getVariable("destination_number")
g_caller_context = session:getVariable("context")
g_caller_uuid = session:getVariable("uuid")
...
session:flushDigits()
digits = session:playAndGetDigits(4, 4, 3, 3000, "#",
              wav_base .. langId .. "/" .. prompt_wav, "",
      "[" .. allowed_digit1 .. "," .. allowed_digitN .. "]");
...
session:transfer(tostring(dest_ext), "XML", "public")

Regards,
Nazim


On 12/16/2010 07:29 AM, Sam wrote:
> hi,
>
> Its not an XML IVR but trying to collect digits when the user punches in
> some digits after a playback sound file,
> so that the digits would get stored into $(digits) and i can use those
> digits for further processing or transferring
> to other dial-plan as per the digits punched in.
>
> Suppose a digits punched are 4567, I collect them in $(digits) and by
> transfer function i do the transfer.
>
> <action application="transfer" data="$(digits) default XML"/>
>
> -----------------------------------------------------------------------------------------------
>
> Also i could see that in IVR.conf can do this by pressing 4567 and executing
> a statement , but here also question
> is how can i use it by storing those collected digits in an variable and the
> using it further.
>
> ----------------------------------------------------------------------------------------------
>
> I was thinking how in both the cases it could be done.
>
> Thnx & Regds
> Sam
>
>
>
> On Thu, Dec 16, 2010 at 4:35 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>> Is this an XML IVR?
>> -MC
>>
>>
>> On Tue, Dec 14, 2010 at 1:43 AM, samir <u2nsam at gmail.com> wrote:
>>
>>> hello,
>>>
>>> Is there any method to collect digits by a variable clause ?
>>>
>>> Suppose i have a ivr playing and user inputs digits , i want to collect
>>> the
>>> dtmf digits and send it to a different application where that digits will
>>> be
>>> used for routing purpose.
>>>
>>> Any ideas !
>>>
>>> Regards
>>> Sam
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5833942.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From cliff at develix.com  Fri Dec 17 01:55:49 2010
From: cliff at develix.com (Cliff Wells)
Date: Thu, 16 Dec 2010 14:55:49 -0800
Subject: [Freeswitch-users] FS ends call on DTMF **
In-Reply-To: <AANLkTimWkpnSRvcc2dsPNtRNQA_DhKczFDBxpPDTfubx@mail.gmail.com>
References: <1292537950.5800.147.camel@portable-evil>
	<AANLkTimWkpnSRvcc2dsPNtRNQA_DhKczFDBxpPDTfubx@mail.gmail.com>
Message-ID: <1292540149.5800.157.camel@portable-evil>

On Thu, 2010-12-16 at 16:32 -0600, Anthony Minessale wrote:

> the playback app has default terminate keys
> 
> use session:streamFile(file); or set the variable playback_terminators to "none"



Thanks so much.    I was pricing Hair Club for Men and it's still too
expensive for me.

Cliff

-- 
Cliff Wells <cliff at develix.com>
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From abubacker at bksystems.co.in  Fri Dec 17 08:49:17 2010
From: abubacker at bksystems.co.in (abubacker)
Date: Fri, 17 Dec 2010 11:19:17 +0530
Subject: [Freeswitch-users] mod call center tier behaviour
Message-ID: <4D0AF9DD.6010703@bksys.co.in>

I am working in mod_callcenter

Before my call center configuration is like this
/referene : callcenter.conf.xml/

/<agents>
<agent name="1000 at default" type="callback" 
contact="[call_timeout=10]user/1000" status="Available" 
max-no-answer="0" wrap-up-time="0" reject-delay-time="1" 
busy-delay-time="1" />
<agent name="1001 at default" type="callback" 
contact="[call_timeout=10]user/1001" status="Available" 
max-no-answer="0" wrap-up-time="0" reject-delay-time="1" 
busy-delay-time="1" />
<agent name="1003 at default" type="callback" 
contact="[call_timeout=10]user/1006" status="Available" 
max-no-answer="0" wrap-up-time="0" reject-delay-time="1" 
busy-delay-time="1" />
</agents>
<tiers>
<!-- If no level or position is provided, they will default to 1.  You 
should do this to keep db value on restart. -->
<tier agent="1000 at default" queue="support at default" level="1" position="1"/>
<tier agent="1001 at default" queue="support at default" level="1" position="2"/>
<tier agent="1003 at default" queue="support at default" level="1" position="3"/>
<!-- secondary tier up -->
<tier agent="1003 at default" queue="secondary" level="1" position="3"/>
<tier agent="1001 at default" queue="secondary" level="1" position="2"/>
<tier agent="1000 at default" queue="secondary" level="1" position="1"/>
</tiers>

</configuration>
/
Tier list :
/freeswitch at FMS-Queue> callcenter_config tier list support at default

queue|agent|state|level|position
support at default|1000 at default|Ready|1|1
support at default|1001 at default|Ready|1|2
support at default|1003 at default|Ready|1|3
+OK/

When I renamed the 1003 at default as 1006 at default
/ <agents>
<agent name="1000 at default" type="callback" 
contact="[call_timeout=10]user/1000" status="Available" 
max-no-answer="0" wrap-up-time="0" reject-delay-time="1" 
busy-delay-time="1" />
<agent name="1001 at default" type="callback" 
contact="[call_timeout=10]user/1001" status="Available" 
max-no-answer="0" wrap-up-time="0" reject-delay-time="1" 
busy-delay-time="1" />
<agent name="1006 at default" type="callback" 
contact="[call_timeout=10]user/1006" status="Available" 
max-no-answer="0" wrap-up-time="0" reject-delay-time="1" 
busy-delay-time="1" />
</agents>
<tiers>
<!-- If no level or position is provided, they will default to 1.  You 
should do this to keep db value on restart. -->
<tier agent="1000 at default" queue="support at default" level="1" position="1"/>
<tier agent="1001 at default" queue="support at default" level="1" position="2"/>
<tier agent="1006 at default" queue="support at default" level="1" position="3"/>
<!-- secondary tier up -->
<tier agent="1006 at default" queue="secondary" level="1" position="3"/>
<tier agent="1001 at default" queue="secondary" level="1" position="2"/>
<tier agent="1000 at default" queue="secondary" level="1" position="1"/>
</tiers>/

After that I have reload the mod_callcenter
http://pastebin.freeswitch.org/14810

The 1003 at default is not overwritten , but 1006 at default has added newly
why 1003 at default is still exist
/freeswitch at FMS-Queue> callcenter_config tier list support at default

queue|agent|state|level|position
support at default|1000 at default|Ready|1|1
support at default|1001 at default|Ready|1|2
support at default|1003 at default|Ready|1|3
support at default|1006 at default|Ready|1|3
+OK/

Is it a bug or Is it a default behaviour ?

-- 
Best regards,
N.Abubacker ,
Associate system engineer ,
bk systems pvt ltd ,
Ph : 9144-43902701

Disclaimer:http://www.bksystems.co.in/email-policy

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From freeswitch at tlainvestments.com  Fri Dec 17 10:15:54 2010
From: freeswitch at tlainvestments.com (Troy Anderson)
Date: Fri, 17 Dec 2010 00:15:54 -0700
Subject: [Freeswitch-users] CallCenter Agent UUID-Standby
In-Reply-To: <AANLkTikNKi7Rzo8tKdMq+=iG7MaO304Q2CD15e6CJTs9@mail.gmail.com>
References: <AANLkTikNKi7Rzo8tKdMq+=iG7MaO304Q2CD15e6CJTs9@mail.gmail.com>
Message-ID: <E9D43200-E2B4-4272-86AD-4D642E3165ED@tlainvestments.com>

Hi Ant?nio,

Take a look at this part of the wiki: http://wiki.freeswitch.org/wiki/Mod_callcenter#Callback

Note the last line of the sample dial plan:
    <action application="transfer" data="4099"/> <!-- Remove this if you just want to get a single call -->
If you want to throw them back in the loop, you have to explicitly do it in your dial plan.

-Troy

On Dec 16, 2010, at 10:30 AM, Antonio Teixeira wrote:

> Good Afternoon.
> 
> I'm leaving Asterisk ( yayyyyyyyyyyyyy) and was curious in trying out the Call Center feature,  uuid-standby , now i created a dialplan and I'm able to connect a caller to an agent.
> Now by my assumption as soon as the caller disconnects the agent should be trow back ( transfer) into the loop (4099 ext.) but instead i get an hangup and the sip client terminates the call.
> 
> Any ideas ?
> 
> Sample Dialplan
> 
> <extension>
>   <condition field="destination_number" expression="^(4099)$">
>     <action application="set" data="res=${callcenter_config(agent set uuid ${caller_id_number}@${domain_name} '${uuid}')}" />
> 
>     <action application="set" data="res=${callcenter_config(agent set type ${caller_id_number}@${domain_name} 'uuid-standby')}" />
>     <action application="set" data="res=${callcenter_config(agent set status ${caller_id_number}@${domain_name} 'Available (On Demand)')}" />
> 
>     <action application="set" data="res=${callcenter_config(agent set state ${caller_id_number}@${domain_name} 'Waiting')}" />
>     <action application="set" data="cc_warning_tone=tone_stream://%(200,0,500,600,700)"/>
> 
>     <action application="answer" />
>     <action application="playback" data="$${hold_music}"/>
>     <action application="transfer" data="4099"/> <!-- Remove this if you just want to get a single call -->
> 
>   </condition>
> </extension>
> 
> Thanks For Your Time
> Ant?nio Teixeira
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From erik.dekkers at wvds.nl  Fri Dec 17 10:28:38 2010
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Fri, 17 Dec 2010 08:28:38 +0100
Subject: [Freeswitch-users] FS G729 Release: 147
In-Reply-To: <AANLkTinfC8cO5iW7bzw41NRfwsVGS4g261Yx81a0ubLY@mail.gmail.com>
References: <6380BBD4-ACDF-43B1-8958-FEFD8A5BC886@freeswitch.org>
	<4C8BC51C.30804@puzzled.xs4all.nl>
	<AANLkTi=wUcO0REk_-a5hz4dH1cmoZes-EOkzebo5CqJ+@mail.gmail.com>
	<AANLkTinciH42m2GzY3Ycb4dPiOnrzuneLmNyUeEtwph6@mail.gmail.com>
	<005348A1-1945-4F38-8A58-78AB5DA836FE@ipeva.fr>
	<66B0274C-5240-4083-8942-7D0808938854@freeswitch.org>
	<B162BFD4-FC00-45C7-A4FB-AB3BD1726B1C@ipeva.fr>
	<69A9A2CA-787F-4494-B750-86A296E388AB@freeswitch.org>
	<6ADDA2AC9C82F04AAFF4CD5C6884888F54800A0C69@SRV.o.1adm.ru>
	<B6C50173-AC9B-4580-849C-678131F12E67@freeswitch.org>
	<AANLkTinfC8cO5iW7bzw41NRfwsVGS4g261Yx81a0ubLY@mail.gmail.com>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE077704C795@FS1.wvds.local>

Here you've got another tester (wvds-nl on #freeswitch)

-----Oorspronkelijk bericht-----
Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Tony Graziano
Verzonden: donderdag 16 december 2010 21:27
Aan: FreeSWITCH Users Help
Onderwerp: Re: [Freeswitch-users] FS G729 Release: 147

2010/12/16 Brian West <brian at freeswitch.org>:
> I currently do not have a build for FreeBSD but its on my roadmap at some point in the future... are you 64bit or 32bit?
>
> /b
I am also interested in seeing this available for freebsd .

Tony

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


From u2nsam at gmail.com  Fri Dec 17 10:56:33 2010
From: u2nsam at gmail.com (samir)
Date: Thu, 16 Dec 2010 23:56:33 -0800 (PST)
Subject: [Freeswitch-users] collecting dtmf digits
In-Reply-To: <4D0A9655.7070007@gmail.com>
References: <1292319831757-5833942.post@n2.nabble.com>
	<AANLkTimLDbh1tv8uKbWDRTbGfV5usVmqW8eQ8zFjz=0t@mail.gmail.com>
	<AANLkTi=o-6h9q2UWwaYZ9hHuC7sMNTNYQs=nJm6yCaBH@mail.gmail.com>
	<4D0A9655.7070007@gmail.com>
Message-ID: <AANLkTi=VkoUNZktvTFcg-NtqkyEw4C7ZznK-Oo1ZC1t_@mail.gmail.com>


Thanks for this Nazim,

Also where i can look for all the updated functions documented for Lua.

Regads
Sam

On Fri, Dec 17, 2010 at 4:17 AM, Nazim Aghabayov [via freeswitch-users] <
ml-node+5843837-1908927474-292429 at n2.nabble.com<ml-node%2B5843837-1908927474-292429 at n2.nabble.com>
> wrote:

> Why not using Lua script? Lua is quite powerful and you could benefit of
> using the luasql for db connectivity.
>
> g_caller_id = session:getVariable("caller_id_number")
> g_caller_destination_number = session:getVariable("destination_number")
> g_caller_context = session:getVariable("context")
> g_caller_uuid = session:getVariable("uuid")
> ...
> session:flushDigits()
> digits = session:playAndGetDigits(4, 4, 3, 3000, "#",
>               wav_base .. langId .. "/" .. prompt_wav, "",
>       "[" .. allowed_digit1 .. "," .. allowed_digitN .. "]");
> ...
> session:transfer(tostring(dest_ext), "XML", "public")
>
> Regards,
> Nazim
>
>
> On 12/16/2010 07:29 AM, Sam wrote:
>
> > hi,
> >
> > Its not an XML IVR but trying to collect digits when the user punches in
> > some digits after a playback sound file,
> > so that the digits would get stored into $(digits) and i can use those
> > digits for further processing or transferring
> > to other dial-plan as per the digits punched in.
> >
> > Suppose a digits punched are 4567, I collect them in $(digits) and by
> > transfer function i do the transfer.
> >
> > <action application="transfer" data="$(digits) default XML"/>
> >
> >
> -----------------------------------------------------------------------------------------------
>
> >
> > Also i could see that in IVR.conf can do this by pressing 4567 and
> executing
> > a statement , but here also question
> > is how can i use it by storing those collected digits in an variable and
> the
> > using it further.
> >
> >
> ----------------------------------------------------------------------------------------------
>
> >
> > I was thinking how in both the cases it could be done.
> >
> > Thnx & Regds
> > Sam
> >
> >
> >
> > On Thu, Dec 16, 2010 at 4:35 AM, Michael Collins <[hidden email]<http://user/SendEmail.jtp?type=node&node=5843837&i=0>>
> wrote:
> >
> >> Is this an XML IVR?
> >> -MC
> >>
> >>
> >> On Tue, Dec 14, 2010 at 1:43 AM, samir <[hidden email]<http://user/SendEmail.jtp?type=node&node=5843837&i=1>>
> wrote:
> >>
> >>> hello,
> >>>
> >>> Is there any method to collect digits by a variable clause ?
> >>>
> >>> Suppose i have a ivr playing and user inputs digits , i want to collect
>
> >>> the
> >>> dtmf digits and send it to a different application where that digits
> will
> >>> be
> >>> used for routing purpose.
> >>>
> >>> Any ideas !
> >>>
> >>> Regards
> >>> Sam
> >>> --
> >>> View this message in context:
> >>>
> http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5833942.html<http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5833942.html?by-user=t>
> >>> Sent from the freeswitch-users mailing list archive at Nabble.com.
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5843837&i=2>
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> [hidden email] <http://user/SendEmail.jtp?type=node&node=5843837&i=3>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > [hidden email] <http://user/SendEmail.jtp?type=node&node=5843837&i=4>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> [hidden email] <http://user/SendEmail.jtp?type=node&node=5843837&i=5>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
>  View message @
> http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5843837.html
>
> To unsubscribe from collecting dtmf digits, click here<http://freeswitch-users.2379917.n2.nabble.com/template/NamlServlet.jtp?macro=unsubscribe_by_code&node=5833942&code=dTJuc2FtQGdtYWlsLmNvbXw1ODMzOTQyfC0xMTk1MzYwMDk5>.
>
>

-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5844699.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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From u2nsam at gmail.com  Fri Dec 17 10:57:21 2010
From: u2nsam at gmail.com (Sam)
Date: Fri, 17 Dec 2010 13:27:21 +0530
Subject: [Freeswitch-users] collecting dtmf digits
In-Reply-To: <4D0A9655.7070007@gmail.com>
References: <1292319831757-5833942.post@n2.nabble.com>
	<AANLkTimLDbh1tv8uKbWDRTbGfV5usVmqW8eQ8zFjz=0t@mail.gmail.com>
	<AANLkTi=o-6h9q2UWwaYZ9hHuC7sMNTNYQs=nJm6yCaBH@mail.gmail.com>
	<4D0A9655.7070007@gmail.com>
Message-ID: <AANLkTimhM9XwaHfeunv_GtywP5axNQO3iMwi9swqhRHZ@mail.gmail.com>

Thanks for this Nazim,

Also where i can look for all the updated functions documented for Lua.

Regads
Sam



On Fri, Dec 17, 2010 at 4:14 AM, Nazim Aghabayov
<nazim.aghabayov at gmail.com>wrote:

> Why not using Lua script? Lua is quite powerful and you could benefit of
> using the luasql for db connectivity.
>
> g_caller_id = session:getVariable("caller_id_number")
> g_caller_destination_number = session:getVariable("destination_number")
> g_caller_context = session:getVariable("context")
> g_caller_uuid = session:getVariable("uuid")
> ...
> session:flushDigits()
> digits = session:playAndGetDigits(4, 4, 3, 3000, "#",
>              wav_base .. langId .. "/" .. prompt_wav, "",
>      "[" .. allowed_digit1 .. "," .. allowed_digitN .. "]");
> ...
> session:transfer(tostring(dest_ext), "XML", "public")
>
> Regards,
> Nazim
>
>
> On 12/16/2010 07:29 AM, Sam wrote:
> > hi,
> >
> > Its not an XML IVR but trying to collect digits when the user punches in
> > some digits after a playback sound file,
> > so that the digits would get stored into $(digits) and i can use those
> > digits for further processing or transferring
> > to other dial-plan as per the digits punched in.
> >
> > Suppose a digits punched are 4567, I collect them in $(digits) and by
> > transfer function i do the transfer.
> >
> > <action application="transfer" data="$(digits) default XML"/>
> >
> >
> -----------------------------------------------------------------------------------------------
> >
> > Also i could see that in IVR.conf can do this by pressing 4567 and
> executing
> > a statement , but here also question
> > is how can i use it by storing those collected digits in an variable and
> the
> > using it further.
> >
> >
> ----------------------------------------------------------------------------------------------
> >
> > I was thinking how in both the cases it could be done.
> >
> > Thnx & Regds
> > Sam
> >
> >
> >
> > On Thu, Dec 16, 2010 at 4:35 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> >
> >> Is this an XML IVR?
> >> -MC
> >>
> >>
> >> On Tue, Dec 14, 2010 at 1:43 AM, samir <u2nsam at gmail.com> wrote:
> >>
> >>> hello,
> >>>
> >>> Is there any method to collect digits by a variable clause ?
> >>>
> >>> Suppose i have a ivr playing and user inputs digits , i want to collect
> >>> the
> >>> dtmf digits and send it to a different application where that digits
> will
> >>> be
> >>> used for routing purpose.
> >>>
> >>> Any ideas !
> >>>
> >>> Regards
> >>> Sam
> >>> --
> >>> View this message in context:
> >>>
> http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5833942.html
> >>> Sent from the freeswitch-users mailing list archive at Nabble.com.
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From u2nsam at gmail.com  Fri Dec 17 11:07:18 2010
From: u2nsam at gmail.com (samir)
Date: Fri, 17 Dec 2010 00:07:18 -0800 (PST)
Subject: [Freeswitch-users] collecting dtmf digits
In-Reply-To: <4D0A9655.7070007@gmail.com>
References: <1292319831757-5833942.post@n2.nabble.com>
	<AANLkTimLDbh1tv8uKbWDRTbGfV5usVmqW8eQ8zFjz=0t@mail.gmail.com>
	<AANLkTi=o-6h9q2UWwaYZ9hHuC7sMNTNYQs=nJm6yCaBH@mail.gmail.com>
	<4D0A9655.7070007@gmail.com>
Message-ID: <AANLkTimHAZF8eZ=h4mrPcPRU9qB9j2GHvGb0rBsE86n7@mail.gmail.com>


Hello Nazim,

When i use originate command it gives me below error :-

2010-12-17 13:29:30.964039 [ERR] mod_lua.cpp:182 Error in originate expected
4..4 args, got 1
stack traceback:
        [C]: in function 'originate'
        /usr/local/freeswitch/scripts/hello.lua:14: in main chunk

I am using

digits = session:getDigits(5, "#", 3000);
freeswitch.consoleLog("info", "Got dtmf: ".. digits .."\n");
session.originate("sofia/external/(digits)@192.168.2.3");


any idea why is it ?

Regards
Sam


On Fri, Dec 17, 2010 at 4:17 AM, Nazim Aghabayov [via freeswitch-users] <
ml-node+5843837-1908927474-292429 at n2.nabble.com<ml-node%2B5843837-1908927474-292429 at n2.nabble.com>
> wrote:

> Why not using Lua script? Lua is quite powerful and you could benefit of
> using the luasql for db connectivity.
>
> g_caller_id = session:getVariable("caller_id_number")
> g_caller_destination_number = session:getVariable("destination_number")
> g_caller_context = session:getVariable("context")
> g_caller_uuid = session:getVariable("uuid")
> ...
> session:flushDigits()
> digits = session:playAndGetDigits(4, 4, 3, 3000, "#",
>               wav_base .. langId .. "/" .. prompt_wav, "",
>       "[" .. allowed_digit1 .. "," .. allowed_digitN .. "]");
> ...
> session:transfer(tostring(dest_ext), "XML", "public")
>
> Regards,
> Nazim
>
>
> On 12/16/2010 07:29 AM, Sam wrote:
>
> > hi,
> >
> > Its not an XML IVR but trying to collect digits when the user punches in
> > some digits after a playback sound file,
> > so that the digits would get stored into $(digits) and i can use those
> > digits for further processing or transferring
> > to other dial-plan as per the digits punched in.
> >
> > Suppose a digits punched are 4567, I collect them in $(digits) and by
> > transfer function i do the transfer.
> >
> > <action application="transfer" data="$(digits) default XML"/>
> >
> >
> -----------------------------------------------------------------------------------------------
>
> >
> > Also i could see that in IVR.conf can do this by pressing 4567 and
> executing
> > a statement , but here also question
> > is how can i use it by storing those collected digits in an variable and
> the
> > using it further.
> >
> >
> ----------------------------------------------------------------------------------------------
>
> >
> > I was thinking how in both the cases it could be done.
> >
> > Thnx & Regds
> > Sam
> >
> >
> >
> > On Thu, Dec 16, 2010 at 4:35 AM, Michael Collins <[hidden email]<http://user/SendEmail.jtp?type=node&node=5843837&i=0>>
> wrote:
> >
> >> Is this an XML IVR?
> >> -MC
> >>
> >>
> >> On Tue, Dec 14, 2010 at 1:43 AM, samir <[hidden email]<http://user/SendEmail.jtp?type=node&node=5843837&i=1>>
> wrote:
> >>
> >>> hello,
> >>>
> >>> Is there any method to collect digits by a variable clause ?
> >>>
> >>> Suppose i have a ivr playing and user inputs digits , i want to collect
>
> >>> the
> >>> dtmf digits and send it to a different application where that digits
> will
> >>> be
> >>> used for routing purpose.
> >>>
> >>> Any ideas !
> >>>
> >>> Regards
> >>> Sam
> >>> --
> >>> View this message in context:
> >>>
> http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5833942.html<http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5833942.html?by-user=t>
> >>> Sent from the freeswitch-users mailing list archive at Nabble.com.
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5843837&i=2>
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> [hidden email] <http://user/SendEmail.jtp?type=node&node=5843837&i=3>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > [hidden email] <http://user/SendEmail.jtp?type=node&node=5843837&i=4>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> [hidden email] <http://user/SendEmail.jtp?type=node&node=5843837&i=5>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
>  View message @
> http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5843837.html
>
> To unsubscribe from collecting dtmf digits, click here<http://freeswitch-users.2379917.n2.nabble.com/template/NamlServlet.jtp?macro=unsubscribe_by_code&node=5833942&code=dTJuc2FtQGdtYWlsLmNvbXw1ODMzOTQyfC0xMTk1MzYwMDk5>.
>
>

-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5844718.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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From david.ponzone at ipeva.fr  Fri Dec 17 11:24:12 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 17 Dec 2010 09:24:12 +0100
Subject: [Freeswitch-users] collecting dtmf digits
In-Reply-To: <AANLkTimHAZF8eZ=h4mrPcPRU9qB9j2GHvGb0rBsE86n7@mail.gmail.com>
References: <1292319831757-5833942.post@n2.nabble.com>
	<AANLkTimLDbh1tv8uKbWDRTbGfV5usVmqW8eQ8zFjz=0t@mail.gmail.com>
	<AANLkTi=o-6h9q2UWwaYZ9hHuC7sMNTNYQs=nJm6yCaBH@mail.gmail.com>
	<4D0A9655.7070007@gmail.com>
	<AANLkTimHAZF8eZ=h4mrPcPRU9qB9j2GHvGb0rBsE86n7@mail.gmail.com>
Message-ID: <D0BEED38-8178-4A24-AEE6-816C0AA46780@ipeva.fr>

You need to read the wiki intensively, and also, Internet is a very useful source of information to learn LUA.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 17/12/2010 ? 09:07, samir a ?crit :

> Hello Nazim,
> 
> When i use originate command it gives me below error :-
> 
> 2010-12-17 13:29:30.964039 [ERR] mod_lua.cpp:182 Error in originate expected 4..4 args, got 1
> stack traceback:
>         [C]: in function 'originate'
>         /usr/local/freeswitch/scripts/hello.lua:14: in main chunk
> 
> I am using    
> 
> digits = session:getDigits(5, "#", 3000);
> freeswitch.consoleLog("info", "Got dtmf: ".. digits .."\n");
> session.originate("sofia/external/(digits)@192.168.2.3");
> 
> 
> any idea why is it ?
> 
> Regards
> Sam
> 
> 
> On Fri, Dec 17, 2010 at 4:17 AM, Nazim Aghabayov [via freeswitch-users] <[hidden email]> wrote:
> Why not using Lua script? Lua is quite powerful and you could benefit of 
> using the luasql for db connectivity. 
> 
> g_caller_id = session:getVariable("caller_id_number") 
> g_caller_destination_number = session:getVariable("destination_number") 
> g_caller_context = session:getVariable("context") 
> g_caller_uuid = session:getVariable("uuid") 
> ... 
> session:flushDigits() 
> digits = session:playAndGetDigits(4, 4, 3, 3000, "#", 
>               wav_base .. langId .. "/" .. prompt_wav, "", 
>       "[" .. allowed_digit1 .. "," .. allowed_digitN .. "]"); 
> ... 
> session:transfer(tostring(dest_ext), "XML", "public") 
> 
> Regards, 
> Nazim 
> 
> 
> On 12/16/2010 07:29 AM, Sam wrote:
> 
> > hi, 
> > 
> > Its not an XML IVR but trying to collect digits when the user punches in 
> > some digits after a playback sound file, 
> > so that the digits would get stored into $(digits) and i can use those 
> > digits for further processing or transferring 
> > to other dial-plan as per the digits punched in. 
> > 
> > Suppose a digits punched are 4567, I collect them in $(digits) and by 
> > transfer function i do the transfer. 
> > 
> > <action application="transfer" data="$(digits) default XML"/> 
> > 
> > ----------------------------------------------------------------------------------------------- 
> > 
> > Also i could see that in IVR.conf can do this by pressing 4567 and executing 
> > a statement , but here also question 
> > is how can i use it by storing those collected digits in an variable and the 
> > using it further. 
> > 
> > ---------------------------------------------------------------------------------------------- 
> > 
> > I was thinking how in both the cases it could be done. 
> > 
> > Thnx & Regds 
> > Sam 
> > 
> > 
> >
> > On Thu, Dec 16, 2010 at 4:35 AM, Michael Collins <[hidden email]> wrote: 
> > 
> >> Is this an XML IVR? 
> >> -MC 
> >> 
> >> 
> >> On Tue, Dec 14, 2010 at 1:43 AM, samir <[hidden email]> wrote:
> 
> >> 
> >>> hello, 
> >>> 
> >>> Is there any method to collect digits by a variable clause ? 
> >>> 
> >>> Suppose i have a ivr playing and user inputs digits , i want to collect 
> >>> the 
> >>> dtmf digits and send it to a different application where that digits will 
> >>> be 
> >>> used for routing purpose. 
> >>> 
> >>> Any ideas ! 
> >>> 
> >>> Regards 
> >>> Sam 
> >>> -- 
> >>> View this message in context: 
> >>> http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5833942.html
> >>> Sent from the freeswitch-users mailing list archive at Nabble.com. 
> >>> 
> >>> _______________________________________________ 
> >>> FreeSWITCH-users mailing list 
> >>> [hidden email] 
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>> 
> >> 
> >> _______________________________________________ 
> >> FreeSWITCH-users mailing list 
> >> [hidden email] 
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >> 
> >> 
> > 
> > 
> > _______________________________________________ 
> > FreeSWITCH-users mailing list 
> > [hidden email] 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________ 
> FreeSWITCH-users mailing list 
> [hidden email] 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> View message @ http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5843837.html
> 
> To unsubscribe from collecting dtmf digits, click here.
> 
> 
> View this message in context: Re: collecting dtmf digits
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From nazim.aghabayov at gmail.com  Fri Dec 17 12:09:51 2010
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Fri, 17 Dec 2010 13:09:51 +0400
Subject: [Freeswitch-users] collecting dtmf digits
In-Reply-To: <D0BEED38-8178-4A24-AEE6-816C0AA46780@ipeva.fr>
References: <1292319831757-5833942.post@n2.nabble.com>	<AANLkTimLDbh1tv8uKbWDRTbGfV5usVmqW8eQ8zFjz=0t@mail.gmail.com>	<AANLkTi=o-6h9q2UWwaYZ9hHuC7sMNTNYQs=nJm6yCaBH@mail.gmail.com>	<4D0A9655.7070007@gmail.com>	<AANLkTimHAZF8eZ=h4mrPcPRU9qB9j2GHvGb0rBsE86n7@mail.gmail.com>
	<D0BEED38-8178-4A24-AEE6-816C0AA46780@ipeva.fr>
Message-ID: <4D0B28DF.9070708@gmail.com>

Hello! You may try something like:

 session:execute("bridge",
                            (string.format("{ignore_early_media='true',
ringback='home/app/sounds/ringback-alaw.wav',
transfer_ringback='/home/app/sounds/ringback-alaw.wav',
hangup_after_bridge='true',
continue_on_fail='true'}sofia/external/%s at xxx.xxx.xxx.xxx",
                                           tostring(did) )) )

There are a lot of nice examples in lua&dialplan wiki pages.

On 12/17/2010 12:24 PM, David Ponzone wrote:
> You need to read the wiki intensively, and also, Internet is a very useful source of information to learn LUA.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 17/12/2010 ? 09:07, samir a ?crit :
>
>> Hello Nazim,
>>
>> When i use originate command it gives me below error :-
>>
>> 2010-12-17 13:29:30.964039 [ERR] mod_lua.cpp:182 Error in originate expected 4..4 args, got 1
>> stack traceback:
>>         [C]: in function 'originate'
>>         /usr/local/freeswitch/scripts/hello.lua:14: in main chunk
>>
>> I am using    
>>
>> digits = session:getDigits(5, "#", 3000);
>> freeswitch.consoleLog("info", "Got dtmf: ".. digits .."\n");
>> session.originate("sofia/external/(digits)@192.168.2.3");
>>
>>
>> any idea why is it ?
>>
>> Regards
>> Sam
>>
>>
>> On Fri, Dec 17, 2010 at 4:17 AM, Nazim Aghabayov [via freeswitch-users] <[hidden email]> wrote:
>> Why not using Lua script? Lua is quite powerful and you could benefit of 
>> using the luasql for db connectivity. 
>>
>> g_caller_id = session:getVariable("caller_id_number") 
>> g_caller_destination_number = session:getVariable("destination_number") 
>> g_caller_context = session:getVariable("context") 
>> g_caller_uuid = session:getVariable("uuid") 
>> ... 
>> session:flushDigits() 
>> digits = session:playAndGetDigits(4, 4, 3, 3000, "#", 
>>               wav_base .. langId .. "/" .. prompt_wav, "", 
>>       "[" .. allowed_digit1 .. "," .. allowed_digitN .. "]"); 
>> ... 
>> session:transfer(tostring(dest_ext), "XML", "public") 
>>
>> Regards, 
>> Nazim 
>>
>>
>> On 12/16/2010 07:29 AM, Sam wrote:
>>
>>> hi, 
>>>
>>> Its not an XML IVR but trying to collect digits when the user punches in 
>>> some digits after a playback sound file, 
>>> so that the digits would get stored into $(digits) and i can use those 
>>> digits for further processing or transferring 
>>> to other dial-plan as per the digits punched in. 
>>>
>>> Suppose a digits punched are 4567, I collect them in $(digits) and by 
>>> transfer function i do the transfer. 
>>>
>>> <action application="transfer" data="$(digits) default XML"/> 
>>>
>>> ----------------------------------------------------------------------------------------------- 
>>>
>>> Also i could see that in IVR.conf can do this by pressing 4567 and executing 
>>> a statement , but here also question 
>>> is how can i use it by storing those collected digits in an variable and the 
>>> using it further. 
>>>
>>> ---------------------------------------------------------------------------------------------- 
>>>
>>> I was thinking how in both the cases it could be done. 
>>>
>>> Thnx & Regds 
>>> Sam 
>>>
>>>
>>>
>>> On Thu, Dec 16, 2010 at 4:35 AM, Michael Collins <[hidden email]> wrote: 
>>>
>>>> Is this an XML IVR? 
>>>> -MC 
>>>>
>>>>
>>>> On Tue, Dec 14, 2010 at 1:43 AM, samir <[hidden email]> wrote:
>>>>> hello, 
>>>>>
>>>>> Is there any method to collect digits by a variable clause ? 
>>>>>
>>>>> Suppose i have a ivr playing and user inputs digits , i want to collect 
>>>>> the 
>>>>> dtmf digits and send it to a different application where that digits will 
>>>>> be 
>>>>> used for routing purpose. 
>>>>>
>>>>> Any ideas ! 
>>>>>
>>>>> Regards 
>>>>> Sam 
>>>>> -- 
>>>>> View this message in context: 
>>>>> http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5833942.html
>>>>> Sent from the freeswitch-users mailing list archive at Nabble.com. 
>>>>>
>>>>> _______________________________________________ 
>>>>> FreeSWITCH-users mailing list 
>>>>> [hidden email] 
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>> _______________________________________________ 
>>>> FreeSWITCH-users mailing list 
>>>> [hidden email] 
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________ 
>>> FreeSWITCH-users mailing list 
>>> [hidden email] 
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________ 
>> FreeSWITCH-users mailing list 
>> [hidden email] 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> View message @ http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5843837.html
>>
>> To unsubscribe from collecting dtmf digits, click here.
>>
>>
>> View this message in context: Re: collecting dtmf digits
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From eagle.antonio at gmail.com  Fri Dec 17 12:14:26 2010
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Fri, 17 Dec 2010 09:14:26 +0000
Subject: [Freeswitch-users] CallCenter Agent UUID-Standby
In-Reply-To: <E9D43200-E2B4-4272-86AD-4D642E3165ED@tlainvestments.com>
References: <AANLkTikNKi7Rzo8tKdMq+=iG7MaO304Q2CD15e6CJTs9@mail.gmail.com>
	<E9D43200-E2B4-4272-86AD-4D642E3165ED@tlainvestments.com>
Message-ID: <AANLkTin88ZknYRanO4sY2S56tPM2w7ZRTFKxfz0V4fE6@mail.gmail.com>

Hello Troy ,

But that part is present in the dial plan and as soon as a caller hangups an
hangup is also send to the agent and it doesn't seem to recover from that.

Full Debug Log And Dialplan in paste bin
http://pastebin.com/ADFfLPtR

Thank you all for all your time
Antonio


2010/12/17 Troy Anderson <freeswitch at tlainvestments.com>

> Hi Ant?nio,
>
> Take a look at this part of the wiki:
> http://wiki.freeswitch.org/wiki/Mod_callcenter#Callback
>
> Note the last line of the sample dial plan:
>
>     <action application="transfer" data="4099"/> <!-- Remove this if you just want to get a single call -->
>
> If you want to throw them back in the loop, you have to explicitly do it in
> your dial plan.
>
> -Troy
>
> On Dec 16, 2010, at 10:30 AM, Antonio Teixeira wrote:
>
> Good Afternoon.
>
> I'm leaving Asterisk ( yayyyyyyyyyyyyy) and was curious in trying out the
> Call Center feature,  uuid-standby , now i created a dialplan and I'm able
> to connect a caller to an agent.
> Now by my assumption as soon as the caller disconnects the agent should be
> trow back ( transfer) into the loop (4099 ext.) but instead i get an hangup
> and the sip client terminates the call.
>
> Any ideas ?
>
> Sample Dialplan
>
> <extension>
>   <condition field="destination_number" expression="^(4099)$">
>     <action application="set" data="res=${callcenter_config(agent set uuid ${caller_id_number}@${domain_name} '${uuid}')}" />
>
>
>
>     <action application="set" data="res=${callcenter_config(agent set type ${caller_id_number}@${domain_name} 'uuid-standby')}" />
>     <action application="set" data="res=${callcenter_config(agent set status ${caller_id_number}@${domain_name} 'Available (On Demand)')}" />
>
>
>
>     <action application="set" data="res=${callcenter_config(agent set state ${caller_id_number}@${domain_name} 'Waiting')}" />
>     <action application="set" data="cc_warning_tone=tone_stream://%(200,0,500,600,700)"/>
>
>
>
>     <action application="answer" />
>     <action application="playback" data="$${hold_music}"/>
>     <action application="transfer" data="4099"/> <!-- Remove this if you just want to get a single call -->
>
>
>
>   </condition>
> </extension>
>
>
> Thanks For Your Time
> Ant?nio Teixeira
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From thisjoy0528 at gmail.com  Fri Dec 17 13:07:20 2010
From: thisjoy0528 at gmail.com (joy this)
Date: Fri, 17 Dec 2010 18:07:20 +0800
Subject: [Freeswitch-users]  questions about VAD and echo cancellation
Message-ID: <AANLkTi=G-gm4V9zV1hLtFVJ4B6-tyQY1Rw=+LEPdnaXa@mail.gmail.com>

Dear all:



        I have questions about VAD and echo cancellation. My FS is Version
1.0.head (git-) under Windows XP. My soft-phone is X-Lite. I use earphones
and microphones for sip 1 which means the talking and hearing are separated;
on the other hand, I use NB for sip 2 which makes the talking and hearing in
the same place. Sip 1 (1000) calls sip 2 (1001) via FS. When I say something
via sip 1, the echo will occur, and the echo will only occur on sip 1.

        How do I cancel the echo? Buy a hardware card or something else? I
have two gateways which are Wellgate2644 and ata171m. Please give me some
suggestions.

        When I enabled VAD, I found something strange. In the start of the
session, I can hear the noise and echo on sip 1. If I say something via sip
2, the echo and the noise will disappear. Then the echo will not occur on
sip 1 only in a short time, about several seconds. The echo and the noise
will occur gradually if I say something via sip1. Is it a normal situation?



Sincerely yours,
Thisjoy.
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From u2nsam at gmail.com  Fri Dec 17 13:17:13 2010
From: u2nsam at gmail.com (Sam)
Date: Fri, 17 Dec 2010 15:47:13 +0530
Subject: [Freeswitch-users] collecting dtmf digits
In-Reply-To: <4D0B28DF.9070708@gmail.com>
References: <1292319831757-5833942.post@n2.nabble.com>
	<AANLkTimLDbh1tv8uKbWDRTbGfV5usVmqW8eQ8zFjz=0t@mail.gmail.com>
	<AANLkTi=o-6h9q2UWwaYZ9hHuC7sMNTNYQs=nJm6yCaBH@mail.gmail.com>
	<4D0A9655.7070007@gmail.com>
	<AANLkTimHAZF8eZ=h4mrPcPRU9qB9j2GHvGb0rBsE86n7@mail.gmail.com>
	<D0BEED38-8178-4A24-AEE6-816C0AA46780@ipeva.fr>
	<4D0B28DF.9070708@gmail.com>
Message-ID: <AANLkTinNOJZxrmk38SfeKrbHqJQOTGfXpuxoBUkdHi82@mail.gmail.com>

Thanx Nazim it was of great help,

i used the below syntax

session:execute("bridge",(string.format("{ignore_early_media='true',continue_on_fail='true'}sofia/external/%
s at 192.168.2.3",tostring(digits) )) ) ;

I am now checking out if i can use hangup causes for routing the call

LCause = session:hangupCause() ;

freeswitch.consoleLog("info", "hangupcause: ".. LCause .." );

then if i can use if then else statement for routing ... like

if ( LCause == "NO_USER_RESPONSE" ) then
       session:transfer(tostring(digits), "XML", "8888")

how is the idea ?

Regds



On Fri, Dec 17, 2010 at 2:39 PM, Nazim Aghabayov
<nazim.aghabayov at gmail.com>wrote:

> Hello! You may try something like:
>
>  session:execute("bridge",
>                            (string.format("{ignore_early_media='true',
> ringback='home/app/sounds/ringback-alaw.wav',
> transfer_ringback='/home/app/sounds/ringback-alaw.wav',
> hangup_after_bridge='true',
> continue_on_fail='true'}sofia/external/%s at xxx.xxx.xxx.xxx",
>                                           tostring(did) )) )
>
> There are a lot of nice examples in lua&dialplan wiki pages.
>
> On 12/17/2010 12:24 PM, David Ponzone wrote:
> > You need to read the wiki intensively, and also, Internet is a very
> useful source of information to learn LUA.
> >
> > David Ponzone  Direction Technique
> > email: david.ponzone at ipeva.fr
> > tel:      01 74 03 18 97
> > gsm:   06 66 98 76 34
> >
> > Service Client IPeva
> > tel:      0811 46 26 26
> > www.ipeva.fr  -   www.ipeva-studio.com
> >
> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >
> >
> >
> >
> > Le 17/12/2010 ? 09:07, samir a ?crit :
> >
> >> Hello Nazim,
> >>
> >> When i use originate command it gives me below error :-
> >>
> >> 2010-12-17 13:29:30.964039 [ERR] mod_lua.cpp:182 Error in originate
> expected 4..4 args, got 1
> >> stack traceback:
> >>         [C]: in function 'originate'
> >>         /usr/local/freeswitch/scripts/hello.lua:14: in main chunk
> >>
> >> I am using
> >>
> >> digits = session:getDigits(5, "#", 3000);
> >> freeswitch.consoleLog("info", "Got dtmf: ".. digits .."\n");
> >> session.originate("sofia/external/(digits)@192.168.2.3");
> >>
> >>
> >> any idea why is it ?
> >>
> >> Regards
> >> Sam
> >>
> >>
> >> On Fri, Dec 17, 2010 at 4:17 AM, Nazim Aghabayov [via freeswitch-users]
> <[hidden email]> wrote:
> >> Why not using Lua script? Lua is quite powerful and you could benefit of
> >> using the luasql for db connectivity.
> >>
> >> g_caller_id = session:getVariable("caller_id_number")
> >> g_caller_destination_number = session:getVariable("destination_number")
> >> g_caller_context = session:getVariable("context")
> >> g_caller_uuid = session:getVariable("uuid")
> >> ...
> >> session:flushDigits()
> >> digits = session:playAndGetDigits(4, 4, 3, 3000, "#",
> >>               wav_base .. langId .. "/" .. prompt_wav, "",
> >>       "[" .. allowed_digit1 .. "," .. allowed_digitN .. "]");
> >> ...
> >> session:transfer(tostring(dest_ext), "XML", "public")
> >>
> >> Regards,
> >> Nazim
> >>
> >>
> >> On 12/16/2010 07:29 AM, Sam wrote:
> >>
> >>> hi,
> >>>
> >>> Its not an XML IVR but trying to collect digits when the user punches
> in
> >>> some digits after a playback sound file,
> >>> so that the digits would get stored into $(digits) and i can use those
> >>> digits for further processing or transferring
> >>> to other dial-plan as per the digits punched in.
> >>>
> >>> Suppose a digits punched are 4567, I collect them in $(digits) and by
> >>> transfer function i do the transfer.
> >>>
> >>> <action application="transfer" data="$(digits) default XML"/>
> >>>
> >>>
> -----------------------------------------------------------------------------------------------
> >>>
> >>> Also i could see that in IVR.conf can do this by pressing 4567 and
> executing
> >>> a statement , but here also question
> >>> is how can i use it by storing those collected digits in an variable
> and the
> >>> using it further.
> >>>
> >>>
> ----------------------------------------------------------------------------------------------
> >>>
> >>> I was thinking how in both the cases it could be done.
> >>>
> >>> Thnx & Regds
> >>> Sam
> >>>
> >>>
> >>>
> >>> On Thu, Dec 16, 2010 at 4:35 AM, Michael Collins <[hidden email]>
> wrote:
> >>>
> >>>> Is this an XML IVR?
> >>>> -MC
> >>>>
> >>>>
> >>>> On Tue, Dec 14, 2010 at 1:43 AM, samir <[hidden email]> wrote:
> >>>>> hello,
> >>>>>
> >>>>> Is there any method to collect digits by a variable clause ?
> >>>>>
> >>>>> Suppose i have a ivr playing and user inputs digits , i want to
> collect
> >>>>> the
> >>>>> dtmf digits and send it to a different application where that digits
> will
> >>>>> be
> >>>>> used for routing purpose.
> >>>>>
> >>>>> Any ideas !
> >>>>>
> >>>>> Regards
> >>>>> Sam
> >>>>> --
> >>>>> View this message in context:
> >>>>>
> http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5833942.html
> >>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> [hidden email]
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> [hidden email]
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> [hidden email]
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> [hidden email]
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> View message @
> http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5843837.html
> >>
> >> To unsubscribe from collecting dtmf digits, click here.
> >>
> >>
> >> View this message in context: Re: collecting dtmf digits
> >> Sent from the freeswitch-users mailing list archive at Nabble.com.
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From thisjoy0528 at gmail.com  Fri Dec 17 13:20:05 2010
From: thisjoy0528 at gmail.com (joy this)
Date: Fri, 17 Dec 2010 18:20:05 +0800
Subject: [Freeswitch-users]  questions about google voice and dingaling
Message-ID: <AANLkTimyXDinQiDZs-THu_qVwx_nKhsMtov6sp-KuGKA@mail.gmail.com>

Dear all:



        I have a question about google voice. I can call a number via google
voice as following: sip => FS => GVoice => Phone. The question is that, I
can call two different phone numbers with two different sip accounts via the
same GVoice account. Is it a normal situation? I am worried about that
google will ban my IP or account if I call several numbers simultaneously
via one GVoice account.

        I tried to login with two GVoice accounts. My client.xml is as
following:

*<include>*

        <profile type="client">

                <param name="name" value="GV1"/>

                <param name="login" value=" GV1 at gmail.com/gtalk"/>

                <param name="password" value="PWD1"/>

                Other settings?

        </profile>

        <profile type="client">

                <param name="name" value="GV2"/>

                <param name="login" value=" GV2 at gmail.com/gtalk"/>

                <param name="password" value="PWD2"/>

                Other settings?

        </profile>

*</include>*

*
*

FS will crash. The error message is:

Assertion failed!

Program: C:\FreeSWITCH\FreeSwitch.exe

File: ../../../src/libgcrypt-1.4.6/src/ath.c

Line: 193

Expression: *lock==MUTEX_UNLOCKED

Sincerely yours,

Thisjoy.
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From Nabble at slickdeals.endjunk.com  Fri Dec 17 16:31:40 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 17 Dec 2010 05:31:40 -0800 (PST)
Subject: [Freeswitch-users] questions about google voice and dingaling
In-Reply-To: <AANLkTimyXDinQiDZs-THu_qVwx_nKhsMtov6sp-KuGKA@mail.gmail.com>
References: <AANLkTimyXDinQiDZs-THu_qVwx_nKhsMtov6sp-KuGKA@mail.gmail.com>
Message-ID: <1292592700991-5845263.post@n2.nabble.com>



joy this wrote:
> The question is that, I can call two different phone numbers with two
> different sip accounts via the same GVoice account.
Your configuration showed you have two GV accounts. R U sure you are ONLY
using a single GV account to place two concurrent calls?

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/questions-about-google-voice-and-dingaling-tp5845051p5845263.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From frank at telonium.com  Fri Dec 17 17:14:41 2010
From: frank at telonium.com (Frank Park)
Date: Fri, 17 Dec 2010 09:14:41 -0500
Subject: [Freeswitch-users] call pickup
In-Reply-To: <AANLkTimZeMcBbDkW-W=O8661Vx4ZXququv8qPtLFTDgJ@mail.gmail.com>
References: <1292391773589-5837233.post@n2.nabble.com>
	<665DE8BC-65A7-46D2-8F8E-002649C59C21@ipeva.fr>
	<AANLkTimZeMcBbDkW-W=O8661Vx4ZXququv8qPtLFTDgJ@mail.gmail.com>
Message-ID: <AANLkTi=KM=Nhj-D7SyyVi1hMfT8Luja3chcUfg=NwByd@mail.gmail.com>

Have you defined the "callgroup" variable in the directory?
<variable name="callgroup" value="VALUE"/>



----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----


On Wed, Dec 15, 2010 at 10:34 PM, Sam <u2nsam at gmail.com> wrote:
>
> Yes David ,
>
> You are right from yesterday i was trying to post it to the forum and it was giving me message that your post is not posted in the forum.
> And at last it did with multiple emails !! may be it could have happened as i created the account day before yesterday and it takes time.
> Sorry all for any inconvenience caused !
>
> Regds
>
> On Wed, Dec 15, 2010 at 3:47 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
>>
>> Samir,
>> Perhaps next time, you could think before sending your mail.
>> It will avoid you sending multiple emails for the same question.
>> David Ponzone ?Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel: ? ? ?01 74 03 18 97
>> gsm: ? 06 66 98 76 34
>> Service Client?IPeva
>> tel: ? ? ?0811 46 26 26
>> www.ipeva.fr? -? ?www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>> Le 15/12/2010 ? 06:42, samir a ?crit :
>>
>> hello, I am trying call "pickup" for the incoming calls with intercept function but it do not seems to work, am i missing something ? Regards Sam
>> ________________________________
>> View this message in context: call pickup
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From moises.silva at gmail.com  Fri Dec 17 17:21:46 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 17 Dec 2010 09:21:46 -0500
Subject: [Freeswitch-users] freetdm and ring_ready
In-Reply-To: <AANLkTim3DVdUnYVsGprgXggQeGU=Pq6Lyp9WNeQ_mB8L@mail.gmail.com>
References: <AANLkTikfyzKUZhb=2KS9=2v_Fmk=mj0=ruz328aTceM3@mail.gmail.com>
	<AANLkTimO_0JV30SoC62we=aX3Lh2-nV5wQ+UsL7oN5Gm@mail.gmail.com>
	<AANLkTin2FNYvy9_VXo_rp9FvZJ1a7avWGBwBk1VU-T5D@mail.gmail.com>
	<AANLkTiniZ_6utTHyWQQJx2V8QnKur27JaaP2NBsmu+rS@mail.gmail.com>
	<AANLkTim3DVdUnYVsGprgXggQeGU=Pq6Lyp9WNeQ_mB8L@mail.gmail.com>
Message-ID: <AANLkTikqwHiLOJRzf0m_7qjxEtYgsj9xTwWW9roNDytw@mail.gmail.com>

On Thu, Dec 16, 2010 at 1:14 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
Moy,
> So is this a case of relying on incorrect behavior as a "feature"? Just
> confirming. I'll make sure that the pre_answer app is mentioned in the
> FreeTDM wiki pages where appropriate.
>
>
Yes, that's what it looks like to me. I just would like confirmation from
Neil that "pre_answer" works as expected.

Neil: If "pre_answer" does not work for FreeTDM, please open a bug in jira.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com
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From infos at madovsky.org  Fri Dec 17 18:35:04 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 17 Dec 2010 10:35:04 -0500
Subject: [Freeswitch-users] members audio conference
Message-ID: <223AD13D879E4B19B15AEEDFC387386A@e1705>

when the first member creates and enters in a new conference
everything is ok. but if a new memeber enters there is no audio
in the conference, unless the ivr.

I have a very simple conference dialplan like this

        <extension name="create_conference">
                <condition field="${sip_to_uri}" expression="^000(\d{10,15})@$${domain}$">
                        <action application="set" data="instant_ringback=true"/>
                        <action application="ring_ready"/>
                        <action application="sleep" data="3000"/>
                        <action application="answer"/>
                        <action application="conference" data="$1-${domain_name}@wideband"/>
                </condition>
        </extension>


I tried to add
<action application="export" data="absolute_codec_string=speex at 16000k"/>

but no success

Thanks
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From thisjoy0528 at gmail.com  Fri Dec 17 18:56:19 2010
From: thisjoy0528 at gmail.com (joy this)
Date: Fri, 17 Dec 2010 23:56:19 +0800
Subject: [Freeswitch-users] questions about google voice and dingaling
In-Reply-To: <1292592700991-5845263.post@n2.nabble.com>
References: <AANLkTimyXDinQiDZs-THu_qVwx_nKhsMtov6sp-KuGKA@mail.gmail.com>
	<1292592700991-5845263.post@n2.nabble.com>
Message-ID: <AANLkTin9rdRQcSB2yNcROS4RpBSOSdOrMQo+01CoZrWc@mail.gmail.com>

I found the situation yesterday. I am sure that there is only one account.
I try to add another account today, and FS crashed.

Sincerely yours,

Thisjoy.
2010/12/17 mazilo <Nabble at slickdeals.endjunk.com>

>
>
> joy this wrote:
> > The question is that, I can call two different phone numbers with two
> > different sip accounts via the same GVoice account.
> Your configuration showed you have two GV accounts. R U sure you are ONLY
> using a single GV account to place two concurrent calls?
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/questions-about-google-voice-and-dingaling-tp5845051p5845263.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From u2nsam at gmail.com  Fri Dec 17 19:03:17 2010
From: u2nsam at gmail.com (Sam)
Date: Fri, 17 Dec 2010 21:33:17 +0530
Subject: [Freeswitch-users] call pickup
In-Reply-To: <AANLkTi=KM=Nhj-D7SyyVi1hMfT8Luja3chcUfg=NwByd@mail.gmail.com>
References: <1292391773589-5837233.post@n2.nabble.com>
	<665DE8BC-65A7-46D2-8F8E-002649C59C21@ipeva.fr>
	<AANLkTimZeMcBbDkW-W=O8661Vx4ZXququv8qPtLFTDgJ@mail.gmail.com>
	<AANLkTi=KM=Nhj-D7SyyVi1hMfT8Luja3chcUfg=NwByd@mail.gmail.com>
Message-ID: <AANLkTimsAshk3=k8BvbWX=XeiAoVXCO5vC8kG50T3_KQ@mail.gmail.com>

Yes those are define for respective extension groups.

Was the dialplan correct or anything needs to be updated ?

Thanks & regds
Sam

On Fri, Dec 17, 2010 at 7:44 PM, Frank Park <frank at telonium.com> wrote:

> Have you defined the "callgroup" variable in the directory?
> <variable name="callgroup" value="VALUE"/>
>
>
>
> ----=======================----
> Frank Park
> Telonium Communications, LLC
> frank at telonium.com
> http://www.telonium.com
> Follow Us on Twitter: @GetTelonium
> 404-566-8888 x1001 Office
> 404-939-4242 Cell
> ----=======================----
>
>
> On Wed, Dec 15, 2010 at 10:34 PM, Sam <u2nsam at gmail.com> wrote:
> >
> > Yes David ,
> >
> > You are right from yesterday i was trying to post it to the forum and it
> was giving me message that your post is not posted in the forum.
> > And at last it did with multiple emails !! may be it could have happened
> as i created the account day before yesterday and it takes time.
> > Sorry all for any inconvenience caused !
> >
> > Regds
> >
> > On Wed, Dec 15, 2010 at 3:47 PM, David Ponzone <david.ponzone at ipeva.fr>
> wrote:
> >>
> >> Samir,
> >> Perhaps next time, you could think before sending your mail.
> >> It will avoid you sending multiple emails for the same question.
> >> David Ponzone  Direction Technique
> >> email: david.ponzone at ipeva.fr
> >> tel:      01 74 03 18 97
> >> gsm:   06 66 98 76 34
> >> Service Client IPeva
> >> tel:      0811 46 26 26
> >> www.ipeva.fr  -   www.ipeva-studio.com
> >> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >>
> >>
> >>
> >> Le 15/12/2010 ? 06:42, samir a ?crit :
> >>
> >> hello, I am trying call "pickup" for the incoming calls with intercept
> function but it do not seems to work, am i missing something ? Regards Sam
> >> ________________________________
> >> View this message in context: call pickup
> >> Sent from the freeswitch-users mailing list archive at Nabble.com.
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From u2nsam at gmail.com  Fri Dec 17 19:17:23 2010
From: u2nsam at gmail.com (Sam)
Date: Fri, 17 Dec 2010 21:47:23 +0530
Subject: [Freeswitch-users] routing via hangup_cause
In-Reply-To: <AANLkTiko+9M0MNeKgZO2aprbZg8LevA_7pHMSsKqn=AE@mail.gmail.com>
References: <1292386719584-5837136.post@n2.nabble.com>
	<AANLkTiko+9M0MNeKgZO2aprbZg8LevA_7pHMSsKqn=AE@mail.gmail.com>
Message-ID: <AANLkTikXHGsM_dqz2VGLVNmbv74sXMfShGAWBaE6cT7E@mail.gmail.com>

Hello,

How to use log function in lua, i am using as below.

LCause = session:hangupCause() ;

freeswitch.consoleLog("info", "hangupcause: ".. LCause .." );

I know i am wrong somewhere .

Regards
Sam



On Fri, Dec 17, 2010 at 1:28 AM, samir [via freeswitch-users] <
ml-node+5843222-471532028-292429 at n2.nabble.com<ml-node%2B5843222-471532028-292429 at n2.nabble.com>
> wrote:

>
> I am not able to fetch the hangup cause in the dial-plan by using log
> function .
> I would like to use that hangup cause variable for further routing so that
> the routing would be conditional to hangup cause.
>
> Regards
> Sam
>
> On Wed, Dec 15, 2010 at 9:48 AM, samir <[hidden email]<http://user/SendEmail.jtp?type=node&node=5843222&i=0>
> > wrote:
>
>>
>> Hello friends,
>>
>> was trying to create a routing rule to to route calls by accounting to the
>> hangup causes,
>>
>> I have written below syntax but it fails to give the cause code to the
>> varriable for routing.
>>
>> <extension name="refer">
>>               <condition field="destination_number" expression="^1234$">
>>
>>
>>       <action application="bridge" data="[hidden email]<http://user/SendEmail.jtp?type=node&node=5843222&i=1>
>> "/>
>>
>>       <action application="set" data="hangup_after_bridge=false"/>
>>       <action application="set" data="continue_on_fail=true"/>
>>       <action application="log" data="1 B-leg hangup cause:
>> ${bridge_hangup_cause}"/>
>>       <action application="transfer" data="${bridge_hangup_cause} XML
>> conbridge"/>
>>
>>               </condition>
>>
>>
>>   </extension>
>>
>>       <extension name="no user">
>>               <condition field="destination_number"
>> expression="^(NO_USER_RESPONSE)$">
>>               <action application="bridge"
>> data="[hidden email]<http://user/SendEmail.jtp?type=node&node=5843222&i=2>
>> "/>
>>
>>
>>               </condition>
>>       </extension>
>>
>>
>> here the ${bridge_hangup_cause} is not getting executed. Am I doing it
>> right
>> or is there any other way to do it.
>>
>> Regards
>> Sam
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/routing-via-hangup-cause-tp5837136p5837136.html<http://freeswitch-users.2379917.n2.nabble.com/routing-via-hangup-cause-tp5837136p5837136.html?by-user=t>
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5843222&i=3>
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> [hidden email] <http://user/SendEmail.jtp?type=node&node=5843222&i=4>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
>  View message @
> http://freeswitch-users.2379917.n2.nabble.com/routing-via-hangup-cause-tp5837136p5843222.html
> To unsubscribe from routing via hangup_cause, click here<http://freeswitch-users.2379917.n2.nabble.com/template/NamlServlet.jtp?macro=unsubscribe_by_code&node=5837136&code=dTJuc2FtQGdtYWlsLmNvbXw1ODM3MTM2fC0xMTk1MzYwMDk5>.
>
>
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From joaocarlosleme at gmail.com  Fri Dec 17 19:48:16 2010
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Fri, 17 Dec 2010 08:48:16 -0800
Subject: [Freeswitch-users] Having problems (no sound) when trying to LogIn
	from Home to Freeswitch
Message-ID: <AANLkTinSYxPCBNEMM1OuPcKw7LKA266XVuQW7m-P1LN1@mail.gmail.com>

Hi there! First time here.
I'm having problems (no sound) when trying to LogIn from Home to Freeswitch.
In the office it all works fine, even when I use the external IP as Domain
on X-Lite. I'm trying to go for internal profile on port 5060 as I want to
be just like if I was there in the office. It actually works, I can make
calls, receive etc but NO SOUND.
What I also did was to change the domain on var.xml to the external ip. I
also forwarded the ports to the computer where Freeswitch according to
Firewall info on docs.
Thanks a lot,
John
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From rafonline at hotmail.com  Sat Dec 18 00:19:11 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Fri, 17 Dec 2010 21:19:11 +0000
Subject: [Freeswitch-users] audio quality issue
Message-ID: <COL122-W33885B610991D11E824AB6B5160@phx.gbl>




Hi, 

I recently installed freeswitch 1.0.6 on centos 5.5 (hosted on xen server).  I feel as if the audio quality is not as good as what it should be (6-7 out of 10).  This is apparent in calling the default voicemail IVR, sometimes I have issues leaving a message, sometimes I hear some weird noises.  However, I have no issues with the quality of the recorded voicemail message I leave, it sounds fine on playback.

Is this an issue with my phones (Flexor 500)?  I have a softphone which doesn't seem as bad but still has issues.

I also get the following message on startup of freeswitch:

2010-12-17 21:17:39.138958 [WARNING] switch_time.c:206 Timer resolution of 4000 microseconds detected!
Do you have your kernel timer frequency set to lower than 1,000Hz? You may experience audio problems.

I assume its related. 

Any help will be much appreciated.

Cheers

Raf
 		 	   		  
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From anthony.minessale at gmail.com  Sat Dec 18 00:22:25 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 17 Dec 2010 15:22:25 -0600
Subject: [Freeswitch-users] audio quality issue
In-Reply-To: <COL122-W33885B610991D11E824AB6B5160@phx.gbl>
References: <COL122-W33885B610991D11E824AB6B5160@phx.gbl>
Message-ID: <AANLkTi=JQiiySBzLfU-vWM7_LRQd73Zy=FyCTj-aebxM@mail.gmail.com>

1) Please try the development snapshot.
2) You need a kernel running at 1khz for best results, we do not
officially support xen, (openVZ works fine)


On Fri, Dec 17, 2010 at 3:19 PM, Rafqat . <rafonline at hotmail.com> wrote:
>
>
> Hi,
>
> I recently installed freeswitch 1.0.6 on centos 5.5 (hosted on xen server).
> I feel as if the audio quality is not as good as what it should be (6-7 out
> of 10).? This is apparent in calling the default voicemail IVR, sometimes I
> have issues leaving a message, sometimes I hear some weird noises.? However,
> I have no issues with the quality of the recorded voicemail message I leave,
> it sounds fine on playback.
>
> Is this an issue with my phones (Flexor 500)?? I have a softphone which
> doesn't seem as bad but still has issues.
>
> I also get the following message on startup of freeswitch:
>
> 2010-12-17 21:17:39.138958 [WARNING] switch_time.c:206 Timer resolution of
> 4000 microseconds detected!
> Do you have your kernel timer frequency set to lower than 1,000Hz? You may
> experience audio problems.
>
> I assume its related.
>
> Any help will be much appreciated.
>
> Cheers
>
> Raf
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From rafonline at hotmail.com  Sat Dec 18 00:22:48 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Fri, 17 Dec 2010 21:22:48 +0000
Subject: [Freeswitch-users] audio quality issue
In-Reply-To: <COL122-W33885B610991D11E824AB6B5160@phx.gbl>
References: <COL122-W33885B610991D11E824AB6B5160@phx.gbl>
Message-ID: <COL122-W5833A44AAF7FDFD440BCBB5160@phx.gbl>



btw.  I have no audio issues when making calls between phones through freeswitch (voice quality is very good).



From: rafonline at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Date: Fri, 17 Dec 2010 21:19:11 +0000
Subject: [Freeswitch-users] audio quality issue










Hi, 

I recently installed freeswitch 1.0.6 on centos 5.5 (hosted on xen server).  I feel as if the audio quality is not as good as what it should be (6-7 out of 10).  This is apparent in calling the default voicemail IVR, sometimes I have issues leaving a message, sometimes I hear some weird noises.  However, I have no issues with the quality of the recorded voicemail message I leave, it sounds fine on playback.

Is this an issue with my phones (Flexor 500)?  I have a softphone which doesn't seem as bad but still has issues.

I also get the following message on startup of freeswitch:

2010-12-17 21:17:39.138958 [WARNING] switch_time.c:206 Timer resolution of 4000 microseconds detected!
Do you have your kernel timer frequency set to lower than 1,000Hz? You may experience audio problems.

I assume its related. 

Any help will be much appreciated.

Cheers

Raf
 		 	   		  

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
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From anthony.minessale at gmail.com  Sat Dec 18 00:28:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 17 Dec 2010 15:28:49 -0600
Subject: [Freeswitch-users] audio quality issue
In-Reply-To: <COL122-W5833A44AAF7FDFD440BCBB5160@phx.gbl>
References: <COL122-W33885B610991D11E824AB6B5160@phx.gbl>
	<COL122-W5833A44AAF7FDFD440BCBB5160@phx.gbl>
Message-ID: <AANLkTinEhrDUHJCfr7RC=9Qm-uYhpFAsx-RJcc7bD4FC@mail.gmail.com>

if you insist to remain on 1.0.6 try it with -vm -nocal cmd line options
as i already stated you should try the development snapshot

On Fri, Dec 17, 2010 at 3:22 PM, Rafqat . <rafonline at hotmail.com> wrote:
>
> btw.? I have no audio issues when making calls between phones through
> freeswitch (voice quality is very good).
>
>
>
> ________________________________
> From: rafonline at hotmail.com
> To: freeswitch-users at lists.freeswitch.org
> Date: Fri, 17 Dec 2010 21:19:11 +0000
> Subject: [Freeswitch-users] audio quality issue
>
>
>
> Hi,
>
> I recently installed freeswitch 1.0.6 on centos 5.5 (hosted on xen server).
> I feel as if the audio quality is not as good as what it should be (6-7 out
> of 10).? This is apparent in calling the default voicemail IVR, sometimes I
> have issues leaving a message, sometimes I hear some weird noises.? However,
> I have no issues with the quality of the recorded voicemail message I leave,
> it sounds fine on playback.
>
> Is this an issue with my phones (Flexor 500)?? I have a softphone which
> doesn't seem as bad but still has issues.
>
> I also get the following message on startup of freeswitch:
>
> 2010-12-17 21:17:39.138958 [WARNING] switch_time.c:206 Timer resolution of
> 4000 microseconds detected!
> Do you have your kernel timer frequency set to lower than 1,000Hz? You may
> experience audio problems.
>
> I assume its related.
>
> Any help will be much appreciated.
>
> Cheers
>
> Raf
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From tgraziano at myitdepartment.net  Sat Dec 18 00:29:31 2010
From: tgraziano at myitdepartment.net (Tony Graziano)
Date: Fri, 17 Dec 2010 16:29:31 -0500
Subject: [Freeswitch-users] audio quality issue
In-Reply-To: <COL122-W5833A44AAF7FDFD440BCBB5160@phx.gbl>
References: <COL122-W33885B610991D11E824AB6B5160@phx.gbl>
	<COL122-W5833A44AAF7FDFD440BCBB5160@phx.gbl>
Message-ID: <AANLkTinCWk=fPmq09vkpdt4wfm=a6sO3=MBUWEM4epDd@mail.gmail.com>

On Fri, Dec 17, 2010 at 4:22 PM, Rafqat . <rafonline at hotmail.com> wrote:
>
> btw.? I have no audio issues when making calls between phones through
> freeswitch (voice quality is very good).
>
>
>
Between phones, when the media is peer to peer will be fine. When the
media is anchored to a media server running on hardware you need to
ensure the media server is not having any latency issues. Hence
Anthony's 2 points.

Media services (any platform/engine) having any latency or jitter
issues (due to timing especially) will result in what you have
experienced.

Tony


From rafonline at hotmail.com  Sat Dec 18 00:30:11 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Fri, 17 Dec 2010 21:30:11 +0000
Subject: [Freeswitch-users] audio quality issue
In-Reply-To: <AANLkTinEhrDUHJCfr7RC=9Qm-uYhpFAsx-RJcc7bD4FC@mail.gmail.com>
References: <COL122-W33885B610991D11E824AB6B5160@phx.gbl>,
	<COL122-W5833A44AAF7FDFD440BCBB5160@phx.gbl>,
	<AANLkTinEhrDUHJCfr7RC=9Qm-uYhpFAsx-RJcc7bD4FC@mail.gmail.com>
Message-ID: <COL122-W5661F836F7A1ED2E6D4FB6B5160@phx.gbl>



I will try the dev snapshot. 

cheers

Raf

> Date: Fri, 17 Dec 2010 15:28:49 -0600
> From: anthony.minessale at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] audio quality issue
> 
> if you insist to remain on 1.0.6 try it with -vm -nocal cmd line options
> as i already stated you should try the development snapshot
> 
> On Fri, Dec 17, 2010 at 3:22 PM, Rafqat . <rafonline at hotmail.com> wrote:
> >
> > btw.  I have no audio issues when making calls between phones through
> > freeswitch (voice quality is very good).
> >
> >
> >
> > ________________________________
> > From: rafonline at hotmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Fri, 17 Dec 2010 21:19:11 +0000
> > Subject: [Freeswitch-users] audio quality issue
> >
> >
> >
> > Hi,
> >
> > I recently installed freeswitch 1.0.6 on centos 5.5 (hosted on xen server).
> > I feel as if the audio quality is not as good as what it should be (6-7 out
> > of 10).  This is apparent in calling the default voicemail IVR, sometimes I
> > have issues leaving a message, sometimes I hear some weird noises.  However,
> > I have no issues with the quality of the recorded voicemail message I leave,
> > it sounds fine on playback.
> >
> > Is this an issue with my phones (Flexor 500)?  I have a softphone which
> > doesn't seem as bad but still has issues.
> >
> > I also get the following message on startup of freeswitch:
> >
> > 2010-12-17 21:17:39.138958 [WARNING] switch_time.c:206 Timer resolution of
> > 4000 microseconds detected!
> > Do you have your kernel timer frequency set to lower than 1,000Hz? You may
> > experience audio problems.
> >
> > I assume its related.
> >
> > Any help will be much appreciated.
> >
> > Cheers
> >
> > Raf
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
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From msc at freeswitch.org  Sat Dec 18 00:51:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 17 Dec 2010 13:51:00 -0800
Subject: [Freeswitch-users] routing via hangup_cause
In-Reply-To: <AANLkTiko+9M0MNeKgZO2aprbZg8LevA_7pHMSsKqn=AE@mail.gmail.com>
References: <1292386719584-5837136.post@n2.nabble.com>
	<AANLkTiko+9M0MNeKgZO2aprbZg8LevA_7pHMSsKqn=AE@mail.gmail.com>
Message-ID: <AANLkTi=-7N=f7PdSurgmxvr-9Y20Wk0G5N4v1+Z3zcmV@mail.gmail.com>

You're almost there. First off, you can delete this line:
<action application="set" data="hangup_after_bridge=false"/>
That's the default behavior and you have to set it prior to the bridge
anyway.

Move this line before the bridge:
<action application="set" data="continue_on_fail=true"/>
Otherwise it won't have any effect.

The other stuff should work if the b-leg hangs up first, but not if the
a-leg does.

-MC


On Thu, Dec 16, 2010 at 2:28 AM, Sam <u2nsam at gmail.com> wrote:

>
> I am not able to fetch the hangup cause in the dial-plan by using log
> function .
> I would like to use that hangup cause variable for further routing so that
> the routing would be conditional to hangup cause.
>
> Regards
> Sam
>
> On Wed, Dec 15, 2010 at 9:48 AM, samir <u2nsam at gmail.com> wrote:
>
>>
>> Hello friends,
>>
>> was trying to create a routing rule to to route calls by accounting to the
>> hangup causes,
>>
>> I have written below syntax but it fails to give the cause code to the
>> varriable for routing.
>>
>> <extension name="refer">
>>               <condition field="destination_number" expression="^1234$">
>>
>>
>>       <action application="bridge" data="sofia/external/1234 at X.X.X.X"/>
>>       <action application="set" data="hangup_after_bridge=false"/>
>>       <action application="set" data="continue_on_fail=true"/>
>>       <action application="log" data="1 B-leg hangup cause:
>> ${bridge_hangup_cause}"/>
>>       <action application="transfer" data="${bridge_hangup_cause} XML
>> conbridge"/>
>>
>>               </condition>
>>
>>
>>   </extension>
>>
>>       <extension name="no user">
>>               <condition field="destination_number"
>> expression="^(NO_USER_RESPONSE)$">
>>               <action application="bridge"
>> data="sofia/external/4567 at X.X.X.X"/>
>>
>>               </condition>
>>       </extension>
>>
>>
>> here the ${bridge_hangup_cause} is not getting executed. Am I doing it
>> right
>> or is there any other way to do it.
>>
>> Regards
>> Sam
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/routing-via-hangup-cause-tp5837136p5837136.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris at cloudtel.com  Sat Dec 18 02:05:09 2010
From: chris at cloudtel.com (Chris Burns)
Date: Fri, 17 Dec 2010 18:05:09 -0500
Subject: [Freeswitch-users] audio quality issue
In-Reply-To: <COL122-W5661F836F7A1ED2E6D4FB6B5160@phx.gbl>
References: <COL122-W33885B610991D11E824AB6B5160@phx.gbl>
	<COL122-W5833A44AAF7FDFD440BCBB5160@phx.gbl>
	<AANLkTinEhrDUHJCfr7RC=9Qm-uYhpFAsx-RJcc7bD4FC@mail.gmail.com>
	<COL122-W5661F836F7A1ED2E6D4FB6B5160@phx.gbl>
Message-ID: <AANLkTindj9jGDMYn9tLuuN1c3uSX=vC19s_F7pCmBHvC@mail.gmail.com>

Improve the kernel and in your situation you will most likely improve your
audio. You can't expect perfect audio when the thread handling RTP is waking
up later than its designated time (after sleeping for the codec timing). The
switch tests for the possibility of this issue on startup when it calibrates
your clock offset ... thus it warned you :)

"-nocal" will simply skip the process of recognizing and warning that your
kernel timer is slow, but will not alter performance in your situation
(unless I am missing something)

On Fri, Dec 17, 2010 at 4:30 PM, Rafqat . <rafonline at hotmail.com> wrote:

>
> I will try the dev snapshot.
>
> cheers
>
> Raf
>
> > Date: Fri, 17 Dec 2010 15:28:49 -0600
> > From: anthony.minessale at gmail.com
>
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] audio quality issue
>
> >
> > if you insist to remain on 1.0.6 try it with -vm -nocal cmd line options
> > as i already stated you should try the development snapshot
> >
> > On Fri, Dec 17, 2010 at 3:22 PM, Rafqat . <rafonline at hotmail.com> wrote:
> > >
> > > btw.  I have no audio issues when making calls between phones through
> > > freeswitch (voice quality is very good).
> > >
> > >
> > >
> > > ________________________________
> > > From: rafonline at hotmail.com
> > > To: freeswitch-users at lists.freeswitch.org
> > > Date: Fri, 17 Dec 2010 21:19:11 +0000
> > > Subject: [Freeswitch-users] audio quality issue
> > >
> > >
> > >
> > > Hi,
> > >
> > > I recently installed freeswitch 1.0.6 on centos 5.5 (hosted on xen
> server).
> > > I feel as if the audio quality is not as good as what it should be (6-7
> out
> > > of 10).  This is apparent in calling the default voicemail IVR,
> sometimes I
> > > have issues leaving a message, sometimes I hear some weird noises.
> However,
> > > I have no issues with the quality of the recorded voicemail message I
> leave,
> > > it sounds fine on playback.
> > >
> > > Is this an issue with my phones (Flexor 500)?  I have a softphone which
> > > doesn't seem as bad but still has issues.
> > >
> > > I also get the following message on startup of freeswitch:
> > >
> > > 2010-12-17 21:17:39.138958 [WARNING] switch_time.c:206 Timer resolution
> of
> > > 4000 microseconds detected!
> > > Do you have your kernel timer frequency set to lower than 1,000Hz? You
> may
> > > experience audio problems.
> > >
> > > I assume its related.
> > >
> > > Any help will be much appreciated.
> > >
> > > Cheers
> > >
> > > Raf
> > >
> > > _______________________________________________ FreeSWITCH-users
> mailing
> > > list FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <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
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris at cloudtel.com  Sat Dec 18 02:27:36 2010
From: chris at cloudtel.com (Chris Burns)
Date: Fri, 17 Dec 2010 18:27:36 -0500
Subject: [Freeswitch-users] call pickup
In-Reply-To: <AANLkTimsAshk3=k8BvbWX=XeiAoVXCO5vC8kG50T3_KQ@mail.gmail.com>
References: <1292391773589-5837233.post@n2.nabble.com>
	<665DE8BC-65A7-46D2-8F8E-002649C59C21@ipeva.fr>
	<AANLkTimZeMcBbDkW-W=O8661Vx4ZXququv8qPtLFTDgJ@mail.gmail.com>
	<AANLkTi=KM=Nhj-D7SyyVi1hMfT8Luja3chcUfg=NwByd@mail.gmail.com>
	<AANLkTimsAshk3=k8BvbWX=XeiAoVXCO5vC8kG50T3_KQ@mail.gmail.com>
Message-ID: <AANLkTinHoVMzeP6eq34grTQoqB-7u3FJrrn+1=7os9SD@mail.gmail.com>

You are doing a hash select to determine the data sent the the intercept app
... is it selecting out meaningful data that you are inserting elsewhere?
The intercept example in the default dialplan works in concert with other
elements which may not be present in your own.

On Fri, Dec 17, 2010 at 11:03 AM, Sam <u2nsam at gmail.com> wrote:

> Yes those are define for respective extension groups.
>
> Was the dialplan correct or anything needs to be updated ?
>
> Thanks & regds
> Sam
>
>
> On Fri, Dec 17, 2010 at 7:44 PM, Frank Park <frank at telonium.com> wrote:
>
>> Have you defined the "callgroup" variable in the directory?
>> <variable name="callgroup" value="VALUE"/>
>>
>>
>>
>> ----=======================----
>> Frank Park
>> Telonium Communications, LLC
>> frank at telonium.com
>> http://www.telonium.com
>> Follow Us on Twitter: @GetTelonium
>> 404-566-8888 x1001 Office
>> 404-939-4242 Cell
>> ----=======================----
>>
>>
>> On Wed, Dec 15, 2010 at 10:34 PM, Sam <u2nsam at gmail.com> wrote:
>> >
>> > Yes David ,
>> >
>> > You are right from yesterday i was trying to post it to the forum and it
>> was giving me message that your post is not posted in the forum.
>> > And at last it did with multiple emails !! may be it could have happened
>> as i created the account day before yesterday and it takes time.
>> > Sorry all for any inconvenience caused !
>> >
>> > Regds
>> >
>> > On Wed, Dec 15, 2010 at 3:47 PM, David Ponzone <david.ponzone at ipeva.fr>
>> wrote:
>> >>
>> >> Samir,
>> >> Perhaps next time, you could think before sending your mail.
>> >> It will avoid you sending multiple emails for the same question.
>> >> David Ponzone  Direction Technique
>> >> email: david.ponzone at ipeva.fr
>> >> tel:      01 74 03 18 97
>> >> gsm:   06 66 98 76 34
>> >> Service Client IPeva
>> >> tel:      0811 46 26 26
>> >> www.ipeva.fr  -   www.ipeva-studio.com
>> >> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> >>
>> >>
>> >>
>> >> Le 15/12/2010 ? 06:42, samir a ?crit :
>> >>
>> >> hello, I am trying call "pickup" for the incoming calls with intercept
>> function but it do not seems to work, am i missing something ? Regards Sam
>> >> ________________________________
>> >> View this message in context: call pickup
>> >> Sent from the freeswitch-users mailing list archive at Nabble.com.
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat Dec 18 03:52:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 17 Dec 2010 16:52:07 -0800
Subject: [Freeswitch-users] collecting dtmf digits
In-Reply-To: <AANLkTinNOJZxrmk38SfeKrbHqJQOTGfXpuxoBUkdHi82@mail.gmail.com>
References: <1292319831757-5833942.post@n2.nabble.com>
	<AANLkTimLDbh1tv8uKbWDRTbGfV5usVmqW8eQ8zFjz=0t@mail.gmail.com>
	<AANLkTi=o-6h9q2UWwaYZ9hHuC7sMNTNYQs=nJm6yCaBH@mail.gmail.com>
	<4D0A9655.7070007@gmail.com>
	<AANLkTimHAZF8eZ=h4mrPcPRU9qB9j2GHvGb0rBsE86n7@mail.gmail.com>
	<D0BEED38-8178-4A24-AEE6-816C0AA46780@ipeva.fr>
	<4D0B28DF.9070708@gmail.com>
	<AANLkTinNOJZxrmk38SfeKrbHqJQOTGfXpuxoBUkdHi82@mail.gmail.com>
Message-ID: <AANLkTikKuHtRfACfa5C1DKVevb9=iS4w-u-7KFMRsVGz@mail.gmail.com>

It depends on what you want to do with the call in the case that the other
end does not answer. If you are doing some sort of billing then STOP! Don't
do that. Do your billing offline via CDRs or use api_hangup_hook.

If you just want to send the call to an alternate destination then you need
to look at the bridge app on the wiki, specifically for doing pipe-separated
lists of dialstrings:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Description

-MC

On Fri, Dec 17, 2010 at 2:17 AM, Sam <u2nsam at gmail.com> wrote:

> Thanx Nazim it was of great help,
>
> i used the below syntax
>
>
> session:execute("bridge",(string.format("{ignore_early_media='true',continue_on_fail='true'}sofia/external/%
> s at 192.168.2.3",tostring(digits) )) ) ;
>
> I am now checking out if i can use hangup causes for routing the call
>
> LCause = session:hangupCause() ;
>
> freeswitch.consoleLog("info", "hangupcause: ".. LCause .." );
>
> then if i can use if then else statement for routing ... like
>
> if ( LCause == "NO_USER_RESPONSE" ) then
>        session:transfer(tostring(digits), "XML", "8888")
>
> how is the idea ?
>
> Regds
>
>
>
>
> On Fri, Dec 17, 2010 at 2:39 PM, Nazim Aghabayov <
> nazim.aghabayov at gmail.com> wrote:
>
>> Hello! You may try something like:
>>
>>  session:execute("bridge",
>>                            (string.format("{ignore_early_media='true',
>> ringback='home/app/sounds/ringback-alaw.wav',
>> transfer_ringback='/home/app/sounds/ringback-alaw.wav',
>> hangup_after_bridge='true',
>> continue_on_fail='true'}sofia/external/%s at xxx.xxx.xxx.xxx",
>>                                           tostring(did) )) )
>>
>> There are a lot of nice examples in lua&dialplan wiki pages.
>>
>> On 12/17/2010 12:24 PM, David Ponzone wrote:
>> > You need to read the wiki intensively, and also, Internet is a very
>> useful source of information to learn LUA.
>> >
>> > David Ponzone  Direction Technique
>> > email: david.ponzone at ipeva.fr
>> > tel:      01 74 03 18 97
>> > gsm:   06 66 98 76 34
>> >
>> > Service Client IPeva
>> > tel:      0811 46 26 26
>> > www.ipeva.fr  -   www.ipeva-studio.com
>> >
>> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> >
>> >
>> >
>> >
>> > Le 17/12/2010 ? 09:07, samir a ?crit :
>> >
>> >> Hello Nazim,
>> >>
>> >> When i use originate command it gives me below error :-
>> >>
>> >> 2010-12-17 13:29:30.964039 [ERR] mod_lua.cpp:182 Error in originate
>> expected 4..4 args, got 1
>> >> stack traceback:
>> >>         [C]: in function 'originate'
>> >>         /usr/local/freeswitch/scripts/hello.lua:14: in main chunk
>> >>
>> >> I am using
>> >>
>> >> digits = session:getDigits(5, "#", 3000);
>> >> freeswitch.consoleLog("info", "Got dtmf: ".. digits .."\n");
>> >> session.originate("sofia/external/(digits)@192.168.2.3");
>> >>
>> >>
>> >> any idea why is it ?
>> >>
>> >> Regards
>> >> Sam
>> >>
>> >>
>> >> On Fri, Dec 17, 2010 at 4:17 AM, Nazim Aghabayov [via freeswitch-users]
>> <[hidden email]> wrote:
>> >> Why not using Lua script? Lua is quite powerful and you could benefit
>> of
>> >> using the luasql for db connectivity.
>> >>
>> >> g_caller_id = session:getVariable("caller_id_number")
>> >> g_caller_destination_number = session:getVariable("destination_number")
>> >> g_caller_context = session:getVariable("context")
>> >> g_caller_uuid = session:getVariable("uuid")
>> >> ...
>> >> session:flushDigits()
>> >> digits = session:playAndGetDigits(4, 4, 3, 3000, "#",
>> >>               wav_base .. langId .. "/" .. prompt_wav, "",
>> >>       "[" .. allowed_digit1 .. "," .. allowed_digitN .. "]");
>> >> ...
>> >> session:transfer(tostring(dest_ext), "XML", "public")
>> >>
>> >> Regards,
>> >> Nazim
>> >>
>> >>
>> >> On 12/16/2010 07:29 AM, Sam wrote:
>> >>
>> >>> hi,
>> >>>
>> >>> Its not an XML IVR but trying to collect digits when the user punches
>> in
>> >>> some digits after a playback sound file,
>> >>> so that the digits would get stored into $(digits) and i can use those
>> >>> digits for further processing or transferring
>> >>> to other dial-plan as per the digits punched in.
>> >>>
>> >>> Suppose a digits punched are 4567, I collect them in $(digits) and by
>> >>> transfer function i do the transfer.
>> >>>
>> >>> <action application="transfer" data="$(digits) default XML"/>
>> >>>
>> >>>
>> -----------------------------------------------------------------------------------------------
>> >>>
>> >>> Also i could see that in IVR.conf can do this by pressing 4567 and
>> executing
>> >>> a statement , but here also question
>> >>> is how can i use it by storing those collected digits in an variable
>> and the
>> >>> using it further.
>> >>>
>> >>>
>> ----------------------------------------------------------------------------------------------
>> >>>
>> >>> I was thinking how in both the cases it could be done.
>> >>>
>> >>> Thnx & Regds
>> >>> Sam
>> >>>
>> >>>
>> >>>
>> >>> On Thu, Dec 16, 2010 at 4:35 AM, Michael Collins <[hidden email]>
>> wrote:
>> >>>
>> >>>> Is this an XML IVR?
>> >>>> -MC
>> >>>>
>> >>>>
>> >>>> On Tue, Dec 14, 2010 at 1:43 AM, samir <[hidden email]> wrote:
>> >>>>> hello,
>> >>>>>
>> >>>>> Is there any method to collect digits by a variable clause ?
>> >>>>>
>> >>>>> Suppose i have a ivr playing and user inputs digits , i want to
>> collect
>> >>>>> the
>> >>>>> dtmf digits and send it to a different application where that digits
>> will
>> >>>>> be
>> >>>>> used for routing purpose.
>> >>>>>
>> >>>>> Any ideas !
>> >>>>>
>> >>>>> Regards
>> >>>>> Sam
>> >>>>> --
>> >>>>> View this message in context:
>> >>>>>
>> http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5833942.html
>> >>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>> >>>>>
>> >>>>> _______________________________________________
>> >>>>> FreeSWITCH-users mailing list
>> >>>>> [hidden email]
>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>> http://www.freeswitch.org
>> >>>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> [hidden email]
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> [hidden email]
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> [hidden email]
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >> View message @
>> http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5843837.html
>> >>
>> >> To unsubscribe from collecting dtmf digits, click here.
>> >>
>> >>
>> >> View this message in context: Re: collecting dtmf digits
>> >> Sent from the freeswitch-users mailing list archive at Nabble.com.
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat Dec 18 03:54:55 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 17 Dec 2010 16:54:55 -0800
Subject: [Freeswitch-users] members audio conference
In-Reply-To: <223AD13D879E4B19B15AEEDFC387386A@e1705>
References: <223AD13D879E4B19B15AEEDFC387386A@e1705>
Message-ID: <AANLkTikydvXoJOk5kkWPYpaDqCGYTJgXbzj2=ffTB07p@mail.gmail.com>

What do you see in the debug logs? Did you compare the logs for a working
vs. non-working call? Anything different?

-MC

On Fri, Dec 17, 2010 at 7:35 AM, Madovsky <infos at madovsky.org> wrote:

>  when the first member creates and enters in a new conference
> everything is ok. but if a new memeber enters there is no audio
> in the conference, unless the ivr.
>
> I have a very simple conference dialplan like this
>
>         <extension name="create_conference">
>                 <condition field="${sip_to_uri}"
> expression="^000(\d{10,15})@$${domain}$">
>                         <action application="set"
> data="instant_ringback=true"/>
>                         <action application="ring_ready"/>
>                         <action application="sleep" data="3000"/>
>                         <action application="answer"/>
>                         <action application="conference" data="
> $1-${domain_name}@wideband"/ <$1-$%7Bdomain_name%7D at wideband%22/>>
>                 </condition>
>         </extension>
>
> I tried to add
> <action application="export" data="absolute_codec_string=speex at 16000k"/<absolute_codec_string=speex at 16000k%22/>
> >
>
> but no success
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat Dec 18 04:01:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 17 Dec 2010 17:01:07 -0800
Subject: [Freeswitch-users] Having problems (no sound) when trying to
 LogIn from Home to Freeswitch
In-Reply-To: <AANLkTinSYxPCBNEMM1OuPcKw7LKA266XVuQW7m-P1LN1@mail.gmail.com>
References: <AANLkTinSYxPCBNEMM1OuPcKw7LKA266XVuQW7m-P1LN1@mail.gmail.com>
Message-ID: <AANLkTinM092V0rCYhYkV3=fkrMZm-gwo0r_t24FE4owy@mail.gmail.com>

Smells like a NAT or firewall problem. What is the model of the firewall?
Make sure that you disable any SIP ALG for your device. Once that is done
then you'll need to compare the SIP INVITE on FS and compare to INVITE
coming from your xlite at home. Chances are something somewhere is futzing
with your SDP.

-MC

On Fri, Dec 17, 2010 at 8:48 AM, Joao Leme <joaocarlosleme at gmail.com> wrote:

> Hi there! First time here.
> I'm having problems (no sound) when trying to LogIn from Home to
> Freeswitch. In the office it all works fine, even when I use the external IP
> as Domain on X-Lite. I'm trying to go for internal profile on port 5060 as I
> want to be just like if I was there in the office. It actually works, I can
> make calls, receive etc but NO SOUND.
> What I also did was to change the domain on var.xml to the external ip. I
> also forwarded the ports to the computer where Freeswitch according to
> Firewall info on docs.
> Thanks a lot,
> John
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Sat Dec 18 05:49:21 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 17 Dec 2010 21:49:21 -0500
Subject: [Freeswitch-users] members audio conference
References: <223AD13D879E4B19B15AEEDFC387386A@e1705>
	<AANLkTikydvXoJOk5kkWPYpaDqCGYTJgXbzj2=ffTB07p@mail.gmail.com>
Message-ID: <532EC039F378476A9515E88F30BBB432@e1705>

nohgint strange on logs.
but I guess it's a codec and rate problem.
tried with different SIP phones and it works.
how a conference manage the codecs  ?
I know the rate can be set in profile,
but how conference codec is managed
 if all members have different codec and rate ?



  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Friday, December 17, 2010 7:54 PM
  Subject: Re: [Freeswitch-users] members audio conference


  What do you see in the debug logs? Did you compare the logs for a working vs. non-working call? Anything different?


  -MC


  On Fri, Dec 17, 2010 at 7:35 AM, Madovsky <infos at madovsky.org> wrote:

    when the first member creates and enters in a new conference
    everything is ok. but if a new memeber enters there is no audio
    in the conference, unless the ivr.

    I have a very simple conference dialplan like this

            <extension name="create_conference">
                    <condition field="${sip_to_uri}" expression="^000(\d{10,15})@$${domain}$">
                            <action application="set" data="instant_ringback=true"/>
                            <action application="ring_ready"/>
                            <action application="sleep" data="3000"/>
                            <action application="answer"/>
                            <action application="conference" data="$1-${domain_name}@wideband"/>
                    </condition>
            </extension>


    I tried to add
    <action application="export" data="absolute_codec_string=speex at 16000k"/>

    but no success

    Thanks

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






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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From u2nsam at gmail.com  Sat Dec 18 07:12:28 2010
From: u2nsam at gmail.com (Sam)
Date: Sat, 18 Dec 2010 09:42:28 +0530
Subject: [Freeswitch-users] routing via hangup_cause
In-Reply-To: <AANLkTi=-7N=f7PdSurgmxvr-9Y20Wk0G5N4v1+Z3zcmV@mail.gmail.com>
References: <1292386719584-5837136.post@n2.nabble.com>
	<AANLkTiko+9M0MNeKgZO2aprbZg8LevA_7pHMSsKqn=AE@mail.gmail.com>
	<AANLkTi=-7N=f7PdSurgmxvr-9Y20Wk0G5N4v1+Z3zcmV@mail.gmail.com>
Message-ID: <AANLkTiniD+yjP7pmeeGQSnFVEy_42MXBeEAxSVwLaTwm@mail.gmail.com>

Yes Bleg hangs up first but i am not able to fetch the hangupcause here,

hangup cause: ${bridge_hangup_cause}

in logs i get as :-

EXECUTE sofia/internal/7001 at 203.153.53.181 log(1 B-leg hangup Q850 cause: )
2010-12-18 09:37:08.342225 [DEBUG] switch_core_state_machine.c:449
(sofia/external/1234 at 203.153.53.188) Callstate Change HANGUP -> DOWN
2010-12-18 09:37:08.342225 [ALERT] mod_dptools.c:1152 B-leg hangup Q850
cause:
2010-12-18 09:37:08.342225 [DEBUG] switch_core_state_machine.c:452
(sofia/external/1234 at 203.153.53.188) Running State Change CS_DESTROY


Regds
Sam



On Sat, Dec 18, 2010 at 3:21 AM, Michael Collins <msc at freeswitch.org> wrote:

> You're almost there. First off, you can delete this line:
> <action application="set" data="hangup_after_bridge=false"/>
> That's the default behavior and you have to set it prior to the bridge
> anyway.
>
> Move this line before the bridge:
> <action application="set" data="continue_on_fail=true"/>
> Otherwise it won't have any effect.
>
> The other stuff should work if the b-leg hangs up first, but not if the
> a-leg does.
>
> -MC
>
>
> On Thu, Dec 16, 2010 at 2:28 AM, Sam <u2nsam at gmail.com> wrote:
>
>>
>> I am not able to fetch the hangup cause in the dial-plan by using log
>> function .
>> I would like to use that hangup cause variable for further routing so that
>> the routing would be conditional to hangup cause.
>>
>> Regards
>> Sam
>>
>> On Wed, Dec 15, 2010 at 9:48 AM, samir <u2nsam at gmail.com> wrote:
>>
>>>
>>> Hello friends,
>>>
>>> was trying to create a routing rule to to route calls by accounting to
>>> the
>>> hangup causes,
>>>
>>> I have written below syntax but it fails to give the cause code to the
>>> varriable for routing.
>>>
>>> <extension name="refer">
>>>               <condition field="destination_number" expression="^1234$">
>>>
>>>
>>>       <action application="bridge" data="sofia/external/1234 at X.X.X.X"/>
>>>       <action application="set" data="hangup_after_bridge=false"/>
>>>       <action application="set" data="continue_on_fail=true"/>
>>>       <action application="log" data="1 B-leg hangup cause:
>>> ${bridge_hangup_cause}"/>
>>>       <action application="transfer" data="${bridge_hangup_cause} XML
>>> conbridge"/>
>>>
>>>               </condition>
>>>
>>>
>>>   </extension>
>>>
>>>       <extension name="no user">
>>>               <condition field="destination_number"
>>> expression="^(NO_USER_RESPONSE)$">
>>>               <action application="bridge"
>>> data="sofia/external/4567 at X.X.X.X"/>
>>>
>>>               </condition>
>>>       </extension>
>>>
>>>
>>> here the ${bridge_hangup_cause} is not getting executed. Am I doing it
>>> right
>>> or is there any other way to do it.
>>>
>>> Regards
>>> Sam
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/routing-via-hangup-cause-tp5837136p5837136.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Sat Dec 18 08:10:32 2010
From: u2nsam at gmail.com (Sam)
Date: Sat, 18 Dec 2010 10:40:32 +0530
Subject: [Freeswitch-users] collecting dtmf digits
In-Reply-To: <AANLkTikKuHtRfACfa5C1DKVevb9=iS4w-u-7KFMRsVGz@mail.gmail.com>
References: <1292319831757-5833942.post@n2.nabble.com>
	<AANLkTimLDbh1tv8uKbWDRTbGfV5usVmqW8eQ8zFjz=0t@mail.gmail.com>
	<AANLkTi=o-6h9q2UWwaYZ9hHuC7sMNTNYQs=nJm6yCaBH@mail.gmail.com>
	<4D0A9655.7070007@gmail.com>
	<AANLkTimHAZF8eZ=h4mrPcPRU9qB9j2GHvGb0rBsE86n7@mail.gmail.com>
	<D0BEED38-8178-4A24-AEE6-816C0AA46780@ipeva.fr>
	<4D0B28DF.9070708@gmail.com>
	<AANLkTinNOJZxrmk38SfeKrbHqJQOTGfXpuxoBUkdHi82@mail.gmail.com>
	<AANLkTikKuHtRfACfa5C1DKVevb9=iS4w-u-7KFMRsVGz@mail.gmail.com>
Message-ID: <AANLkTimxSgthRT=quw90vdvSPz2v8WX07Wjjks0czPuF@mail.gmail.com>

If i a get a hangup cause into some variable then i can do some routing
depending upon that hangup-cause,

if (session:ready()) then

session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/zrtp/8000/zrtp-somethings_wrong.wav")

    disposition = session:getVariable("hangup_cause")
    freeswitch.consoleLog("info","Hangup cause = " .. disposition .. "\n")
    disposition = session:hangupCause()
    freeswitch.consoleLog("info","Hangup cause = " .. disposition .. "\n")

then using the "disposition" i can do the routing by if ... then ...

after executing getting below logs :-

2010-12-18 10:34:55.581403 [ERR] mod_lua.cpp:182
/usr/local/freeswitch/scripts/hello.lua:29: attempt to concatenate global
'disposition' (a nil value)
stack traceback:
        /usr/local/freeswitch/scripts/hello.lua:29: in main chunk


Regds
Sam

On Sat, Dec 18, 2010 at 6:26 AM, mercutioviz [via freeswitch-users] <
ml-node+5847277-1438596481-292429 at n2.nabble.com<ml-node%2B5847277-1438596481-292429 at n2.nabble.com>
> wrote:

> It depends on what you want to do with the call in the case that the other
> end does not answer. If you are doing some sort of billing then STOP! Don't
> do that. Do your billing offline via CDRs or use api_hangup_hook.
>
> If you just want to send the call to an alternate destination then you need
> to look at the bridge app on the wiki, specifically for doing pipe-separated
> lists of dialstrings:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Description
>
> -MC
>
> On Fri, Dec 17, 2010 at 2:17 AM, Sam <[hidden email]<http://user/SendEmail.jtp?type=node&node=5847277&i=0>
> > wrote:
>
>> Thanx Nazim it was of great help,
>>
>> i used the below syntax
>>
>>
>> session:execute("bridge",(string.format("{ignore_early_media='true',continue_on_fail='true'}sofia/external/%[hidden
>> email] <http://user/SendEmail.jtp?type=node&node=5847277&i=1>",tostring(digits)
>> )) ) ;
>>
>>
>> I am now checking out if i can use hangup causes for routing the call
>>
>> LCause = session:hangupCause() ;
>>
>> freeswitch.consoleLog("info", "hangupcause: ".. LCause .." );
>>
>> then if i can use if then else statement for routing ... like
>>
>> if ( LCause == "NO_USER_RESPONSE" ) then
>>        session:transfer(tostring(digits), "XML", "8888")
>>
>> how is the idea ?
>>
>> Regds
>>
>>
>>
>>
>> On Fri, Dec 17, 2010 at 2:39 PM, Nazim Aghabayov <[hidden email]<http://user/SendEmail.jtp?type=node&node=5847277&i=2>
>> > wrote:
>>
>>> Hello! You may try something like:
>>>
>>>  session:execute("bridge",
>>>                            (string.format("{ignore_early_media='true',
>>> ringback='home/app/sounds/ringback-alaw.wav',
>>> transfer_ringback='/home/app/sounds/ringback-alaw.wav',
>>> hangup_after_bridge='true',
>>> continue_on_fail='true'}[hidden email]<http://user/SendEmail.jtp?type=node&node=5847277&i=3>
>>> ",
>>>
>>>                                           tostring(did) )) )
>>>
>>> There are a lot of nice examples in lua&dialplan wiki pages.
>>>
>>> On 12/17/2010 12:24 PM, David Ponzone wrote:
>>> > You need to read the wiki intensively, and also, Internet is a very
>>> useful source of information to learn LUA.
>>> >
>>> > David Ponzone  Direction Technique
>>> > email: [hidden email]<http://user/SendEmail.jtp?type=node&node=5847277&i=4>
>>>
>>> > tel:      01 74 03 18 97
>>> > gsm:   06 66 98 76 34
>>> >
>>> > Service Client IPeva
>>> > tel:      0811 46 26 26
>>> > www.ipeva.fr  -   www.ipeva-studio.com
>>> >
>>> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>> >
>>> >
>>> >
>>> >
>>> > Le 17/12/2010 ? 09:07, samir a ?crit :
>>> >
>>> >> Hello Nazim,
>>> >>
>>> >> When i use originate command it gives me below error :-
>>> >>
>>> >> 2010-12-17 13:29:30.964039 [ERR] mod_lua.cpp:182 Error in originate
>>> expected 4..4 args, got 1
>>> >> stack traceback:
>>> >>         [C]: in function 'originate'
>>> >>         /usr/local/freeswitch/scripts/hello.lua:14: in main chunk
>>> >>
>>> >> I am using
>>> >>
>>> >> digits = session:getDigits(5, "#", 3000);
>>> >> freeswitch.consoleLog("info", "Got dtmf: ".. digits .."\n");
>>> >> session.originate("sofia/external/(digits)@192.168.2.3");
>>> >>
>>> >>
>>> >> any idea why is it ?
>>> >>
>>> >> Regards
>>> >> Sam
>>> >>
>>> >>
>>> >> On Fri, Dec 17, 2010 at 4:17 AM, Nazim Aghabayov [via
>>> freeswitch-users] <[hidden email]> wrote:
>>> >> Why not using Lua script? Lua is quite powerful and you could benefit
>>> of
>>> >> using the luasql for db connectivity.
>>> >>
>>> >> g_caller_id = session:getVariable("caller_id_number")
>>> >> g_caller_destination_number =
>>> session:getVariable("destination_number")
>>> >> g_caller_context = session:getVariable("context")
>>> >> g_caller_uuid = session:getVariable("uuid")
>>> >> ...
>>> >> session:flushDigits()
>>> >> digits = session:playAndGetDigits(4, 4, 3, 3000, "#",
>>> >>               wav_base .. langId .. "/" .. prompt_wav, "",
>>> >>       "[" .. allowed_digit1 .. "," .. allowed_digitN .. "]");
>>> >> ...
>>> >> session:transfer(tostring(dest_ext), "XML", "public")
>>> >>
>>> >> Regards,
>>> >> Nazim
>>> >>
>>> >>
>>> >> On 12/16/2010 07:29 AM, Sam wrote:
>>> >>
>>> >>> hi,
>>> >>>
>>> >>> Its not an XML IVR but trying to collect digits when the user punches
>>> in
>>> >>> some digits after a playback sound file,
>>> >>> so that the digits would get stored into $(digits) and i can use
>>> those
>>> >>> digits for further processing or transferring
>>> >>> to other dial-plan as per the digits punched in.
>>> >>>
>>> >>> Suppose a digits punched are 4567, I collect them in $(digits) and by
>>> >>> transfer function i do the transfer.
>>> >>>
>>> >>> <action application="transfer" data="$(digits) default XML"/>
>>> >>>
>>> >>>
>>> -----------------------------------------------------------------------------------------------
>>> >>>
>>> >>> Also i could see that in IVR.conf can do this by pressing 4567 and
>>> executing
>>> >>> a statement , but here also question
>>> >>> is how can i use it by storing those collected digits in an variable
>>> and the
>>> >>> using it further.
>>> >>>
>>> >>>
>>> ----------------------------------------------------------------------------------------------
>>> >>>
>>> >>> I was thinking how in both the cases it could be done.
>>> >>>
>>> >>> Thnx & Regds
>>> >>> Sam
>>> >>>
>>> >>>
>>> >>>
>>> >>> On Thu, Dec 16, 2010 at 4:35 AM, Michael Collins <[hidden email]>
>>> wrote:
>>> >>>
>>> >>>> Is this an XML IVR?
>>> >>>> -MC
>>> >>>>
>>> >>>>
>>> >>>> On Tue, Dec 14, 2010 at 1:43 AM, samir <[hidden email]> wrote:
>>> >>>>> hello,
>>> >>>>>
>>> >>>>> Is there any method to collect digits by a variable clause ?
>>> >>>>>
>>> >>>>> Suppose i have a ivr playing and user inputs digits , i want to
>>> collect
>>> >>>>> the
>>> >>>>> dtmf digits and send it to a different application where that
>>> digits will
>>> >>>>> be
>>> >>>>> used for routing purpose.
>>> >>>>>
>>> >>>>> Any ideas !
>>> >>>>>
>>> >>>>> Regards
>>> >>>>> Sam
>>> >>>>> --
>>> >>>>> View this message in context:
>>> >>>>>
>>> http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5833942.html<http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5833942.html?by-user=t>
>>> >>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>> >>>>>
>>> >>>>> _______________________________________________
>>> >>>>> FreeSWITCH-users mailing list
>>> >>>>> [hidden email]
>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>>> http://www.freeswitch.org
>>> >>>>>
>>> >>>> _______________________________________________
>>> >>>> FreeSWITCH-users mailing list
>>> >>>> [hidden email]
>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>> http://www.freeswitch.org
>>> >>>>
>>> >>>>
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> [hidden email]
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> [hidden email]
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >>
>>> >> View message @
>>> http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5843837.html<http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5843837.html?by-user=t>
>>> >>
>>> >> To unsubscribe from collecting dtmf digits, click here.
>>> >>
>>> >>
>>> >> View this message in context: Re: collecting dtmf digits
>>> >> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> [hidden email] <http://user/SendEmail.jtp?type=node&node=5847277&i=5>
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > [hidden email] <http://user/SendEmail.jtp?type=node&node=5847277&i=6>
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5847277&i=7>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5847277&i=8>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> [hidden email] <http://user/SendEmail.jtp?type=node&node=5847277&i=9>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
>  View message @
> http://freeswitch-users.2379917.n2.nabble.com/collecting-dtmf-digits-tp5833942p5847277.html
>
> To unsubscribe from collecting dtmf digits, click here<http://freeswitch-users.2379917.n2.nabble.com/template/NamlServlet.jtp?macro=unsubscribe_by_code&node=5833942&code=dTJuc2FtQGdtYWlsLmNvbXw1ODMzOTQyfC0xMTk1MzYwMDk5>.
>
>
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From abubacker at bksystems.co.in  Sat Dec 18 09:30:18 2010
From: abubacker at bksystems.co.in (abubacker)
Date: Sat, 18 Dec 2010 12:00:18 +0530
Subject: [Freeswitch-users] mod_callcenter: member table does not have any
	value
Message-ID: <4D0C54FA.6020709@bksys.co.in>

Dear all,
     I am working in mod_callcenter ,

using sqlite3 I have opened the callcenter.db database
/root at FMS-Queue:/usr/local/freeswitch/db# sqlite3 callcenter.db /

In that I found the member table always has no row , even the customer 
is talking with an
agent.

my question is when this table will have an entry ?

-- 
Best regards,
N.Abubacker ,
Associate system engineer ,
bk systems pvt ltd ,
Ph : 9144-43902701

Disclaimer: http://www.bksystems.co.in/email-policy

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From abubacker at bksystems.co.in  Sat Dec 18 09:47:20 2010
From: abubacker at bksystems.co.in (abubacker)
Date: Sat, 18 Dec 2010 12:17:20 +0530
Subject: [Freeswitch-users] mod_callcenter: member table does not have
 any	value
In-Reply-To: <4D0C54FA.6020709@bksys.co.in>
References: <4D0C54FA.6020709@bksys.co.in>
Message-ID: <4D0C58F8.2030809@bksys.co.in>

On Saturday 18 December 2010 12:00 PM, abubacker wrote:
> Dear all,
>     I am working in mod_callcenter ,
>
> using sqlite3 I have opened the callcenter.db database
> /root at FMS-Queue:/usr/local/freeswitch/db# sqlite3 callcenter.db /
>
> In that I found the member table always has no row , even the customer 
> is talking with an
> agent.
>
> my question is when this table will have an entry ?
> -- 
> Best regards,
> N.Abubacker ,
> Associate system engineer ,
> bk systems pvt ltd ,
> Ph : 9144-43902701
>
> Disclaimer:http://www.bksystems.co.in/email-policy
>    
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
I found the issue and resolved
issue is not in a freeswitch side !


-- 
Best regards,
N.Abubacker ,
Associate system engineer ,
bk systems pvt ltd ,
Ph : 9144-43902701

Disclaimer: http://www.bksystems.co.in/email-policy

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From u2nsam at gmail.com  Sat Dec 18 09:59:40 2010
From: u2nsam at gmail.com (Sam)
Date: Sat, 18 Dec 2010 12:29:40 +0530
Subject: [Freeswitch-users] hangup cause
Message-ID: <AANLkTi=gGs8iDB-GO7twdergCEK2+V6bObtsJS-FeVBa@mail.gmail.com>

hello,

How can i fetch hangup cause and store it in an variable.


[INFO] mod_dptools.c:2579 Originate Failed.  Cause: NO_USER_RESPONSE



Regards
Sam
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From neilp at cs.stanford.edu  Sat Dec 18 10:20:24 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Sat, 18 Dec 2010 12:50:24 +0530
Subject: [Freeswitch-users] freetdm and ring_ready
In-Reply-To: <AANLkTikqwHiLOJRzf0m_7qjxEtYgsj9xTwWW9roNDytw@mail.gmail.com>
References: <AANLkTikfyzKUZhb=2KS9=2v_Fmk=mj0=ruz328aTceM3@mail.gmail.com>
	<AANLkTimO_0JV30SoC62we=aX3Lh2-nV5wQ+UsL7oN5Gm@mail.gmail.com>
	<AANLkTin2FNYvy9_VXo_rp9FvZJ1a7avWGBwBk1VU-T5D@mail.gmail.com>
	<AANLkTiniZ_6utTHyWQQJx2V8QnKur27JaaP2NBsmu+rS@mail.gmail.com>
	<AANLkTim3DVdUnYVsGprgXggQeGU=Pq6Lyp9WNeQ_mB8L@mail.gmail.com>
	<AANLkTikqwHiLOJRzf0m_7qjxEtYgsj9xTwWW9roNDytw@mail.gmail.com>
Message-ID: <AANLkTi=fqA7tMcWAMUCHNfH=bFaY6b1wzQKsj6SnZpXp@mail.gmail.com>

Hey Moy and Michael,

Looks like pre_answer doesn't work either. Bug filed on Jira:
http://jira.freeswitch.org/browse/OPENZAP-129

Thanks for your help! Our project <http://hci.stanford.edu/research/otalo>,
powered by FS, is going quite well. Appreciate all of your help in making it
possible.

-Neil

On Fri, Dec 17, 2010 at 7:51 PM, Moises Silva <moises.silva at gmail.com>wrote:

> On Thu, Dec 16, 2010 at 1:14 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
> Moy,
>> So is this a case of relying on incorrect behavior as a "feature"? Just
>> confirming. I'll make sure that the pre_answer app is mentioned in the
>> FreeTDM wiki pages where appropriate.
>>
>>
> Yes, that's what it looks like to me. I just would like confirmation from
> Neil that "pre_answer" works as expected.
>
> Neil: If "pre_answer" does not work for FreeTDM, please open a bug in jira.
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R
> 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris at cloudtel.com  Sat Dec 18 10:21:26 2010
From: chris at cloudtel.com (Chris Burns)
Date: Sat, 18 Dec 2010 02:21:26 -0500
Subject: [Freeswitch-users] routing via hangup_cause
In-Reply-To: <AANLkTiniD+yjP7pmeeGQSnFVEy_42MXBeEAxSVwLaTwm@mail.gmail.com>
References: <1292386719584-5837136.post@n2.nabble.com>
	<AANLkTiko+9M0MNeKgZO2aprbZg8LevA_7pHMSsKqn=AE@mail.gmail.com>
	<AANLkTi=-7N=f7PdSurgmxvr-9Y20Wk0G5N4v1+Z3zcmV@mail.gmail.com>
	<AANLkTiniD+yjP7pmeeGQSnFVEy_42MXBeEAxSVwLaTwm@mail.gmail.com>
Message-ID: <AANLkTinXEWcS1LfDc4Nvz7BtosvYPWXjUFD9rrA1mGVe@mail.gmail.com>

First of all, your log output is different than your dialplan example.
Second, you show only a tiny log snip taken after everything interesting
occurred. All it shows is that you are having the issue you describe. Third,
it seems like you are making multiple posts to the list for the same issue,
and multiple emails within those requests. Slow down, take your time, try to
learn as best you can (using the wiki), and give the list a reasonable
amount of time to answer your questions. If you cant get a hangup cause in
your XML dialplan, you dont need to start a seperate thread for how you cant
do it in LUA either ... you see my point? :)

The other end has to answer the call in order to hang up on you, as far as I
know. If the originate fails you don't get a hangup cause into the
bridge_hangup_cause channel variable. For instance, the other end never
answers the call and originate fails with NO_USER_RESPONSE .... you will not
have any value in bridge_hangup_cause because nothing ever bridged. Check
the variable originate_disposition:
http://wiki.freeswitch.org/wiki/Variable_originate_disposition

Try out this extension using the "info" app to debug variables after your
bridge:
<extension name="funwithvariables">
  <condition field="destination_number" expression="^1234$">
    <action application="set" data="hangup_after_bridge=false"/>
    <action application="set" data="continue_on_fail=true"/>
    <action application="bridge"
data="sofia/external/myfriend at myfriendsSIPserver.com"/>
    <action application="info"/>
  </condition>
</extension>


On Fri, Dec 17, 2010 at 11:12 PM, Sam <u2nsam at gmail.com> wrote:

> Yes Bleg hangs up first but i am not able to fetch the hangupcause here,
>
>
> hangup cause: ${bridge_hangup_cause}
>
> in logs i get as :-
>
> EXECUTE sofia/internal/7001 at 203.153.53.181 log(1 B-leg hangup Q850 cause:
> )
> 2010-12-18 09:37:08.342225 [DEBUG] switch_core_state_machine.c:449
> (sofia/external/1234 at 203.153.53.188) Callstate Change HANGUP -> DOWN
> 2010-12-18 09:37:08.342225 [ALERT] mod_dptools.c:1152 B-leg hangup Q850
> cause:
> 2010-12-18 09:37:08.342225 [DEBUG] switch_core_state_machine.c:452
> (sofia/external/1234 at 203.153.53.188) Running State Change CS_DESTROY
>
>
> Regds
> Sam
>
>
>
>
> On Sat, Dec 18, 2010 at 3:21 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> You're almost there. First off, you can delete this line:
>> <action application="set" data="hangup_after_bridge=false"/>
>> That's the default behavior and you have to set it prior to the bridge
>> anyway.
>>
>> Move this line before the bridge:
>> <action application="set" data="continue_on_fail=true"/>
>> Otherwise it won't have any effect.
>>
>> The other stuff should work if the b-leg hangs up first, but not if the
>> a-leg does.
>>
>> -MC
>>
>>
>> On Thu, Dec 16, 2010 at 2:28 AM, Sam <u2nsam at gmail.com> wrote:
>>
>>>
>>> I am not able to fetch the hangup cause in the dial-plan by using log
>>> function .
>>> I would like to use that hangup cause variable for further routing so
>>> that
>>> the routing would be conditional to hangup cause.
>>>
>>> Regards
>>> Sam
>>>
>>> On Wed, Dec 15, 2010 at 9:48 AM, samir <u2nsam at gmail.com> wrote:
>>>
>>>>
>>>> Hello friends,
>>>>
>>>> was trying to create a routing rule to to route calls by accounting to
>>>> the
>>>> hangup causes,
>>>>
>>>> I have written below syntax but it fails to give the cause code to the
>>>> varriable for routing.
>>>>
>>>> <extension name="refer">
>>>>               <condition field="destination_number" expression="^1234$">
>>>>
>>>>
>>>>       <action application="bridge" data="sofia/external/1234 at X.X.X.X"/>
>>>>       <action application="set" data="hangup_after_bridge=false"/>
>>>>       <action application="set" data="continue_on_fail=true"/>
>>>>       <action application="log" data="1 B-leg hangup cause:
>>>> ${bridge_hangup_cause}"/>
>>>>       <action application="transfer" data="${bridge_hangup_cause} XML
>>>> conbridge"/>
>>>>
>>>>               </condition>
>>>>
>>>>
>>>>   </extension>
>>>>
>>>>       <extension name="no user">
>>>>               <condition field="destination_number"
>>>> expression="^(NO_USER_RESPONSE)$">
>>>>               <action application="bridge"
>>>> data="sofia/external/4567 at X.X.X.X"/>
>>>>
>>>>               </condition>
>>>>       </extension>
>>>>
>>>>
>>>> here the ${bridge_hangup_cause} is not getting executed. Am I doing it
>>>> right
>>>> or is there any other way to do it.
>>>>
>>>> Regards
>>>> Sam
>>>> --
>>>> View this message in context:
>>>> http://freeswitch-users.2379917.n2.nabble.com/routing-via-hangup-cause-tp5837136p5837136.html
>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris at cloudtel.com  Sat Dec 18 10:35:23 2010
From: chris at cloudtel.com (Chris Burns)
Date: Sat, 18 Dec 2010 02:35:23 -0500
Subject: [Freeswitch-users] hangup cause
In-Reply-To: <AANLkTi=gGs8iDB-GO7twdergCEK2+V6bObtsJS-FeVBa@mail.gmail.com>
References: <AANLkTi=gGs8iDB-GO7twdergCEK2+V6bObtsJS-FeVBa@mail.gmail.com>
Message-ID: <AANLkTikn29nN1apGdyNsa-mybLmVzJyXTtjV3+yHBmX1@mail.gmail.com>

Please stop creating new threads for the same issue. Stick to 1 of the 2
threads you already have.

We get the message ... you are desperate for support. Perhaps you require
some kind of official training or support contract? Otherwise have patience
:)

On Sat, Dec 18, 2010 at 1:59 AM, Sam <u2nsam at gmail.com> wrote:

> hello,
>
> How can i fetch hangup cause and store it in an variable.
>
>
> [INFO] mod_dptools.c:2579 Originate Failed.  Cause: NO_USER_RESPONSE
>
>
>
> Regards
> Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris at cloudtel.com  Sat Dec 18 11:42:53 2010
From: chris at cloudtel.com (Chris Burns)
Date: Sat, 18 Dec 2010 03:42:53 -0500
Subject: [Freeswitch-users] Lua not playing any wav files
In-Reply-To: <AANLkTimHrCzzzg8S=PVTVDmH1sd4iv4jQiMFbFHR7teo@mail.gmail.com>
References: <AANLkTimHrCzzzg8S=PVTVDmH1sd4iv4jQiMFbFHR7teo@mail.gmail.com>
Message-ID: <AANLkTik3MPkku8-Zk-JKRezUs=tMmrrG=xmCvZPcQ3wt@mail.gmail.com>

Hmmm no help for you yet huh ... you may have solved it on your own already,
but ...

You want to answer the call there, and not pre-answer. Pre-answer is for
early media, which is for exchanging media before committing to answer the
call. Admittedly you should hear something either way, but you definitely
want to answer the call in your case. If your XML dialplan works as you
said, you should compare the log output between these 2 extensions:

<extension name="testXML">
  <condition field="destination_number" expression="^9001$">
    <action application="answer"/>
    <action application="playback" data="local_stream://moh"/>
  </condition>
</extension>

<extension name="testLUA">
  <condition field="destination_number" expression="^9002$">
    <action application="lua" data="test.lua"/>
  </condition>
</extension>

Contents of test.lua:
session:answer();
session:execute("playback","local_stream://moh");


On Wed, Dec 15, 2010 at 1:59 AM, Marc de Corny <marcdecorny at gmail.com>wrote:

> Hi all,
>
> I have run into an issue on something so basic that I must be as simple as
> enabling a feature somewhere.
>
> I have been trying to get lua to play a message from a WAV file. I have
> tried session:execute("playback", main_msg) and
> session:streamFile(ivr_invalid_msg) but neither of them play any music to
> the caller. I tried both to answer and preAnswer the call first but it made
> no difference. However if I put the same file into the XML dialplan and play
> it with the commands below I hear the music fine.
> <action application="set" data="playback_terminators=#"/>
> <action application="playback" data="${401_sound_file_path}"/>
>
> The issue only seems to be from lua when playing any type of wav file and
> those files are definitelly there as can be read by the XML
>
> The error message is below for the execute(playback) command, but nothing
> can be seen for the
> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application
> playback Requires media! pre_answering channel
> sofia/external/2031701665 at 194.0.147.16:5060 EXECUTE
> sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswitch/sounds/svc_sound_files/default_autoattendant.wav)
> But there is no mention of the streamFile command. I have had similar issue
> with the PlayAndGetDigits command.
> Is there something that I need to enable in lua so that is can playback
> messages to the caller.
>
> Many thanks to anyone who can help.
> Marc
>
>
> below is the XML dialplan and lua script as well as the log at the very
> end.
>
> XML DIALPLAN:
> <include>
> <extension name="IVR_FROM_MYSQL">
> <condition field="caller_id_number" expression="Anonymous" break="never">
> <action application="set" data="effective_caller_id_number=0000000000"/>
> </condition>
> <condition field="destination_number" expression="^(404)">
> <action inline="true" application="lua" data="ivr_mysql.lua
> ${destination_number:3}"/>
> <action application="set" data="effective_caller_id_name=${404_tag}"/>
> <action application="bridge" data="${404_dial}"/>
> </condition>
> </extension>
> </include>
>
> The LUA script ivr_mysql.lua is callsed and this is it.
> -- IVR : PLAY IVR WAV FILES
> -- Global Variables:
> local dialstr_prefix = "sofia/gateway/CS2k/"
> local dialstr_main = ""
> local breakoutcode = "184"
> local sound_file_folder = "/usr/local/freeswitch/sounds/svc_sound_files/"
> local ddi = argv[1]
> -- answer the call
> session:preAnswer();
> freeswitch.consoleLog("info", "All Answered\n");
> ivr_invalid_msg = sound_file_folder .. "invalid_msg.wav"
> main_msg = sound_file_folder .. "default_autoattendant.wav"
> -- Play with Execute
> session:execute("playback", main_msg)
> -- Play with StreamFile
> session:streamFile(ivr_invalid_msg);
> dialstr_main  = dialstr_main .. dialstr_prefix .. breakoutcode ..
> "02031701665"
> session:setVariable("404_dial",dialstr_main)
> session:setVariable("404_tag","IVR")
>
>
> RELEVANT LOGS :
> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Regex (PASS)
> [IVR_FROM_MYS                          QL] destination_number(4042031956241)
> =~ /^(404)/ break=on-false
> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
> lua(ivr_mysql.lua $                          {destination_number:3}) INLINE
> EXECUTE sofia/external/2031701665 at 194.0.147.16:5060 lua(ivr_mysql.lua
> 2031956241                          )
> 2010-12-01 11:56:01.525426 [INFO] switch_cpp.cpp:584 Sending early media
> 2010-12-01 11:56:01.525426 [DEBUG] sofia_glue.c:2972 AUDIO RTP
> [sofia/external/2                          031701665 at 194.0.147.16:5060]
> 10.5.2.105 port 29900 -> 194.0.147.164 port 50202 c
> odec: 8 ms: 20
> 2010-12-01 11:56:01.525426 [DEBUG] switch_rtp.c:1418 Starting timer [soft]
> 160 b                          ytes per 20ms
> 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3190 Set 2833 dtmf send
> payload                           to 101
> 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3195 Set 2833 dtmf receive
> paylo                          ad to 101
> 2010-12-01 11:56:01.532280 [DEBUG] mod_sofia.c:2172 Ring SDP:
> v=0
> o=FreeSWITCH 1291174661 1291174662 IN IP4 10.5.2.105
> s=FreeSWITCH
> c=IN IP4 10.5.2.105
> t=0 0
> m=audio 29900 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 2010-12-01 11:56:01.532280 [NOTICE] mod_sofia.c:2175 Pre-Answer
> sofia/external/2                          031701665 at 194.0.147.16:5060!
> 2010-12-01 11:56:01.532280 [DEBUG] switch_channel.c:2544
> (sofia/external/2031701                          665 at 194.0.147.16:5060)
> Callstate Change RINGING -> EARLY
> 2010-12-01 11:56:01.534727 [DEBUG] sofia.c:4576 Channel
> sofia/external/203170166                          5 at 194.0.147.16:5060skipping state [early][183]
> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:676 Send signal
> sofia/e                          xternal/2031701665 at 194.0.147.16:5060[BREAK]
> 2010-12-01 11:56:01.534727 [INFO] switch_cpp.cpp:1181 All Answered
> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application
> playba                          ck Requires media! pre_answering channel
> sofia/external/2031701665 at 194.0.147.16:                          5060
> EXECUTE sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswit
> ch/sounds/svc_sound_files/default_autoattendant.wav)
> 2010-12-01 11:56:01.537644 [DEBUG] switch_cpp.cpp:972
> sofia/external/2031701665@                          194.0.147.16:5060destroy/unlink session from object
> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
> set(effective_calle                          r_id_name=${404_tag})
> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
> bridge(${404_dial})
> 2010-12-01 11:56:01.537644 [DEBUG] switch_core_state_machine.c:119
> (sofia/extern                          al/2031701665 at 194.0.147.16:5060)
> State Change CS_ROUTING -> CS_EXECUTE
> 2010-12-01 11:56:01.537644 [DEBUG] switch_core_session.c:1057 Send signal
> sofia/                          external/2031701665 at 194.0.147.16:5060[BREAK]
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From marcdecorny at gmail.com  Sat Dec 18 12:03:57 2010
From: marcdecorny at gmail.com (Marc De Corny)
Date: Sat, 18 Dec 2010 09:03:57 +0000
Subject: [Freeswitch-users] Lua not playing any wav files
In-Reply-To: <AANLkTik3MPkku8-Zk-JKRezUs=tMmrrG=xmCvZPcQ3wt@mail.gmail.com>
References: <AANLkTimHrCzzzg8S=PVTVDmH1sd4iv4jQiMFbFHR7teo@mail.gmail.com>
	<AANLkTik3MPkku8-Zk-JKRezUs=tMmrrG=xmCvZPcQ3wt@mail.gmail.com>
Message-ID: <C4D7BFCA-3104-4D35-8F47-E45064DA2D30@gmail.com>

Thanks Chris, 
Stil got issue unfortunately
i tried with both answer and preAnswer and got same results. I will post the logs of the two examples and see.
Thanks for following up on this.
Marc

On 18 Dec 2010, at 08:42, Chris Burns <chris at cloudtel.com> wrote:

> Hmmm no help for you yet huh ... you may have solved it on your own already, but ...
> 
> You want to answer the call there, and not pre-answer. Pre-answer is for early media, which is for exchanging media before committing to answer the call. Admittedly you should hear something either way, but you definitely want to answer the call in your case. If your XML dialplan works as you said, you should compare the log output between these 2 extensions:
> 
> <extension name="testXML">
>   <condition field="destination_number" expression="^9001$">
>     <action application="answer"/>
>     <action application="playback" data="local_stream://moh"/>
>   </condition>
> </extension>
> 
> <extension name="testLUA">
>   <condition field="destination_number" expression="^9002$">
>     <action application="lua" data="test.lua"/>
>   </condition>
> </extension>
> 
> Contents of test.lua:
> session:answer();
> session:execute("playback","local_stream://moh");
> 
> 
> On Wed, Dec 15, 2010 at 1:59 AM, Marc de Corny <marcdecorny at gmail.com> wrote:
> Hi all,
>  
> I have run into an issue on something so basic that I must be as simple as enabling a feature somewhere.
>  
> I have been trying to get lua to play a message from a WAV file. I have tried session:execute("playback", main_msg) and session:streamFile(ivr_invalid_msg) but neither of them play any music to the caller. I tried both to answer and preAnswer the call first but it made no difference. However if I put the same file into the XML dialplan and play it with the commands below I hear the music fine. 
> <action application="set" data="playback_terminators=#"/>
> <action application="playback" data="${401_sound_file_path}"/>
>  
> The issue only seems to be from lua when playing any type of wav file and those files are definitelly there as can be read by the XML
>  
> The error message is below for the execute(playback) command, but nothing can be seen for the 
> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application playback Requires media! pre_answering channel sofia/external/2031701665 at 194.0.147.16:5060 EXECUTE sofia/external/2031701665 at 194.0.147.16:5060 playback(/usr/local/freeswitch/sounds/svc_sound_files/default_autoattendant.wav)
> But there is no mention of the streamFile command. I have had similar issue with the PlayAndGetDigits command.
> Is there something that I need to enable in lua so that is can playback messages to the caller.
>  
> Many thanks to anyone who can help.
> Marc
>  
>  
> below is the XML dialplan and lua script as well as the log at the very end.
>  
> XML DIALPLAN:
> <include>
> <extension name="IVR_FROM_MYSQL">
> <condition field="caller_id_number" expression="Anonymous" break="never">
> <action application="set" data="effective_caller_id_number=0000000000"/>
> </condition>
> <condition field="destination_number" expression="^(404)">
> <action inline="true" application="lua" data="ivr_mysql.lua ${destination_number:3}"/>
> <action application="set" data="effective_caller_id_name=${404_tag}"/>
> <action application="bridge" data="${404_dial}"/>
> </condition>
> </extension>
> </include>
>  
> The LUA script ivr_mysql.lua is callsed and this is it.
> -- IVR : PLAY IVR WAV FILES 
> -- Global Variables:
> local dialstr_prefix = "sofia/gateway/CS2k/"
> local dialstr_main = ""
> local breakoutcode = "184"
> local sound_file_folder = "/usr/local/freeswitch/sounds/svc_sound_files/"
> local ddi = argv[1]
> -- answer the call
> session:preAnswer();
> freeswitch.consoleLog("info", "All Answered\n"); 
> ivr_invalid_msg = sound_file_folder .. "invalid_msg.wav"
> main_msg = sound_file_folder .. "default_autoattendant.wav"
> -- Play with Execute
> session:execute("playback", main_msg)
> -- Play with StreamFile
> session:streamFile(ivr_invalid_msg);
> dialstr_main  = dialstr_main .. dialstr_prefix .. breakoutcode .. "02031701665"
> session:setVariable("404_dial",dialstr_main)
> session:setVariable("404_tag","IVR")
>  
>  
> RELEVANT LOGS :
> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Regex (PASS) [IVR_FROM_MYS                          QL] destination_number(4042031956241) =~ /^(404)/ break=on-false
> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action lua(ivr_mysql.lua $                          {destination_number:3}) INLINE
> EXECUTE sofia/external/2031701665 at 194.0.147.16:5060 lua(ivr_mysql.lua 2031956241                          )
> 2010-12-01 11:56:01.525426 [INFO] switch_cpp.cpp:584 Sending early media
> 2010-12-01 11:56:01.525426 [DEBUG] sofia_glue.c:2972 AUDIO RTP [sofia/external/2                          031701665 at 194.0.147.16:5060] 10.5.2.105 port 29900 -> 194.0.147.164 port 50202 c                          odec: 8 ms: 20
> 2010-12-01 11:56:01.525426 [DEBUG] switch_rtp.c:1418 Starting timer [soft] 160 b                          ytes per 20ms
> 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3190 Set 2833 dtmf send payload                           to 101
> 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3195 Set 2833 dtmf receive paylo                          ad to 101
> 2010-12-01 11:56:01.532280 [DEBUG] mod_sofia.c:2172 Ring SDP:
> v=0
> o=FreeSWITCH 1291174661 1291174662 IN IP4 10.5.2.105
> s=FreeSWITCH
> c=IN IP4 10.5.2.105
> t=0 0
> m=audio 29900 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 2010-12-01 11:56:01.532280 [NOTICE] mod_sofia.c:2175 Pre-Answer sofia/external/2                          031701665 at 194.0.147.16:5060!
> 2010-12-01 11:56:01.532280 [DEBUG] switch_channel.c:2544 (sofia/external/2031701                          665 at 194.0.147.16:5060) Callstate Change RINGING -> EARLY
> 2010-12-01 11:56:01.534727 [DEBUG] sofia.c:4576 Channel sofia/external/203170166                          5 at 194.0.147.16:5060 skipping state [early][183]
> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:676 Send signal sofia/e                          xternal/2031701665 at 194.0.147.16:5060 [BREAK]
> 2010-12-01 11:56:01.534727 [INFO] switch_cpp.cpp:1181 All Answered
> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application playba                          ck Requires media! pre_answering channel sofia/external/2031701665 at 194.0.147.16:                          5060
> EXECUTE sofia/external/2031701665 at 194.0.147.16:5060 playback(/usr/local/freeswit                          ch/sounds/svc_sound_files/default_autoattendant.wav)
> 2010-12-01 11:56:01.537644 [DEBUG] switch_cpp.cpp:972 sofia/external/2031701665@                          194.0.147.16:5060 destroy/unlink session from object
> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action set(effective_calle                          r_id_name=${404_tag})
> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action bridge(${404_dial})                          
> 2010-12-01 11:56:01.537644 [DEBUG] switch_core_state_machine.c:119 (sofia/extern                          al/2031701665 at 194.0.147.16:5060) State Change CS_ROUTING -> CS_EXECUTE
> 2010-12-01 11:56:01.537644 [DEBUG] switch_core_session.c:1057 Send signal sofia/                          external/2031701665 at 194.0.147.16:5060 [BREAK] 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From u2nsam at gmail.com  Sat Dec 18 12:16:26 2010
From: u2nsam at gmail.com (Sam)
Date: Sat, 18 Dec 2010 14:46:26 +0530
Subject: [Freeswitch-users] Lua not playing any wav files
In-Reply-To: <C4D7BFCA-3104-4D35-8F47-E45064DA2D30@gmail.com>
References: <AANLkTimHrCzzzg8S=PVTVDmH1sd4iv4jQiMFbFHR7teo@mail.gmail.com>
	<AANLkTik3MPkku8-Zk-JKRezUs=tMmrrG=xmCvZPcQ3wt@mail.gmail.com>
	<C4D7BFCA-3104-4D35-8F47-E45064DA2D30@gmail.com>
Message-ID: <AANLkTimKUWvBgo9M980xDJqCSHN8cCePGffPw-M2varx@mail.gmail.com>

you can try this ...


session:answer();

-- sleep a second
session:sleep(100);

-- play a file
session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav");

Regards
Sam


On Sat, Dec 18, 2010 at 2:33 PM, Marc De Corny <marcdecorny at gmail.com>wrote:

> Thanks Chris,
> Stil got issue unfortunately
> i tried with both answer and preAnswer and got same results. I will post
> the logs of the two examples and see.
> Thanks for following up on this.
> Marc
>
>
> On 18 Dec 2010, at 08:42, Chris Burns <chris at cloudtel.com> wrote:
>
> Hmmm no help for you yet huh ... you may have solved it on your own
> already, but ...
>
> You want to answer the call there, and not pre-answer. Pre-answer is for
> early media, which is for exchanging media before committing to answer the
> call. Admittedly you should hear something either way, but you definitely
> want to answer the call in your case. If your XML dialplan works as you
> said, you should compare the log output between these 2 extensions:
>
> <extension name="testXML">
>   <condition field="destination_number" expression="^9001$">
>     <action application="answer"/>
>     <action application="playback" data="local_stream://moh"/>
>   </condition>
> </extension>
>
> <extension name="testLUA">
>   <condition field="destination_number" expression="^9002$">
>     <action application="lua" data="test.lua"/>
>   </condition>
> </extension>
>
> Contents of test.lua:
> session:answer();
> session:execute("playback","local_stream://moh");
>
>
> On Wed, Dec 15, 2010 at 1:59 AM, Marc de Corny < <marcdecorny at gmail.com>
> marcdecorny at gmail.com> wrote:
>
>> Hi all,
>>
>> I have run into an issue on something so basic that I must be as simple as
>> enabling a feature somewhere.
>>
>> I have been trying to get lua to play a message from a WAV file. I have
>> tried session:execute("playback", main_msg) and
>> session:streamFile(ivr_invalid_msg) but neither of them play any music to
>> the caller. I tried both to answer and preAnswer the call first but it made
>> no difference. However if I put the same file into the XML dialplan and play
>> it with the commands below I hear the music fine.
>> <action application="set" data="playback_terminators=#"/>
>> <action application="playback" data="${401_sound_file_path}"/>
>>
>> The issue only seems to be from lua when playing any type of wav file and
>> those files are definitelly there as can be read by the XML
>>
>> The error message is below for the execute(playback) command, but nothing
>> can be seen for the
>> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application
>> playback Requires media! pre_answering channel
>> sofia/external/2031701665 at 194.0.147.16:5060 EXECUTE
>> sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswitch/sounds/svc_sound_files/default_autoattendant.wav)
>> But there is no mention of the streamFile command. I have had similar
>> issue with the PlayAndGetDigits command.
>> Is there something that I need to enable in lua so that is can playback
>> messages to the caller.
>>
>> Many thanks to anyone who can help.
>> Marc
>>
>>
>> below is the XML dialplan and lua script as well as the log at the very
>> end.
>>
>> XML DIALPLAN:
>> <include>
>> <extension name="IVR_FROM_MYSQL">
>> <condition field="caller_id_number" expression="Anonymous" break="never">
>> <action application="set" data="effective_caller_id_number=0000000000"/>
>> </condition>
>> <condition field="destination_number" expression="^(404)">
>> <action inline="true" application="lua" data="ivr_mysql.lua
>> ${destination_number:3}"/>
>> <action application="set" data="effective_caller_id_name=${404_tag}"/>
>> <action application="bridge" data="${404_dial}"/>
>> </condition>
>> </extension>
>> </include>
>>
>> The LUA script ivr_mysql.lua is callsed and this is it.
>> -- IVR : PLAY IVR WAV FILES
>> -- Global Variables:
>> local dialstr_prefix = "sofia/gateway/CS2k/"
>> local dialstr_main = ""
>> local breakoutcode = "184"
>> local sound_file_folder = "/usr/local/freeswitch/sounds/svc_sound_files/"
>> local ddi = argv[1]
>> -- answer the call
>> session:preAnswer();
>> freeswitch.consoleLog("info", "All Answered\n");
>> ivr_invalid_msg = sound_file_folder .. "invalid_msg.wav"
>> main_msg = sound_file_folder .. "default_autoattendant.wav"
>> -- Play with Execute
>> session:execute("playback", main_msg)
>> -- Play with StreamFile
>> session:streamFile(ivr_invalid_msg);
>> dialstr_main  = dialstr_main .. dialstr_prefix .. breakoutcode ..
>> "02031701665"
>> session:setVariable("404_dial",dialstr_main)
>> session:setVariable("404_tag","IVR")
>>
>>
>> RELEVANT LOGS :
>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Regex (PASS)
>> [IVR_FROM_MYS                          QL] destination_number(4042031956241)
>> =~ /^(404)/ break=on-false
>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
>> lua(ivr_mysql.lua $                          {destination_number:3}) INLINE
>> EXECUTE sofia/external/2031701665 at 194.0.147.16:5060 lua(ivr_mysql.lua
>> 2031956241                          )
>> 2010-12-01 11:56:01.525426 [INFO] switch_cpp.cpp:584 Sending early media
>> 2010-12-01 11:56:01.525426 [DEBUG] sofia_glue.c:2972 AUDIO RTP
>> [sofia/external/2                          031701665 at 194.0.147.16:5060]
>> 10.5.2.105 port 29900 -> 194.0.147.164 port 50202 c
>> odec: 8 ms: 20
>> 2010-12-01 11:56:01.525426 [DEBUG] switch_rtp.c:1418 Starting timer [soft]
>> 160 b                          ytes per 20ms
>> 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3190 Set 2833 dtmf send
>> payload                           to 101
>> 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3195 Set 2833 dtmf receive
>> paylo                          ad to 101
>> 2010-12-01 11:56:01.532280 [DEBUG] mod_sofia.c:2172 Ring SDP:
>> v=0
>> o=FreeSWITCH 1291174661 1291174662 IN IP4 10.5.2.105
>> s=FreeSWITCH
>> c=IN IP4 10.5.2.105
>> t=0 0
>> m=audio 29900 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>> 2010-12-01 11:56:01.532280 [NOTICE] mod_sofia.c:2175 Pre-Answer
>> sofia/external/2                          031701665 at 194.0.147.16:5060!
>> 2010-12-01 11:56:01.532280 [DEBUG] switch_channel.c:2544
>> (sofia/external/2031701                          665 at 194.0.147.16:5060)
>> Callstate Change RINGING -> EARLY
>> 2010-12-01 11:56:01.534727 [DEBUG] sofia.c:4576 Channel
>> sofia/external/203170166                          5 at 194.0.147.16:5060skipping state [early][183]
>> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:676 Send signal
>> sofia/e                          xternal/2031701665 at 194.0.147.16:5060[BREAK]
>> 2010-12-01 11:56:01.534727 [INFO] switch_cpp.cpp:1181 All Answered
>> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application
>> playba                          ck Requires media! pre_answering channel
>> <sofia/external/2031701665 at 194.0.147.16>
>> sofia/external/2031701665 at 194.0.147.16:                          5060
>> EXECUTE sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswit
>> ch/sounds/svc_sound_files/default_autoattendant.wav)
>> 2010-12-01 11:56:01.537644 [DEBUG] switch_cpp.cpp:972
>> sofia/external/2031701665@                          194.0.147.16:5060destroy/unlink session from object
>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
>> set(effective_calle                          r_id_name=${404_tag})
>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
>> bridge(${404_dial})
>> 2010-12-01 11:56:01.537644 [DEBUG] switch_core_state_machine.c:119
>> (sofia/extern                          al/2031701665 at 194.0.147.16:5060)
>> State Change CS_ROUTING -> CS_EXECUTE
>> 2010-12-01 11:56:01.537644 [DEBUG] switch_core_session.c:1057 Send signal
>> sofia/                          external/2031701665 at 194.0.147.16:5060[BREAK]
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Sat Dec 18 12:33:02 2010
From: u2nsam at gmail.com (Sam)
Date: Sat, 18 Dec 2010 15:03:02 +0530
Subject: [Freeswitch-users] routing via hangup_cause
In-Reply-To: <AANLkTinXEWcS1LfDc4Nvz7BtosvYPWXjUFD9rrA1mGVe@mail.gmail.com>
References: <1292386719584-5837136.post@n2.nabble.com>
	<AANLkTiko+9M0MNeKgZO2aprbZg8LevA_7pHMSsKqn=AE@mail.gmail.com>
	<AANLkTi=-7N=f7PdSurgmxvr-9Y20Wk0G5N4v1+Z3zcmV@mail.gmail.com>
	<AANLkTiniD+yjP7pmeeGQSnFVEy_42MXBeEAxSVwLaTwm@mail.gmail.com>
	<AANLkTinXEWcS1LfDc4Nvz7BtosvYPWXjUFD9rrA1mGVe@mail.gmail.com>
Message-ID: <AANLkTi=BBHeZOUMk64m+YGBECyCCmR3R4vLmwH-PcOEj@mail.gmail.com>

Thnx Chris,


I was able to do that via ${originate_disposition} in XML, how can i do it
via lua script ?


Regds
Sam

On Sat, Dec 18, 2010 at 12:55 PM, Chris Burns [via freeswitch-users] <
ml-node+5847699-1581430113-292429 at n2.nabble.com<ml-node%2B5847699-1581430113-292429 at n2.nabble.com>
> wrote:

> First of all, your log output is different than your dialplan example.
> Second, you show only a tiny log snip taken after everything interesting
> occurred. All it shows is that you are having the issue you describe. Third,
> it seems like you are making multiple posts to the list for the same issue,
> and multiple emails within those requests. Slow down, take your time, try to
> learn as best you can (using the wiki), and give the list a reasonable
> amount of time to answer your questions. If you cant get a hangup cause in
> your XML dialplan, you dont need to start a seperate thread for how you cant
> do it in LUA either ... you see my point? :)
>
> The other end has to answer the call in order to hang up on you, as far as
> I know. If the originate fails you don't get a hangup cause into the
> bridge_hangup_cause channel variable. For instance, the other end never
> answers the call and originate fails with NO_USER_RESPONSE .... you will not
> have any value in bridge_hangup_cause because nothing ever bridged. Check
> the variable originate_disposition:
> http://wiki.freeswitch.org/wiki/Variable_originate_disposition
>
> Try out this extension using the "info" app to debug variables after your
> bridge:
> <extension name="funwithvariables">
>   <condition field="destination_number" expression="^1234$">
>     <action application="set" data="hangup_after_bridge=false"/>
>     <action application="set" data="continue_on_fail=true"/>
>     <action application="bridge" data="[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=0>
> "/>
>
>     <action application="info"/>
>   </condition>
> </extension>
>
>
> On Fri, Dec 17, 2010 at 11:12 PM, Sam <[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=1>
> > wrote:
>
>> Yes Bleg hangs up first but i am not able to fetch the hangupcause here,
>>
>>
>> hangup cause: ${bridge_hangup_cause}
>>
>> in logs i get as :-
>>
>> EXECUTE sofia/internal/[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=2>log(1 B-leg hangup Q850 cause: )
>> 2010-12-18 09:37:08.342225 [DEBUG] switch_core_state_machine.c:449
>> (sofia/external/[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=3>)
>> Callstate Change HANGUP -> DOWN
>>
>> 2010-12-18 09:37:08.342225 [ALERT] mod_dptools.c:1152 B-leg hangup Q850
>> cause:
>> 2010-12-18 09:37:08.342225 [DEBUG] switch_core_state_machine.c:452
>> (sofia/external/[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=4>)
>> Running State Change CS_DESTROY
>>
>>
>> Regds
>> Sam
>>
>>
>>
>>
>> On Sat, Dec 18, 2010 at 3:21 AM, Michael Collins <[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=5>
>> > wrote:
>>
>>> You're almost there. First off, you can delete this line:
>>> <action application="set" data="hangup_after_bridge=false"/>
>>> That's the default behavior and you have to set it prior to the bridge
>>> anyway.
>>>
>>> Move this line before the bridge:
>>> <action application="set" data="continue_on_fail=true"/>
>>> Otherwise it won't have any effect.
>>>
>>> The other stuff should work if the b-leg hangs up first, but not if the
>>> a-leg does.
>>>
>>> -MC
>>>
>>>
>>> On Thu, Dec 16, 2010 at 2:28 AM, Sam <[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=6>
>>> > wrote:
>>>
>>>>
>>>> I am not able to fetch the hangup cause in the dial-plan by using log
>>>> function .
>>>> I would like to use that hangup cause variable for further routing so
>>>> that
>>>> the routing would be conditional to hangup cause.
>>>>
>>>> Regards
>>>> Sam
>>>>
>>>> On Wed, Dec 15, 2010 at 9:48 AM, samir <[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=7>
>>>> > wrote:
>>>>
>>>>>
>>>>> Hello friends,
>>>>>
>>>>> was trying to create a routing rule to to route calls by accounting to
>>>>> the
>>>>> hangup causes,
>>>>>
>>>>> I have written below syntax but it fails to give the cause code to the
>>>>> varriable for routing.
>>>>>
>>>>> <extension name="refer">
>>>>>               <condition field="destination_number"
>>>>> expression="^1234$">
>>>>>
>>>>>
>>>>>       <action application="bridge" data="[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=8>
>>>>> "/>
>>>>>
>>>>>       <action application="set" data="hangup_after_bridge=false"/>
>>>>>       <action application="set" data="continue_on_fail=true"/>
>>>>>       <action application="log" data="1 B-leg hangup cause:
>>>>> ${bridge_hangup_cause}"/>
>>>>>       <action application="transfer" data="${bridge_hangup_cause} XML
>>>>> conbridge"/>
>>>>>
>>>>>               </condition>
>>>>>
>>>>>
>>>>>   </extension>
>>>>>
>>>>>       <extension name="no user">
>>>>>               <condition field="destination_number"
>>>>> expression="^(NO_USER_RESPONSE)$">
>>>>>               <action application="bridge"
>>>>> data="[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=9>
>>>>> "/>
>>>>>
>>>>>
>>>>>               </condition>
>>>>>       </extension>
>>>>>
>>>>>
>>>>> here the ${bridge_hangup_cause} is not getting executed. Am I doing it
>>>>> right
>>>>> or is there any other way to do it.
>>>>>
>>>>> Regards
>>>>> Sam
>>>>> --
>>>>> View this message in context:
>>>>> http://freeswitch-users.2379917.n2.nabble.com/routing-via-hangup-cause-tp5837136p5837136.html<http://freeswitch-users.2379917.n2.nabble.com/routing-via-hangup-cause-tp5837136p5837136.html?by-user=t>
>>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5847699&i=10>
>>>>>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5847699&i=11>
>>>>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5847699&i=12>
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5847699&i=13>
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> [hidden email] <http://user/SendEmail.jtp?type=node&node=5847699&i=14>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
>  View message @
> http://freeswitch-users.2379917.n2.nabble.com/routing-via-hangup-cause-tp5837136p5847699.html
>
> To unsubscribe from routing via hangup_cause, click here<http://freeswitch-users.2379917.n2.nabble.com/template/NamlServlet.jtp?macro=unsubscribe_by_code&node=5837136&code=dTJuc2FtQGdtYWlsLmNvbXw1ODM3MTM2fC0xMTk1MzYwMDk5>.
>
>
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From chris at cloudtel.com  Sat Dec 18 12:50:35 2010
From: chris at cloudtel.com (Chris Burns)
Date: Sat, 18 Dec 2010 04:50:35 -0500
Subject: [Freeswitch-users] routing via hangup_cause
In-Reply-To: <AANLkTi=BBHeZOUMk64m+YGBECyCCmR3R4vLmwH-PcOEj@mail.gmail.com>
References: <1292386719584-5837136.post@n2.nabble.com>
	<AANLkTiko+9M0MNeKgZO2aprbZg8LevA_7pHMSsKqn=AE@mail.gmail.com>
	<AANLkTi=-7N=f7PdSurgmxvr-9Y20Wk0G5N4v1+Z3zcmV@mail.gmail.com>
	<AANLkTiniD+yjP7pmeeGQSnFVEy_42MXBeEAxSVwLaTwm@mail.gmail.com>
	<AANLkTinXEWcS1LfDc4Nvz7BtosvYPWXjUFD9rrA1mGVe@mail.gmail.com>
	<AANLkTi=BBHeZOUMk64m+YGBECyCCmR3R4vLmwH-PcOEj@mail.gmail.com>
Message-ID: <AANLkTimFksnC1=_ESK3k4WDOaJ19vpUf1c+27eGSMdeL@mail.gmail.com>

Trust in the wiki

http://wiki.freeswitch.org/wiki/Lua
http://wiki.freeswitch.org/wiki/Lua#session:getVariable
http://wiki.freeswitch.org/wiki/Lua#session:transfer

Use getVariable to retrieve session variables, and then use transfer to send
them somewhere based on the contents of variables. There are plenty of
examples on the wiki for you to check and figure it out. Good luck!


On Sat, Dec 18, 2010 at 4:33 AM, Sam <u2nsam at gmail.com> wrote:

> Thnx Chris,
>
>
> I was able to do that via ${originate_disposition} in XML, how can i do it
> via lua script ?
>
>
> Regds
> Sam
>
> On Sat, Dec 18, 2010 at 12:55 PM, Chris Burns [via freeswitch-users] <
> ml-node+5847699-1581430113-292429 at n2.nabble.com<ml-node%2B5847699-1581430113-292429 at n2.nabble.com>
> > wrote:
>
>> First of all, your log output is different than your dialplan example.
>> Second, you show only a tiny log snip taken after everything interesting
>> occurred. All it shows is that you are having the issue you describe. Third,
>> it seems like you are making multiple posts to the list for the same issue,
>> and multiple emails within those requests. Slow down, take your time, try to
>> learn as best you can (using the wiki), and give the list a reasonable
>> amount of time to answer your questions. If you cant get a hangup cause in
>> your XML dialplan, you dont need to start a seperate thread for how you cant
>> do it in LUA either ... you see my point? :)
>>
>> The other end has to answer the call in order to hang up on you, as far as
>> I know. If the originate fails you don't get a hangup cause into the
>> bridge_hangup_cause channel variable. For instance, the other end never
>> answers the call and originate fails with NO_USER_RESPONSE .... you will not
>> have any value in bridge_hangup_cause because nothing ever bridged. Check
>> the variable originate_disposition:
>> http://wiki.freeswitch.org/wiki/Variable_originate_disposition
>>
>> Try out this extension using the "info" app to debug variables after your
>> bridge:
>> <extension name="funwithvariables">
>>   <condition field="destination_number" expression="^1234$">
>>      <action application="set" data="hangup_after_bridge=false"/>
>>     <action application="set" data="continue_on_fail=true"/>
>>     <action application="bridge" data="[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=0>
>> "/>
>>
>>     <action application="info"/>
>>   </condition>
>> </extension>
>>
>>
>> On Fri, Dec 17, 2010 at 11:12 PM, Sam <[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=1>
>> > wrote:
>>
>>> Yes Bleg hangs up first but i am not able to fetch the hangupcause here,
>>>
>>>
>>> hangup cause: ${bridge_hangup_cause}
>>>
>>> in logs i get as :-
>>>
>>> EXECUTE sofia/internal/[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=2>log(1 B-leg hangup Q850 cause: )
>>> 2010-12-18 09:37:08.342225 [DEBUG] switch_core_state_machine.c:449
>>> (sofia/external/[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=3>)
>>> Callstate Change HANGUP -> DOWN
>>>
>>> 2010-12-18 09:37:08.342225 [ALERT] mod_dptools.c:1152 B-leg hangup Q850
>>> cause:
>>> 2010-12-18 09:37:08.342225 [DEBUG] switch_core_state_machine.c:452
>>> (sofia/external/[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=4>)
>>> Running State Change CS_DESTROY
>>>
>>>
>>> Regds
>>> Sam
>>>
>>>
>>>
>>>
>>> On Sat, Dec 18, 2010 at 3:21 AM, Michael Collins <[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=5>
>>> > wrote:
>>>
>>>> You're almost there. First off, you can delete this line:
>>>> <action application="set" data="hangup_after_bridge=false"/>
>>>> That's the default behavior and you have to set it prior to the bridge
>>>> anyway.
>>>>
>>>> Move this line before the bridge:
>>>> <action application="set" data="continue_on_fail=true"/>
>>>> Otherwise it won't have any effect.
>>>>
>>>> The other stuff should work if the b-leg hangs up first, but not if the
>>>> a-leg does.
>>>>
>>>> -MC
>>>>
>>>>
>>>>  On Thu, Dec 16, 2010 at 2:28 AM, Sam <[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=6>
>>>> > wrote:
>>>>
>>>>>
>>>>> I am not able to fetch the hangup cause in the dial-plan by using log
>>>>> function .
>>>>> I would like to use that hangup cause variable for further routing so
>>>>> that
>>>>> the routing would be conditional to hangup cause.
>>>>>
>>>>> Regards
>>>>> Sam
>>>>>
>>>>> On Wed, Dec 15, 2010 at 9:48 AM, samir <[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=7>
>>>>> > wrote:
>>>>>
>>>>>>
>>>>>> Hello friends,
>>>>>>
>>>>>> was trying to create a routing rule to to route calls by accounting to
>>>>>> the
>>>>>> hangup causes,
>>>>>>
>>>>>> I have written below syntax but it fails to give the cause code to the
>>>>>> varriable for routing.
>>>>>>
>>>>>> <extension name="refer">
>>>>>>               <condition field="destination_number"
>>>>>> expression="^1234$">
>>>>>>
>>>>>>
>>>>>>       <action application="bridge" data="[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=8>
>>>>>> "/>
>>>>>>
>>>>>>       <action application="set" data="hangup_after_bridge=false"/>
>>>>>>       <action application="set" data="continue_on_fail=true"/>
>>>>>>       <action application="log" data="1 B-leg hangup cause:
>>>>>> ${bridge_hangup_cause}"/>
>>>>>>       <action application="transfer" data="${bridge_hangup_cause} XML
>>>>>> conbridge"/>
>>>>>>
>>>>>>               </condition>
>>>>>>
>>>>>>
>>>>>>   </extension>
>>>>>>
>>>>>>       <extension name="no user">
>>>>>>               <condition field="destination_number"
>>>>>> expression="^(NO_USER_RESPONSE)$">
>>>>>>               <action application="bridge"
>>>>>> data="[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=9>
>>>>>> "/>
>>>>>>
>>>>>>
>>>>>>               </condition>
>>>>>>       </extension>
>>>>>>
>>>>>>
>>>>>> here the ${bridge_hangup_cause} is not getting executed. Am I doing it
>>>>>> right
>>>>>> or is there any other way to do it.
>>>>>>
>>>>>> Regards
>>>>>> Sam
>>>>>> --
>>>>>> View this message in context:
>>>>>> http://freeswitch-users.2379917.n2.nabble.com/routing-via-hangup-cause-tp5837136p5837136.html<http://freeswitch-users.2379917.n2.nabble.com/routing-via-hangup-cause-tp5837136p5837136.html?by-user=t>
>>>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> [hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=10>
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5847699&i=11>
>>>>>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5847699&i=12>
>>>>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5847699&i=13>
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5847699&i=14>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> ------------------------------
>>  View message @
>> http://freeswitch-users.2379917.n2.nabble.com/routing-via-hangup-cause-tp5837136p5847699.html
>>
>> To unsubscribe from routing via hangup_cause, click here<http://freeswitch-users.2379917.n2.nabble.com/template/NamlServlet.jtp?macro=unsubscribe_by_code&node=5837136&code=dTJuc2FtQGdtYWlsLmNvbXw1ODM3MTM2fC0xMTk1MzYwMDk5>.
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Sat Dec 18 13:00:36 2010
From: u2nsam at gmail.com (Sam)
Date: Sat, 18 Dec 2010 15:30:36 +0530
Subject: [Freeswitch-users] routing via hangup_cause
In-Reply-To: <AANLkTimFksnC1=_ESK3k4WDOaJ19vpUf1c+27eGSMdeL@mail.gmail.com>
References: <1292386719584-5837136.post@n2.nabble.com>
	<AANLkTiko+9M0MNeKgZO2aprbZg8LevA_7pHMSsKqn=AE@mail.gmail.com>
	<AANLkTi=-7N=f7PdSurgmxvr-9Y20Wk0G5N4v1+Z3zcmV@mail.gmail.com>
	<AANLkTiniD+yjP7pmeeGQSnFVEy_42MXBeEAxSVwLaTwm@mail.gmail.com>
	<AANLkTinXEWcS1LfDc4Nvz7BtosvYPWXjUFD9rrA1mGVe@mail.gmail.com>
	<AANLkTi=BBHeZOUMk64m+YGBECyCCmR3R4vLmwH-PcOEj@mail.gmail.com>
	<AANLkTimFksnC1=_ESK3k4WDOaJ19vpUf1c+27eGSMdeL@mail.gmail.com>
Message-ID: <AANLkTinCS9uFDV4ziQpHG9WBVCq6F2FNgh=9TnugQgDh@mail.gmail.com>

Thanks for helping Chris .. i got the desired !

Regards
Sam



On Sat, Dec 18, 2010 at 3:20 PM, Chris Burns <chris at cloudtel.com> wrote:

> Trust in the wiki
>
> http://wiki.freeswitch.org/wiki/Lua
> http://wiki.freeswitch.org/wiki/Lua#session:getVariable
> http://wiki.freeswitch.org/wiki/Lua#session:transfer
>
> Use getVariable to retrieve session variables, and then use transfer to
> send them somewhere based on the contents of variables. There are plenty of
> examples on the wiki for you to check and figure it out. Good luck!
>
>
> On Sat, Dec 18, 2010 at 4:33 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Thnx Chris,
>>
>>
>> I was able to do that via ${originate_disposition} in XML, how can i do
>> it via lua script ?
>>
>>
>> Regds
>> Sam
>>
>> On Sat, Dec 18, 2010 at 12:55 PM, Chris Burns [via freeswitch-users] <
>> ml-node+5847699-1581430113-292429 at n2.nabble.com<ml-node%2B5847699-1581430113-292429 at n2.nabble.com>
>> > wrote:
>>
>>> First of all, your log output is different than your dialplan example.
>>> Second, you show only a tiny log snip taken after everything interesting
>>> occurred. All it shows is that you are having the issue you describe. Third,
>>> it seems like you are making multiple posts to the list for the same issue,
>>> and multiple emails within those requests. Slow down, take your time, try to
>>> learn as best you can (using the wiki), and give the list a reasonable
>>> amount of time to answer your questions. If you cant get a hangup cause in
>>> your XML dialplan, you dont need to start a seperate thread for how you cant
>>> do it in LUA either ... you see my point? :)
>>>
>>> The other end has to answer the call in order to hang up on you, as far
>>> as I know. If the originate fails you don't get a hangup cause into the
>>> bridge_hangup_cause channel variable. For instance, the other end never
>>> answers the call and originate fails with NO_USER_RESPONSE .... you will not
>>> have any value in bridge_hangup_cause because nothing ever bridged. Check
>>> the variable originate_disposition:
>>> http://wiki.freeswitch.org/wiki/Variable_originate_disposition
>>>
>>> Try out this extension using the "info" app to debug variables after your
>>> bridge:
>>> <extension name="funwithvariables">
>>>   <condition field="destination_number" expression="^1234$">
>>>      <action application="set" data="hangup_after_bridge=false"/>
>>>     <action application="set" data="continue_on_fail=true"/>
>>>     <action application="bridge" data="[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=0>
>>> "/>
>>>
>>>     <action application="info"/>
>>>   </condition>
>>> </extension>
>>>
>>>
>>> On Fri, Dec 17, 2010 at 11:12 PM, Sam <[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=1>
>>> > wrote:
>>>
>>>> Yes Bleg hangs up first but i am not able to fetch the hangupcause here,
>>>>
>>>>
>>>> hangup cause: ${bridge_hangup_cause}
>>>>
>>>> in logs i get as :-
>>>>
>>>> EXECUTE sofia/internal/[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=2>log(1 B-leg hangup Q850 cause: )
>>>> 2010-12-18 09:37:08.342225 [DEBUG] switch_core_state_machine.c:449
>>>> (sofia/external/[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=3>)
>>>> Callstate Change HANGUP -> DOWN
>>>>
>>>> 2010-12-18 09:37:08.342225 [ALERT] mod_dptools.c:1152 B-leg hangup Q850
>>>> cause:
>>>> 2010-12-18 09:37:08.342225 [DEBUG] switch_core_state_machine.c:452
>>>> (sofia/external/[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=4>)
>>>> Running State Change CS_DESTROY
>>>>
>>>>
>>>> Regds
>>>> Sam
>>>>
>>>>
>>>>
>>>>
>>>> On Sat, Dec 18, 2010 at 3:21 AM, Michael Collins <[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=5>
>>>> > wrote:
>>>>
>>>>> You're almost there. First off, you can delete this line:
>>>>> <action application="set" data="hangup_after_bridge=false"/>
>>>>> That's the default behavior and you have to set it prior to the bridge
>>>>> anyway.
>>>>>
>>>>> Move this line before the bridge:
>>>>> <action application="set" data="continue_on_fail=true"/>
>>>>> Otherwise it won't have any effect.
>>>>>
>>>>> The other stuff should work if the b-leg hangs up first, but not if the
>>>>> a-leg does.
>>>>>
>>>>> -MC
>>>>>
>>>>>
>>>>>  On Thu, Dec 16, 2010 at 2:28 AM, Sam <[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=6>
>>>>> > wrote:
>>>>>
>>>>>>
>>>>>> I am not able to fetch the hangup cause in the dial-plan by using log
>>>>>> function .
>>>>>> I would like to use that hangup cause variable for further routing so
>>>>>> that
>>>>>> the routing would be conditional to hangup cause.
>>>>>>
>>>>>> Regards
>>>>>> Sam
>>>>>>
>>>>>> On Wed, Dec 15, 2010 at 9:48 AM, samir <[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=7>
>>>>>> > wrote:
>>>>>>
>>>>>>>
>>>>>>> Hello friends,
>>>>>>>
>>>>>>> was trying to create a routing rule to to route calls by accounting
>>>>>>> to the
>>>>>>> hangup causes,
>>>>>>>
>>>>>>> I have written below syntax but it fails to give the cause code to
>>>>>>> the
>>>>>>> varriable for routing.
>>>>>>>
>>>>>>> <extension name="refer">
>>>>>>>               <condition field="destination_number"
>>>>>>> expression="^1234$">
>>>>>>>
>>>>>>>
>>>>>>>       <action application="bridge" data="[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=8>
>>>>>>> "/>
>>>>>>>
>>>>>>>       <action application="set" data="hangup_after_bridge=false"/>
>>>>>>>       <action application="set" data="continue_on_fail=true"/>
>>>>>>>       <action application="log" data="1 B-leg hangup cause:
>>>>>>> ${bridge_hangup_cause}"/>
>>>>>>>       <action application="transfer" data="${bridge_hangup_cause} XML
>>>>>>> conbridge"/>
>>>>>>>
>>>>>>>               </condition>
>>>>>>>
>>>>>>>
>>>>>>>   </extension>
>>>>>>>
>>>>>>>       <extension name="no user">
>>>>>>>               <condition field="destination_number"
>>>>>>> expression="^(NO_USER_RESPONSE)$">
>>>>>>>               <action application="bridge"
>>>>>>> data="[hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=9>
>>>>>>> "/>
>>>>>>>
>>>>>>>
>>>>>>>               </condition>
>>>>>>>       </extension>
>>>>>>>
>>>>>>>
>>>>>>> here the ${bridge_hangup_cause} is not getting executed. Am I doing
>>>>>>> it right
>>>>>>> or is there any other way to do it.
>>>>>>>
>>>>>>> Regards
>>>>>>> Sam
>>>>>>> --
>>>>>>> View this message in context:
>>>>>>> http://freeswitch-users.2379917.n2.nabble.com/routing-via-hangup-cause-tp5837136p5837136.html<http://freeswitch-users.2379917.n2.nabble.com/routing-via-hangup-cause-tp5837136p5837136.html?by-user=t>
>>>>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> [hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=10>
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> [hidden email]<http://user/SendEmail.jtp?type=node&node=5847699&i=11>
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5847699&i=12>
>>>>>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5847699&i=13>
>>>>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5847699&i=14>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> ------------------------------
>>>  View message @
>>> http://freeswitch-users.2379917.n2.nabble.com/routing-via-hangup-cause-tp5837136p5847699.html
>>>
>>> To unsubscribe from routing via hangup_cause, click here<http://freeswitch-users.2379917.n2.nabble.com/template/NamlServlet.jtp?macro=unsubscribe_by_code&node=5837136&code=dTJuc2FtQGdtYWlsLmNvbXw1ODM3MTM2fC0xMTk1MzYwMDk5>.
>>>
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Sat Dec 18 15:01:19 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 18 Dec 2010 12:01:19 +0000
Subject: [Freeswitch-users] hangup cause
In-Reply-To: <AANLkTi=gGs8iDB-GO7twdergCEK2+V6bObtsJS-FeVBa@mail.gmail.com>
References: <AANLkTi=gGs8iDB-GO7twdergCEK2+V6bObtsJS-FeVBa@mail.gmail.com>
Message-ID: <AANLkTi=somsAp185QqCTzPjf8P_pkCbPC_f2y3_kSt0h@mail.gmail.com>

It is already in a variable:

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

-Steve


On 18 December 2010 06:59, Sam <u2nsam at gmail.com> wrote:
> hello,
>
> How can i fetch hangup cause and store it in an variable.
>
>
> [INFO] mod_dptools.c:2579 Originate Failed.? Cause: NO_USER_RESPONSE
>
>
>
> Regards
> Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From rafonline at hotmail.com  Sat Dec 18 17:11:14 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Sat, 18 Dec 2010 14:11:14 +0000
Subject: [Freeswitch-users] audio quality issue
In-Reply-To: <AANLkTindj9jGDMYn9tLuuN1c3uSX=vC19s_F7pCmBHvC@mail.gmail.com>
References: <COL122-W33885B610991D11E824AB6B5160@phx.gbl>,
	<COL122-W5833A44AAF7FDFD440BCBB5160@phx.gbl>,
	<AANLkTinEhrDUHJCfr7RC=9Qm-uYhpFAsx-RJcc7bD4FC@mail.gmail.com>,
	<COL122-W5661F836F7A1ED2E6D4FB6B5160@phx.gbl>,
	<AANLkTindj9jGDMYn9tLuuN1c3uSX=vC19s_F7pCmBHvC@mail.gmail.com>
Message-ID: <COL122-W35C48A904B9ED67066F154B5170@phx.gbl>



I upgraded to dev snapshot (still on Xen) - still had audio issues.

I got rid of xen server and installed centos 5.5 and freeswitch only - timer frequency warning disappeared - still felt audio was not what it should be.  

What I still dont understand is why when i playback my recorded voicemail messages the quality is good.  

Sometimes I even have issues leaving a message after the voicemail tone, the IVR seems to think i have ended recording the message before even starting.  The voicemail tone itself can be very high pitch at times.

Any help will be much appreciated.

Chris - can you please point me to some docs on how to improve my kernel (sorry but i am newbie to unix).

Cheers

Raf


Date: Fri, 17 Dec 2010 18:05:09 -0500
From: chris at cloudtel.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] audio quality issue

Improve the kernel and in your situation you will most likely improve your audio. You can't expect perfect audio when the thread handling RTP is waking up later than its designated time (after sleeping for the codec timing). The switch tests for the possibility of this issue on startup when it calibrates your clock offset ... thus it warned you :)


"-nocal" will simply skip the process of recognizing and warning that your kernel timer is slow, but will not alter performance in your situation (unless I am missing something)


On Fri, Dec 17, 2010 at 4:30 PM, Rafqat . <rafonline at hotmail.com> wrote:








I will try the dev snapshot. 

cheers

Raf

> Date: Fri, 17 Dec 2010 15:28:49 -0600
> From: anthony.minessale at gmail.com


> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] audio quality issue

> 
> if you insist to remain on 1.0.6 try it with -vm -nocal cmd line options
> as i already stated you should try the development snapshot
> 
> On Fri, Dec 17, 2010 at 3:22 PM, Rafqat . <rafonline at hotmail.com> wrote:


> >
> > btw.  I have no audio issues when making calls between phones through
> > freeswitch (voice quality is very good).
> >
> >
> >
> > ________________________________


> > From: rafonline at hotmail.com
> > To: freeswitch-users at lists.freeswitch.org


> > Date: Fri, 17 Dec 2010 21:19:11 +0000
> > Subject: [Freeswitch-users] audio quality issue
> >
> >
> >
> > Hi,
> >
> > I recently installed freeswitch 1.0.6 on centos 5.5 (hosted on xen server).


> > I feel as if the audio quality is not as good as what it should be (6-7 out
> > of 10).  This is apparent in calling the default voicemail IVR, sometimes I
> > have issues leaving a message, sometimes I hear some weird noises.  However,


> > I have no issues with the quality of the recorded voicemail message I leave,
> > it sounds fine on playback.
> >
> > Is this an issue with my phones (Flexor 500)?  I have a softphone which


> > doesn't seem as bad but still has issues.
> >
> > I also get the following message on startup of freeswitch:
> >
> > 2010-12-17 21:17:39.138958 [WARNING] switch_time.c:206 Timer resolution of


> > 4000 microseconds detected!
> > Do you have your kernel timer frequency set to lower than 1,000Hz? You may
> > experience audio problems.
> >
> > I assume its related.
> >


> > Any help will be much appreciated.
> >
> > Cheers
> >
> > Raf
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org


> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users


> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org


> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users


> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/


> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 

> AIM: anthm

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


> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org


> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________


> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users


> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


 		 	   		  

_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
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From Avi at aMarcus.com  Sat Dec 18 18:54:50 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Sat, 18 Dec 2010 17:54:50 +0200
Subject: [Freeswitch-users] questions about google voice and dingaling
In-Reply-To: <AANLkTin9rdRQcSB2yNcROS4RpBSOSdOrMQo+01CoZrWc@mail.gmail.com>
References: <AANLkTimyXDinQiDZs-THu_qVwx_nKhsMtov6sp-KuGKA@mail.gmail.com>
	<1292592700991-5845263.post@n2.nabble.com>
	<AANLkTin9rdRQcSB2yNcROS4RpBSOSdOrMQo+01CoZrWc@mail.gmail.com>
Message-ID: <AANLkTikcxC35Yw-bCN3h3iF52e1hZ=zKYmwhgOBJ5xQX@mail.gmail.com>

I had no trouble with adding a second google voice account, however
dingaling seems to have a 2-3 second delay on the audio. Setting RTP timers
to none makes the audio drop packets.
-Avi

On Fri, Dec 17, 2010 at 5:56 PM, joy this <thisjoy0528 at gmail.com> wrote:

> I found the situation yesterday. I am sure that there is only one account.
> I try to add another account today, and FS crashed.
>
> Sincerely yours,
>
> Thisjoy.
> 2010/12/17 mazilo <Nabble at slickdeals.endjunk.com>
>
>>
>>
>> joy this wrote:
>> > The question is that, I can call two different phone numbers with two
>> > different sip accounts via the same GVoice account.
>> Your configuration showed you have two GV accounts. R U sure you are ONLY
>> using a single GV account to place two concurrent calls?
>>
>> -----
>> don't and stop are the ONLY two 4-letter words considered offensive to
>> men,
>> but not when used together.
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/questions-about-google-voice-and-dingaling-tp5845051p5845263.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mel0torme at gmail.com  Sat Dec 18 21:21:39 2010
From: mel0torme at gmail.com (Tom C)
Date: Sat, 18 Dec 2010 10:21:39 -0800
Subject: [Freeswitch-users] Linux IDE for debugging FS?
Message-ID: <AANLkTinwG-0StzvFhEYXSwmepNc6fZ0=LqmarPy2=Eab@mail.gmail.com>

What "Integrated Development Environment" do you use for developing and
debugging FreeSwitch on Linux?

I'd like to install an IDE on my Debian Lenny box.  I tried installing
Eclipse, but it said that my machine (P4) was not supported.  And being a
Linux newbie, I have no idea what is out there that's good.

Ultimately I want to be able to connect to my DockStar and debug the
instance of FS running there.
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From david.ponzone at ipeva.fr  Sat Dec 18 21:26:26 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sat, 18 Dec 2010 19:26:26 +0100
Subject: [Freeswitch-users] Linux IDE for debugging FS?
In-Reply-To: <AANLkTinwG-0StzvFhEYXSwmepNc6fZ0=LqmarPy2=Eab@mail.gmail.com>
References: <AANLkTinwG-0StzvFhEYXSwmepNc6fZ0=LqmarPy2=Eab@mail.gmail.com>
Message-ID: <9B704162-95FC-4F36-A29E-438F9AFB7674@ipeva.fr>

You mean for writing and debugging XML ?
Personally, I use vi.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 18/12/2010 ? 19:21, Tom C a ?crit :

> What "Integrated Development Environment" do you use for developing and debugging FreeSwitch on Linux?
> 
> I'd like to install an IDE on my Debian Lenny box.  I tried installing Eclipse, but it said that my machine (P4) was not supported.  And being a Linux newbie, I have no idea what is out there that's good.
> 
> Ultimately I want to be able to connect to my DockStar and debug the instance of FS running there.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mel0torme at gmail.com  Sat Dec 18 21:34:50 2010
From: mel0torme at gmail.com (Tom C)
Date: Sat, 18 Dec 2010 10:34:50 -0800
Subject: [Freeswitch-users] Linux IDE for debugging FS?
In-Reply-To: <9B704162-95FC-4F36-A29E-438F9AFB7674@ipeva.fr>
References: <AANLkTinwG-0StzvFhEYXSwmepNc6fZ0=LqmarPy2=Eab@mail.gmail.com>
	<9B704162-95FC-4F36-A29E-438F9AFB7674@ipeva.fr>
Message-ID: <AANLkTimSXK8T=KQL8xA7RwWzVio__ik+CxAqOLS_N_yR@mail.gmail.com>

I mean debugging the C source and being able to set breakpoints and step
through the code, checking the contents of memory at each step.  And being
able to connect to an instance of FreeSwitch running on another machine and
step through the code.

On Sat, Dec 18, 2010 at 10:26 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> You mean for writing and debugging XML ?
> Personally, I use vi.
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 18/12/2010 ? 19:21, Tom C a ?crit :
>
> What "Integrated Development Environment" do you use for developing and
> debugging FreeSwitch on Linux?
>
> I'd like to install an IDE on my Debian Lenny box.  I tried installing
> Eclipse, but it said that my machine (P4) was not supported.  And being a
> Linux newbie, I have no idea what is out there that's good.
>
> Ultimately I want to be able to connect to my DockStar and debug the
> instance of FS running there.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From freeswitch-list at puzzled.xs4all.nl  Sat Dec 18 21:46:50 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Sat, 18 Dec 2010 19:46:50 +0100
Subject: [Freeswitch-users] Linux IDE for debugging FS?
In-Reply-To: <AANLkTimSXK8T=KQL8xA7RwWzVio__ik+CxAqOLS_N_yR@mail.gmail.com>
References: <AANLkTinwG-0StzvFhEYXSwmepNc6fZ0=LqmarPy2=Eab@mail.gmail.com>	<9B704162-95FC-4F36-A29E-438F9AFB7674@ipeva.fr>
	<AANLkTimSXK8T=KQL8xA7RwWzVio__ik+CxAqOLS_N_yR@mail.gmail.com>
Message-ID: <4D0D019A.2090809@puzzled.xs4all.nl>

On 12/18/2010 07:34 PM, Tom C wrote:
> I mean debugging the C source and being able to set breakpoints and step
> through the code, checking the contents of memory at each step.  And
> being able to connect to an instance of FreeSwitch running on another
> machine and step through the code.

How about gdb and valgrind:
http://www.gnu.org/software/gdb/
http://valgrind.org/

I don't know if you can connect with either to an instance running on
another machine.

Regards,
Patrick


From thisjoy0528 at gmail.com  Sat Dec 18 21:51:23 2010
From: thisjoy0528 at gmail.com (joy this)
Date: Sun, 19 Dec 2010 02:51:23 +0800
Subject: [Freeswitch-users] questions about google voice and dingaling
In-Reply-To: <AANLkTikcxC35Yw-bCN3h3iF52e1hZ=zKYmwhgOBJ5xQX@mail.gmail.com>
References: <AANLkTimyXDinQiDZs-THu_qVwx_nKhsMtov6sp-KuGKA@mail.gmail.com>
	<1292592700991-5845263.post@n2.nabble.com>
	<AANLkTin9rdRQcSB2yNcROS4RpBSOSdOrMQo+01CoZrWc@mail.gmail.com>
	<AANLkTikcxC35Yw-bCN3h3iF52e1hZ=zKYmwhgOBJ5xQX@mail.gmail.com>
Message-ID: <AANLkTinjks+CRFp+i1kbFbGGDvQ2gHAgiupeMXTeOzTe@mail.gmail.com>

If I add another account, it sometimes would not crash. When the error
window appears and I click the "igore" button, FS window will not be closed.
I could input "dingaling status", and there is only one account arthorized.
I tried some different versions of Guntls, and the error message "
libgcrypt-1.4.6" would be "libgcrypt-1.4.4" or others.

Sincerely yours,

Thisjoy.

2010/12/18 Avi Marcus <Avi at amarcus.com>

> I had no trouble with adding a second google voice account, however
> dingaling seems to have a 2-3 second delay on the audio. Setting RTP timers
> to none makes the audio drop packets.
> -Avi
>
>
> On Fri, Dec 17, 2010 at 5:56 PM, joy this <thisjoy0528 at gmail.com> wrote:
>
>> I found the situation yesterday. I am sure that there is only one account.
>> I try to add another account today, and FS crashed.
>>
>> Sincerely yours,
>>
>> Thisjoy.
>> 2010/12/17 mazilo <Nabble at slickdeals.endjunk.com>
>>
>>>
>>>
>>> joy this wrote:
>>> > The question is that, I can call two different phone numbers with two
>>> > different sip accounts via the same GVoice account.
>>> Your configuration showed you have two GV accounts. R U sure you are ONLY
>>> using a single GV account to place two concurrent calls?
>>>
>>> -----
>>> don't and stop are the ONLY two 4-letter words considered offensive to
>>> men,
>>> but not when used together.
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/questions-about-google-voice-and-dingaling-tp5845051p5845263.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris at cloudtel.com  Sat Dec 18 22:48:44 2010
From: chris at cloudtel.com (Chris Burns)
Date: Sat, 18 Dec 2010 14:48:44 -0500
Subject: [Freeswitch-users] Linux IDE for debugging FS?
In-Reply-To: <4D0D019A.2090809@puzzled.xs4all.nl>
References: <AANLkTinwG-0StzvFhEYXSwmepNc6fZ0=LqmarPy2=Eab@mail.gmail.com>
	<9B704162-95FC-4F36-A29E-438F9AFB7674@ipeva.fr>
	<AANLkTimSXK8T=KQL8xA7RwWzVio__ik+CxAqOLS_N_yR@mail.gmail.com>
	<4D0D019A.2090809@puzzled.xs4all.nl>
Message-ID: <AANLkTinLWcn_f6+1AVCDmw1AnjwjsXMQ1LTjrtiRZBrg@mail.gmail.com>

I guess Eclipse CDT is what you are looking for, but I have never used it.
It always seemed efficient to me to just deal with gdb directly, and build
using make + related tools

gdb supports remote debugging via gdbserver but I have never found a reason
to use it

For an editor my personal choice is KDE's kate, just for ease of working
with remote files transparently and using my mouse once in a while (I know
... I'm such a noob). Men of stiffer resolve than myself swear by emacs or
vim ... choosing one over the other defines you forever as a programmer so
be careful (slightly joking). If you want to really go out on a limb you
chould use nano, but everyone will laugh at you :D

On Sat, Dec 18, 2010 at 1:46 PM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 12/18/2010 07:34 PM, Tom C wrote:
> > I mean debugging the C source and being able to set breakpoints and step
> > through the code, checking the contents of memory at each step.  And
> > being able to connect to an instance of FreeSwitch running on another
> > machine and step through the code.
>
> How about gdb and valgrind:
> http://www.gnu.org/software/gdb/
> http://valgrind.org/
>
> I don't know if you can connect with either to an instance running on
> another machine.
>
> Regards,
> Patrick
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mail at jankubr.com  Sat Dec 18 23:14:25 2010
From: mail at jankubr.com (Jan Kubr)
Date: Sat, 18 Dec 2010 21:14:25 +0100
Subject: [Freeswitch-users] SIP Registration DNS Error
In-Reply-To: <AANLkTins8Z6c=dHp=fDta8u9WEmdahz+vqeD7vfvuxOa@mail.gmail.com>
References: <06502C073AD9394AADB3CA7FD94931BC0519DF46@okc1x1.Logixcom.com>
	<AANLkTi=10HbngBxzwzf=i5PpKMhG5tdzWvtcCdA_weJt@mail.gmail.com>
	<49E1E293-BE37-43AA-960C-56B921FEBC9D@ipeva.fr>
	<AANLkTins8Z6c=dHp=fDta8u9WEmdahz+vqeD7vfvuxOa@mail.gmail.com>
Message-ID: <AANLkTimv_dDh5ynwuOD89XMqatw929YMC=PNL1gOM2Gf@mail.gmail.com>

OK they don't want me to register on the IP anymore.

What does this "DNS Error [503]" error mean exactly? When I nslookup the
domain on the same server, it gives me the correct answer. Right now it
doesn't even help when I restart the profile or whole FreeSWITCH, I keep
getting this DNS Error.

Thanks.

Jan

On Wed, Dec 8, 2010 at 4:47 PM, Jan Kubr <mail at jankubr.com> wrote:

> I'm sure they have a DNS issue (although it's a different provider), but I
> was more interested in why the retries fail even when everything is working
> again on their side. And only restart helps.
>
> JK
>
>
> On Wed, Dec 8, 2010 at 4:05 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> I think I remember that Flowroute has a DNS issue.
>> At least that was the case some months ago.
>> Use an IP, that should fix it.
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 08/12/2010 ? 16:01, Jan Kubr a ?crit :
>>
>> I have the same problem with one gateway. What also helps is restarting
>> the profile:
>>
>> sofia profile external restart reloadxml
>>
>> But that means dropping the calls as well.
>>
>> Anyone knows how to fix this?
>>
>> Thanks,
>> Jan
>>
>> On Tue, Oct 12, 2010 at 7:27 PM, Joshua Foshee <
>> Joshua.Foshee at logixcom.com> wrote:
>>
>>> I have setup up two sip providers that I can connect to fine but after a
>>> while I then start to get these messages.
>>>
>>>
>>> 2010-10-12 07:27:57.292742 [NOTICE] sofia_reg.c:342 Registering flowroute
>>>
>>> 2010-10-12 07:27:57.295138 [ERR] sofia_reg.c:1611 flowroute Registration
>>> Failed with status DNS Error [503]. failure #1391
>>>
>>> 2010-10-12 07:27:59.959022 [WARNING] sofia_reg.c:387 flowroute Failed
>>> Registration, setting retry to 30 seconds.
>>>
>>>
>>> Here is the output of the gateway status
>>>
>>>
>>> Name            flowroute
>>>
>>> Profile         external
>>>
>>> Scheme          Digest
>>>
>>> Realm           sip.flowroute.com
>>>
>>> Username        xxxxxxx
>>>
>>> Password        yes
>>>
>>> From            <sip:xxxxxxxxx at sip.flowroute.com<sip%3Axxxxxxxxx at sip.flowroute.com>
>>> ;transport=udp>
>>>
>>> Contact         <
>>> sip:gw+flowroute at xxxxxxxx:5080;transport=udp;gw=flowroute>
>>>
>>> Exten           xxxxxxx
>>>
>>> To              sip:xxxxxxxxx at sip.flowroute.com<sip%3Axxxxxxxxx at sip.flowroute.com>
>>>
>>> Proxy           sip:sip.flowroute.com
>>>
>>> Context         public
>>>
>>> Expires         600
>>>
>>> Freq            600
>>>
>>> Ping            1286835829
>>>
>>> PingFreq        25
>>>
>>> PingState       -1/0/1
>>>
>>> State           FAIL_WAIT
>>>
>>> Status          DOWN
>>>
>>> CallsIN         0
>>>
>>> CallsOUT        5
>>>
>>> FailedCallsIN   0
>>>
>>> FailedCallsOUT  5
>>>
>>>
>>>
>>> Name            broadvoice
>>>
>>> Profile         external
>>>
>>> Scheme          Digest
>>>
>>> Realm           BroadWorks
>>>
>>> Username        xxxxxxxx
>>>
>>> Password        yes
>>>
>>> From            <sip:xxxxxxxxxxxx at sip.broadvoice.com<sip%3Axxxxxxxxxxxx at sip.broadvoice.com>
>>> ;transport=udp>
>>>
>>> Contact         <
>>> sip:gw+broadvoice at xxxxxxxxxx:5080;transport=udp;gw=broadvoice>
>>>
>>> Exten           xxxxxxxx
>>>
>>> To              sip:xxxxxxxxxx at sip.broadvoice.com<sip%3Axxxxxxxxxx at sip.broadvoice.com>
>>>
>>> Proxy           sip:sip.broadvoice.com
>>>
>>> Context         public
>>>
>>> Expires         30
>>>
>>> Freq            30
>>>
>>> Ping            0
>>>
>>> PingFreq        0
>>>
>>> PingState       0/0/0
>>>
>>> State           FAIL_WAIT
>>>
>>> Status          DOWN
>>>
>>> CallsIN         0
>>>
>>> CallsOUT        2
>>>
>>> FailedCallsIN   0
>>>
>>> FailedCallsOUT  3
>>>
>>>
>>> If I restart Freeswitch process they both come up and Reg just fine for a
>>> while till it fails again.
>>>
>>>
>>> Thanks in advance,
>>>
>>> Josh
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From shamun.toha at gmail.com  Sat Dec 18 23:48:34 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Sat, 18 Dec 2010 21:48:34 +0100
Subject: [Freeswitch-users] FreeSwitch cant handle Pre-Paid ?
Message-ID: <AANLkTi=2bJiYVsTVD6Q+mtG7_RcRqRGLWe9gbPSThu+q@mail.gmail.com>

Hello,

FreeSwitch gets freezed when i try to make it for Pre-Paid track.

ex:
while(session.ready())
{
   curl(....balance...);
}

I tried many possibilities but still i cant make this setTimeout() function
working. How can i track when call is connected, seperate option to allow me
tracking that call.

Is it really not possible ? is it better to have other switch for this,
because FreeSwitch cant allow developers to put there own javascript
function while call is connected ? Should we use for this other switch
? ex: OpenSIPS
or Kamilio.

Please kindly guide, project is taking way too long only for this feature,
dont found any wiki guide about this info.

Thanks & Regards
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From chris at cloudtel.com  Sun Dec 19 01:15:19 2010
From: chris at cloudtel.com (Chris Burns)
Date: Sat, 18 Dec 2010 17:15:19 -0500
Subject: [Freeswitch-users] FreeSwitch cant handle Pre-Paid ?
In-Reply-To: <AANLkTi=2bJiYVsTVD6Q+mtG7_RcRqRGLWe9gbPSThu+q@mail.gmail.com>
References: <AANLkTi=2bJiYVsTVD6Q+mtG7_RcRqRGLWe9gbPSThu+q@mail.gmail.com>
Message-ID: <AANLkTin-3pSBKmx+xfks7vL=+=F3aBZ6rWHBvwFK7Y4Z@mail.gmail.com>

mod_nibblebill does prepaid billing ... if you search the wiki for "billing"
it is the first result :D

If you wanted to properly implement your own billing, you should consider
using the event socket and subscribing to events. Trying to create a prepaid
billing system using lua or javascript is going to be an uphill battle

On Sat, Dec 18, 2010 at 3:48 PM, Shamun toha md <shamun.toha at gmail.com>wrote:

> Hello,
>
> FreeSwitch gets freezed when i try to make it for Pre-Paid track.
>
> ex:
> while(session.ready())
> {
>    curl(....balance...);
> }
>
> I tried many possibilities but still i cant make this setTimeout() function
> working. How can i track when call is connected, seperate option to allow me
> tracking that call.
>
> Is it really not possible ? is it better to have other switch for this,
> because FreeSwitch cant allow developers to put there own javascript
> function while call is connected ? Should we use for this other switch ? ex: OpenSIPS
> or Kamilio.
>
> Please kindly guide, project is taking way too long only for this feature,
> dont found any wiki guide about this info.
>
> Thanks & Regards
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From shamun.toha at gmail.com  Sun Dec 19 01:54:12 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Sat, 18 Dec 2010 23:54:12 +0100
Subject: [Freeswitch-users] FreeSwitch cant handle Pre-Paid ?
In-Reply-To: <AANLkTin-3pSBKmx+xfks7vL=+=F3aBZ6rWHBvwFK7Y4Z@mail.gmail.com>
References: <AANLkTi=2bJiYVsTVD6Q+mtG7_RcRqRGLWe9gbPSThu+q@mail.gmail.com>
	<AANLkTin-3pSBKmx+xfks7vL=+=F3aBZ6rWHBvwFK7Y4Z@mail.gmail.com>
Message-ID: <AANLkTinNS8zHfwGAfnzVYk95UW1LbSoHs-JgC7yesZ1e@mail.gmail.com>

Sorry, first of all thanks, for the info. Few months ago i need to know few
suggestion via mailing because my scenario was completely ugly, and involves
lot of non standard billing methods. Therefore, with small amount of
suggestion to my email, i came to my own customized solution now, which
somehow does not work.

I know there is our greatest mod_nibblebill, but in my case i cant use it,
its not working for me, as i mentioned its a ugly and hated project that i
have to ever submit.

- Lua i cant in my small brain, its not getting fit. Again another new
language...
- Instead of Lua, 6 months i was writing C (basic part) + Javascript
(largest part) + Java (basic part) + PHP (elephant part), now after 6 month
its a giant code base with all the cocktail.

I cant simply change or move in to Lua as of this moment.

With javascript it would have taken 30 minute to build a prepaid billing
system. Alas i dont still understand why our greatest
FreeSwitch-> Mr. Spider Monkey, is not allowing me the simple ex:
setTimeout(function() { } , 1000); function....


>From my point of view its a nightmare, its really not easy now to change
simply move all those SOAP/REST/PLAN etc etc..

Thanks & Regards




On Sat, Dec 18, 2010 at 11:15 PM, Chris Burns <chris at cloudtel.com> wrote:

> mod_nibblebill does prepaid billing ... if you search the wiki for
> "billing" it is the first result :D
>
> If you wanted to properly implement your own billing, you should consider
> using the event socket and subscribing to events. Trying to create a prepaid
> billing system using lua or javascript is going to be an uphill battle
>
> On Sat, Dec 18, 2010 at 3:48 PM, Shamun toha md <shamun.toha at gmail.com>wrote:
>
>> Hello,
>>
>> FreeSwitch gets freezed when i try to make it for Pre-Paid track.
>>
>> ex:
>> while(session.ready())
>> {
>>    curl(....balance...);
>> }
>>
>> I tried many possibilities but still i cant make this setTimeout()
>> function working. How can i track when call is connected, seperate option to
>> allow me tracking that call.
>>
>> Is it really not possible ? is it better to have other switch for this,
>> because FreeSwitch cant allow developers to put there own javascript
>> function while call is connected ? Should we use for this other switch ? ex: OpenSIPS
>> or Kamilio.
>>
>> Please kindly guide, project is taking way too long only for this feature,
>> dont found any wiki guide about this info.
>>
>> Thanks & Regards
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris at cloudtel.com  Sun Dec 19 01:56:17 2010
From: chris at cloudtel.com (Chris Burns)
Date: Sat, 18 Dec 2010 17:56:17 -0500
Subject: [Freeswitch-users] audio quality issue
In-Reply-To: <COL122-W35C48A904B9ED67066F154B5170@phx.gbl>
References: <COL122-W33885B610991D11E824AB6B5160@phx.gbl>
	<COL122-W5833A44AAF7FDFD440BCBB5160@phx.gbl>
	<AANLkTinEhrDUHJCfr7RC=9Qm-uYhpFAsx-RJcc7bD4FC@mail.gmail.com>
	<COL122-W5661F836F7A1ED2E6D4FB6B5160@phx.gbl>
	<AANLkTindj9jGDMYn9tLuuN1c3uSX=vC19s_F7pCmBHvC@mail.gmail.com>
	<COL122-W35C48A904B9ED67066F154B5170@phx.gbl>
Message-ID: <AANLkTinfJDCjnY1CxAFAqHC7p5TQf4bKu1HYS7caa7W3@mail.gmail.com>

By "improve" your kernel I just meant resolve the timer frequency issue or
even better stop using xen altogether until you figure out why you have
audio issues.

I assume you mean the voicemail messages sound good when you listen to them
as an attachment in your email, and that your audio quality is always bad in
real time from the server. Spitting an audio file into a codec bit by bit in
real time requires decent precision to pull off ... recording each frame of
audio you recieve into a file does not require precision timing. That is why
the switch could sound poor and create perfect recordings ... although it is
only a guess.

Are you testing with the phone and switch on the same LAN? Do you have
physical access to the switch system and is it a regular old unvirtualized
OS now? There seems to be either an issue with your FS system, or an issue
with the network conditions between you and it. Sadly, I have not
encountered enough RTP quality issues to be good at troubleshooting them or
I might have great advice at this point. I just happen to be bored and
feeling holiday helpful :)


On Sat, Dec 18, 2010 at 9:11 AM, Rafqat . <rafonline at hotmail.com> wrote:

>
> I upgraded to dev snapshot (still on Xen) - still had audio issues.
>
> I got rid of xen server and installed centos 5.5 and freeswitch only -
> timer frequency warning disappeared - still felt audio was not what it
> should be.
>
> What I still dont understand is why when i playback my recorded voicemail
> messages the quality is good.
>
> Sometimes I even have issues leaving a message after the voicemail tone,
> the IVR seems to think i have ended recording the message before even
> starting.  The voicemail tone itself can be very high pitch at times.
>
>
> Any help will be much appreciated.
>
> Chris - can you please point me to some docs on how to improve my kernel
> (sorry but i am newbie to unix).
>
> Cheers
>
> Raf
>
>
> ------------------------------
> Date: Fri, 17 Dec 2010 18:05:09 -0500
> From: chris at cloudtel.com
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] audio quality issue
>
> Improve the kernel and in your situation you will most likely improve your
> audio. You can't expect perfect audio when the thread handling RTP is waking
> up later than its designated time (after sleeping for the codec timing). The
> switch tests for the possibility of this issue on startup when it calibrates
> your clock offset ... thus it warned you :)
>
> "-nocal" will simply skip the process of recognizing and warning that your
> kernel timer is slow, but will not alter performance in your situation
> (unless I am missing something)
>
> On Fri, Dec 17, 2010 at 4:30 PM, Rafqat . <rafonline at hotmail.com> wrote:
>
>
> I will try the dev snapshot.
>
> cheers
>
> Raf
>
> > Date: Fri, 17 Dec 2010 15:28:49 -0600
> > From: anthony.minessale at gmail.com
>
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] audio quality issue
>
> >
> > if you insist to remain on 1.0.6 try it with -vm -nocal cmd line options
> > as i already stated you should try the development snapshot
> >
> > On Fri, Dec 17, 2010 at 3:22 PM, Rafqat . <rafonline at hotmail.com> wrote:
> > >
> > > btw.  I have no audio issues when making calls between phones through
> > > freeswitch (voice quality is very good).
> > >
> > >
> > >
> > > ________________________________
> > > From: rafonline at hotmail.com
> > > To: freeswitch-users at lists.freeswitch.org
> > > Date: Fri, 17 Dec 2010 21:19:11 +0000
> > > Subject: [Freeswitch-users] audio quality issue
> > >
> > >
> > >
> > > Hi,
> > >
> > > I recently installed freeswitch 1.0.6 on centos 5.5 (hosted on xen
> server).
> > > I feel as if the audio quality is not as good as what it should be (6-7
> out
> > > of 10).  This is apparent in calling the default voicemail IVR,
> sometimes I
> > > have issues leaving a message, sometimes I hear some weird noises.
> However,
> > > I have no issues with the quality of the recorded voicemail message I
> leave,
> > > it sounds fine on playback.
> > >
> > > Is this an issue with my phones (Flexor 500)?  I have a softphone which
> > > doesn't seem as bad but still has issues.
> > >
> > > I also get the following message on startup of freeswitch:
> > >
> > > 2010-12-17 21:17:39.138958 [WARNING] switch_time.c:206 Timer resolution
> of
> > > 4000 microseconds detected!
> > > Do you have your kernel timer frequency set to lower than 1,000Hz? You
> may
> > > experience audio problems.
> > >
> > > I assume its related.
> > >
> > > Any help will be much appreciated.
> > >
> > > Cheers
> > >
> > > Raf
> > >
> > > _______________________________________________ FreeSWITCH-users
> mailing
> > > list FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > googletalk:conf+888 at conference.freeswitch.org<googletalk:conf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris at cloudtel.com  Sun Dec 19 03:49:26 2010
From: chris at cloudtel.com (Chris Burns)
Date: Sat, 18 Dec 2010 19:49:26 -0500
Subject: [Freeswitch-users] FreeSwitch cant handle Pre-Paid ?
In-Reply-To: <AANLkTinNS8zHfwGAfnzVYk95UW1LbSoHs-JgC7yesZ1e@mail.gmail.com>
References: <AANLkTi=2bJiYVsTVD6Q+mtG7_RcRqRGLWe9gbPSThu+q@mail.gmail.com>
	<AANLkTin-3pSBKmx+xfks7vL=+=F3aBZ6rWHBvwFK7Y4Z@mail.gmail.com>
	<AANLkTinNS8zHfwGAfnzVYk95UW1LbSoHs-JgC7yesZ1e@mail.gmail.com>
Message-ID: <AANLkTikR9nRxx6rnKcp9CtxO2AHF2QT7yjO6QyrcQU-K@mail.gmail.com>

If I was forced at gunpoint to use javascript to implement pre-paid billing,
it would start something like the example below

<extension name="testJSprepay">
  <condition field="destination_number" expression="^9001$">
    <action application="answer"/>
    <action application="export" data="nolocal:api_on_answer=jsrun test.js
${uuid}"/>
    <action application="bridge"
data="sofia/gateway/mySBC/myCellPhoneNumber"/>
  </condition>
</extension>


Contents of test.js:

uuid_a = argv[0];
uuid_b = apiExecute("uuid_getvar", uuid_a + "  bridge_uuid");
console_log("warning", "uuid_a=" + uuid_a + "\n");
console_log("warning", "uuid_b=" + uuid_b + "\n");

for (;;) {
  msleep(1000);
  exists_a = apiExecute("uuid_exists", uuid_a);
  exists_b = apiExecute("uuid_exists", uuid_b);

  if (exists_a == "true" && exists_b == "true") {
    console_log("warning", "call is still connected\n");
  } else {
    console_log("warning", "call has ended\n");
    break;
  }
}

On Sat, Dec 18, 2010 at 5:54 PM, Shamun toha md <shamun.toha at gmail.com>wrote:

> Sorry, first of all thanks, for the info. Few months ago i need to know few
> suggestion via mailing because my scenario was completely ugly, and involves
> lot of non standard billing methods. Therefore, with small amount of
> suggestion to my email, i came to my own customized solution now, which
> somehow does not work.
>
> I know there is our greatest mod_nibblebill, but in my case i cant use it,
> its not working for me, as i mentioned its a ugly and hated project that i
> have to ever submit.
>
> - Lua i cant in my small brain, its not getting fit. Again another new
> language...
> - Instead of Lua, 6 months i was writing C (basic part) + Javascript
> (largest part) + Java (basic part) + PHP (elephant part), now after 6 month
> its a giant code base with all the cocktail.
>
> I cant simply change or move in to Lua as of this moment.
>
> With javascript it would have taken 30 minute to build a prepaid billing
> system. Alas i dont still understand why our greatest
> FreeSwitch-> Mr. Spider Monkey, is not allowing me the simple ex:
> setTimeout(function() { } , 1000); function....
>
>
> From my point of view its a nightmare, its really not easy now to change
> simply move all those SOAP/REST/PLAN etc etc..
>
> Thanks & Regards
>
>
>
>
> On Sat, Dec 18, 2010 at 11:15 PM, Chris Burns <chris at cloudtel.com> wrote:
>
>> mod_nibblebill does prepaid billing ... if you search the wiki for
>> "billing" it is the first result :D
>>
>> If you wanted to properly implement your own billing, you should consider
>> using the event socket and subscribing to events. Trying to create a prepaid
>> billing system using lua or javascript is going to be an uphill battle
>>
>> On Sat, Dec 18, 2010 at 3:48 PM, Shamun toha md <shamun.toha at gmail.com>wrote:
>>
>>> Hello,
>>>
>>> FreeSwitch gets freezed when i try to make it for Pre-Paid track.
>>>
>>> ex:
>>> while(session.ready())
>>> {
>>>    curl(....balance...);
>>> }
>>>
>>> I tried many possibilities but still i cant make this setTimeout()
>>> function working. How can i track when call is connected, seperate option to
>>> allow me tracking that call.
>>>
>>> Is it really not possible ? is it better to have other switch for this,
>>> because FreeSwitch cant allow developers to put there own javascript
>>> function while call is connected ? Should we use for this other switch ? ex: OpenSIPS
>>> or Kamilio.
>>>
>>> Please kindly guide, project is taking way too long only for this
>>> feature, dont found any wiki guide about this info.
>>>
>>> Thanks & Regards
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Sun Dec 19 04:56:57 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 18 Dec 2010 19:56:57 -0600
Subject: [Freeswitch-users] SIP Registration DNS Error
In-Reply-To: <AANLkTimv_dDh5ynwuOD89XMqatw929YMC=PNL1gOM2Gf@mail.gmail.com>
References: <06502C073AD9394AADB3CA7FD94931BC0519DF46@okc1x1.Logixcom.com>
	<AANLkTi=10HbngBxzwzf=i5PpKMhG5tdzWvtcCdA_weJt@mail.gmail.com>
	<49E1E293-BE37-43AA-960C-56B921FEBC9D@ipeva.fr>
	<AANLkTins8Z6c=dHp=fDta8u9WEmdahz+vqeD7vfvuxOa@mail.gmail.com>
	<AANLkTimv_dDh5ynwuOD89XMqatw929YMC=PNL1gOM2Gf@mail.gmail.com>
Message-ID: <D1456321-8983-47F8-A662-8F6183C26E65@freeswitch.org>

sofia loglevel all 9
watch it and paste that in pastebin... send us the link.

/b

On Dec 18, 2010, at 2:14 PM, Jan Kubr wrote:

> OK they don't want me to register on the IP anymore.
> 
> What does this "DNS Error [503]" error mean exactly? When I nslookup the domain on the same server, it gives me the correct answer. Right now it doesn't even help when I restart the profile or whole FreeSWITCH, I keep getting this DNS Error.
> 
> Thanks.
> 
> Jan



From chad at apartmentlines.com  Sun Dec 19 05:42:37 2010
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Sat, 18 Dec 2010 18:42:37 -0800
Subject: [Freeswitch-users] Jester pre-alpha released
Message-ID: <88A5BB12-CEE5-4041-8E6C-30AC37B71568@apartmentlines.com>

Jester, the new Lua scripting toolkit for FreeSWITCH, is now officially released in the pre-alpha phase.  Code is available in the freeswitch-contrib repository under hunmonk/jester.

To learn more, you can start here: http://wiki.freeswitch.org/wiki/Jester



From u2nsam at gmail.com  Sun Dec 19 10:40:41 2010
From: u2nsam at gmail.com (Sam)
Date: Sun, 19 Dec 2010 13:10:41 +0530
Subject: [Freeswitch-users] audio quality issue
In-Reply-To: <COL122-W5833A44AAF7FDFD440BCBB5160@phx.gbl>
References: <COL122-W33885B610991D11E824AB6B5160@phx.gbl>
	<COL122-W5833A44AAF7FDFD440BCBB5160@phx.gbl>
Message-ID: <AANLkTi=hyQP2g9Wam-47kTw5BED0yeurmW0O0Au52Ji_@mail.gmail.com>

Hi ,

Do you have issues only with hearing the IVR originating from the switch ?
Phone to phone works fine when RTP pass through is enabled ? which codec
gets negotiated when IVR plays and what are the IVR file properties which is
negotiated?
Your voice mail gets recorded properly means the RTP is passing well to the
switch ...

Regads
Sam


On Sat, Dec 18, 2010 at 2:52 AM, Rafqat . <rafonline at hotmail.com> wrote:

>
> btw.  I have no audio issues when making calls between phones through
> freeswitch (voice quality is very good).
>
>
>
> ------------------------------
> From: rafonline at hotmail.com
> To: freeswitch-users at lists.freeswitch.org
> Date: Fri, 17 Dec 2010 21:19:11 +0000
> Subject: [Freeswitch-users] audio quality issue
>
>
>
>
> Hi,
>
> I recently installed freeswitch 1.0.6 on centos 5.5 (hosted on xen
> server).  I feel as if the audio quality is not as good as what it should be
> (6-7 out of 10).  This is apparent in calling the default voicemail IVR,
> sometimes I have issues leaving a message, sometimes I hear some weird
> noises.  However, I have no issues with the quality of the recorded
> voicemail message I leave, it sounds fine on playback.
>
> Is this an issue with my phones (Flexor 500)?  I have a softphone which
> doesn't seem as bad but still has issues.
>
> I also get the following message on startup of freeswitch:
>
> 2010-12-17 21:17:39.138958 [WARNING] switch_time.c:206 Timer resolution of
> 4000 microseconds detected!
> Do you have your kernel timer frequency set to lower than 1,000Hz? You may
> experience audio problems.
>
> I assume its related.
>
> Any help will be much appreciated.
>
> Cheers
>
> Raf
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From noel at voipscience.com  Sat Dec 18 21:49:04 2010
From: noel at voipscience.com (Noel Morgan)
Date: Sat, 18 Dec 2010 12:49:04 -0600
Subject: [Freeswitch-users] Linux IDE for debugging FS?
In-Reply-To: <AANLkTimSXK8T=KQL8xA7RwWzVio__ik+CxAqOLS_N_yR@mail.gmail.com>
Message-ID: <C9325DE1.B82C%noel@voipscience.com>

I like Anjuta

From:  Tom C <mel0torme at gmail.com>
Reply-To:  FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date:  Sat, 18 Dec 2010 10:34:50 -0800
To:  FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] Linux IDE for debugging FS?

I mean debugging the C source and being able to set breakpoints and step
through the code, checking the contents of memory at each step.  And being
able to connect to an instance of FreeSwitch running on another machine and
step through the code.

On Sat, Dec 18, 2010 at 10:26 AM, David Ponzone <david.ponzone at ipeva.fr>
wrote:
> You mean for writing and debugging XML ?
> Personally, I use vi.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non
> autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a
> ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 18/12/2010 ? 19:21, Tom C a ?crit :
> 
>> What "Integrated Development Environment" do you use for developing and
>> debugging FreeSwitch on Linux?
>> 
>> I'd like to install an IDE on my Debian Lenny box.  I tried installing
>> Eclipse, but it said that my machine (P4) was not supported.  And being a
>> Linux newbie, I have no idea what is out there that's good.
>> 
>> Ultimately I want to be able to connect to my DockStar and debug the instance
>> of FS running there.
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 

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

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From simenov at me.com  Sun Dec 19 04:59:12 2010
From: simenov at me.com (=?utf-8?Q?Simen_=C3=98vreb=C3=B8?=)
Date: Sun, 19 Dec 2010 02:59:12 +0100
Subject: [Freeswitch-users] No contact header and F.729AB
Message-ID: <3E7186CF-3BE3-4E3B-ADE2-3C758703068F@me.com>

Hi,

I'm having these two Fritz!Boxes that keep annoying me wit their missing contact headers. Any workarounds?

I also wonder if G.729AB for Mac OS X is planned in the near future? At least ten licenses planned in the start phase. Cellphone remote workers and G.711a not very stable.


Regards,

Simen ?vreb?

From mail at jankubr.com  Sun Dec 19 14:30:53 2010
From: mail at jankubr.com (Jan Kubr)
Date: Sun, 19 Dec 2010 12:30:53 +0100
Subject: [Freeswitch-users] SIP Registration DNS Error
In-Reply-To: <D1456321-8983-47F8-A662-8F6183C26E65@freeswitch.org>
References: <06502C073AD9394AADB3CA7FD94931BC0519DF46@okc1x1.Logixcom.com>
	<AANLkTi=10HbngBxzwzf=i5PpKMhG5tdzWvtcCdA_weJt@mail.gmail.com>
	<49E1E293-BE37-43AA-960C-56B921FEBC9D@ipeva.fr>
	<AANLkTins8Z6c=dHp=fDta8u9WEmdahz+vqeD7vfvuxOa@mail.gmail.com>
	<AANLkTimv_dDh5ynwuOD89XMqatw929YMC=PNL1gOM2Gf@mail.gmail.com>
	<D1456321-8983-47F8-A662-8F6183C26E65@freeswitch.org>
Message-ID: <AANLkTimqApc_gXv6+0LywmqPOn4Z5aPMXZeHvy-eiyP5@mail.gmail.com>

Here it is: http://pastebin.freeswitch.org/14824

On Sun, Dec 19, 2010 at 2:56 AM, Brian West <brian at freeswitch.org> wrote:

> sofia loglevel all 9
> watch it and paste that in pastebin... send us the link.
>
> /b
>
> On Dec 18, 2010, at 2:14 PM, Jan Kubr wrote:
>
> > OK they don't want me to register on the IP anymore.
> >
> > What does this "DNS Error [503]" error mean exactly? When I nslookup the
> domain on the same server, it gives me the correct answer. Right now it
> doesn't even help when I restart the profile or whole FreeSWITCH, I keep
> getting this DNS Error.
> >
> > Thanks.
> >
> > Jan
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mail at jankubr.com  Sun Dec 19 15:05:53 2010
From: mail at jankubr.com (Jan Kubr)
Date: Sun, 19 Dec 2010 13:05:53 +0100
Subject: [Freeswitch-users] SIP Registration DNS Error
In-Reply-To: <AANLkTimqApc_gXv6+0LywmqPOn4Z5aPMXZeHvy-eiyP5@mail.gmail.com>
References: <06502C073AD9394AADB3CA7FD94931BC0519DF46@okc1x1.Logixcom.com>
	<AANLkTi=10HbngBxzwzf=i5PpKMhG5tdzWvtcCdA_weJt@mail.gmail.com>
	<49E1E293-BE37-43AA-960C-56B921FEBC9D@ipeva.fr>
	<AANLkTins8Z6c=dHp=fDta8u9WEmdahz+vqeD7vfvuxOa@mail.gmail.com>
	<AANLkTimv_dDh5ynwuOD89XMqatw929YMC=PNL1gOM2Gf@mail.gmail.com>
	<D1456321-8983-47F8-A662-8F6183C26E65@freeswitch.org>
	<AANLkTimqApc_gXv6+0LywmqPOn4Z5aPMXZeHvy-eiyP5@mail.gmail.com>
Message-ID: <AANLkTimw2t5P_ELc3PMGf+HB25q-_fW-NtYMPQeJsL1z@mail.gmail.com>

So it's their incorrect NAPTR record? I'm only learning about this
mechanism. I'll be interested to see what the NAPTR record is when the
registration stops failing again.

On Sun, Dec 19, 2010 at 12:30 PM, Jan Kubr <mail at jankubr.com> wrote:

> Here it is: http://pastebin.freeswitch.org/14824
>
> On Sun, Dec 19, 2010 at 2:56 AM, Brian West <brian at freeswitch.org> wrote:
>
>> sofia loglevel all 9
>> watch it and paste that in pastebin... send us the link.
>>
>> /b
>>
>> On Dec 18, 2010, at 2:14 PM, Jan Kubr wrote:
>>
>> > OK they don't want me to register on the IP anymore.
>> >
>> > What does this "DNS Error [503]" error mean exactly? When I nslookup the
>> domain on the same server, it gives me the correct answer. Right now it
>> doesn't even help when I restart the profile or whole FreeSWITCH, I keep
>> getting this DNS Error.
>> >
>> > Thanks.
>> >
>> > Jan
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From covici at ccs.covici.com  Sun Dec 19 16:28:30 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sun, 19 Dec 2010 08:28:30 -0500
Subject: [Freeswitch-users] rebinding meta_app digit does not work on b-leg
Message-ID: <10356.1292765310@ccs.covici.com>

Hi.  I have some dialplan code which allows me to press *2 to record and
*2 to end the recording.  The way I do this is to rebind the digit by
executing an extension.  Now if I call, it works fine, but if soneone
calls me, it does not work -- looking at the logs I found out that even
though it says it rebound the digit 2 on the b-leg, it actually still
executes the previous bind of meta-app.

Here is the pastebin to show the log:
http://pastebin.freeswitch.org/14825

Thanks in advance for any help.
-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From lloyd.aloysius at gmail.com  Sun Dec 19 20:04:48 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 19 Dec 2010 12:04:48 -0500
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
	Permission denied
Message-ID: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>

Hi All,

I follow the instructions in the install.txt

cd /usr/tmp
wget http://files.freeswitch.org/g729/fsg729-167-installer
chmod 755 fsg729-167-installer
./fsg729-167-installer

I get the following Error. I am using CentOS.

FreeSWITCH G729 Self Extracting Installer

./fsg729-167-installer: ./installer: /bin/bash: bad interpreter: Permission
denied


Any help is appreciated.

Thanks
Lloyd
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From tgraziano at myitdepartment.net  Sun Dec 19 20:19:04 2010
From: tgraziano at myitdepartment.net (Tony Graziano)
Date: Sun, 19 Dec 2010 12:19:04 -0500
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
 Permission denied
In-Reply-To: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
Message-ID: <AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>

you didnt make it executable

chmod +x fsg729-167-installer

On Sun, Dec 19, 2010 at 12:04 PM, Aloysius Lloyd
<lloyd.aloysius at gmail.com> wrote:
> Hi All,
> I follow the instructions in the install.txt
> cd /usr/tmp
> wget http://files.freeswitch.org/g729/fsg729-167-installer
> chmod 755 fsg729-167-installer
> ./fsg729-167-installer
> I get the following Error. I am using CentOS.
>
> FreeSWITCH G729 Self Extracting Installer
>
> ./fsg729-167-installer: ./installer: /bin/bash: bad interpreter: Permission
> denied
>
> Any help is appreciated.
> Thanks
> Lloyd
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgraziano at voice.myitdepartment.net
Fax: 434.326.5325

Email: tgraziano at myitdepartment.net

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpdesk at voice.myitdepartment.net

Helpdesk Contract Customers:
http://support.myitdepartment.net
Blog:
http://blog.myitdepartment.net

Linked-In Profile:?http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4


From lloyd.aloysius at sunteltech.ca  Sun Dec 19 20:21:27 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Sun, 19 Dec 2010 12:21:27 -0500
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
 Permission denied
In-Reply-To: <AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
	<AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
Message-ID: <AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com>

same results

Thanks
Lloyd


On Sun, Dec 19, 2010 at 12:19 PM, Tony Graziano <
tgraziano at myitdepartment.net> wrote:

> you didnt make it executable
>
> chmod +x fsg729-167-installer
>
> On Sun, Dec 19, 2010 at 12:04 PM, Aloysius Lloyd
> <lloyd.aloysius at gmail.com> wrote:
> > Hi All,
> > I follow the instructions in the install.txt
> > cd /usr/tmp
> > wget http://files.freeswitch.org/g729/fsg729-167-installer
> > chmod 755 fsg729-167-installer
> > ./fsg729-167-installer
> > I get the following Error. I am using CentOS.
> >
> > FreeSWITCH G729 Self Extracting Installer
> >
> > ./fsg729-167-installer: ./installer: /bin/bash: bad interpreter:
> Permission
> > denied
> >
> > Any help is appreciated.
> > Thanks
> > Lloyd
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> ======================
> Tony Graziano, Manager
> Telephone: 434.984.8430
> sip: tgraziano at voice.myitdepartment.net
> Fax: 434.326.5325
>
> Email: tgraziano at myitdepartment.net
>
> LAN/Telephony/Security and Control Systems Helpdesk:
> Telephone: 434.984.8426
> sip: helpdesk at voice.myitdepartment.net
>
> Helpdesk Contract Customers:
> http://support.myitdepartment.net
> Blog:
> http://blog.myitdepartment.net
>
> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.ponzone at ipeva.fr  Sun Dec 19 20:23:01 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 19 Dec 2010 18:23:01 +0100
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
	Permission denied
In-Reply-To: <AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
	<AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
Message-ID: <7FEDC10E-B47B-4D92-90DC-94FA4F7A6710@ipeva.fr>

Tony,

Last time I checked, chmod 755 was setting the x bit for everyone :)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 19/12/2010 ? 18:19, Tony Graziano a ?crit :

> you didnt make it executable
> 
> chmod +x fsg729-167-installer
> 
> On Sun, Dec 19, 2010 at 12:04 PM, Aloysius Lloyd
> <lloyd.aloysius at gmail.com> wrote:
>> Hi All,
>> I follow the instructions in the install.txt
>> cd /usr/tmp
>> wget http://files.freeswitch.org/g729/fsg729-167-installer
>> chmod 755 fsg729-167-installer
>> ./fsg729-167-installer
>> I get the following Error. I am using CentOS.
>> 
>> FreeSWITCH G729 Self Extracting Installer
>> 
>> ./fsg729-167-installer: ./installer: /bin/bash: bad interpreter: Permission
>> denied
>> 
>> Any help is appreciated.
>> Thanks
>> Lloyd
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 
> 
> -- 
> ======================
> Tony Graziano, Manager
> Telephone: 434.984.8430
> sip: tgraziano at voice.myitdepartment.net
> Fax: 434.326.5325
> 
> Email: tgraziano at myitdepartment.net
> 
> LAN/Telephony/Security and Control Systems Helpdesk:
> Telephone: 434.984.8426
> sip: helpdesk at voice.myitdepartment.net
> 
> Helpdesk Contract Customers:
> http://support.myitdepartment.net
> Blog:
> http://blog.myitdepartment.net
> 
> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Sun Dec 19 20:24:19 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 19 Dec 2010 11:24:19 -0600
Subject: [Freeswitch-users] SIP Registration DNS Error
In-Reply-To: <AANLkTimw2t5P_ELc3PMGf+HB25q-_fW-NtYMPQeJsL1z@mail.gmail.com>
References: <06502C073AD9394AADB3CA7FD94931BC0519DF46@okc1x1.Logixcom.com>
	<AANLkTi=10HbngBxzwzf=i5PpKMhG5tdzWvtcCdA_weJt@mail.gmail.com>
	<49E1E293-BE37-43AA-960C-56B921FEBC9D@ipeva.fr>
	<AANLkTins8Z6c=dHp=fDta8u9WEmdahz+vqeD7vfvuxOa@mail.gmail.com>
	<AANLkTimv_dDh5ynwuOD89XMqatw929YMC=PNL1gOM2Gf@mail.gmail.com>
	<D1456321-8983-47F8-A662-8F6183C26E65@freeswitch.org>
	<AANLkTimqApc_gXv6+0LywmqPOn4Z5aPMXZeHvy-eiyP5@mail.gmail.com>
	<AANLkTimw2t5P_ELc3PMGf+HB25q-_fW-NtYMPQeJsL1z@mail.gmail.com>
Message-ID: <A27AF231-08B8-44D2-85A7-2A6353C73AD5@freeswitch.org>

Thats what it looks like sofia follows the rules to the letter of the law.  they need to fix their NAPTR records.

/b

On Dec 19, 2010, at 6:05 AM, Jan Kubr wrote:

> So it's their incorrect NAPTR record? I'm only learning about this mechanism. I'll be interested to see what the NAPTR record is when the registration stops failing again.
> 
> On Sun, Dec 19, 2010 at 12:30 PM, Jan Kubr <mail at jankubr.com> wrote:
> Here it is: http://pastebin.freeswitch.org/14824
> 
> On Sun, Dec 19, 2010 at 2:56 AM, Brian West <brian at freeswitch.org> wrote:
> sofia loglevel all 9
> watch it and paste that in pastebin... send us the link.
> 
> /b

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From brian at freeswitch.org  Sun Dec 19 20:25:43 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 19 Dec 2010 11:25:43 -0600
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
	Permission denied
In-Reply-To: <AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com>
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
	<AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
	<AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com>
Message-ID: <FD03B9A4-EF89-4D88-B353-6CBAFBB6EF83@freeswitch.org>

You don't have /bin/bash installed do you?

/b

On Dec 19, 2010, at 11:21 AM, Aloysius Lloyd wrote:

> same results
> 
> Thanks
> Lloyd

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From brian at freeswitch.org  Sun Dec 19 20:26:59 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 19 Dec 2010 11:26:59 -0600
Subject: [Freeswitch-users] No contact header and F.729AB
In-Reply-To: <3E7186CF-3BE3-4E3B-ADE2-3C758703068F@me.com>
References: <3E7186CF-3BE3-4E3B-ADE2-3C758703068F@me.com>
Message-ID: <49F685A5-9645-4DB0-93EA-24D7E6566D69@freeswitch.org>

Well if your device is sending broken packets its a toss up here... what does the sip trace look like?

/b

On Dec 18, 2010, at 7:59 PM, Simen ?vreb? wrote:

> Hi,
> 
> I'm having these two Fritz!Boxes that keep annoying me wit their missing contact headers. Any workarounds?
> 
> I also wonder if G.729AB for Mac OS X is planned in the near future? At least ten licenses planned in the start phase. Cellphone remote workers and G.711a not very stable.
> 
> 
> Regards,
> 
> Simen ?vreb?



From lloyd.aloysius at sunteltech.ca  Sun Dec 19 20:29:50 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Sun, 19 Dec 2010 12:29:50 -0500
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
 Permission denied
In-Reply-To: <FD03B9A4-EF89-4D88-B353-6CBAFBB6EF83@freeswitch.org>
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
	<AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
	<AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com>
	<FD03B9A4-EF89-4D88-B353-6CBAFBB6EF83@freeswitch.org>
Message-ID: <AANLkTikfHjX9a9ue+9-JbXbY-3TNZ1bEZVjxhCe8Ty7-@mail.gmail.com>

OS : CentOS release 5.5 (Final)

Default CentOS Installation. /bin/bash is installed.

 ls -al /bin/bash
-rwxr-xr-x 1 root root 735004 Jan 21  2009 /bin/bash

Thanks
Lloyd


On Sun, Dec 19, 2010 at 12:25 PM, Brian West <brian at freeswitch.org> wrote:

> You don't have /bin/bash installed do you?
>
> /b
>
> On Dec 19, 2010, at 11:21 AM, Aloysius Lloyd wrote:
>
> same results
>
> Thanks
> Lloyd
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Sun Dec 19 20:34:59 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 19 Dec 2010 18:34:59 +0100
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
	Permission denied
In-Reply-To: <AANLkTikfHjX9a9ue+9-JbXbY-3TNZ1bEZVjxhCe8Ty7-@mail.gmail.com>
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
	<AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
	<AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com>
	<FD03B9A4-EF89-4D88-B353-6CBAFBB6EF83@freeswitch.org>
	<AANLkTikfHjX9a9ue+9-JbXbY-3TNZ1bEZVjxhCe8Ty7-@mail.gmail.com>
Message-ID: <2AD4C3DA-5C91-4FE4-AF9E-A9B3D572C21F@ipeva.fr>

Are you trying to run it from a partition mounted with the noexec flag ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 19/12/2010 ? 18:29, Aloysius Lloyd a ?crit :

> OS : CentOS release 5.5 (Final)
> 
> Default CentOS Installation. /bin/bash is installed.
> 
>  ls -al /bin/bash
> -rwxr-xr-x 1 root root 735004 Jan 21  2009 /bin/bash
> 
> Thanks
> Lloyd
> 
> 
> On Sun, Dec 19, 2010 at 12:25 PM, Brian West <brian at freeswitch.org> wrote:
> You don't have /bin/bash installed do you?
> 
> /b
> 
> On Dec 19, 2010, at 11:21 AM, Aloysius Lloyd wrote:
> 
>> same results
>> 
>> Thanks
>> Lloyd
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lloyd.aloysius at sunteltech.ca  Sun Dec 19 20:43:51 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Sun, 19 Dec 2010 12:43:51 -0500
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
 Permission denied
In-Reply-To: <2AD4C3DA-5C91-4FE4-AF9E-A9B3D572C21F@ipeva.fr>
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
	<AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
	<AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com>
	<FD03B9A4-EF89-4D88-B353-6CBAFBB6EF83@freeswitch.org>
	<AANLkTikfHjX9a9ue+9-JbXbY-3TNZ1bEZVjxhCe8Ty7-@mail.gmail.com>
	<2AD4C3DA-5C91-4FE4-AF9E-A9B3D572C21F@ipeva.fr>
Message-ID: <AANLkTi=V+TxN55mEfk-BDN8D8aP=fhgwsXMtZUJ8wKM_@mail.gmail.com>

Hi David,

I Google it for the error , there is suggestion for the fstab modifications.


Here is fstab

 more /etc/fstab

/dev/md2                /                       ext3    defaults        1 1
/dev/md1                /tmp                    ext3
 defaults,nosuid,noexec,nodev        1 2
/dev/md0                /boot                   ext3    defaults        1 2
tmpfs                   /dev/shm                tmpfs   defaults        0 0
devpts                  /dev/pts                devpts  gid=5,mode=620  0 0
sysfs                   /sys                    sysfs   defaults        0 0
proc                    /proc                   proc    defaults        0 0
LABEL=SWAP-sda3         swap                    swap    defaults        0 0
LABEL=SWAP-sdb3         swap                    swap    defaults        0 0


Thanks
Lloyd

2010/12/19 David Ponzone <david.ponzone at ipeva.fr>

> Are you trying to run it from a partition mounted with the noexec flag ?
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 19/12/2010 ? 18:29, Aloysius Lloyd a ?crit :
>
> OS : CentOS release 5.5 (Final)
>
> Default CentOS Installation. /bin/bash is installed.
>
>  ls -al /bin/bash
> -rwxr-xr-x 1 root root 735004 Jan 21  2009 /bin/bash
>
> Thanks
> Lloyd
>
>
> On Sun, Dec 19, 2010 at 12:25 PM, Brian West <brian at freeswitch.org> wrote:
>
>> You don't have /bin/bash installed do you?
>>
>> /b
>>
>> On Dec 19, 2010, at 11:21 AM, Aloysius Lloyd wrote:
>>
>> same results
>>
>> Thanks
>> Lloyd
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Sun Dec 19 20:51:40 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 19 Dec 2010 18:51:40 +0100
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
	Permission denied
In-Reply-To: <AANLkTi=V+TxN55mEfk-BDN8D8aP=fhgwsXMtZUJ8wKM_@mail.gmail.com>
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
	<AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
	<AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com>
	<FD03B9A4-EF89-4D88-B353-6CBAFBB6EF83@freeswitch.org>
	<AANLkTikfHjX9a9ue+9-JbXbY-3TNZ1bEZVjxhCe8Ty7-@mail.gmail.com>
	<2AD4C3DA-5C91-4FE4-AF9E-A9B3D572C21F@ipeva.fr>
	<AANLkTi=V+TxN55mEfk-BDN8D8aP=fhgwsXMtZUJ8wKM_@mail.gmail.com>
Message-ID: <EF669B87-27CB-4B84-AB1F-44DBE350A259@ipeva.fr>

check if /usr/tmp is a symlink to /tmp
your /tmp is mounted noexec, so that would be it

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 19/12/2010 ? 18:43, Aloysius Lloyd a ?crit :

> Hi David,
> 
> I Google it for the error , there is suggestion for the fstab modifications.
> 
> 
> Here is fstab
> 
>  more /etc/fstab
> 
> /dev/md2                /                       ext3    defaults        1 1
> /dev/md1                /tmp                    ext3    defaults,nosuid,noexec,nodev        1 2
> /dev/md0                /boot                   ext3    defaults        1 2
> tmpfs                   /dev/shm                tmpfs   defaults        0 0
> devpts                  /dev/pts                devpts  gid=5,mode=620  0 0
> sysfs                   /sys                    sysfs   defaults        0 0
> proc                    /proc                   proc    defaults        0 0
> LABEL=SWAP-sda3         swap                    swap    defaults        0 0
> LABEL=SWAP-sdb3         swap                    swap    defaults        0 0
> 
> 
> Thanks
> Lloyd
> 
> 2010/12/19 David Ponzone <david.ponzone at ipeva.fr>
> Are you trying to run it from a partition mounted with the noexec flag ?
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 19/12/2010 ? 18:29, Aloysius Lloyd a ?crit :
> 
>> OS : CentOS release 5.5 (Final)
>> 
>> Default CentOS Installation. /bin/bash is installed.
>> 
>>  ls -al /bin/bash
>> -rwxr-xr-x 1 root root 735004 Jan 21  2009 /bin/bash
>> 
>> Thanks
>> Lloyd
>> 
>> 
>> On Sun, Dec 19, 2010 at 12:25 PM, Brian West <brian at freeswitch.org> wrote:
>> You don't have /bin/bash installed do you?
>> 
>> /b
>> 
>> On Dec 19, 2010, at 11:21 AM, Aloysius Lloyd wrote:
>> 
>>> same results
>>> 
>>> Thanks
>>> Lloyd
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lloyd.aloysius at sunteltech.ca  Sun Dec 19 20:56:51 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Sun, 19 Dec 2010 12:56:51 -0500
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
 Permission denied
In-Reply-To: <EF669B87-27CB-4B84-AB1F-44DBE350A259@ipeva.fr>
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
	<AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
	<AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com>
	<FD03B9A4-EF89-4D88-B353-6CBAFBB6EF83@freeswitch.org>
	<AANLkTikfHjX9a9ue+9-JbXbY-3TNZ1bEZVjxhCe8Ty7-@mail.gmail.com>
	<2AD4C3DA-5C91-4FE4-AF9E-A9B3D572C21F@ipeva.fr>
	<AANLkTi=V+TxN55mEfk-BDN8D8aP=fhgwsXMtZUJ8wKM_@mail.gmail.com>
	<EF669B87-27CB-4B84-AB1F-44DBE350A259@ipeva.fr>
Message-ID: <AANLkTik49Xae=8M3PxBahnJMnuZ+i5EqPZBazm427FPE@mail.gmail.com>

David,

I check the /usr/tmp here is the output

 ls -al /usr/tmp
lrwxrwxrwx 1 root root 10 Sep 18 05:24 /usr/tmp -> ../var/tmp

This is a production box CentOS , Apache ,MySql, PHP and FreeSWITCH running
for months.

I do not quite understand why this installer is not working.

Also i try in several other directories ... same results.

I try in the following directories

/usr/src

/root


Thanks
Lloyd




2010/12/19 David Ponzone <david.ponzone at ipeva.fr>

> check if /usr/tmp is a symlink to /tmp
> your /tmp is mounted noexec, so that would be it
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 19/12/2010 ? 18:43, Aloysius Lloyd a ?crit :
>
> Hi David,
>
> I Google it for the error , there is suggestion for the fstab
> modifications.
>
>
> Here is fstab
>
>  more /etc/fstab
>
> /dev/md2                /                       ext3    defaults        1 1
> /dev/md1                /tmp                    ext3
>  defaults,nosuid,noexec,nodev        1 2
> /dev/md0                /boot                   ext3    defaults        1 2
> tmpfs                   /dev/shm                tmpfs   defaults        0 0
> devpts                  /dev/pts                devpts  gid=5,mode=620  0 0
> sysfs                   /sys                    sysfs   defaults        0 0
> proc                    /proc                   proc    defaults        0 0
> LABEL=SWAP-sda3         swap                    swap    defaults        0 0
> LABEL=SWAP-sdb3         swap                    swap    defaults        0 0
>
>
> Thanks
> Lloyd
>
> 2010/12/19 David Ponzone <david.ponzone at ipeva.fr>
>
>> Are you trying to run it from a partition mounted with the noexec flag ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 19/12/2010 ? 18:29, Aloysius Lloyd a ?crit :
>>
>> OS : CentOS release 5.5 (Final)
>>
>> Default CentOS Installation. /bin/bash is installed.
>>
>>  ls -al /bin/bash
>> -rwxr-xr-x 1 root root 735004 Jan 21  2009 /bin/bash
>>
>> Thanks
>> Lloyd
>>
>>
>> On Sun, Dec 19, 2010 at 12:25 PM, Brian West <brian at freeswitch.org>wrote:
>>
>>> You don't have /bin/bash installed do you?
>>>
>>> /b
>>>
>>> On Dec 19, 2010, at 11:21 AM, Aloysius Lloyd wrote:
>>>
>>> same results
>>>
>>> Thanks
>>> Lloyd
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Sun Dec 19 21:05:58 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 19 Dec 2010 19:05:58 +0100
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
	Permission denied
In-Reply-To: <AANLkTik49Xae=8M3PxBahnJMnuZ+i5EqPZBazm427FPE@mail.gmail.com>
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
	<AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
	<AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com>
	<FD03B9A4-EF89-4D88-B353-6CBAFBB6EF83@freeswitch.org>
	<AANLkTikfHjX9a9ue+9-JbXbY-3TNZ1bEZVjxhCe8Ty7-@mail.gmail.com>
	<2AD4C3DA-5C91-4FE4-AF9E-A9B3D572C21F@ipeva.fr>
	<AANLkTi=V+TxN55mEfk-BDN8D8aP=fhgwsXMtZUJ8wKM_@mail.gmail.com>
	<EF669B87-27CB-4B84-AB1F-44DBE350A259@ipeva.fr>
	<AANLkTik49Xae=8M3PxBahnJMnuZ+i5EqPZBazm427FPE@mail.gmail.com>
Message-ID: <52A917AD-90F9-45E8-9FB1-45CB1CF080A1@ipeva.fr>

that's weird indeed, but there is a reason for everything
redownload the installer perhaps

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 19/12/2010 ? 18:56, Aloysius Lloyd a ?crit :

> David,
> 
> I check the /usr/tmp here is the output
> 
>  ls -al /usr/tmp
> lrwxrwxrwx 1 root root 10 Sep 18 05:24 /usr/tmp -> ../var/tmp
> 
> This is a production box CentOS , Apache ,MySql, PHP and FreeSWITCH running for months.
> 
> I do not quite understand why this installer is not working.
> 
> Also i try in several other directories ... same results.
> 
> I try in the following directories
> 
> /usr/src
> 
> /root
> 
> 
> Thanks
> Lloyd
> 
> 
> 
> 
> 2010/12/19 David Ponzone <david.ponzone at ipeva.fr>
> check if /usr/tmp is a symlink to /tmp
> your /tmp is mounted noexec, so that would be it
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 19/12/2010 ? 18:43, Aloysius Lloyd a ?crit :
> 
>> Hi David,
>> 
>> I Google it for the error , there is suggestion for the fstab modifications.
>> 
>> 
>> Here is fstab
>> 
>>  more /etc/fstab
>> 
>> /dev/md2                /                       ext3    defaults        1 1
>> /dev/md1                /tmp                    ext3    defaults,nosuid,noexec,nodev        1 2
>> /dev/md0                /boot                   ext3    defaults        1 2
>> tmpfs                   /dev/shm                tmpfs   defaults        0 0
>> devpts                  /dev/pts                devpts  gid=5,mode=620  0 0
>> sysfs                   /sys                    sysfs   defaults        0 0
>> proc                    /proc                   proc    defaults        0 0
>> LABEL=SWAP-sda3         swap                    swap    defaults        0 0
>> LABEL=SWAP-sdb3         swap                    swap    defaults        0 0
>> 
>> 
>> Thanks
>> Lloyd
>> 
>> 2010/12/19 David Ponzone <david.ponzone at ipeva.fr>
>> Are you trying to run it from a partition mounted with the noexec flag ?
>> 
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> 
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> 
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> 
>> Le 19/12/2010 ? 18:29, Aloysius Lloyd a ?crit :
>> 
>>> OS : CentOS release 5.5 (Final)
>>> 
>>> Default CentOS Installation. /bin/bash is installed.
>>> 
>>>  ls -al /bin/bash
>>> -rwxr-xr-x 1 root root 735004 Jan 21  2009 /bin/bash
>>> 
>>> Thanks
>>> Lloyd
>>> 
>>> 
>>> On Sun, Dec 19, 2010 at 12:25 PM, Brian West <brian at freeswitch.org> wrote:
>>> You don't have /bin/bash installed do you?
>>> 
>>> /b
>>> 
>>> On Dec 19, 2010, at 11:21 AM, Aloysius Lloyd wrote:
>>> 
>>>> same results
>>>> 
>>>> Thanks
>>>> Lloyd
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tgraziano at myitdepartment.net  Sun Dec 19 21:11:53 2010
From: tgraziano at myitdepartment.net (Tony Graziano)
Date: Sun, 19 Dec 2010 13:11:53 -0500
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
 Permission denied
In-Reply-To: <AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com>
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
	<AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
	<AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com>
Message-ID: <AANLkTik9qMKo4GAtDN8e19M=8EnK3N+invC902wdkyW8@mail.gmail.com>

Are you sure you are running the executable with the proper
permissions or as the correct user?

I downloaded and was able to run after I set the file to executable.

What is the OS? What user are you trying to run this as? Did you check
to ensure the file is executable after you did the chmod?



On 12/19/10, Aloysius Lloyd <lloyd.aloysius at sunteltech.ca> wrote:
> same results
>
> Thanks
> Lloyd
>
>
> On Sun, Dec 19, 2010 at 12:19 PM, Tony Graziano <
> tgraziano at myitdepartment.net> wrote:
>
>> you didnt make it executable
>>
>> chmod +x fsg729-167-installer
>>
>> On Sun, Dec 19, 2010 at 12:04 PM, Aloysius Lloyd
>> <lloyd.aloysius at gmail.com> wrote:
>> > Hi All,
>> > I follow the instructions in the install.txt
>> > cd /usr/tmp
>> > wget http://files.freeswitch.org/g729/fsg729-167-installer
>> > chmod 755 fsg729-167-installer
>> > ./fsg729-167-installer
>> > I get the following Error. I am using CentOS.
>> >
>> > FreeSWITCH G729 Self Extracting Installer
>> >
>> > ./fsg729-167-installer: ./installer: /bin/bash: bad interpreter:
>> Permission
>> > denied
>> >
>> > Any help is appreciated.
>> > Thanks
>> > Lloyd
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> ======================
>> Tony Graziano, Manager
>> Telephone: 434.984.8430
>> sip: tgraziano at voice.myitdepartment.net
>> Fax: 434.326.5325
>>
>> Email: tgraziano at myitdepartment.net
>>
>> LAN/Telephony/Security and Control Systems Helpdesk:
>> Telephone: 434.984.8426
>> sip: helpdesk at voice.myitdepartment.net
>>
>> Helpdesk Contract Customers:
>> http://support.myitdepartment.net
>> Blog:
>> http://blog.myitdepartment.net
>>
>> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>

-- 
Sent from my mobile device

======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgraziano at voice.myitdepartment.net
Fax: 434.326.5325

Email: tgraziano at myitdepartment.net

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpdesk at voice.myitdepartment.net

Helpdesk Contract Customers:
http://support.myitdepartment.net

<http://support.myitdepartment.net>Blog:
http://blog.myitdepartment.net

Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4


From lloyd.aloysius at sunteltech.ca  Sun Dec 19 21:16:29 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Sun, 19 Dec 2010 13:16:29 -0500
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
 Permission denied
In-Reply-To: <AANLkTik9qMKo4GAtDN8e19M=8EnK3N+invC902wdkyW8@mail.gmail.com>
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
	<AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
	<AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com>
	<AANLkTik9qMKo4GAtDN8e19M=8EnK3N+invC902wdkyW8@mail.gmail.com>
Message-ID: <AANLkTikAoeGo410KS-KFi+2q-vGS9jbGuJcpYpOhndAU@mail.gmail.com>

Tony,

Here is my configuration ....

OS : CentOS 5.5

User : root

Permission: -rwxr-xr-x  1 root root 1248017 Nov 19 11:47
fsg729-167-installer

Also as I said in my last email this is a FreeSWITCH production BOX .

Thanks
Lloyd


On Sun, Dec 19, 2010 at 1:11 PM, Tony Graziano <tgraziano at myitdepartment.net
> wrote:

> Are you sure you are running the executable with the proper
> permissions or as the correct user?
>
> I downloaded and was able to run after I set the file to executable.
>
> What is the OS? What user are you trying to run this as? Did you check
> to ensure the file is executable after you did the chmod?
>
>
>
> On 12/19/10, Aloysius Lloyd <lloyd.aloysius at sunteltech.ca> wrote:
> > same results
> >
> > Thanks
> > Lloyd
> >
> >
> > On Sun, Dec 19, 2010 at 12:19 PM, Tony Graziano <
> > tgraziano at myitdepartment.net> wrote:
> >
> >> you didnt make it executable
> >>
> >> chmod +x fsg729-167-installer
> >>
> >> On Sun, Dec 19, 2010 at 12:04 PM, Aloysius Lloyd
> >> <lloyd.aloysius at gmail.com> wrote:
> >> > Hi All,
> >> > I follow the instructions in the install.txt
> >> > cd /usr/tmp
> >> > wget http://files.freeswitch.org/g729/fsg729-167-installer
> >> > chmod 755 fsg729-167-installer
> >> > ./fsg729-167-installer
> >> > I get the following Error. I am using CentOS.
> >> >
> >> > FreeSWITCH G729 Self Extracting Installer
> >> >
> >> > ./fsg729-167-installer: ./installer: /bin/bash: bad interpreter:
> >> Permission
> >> > denied
> >> >
> >> > Any help is appreciated.
> >> > Thanks
> >> > Lloyd
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> ======================
> >> Tony Graziano, Manager
> >> Telephone: 434.984.8430
> >> sip: tgraziano at voice.myitdepartment.net
> >> Fax: 434.326.5325
> >>
> >> Email: tgraziano at myitdepartment.net
> >>
> >> LAN/Telephony/Security and Control Systems Helpdesk:
> >> Telephone: 434.984.8426
> >> sip: helpdesk at voice.myitdepartment.net
> >>
> >> Helpdesk Contract Customers:
> >> http://support.myitdepartment.net
> >> Blog:
> >> http://blog.myitdepartment.net
> >>
> >> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
>
> --
> Sent from my mobile device
>
> ======================
> Tony Graziano, Manager
> Telephone: 434.984.8430
> sip: tgraziano at voice.myitdepartment.net
> Fax: 434.326.5325
>
> Email: tgraziano at myitdepartment.net
>
> LAN/Telephony/Security and Control Systems Helpdesk:
> Telephone: 434.984.8426
> sip: helpdesk at voice.myitdepartment.net
>
> Helpdesk Contract Customers:
> http://support.myitdepartment.net
>
> <http://support.myitdepartment.net>Blog:
> http://blog.myitdepartment.net
>
> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tgraziano at myitdepartment.net  Sun Dec 19 21:37:11 2010
From: tgraziano at myitdepartment.net (Tony Graziano)
Date: Sun, 19 Dec 2010 13:37:11 -0500
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
 Permission denied
In-Reply-To: <AANLkTikAoeGo410KS-KFi+2q-vGS9jbGuJcpYpOhndAU@mail.gmail.com>
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
	<AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
	<AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com>
	<AANLkTik9qMKo4GAtDN8e19M=8EnK3N+invC902wdkyW8@mail.gmail.com>
	<AANLkTikAoeGo410KS-KFi+2q-vGS9jbGuJcpYpOhndAU@mail.gmail.com>
Message-ID: <AANLkTin3ccd_JD9e-GEpGPjhZ0F9ca-E16TZ=_hwk4z0@mail.gmail.com>

On Sun, Dec 19, 2010 at 1:16 PM, Aloysius Lloyd
<lloyd.aloysius at sunteltech.ca> wrote:
> Tony,
> Here is my configuration ....
> OS : CentOS 5.5
> User : root
> Permission:?-rwxr-xr-x ?1 root root 1248017 Nov 19 11:47
> fsg729-167-installer
> Also as I said in my last email this is a FreeSWITCH production BOX .
> Thanks
> Lloyd
>
My box is not running only FS, but has no issues running the
installer. It is also Centos 5.5 and matches what you have sent.

Is SELinux running in permissive or strict mode?


From rafonline at hotmail.com  Sun Dec 19 21:50:47 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Sun, 19 Dec 2010 18:50:47 +0000
Subject: [Freeswitch-users] audio quality issue
In-Reply-To: <AANLkTi=hyQP2g9Wam-47kTw5BED0yeurmW0O0Au52Ji_@mail.gmail.com>
References: <COL122-W33885B610991D11E824AB6B5160@phx.gbl>,
	<COL122-W5833A44AAF7FDFD440BCBB5160@phx.gbl>,
	<AANLkTi=hyQP2g9Wam-47kTw5BED0yeurmW0O0Au52Ji_@mail.gmail.com>
Message-ID: <COL122-W5133EEE8694618648B675B5180@phx.gbl>



Hi Sam

Sorry for the late reply.

My issue is with only hearing the IVR originating from the switch. 

Phone call debug:

2010-12-19 18:47:16.057209 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMU:0:8000:20]/[G7221:115:32000:20]
2010-12-19 18:47:16.057209 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMU:0:8000:20]/[G7221:107:16000:20]
2010-12-19 18:47:16.057209 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20]
2010-12-19 18:47:16.057209 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
2010-12-19 18:47:16.057209 [DEBUG] sofia_glue.c:2354 Set Codec sofia/internal/1006 at 192.168.0.10 PCMU/8000 20 ms 160 samples
.
.
.
.
EXECUTE sofia/internal/1006 at 192.168.0.10 sleep(1000)
2010-12-19 18:28:22.618376 [DEBUG] sofia.c:4153 Channel sofia/internal/1006 at 192.168.0.10 entering state [ready][200]
2010-12-19 18:28:22.658376 [DEBUG] switch_rtp.c:2066 Correct ip/port confirmed.
EXECUTE sofia/internal/1006 at 192.168.0.10 voicemail(check default 192.168.0.10)
2010-12-19 18:28:23.618435 [DEBUG] switch_ivr_play_say.c:63 No language specified - Using [en]
2010-12-19 18:28:23.818446 [DEBUG] switch_ivr_play_say.c:63 No language specified - Using [en]
2010-12-19 18:28:23.818446 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-enter_id.wav] (en:en)
2010-12-19 18:28:23.818446 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16 at 8000hz 1 channels 20ms
2010-12-19 18:28:25.778567 [DEBUG] switch_ivr_play_say.c:1444 done playing file
2010-12-19 18:28:25.878572 [DEBUG] switch_ivr_play_say.c:244 Handle say:[#] (en:en)
2010-12-19 18:28:25.878572 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16 at 8000hz 1 channels 20ms
2010-12-19 18:28:26.298600 [DEBUG] switch_ivr_play_say.c:1444 done playing file


The phone when playing 'vm-enter_id.wav' seems to jump in volume (initially low volume then high).  It also makes a high pitch creaking type noise after 'say:[#] '.

Also when it asks me to leave a message after the tone, it immediatley says 'my message was too short' without allowing me to record anything.  Could the phone be sending an echo which causes freeeswitch to think i have ended my recording?

I was wondering if i should invest in some new phones?  Any recommendations?

Cheers

Raf






Date: Sun, 19 Dec 2010 13:10:41 +0530
From: u2nsam at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] audio quality issue

Hi ,

Do you have issues only with hearing the IVR originating from the switch ? 
Phone to phone works fine when RTP pass through is enabled ? which codec gets negotiated when IVR plays and what are the IVR file properties which is negotiated?

Your voice mail gets recorded properly means the RTP is passing well to the switch ...

Regads
Sam


On Sat, Dec 18, 2010 at 2:52 AM, Rafqat . <rafonline at hotmail.com> wrote:







btw.  I have no audio issues when making calls between phones through freeswitch (voice quality is very good).



From: rafonline at hotmail.com

To: freeswitch-users at lists.freeswitch.org
Date: Fri, 17 Dec 2010 21:19:11 +0000
Subject: [Freeswitch-users] audio quality issue











Hi, 

I recently installed freeswitch 1.0.6 on centos 5.5 (hosted on xen server).  I feel as if the audio quality is not as good as what it should be (6-7 out of 10).  This is apparent in calling the default voicemail IVR, sometimes I have issues leaving a message, sometimes I hear some weird noises.  However, I have no issues with the quality of the recorded voicemail message I leave, it sounds fine on playback.


Is this an issue with my phones (Flexor 500)?  I have a softphone which doesn't seem as bad but still has issues.

I also get the following message on startup of freeswitch:

2010-12-17 21:17:39.138958 [WARNING] switch_time.c:206 Timer resolution of 4000 microseconds detected!

Do you have your kernel timer frequency set to lower than 1,000Hz? You may experience audio problems.

I assume its related. 

Any help will be much appreciated.

Cheers

Raf
 		 	   		  

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

_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
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From thisjoy0528 at gmail.com  Sun Dec 19 22:00:45 2010
From: thisjoy0528 at gmail.com (joy this)
Date: Mon, 20 Dec 2010 03:00:45 +0800
Subject: [Freeswitch-users] questions about VAD and echo cancellation
In-Reply-To: <AANLkTi=G-gm4V9zV1hLtFVJ4B6-tyQY1Rw=+LEPdnaXa@mail.gmail.com>
References: <AANLkTi=G-gm4V9zV1hLtFVJ4B6-tyQY1Rw=+LEPdnaXa@mail.gmail.com>
Message-ID: <AANLkTin0Q_NNJXdoFX4d-PYpkBKj2ct84Er5GAWCfoPf@mail.gmail.com>

Could you give me any suggestions?

Sincerely yours,
Thisjoy.
2010/12/17 joy this <thisjoy0528 at gmail.com>

> Dear all:
>
>
>
>         I have questions about VAD and echo cancellation. My FS is Version
> 1.0.head (git-) under Windows XP. My soft-phone is X-Lite. I use earphones
> and microphones for sip 1 which means the talking and hearing are separated;
> on the other hand, I use NB for sip 2 which makes the talking and hearing in
> the same place. Sip 1 (1000) calls sip 2 (1001) via FS. When I say something
> via sip 1, the echo will occur, and the echo will only occur on sip 1.
>
>         How do I cancel the echo? Buy a hardware card or something else? I
> have two gateways which are Wellgate2644 and ata171m. Please give me some
> suggestions.
>
>         When I enabled VAD, I found something strange. In the start of the
> session, I can hear the noise and echo on sip 1. If I say something via sip
> 2, the echo and the noise will disappear. Then the echo will not occur on
> sip 1 only in a short time, about several seconds. The echo and the noise
> will occur gradually if I say something via sip1. Is it a normal situation?
>
>
>
> Sincerely yours,
> Thisjoy.
>
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From brian at freeswitch.org  Sun Dec 19 22:40:05 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 19 Dec 2010 13:40:05 -0600
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
	Permission denied
In-Reply-To: <AANLkTin3ccd_JD9e-GEpGPjhZ0F9ca-E16TZ=_hwk4z0@mail.gmail.com>
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
	<AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
	<AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com>
	<AANLkTik9qMKo4GAtDN8e19M=8EnK3N+invC902wdkyW8@mail.gmail.com>
	<AANLkTikAoeGo410KS-KFi+2q-vGS9jbGuJcpYpOhndAU@mail.gmail.com>
	<AANLkTin3ccd_JD9e-GEpGPjhZ0F9ca-E16TZ=_hwk4z0@mail.gmail.com>
Message-ID: <E1173A0B-CC37-4A29-A156-229FFA7326C5@freeswitch.org>

That is the problem if /tmp/ is mounted noexec then it will not run.

/b

On Dec 19, 2010, at 12:37 PM, Tony Graziano wrote:

> My box is not running only FS, but has no issues running the
> installer. It is also Centos 5.5 and matches what you have sent.
> 
> Is SELinux running in permissive or strict mode?
> 
> _______



From lloyd.aloysius at sunteltech.ca  Sun Dec 19 22:54:43 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Sun, 19 Dec 2010 14:54:43 -0500
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
 Permission denied
In-Reply-To: <E1173A0B-CC37-4A29-A156-229FFA7326C5@freeswitch.org>
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
	<AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
	<AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com>
	<AANLkTik9qMKo4GAtDN8e19M=8EnK3N+invC902wdkyW8@mail.gmail.com>
	<AANLkTikAoeGo410KS-KFi+2q-vGS9jbGuJcpYpOhndAU@mail.gmail.com>
	<AANLkTin3ccd_JD9e-GEpGPjhZ0F9ca-E16TZ=_hwk4z0@mail.gmail.com>
	<E1173A0B-CC37-4A29-A156-229FFA7326C5@freeswitch.org>
Message-ID: <AANLkTimUEBFu0y6Ldbo=haQP=kSjzpd1dBe9WH5Cd51n@mail.gmail.com>

Brian,

Thank you.

I figured out that is the problem. I try the installer with my development
box , no issues.

Do you know any workaround for this situation?

Can I do this way

remount the /tmp as exec

once install finished the install

remount /tmp to nonexec

Thanks
Lloyd


On Sun, Dec 19, 2010 at 2:40 PM, Brian West <brian at freeswitch.org> wrote:

> That is the problem if /tmp/ is mounted noexec then it will not run.
>
> /b
>
> On Dec 19, 2010, at 12:37 PM, Tony Graziano wrote:
>
> > My box is not running only FS, but has no issues running the
> > installer. It is also Centos 5.5 and matches what you have sent.
> >
> > Is SELinux running in permissive or strict mode?
> >
> > _______
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From xduvox at gmail.com  Sun Dec 19 23:17:35 2010
From: xduvox at gmail.com (Octavio Duarte)
Date: Sun, 19 Dec 2010 14:17:35 -0600
Subject: [Freeswitch-users] gateway without static IP
Message-ID: <AANLkTi=kDe23KtpwxMd9S66JjrXhga1RK7+BWE5WAqV5@mail.gmail.com>

***Hello everyone
*
i need some help with a gateway, i can receive calls from it because i
can use it as a user so i can see it by sofia status profile internal
but when i try to make a call to the PSTN like this

*<action application="bridge"** data="sofia/internal/userGateway%{domain}^$1"/>

the gateway doesn't dial the number on variable $1

so when to do like this

**<action application="bridge" **data="sofia/internal/$1@<gateway_IP>"/>

the gateway connect the calls to the PSTN

my problem its that the gateway does not have an static IP and i want to know
**if it is possible to make calls through this gateway if
**it has a variable IP, and how to do it if so;
**the gateway uses SIP accounts to register to FS so i think it might be
**possible to know the IP's gateway by using one of the registered
extension or tell fs to use the ip of one user for the gateway set up?
**
**Anyone has ideas or knows how to do this?, help is greatly appreciated!
*
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From lloyd.aloysius at sunteltech.ca  Sun Dec 19 23:46:14 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Sun, 19 Dec 2010 15:46:14 -0500
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
 Permission denied
In-Reply-To: <AANLkTimUEBFu0y6Ldbo=haQP=kSjzpd1dBe9WH5Cd51n@mail.gmail.com>
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
	<AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
	<AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com>
	<AANLkTik9qMKo4GAtDN8e19M=8EnK3N+invC902wdkyW8@mail.gmail.com>
	<AANLkTikAoeGo410KS-KFi+2q-vGS9jbGuJcpYpOhndAU@mail.gmail.com>
	<AANLkTin3ccd_JD9e-GEpGPjhZ0F9ca-E16TZ=_hwk4z0@mail.gmail.com>
	<E1173A0B-CC37-4A29-A156-229FFA7326C5@freeswitch.org>
	<AANLkTimUEBFu0y6Ldbo=haQP=kSjzpd1dBe9WH5Cd51n@mail.gmail.com>
Message-ID: <AANLkTinA1RGjrXbgmavP_Q-DjZjLDE-TF6irc-4qudfG@mail.gmail.com>

Finally I manged to install the g729. This may helpful if someone face the
same problem like mine

mount -o remount,exec /tmp

run the fsg729-167-installer

mount -o remount /tmp


Thank you for the help.

Regards,
Lloyd

On Sun, Dec 19, 2010 at 2:54 PM, Aloysius Lloyd <
lloyd.aloysius at sunteltech.ca> wrote:

> Brian,
>
> Thank you.
>
> I figured out that is the problem. I try the installer with my development
> box , no issues.
>
> Do you know any workaround for this situation?
>
> Can I do this way
>
> remount the /tmp as exec
>
> once install finished the install
>
> remount /tmp to nonexec
>
> Thanks
> Lloyd
>
>
> On Sun, Dec 19, 2010 at 2:40 PM, Brian West <brian at freeswitch.org> wrote:
>
>> That is the problem if /tmp/ is mounted noexec then it will not run.
>>
>> /b
>>
>> On Dec 19, 2010, at 12:37 PM, Tony Graziano wrote:
>>
>> > My box is not running only FS, but has no issues running the
>> > installer. It is also Centos 5.5 and matches what you have sent.
>> >
>> > Is SELinux running in permissive or strict mode?
>> >
>> > _______
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From steveayre at gmail.com  Mon Dec 20 01:00:10 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 19 Dec 2010 22:00:10 +0000
Subject: [Freeswitch-users] gateway without static IP
In-Reply-To: <AANLkTi=kDe23KtpwxMd9S66JjrXhga1RK7+BWE5WAqV5@mail.gmail.com>
References: <AANLkTi=kDe23KtpwxMd9S66JjrXhga1RK7+BWE5WAqV5@mail.gmail.com>
Message-ID: <AANLkTi=KTyAGuqQMRTNcqxkyN1fqxP-z9NXikzeepKPM@mail.gmail.com>

First of all, I would  recommend you configure the gateway within the
<gateways> section of the SIP profile. The dialstring will then be
sofia/gwname/$1. That format has an advantage because FS will monitor
the status of the gateway and automatically mark it as offline and
stop sending traffic to it if it stops responding.

When configuring a gateway you can specify a domain name instead of an
IP. If you combine that with a service such as www.dyndns.org you can
run the gateway on a dynamic IP.

Also, have you tried:
<action application="bridge" data="sofia/internal/$1@<domain>"/>
that might work, but I've never tried it with anything but an IP so
can't say for sure without trying it.

This does not do what you think it does:
<action application="bridge" data="sofia/internal/userGateway%{domain}^$1"/>
The % syntax is for dialing an extension on the *local* server, on the
{domain} handled by the current server. That won't send a call out to
a gateway.

-Steve


On 19 December 2010 20:17, Octavio Duarte <xduvox at gmail.com> wrote:
> Hello everyone
>
> i need some help with a gateway, i can receive calls from it because i can
> use it as a user so i can see it by sofia status profile internal
> but when i try to make a call to the PSTN like this
>
>
> <action application="bridge" data="sofia/internal/userGateway%{domain}^$1"/>
>
> the gateway doesn't dial the number on variable $1
>
> so when to do like this
>
> <action application="bridge" data="sofia/internal/$1@<gateway_IP>"/>
>
>
> the gateway connect the calls to the PSTN
>
> my problem its that the gateway does not have an static IP and i want to
> know
> if it is possible to make calls through this gateway if
> it has a variable IP, and how to do it if so;
>
> the gateway uses SIP accounts to register to FS so i think it might be
> possible to know the IP's gateway by using one of the registered extension
> or tell fs to use the ip of one user for the gateway set up?
>
>
> Anyone has ideas or knows how to do this?, help is greatly appreciated!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Mon Dec 20 01:07:17 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 19 Dec 2010 16:07:17 -0600
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
	Permission denied
In-Reply-To: <AANLkTinA1RGjrXbgmavP_Q-DjZjLDE-TF6irc-4qudfG@mail.gmail.com>
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com>
	<AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com>
	<AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com>
	<AANLkTik9qMKo4GAtDN8e19M=8EnK3N+invC902wdkyW8@mail.gmail.com>
	<AANLkTikAoeGo410KS-KFi+2q-vGS9jbGuJcpYpOhndAU@mail.gmail.com>
	<AANLkTin3ccd_JD9e-GEpGPjhZ0F9ca-E16TZ=_hwk4z0@mail.gmail.com>
	<E1173A0B-CC37-4A29-A156-229FFA7326C5@freeswitch.org>
	<AANLkTimUEBFu0y6Ldbo=haQP=kSjzpd1dBe9WH5Cd51n@mail.gmail.com>
	<AANLkTinA1RGjrXbgmavP_Q-DjZjLDE-TF6irc-4qudfG@mail.gmail.com>
Message-ID: <E39CCC9F-AB0A-460E-841B-F65FD0BBDEFB@freeswitch.org>

I would add this to the installer but I don't want to be responsible for changing something on someones system in that manner.

/b

On Dec 19, 2010, at 2:46 PM, Aloysius Lloyd wrote:

> Finally I manged to install the g729. This may helpful if someone face the same problem like mine
> 
> mount -o remount,exec /tmp
> 
> run the fsg729-167-installer
> 
> mount -o remount /tmp
> 
> 
> Thank you for the help.
> 
> Regards,
> Lloyd

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From infos at madovsky.org  Mon Dec 20 01:42:01 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 19 Dec 2010 17:42:01 -0500
Subject: [Freeswitch-users] fsg729-167-installer - bad interpreter:
	Permission denied
References: <AANLkTi=_hkwRSOveEerbP2usFYph9oO2Ww01HR3DfFCq@mail.gmail.com><AANLkTikziq9voT0x_jeCWPcER7dcGFpJcxk4P=pfsBzC@mail.gmail.com><AANLkTimGrRdjcGsUYTzVDLss7UeaJ7J=KMtyUZDPXtL9@mail.gmail.com><AANLkTik9qMKo4GAtDN8e19M=8EnK3N+invC902wdkyW8@mail.gmail.com><AANLkTikAoeGo410KS-KFi+2q-vGS9jbGuJcpYpOhndAU@mail.gmail.com><AANLkTin3ccd_JD9e-GEpGPjhZ0F9ca-E16TZ=_hwk4z0@mail.gmail.com><E1173A0B-CC37-4A29-A156-229FFA7326C5@freeswitch.org><AANLkTimUEBFu0y6Ldbo=haQP=kSjzpd1dBe9WH5Cd51n@mail.gmail.com>
	<AANLkTinA1RGjrXbgmavP_Q-DjZjLDE-TF6irc-4qudfG@mail.gmail.com>
Message-ID: <5C579D218B6F4B82BECD2BADC08D5E98@e1705>

of course, /tmp is usually done for read , security...
  ----- Original Message ----- 
  From: Aloysius Lloyd 
  To: FreeSWITCH Users Help 
  Sent: Sunday, December 19, 2010 3:46 PM
  Subject: Re: [Freeswitch-users] fsg729-167-installer - bad interpreter: Permission denied


  Finally I manged to install the g729. This may helpful if someone face the same problem like mine


  mount -o remount,exec /tmp


  run the fsg729-167-installer


  mount -o remount /tmp




  Thank you for the help.


  Regards,
  Lloyd


  On Sun, Dec 19, 2010 at 2:54 PM, Aloysius Lloyd <lloyd.aloysius at sunteltech.ca> wrote:

    Brian,


    Thank you.


    I figured out that is the problem. I try the installer with my development box , no issues.


    Do you know any workaround for this situation?


    Can I do this way


    remount the /tmp as exec 


    once install finished the install


    remount /tmp to nonexec


    Thanks
    Lloyd




    On Sun, Dec 19, 2010 at 2:40 PM, Brian West <brian at freeswitch.org> wrote:

      That is the problem if /tmp/ is mounted noexec then it will not run.

      /b


      On Dec 19, 2010, at 12:37 PM, Tony Graziano wrote:

      > My box is not running only FS, but has no issues running the
      > installer. It is also Centos 5.5 and matches what you have sent.
      >
      > Is SELinux running in permissive or strict mode?
      >
      > _______


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







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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From u2nsam at gmail.com  Mon Dec 20 08:43:47 2010
From: u2nsam at gmail.com (Sam)
Date: Mon, 20 Dec 2010 11:13:47 +0530
Subject: [Freeswitch-users] audio quality issue
In-Reply-To: <COL122-W5133EEE8694618648B675B5180@phx.gbl>
References: <COL122-W33885B610991D11E824AB6B5160@phx.gbl>
	<COL122-W5833A44AAF7FDFD440BCBB5160@phx.gbl>
	<AANLkTi=hyQP2g9Wam-47kTw5BED0yeurmW0O0Au52Ji_@mail.gmail.com>
	<COL122-W5133EEE8694618648B675B5180@phx.gbl>
Message-ID: <AANLkTikP54tggoF8qWzN8q0+U4WuZkCiBegBzgx2FFE9@mail.gmail.com>

Try using a IVR sound file with 64kbps instead of 128kbps and check.

Regds
Sam

On Mon, Dec 20, 2010 at 12:20 AM, Rafqat . <rafonline at hotmail.com> wrote:

>
> Hi Sam
>
> Sorry for the late reply.
>
> My issue is with only hearing the IVR originating from the switch.
>
> Phone call debug:
>
> 2010-12-19 18:47:16.057209 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
> [PCMU:0:8000:20]/[G7221:115:32000:20]
> 2010-12-19 18:47:16.057209 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
> [PCMU:0:8000:20]/[G7221:107:16000:20]
> 2010-12-19 18:47:16.057209 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
> [PCMU:0:8000:20]/[G722:9:8000:20]
> 2010-12-19 18:47:16.057209 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
> [PCMU:0:8000:20]/[PCMU:0:8000:20]
> 2010-12-19 18:47:16.057209 [DEBUG] sofia_glue.c:2354 Set Codec
> sofia/internal/1006 at 192.168.0.10 PCMU/8000 20 ms 160 samples
> .
> .
> .
> .
> EXECUTE sofia/internal/1006 at 192.168.0.10 sleep(1000)
> 2010-12-19 18:28:22.618376 [DEBUG] sofia.c:4153 Channel sofia/internal/
> 1006 at 192.168.0.10 entering state [ready][200]
> 2010-12-19 18:28:22.658376 [DEBUG] switch_rtp.c:2066 Correct ip/port
> confirmed.
> EXECUTE sofia/internal/1006 at 192.168.0.10 voicemail(check default
> 192.168.0.10)
> 2010-12-19 18:28:23.618435 [DEBUG] switch_ivr_play_say.c:63 No language
> specified - Using [en]
> 2010-12-19 18:28:23.818446 [DEBUG] switch_ivr_play_say.c:63 No language
> specified - Using [en]
> 2010-12-19 18:28:23.818446 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-enter_id.wav] (en:en)
> 2010-12-19 18:28:23.818446 [DEBUG] switch_ivr_play_say.c:1152 Codec
> Activated L16 at 8000hz 1 channels 20ms
> 2010-12-19 18:28:25.778567 [DEBUG] switch_ivr_play_say.c:1444 done playing
> file
> 2010-12-19 18:28:25.878572 [DEBUG] switch_ivr_play_say.c:244 Handle say:[#]
> (en:en)
> 2010-12-19 18:28:25.878572 [DEBUG] switch_ivr_play_say.c:1152 Codec
> Activated L16 at 8000hz 1 channels 20ms
> 2010-12-19 18:28:26.298600 [DEBUG] switch_ivr_play_say.c:1444 done playing
> file
>
>
> The phone when playing 'vm-enter_id.wav' seems to jump in volume (initially
> low volume then high).  It also makes a high pitch creaking type noise after
> 'say:[#] '.
>
> Also when it asks me to leave a message after the tone, it immediatley says
> 'my message was too short' without allowing me to record anything.  Could
> the phone be sending an echo which causes freeeswitch to think i have ended
> my recording?
>
> I was wondering if i should invest in some new phones?  Any
> recommendations?
>
> Cheers
>
> Raf
>
>
>
>
>
>
> ------------------------------
> Date: Sun, 19 Dec 2010 13:10:41 +0530
> From: u2nsam at gmail.com
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] audio quality issue
>
>
> Hi ,
>
> Do you have issues only with hearing the IVR originating from the switch ?
> Phone to phone works fine when RTP pass through is enabled ? which codec
> gets negotiated when IVR plays and what are the IVR file properties which is
> negotiated?
> Your voice mail gets recorded properly means the RTP is passing well to the
> switch ...
>
> Regads
> Sam
>
>
> On Sat, Dec 18, 2010 at 2:52 AM, Rafqat . <rafonline at hotmail.com> wrote:
>
>
> btw.  I have no audio issues when making calls between phones through
> freeswitch (voice quality is very good).
>
>
>
> ------------------------------
> From: rafonline at hotmail.com
> To: freeswitch-users at lists.freeswitch.org
> Date: Fri, 17 Dec 2010 21:19:11 +0000
> Subject: [Freeswitch-users] audio quality issue
>
>
>
>
> Hi,
>
> I recently installed freeswitch 1.0.6 on centos 5.5 (hosted on xen
> server).  I feel as if the audio quality is not as good as what it should be
> (6-7 out of 10).  This is apparent in calling the default voicemail IVR,
> sometimes I have issues leaving a message, sometimes I hear some weird
> noises.  However, I have no issues with the quality of the recorded
> voicemail message I leave, it sounds fine on playback.
>
> Is this an issue with my phones (Flexor 500)?  I have a softphone which
> doesn't seem as bad but still has issues.
>
> I also get the following message on startup of freeswitch:
>
> 2010-12-17 21:17:39.138958 [WARNING] switch_time.c:206 Timer resolution of
> 4000 microseconds detected!
> Do you have your kernel timer frequency set to lower than 1,000Hz? You may
> experience audio problems.
>
> I assume its related.
>
> Any help will be much appreciated.
>
> Cheers
>
> Raf
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mail at jankubr.com  Mon Dec 20 13:10:18 2010
From: mail at jankubr.com (Jan Kubr)
Date: Mon, 20 Dec 2010 11:10:18 +0100
Subject: [Freeswitch-users] SIP Registration DNS Error
In-Reply-To: <A27AF231-08B8-44D2-85A7-2A6353C73AD5@freeswitch.org>
References: <06502C073AD9394AADB3CA7FD94931BC0519DF46@okc1x1.Logixcom.com>
	<AANLkTi=10HbngBxzwzf=i5PpKMhG5tdzWvtcCdA_weJt@mail.gmail.com>
	<49E1E293-BE37-43AA-960C-56B921FEBC9D@ipeva.fr>
	<AANLkTins8Z6c=dHp=fDta8u9WEmdahz+vqeD7vfvuxOa@mail.gmail.com>
	<AANLkTimv_dDh5ynwuOD89XMqatw929YMC=PNL1gOM2Gf@mail.gmail.com>
	<D1456321-8983-47F8-A662-8F6183C26E65@freeswitch.org>
	<AANLkTimqApc_gXv6+0LywmqPOn4Z5aPMXZeHvy-eiyP5@mail.gmail.com>
	<AANLkTimw2t5P_ELc3PMGf+HB25q-_fW-NtYMPQeJsL1z@mail.gmail.com>
	<A27AF231-08B8-44D2-85A7-2A6353C73AD5@freeswitch.org>
Message-ID: <AANLkTi=C_wHx4yE9gJ7eckaUfFGXB4evcMEvsM8WwdWZ@mail.gmail.com>

Makes sense, thanks!

Jan

On Sun, Dec 19, 2010 at 6:24 PM, Brian West <brian at freeswitch.org> wrote:

> Thats what it looks like sofia follows the rules to the letter of the law.
>  they need to fix their NAPTR records.
>
> /b
>
> On Dec 19, 2010, at 6:05 AM, Jan Kubr wrote:
>
> So it's their incorrect NAPTR record? I'm only learning about this
> mechanism. I'll be interested to see what the NAPTR record is when the
> registration stops failing again.
>
> On Sun, Dec 19, 2010 at 12:30 PM, Jan Kubr <mail at jankubr.com> wrote:
>
>> Here it is: http://pastebin.freeswitch.org/14824
>>
>> On Sun, Dec 19, 2010 at 2:56 AM, Brian West <brian at freeswitch.org> wrote:
>>
>>> sofia loglevel all 9
>>> watch it and paste that in pastebin... send us the link.
>>>
>>> /b
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From hwnorman at hotmail.com  Mon Dec 20 13:20:12 2010
From: hwnorman at hotmail.com (Norman Lam)
Date: Mon, 20 Dec 2010 18:20:12 +0800
Subject: [Freeswitch-users] Error in compiling tar ball
	freeswitch-1.0.6.tar.gz on windows
Message-ID: <SNT109-DS5FB277D37D16C78A3FF61C1190@phx.gbl>

Hi , I am newbie , can't compile on AMD Athon X2, winxp sp3, vc++2008
express, follow the packt book , and the freeswitch website 

 

Please advise, thanks advance

 

Norman

 

 

1>------ Build started: Project: libpcre Generate pcre_chartables.c,
Configuration: Debug Win32 ------

2>------ Build started: Project: libapr, Configuration: Debug Win32 ------

1>Compiling...

2>Performing Pre-Build Event...

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_alloca
tor.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_atomic
.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_dso.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_env.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_errno.
h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_file_i
nfo.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_file_i
o.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_fnmatc
h.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_genera
l.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_getopt
.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_global
_mutex.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_hash.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_inheri
t.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_lib.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_mmap.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_networ
k_io.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_poll.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_pools.
h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_portab
le.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_proc_m
utex.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_random
.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_ring.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_shm.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_signal
.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_string
s.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_suppor
t.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_tables
.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_thread
_cond.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_thread
_mutex.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_thread
_proc.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_thread
_rwlock.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_time.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_user.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_versio
n.h

2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_want.h

2>36 File(s) copied

2>Compiling...

1>dftables.c

2>userinfo.c

1>Compiling manifest to resources...

1>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

1>Copyright (C) Microsoft Corporation.  All rights reserved.

1>Linking...

1>   Creating library Debug\dftables.lib and object Debug\dftables.exp

2>groupinfo.c

1>Embedding manifest...

2>timestr.c

1>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

1>Copyright (C) Microsoft Corporation.  All rights reserved.

2>time.c

1>Performing Post-Build Event...

2>access.c

1>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\pcre\Debug\BuildLog.htm"

1>libpcre Generate pcre_chartables.c - 0 error(s), 0 warning(s)

2>threadpriv.c

3>------ Build started: Project: libpcre, Configuration: Debug Win32 ------

3>libpcre : warning PRJ0009 : Build log could not be opened for writing.

3>Make sure that the file is not open by another process and is not
write-protected.

3>Compiling...

3>pcre_compile.c

2>thread.c

3>pcre_config.c

3>pcre_dfa_exec.c

2>signals.c

3>pcre_exec.c

2>proc.c

3>pcre_fullinfo.c

3>pcre_get.c

2>apr_tables.c

3>pcre_globals.c

2>apr_hash.c

3>pcre_info.c

2>apr_strtok.c

3>pcre_maketables.c

3>pcre_ord2utf8.c

2>apr_strnatcmp.c

3>pcre_refcount.c

2>apr_strings.c

3>pcre_study.c

3>pcre_tables.c

2>apr_snprintf.c

3>pcre_try_flipped.c

3>pcre_newline.c

2>apr_fnmatch.c

3>pcre_ucd.c

3>pcre_ucp_searchfuncs.c

2>apr_cpystrn.c

3>..\..\pcre\pcre_ucp_searchfuncs.c(158) : warning C4018: '<' :
signed/unsigned mismatch

3>..\..\pcre\pcre_ucp_searchfuncs.c(163) : warning C4018: '<=' :
signed/unsigned mismatch

3>pcre_valid_utf8.c

2>shm.c

3>pcre_version.c

3>pcre_xclass.c

2>sha2_glue.c

3>Generating Code...

2>sha2.c

2>Generating Code...

2>Compiling...

2>apr_random.c

3>Compiling...

3>pcre_chartables.c

3>Generating Code...

3>Creating library...

2>apr_getpass.c

3>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\pcre\Debug\BuildLog.htm"

3>libpcre - 0 error(s), 3 warning(s)

4>------ Build started: Project: libsqlite, Configuration: Debug Win32
------

4>Performing Pre-Build Event...

2>sockopt.c

4>Compiling...

4>analyze.c

4>attach.c

2>sockets.c

4>auth.c

4>btree.c

2>sockaddr.c

4>build.c

4>callback.c

4>complete.c

2>sendrecv.c

4>date.c

4>delete.c

2>select.c

4>expr.c

4>func.c

2>multicast.c

4>hash.c

4>insert.c

4>legacy.c

2>inet_pton.c

4>loadext.c

2>inet_ntop.c

2>mmap.c

4>main.c

4>opcodes.c

4>os.c

2>common.c

4>os_win.c

2>version.c

2>utf8.c

4>pager.c

2>start.c

4>Generating Code...

4>Compiling...

4>parse.c

2>rand.c

4>pragma.c

2>otherchild.c

4>prepare.c

2>misc.c

4>printf.c

4>random.c

2>internal.c

4>select.c

4>shell.c

4>table.c

4>tokenize.c

2>getopt.c

2>Generating Code...

4>trigger.c

2>Compiling...

2>errorcodes.c

4>update.c

4>utf.c

4>util.c

2>env.c

4>vacuum.c

2>charset.c

4>vdbe.c

4>vdbeapi.c

2>apr_pools.c

4>vdbeaux.c

2>thread_rwlock.c

4>vdbefifo.c

4>vdbemem.c

2>thread_mutex.c

4>vtab.c

2>thread_cond.c

2>proc_mutex.c

4>Generating Code...

4>Compiling...

4>where.c

4>alter.c

2>tempdir.c

4>Generating Code...

4>Creating library...

4>Performing Post-Build Event...

4>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\sqlite\Debug\BuildLog.htm"

4>libsqlite - 0 error(s), 0 warning(s)

2>seek.c

5>------ Build started: Project: libsrtp, Configuration: Debug Win32 ------

2>readwrite.c

5>Creating config.h from config.hw

5>Compiling...

5>alloc.c

2>pipe.c

2>open.c

5>crypto_kernel.c

2>mktemp.c

2>fullrw.c

5>ctr_prng.c

2>flock.c

5>err.c

2>filesys.c

2>filestat.c

5>key.c

5>prng.c

2>filepath_util.c

2>filepath.c

5>rand_source.c

2>Generating Code...

5>aes.c

2>Compiling...

2>filedup.c

5>aes_cbc.c

2>fileacc.c

5>aes_icm.c

2>dir.c

2>copy.c

5>cipher.c

2>dso.c

5>null_cipher.c

2>apr_atomic.c

2>Generating Code...

5>auth.c

2>Compiling resources...

2>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

2>Copyright (C) Microsoft Corporation.  All rights reserved.

2>Linking...

2>   Creating library .\Debug/libapr-1.lib and object .\Debug/libapr-1.exp

5>hmac.c

5>null_auth.c

2>Embedding manifest...

2>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\Debug\BuildLog.htm"

2>libapr - 0 error(s), 0 warning(s)

5>sha1.c

6>------ Build started: Project: libteletone, Configuration: Debug Win32
------

5>rdb.c

5>rdbx.c

6>Compiling...

6>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

6>libteletone_detect.c

6>libteletone_generate.c

6>Generating Code...

6>Linking...

6>   Creating library Debug/libteletone.lib and object Debug/libteletone.exp

6>Embedding manifest...

5>ut_sim.c

5>datatypes.c

6>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\libteletone\Debug\BuildLog.htm"

6>libteletone - 0 error(s), 1 warning(s)

7>------ Build started: Project: libaprutil, Configuration: Debug Win32
------

7>Performing Pre-Build Event...

7>The system cannot find the file specified.

7>The system cannot find the file specified.

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_anylock.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_base64.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_buckets.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_date.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_dbd.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_dbm.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_hooks.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_ldap.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_ldap_init.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_ldap_option.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_ldap_url.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_md4.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_md5.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_optional.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_optional_hooks.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_queue.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_reslist.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_rmm.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_sdbm.h

5>Generating Code...

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_sha1.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_strmatch.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_uri.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_uuid.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_xlate.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apr_xml.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apu.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apu_config.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apu_select_dbm.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apu_version.h

7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\
apu_want.h

7>30 File(s) copied

7>Creating apu_want.h from apu_want.hw

7>Creating apu_select_dbm.h from apu_select_dbm.hw

7>Creating apu_config.h from apu_config.hw

7>Creating apu.h from apu.hw

5>Compiling...

5>stat.c

7>Creating apr_ldap.h from apr_ldap.hw

7>Compiling...

7>xlate.c

7>apr_queue.c

7>apr_base64.c

5>srtp.c

7>uuid.c

7>getuuid.c

7>apr_sha1.c

7>apr_md5.c

5>Generating Code...

5>Creating library...

7>apr_md4.c

5>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\srtp\Debug\BuildLog.htm"

5>libsrtp - 0 error(s), 0 warning(s)

7>Generating Code...

8>------ Build started: Project: Download sphinxmodel, Configuration: Debug
Win32 ------

7>Compiling resources...

8>Downloading sphinxmodel.

7>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

7>Copyright (C) Microsoft Corporation.  All rights reserved.

8>Downloading: http://files.freeswitch.org/downloads/win32/7za.exe

8>Downloading:
http://files.freeswitch.org/downloads/libs/communicator_semi_6000_20080321.t
ar.gz

8>Extracting:
C:\FreeSWITCH\freeswitch-1.0.6\libs\communicator_semi_6000_20080321.tar.gz

8>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

8>Processing archive:
C:\FreeSWITCH\freeswitch-1.0.6\libs\communicator_semi_6000_20080321.tar.gz

8>Extracting  communicator_semi_6000_20080321.tar

8>Everything is Ok

8>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

8>Processing archive:
C:\FreeSWITCH\freeswitch-1.0.6\libs\communicator_semi_6000_20080321.tar

8>Extracting  Communicator_semi_40.cd_semi_6000

8>Extracting  Communicator_semi_40.cd_semi_6000\transition_matrices

8>Extracting  Communicator_semi_40.cd_semi_6000\mdef

8>Extracting  Communicator_semi_40.cd_semi_6000\feat.params

8>Extracting  Communicator_semi_40.cd_semi_6000\means

8>Extracting  Communicator_semi_40.cd_semi_6000\noisedict

8>Extracting  Communicator_semi_40.cd_semi_6000\variances

8>Extracting  Communicator_semi_40.cd_semi_6000\sendump

8>Extracting  Communicator_semi_40.cd_semi_6000\COPYING

8>Everything is Ok

8>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download
sphinxmodel.htm"

8>Download sphinxmodel - 0 error(s), 0 warning(s)

7>Linking...

7>   Creating library .\Debug/libaprutil-1.lib and object
.\Debug/libaprutil-1.exp

7>Embedding manifest...

9>------ Build started: Project: Download sphinxbase, Configuration: Debug
Win32 ------

9>Downloading sphinxbase.

7>Performing Post-Build Event...

9>Downloading:
http://files.freeswitch.org/downloads/libs/sphinxbase-0.4.99-20091212.tar.gz

7>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\Debug\BuildLog.ht
m"

7>libaprutil - 0 error(s), 0 warning(s)

10>------ Build started: Project: libspeexdsp, Configuration: Debug Win32
------

10>Compiling...

10>fftwrap.c

9>Extracting:
C:\FreeSWITCH\freeswitch-1.0.6\libs\sphinxbase-0.4.99-20091212.tar.gz

10>filterbank.c

9>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

9>Processing archive:
C:\FreeSWITCH\freeswitch-1.0.6\libs\sphinxbase-0.4.99-20091212.tar.gz

9>Extracting  sphinxbase-0.4.99-20091212.tar

9>Everything is Ok

10>jitter.c

10>kiss_fft.c

10>kiss_fftr.c

10>mdf.c

10>preprocess.c

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11>Everything is Ok

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12>Downloading Lame.

11>Build log was saved at
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11>Download PTHREAD - 0 error(s), 0 warning(s)

13>------ Build started: Project: make_at_dictionary, Configuration: All
Win32 ------

13>Compiling...

12>Downloading: http://downloads.xiph.org/releases/ogg/libogg-1.1.3.tar.gz

13>make_at_dictionary.c

13>Linking...

13>Embedding manifest...

13>Performing Post-Build Event...

13>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\spandsp\src\msvc\All\BuildLog
make_at_dictionary.htm"

13>make_at_dictionary - 0 error(s), 0 warning(s)

12>Extracting: C:\FreeSWITCH\freeswitch-1.0.6\libs\libogg-1.1.3.tar.gz

12>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

12>Processing archive:
C:\FreeSWITCH\freeswitch-1.0.6\libs\libogg-1.1.3.tar.gz

12>Extracting  libogg-1.1.3.tar

12>Everything is Ok

14>------ Build started: Project: pthread, Configuration: Debug DLL Win32
------

14>Compiling...

14>pthread.c

12>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

12>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\libogg-1.1.3.tar

12>Extracting  libogg-1.1.3

12>Extracting  libogg-1.1.3\doc

12>Extracting  libogg-1.1.3\doc\white-ogg.png

12>Extracting  libogg-1.1.3\doc\framing.html

12>Extracting  libogg-1.1.3\doc\index.html

12>Extracting  libogg-1.1.3\doc\ogg-multiplex.html

12>Extracting  libogg-1.1.3\doc\Makefile.am

12>Extracting  libogg-1.1.3\doc\Makefile.in

12>Extracting  libogg-1.1.3\doc\white-xifish.png

12>Extracting  libogg-1.1.3\doc\libogg

12>Extracting  libogg-1.1.3\doc\libogg\ogg_page_continued.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_sync_pageout.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_page_pageno.html

12>Extracting  libogg-1.1.3\doc\libogg\encoding.html

12>Extracting  libogg-1.1.3\doc\libogg\datastructures.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_packetin.html

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_look1.html

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_writeclear.html

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_read.html

12>Extracting  libogg-1.1.3\doc\libogg\style.css

12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_eos.html

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_write.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_reset_serialno.html

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_adv1.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_page_eos.html

12>Extracting  libogg-1.1.3\doc\libogg\vorbis_info.html

12>Extracting  libogg-1.1.3\doc\libogg\general.html

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_readinit.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_sync_buffer.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_page_packets.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_sync_reset.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_sync_state.html

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_writetrunc.html

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_bits.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_packet.html

12>Extracting  libogg-1.1.3\doc\libogg\index.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_flush.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_packet_clear.html

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_writeinit.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_reset.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_sync_clear.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_state.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_page_checksum_set.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_page_serialno.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_page_bos.html

12>Extracting  libogg-1.1.3\doc\libogg\overview.html

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_adv.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_clear.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_sync_wrote.html

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_bytes.html

12>Extracting  libogg-1.1.3\doc\libogg\Makefile.am

12>Extracting  libogg-1.1.3\doc\libogg\Makefile.in

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_reset.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_packetpeek.html

12>Extracting  libogg-1.1.3\doc\libogg\reference.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_sync_pageseek.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_destroy.html

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_read1.html

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_writecopy.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_pageout.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_page_granulepos.html

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_look.html

12>Extracting  libogg-1.1.3\doc\libogg\bitpacking.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_page.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_packetout.html

12>Extracting  libogg-1.1.3\doc\libogg\decoding.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_sync_init.html

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_writealign.html

12>Extracting  libogg-1.1.3\doc\libogg\vorbis_comment.html

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_get_buffer.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_pagein.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_page_version.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_sync_destroy.html

14>Compiling resources...

14>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

14>Copyright (C) Microsoft Corporation.  All rights reserved.

14>Compiling manifest to resources...

14>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

14>Copyright (C) Microsoft Corporation.  All rights reserved.

14>Linking...

12>Extracting  libogg-1.1.3\doc\libogg\oggpack_buffer.html

12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_init.html

12>Extracting  libogg-1.1.3\doc\vorbisword2.png

12>Extracting  libogg-1.1.3\doc\stream.png

12>Extracting  libogg-1.1.3\doc\oggstream.html

12>Extracting  libogg-1.1.3\doc\rfc3533.txt

12>Extracting  libogg-1.1.3\doc\rfc3534.txt

12>Extracting  libogg-1.1.3\src

12>Extracting  libogg-1.1.3\src\framing.c

12>Extracting  libogg-1.1.3\src\bitwise.c

12>Extracting  libogg-1.1.3\src\Makefile.am

12>Extracting  libogg-1.1.3\src\Makefile.in

12>Extracting  libogg-1.1.3\compile

12>Extracting  libogg-1.1.3\depcomp

12>Extracting  libogg-1.1.3\aclocal.m4

12>Extracting  libogg-1.1.3\macos

12>Extracting  libogg-1.1.3\macos\libogg.mcp

12>Extracting  libogg-1.1.3\macos\libogg.mcp.exp

12>Extracting  libogg-1.1.3\macos\compat

12>Extracting  libogg-1.1.3\macos\compat\sys

12>Extracting  libogg-1.1.3\macos\compat\sys\types.h

12>Extracting  libogg-1.1.3\macos\compat\strdup.c

12>Extracting  libogg-1.1.3\win32

12>Extracting  libogg-1.1.3\win32\ogg_dynamic.dsp

12>Extracting  libogg-1.1.3\win32\build_ogg_dynamic_debug.bat

12>Extracting  libogg-1.1.3\win32\build_ogg_dynamic.bat

12>Extracting  libogg-1.1.3\win32\Makefile.am

12>Extracting  libogg-1.1.3\win32\Makefile.in

12>Extracting  libogg-1.1.3\win32\build_ogg_static_debug.bat

12>Extracting  libogg-1.1.3\win32\ogg_static.dsp

12>Extracting  libogg-1.1.3\win32\ogg.def

12>Extracting  libogg-1.1.3\win32\ogg.dsw

12>Extracting  libogg-1.1.3\win32\build_ogg_static.bat

12>Extracting  libogg-1.1.3\README

12>Extracting  libogg-1.1.3\ltmain.sh

12>Extracting  libogg-1.1.3\configure

12>Extracting  libogg-1.1.3\configure.in

12>Extracting  libogg-1.1.3\config.guess

12>Extracting  libogg-1.1.3\install-sh

12>Extracting  libogg-1.1.3\config.sub

12>Extracting  libogg-1.1.3\missing

12>Extracting  libogg-1.1.3\debian

12>Extracting  libogg-1.1.3\debian\control

12>Extracting  libogg-1.1.3\debian\libogg-dev.docs

12>Extracting  libogg-1.1.3\debian\rules

12>Extracting  libogg-1.1.3\debian\watch

12>Extracting  libogg-1.1.3\debian\changelog

12>Extracting  libogg-1.1.3\debian\libogg0.README.Debian

12>Extracting  libogg-1.1.3\debian\libogg-dev.install

12>Extracting  libogg-1.1.3\debian\copyright

12>Extracting  libogg-1.1.3\debian\libogg0.install

12>Extracting  libogg-1.1.3\debian\.cvsignore

12>Extracting  libogg-1.1.3\libogg.spec.in

12>Extracting  libogg-1.1.3\ogg.pc.in

12>Extracting  libogg-1.1.3\Makefile.am

12>Extracting  libogg-1.1.3\Makefile.in

12>Extracting  libogg-1.1.3\macosx

12>Extracting  libogg-1.1.3\macosx\Ogg.xcodeproj

12>Extracting  libogg-1.1.3\macosx\Ogg.xcodeproj\project.pbxproj

12>Extracting  libogg-1.1.3\macosx\Ogg_Prefix.pch

12>Extracting  libogg-1.1.3\macosx\English.lproj

12>Extracting  libogg-1.1.3\macosx\English.lproj\InfoPlist.strings

12>Extracting  libogg-1.1.3\macosx\Info.plist

12>Extracting  libogg-1.1.3\config.h.in

12>Extracting  libogg-1.1.3\ogg-uninstalled.pc.in

12>Extracting  libogg-1.1.3\ogg.m4

12>Extracting  libogg-1.1.3\AUTHORS

12>Extracting  libogg-1.1.3\CHANGES

12>Extracting  libogg-1.1.3\include

12>Extracting  libogg-1.1.3\include\ogg

12>Extracting  libogg-1.1.3\include\ogg\ogg.h

12>Extracting  libogg-1.1.3\include\ogg\Makefile.am

12>Extracting  libogg-1.1.3\include\ogg\Makefile.in

12>Extracting  libogg-1.1.3\include\ogg\config_types.h.in

12>Extracting  libogg-1.1.3\include\ogg\os_types.h

12>Extracting  libogg-1.1.3\include\Makefile.am

12>Extracting  libogg-1.1.3\include\Makefile.in

12>Extracting  libogg-1.1.3\COPYING

12>Extracting  libogg-1.1.3\libogg.spec

12>Everything is Ok

14>   Creating library .\./pthreadVC2.lib and object .\./pthreadVC2.exp

14>Embedding manifest...

14>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

14>Copyright (C) Microsoft Corporation.  All rights reserved.

14>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\pthread\Debug
DLL\BuildLog.htm"

14>pthread - 0 error(s), 0 warning(s)

15>------ Build started: Project: make_modem_filter, Configuration: All
Win32 ------

15>Compiling...

15>filter_tools.c

15>getopt.c

15>make_modem_filter.c

12>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download
OGG.htm"

12>Download OGG - 0 error(s), 0 warning(s)

15>Generating Code...

15>Linking...

16>------ Build started: Project: FreeSwitchCoreLib, Configuration: Debug
Win32 ------

15>Embedding manifest...

16>Generating switch_version.h

15>Performing Post-Build Event...

16>Downloading:
http://files.freeswitch.org/downloads/win32/fs_svnversion.exe

15>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\spandsp\src\msvc\All\BuildLog
make_modem_filter.htm"

15>make_modem_filter - 0 error(s), 0 warning(s)

17>------ Build started: Project: curllib, Configuration: Debug Win32 ------

17>Compiling...

16>Downloading: http://files.freeswitch.org/downloads/win32/libdb44.dll

17>connect.c

17>content_encoding.c

16>Downloading:
http://files.freeswitch.org/downloads/win32/libsvn_diff-1.dll

17>cookie.c

17>dict.c

16>Downloading:
http://files.freeswitch.org/downloads/win32/libsvn_subr-1.dll

17>easy.c

16>Downloading: http://files.freeswitch.org/downloads/win32/libsvn_wc-1.dll

17>escape.c

16>Downloading: http://files.freeswitch.org/downloads/win32/intl3_svn.dll

17>file.c

16>Downloading: http://files.freeswitch.org/downloads/win32/libapr-1.dll

17>formdata.c

16>Downloading: http://files.freeswitch.org/downloads/win32/libaprutil-1.dll

17>ftp.c

16>Downloading:
http://files.freeswitch.org/downloads/win32/libapriconv-1.dll

17>getenv.c

16>Downloading:
http://files.freeswitch.org/downloads/win32/libsvn_delta-1.dll

17>getinfo.c

17>gtls.c

17>hash.c

17>hostares.c

17>hostasyn.c

16>Compiling...

16>switch_buffer.c

17>hostip.c

17>hostip4.c

17>hostip6.c

17>hostsyn.c

17>hostthre.c

17>Generating Code...

17>Compiling...

17>http.c

17>http_chunks.c

17>http_digest.c

17>http_negotiate.c

17>http_ntlm.c

17>if2ip.c

17>inet_ntop.c

17>inet_pton.c

17>krb4.c

17>ldap.c

17>llist.c

17>md5.c

17>memdebug.c

17>mprintf.c

17>multi.c

17>netrc.c

17>parsedate.c

17>progress.c

17>security.c

17>select.c

16>Compiling...

16>switch_apr.c

17>Generating Code...

17>Compiling...

17>sendf.c

16>switch_caller.c

17>share.c

16>switch_channel.c

17>socks.c

16>switch_config.c

17>speedcheck.c

16>switch_console.c

17>splay.c

16>switch_core.c

17>sslgen.c

17>ssluse.c

16>switch_core_asr.c

17>strequal.c

16>switch_core_codec.c

17>strerror.c

16>switch_core_db.c

17>strtok.c

16>switch_core_directory.c

17>strtoofft.c

16>switch_core_event_hook.c

17>telnet.c

16>switch_core_file.c

17>tftp.c

17>timeval.c

16>switch_core_hash.c

17>transfer.c

17>url.c

16>switch_core_io.c

17>version.c

16>switch_core_media_bug.c

17>base64.c

17>Generating Code...

16>switch_core_memory.c

17>Creating library...

16>switch_core_port_allocator.c

17>Creating browse information file...

16>switch_core_rwlock.c

16>switch_core_speech.c

16>switch_core_sqldb.c

16>Generating Code...

16>Compiling...

16>switch_core_state_machine.c

16>switch_core_timer.c

16>switch_dso.c

16>switch_event.c

16>switch_ivr.c

16>switch_ivr_async.c

16>switch_ivr_bridge.c

16>switch_ivr_menu.c

16>switch_ivr_play_say.c

16>switch_ivr_say.c

16>switch_loadable_module.c

16>switch_log.c

16>switch_mprintf.c

16>switch_odbc.c

16>switch_pcm.c

16>switch_profile.c

16>switch_regex.c

16>switch_resample.c

16>switch_rtp.c

16>switch_scheduler.c

16>Generating Code...

16>Compiling...

16>switch_stun.c

16>switch_time.c

16>switch_utils.c

16>switch_xml.c

16>switch_xml_config.c

16>Generating Code...

16>Compiling...

16>getgateway.c

16>natpmp.c

16>Generating Code...

16>Compiling...

16>igd_desc_parse.c

16>minisoap.c

16>minissdpc.c

16>miniwget.c

17>Microsoft Browse Information Maintenance Utility Version 9.00.30729

17>Copyright (C) Microsoft Corporation. All rights reserved.

17>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\curl\Debug\BuildLog.htm"

17>curllib - 0 error(s), 0 warning(s)

16>minixml.c

16>upnpcommands.c

16>upnperrors.c

16>upnpreplyparse.c

18>------ Build started: Project: libtiff, Configuration: Debug Win32 ------

18>Performing Pre-Build Event...

16>Generating Code...

18>        1 file(s) copied.

18>        1 file(s) copied.

18>Compiling...

16>Compiling...

18>tif_close.c

16>miniupnpc.c

18>tif_codec.c

18>tif_color.c

18>tif_compress.c

18>tif_dir.c

16>Compiling...

18>tif_dirinfo.c

16>switch_nat.c

18>tif_dirread.c

18>tif_dirwrite.c

18>tif_dumpmode.c

16>Compiling...

18>tif_error.c

16>switch_ivr_originate.c

18>tif_extension.c

18>tif_fax3.c

18>tif_fax3sm.c

18>tif_flush.c

18>tif_getimage.c

16>Compiling...

16>switch_cpp.cpp

18>tif_jpeg.c

18>tif_luv.c

18>tif_lzw.c

18>tif_next.c

18>tif_ojpeg.c

18>Generating Code...

18>Compiling...

18>tif_open.c

18>tif_packbits.c

18>tif_pixarlog.c

18>tif_predict.c

18>tif_print.c

18>tif_read.c

18>tif_strip.c

18>tif_swab.c

18>tif_thunder.c

18>tif_tile.c

18>tif_unix.c

18>tif_version.c

18>tif_warning.c

18>tif_write.c

18>tif_zip.c

18>tif_aux.c

18>Generating Code...

18>Creating library...

18>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\spandsp\src\Debug\BuildLog
libtiff.htm"

18>libtiff - 0 error(s), 0 warning(s)

19>------ Build started: Project: iksemel, Configuration: Debug Win32 ------

19>Compiling...

19>dom.c

19>filter.c

19>iks.c

19>ikstack.c

19>io-posix.c

16>Compiling...

16>switch_core_session.c

19>jabber.c

16>Compiling...

19>md5.c

16>stfu.c

19>sax.c

19>sha.c

16>Compiling...

16>g711.c

19>stream.c

16>inet_pton.c

19>utility.c

19>base64.c

19>Generating Code...

19>Creating library...

19>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\iksemel\Debug\BuildLog.htm
"

19>iksemel - 0 error(s), 0 warning(s)

16>Generating Code...

16>Compiling manifest to resources...

16>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

16>Copyright (C) Microsoft Corporation.  All rights reserved.

20>------ Build started: Project: Download pocketsphinx, Configuration:
Debug Win32 ------

16>Linking...

20>Downloading pocketsphinx.

20>Downloading:
http://files.freeswitch.org/downloads/libs/pocketsphinx-0.5.99-20091212.tar.
gz

16>   Creating library Debug/FreeSwitchCore.lib and object
Debug/FreeSwitchCore.exp

16>Embedding manifest...

16>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

16>Copyright (C) Microsoft Corporation.  All rights reserved.

16>Performing Post-Build Event...

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\extensions.conf

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\freeswitch.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\fur_elise.ttml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\mime.types

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\notify-voicemail.tpl

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\tetris.ttml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\vars.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\voicemail.tpl

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\web-vm.tpl

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\acl.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\alsa.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\cdr_csv.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\cdr_pg_csv.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\cidlookup.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\conference.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\console.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\dialplan_directory.c
onf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\dingaling.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\directory.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\distributor.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\easyroute.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\enum.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\erlang_event.conf.xm
l

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\event_multicast.conf
.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\event_socket.conf.xm
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16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\fax.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\fifo.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\ivr.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\java.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\lcr.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\limit.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\local_stream.conf.xm
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16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\logfile.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\lua.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\memcache.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\modules.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\nibblebill.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\opal.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\perl.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\pocketsphinx.conf.xm
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16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\portaudio.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\post_load_modules.co
nf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\python.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\rss.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\sangoma_codec.conf.x
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16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\shout.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\skinny.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\sofia.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\spidermonkey.conf.xm
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16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\switch.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\syslog.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\timezones.conf.xml

20>Extracting:
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16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\tts_commandline.conf
.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\unicall.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\unimrcp.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\voicemail.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\xml_cdr.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\xml_curl.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\xml_rpc.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\zeroconf.conf.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\features.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\public.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default\00_pizza_demo.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default\01_example.com.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default\99999_enum.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default\ideasip.com.noload

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default\pulver.com.noload

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default\sipbroker.com.noload

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default\sipphone.com.noload

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default\tollfreegateway.com.
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16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\public\00_inbound_did.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1000.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1001.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1002.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1003.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1004.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1005.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1006.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1007.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1008.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1009.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1010.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1011.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1012.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1013.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1014.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1015.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1016.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1017.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1018.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1019.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\brian.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\default.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\example.com.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\ivr_menus\demo_ivr.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\jingle_profiles\client.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\jingle_profiles\server.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\de\de.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\de\demo\demo.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\de\vm\tts.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\en\en.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\en\demo\demo-ivr.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\en\demo\demo.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\en\dir\sounds.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\en\dir\tts.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\en\vm\sounds.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\en\vm\tts.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\fr\fr.xml

20>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

20>Processing archive:
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20>Extracting  pocketsphinx-0.5.99-20091212.tar

20>Everything is Ok

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\fr\demo\demo.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\fr\dir\sounds.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\fr\dir\tts.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\fr\vm\sounds.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\ru\ru.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\ru\demo\demo-ivr.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\ru\demo\demo.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\ru\dir\sounds.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\ru\dir\tts.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\ru\vm\sounds.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\ru\vm\tts.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\mrcp_profiles\loquendo-7-mrcp-v2.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\mrcp_profiles\nuance-1.0.0-mrcp-v1.xm
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16>C:\FreeSWITCH\freeswitch-1.0.6\conf\mrcp_profiles\nuance-5.0-mrcp-v1.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\mrcp_profiles\nuance-5.0-mrcp-v2.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\mrcp_profiles\unimrcpserver-mrcp-v1.x
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16>C:\FreeSWITCH\freeswitch-1.0.6\conf\mrcp_profiles\voxeo-prophecy-8.0-mrcp
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16>C:\FreeSWITCH\freeswitch-1.0.6\conf\sip_profiles\external.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\sip_profiles\internal-ipv6.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\sip_profiles\internal.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\sip_profiles\external\example.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\sip_profiles\internal\example.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\skinny_profiles\internal.xml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\yaml\extensions.yaml

16>C:\FreeSWITCH\freeswitch-1.0.6\conf\yaml\mod_yaml.yaml

16>135 File(s) copied

16>Build log was saved at
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16>FreeSwitchCoreLib - 0 error(s), 0 warning(s)

20>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

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20>Everything is Ok

21>dtoa.c

21>err.c

21>f2c_lite.c

21>fe_interface.c

21>fe_sigproc.c

21>fe_warp.c

21>fe_warp_affine.c

21>fe_warp_inverse_linear.c

20>Build log was saved at
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20>Download pocketsphinx - 0 error(s), 0 warning(s)

21>fe_warp_piecewise_linear.c

21>feat.c

22>------ Build started: Project: Download FLITE, Configuration: Debug Win32
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22>Downloading Flite.

21>Generating Code...

22>Downloading:
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21>Compiling...

21>filename.c

21>fixlog.c

21>fsg_model.c

21>genrand.c

21>glist.c

21>hash_table.c

21>heap.c

21>info.c

21>jsgf.c

21>jsgf_parser.c

21>jsgf_scanner.c

21>lda.c

21>listelem_alloc.c

21>lm3g_model.c

21>logmath.c

21>matrix.c

21>mmio.c

21>ngram_model.c

21>ngram_model_arpa.c

21>ngram_model_dmp.c

21>Generating Code...

21>Compiling...

21>ngram_model_dmp32.c

21>ngram_model_set.c

21>pio.c

21>play_win32.c

21>profile.c

21>rec_win32.c

21>sbthread.c

21>slamch.c

21>slapack_lite.c

21>strfuncs.c

21>unlimit.c

21>yin.c

21>Generating Code...

21>Compiling manifest to resources...

21>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

21>Copyright (C) Microsoft Corporation.  All rights reserved.

21>Linking...

21>   Creating library .\../../lib/Debug/sphinxbase.lib and object
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21>Embedding manifest...

21>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

21>Copyright (C) Microsoft Corporation.  All rights reserved.

22>Extracting:
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21>Build log was saved at
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21>sphinxbase - 0 error(s), 0 warning(s)

23>------ Build started: Project: Download LAME, Configuration: Debug Win32
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23>Downloading Lame.

23>Downloading: http://files.freeswitch.org/downloads/libs/lame-3.97.tar.gz

23>Extracting: C:\FreeSWITCH\freeswitch-1.0.6\libs\lame-3.97.tar.gz

22>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

22>Processing archive:
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22>Extracting  flite-1.3.99-latest.tar

22>Everything is Ok

23>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

23>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\lame-3.97.tar.gz

23>Extracting  lame-3.97.tar

23>Everything is Ok

23>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

23>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\lame-3.97.tar

23>Extracting  lame-3.97

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22>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

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22>Extracting  flite-1.3.99\palm\pocore\po_StrPrintF.c

22>Extracting  flite-1.3.99\palm\pocore\Makefile

22>Extracting  flite-1.3.99\palm\flop

22>Extracting  flite-1.3.99\palm\flop\flop.def

22>Extracting  flite-1.3.99\palm\flop\flop.rcp

22>Extracting  flite-1.3.99\palm\flop\flop.bmp

22>Extracting  flite-1.3.99\palm\flop\flop.c

22>Extracting  flite-1.3.99\palm\flop\flop.h

22>Extracting  flite-1.3.99\palm\flop\flopsmall.bmp

22>Extracting  flite-1.3.99\palm\flop\Makefile

22>Extracting  flite-1.3.99\palm\m68k_flite

22>Extracting  flite-1.3.99\palm\m68k_flite\m68k_flite.c

22>Extracting  flite-1.3.99\palm\m68k_flite\peal.c

22>Extracting  flite-1.3.99\palm\m68k_flite\fms.c

22>Extracting  flite-1.3.99\palm\m68k_flite\Makefile

22>Extracting  flite-1.3.99\palm\Makefile

22>Extracting  flite-1.3.99\README

22>Extracting  flite-1.3.99\doc

22>Extracting  flite-1.3.99\doc\stuff.ed

22>Extracting  flite-1.3.99\doc\flite.texi

22>Extracting  flite-1.3.99\doc\intro.txt

22>Extracting  flite-1.3.99\doc\alice

22>Extracting  flite-1.3.99\doc\Makefile

22>Extracting  flite-1.3.99\config.guess

22>Extracting  flite-1.3.99\tools

22>Extracting  flite-1.3.99\tools\play_sync.scm

22>Extracting  flite-1.3.99\tools\make_clunits.scm

22>Extracting  flite-1.3.99\tools\make_lts_rewrite.scm

22>Extracting  flite-1.3.99\tools\make_cg.scm

22>Extracting  flite-1.3.99\tools\make_cart.scm

22>Extracting  flite-1.3.99\tools\make_vallist.scm

22>Extracting  flite-1.3.99\tools\make_f0lr.scm

22>Extracting  flite-1.3.99\tools\build_flite

22>Extracting  flite-1.3.99\tools\flite_sort_main.c

22>Extracting  flite-1.3.99\tools\make_didb2.scm

22>Extracting  flite-1.3.99\tools\VOICE_diphone.c

22>Extracting  flite-1.3.99\tools\VOICE_cg.c

22>Extracting  flite-1.3.99\tools\make_voice_list

22>Extracting  flite-1.3.99\tools\find_sts_main.c

22>Extracting  flite-1.3.99\tools\make_lex.scm

22>Extracting  flite-1.3.99\tools\make_lts_wfst.scm

22>Extracting  flite-1.3.99\tools\VOICE_clunits.c

22>Extracting  flite-1.3.99\tools\flite_test

22>Extracting  flite-1.3.99\tools\VOICE_ldom.c

22>Extracting  flite-1.3.99\tools\make_didb.scm

22>Extracting  flite-1.3.99\tools\huff_table

22>Extracting  flite-1.3.99\tools\find_cmimax

22>Extracting  flite-1.3.99\tools\Makefile.flite

22>Extracting  flite-1.3.99\tools\make_lts.scm

22>Extracting  flite-1.3.99\tools\make_phoneset.scm

22>Extracting  flite-1.3.99\tools\setup_flite

22>Extracting  flite-1.3.99\tools\Makefile

22>Extracting  flite-1.3.99\Makefile

22>Everything is Ok

23>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download
LAME.htm"

23>Download LAME - 0 error(s), 0 warning(s)

22>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download
FLITE.htm"

22>Download FLITE - 0 error(s), 0 warning(s)

24>------ Build started: Project: Download LIBSHOUT, Configuration: Debug
Win32 ------

24>Downloading Flite.

24>Downloading:
http://files.freeswitch.org/downloads/libs/libshout-2.2.2.tar.gz

25>------ Build started: Project: js, Configuration: Debug Win32 ------

25>Performing Pre-Build Event...

25>Compiling...

25>e_acos.c

25>e_acosh.c

25>e_asin.c

25>e_atan2.c

25>e_atanh.c

25>e_cosh.c

25>e_exp.c

25>e_fmod.c

25>e_gamma.c

25>e_gamma_r.c

25>e_hypot.c

25>e_j0.c

25>e_j1.c

25>e_jn.c

24>Extracting: C:\FreeSWITCH\freeswitch-1.0.6\libs\libshout-2.2.2.tar.gz

25>e_lgamma.c

24>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

24>Processing archive:
C:\FreeSWITCH\freeswitch-1.0.6\libs\libshout-2.2.2.tar.gz

24>Extracting  libshout-2.2.2.tar

24>Everything is Ok

25>e_lgamma_r.c

25>e_log.c

25>e_log10.c

25>e_pow.c

25>e_rem_pio2.c

25>Generating Code...

25>Compiling...

25>e_remainder.c

25>e_scalb.c

25>e_sinh.c

25>e_sqrt.c

25>jsapi.c

24>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

24>Processing archive:
C:\FreeSWITCH\freeswitch-1.0.6\libs\libshout-2.2.2.tar

24>Extracting  libshout-2.2.2

24>Extracting  libshout-2.2.2\m4

24>Extracting  libshout-2.2.2\m4\vorbis.m4

24>Extracting  libshout-2.2.2\m4\ac_config_libconfig_in.m4

24>Extracting  libshout-2.2.2\m4\xiph_compiler.m4

24>Extracting  libshout-2.2.2\m4\acx_pthread.m4

24>Extracting  libshout-2.2.2\m4\ogg.m4

24>Extracting  libshout-2.2.2\m4\xiph_net.m4

24>Extracting  libshout-2.2.2\m4\speex.m4

24>Extracting  libshout-2.2.2\m4\shout.m4

24>Extracting  libshout-2.2.2\m4\xiph_types.m4

24>Extracting  libshout-2.2.2\m4\theora.m4

24>Extracting  libshout-2.2.2\ltmain.sh

24>Extracting  libshout-2.2.2\src

24>Extracting  libshout-2.2.2\src\Makefile.in

24>Extracting  libshout-2.2.2\src\shout_private.h

24>Extracting  libshout-2.2.2\src\Makefile.am

24>Extracting  libshout-2.2.2\src\shout_ogg.h

24>Extracting  libshout-2.2.2\src\mp3.c

24>Extracting  libshout-2.2.2\src\vorbis.c

24>Extracting  libshout-2.2.2\src\ogg.c

24>Extracting  libshout-2.2.2\src\theora.c

24>Extracting  libshout-2.2.2\src\speex.c

24>Extracting  libshout-2.2.2\src\thread

24>Extracting  libshout-2.2.2\src\thread\Makefile.in

24>Extracting  libshout-2.2.2\src\thread\Makefile.am

24>Extracting  libshout-2.2.2\src\thread\TODO

24>Extracting  libshout-2.2.2\src\thread\thread.c

24>Extracting  libshout-2.2.2\src\thread\BUILDING

24>Extracting  libshout-2.2.2\src\thread\COPYING

24>Extracting  libshout-2.2.2\src\thread\README

24>Extracting  libshout-2.2.2\src\thread\thread.h

24>Extracting  libshout-2.2.2\src\util.c

24>Extracting  libshout-2.2.2\src\shout.c

24>Extracting  libshout-2.2.2\src\net

24>Extracting  libshout-2.2.2\src\net\Makefile.in

24>Extracting  libshout-2.2.2\src\net\resolver.c

24>Extracting  libshout-2.2.2\src\net\Makefile.am

24>Extracting  libshout-2.2.2\src\net\TODO

24>Extracting  libshout-2.2.2\src\net\sock.c

24>Extracting  libshout-2.2.2\src\net\test_resolver.c

24>Extracting  libshout-2.2.2\src\net\resolver.h

24>Extracting  libshout-2.2.2\src\net\BUILDING

24>Extracting  libshout-2.2.2\src\net\COPYING

24>Extracting  libshout-2.2.2\src\net\README

24>Extracting  libshout-2.2.2\src\net\sock.h

24>Extracting  libshout-2.2.2\src\timing

24>Extracting  libshout-2.2.2\src\timing\timing.h

24>Extracting  libshout-2.2.2\src\timing\Makefile.in

24>Extracting  libshout-2.2.2\src\timing\Makefile.am

24>Extracting  libshout-2.2.2\src\timing\TODO

24>Extracting  libshout-2.2.2\src\timing\timing.c

24>Extracting  libshout-2.2.2\src\timing\BUILDING

24>Extracting  libshout-2.2.2\src\timing\COPYING

24>Extracting  libshout-2.2.2\src\timing\README

24>Extracting  libshout-2.2.2\src\util.h

24>Extracting  libshout-2.2.2\src\avl

24>Extracting  libshout-2.2.2\src\avl\Makefile.in

24>Extracting  libshout-2.2.2\src\avl\test.c

24>Extracting  libshout-2.2.2\src\avl\Makefile.am

24>Extracting  libshout-2.2.2\src\avl\TODO

24>Extracting  libshout-2.2.2\src\avl\avl.dsp

24>Extracting  libshout-2.2.2\src\avl\avl.c

24>Extracting  libshout-2.2.2\src\avl\avl.h

24>Extracting  libshout-2.2.2\src\avl\BUILDING

24>Extracting  libshout-2.2.2\src\avl\COPYING

24>Extracting  libshout-2.2.2\src\avl\README

24>Extracting  libshout-2.2.2\src\httpp

24>Extracting  libshout-2.2.2\src\httpp\Makefile.in

24>Extracting  libshout-2.2.2\src\httpp\Makefile.am

24>Extracting  libshout-2.2.2\src\httpp\httpp.h

24>Extracting  libshout-2.2.2\src\httpp\TODO

24>Extracting  libshout-2.2.2\src\httpp\httpp.c

24>Extracting  libshout-2.2.2\src\httpp\COPYING

24>Extracting  libshout-2.2.2\src\httpp\README

24>Extracting  libshout-2.2.2\examples

24>Extracting  libshout-2.2.2\examples\Makefile.in

24>Extracting  libshout-2.2.2\examples\Makefile.am

24>Extracting  libshout-2.2.2\examples\nonblocking.c

24>Extracting  libshout-2.2.2\examples\example.c

24>Extracting  libshout-2.2.2\Makefile.in

24>Extracting  libshout-2.2.2\compile

24>Extracting  libshout-2.2.2\debian

24>Extracting  libshout-2.2.2\debian\rules

24>Extracting  libshout-2.2.2\debian\Makefile.in

24>Extracting  libshout-2.2.2\debian\watch

24>Extracting  libshout-2.2.2\debian\libshout3-dev.examples

24>Extracting  libshout-2.2.2\debian\Makefile.am

24>Extracting  libshout-2.2.2\debian\libshout3.install

24>Extracting  libshout-2.2.2\debian\copyright

24>Extracting  libshout-2.2.2\debian\compat

24>Extracting  libshout-2.2.2\debian\control

24>Extracting  libshout-2.2.2\debian\changelog

24>Extracting  libshout-2.2.2\debian\libshout3-dev.install

24>Extracting  libshout-2.2.2\configure

24>Extracting  libshout-2.2.2\configure.ac

24>Extracting  libshout-2.2.2\Makefile.am

24>Extracting  libshout-2.2.2\aclocal.m4

24>Extracting  libshout-2.2.2\shout-config.in

24>Extracting  libshout-2.2.2\install-sh

24>Extracting  libshout-2.2.2\missing

24>Extracting  libshout-2.2.2\config.h.in

24>Extracting  libshout-2.2.2\NEWS

24>Extracting  libshout-2.2.2\config.guess

24>Extracting  libshout-2.2.2\config.sub

24>Extracting  libshout-2.2.2\doc

24>Extracting  libshout-2.2.2\doc\Makefile.in

24>Extracting  libshout-2.2.2\doc\Makefile.am

24>Extracting  libshout-2.2.2\doc\libshout.xml

24>Extracting  libshout-2.2.2\doc\spec-html.xsl

24>Extracting  libshout-2.2.2\shout.pc.in

24>Extracting  libshout-2.2.2\COPYING

24>Extracting  libshout-2.2.2\include

24>Extracting  libshout-2.2.2\include\Makefile.in

24>Extracting  libshout-2.2.2\include\os.h

24>Extracting  libshout-2.2.2\include\Makefile.am

24>Extracting  libshout-2.2.2\include\shout

24>Extracting  libshout-2.2.2\include\shout\Makefile.in

24>Extracting  libshout-2.2.2\include\shout\Makefile.am

24>Extracting  libshout-2.2.2\include\shout\shout.h.in

24>Extracting  libshout-2.2.2\README

24>Extracting  libshout-2.2.2\INSTALL

24>Extracting  libshout-2.2.2\win32

24>Extracting  libshout-2.2.2\win32\Makefile.in

24>Extracting  libshout-2.2.2\win32\Makefile.am

24>Extracting  libshout-2.2.2\win32\libshout.dsw

24>Extracting  libshout-2.2.2\win32\libshout.dsp

24>Extracting  libshout-2.2.2\depcomp

24>Everything is Ok

25>jsarena.c

25>jsarray.c

25>jsatom.c

25>jsbool.c

25>jscntxt.c

25>jsdate.c

25>jsdbgapi.c

25>jsdhash.c

24>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download
LIBSHOUT.htm"

24>Download LIBSHOUT - 0 error(s), 0 warning(s)

25>jsdso.c

25>jsdtoa.c

26>------ Build started: Project: libogg, Configuration: Debug Win32 ------

26>Compiling...

26>framing.c

25>jsemit.c

26>bitwise.c

26>Generating Code...

26>Creating library...

26>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\libogg\Debug\BuildLog.htm"

26>libogg - 0 error(s), 0 warning(s)

25>jsexn.c

27>------ Build started: Project: Download mpg123, Configuration: Debug
Win32 ------

27>Downloading Flite.

25>jsfile.c

27>Downloading: http://files.freeswitch.org/downloads/libs/mpg123.tar.gz

25>jsfun.c

25>jsgc.c

25>Generating Code...

27>Extracting: C:\FreeSWITCH\freeswitch-1.0.6\libs\mpg123.tar.gz

27>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

27>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\mpg123.tar.gz

27>Extracting  mpg123.tar

27>Everything is Ok

25>Compiling...

25>jshash.c

25>jsinterp.c

25>jslock.c

27>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

27>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\mpg123.tar

27>Extracting  mpg123

27>Extracting  mpg123\build

27>Extracting  mpg123\build\compile

27>Extracting  mpg123\build\config.guess

27>Extracting  mpg123\build\config.sub

27>Extracting  mpg123\build\depcomp

27>Extracting  mpg123\build\install-sh

27>Extracting  mpg123\build\ltmain.sh

27>Extracting  mpg123\build\missing

27>Extracting  mpg123\man1

27>Extracting  mpg123\man1\mpg123.1

27>Extracting  mpg123\ports

27>Extracting  mpg123\ports\MSVC++

27>Extracting  mpg123\ports\MSVC++\INCLUDE

27>Extracting  mpg123\ports\MSVC++\INCLUDE\CORE

27>Extracting  mpg123\ports\MSVC++\INCLUDE\CORE\CORE_FileIn.H

27>Extracting  mpg123\ports\MSVC++\INCLUDE\CORE\SourceFilter_MP3.H

27>Extracting  mpg123\ports\MSVC++\INCLUDE\IIEP_FileIn.H

27>Extracting  mpg123\ports\MSVC++\INCLUDE\IIEP_Def.H

27>Extracting  mpg123\ports\MSVC++\SOURCE

27>Extracting  mpg123\ports\MSVC++\SOURCE\CORE_Log.CPP

27>Extracting  mpg123\ports\MSVC++\SOURCE\CORE_FileIn.CPP

27>Extracting  mpg123\ports\MSVC++\SOURCE\SourceFilter_MP3Stream.CPP

27>Extracting  mpg123\ports\MSVC++\SOURCE\CORE_Mutex.CPP

27>Extracting  mpg123\ports\MSVC++\libMPG123

27>Extracting  mpg123\ports\MSVC++\libMPG123\libMPG123.vcproj

27>Extracting  mpg123\ports\MSVC++\libMPG123\PLACE_LIBMPG123_SOURCES_HERE

27>Extracting  mpg123\ports\MSVC++\README

27>Extracting  mpg123\ports\Sony_PSP

27>Extracting  mpg123\ports\Sony_PSP\config.h

27>Extracting  mpg123\ports\Sony_PSP\README

27>Extracting  mpg123\ports\Sony_PSP\Makefile.psp

27>Extracting  mpg123\ports\Sony_PSP\readers.c.patch

27>Extracting  mpg123\ports\README

27>Extracting  mpg123\ports\mpg123_.pas

27>Extracting  mpg123\src

27>Extracting  mpg123\src\libmpg123

27>Extracting  mpg123\src\libmpg123\compat.c

27>Extracting  mpg123\src\libmpg123\compat.h

27>Extracting  mpg123\src\libmpg123\Makefile.am

27>Extracting  mpg123\src\libmpg123\Makefile.in

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27>Extracting  mpg123\src\libmpg123\parse.c

27>Extracting  mpg123\src\libmpg123\parse.h

27>Extracting  mpg123\src\libmpg123\frame.c

27>Extracting  mpg123\src\libmpg123\format.c

27>Extracting  mpg123\src\libmpg123\frame.h

27>Extracting  mpg123\src\libmpg123\reader.h

27>Extracting  mpg123\src\libmpg123\debug.h

27>Extracting  mpg123\src\libmpg123\decode.h

27>Extracting  mpg123\src\libmpg123\decode_2to1.c

27>Extracting  mpg123\src\libmpg123\decode_4to1.c

27>Extracting  mpg123\src\libmpg123\decode_ntom.c

27>Extracting  mpg123\src\libmpg123\equalizer.c

27>Extracting  mpg123\src\libmpg123\huffman.h

27>Extracting  mpg123\src\libmpg123\icy.c

27>Extracting  mpg123\src\libmpg123\icy.h

27>Extracting  mpg123\src\libmpg123\icy2utf8.c

27>Extracting  mpg123\src\libmpg123\icy2utf8.h

27>Extracting  mpg123\src\libmpg123\id3.c

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27>Extracting  mpg123\src\libmpg123\layer1.c

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27>Extracting  mpg123\src\libmpg123\getbits.h

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27>Extracting  mpg123\src\libmpg123\optimize.c

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27>Extracting  mpg123\src\libmpg123\libmpg123.c

27>Extracting  mpg123\src\libmpg123\mpg123lib_intern.h

27>Extracting  mpg123\src\libmpg123\mangle.h

27>Extracting  mpg123\src\libmpg123\getcpuflags.h

27>Extracting  mpg123\src\libmpg123\libmpg123.sym

27>Extracting  mpg123\src\libmpg123\dct36_3dnowext.S

27>Extracting  mpg123\src\libmpg123\dct36_3dnow.S

27>Extracting  mpg123\src\libmpg123\dct64_3dnowext.S

27>Extracting  mpg123\src\libmpg123\dct64_3dnow.S

27>Extracting  mpg123\src\libmpg123\dct64_altivec.c

27>Extracting  mpg123\src\libmpg123\dct64.c

27>Extracting  mpg123\src\libmpg123\dct64_i386.c

27>Extracting  mpg123\src\libmpg123\dct64_i486.c

27>Extracting  mpg123\src\libmpg123\dct64_mmx.S

27>Extracting  mpg123\src\libmpg123\dct64_sse.S

27>Extracting  mpg123\src\libmpg123\decode_3dnowext.S

27>Extracting  mpg123\src\libmpg123\decode_3dnow.S

27>Extracting  mpg123\src\libmpg123\decode_altivec.c

27>Extracting  mpg123\src\libmpg123\decode.c

25>jslog2.c

25>jslong.c

25>jsmath.c

25>jsnum.c

25>jsobj.c

27>Extracting  mpg123\src\libmpg123\decode_i386.c

27>Extracting  mpg123\src\libmpg123\decode_i486.c

27>Extracting  mpg123\src\libmpg123\decode_i586_dither.S

27>Extracting  mpg123\src\libmpg123\decode_i586.S

27>Extracting  mpg123\src\libmpg123\decode_mmx.S

27>Extracting  mpg123\src\libmpg123\decode_sse3d.h

27>Extracting  mpg123\src\libmpg123\decode_sse.S

27>Extracting  mpg123\src\libmpg123\equalizer_3dnow.S

27>Extracting  mpg123\src\libmpg123\tabinit_mmx.S

27>Extracting  mpg123\src\libmpg123\getcpuflags.S

27>Extracting  mpg123\src\libmpg123\testcpu.c

27>Extracting  mpg123\src\libmpg123\dnoise.sh

27>Extracting  mpg123\src\libmpg123\dnoise.dat

27>Extracting  mpg123\src\Makefile.am

27>Extracting  mpg123\src\Makefile.in

27>Extracting  mpg123\src\config.h.in

27>Extracting  mpg123\src\audio.c

27>Extracting  mpg123\src\audio.h

27>Extracting  mpg123\src\buffer.c

27>Extracting  mpg123\src\buffer.h

27>Extracting  mpg123\src\common.c

27>Extracting  mpg123\src\common.h

27>Extracting  mpg123\src\control_generic.c

27>Extracting  mpg123\src\getlopt.c

27>Extracting  mpg123\src\getlopt.h

27>Extracting  mpg123\src\httpget.c

27>Extracting  mpg123\src\httpget.h

27>Extracting  mpg123\src\resolver.c

27>Extracting  mpg123\src\resolver.h

27>Extracting  mpg123\src\genre.h

27>Extracting  mpg123\src\genre.c

27>Extracting  mpg123\src\module.h

27>Extracting  mpg123\src\mpg123.c

27>Extracting  mpg123\src\mpg123app.h

27>Extracting  mpg123\src\metaprint.c

27>Extracting  mpg123\src\metaprint.h

27>Extracting  mpg123\src\playlist.c

27>Extracting  mpg123\src\playlist.h

27>Extracting  mpg123\src\sfifo.c

27>Extracting  mpg123\src\sfifo.h

27>Extracting  mpg123\src\term.c

27>Extracting  mpg123\src\term.h

27>Extracting  mpg123\src\wav.c

27>Extracting  mpg123\src\xfermem.c

27>Extracting  mpg123\src\xfermem.h

27>Extracting  mpg123\src\Makefile.legacy

27>Extracting  mpg123\src\config.h.legacy

27>Extracting  mpg123\src\legacy_module.c

27>Extracting  mpg123\src\module.c

27>Extracting  mpg123\src\output

27>Extracting  mpg123\src\output\Makefile.am

27>Extracting  mpg123\src\output\Makefile.in

27>Extracting  mpg123\src\output\aix.c

27>Extracting  mpg123\src\output\alib.c

27>Extracting  mpg123\src\output\alsa.c

27>Extracting  mpg123\src\output\arts.c

27>Extracting  mpg123\src\output\coreaudio.c

27>Extracting  mpg123\src\output\dummy.c

27>Extracting  mpg123\src\output\esd.c

27>Extracting  mpg123\src\output\hp.c

27>Extracting  mpg123\src\output\jack.c

27>Extracting  mpg123\src\output\mint.c

27>Extracting  mpg123\src\output\nas.c

27>Extracting  mpg123\src\output\os2.c

27>Extracting  mpg123\src\output\oss.c

27>Extracting  mpg123\src\output\portaudio.c

27>Extracting  mpg123\src\output\pulse.c

27>Extracting  mpg123\src\output\sdl.c

27>Extracting  mpg123\src\output\sgi.c

27>Extracting  mpg123\src\output\sun.c

27>Extracting  mpg123\src\output\win32.c

27>Extracting  mpg123\test

27>Extracting  mpg123\test\forkfaint.c

27>Extracting  mpg123\test\rms16.c

27>Extracting  mpg123\xmms2-plugin

27>Extracting  mpg123\xmms2-plugin\mpg123

27>Extracting  mpg123\xmms2-plugin\mpg123\mpg123.c

27>Extracting  mpg123\xmms2-plugin\mpg123\wscript

27>Extracting  mpg123\xmms2-plugin\README

27>Extracting  mpg123\README

27>Extracting  mpg123\configure.ac

27>Extracting  mpg123\aclocal.m4

27>Extracting  mpg123\Makefile.am

27>Extracting  mpg123\Makefile.in

27>Extracting  mpg123\libmpg123.pc.in

27>Extracting  mpg123\configure

27>Extracting  mpg123\AUTHORS

27>Extracting  mpg123\COPYING

27>Extracting  mpg123\ChangeLog

27>Extracting  mpg123\INSTALL

27>Extracting  mpg123\NEWS

27>Extracting  mpg123\TODO

27>Extracting  mpg123\MakeLegacy.sh

27>Extracting  mpg123\mpg123.spec.in

27>Extracting  mpg123\mpg123.spec

27>Extracting  mpg123\makedll.sh

27>Extracting  mpg123\NEWS.libmpg123

27>Extracting  mpg123\autogen.sh

27>Extracting  mpg123\libltdl

27>Extracting  mpg123\libltdl\README

27>Extracting  mpg123\libltdl\acinclude.m4

27>Extracting  mpg123\libltdl\configure.ac

27>Extracting  mpg123\libltdl\aclocal.m4

27>Extracting  mpg123\libltdl\ltdl.h

27>Extracting  mpg123\libltdl\Makefile.am

27>Extracting  mpg123\libltdl\Makefile.in

27>Extracting  mpg123\libltdl\config-h.in

27>Extracting  mpg123\libltdl\configure

27>Extracting  mpg123\libltdl\COPYING.LIB

27>Extracting  mpg123\libltdl\config.guess

27>Extracting  mpg123\libltdl\config.sub

27>Extracting  mpg123\libltdl\install-sh

27>Extracting  mpg123\libltdl\ltmain.sh

27>Extracting  mpg123\libltdl\missing

27>Extracting  mpg123\libltdl\ltdl.c

27>Extracting  mpg123\doc

27>Extracting  mpg123\doc\examples

27>Extracting  mpg123\doc\examples\mpg123_to_wav.c

27>Extracting  mpg123\doc\examples\scan.c

27>Extracting  mpg123\doc\examples\mpglib.c

27>Extracting  mpg123\doc\examples\id3dump.c

27>Extracting  mpg123\doc\examples\Makefile

27>Extracting  mpg123\doc\Makefile.am

27>Extracting  mpg123\doc\Makefile.in

27>Extracting  mpg123\doc\THANKS

27>Extracting  mpg123\doc\TODO

27>Extracting  mpg123\doc\BENCHMARKING

27>Extracting  mpg123\doc\BUGS

27>Extracting  mpg123\doc\CONTACT

27>Extracting  mpg123\doc\PATENTS

27>Extracting  mpg123\doc\README.3DNOW

27>Extracting  mpg123\doc\README.WIN32

27>Extracting  mpg123\doc\README.gain

27>Extracting  mpg123\doc\README.remote

27>Extracting  mpg123\doc\ROAD_TO_LGPL

27>Extracting  mpg123\doc\LICENSE

27>Extracting  mpg123\doc\ACCURACY

27>Extracting  mpg123\doc\libmpg123_speed.txt

27>Extracting  mpg123\doc\doxyhead.xhtml

27>Extracting  mpg123\doc\doxy_examples.c

27>Extracting  mpg123\doc\doxygen.conf

27>Everything is Ok

25>jsopcode.c

25>jsparse.c

25>jsprf.c

25>jsregexp.c

25>jsscan.c

25>jsscope.c

25>jsscript.c

25>jsstr.c

25>jsutil.c

27>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download
mpg123.htm"

27>Download mpg123 - 0 error(s), 0 warning(s)

28>------ Build started: Project: libmpg123, Configuration: Debug Win32
------

28>Compiling...

28>dct64.c

25>jsxdrapi.c

28>decode.c

25>jsxml.c

25>k_cos.c

28>decode_2to1.c

25>Generating Code...

28>decode_4to1.c

28>decode_ntom.c

28>equalizer.c

28>format.c

28>frame.c

25>Compiling...

25>k_rem_pio2.c

25>k_sin.c

25>k_standard.c

28>icy.c

25>k_tan.c

28>icy2utf8.c

25>prmjtime.c

28>id3.c

28>layer1.c

25>s_asinh.c

25>s_atan.c

25>s_cbrt.c

28>layer2.c

25>s_ceil.c

25>s_copysign.c

25>s_cos.c

28>layer3.c

25>s_erf.c

25>s_expm1.c

25>s_fabs.c

25>s_finite.c

28>libmpg123.c

25>s_floor.c

25>s_frexp.c

25>s_ilogb.c

25>s_isnan.c

25>s_ldexp.c

28>optimize.c

25>Generating Code...

28>parse.c

25>Compiling...

25>s_lib_version.c

25>s_log1p.c

25>s_logb.c

25>s_matherr.c

25>s_modf.c

25>s_nextafter.c

25>s_rint.c

25>s_scalbn.c

25>s_signgam.c

25>s_significand.c

25>s_sin.c

28>readers.c

25>s_tan.c

25>s_tanh.c

25>w_acos.c

28>stringbuf.c

25>w_acosh.c

25>w_asin.c

25>w_atan2.c

28>tabinit.c

25>w_atanh.c

25>w_cosh.c

25>w_exp.c

28>Generating Code...

25>Generating Code...

28>Compiling...

28>compat.c

25>Compiling...

25>w_fmod.c

28>Generating Code...

25>w_gamma.c

25>w_gamma_r.c

28>Creating library...

25>w_hypot.c

25>w_j0.c

25>w_j1.c

25>w_jn.c

25>w_lgamma.c

25>w_lgamma_r.c

25>w_log.c

25>w_log10.c

25>w_pow.c

25>w_remainder.c

25>w_scalb.c

25>w_sinh.c

25>w_sqrt.c

25>ntinrval.c

28>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\mpg123\Debug\BuildLog.htm"

28>libmpg123 - 0 error(s), 0 warning(s)

29>------ Build started: Project: libdingaling, Configuration: Debug Win32
------

29>Compiling...

29>sha1.c

25>ntio.c

29>libdingaling.c

25>ntmisc.c

29>Generating Code...

29>Creating library...

29>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\libdingaling\Debug\BuildLog.htm"

29>libdingaling - 0 error(s), 0 warning(s)

30>------ Skipped Build: Project: Download 16khz music, Configuration: Debug
Win32 ------

30>Project not selected to build for this solution configuration 

31>------ Build started: Project: Download 8khz music, Configuration: Debug
Win32 ------

31>Downloading 8khzsound.

25>ntsec.c

31>Downloading:
http://files.freeswitch.org/freeswitch-sounds-music-8000-1.0.8.tar.gz

25>Generating Code...

25>Compiling...

25>ntthread.c

25>pratom.c

25>prcthr.c

25>prdir.c

25>prerror.c

25>prfdcach.c

25>prfile.c

25>prinit.c

25>prinrval.c

31>Extracting:
C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\freeswitch-sounds-music-8000-1.0.
8.tar.gz

25>prio.c

25>priometh.c

31>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

31>Processing archive:
C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\freeswitch-sounds-music-8000-1.0.
8.tar.gz

31>Extracting  freeswitch-sounds-music-8000-1.0.8.tar

31>Everything is Ok

31>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

31>Processing archive:
C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\freeswitch-sounds-music-8000-1.0.
8.tar

31>Extracting  music\8000

31>Extracting  music\8000\partita-no-3-in-e-major-bwv-1006-1-preludio.wav

31>Extracting  music\8000\ponce-preludio-in-e-major.wav

31>Extracting  music\8000\suite-espanola-op-47-leyenda.wav

31>Extracting  music\8000\danza-espanola-op-37-h-142-xii-arabesca.wav

31>Everything is Ok

25>prlayer.c

25>prlog.c

25>prmem.c

31>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download
8khz music.htm"

31>Download 8khz music - 0 error(s), 0 warning(s)

32>------ Build started: Project: libsndfile, Configuration: Debug Win32
------

32>Compiling...

25>prmmap.c

32>alaw.c

32>au.c

25>prmwait.c

32>audio_detect.c

32>avr.c

32>broadcast.c

25>prolock.c

25>prosdep.c

32>caf.c

32>chunk.c

25>prprf.c

32>command.c

32>common.c

32>dither.c

25>prseg.c

32>double64.c

32>dwd.c

25>Generating Code...

32>dwvw.c

25>Compiling...

25>prtime.c

32>file_io.c

25>prtpd.c

32>flac.c

25>prucpu.c

32>float32.c

25>prucv.c

32>gsm610.c

32>htk.c

32>ima_adpcm.c

25>prulock.c

32>ima_oki_adpcm.c

32>Generating Code...

25>prustack.c

32>Compiling...

32>ircam.c

25>pruthr.c

32>mat4.c

32>mat5.c

25>w32poll.c

32>mpc2k.c

32>ms_adpcm.c

32>nist.c

25>win32_errors.c

32>ogg.c

25>Generating Code...

32>paf.c

25>Linking...

32>pcm.c

25>   Creating library .\Debug/js32.lib and object .\Debug/js32.exp

32>pvf.c

32>raw.c

32>rf64.c

32>rx2.c

32>sd2.c

25>Embedding manifest...

32>sds.c

32>sndfile.c

32>strings.c

32>svx.c

25>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\js\Debug\BuildLog.htm"

25>js - 0 error(s), 0 warning(s)

33>------ Skipped Build: Project: Download 32khz music, Configuration: Debug
Win32 ------

33>Project not selected to build for this solution configuration 

34>------ Build started: Project: lua51, Configuration: Debug Win32 ------

32>txw.c

34>Compiling...

34>lzio.c

32>ulaw.c

32>Generating Code...

34>lvm.c

32>Compiling...

32>voc.c

34>lundump.c

34>ltm.c

32>vox_adpcm.c

34>ltablib.c

32>w64.c

34>ltable.c

32>wav.c

34>lstrlib.c

34>lstring.c

32>wav_w64.c

34>lstate.c

32>wve.c

34>lparser.c

32>xi.c

32>add.c

34>loslib.c

32>code.c

34>lopcodes.c

32>decode.c

34>lobject.c

32>gsm_create.c

32>gsm_decode.c

34>loadlib.c

32>gsm_destroy.c

32>gsm_encode.c

32>gsm_option.c

32>long_term.c

32>lpc.c

32>preprocess.c

32>rpe.c

32>short_term.c

34>lmem.c

32>Generating Code...

34>lmathlib.c

34>llex.c

32>Compiling...

34>liolib.c

32>table.c

32>g721.c

34>linit.c

34>lgc.c

32>g723_16.c

32>g723_24.c

32>g723_40.c

32>g72x.c

32>aiff.c

34>Generating Code...

32>Generating Code...

34>Compiling...

34>lfunc.c

32>Compiling...

34>ldump.c

32>g72x.c

34>ldo.c

32>Creating library...

34>ldebug.c

32>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\libsndfile\Debug\BuildLog.
htm"

32>libsndfile - 0 error(s), 0 warning(s)

34>ldblib.c

34>lcode.c

34>lbaselib.c

34>lauxlib.c

34>lapi.c

35>------ Build started: Project: libshout, Configuration: Debug Win32
------

34>Generating Code...

35>Compiling...

35>httpp.c

34>Linking...

35>mp3.c

34>   Creating library .\Debug/lua5.1.lib and object .\Debug/lua5.1.exp

35>ogg.c

34>Embedding manifest...

35>resolver.c

34>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_lua\lua\Debug\B
uildLog.htm"

34>lua51 - 0 error(s), 0 warning(s)

36>------ Build started: Project: libmp3lame, Configuration: Debug Win32
------

36>Compiling...

36>bitstream.c

35>shout.c

36>version.c

35>sock.c

36>VbrTag.c

36>vbrquantize.c

35>thread.c

36>util.c

35>timing.c

36>takehiro.c

35>util.c

36>set_get.c

35>avl.c

36>reservoir.c

35>Generating Code...

35>Creating library...

36>quantize_pvt.c

35>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\libshout\Debug\BuildLog.ht
m"

35>libshout - 0 error(s), 0 warning(s)

36>quantize.c

36>psymodel.c

36>presets.c

37>------ Build started: Project: libg722_1, Configuration: Debug Win32
------

37>Compiling...

37>bitstream.c

36>newmdct.c

36>mpglib_interface.c

36>lame.c

36>id3tag.c

37>coef2sam.c

36>gain_analysis.c

36>fft.c

37>common.c

36>encoder.c

36>Generating Code...

37>commonf.c

36>Compiling...

36>tables.c

37>dct4.c

36>Creating library...

36>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\libmp3lame\Debug\BuildLog.
htm"

36>libmp3lame - 0 error(s), 0 warning(s)

38>------ Build started: Project: libsofia_sip_ua_static, Configuration:
Debug Win32 ------

38>Performing Pre-Build Event...

37>dct4_a.c

37>dct4_s.c

38>Downloading: http://files.freeswitch.org/downloads/win32/gawk.exe

37>decoder.c

37>decoderf.c

38>multipart mismatch with Recursive multipart ()

37>encoder.c

37>encoderf.c

38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""

38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""

38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""

38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""

38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""

38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""

38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""

38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""

38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""

38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""

38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""

38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""

38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""

38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""

37>huff_tab.c

38>NOTE:

38>NOTE: Remember to install pthreadVC2.dll to your path, too!

38>NOTE:

38>Compiling...

38>inet_pton.c

37>sam2coef.c

38>smoothsort.c

38>string0.c

38>su.c

37>tables.c

38>su_addrinfo.c

37>basop32.c

37>Generating Code...

38>su_alloc.c

37>Creating library...

38>su_alloc_lock.c

37>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\libg722_1\Debug\BuildLog.h
tm"

37>libg722_1 - 0 error(s), 0 warning(s)

39>------ Build started: Project: esl, Configuration: Debug Win32 ------

38>su_base_port.c

39>Compiling...

39>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

39>esl_config.c

39>esl_event.c

38>su_bm.c

39>esl_threadmutex.c

38>su_default_log.c

38>su_errno.c

39>esl.c

39>Generating Code...

39>Creating library...

38>su_global_log.c

39>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\esl\src\Debug\BuildLog.htm"

38>su_localinfo.c

39>esl - 0 error(s), 1 warning(s)

38>su_log.c

38>su_md5.c

38>su_os_nw.c

40>------ Build started: Project: libilbc, Configuration: Debug Win32 ------

40>Compiling...

40>constants.c

38>su_port.c

38>su_pthread_port.c

40>createCB.c

38>su_root.c

40>doCPLC.c

38>su_socket_port.c

40>enhancer.c

38>Generating Code...

40>filter.c

38>Compiling...

38>su_sprintf.c

38>su_strdup.c

38>su_string.c

40>FrameClassify.c

38>su_strlst.c

38>su_tag.c

38>su_tag_io.c

40>gainquant.c

38>su_taglist.c

40>getCBvec.c

38>su_time.c

40>helpfun.c

38>su_time0.c

40>hpInput.c

38>su_timer.c

40>hpOutput.c

38>su_uniqueid.c

40>iCBConstruct.c

38>su_vector.c

38>su_wait.c

40>iCBSearch.c

38>su_win32_port.c

40>iLBC_decode.c

38>base64.c

38>rc4.c

38>token64.c

38>url.c

40>iLBC_encode.c

38>url_tag.c

40>LPCdecode.c

38>url_tag_ref.c

38>Generating Code...

40>LPCencode.c

38>Compiling...

40>lsf.c

38>features.c

40>packing.c

38>bnf.c

38>msg.c

40>StateConstructW.c

38>msg_auth.c

38>msg_basic.c

40>Generating Code...

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065:
'msg_error_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065:
'msg_error_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065:
'msg_error_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2065:
'msg_unknown_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2065:
'msg_unknown_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2065:
'msg_unknown_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : error C2065:
'msg_payload_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : error C2065:
'msg_separator_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>msg_date.c

40>Compiling...

40>StateSearchW.c

38>msg_generic.c

38>msg_header_copy.c

40>syntFilter.c

38>msg_header_make.c

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_header_make.c(71) : error C2065:
'msg_payload_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_header_make.c(72) : error C2065:
'msg_separator_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_header_make.c(73) : error C2065:
'msg_error_hash' : undeclared identifier

40>anaFilter.c

38>msg_mclass.c

40>Generating Code...

40>Creating library...

38>msg_mime.c

40>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\ilbc\Debug\BuildLog.htm"

40>libilbc - 0 error(s), 0 warning(s)

41>------ Build started: Project: libudns, Configuration: Debug Win32 ------

41>Compiling...

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(246) : error C2065:
'msg_multipart_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(246) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(246) : warning C4024:
'msg_header_alloc' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(250) : warning C4013:
'msg_content_type_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(250) : warning C4047: '='
: 'msg_content_type_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(387) : error C2065:
'msg_multipart_mclass' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(387) : warning C4047: '='
: 'const msg_mclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(435) : error C2065:
'msg_multipart_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(435) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(435) : warning C4024:
'msg_header_alloc' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(454) : error C2065:
'msg_payload_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(454) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

41>udns_bl.c

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(454) : warning C4024:
'msg_header_alloc' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(635) : warning C4013:
'msg_payload_format' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(636) : warning C4047: '='
: 'msg_payload_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(655) : warning C4013:
'msg_separator_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(655) : warning C4047: '='
: 'msg_separator_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(830) : warning C4013:
'msg_is_multipart' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(832) : warning C4013:
'msg_payload_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2065:
'msg_accept_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2065:
'msg_accept_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2065:
'msg_accept_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2065:
'msg_accept_dup_xtra' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2065:
'msg_accept_dup_one' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2065:
'msg_accept_update' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'char [7]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : warning C4047:
'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'size_t'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1095) : warning C4013:
'msg_is_accept' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : error C2065:
'msg_accept_charset_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : error C2065:
'msg_accept_charset_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : error C2065:
'msg_accept_charset_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [15]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1287) : warning C4013:
'msg_is_accept_charset' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : error C2065:
'msg_accept_encoding_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : error C2065:
'msg_accept_encoding_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : error C2065:
'msg_accept_encoding_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [16]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : error C2065:
'msg_accept_language_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : error C2065:
'msg_accept_language_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : error C2065:
'msg_accept_language_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [16]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1389) : warning C4013:
'msg_is_accept_language' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2065:
'msg_content_disposition_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2065:
'msg_content_disposition_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2065:
'msg_content_disposition_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2065:
'msg_content_disposition_dup_xtra' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2065:
'msg_content_disposition_dup_one' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2065:
'msg_content_disposition_update' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'char [20]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : warning C4047:
'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'size_t'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1477) : warning C4013:
'msg_is_content_disposition' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : error C2065:
'msg_content_encoding_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : error C2065:
'msg_content_encoding_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : error C2065:
'msg_content_encoding_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [17]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [2]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1577) : warning C4013:
'msg_is_content_encoding' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : error C2065:
'msg_content_language_hash' : undeclared identifier

41>udns_codes.c

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : error C2099:
initializer is not a constant

41>udns_dn.c

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : error C2065:
'msg_content_language_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : error C2065:
'msg_content_language_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [17]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1632) : warning C4013:
'msg_is_content_language' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : error C2065:
'msg_content_length_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [15]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [2]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : error C2065:
'msg_content_md5_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [12]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : error C2065:
'msg_content_id_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [11]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2065:
'msg_content_type_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2065:
'msg_content_type_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2065:
'msg_content_type_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2065:
'msg_content_type_dup_xtra' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2065:
'msg_content_type_dup_one' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'void *'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char [13]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : warning C4047:
'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [2]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'size_t'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1844) : warning C4013:
'msg_is_content_type' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : error C2065:
'msg_mime_version_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : error C2065:
'msg_mime_version_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : error C2065:
'msg_mime_version_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [13]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1924) : warning C4013:
'msg_is_mime_version' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : error C2065:
'msg_content_location_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [17]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : error C2065:
'msg_content_transfer_encoding_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [26]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : error C2065:
'msg_warning_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [8]'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>msg_mime_table.c

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime_table.c(6) : error C2143:
syntax error : missing '{' before '<cv-qualifer>'

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime_table.c(6) : error C2059:
syntax error : '<Unknown>'

41>udns_dntosp.c

38>msg_parser.c

41>udns_misc.c

41>udns_parse.c

41>udns_resolver.c

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_parser.c(127) : error C2065:
'msg_request_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_parser.c(132) : error C2065:
'msg_status_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_parser.c(1051) : error C2065:
'msg_unknown_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_parser.c(2151) : warning C4013:
'msg_is_payload' undefined; assuming extern returning int

38>msg_parser_util.c

41>udns_rr_a.c

41>udns_rr_mx.c

41>udns_rr_naptr.c

41>udns_rr_ptr.c

41>udns_rr_srv.c

41>udns_rr_txt.c

38>msg_tag.c

41>inet_pton.c

41>Generating Code...

41>Creating library...

38>memcspn.c

38>memmem.c

41>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\udns\Debug\BuildLog.htm"

38>memspn.c

41>libudns - 0 error(s), 0 warning(s)

38>strcasestr.c

38>strtoull.c

38>Generating Code...

42>------ Build started: Project: flite, Configuration: Debug Win32 ------

42>Compiling...

42>au_none.c

38>Compiling...

38>sip_basic.c

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2065:
'sip_request_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2065:
'sip_request_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2065:
'sip_request_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f
(__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'msg_dup_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2065:
'sip_status_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2065:
'sip_status_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2065:
'sip_status_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f
(__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'msg_dup_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(321) : warning C4013:
'sip_is_status' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : error C2065:
'sip_payload_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(459) : warning C4013:
'sip_header_data' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(459) : warning C4047:
'initializing' : 'char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : error C2065:
'sip_separator_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2065:
'sip_unknown_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2065:
'sip_unknown_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2065:
'sip_unknown_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(564) : warning C4013:
'msg_unknown_d' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(569) : warning C4013:
'msg_unknown_e' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2065:
'sip_error_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2065:
'sip_error_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2065:
'sip_error_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(962) : warning C4047:
'initializing' : 'char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2065:
'sip_call_id_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2065:
'sip_call_id_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2065:
'sip_call_id_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f
(__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'msg_dup_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [8]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [2]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2065:
'sip_cseq_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2065:
'sip_cseq_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2065:
'sip_cseq_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f
(__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'msg_dup_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [5]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1224) : warning C4013:
'sip_is_cseq' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1374) : error C2065:
'sip_contact_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1374) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1375) : error C2065:
'sip_contact_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1375) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1376) : error C2065:
'sip_contact_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1376) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1377) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f
(__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1378) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'msg_dup_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1380) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [8]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1381) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1382) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [2]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1383) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1414) : warning C4013:
'sip_is_contact' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2065:
'sip_content_length_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2065:
'sip_content_length_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2065:
'sip_content_length_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [15]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [2]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1546) : warning C4013:
'sip_is_content_length' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2065:
'sip_date_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2065:
'sip_date_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2065:
'sip_date_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [5]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2065:
'sip_expires_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2065:
'sip_expires_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2065:
'sip_expires_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [8]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2065:
'sip_from_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2065:
'sip_from_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2065:
'sip_from_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const sip_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [5]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [2]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1798) : warning C4013:
'sip_is_from' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2065:
'sip_max_forwards_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2065:
'sip_max_forwards_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2065:
'sip_max_forwards_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [13]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1909) : warning C4013:
'sip_is_max_forwards' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2065:
'sip_min_expires_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2065:
'sip_min_expires_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2065:
'sip_min_expires_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [12]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1955) : warning C4013:
'sip_is_min_expires' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2065:
'sip_retry_after_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2065:
'sip_retry_after_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2065:
'sip_retry_after_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f
(__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'msg_dup_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [12]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2196) : warning C4047:
'=' : 'char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2065:
'sip_route_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2065:
'sip_route_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2065:
'sip_route_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const sip_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [6]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2264) : warning C4013:
'sip_is_route' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2065:
'sip_record_route_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2065:
'sip_record_route_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2065:
'sip_record_route_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const sip_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [13]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2338) : warning C4013:
'sip_is_record_route' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2401) : error C2065:
'sip_to_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2401) : fatal error
C1003: error count exceeds 100; stopping compilation

38>sip_caller_prefs.c

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error
C2065: 'sip_request_disposition_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error
C2099: initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error
C2065: 'sip_request_disposition_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error
C2099: initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error
C2065: 'sip_request_disposition_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error
C2099: initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning
C4047: 'initializing' : 'int' differs in levels of indirection from
'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning
C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of
indirection from 'msg_dup_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning
C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of
indirection from 'char [20]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning
C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of
indirection from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning
C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of
indirection from 'char [2]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning
C4047: 'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(116) : warning
C4013: 'sip_is_request_disposition' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error
C2065: 'sip_accept_contact_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error
C2099: initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error
C2065: 'sip_accept_contact_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error
C2099: initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error
C2065: 'sip_accept_contact_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error
C2099: initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning
C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const sip_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning
C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of
indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning
C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of
indirection from 'char [15]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning
C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of
indirection from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning
C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of
indirection from 'char [2]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning
C4047: 'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error
C2065: 'sip_reject_contact_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error
C2099: initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error
C2065: 'sip_reject_contact_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error
C2099: initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error
C2065: 'sip_reject_contact_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error
C2099: initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning
C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const sip_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning
C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of
indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning
C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of
indirection from 'char [15]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning
C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of
indirection from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning
C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of
indirection from 'char [2]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning
C4047: 'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>sip_event.c

42>au_streaming.c

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2065:
'sip_event_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2065:
'sip_event_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2065:
'sip_event_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f
(__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'msg_dup_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [6]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [2]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(120) : warning C4013:
'sip_is_event' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2065:
'sip_allow_events_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2065:
'sip_allow_events_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2065:
'sip_allow_events_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [13]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [2]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(219) : warning C4013:
'sip_is_allow_events' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2065:
'sip_subscription_state_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2065:
'sip_subscription_state_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2065:
'sip_subscription_state_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f
(__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'msg_dup_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [19]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(331) : warning C4013:
'sip_is_subscription_state' undefined; assuming extern returning int

38>sip_extra.c

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_extra.c(44) : fatal error C1083:
Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory

38>sip_feature.c

42>au_wince.c

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2065:
'sip_allow_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2065:
'sip_allow_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2065:
'sip_allow_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [6]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(105) : warning C4013:
'sip_is_allow' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2065:
'sip_proxy_require_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2065:
'sip_proxy_require_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2065:
'sip_proxy_require_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [14]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(192) : warning C4013:
'sip_is_proxy_require' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2065:
'sip_require_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2065:
'sip_require_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2065:
'sip_require_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [8]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(240) : warning C4013:
'sip_is_require' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2065:
'sip_supported_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2065:
'sip_supported_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2065:
'sip_supported_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [10]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [2]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(289) : warning C4013:
'sip_is_supported' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2065:
'sip_unsupported_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2065:
'sip_unsupported_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2065:
'sip_unsupported_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [12]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(336) : warning C4013:
'sip_is_unsupported' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(450) : warning C4013:
'sip_unsupported_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(450) : warning C4047:
'=' : 'sip_unsupported_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2065:
'sip_path_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2065:
'sip_path_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2065:
'sip_path_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const sip_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [5]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(538) : warning C4013:
'sip_is_path' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2065:
'sip_service_route_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2065:
'sip_service_route_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2065:
'sip_service_route_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const sip_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [14]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(594) : warning C4013:
'sip_is_service_route' undefined; assuming extern returning int

38>sip_header.c

38>sip_mime.c

42>audio.c

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065:
'sip_accept_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065:
'sip_accept_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065:
'sip_accept_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065:
'msg_accept_dup_xtra' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065:
'msg_accept_dup_one' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065:
'msg_accept_update' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'char [7]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047:
'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'size_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(105) : warning C4013:
'msg_accept_d' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(110) : warning C4013:
'msg_accept_e' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2065:
'sip_accept_encoding_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2065:
'sip_accept_encoding_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2065:
'sip_accept_encoding_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2065:
'msg_accept_any_update' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047:
'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection
from 'char [16]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'size_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(220) : warning C4013:
'msg_accept_encoding_d' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(234) : warning C4013:
'msg_accept_encoding_e' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2065:
'sip_accept_language_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2065:
'sip_accept_language_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2065:
'sip_accept_language_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2065:
'msg_accept_any_update' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047:
'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection
from 'char [16]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'size_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(286) : warning C4013:
'msg_accept_language_d' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(299) : warning C4013:
'msg_accept_language_e' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2065:
'sip_content_disposition_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2065:
'sip_content_disposition_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2065:
'sip_content_disposition_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f
(__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'msg_dup_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2065:
'msg_content_disposition_update' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047:
'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection
from 'char [20]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'size_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(376) : warning C4013:
'msg_content_disposition_d' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(382) : warning C4013:
'msg_content_disposition_e' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(388) : warning C4013:
'msg_content_disposition_dup_xtra' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(397) : warning C4013:
'msg_content_disposition_dup_one' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(397) : warning C4047:
'return' : 'char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2065:
'sip_content_encoding_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2065:
'sip_content_encoding_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2065:
'sip_content_encoding_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [17]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [2]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2065:
'sip_content_language_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2065:
'sip_content_language_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2065:
'sip_content_language_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [17]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2065:
'sip_content_type_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2065:
'sip_content_type_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2065:
'sip_content_type_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f
(__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'msg_dup_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [13]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [2]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(557) : warning C4013:
'msg_content_type_d' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(562) : warning C4013:
'msg_content_type_e' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(568) : warning C4013:
'msg_content_type_dup_xtra' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(576) : warning C4013:
'msg_content_type_dup_one' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(578) : warning C4047:
'return' : 'char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2065:
'sip_mime_version_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2065:
'sip_mime_version_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2065:
'sip_mime_version_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [13]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2065:
'sip_warning_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2065:
'sip_warning_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2065:
'sip_warning_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2065:
'msg_warning_dup_xtra' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2065:
'msg_warning_dup_one' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'void *'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char [8]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047:
'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(677) : warning C4013:
'msg_warning_d' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(681) : warning C4013:
'msg_warning_e' undefined; assuming extern returning int

38>sip_parser.c

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser.c(598) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser.c(598) : warning C4013:
'sip_object' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser.c(598) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

42>auserver.c

38>sip_parser_table.c

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(5) : error C2143:
syntax error : missing '{' before 'const'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(5) : warning
C4431: missing type specifier - int assumed. Note: C no longer supports
default-int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(5) : error C2065:
'MC_SHORT_SIZE' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(5) : error C2057:
expected constant expression

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(5) : error C2466:
cannot allocate an array of constant size 0

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : error C2065:
'sip_accept_contact_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : warning
C4013: 'offsetof' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : error C2065:
'sip_t' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : error C2065:
'sip_accept_contact' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(8) : error C2065:
'sip_mask_pref' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(8) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(9) : error C2065:
'sip_referred_by_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(9) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(9) : error C2065:
'sip_t' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(9) : error C2065:
'sip_referred_by' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(9) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(10) : error C2065:
'sip_content_type_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(10) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(10) : error C2065:
'sip_t' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(10) : error C2065:
'sip_content_type' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(10) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(11) : error C2065:
'sip_mask_ua' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(11) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(12) : error C2065:
'sip_request_disposition_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(12) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(12) : error C2065:
'sip_t' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(12) : error C2065:
'sip_request_disposition' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(12) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(13) : error C2065:
'sip_mask_pref' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(13) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(14) : error C2065:
'sip_content_encoding_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(14) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(14) : error C2065:
'sip_t' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(14) : error C2065:
'sip_content_encoding' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(14) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(15) : error C2065:
'sip_mask_ua' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(15) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(16) : error C2065:
'sip_from_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(16) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(16) : error C2065:
'sip_t' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(16) : error C2065:
'sip_from' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(16) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(17) : error C2065:
'sip_mask_request' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(17) : error C2065:
'sip_mask_response' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(17) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(18) : error C2065:
'NULL' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(18) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(19) : error C2065:
'NULL' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(19) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(20) : error C2065:
'sip_call_id_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(20) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(20) : error C2065:
'sip_t' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(20) : error C2065:
'sip_call_id' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(20) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(21) : error C2065:
'sip_mask_request' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(21) : error C2065:
'sip_mask_response' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(21) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(22) : error C2065:
'sip_reject_contact_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(22) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(22) : error C2065:
'sip_t' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(22) : error C2065:
'sip_reject_contact' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(22) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(23) : error C2065:
'sip_mask_pref' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(23) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(24) : error C2065:
'sip_supported_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(24) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(24) : error C2065:
'sip_t' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(24) : error C2065:
'sip_supported' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(24) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(25) : error C2065:
'sip_content_length_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(25) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(25) : error C2065:
'sip_t' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(25) : error C2065:
'sip_content_length' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(25) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(26) : error C2065:
'sip_mask_request' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(26) : error C2065:
'sip_mask_response' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(26) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(27) : error C2065:
'sip_contact_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(27) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(27) : error C2065:
'sip_t' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(27) : error C2065:
'sip_contact' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(27) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(28) : error C2065:
'sip_mask_ua' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(28) : error C2065:
'sip_mask_proxy' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(28) : error C2065:
'sip_mask_registrar' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(28) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(29) : error C2065:
'NULL' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(29) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(30) : error C2065:
'sip_event_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(30) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(30) : error C2065:
'sip_t' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(30) : error C2065:
'sip_event' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(30) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(31) : error C2065:
'sip_mask_events' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(31) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(32) : error C2065:
'NULL' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(32) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(33) : error C2065:
'NULL' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(33) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(34) : error C2065:
'sip_refer_to_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(34) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(34) : fatal error
C1003: error count exceeds 100; stopping compilation

38>sip_prack.c

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2065:
'sip_rack_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2065:
'sip_rack_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2065:
'sip_rack_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f
(__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'msg_dup_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [5]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(122) : warning C4013:
'sip_is_rack' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2065:
'sip_rseq_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2065:
'sip_rseq_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2065:
'sip_rseq_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [5]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(202) : warning C4013:
'sip_is_rseq' undefined; assuming extern returning int

38>sip_pref_util.c

42>cmu_lex.c

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_pref_util.c(529) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_pref_util.c(529) : warning C4013:
'sip_contact_copy' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_pref_util.c(529) : warning C4047:
'=' : 'sip_contact_t *' differs in levels of indirection from 'int'

38>sip_reason.c

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2065:
'sip_reason_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2065:
'sip_reason_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2065:
'sip_reason_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f
(__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'msg_dup_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [7]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(120) : warning C4013:
'sip_is_reason' undefined; assuming extern returning int

38>sip_refer.c

42>c:\freeswitch\freeswitch-1.0.6\libs\flite-1.3.99\lang\cmulex\cmu_lex.c(35
6) : warning C4090: '=' : different 'const' qualifiers

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_refer.c(43) : fatal error C1083:
Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory

38>sip_security.c

42>cmu_lex_data.c

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2065:
'sip_authorization_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2065:
'sip_authorization_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2065:
'sip_authorization_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [14]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(140) : warning C4013:
'sip_is_authorization' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2065:
'sip_proxy_authenticate_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2065:
'sip_proxy_authenticate_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2065:
'sip_proxy_authenticate_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [19]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(211) : warning C4013:
'sip_is_proxy_authenticate' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2065:
'sip_proxy_authorization_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2065:
'sip_proxy_authorization_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2065:
'sip_proxy_authorization_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [20]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(267) : warning C4013:
'sip_is_proxy_authorization' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2065:
'sip_www_authenticate_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2065:
'sip_www_authenticate_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2065:
'sip_www_authenticate_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [17]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(321) : warning C4013:
'sip_is_www_authenticate' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2065:
'sip_authentication_info_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2065:
'sip_authentication_info_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2065:
'sip_authentication_info_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [20]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(380) : warning C4013:
'sip_is_authentication_info' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2065:
'sip_proxy_authentication_info_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2065:
'sip_proxy_authentication_info_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2065:
'sip_proxy_authentication_info_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const msg_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [26]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(447) : warning C4013:
'sip_is_proxy_authentication_info' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2065:
'sip_security_client_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2065:
'sip_security_client_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2065:
'sip_security_client_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const sip_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [16]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2065:
'sip_security_server_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2065:
'sip_security_server_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2065:
'sip_security_server_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const sip_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [16]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2065:
'sip_security_verify_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2065:
'sip_security_verify_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2065:
'sip_security_verify_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'isize_t
(__cdecl *)(const sip_header_t *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [16]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2065:
'sip_privacy_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2065:
'sip_privacy_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2065:
'sip_privacy_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f
(__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'msg_dup_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [8]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>sip_session.c

42>cmu_lex_entries.c

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2065:
'sip_session_expires_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2065:
'sip_session_expires_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2065:
'sip_session_expires_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f
(__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'msg_dup_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [16]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [2]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2065:
'sip_min_se_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2065:
'sip_min_se_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2065:
'sip_min_se_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f
(__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'msg_dup_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [7]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'uintptr_t'

38>sip_status.c

38>sip_tag.c

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag.c(2) : error C2220: warning
treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag.c(2) : warning C4206:
nonstandard extension used : translation unit is empty

38>sip_tag_class.c

42>cmu_lts_model.c

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(219) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(219) : warning C4013:
'sip_object' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(219) : warning C4047:
'=' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(224) : error C2065:
'siptag_end' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(224) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(232) : warning C4013:
'SIPTAG_P' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(247) : error C2065:
'siptag_header' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(247) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(253) : warning C4013:
'SIPTAG_STR_P' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(259) : error C2065:
'siptag_header_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(259) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(364) : error C2065:
'sip_payload_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(454) : error C2065:
'siptag_payload' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(454) : warning C4047:
'=' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(454) : error C2065:
'sip_payload_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(454) : warning C4047:
'=' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(456) : error C2065:
'sip_tag_list' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(456) : error C2109:
subscript requires array or pointer type

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(457) : error C2065:
'sip_tag_list' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(457) : error C2109:
subscript requires array or pointer type

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(483) : error C2065:
'siptag_header_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(483) : warning C4047:
'=' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(493) : error C2065:
'siptag_header_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(493) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>sip_tag_ref.c

38>sip_time.c

42>cmu_lts_rules.c

38>Generating Code...

38>Compiling...

38>sip_util.c

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(130) : error C2220:
warning treated as error - no 'object' file generated

42>cmu_postlex.c

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(130) : warning C4013:
'sip_contact_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(130) : warning C4047: '='
: 'sip_contact_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(196) : warning C4047:
'initializing' : 'sip_contact_t *' differs in levels of indirection from
'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(575) : warning C4013:
'sip_route_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(615) : error C2065:
'sip_route_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(615) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(615) : warning C4024:
'sip_route_reverse_as' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(668) : error C2065:
'sip_route_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(668) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(668) : warning C4024:
'sip_route_fixdup_as' : different types for formal and actual parameter 2

38>http_basic.c

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(161) : error C2065:
'http_request_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(161) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(161) : warning C4024:
'msg_header_alloc' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : error C2065:
'http_request_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4113:
'isize_t (__cdecl *)(const http_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const http_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4113:
'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(http_header_t *,const http_header_t *,char
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(271) : error C2065:
'http_status_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(271) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(271) : warning C4024:
'msg_header_alloc' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : error C2065:
'http_status_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4113:
'isize_t (__cdecl *)(const http_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const http_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4113:
'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(http_header_t *,const http_header_t *,char
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : error C2065:
'http_accept_ranges_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [14]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : error C2065:
'http_age_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [4]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : error C2065:
'http_allow_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [6]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : error C2065:
'http_authentication_info_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [20]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : error C2065:
'http_authorization_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [14]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : error C2065:
'http_cache_control_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [14]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : error C2065:
'http_connection_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [11]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : error C2065:
'http_content_range_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [14]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(584) : error C2065:
'http_date_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(584) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(584) : warning C4024:
'msg_header_alloc' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : error C2065:
'http_date_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [5]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : error C2065:
'http_etag_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [5]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : error C2065:
'http_expect_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [7]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : error C2065:
'http_expires_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [8]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : error C2065:
'http_from_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [5]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(735) : warning C4013:
'http_host_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(740) : warning C4013:
'http_host_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(740) : warning C4047:
'return' : 'http_host_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : error C2065:
'http_host_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4113:
'isize_t (__cdecl *)(const http_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const http_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4113:
'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(http_header_t *,const http_header_t *,char
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [5]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : error C2065:
'http_if_match_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [9]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : error C2065:
'http_if_modified_since_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [18]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : error C2065:
'http_if_none_match_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [14]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : error C2065:
'http_if_range_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4113:
'isize_t (__cdecl *)(const http_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const http_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4113:
'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(http_header_t *,const http_header_t *,char
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [9]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : error C2065:
'http_if_unmodified_since_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [20]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : error C2065:
'http_last_modified_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [14]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : error C2065:
'http_location_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [9]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : error C2065:
'http_max_forwards_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

42>cmu_time_awb_cart.c

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [13]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : error C2065:
'http_pragma_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [7]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : error C2065:
'http_proxy_authenticate_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [19]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : error C2065:
'http_proxy_authorization_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [20]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : error C2065:
'http_range_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [6]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : error C2065:
'http_referer_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [8]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : error C2065:
'http_retry_after_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [12]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : error C2065:
'http_server_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [7]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1402) : warning C4013:
'http_is_te' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : error C2065:
'http_te_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [3]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : error C2065:
'http_trailer_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [8]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : error C2065:
'http_transfer_encoding_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [18]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : error C2065:
'http_upgrade_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [8]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : error C2065:
'http_user_agent_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [11]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : error C2065:
'http_vary_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [5]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : error C2065:
'http_via_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4113:
'isize_t (__cdecl *)(const http_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const http_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4113:
'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(http_header_t *,const http_header_t *,char
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [4]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2065:
'http_warning_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2065:
'msg_warning_d' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2065:
'msg_warning_e' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2065:
'msg_warning_dup_xtra' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2065:
'msg_warning_dup_one' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'void *'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047:
'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection
from 'char [8]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047:
'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection
from 'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047:
'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection
from 'uintptr_t'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : error C2065:
'http_www_authenticate_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [17]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>http_extra.c

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : error C2065:
'http_proxy_connection_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [17]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : error C2065:
'http_cookie_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [7]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : error C2065:
'http_set_cookie_hash' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4047:
'initializing' : 'int' differs in levels of indirection from 'issize_t
(__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4113:
'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter
lists from 'msg_print_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4057:
'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to
slightly different base types from 'isize_t (__cdecl *)(const msg_header_t
*,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4113:
'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
differs in parameter lists from 'msg_xtra_f (__cdecl *)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4047:
'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection
from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4047:
'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection
from 'char [11]'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4047:
'initializing' : 'const char *' differs in levels of indirection from
'unsigned int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4047:
'initializing' : 'short' differs in levels of indirection from 'char [1]'

38>http_header.c

42>cmu_time_awb_clunits.c

38>..\..\sofia-sip\libsofia-sip-ua\http\http_header.c(223) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\http\http_header.c(223) : warning C4013:
'http_header_vformat' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\http\http_header.c(223) : warning C4047:
'=' : 'http_header_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_header.c(274) : warning C4047:
'=' : 'http_header_t *' differs in levels of indirection from 'int'

38>http_parser.c

42>cmu_time_awb_lex_entry.c

38>http_parser_table.c

38>..\..\sofia-sip\libsofia-sip-ua\http\http_parser_table.c(6) : error
C2143: syntax error : missing '{' before '<cv-qualifer>'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_parser_table.c(6) : error
C2059: syntax error : '<Unknown>'

38>http_status.c

38>http_tag.c

38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag.c(2) : error C2220: warning
treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag.c(2) : warning C4206:
nonstandard extension used : translation unit is empty

38>http_tag_class.c

42>cmu_time_awb_lpc.c

38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(177) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(177) : warning
C4013: 'HTTPTAG_P' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(196) : error C2065:
'httptag_header' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(196) : warning
C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(203) : warning
C4013: 'HTTPTAG_STR_P' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(209) : error C2065:
'httptag_header_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(209) : warning
C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'

38>http_tag_ref.c

38>nth_client.c

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_client.c(652) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_client.c(652) : warning C4013:
'HTTPTAG_VERSION_REF' undefined; assuming extern returning int

38>nth_server.c

42>cmu_time_awb_mcep.c

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(718) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(718) : warning C4013:
'HTTPTAG_SERVER_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(719) : warning C4013:
'HTTPTAG_SERVER_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(751) : warning C4013:
'http_server_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(751) : warning C4047:
'=' : 'http_server_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(753) : warning C4013:
'http_server_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(753) : warning C4047:
'=' : 'http_server_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(892) : warning C4013:
'http_location_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(966) : warning C4013:
'http_payload_format' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(970) : warning C4047:
'=' : 'http_payload_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(975) : warning C4013:
'http_status_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(981) : warning C4013:
'HTTPTAG_STATUS' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(981) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(981) : warning C4024:
'http_add_tl' : different types for formal and actual parameter 3

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(982) : warning C4013:
'HTTPTAG_SERVER' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(983) : warning C4013:
'HTTPTAG_CONTENT_TYPE_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(984) : warning C4013:
'HTTPTAG_SEPARATOR_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(985) : warning C4013:
'HTTPTAG_CONNECTION_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(985) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1077) : error C2065:
'http_www_authenticate_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1077) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1109) : warning C4013:
'HTTPTAG_HEADER' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1109) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1109) : warning C4024:
'nth_request_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1147) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1147) : warning C4024:
'nth_request_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1148) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1148) : warning C4024:
'nth_request_treply' : different types for formal and actual parameter 5

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1263) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1263) : warning C4024:
'http_add_tl' : different types for formal and actual parameter 3

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1295) : warning C4013:
'http_date_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1295) : error C2223:
left of '->d_time' must point to struct/union

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1311) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1311) : warning C4024:
'http_add_tl' : different types for formal and actual parameter 3

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1311) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1311) : warning C4024:
'http_add_tl' : different types for formal and actual parameter 4

38>nth_tag.c

38>nth_tag_ref.c

42>cmu_us_kal.c

38>sres.c

42>cmu_us_kal_diphone.c

42>cmu_us_kal_lpc.c

38>sres_blocking.c

42>cmu_us_kal_res.c

38>sres_cache.c

38>sres_sip.c

38>sresolv.c

38>nea.c

42>Generating Code...

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : error C2065: 'siptag_to'
: undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : warning C4024: 'tl_find'
: different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : error C2065:
'siptag_to_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : warning C4024: 'tl_find'
: different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : error C2065:
'siptag_from' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : warning C4024: 'tl_find'
: different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : error C2065:
'siptag_from_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : warning C4024: 'tl_find'
: different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(135) : error C2065:
'siptag_contact' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(135) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(135) : warning C4024: 'tl_find'
: different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(136) : error C2065:
'siptag_contact_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(136) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(136) : warning C4024: 'tl_find'
: different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(155) : warning C4013:
'SIPTAG_CONTACT' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(155) : warning C4047: ':' :
'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(156) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(156) : warning C4024:
'tl_tlist' : different types for formal and actual parameter 3

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(160) : warning C4013:
'SIPTAG_EXPIRES_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(160) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(160) : warning C4024: 'tl_gets'
: different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(161) : warning C4013:
'SIPTAG_EXPIRES_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(162) : warning C4013:
'SIPTAG_TO_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(170) : warning C4013:
'sip_to_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(170) : warning C4047: '=' :
'sip_to_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(173) : warning C4013:
'sip_expires_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(173) : warning C4047: '=' :
'sip_expires_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(175) : warning C4013:
'sip_expires_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(175) : warning C4047: '=' :
'sip_expires_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(180) : warning C4013:
'SIPTAG_EXPIRES' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(180) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(180) : warning C4024:
'tl_tremove' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(181) : warning C4013:
'SIPTAG_EXPIRES_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(195) : warning C4013:
'SIPTAG_FROM' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(195) : warning C4047: ':' :
'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(196) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_typedef_t'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(196) : warning C4024:
'nta_leg_tcreate' : different types for formal and actual parameter 5

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(205) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(205) : warning C4024:
'nta_outgoing_tcreate' : different types for formal and actual parameter 8

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(206) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_typedef_t'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(206) : warning C4024:
'nta_outgoing_tcreate' : different types for formal and actual parameter 9

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(242) : warning C4013:
'SIPTAG_CONTENT_TYPE_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(242) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(242) : warning C4024: 'tl_gets'
: different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(243) : warning C4013:
'SIPTAG_CONTENT_TYPE_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(244) : warning C4013:
'SIPTAG_PAYLOAD_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(254) : warning C4013:
'SIPTAG_CONTENT_TYPE' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(254) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(254) : warning C4024:
'tl_tremove' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(255) : warning C4013:
'SIPTAG_CONTENT_TYPE_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(256) : warning C4013:
'SIPTAG_PAYLOAD' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(257) : warning C4013:
'SIPTAG_PAYLOAD_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(263) : warning C4047: '=' :
'sip_expires_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(273) : warning C4013:
'SIPTAG_TO' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(273) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(273) : warning C4024:
'nta_outgoing_tcreate' : different types for formal and actual parameter 8

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(275) : warning C4047: ':' :
'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(276) : warning C4047: ':' :
'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(307) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(307) : warning C4024:
'nta_outgoing_tcreate' : different types for formal and actual parameter 8

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(308) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_typedef_t'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(308) : warning C4024:
'nta_outgoing_tcreate' : different types for formal and actual parameter 9

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(346) : warning C4013:
'SIPTAG_ALLOW_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(346) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(346) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(346) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(346) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 5

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(410) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(410) : warning C4024:
'nta_outgoing_tcreate' : different types for formal and actual parameter 8

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(411) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_typedef_t'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(411) : warning C4024:
'nta_outgoing_tcreate' : different types for formal and actual parameter 9

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(416) : warning C4013:
'sip_expires_format' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(416) : warning C4047: '=' :
'sip_expires_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(425) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(425) : warning C4024:
'nta_outgoing_tcreate' : different types for formal and actual parameter 8

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(426) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_typedef_t'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(426) : warning C4024:
'nta_outgoing_tcreate' : different types for formal and actual parameter 9

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(496) : warning C4013:
'sip_subscription_state_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(594) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(594) : warning C4024:
'nta_outgoing_tcreate' : different types for formal and actual parameter 8

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(595) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_typedef_t'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(595) : warning C4024:
'nta_outgoing_tcreate' : different types for formal and actual parameter 9

38>Generating Code...

38>Compiling...

38>nea_debug.c

38>nea_event.c

42>Compiling...

42>cmu_us_kal_residx.c

42>cst_alloc.c

38>nea_server.c

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(401) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(401) : warning C4013:
'SIPTAG_CONTACT_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(401) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(401) : warning C4024:
'tl_gets' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(402) : warning C4013:
'SIPTAG_CONTACT_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(403) : warning C4013:
'SIPTAG_ALLOW_EVENTS_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(404) : warning C4013:
'SIPTAG_SERVER_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(405) : warning C4013:
'SIPTAG_REQUIRE_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(406) : warning C4013:
'SIPTAG_REQUIRE_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(450) : warning C4013:
'sip_allow_events_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(450) : warning C4047:
'=' : 'sip_allow_events_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(452) : warning C4013:
'sip_allow_events_make' undefined; assuming extern returning int

42>cst_args.c

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(452) : warning C4047:
'=' : 'sip_allow_events_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(454) : warning C4013:
'sip_allow_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(454) : warning C4047:
'=' : 'sip_allow_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(463) : warning C4013:
'sip_contact_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(463) : warning C4047:
'=' : 'sip_contact_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(465) : warning C4013:
'sip_contact_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(465) : warning C4047:
'=' : 'sip_contact_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(488) : warning C4013:
'sip_require_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(488) : warning C4047:
'=' : 'sip_require_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(490) : warning C4013:
'sip_require_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(490) : warning C4047:
'=' : 'sip_require_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(726) : warning C4013:
'SIPTAG_CONTENT_TYPE_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(726) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(726) : warning C4024:
'tl_gets' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(727) : warning C4013:
'SIPTAG_CONTENT_TYPE_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(728) : warning C4013:
'SIPTAG_PAYLOAD_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(729) : warning C4013:
'SIPTAG_PAYLOAD_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(785) : warning C4013:
'sip_payload_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(786) : warning C4013:
'sip_payload_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(786) : warning C4047:
'=' : 'sip_payload_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(788) : warning C4013:
'sip_content_type_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(789) : warning C4013:
'sip_content_type_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(789) : warning C4047:
'=' : 'sip_content_type_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(825) : warning C4047:
'=' : 'sip_payload_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1208) : warning C4013:
'SIPTAG_CONTENT_TYPE_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1208) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1208) : warning C4024:
'nea_event_tcreate' : different types for formal and actual parameter 6

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1209) : warning C4013:
'SIPTAG_ACCEPT_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1270) : warning C4013:
'SIPTAG_ACCEPT_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1271) : warning C4013:
'SIPTAG_ACCEPT_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1272) : warning C4013:
'SIPTAG_SUPPORTED_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1273) : warning C4013:
'SIPTAG_SUPPORTED_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1280) : warning C4013:
'sip_event_format' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1283) : warning C4047:
'=' : 'sip_event_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1294) : warning C4047:
'=' : 'sip_require_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1296) : warning C4047:
'=' : 'sip_require_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1307) : warning C4013:
'sip_accept_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1307) : warning C4047:
'=' : 'const sip_accept_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1309) : warning C4047:
'=' : 'const sip_accept_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1315) : warning C4013:
'sip_accept_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1315) : warning C4047:
'=' : 'const sip_accept_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1317) : warning C4047:
'=' : 'const sip_accept_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1320) : warning C4013:
'sip_supported_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1320) : warning C4047:
'=' : 'sip_supported_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1322) : warning C4013:
'sip_supported_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1322) : warning C4047:
'=' : 'sip_supported_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1503) : warning C4013:
'sip_from_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1503) : warning C4047:
'=' : 'sip_from_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1508) : warning C4047:
'=' : 'sip_contact_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1516) : warning C4013:
'SIPTAG_FROM' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1516) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1516) : warning C4024:
'nta_leg_tcreate' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1517) : warning C4013:
'SIPTAG_TO' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1518) : warning C4013:
'SIPTAG_CALL_ID' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1576) : warning C4013:
'SIPTAG_ALLOW_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1576) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1576) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1577) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1577) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 5

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1619) : warning C4013:
'SIP_EXPIRES_INIT' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1619) : warning C4047:
'initializing' : 'msg_header_t *' differs in levels of indirection from
'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1631) : warning C4013:
'SIPTAG_SERVER_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1631) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1631) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1632) : warning C4013:
'SIPTAG_ALLOW_EVENTS' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1633) : warning C4013:
'SIPTAG_ALLOW' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1643) : warning C4013:
'sip_min_expires_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1648) : warning C4013:
'SIPTAG_ACCEPT' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1648) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1648) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1649) : warning C4013:
'SIPTAG_MIN_EXPIRES' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1666) : warning C4013:
'SIPTAG_REQUIRE' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1666) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1666) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1667) : warning C4013:
'SIPTAG_UNSUPPORTED' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1688) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1688) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1689) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1689) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 5

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1734) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1734) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1746) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1746) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1748) : warning C4013:
'SIPTAG_SUPPORTED' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1763) : warning C4013:
'sip_accept_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1777) : warning C4047:
'=' : 'sip_accept_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1853) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1853) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1879) : warning C4047:
'=' : 'sip_contact_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1884) : warning C4047:
'=' : 'sip_content_type_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1886) : warning C4047:
'=' : 'sip_payload_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1901) : warning C4013:
'sip_event_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1901) : warning C4047:
'=' : 'sip_event_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1909) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1909) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1914) : warning C4013:
'SIPTAG_EXPIRES' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1915) : warning C4013:
'SIPTAG_CONTACT' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1942) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1942) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1943) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1943) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 5

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1946) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1946) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1956) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1956) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2038) : warning C4013:
'sip_subscription_state_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2092) : warning C4013:
'SIPTAG_SUBSCRIPTION_STATE' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2092) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2092) : warning C4024:
'nta_outgoing_tcreate' : different types for formal and actual parameter 8

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2095) : warning C4013:
'SIPTAG_USER_AGENT_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2097) : warning C4013:
'SIPTAG_EVENT' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2099) : warning C4013:
'SIPTAG_CONTENT_TYPE' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2099) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2101) : warning C4013:
'SIPTAG_PAYLOAD' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2101) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>nea_tag.c

42>cst_cart.c

38>nea_tag_ref.c

38>auth_client.c

42>cst_cg.c

38>auth_common.c

42>cst_clunits.c

38>auth_digest.c

38>auth_module.c

42>cst_diphone.c

38>auth_module_http.c

38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_http.c(47) : error
C2065: 'http_www_authenticate_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_http.c(47) : error
C2099: initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_http.c(50) : error
C2065: 'http_proxy_authenticate_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_http.c(50) : error
C2099: initializer is not a constant

42>cst_endian.c

38>auth_module_sip.c

42>cst_error.c

38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(49) : error
C2065: 'sip_www_authenticate_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(49) : error
C2099: initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(51) : error
C2065: 'sip_authentication_info_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(51) : error
C2099: initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(54) : error
C2065: 'sip_proxy_authenticate_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(54) : error
C2099: initializer is not a constant

42>cst_features.c

38>auth_plugin.c

42>cst_ffeature.c

38>auth_plugin_delayed.c

38>auth_tag.c

42>cst_ffeatures.c

38>auth_tag_ref.c

38>iptsec_debug.c

42>cst_file_stdio.c

38>stun.c

42>cst_item.c

38>stun_common.c

42>cst_lexicon.c

38>stun_dns.c

42>cst_lpcres.c

38>stun_mini.c

38>Generating Code...

42>cst_lts.c

38>Compiling...

38>stun_tag.c

42>cst_lts_rewrites.c

38>stun_tag_ref.c

38>nua.c

42>cst_mlpg.c

38>nua_client.c

42>cst_mlsa.c

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(496) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(496) : warning C4013:
'sip_object' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(496) : warning C4047:
'=' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(502) : error C2065:
'siptag_contact' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(502) : warning C4047:
'==' : 'const tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(503) : error C2065:
'siptag_contact_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(503) : warning C4047:
'==' : 'const tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(529) : error C2065:
'siptag_contact' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(529) : warning C4047:
'==' : 'const tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(530) : error C2065:
'siptag_contact_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(530) : warning C4047:
'==' : 'const tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(549) : warning C4013:
'sip_add_tagis' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(569) : warning C4013:
'sip_to_tag' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(573) : warning C4013:
'sip_add_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(577) : warning C4013:
'sip_add_dup_as' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(577) : error C2065:
'sip_to_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(615) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(625) : error C2065:
'siptag_from' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(625) : warning C4047:
'==' : 'const tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(743) : warning C4013:
'SIPTAG_CALL_ID' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(743) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(743) : warning C4024:
'nta_leg_tcreate' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(744) : warning C4013:
'SIPTAG_FROM' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(745) : warning C4013:
'SIPTAG_TO' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(746) : warning C4013:
'SIPTAG_CSEQ' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(754) : warning C4013:
'sip_from_tag' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(768) : warning C4047:
'=' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(781) : warning C4013:
'sip_route_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(781) : warning C4047:
'=' : 'sip_route_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(813) : warning C4013:
'sip_to_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(813) : warning C4047:
'=' : 'const sip_to_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(815) : warning C4013:
'sip_from_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(815) : warning C4047:
'=' : 'const sip_from_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(830) : warning C4013:
'sip_has_feature' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(832) : warning C4013:
'sip_add_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(832) : error C2065:
'sip_supported_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(835) : error C2065:
'sip_organization_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(838) : error C2065:
'sip_user_agent_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(869) : warning C4013:
'sip_contact_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(869) : warning C4047:
'initializing' : 'sip_contact_t *' differs in levels of indirection from
'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1164) : warning C4013:
'sip_expires_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1181) : error C2065:
'sip_authorization_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1181) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1181) : warning C4024:
'auc_challenge' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1186) : error C2065:
'sip_proxy_authorization_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1186) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1186) : warning C4024:
'auc_challenge' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1477) : error C2065:
'sip_authorization_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1477) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1477) : warning C4024:
'auc_info' : different types for formal and actual parameter 3

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1481) : error C2065:
'sip_proxy_authorization_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1481) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1481) : warning C4024:
'auc_info' : different types for formal and actual parameter 3

38>nua_common.c

42>Generating Code...

42>Compiling...

42>cst_mmap_win32.c

38>nua_dialog.c

42>cst_phoneset.c

38>c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_typ
es.h(41) : error C2061: syntax error : identifier 'nua_owner_t'

38>c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_typ
es.h(41) : error C2059: syntax error : ';'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(47) : error C2016: C requires that a struct or union has at least one
member

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(47) : error C2061: syntax error : identifier 'nua_owner_t'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(58) : error C2143: syntax error : missing '{' before ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(58) : error C2059: syntax error : ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(60) : error C2143: syntax error : missing '{' before ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(60) : error C2059: syntax error : ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(61) : error C2143: syntax error : missing '{' before ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(61) : error C2059: syntax error : ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(63) : error C2143: syntax error : missing '{' before ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(63) : error C2059: syntax error : ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(64) : error C2143: syntax error : missing '{' before ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(64) : error C2059: syntax error : ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(65) : error C2143: syntax error : missing '{' before ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(65) : error C2059: syntax error : ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(67) : error C2143: syntax error : missing '{' before ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(67) : error C2059: syntax error : ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(68) : error C2143: syntax error : missing '{' before ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(68) : error C2059: syntax error : ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(70) : error C2059: syntax error : ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(92) : error C2059: syntax error : '}'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(97) : error C2143: syntax error : missing ')' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(97) : error C2143: syntax error : missing ';' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(97) : error C2059: syntax error : '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(99) : error C2059: syntax error : ')'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(100) : error C2143: syntax error : missing ')' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(100) : error C2143: syntax error : missing '{' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(100) : error C2059: syntax error : ','

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(104) : error C2059: syntax error : ')'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(113) : error C2143: syntax error : missing ')' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(113) : error C2143: syntax error : missing '{' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(113) : error C2059: syntax error : ','

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(114) : error C2059: syntax error : ')'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(115) : error C2143: syntax error : missing ')' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(115) : error C2143: syntax error : missing '{' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(115) : error C2059: syntax error : ','

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(116) : error C2059: syntax error : ')'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(117) : error C2059: syntax error : '}'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(123) : error C2061: syntax error : identifier 'nua_usage_class'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(129) : error C2143: syntax error : missing '{' before ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(129) : error C2059: syntax error : ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(130) : error C2143: syntax error : missing '{' before ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(130) : error C2059: syntax error : ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(131) : error C2059: syntax error : ':'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(140) : error C2059: syntax error : '}'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(142) : error C2143: syntax error : missing ')' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(142) : error C2143: syntax error : missing '{' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(142) : error C2059: syntax error : ','

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(143) : error C2059: syntax error : ')'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(144) : error C2143: syntax error : missing ')' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(144) : error C2143: syntax error : missing '{' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(144) : error C2059: syntax error : ','

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(145) : error C2059: syntax error : ')'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(146) : error C2143: syntax error : missing ')' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(146) : error C2143: syntax error : missing '{' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(146) : error C2059: syntax error : ','

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(147) : error C2059: syntax error : ')'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(148) : error C2143: syntax error : missing ')' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(148) : error C2143: syntax error : missing '{' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(148) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(149) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(149) : error C2371: 'nua_dialog_state_t' : redefinition; different
basic types

38>
c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.
h(43) : see declaration of 'nua_dialog_state_t'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(149) : error C2143: syntax error : missing ';' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(149) : warning C4218: nonstandard extension used : must specify at
least a storage class or a type

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(149) : error C2059: syntax error : ')'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(149) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(149) : warning C4218: nonstandard extension used : must specify at
least a storage class or a type

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(150) : error C2143: syntax error : missing ')' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(150) : error C2143: syntax error : missing '{' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(150) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(151) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(151) : error C2371: 'nua_dialog_state_t' : redefinition; different
basic types

38>
c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.
h(43) : see declaration of 'nua_dialog_state_t'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(151) : error C2143: syntax error : missing ';' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(151) : warning C4218: nonstandard extension used : must specify at
least a storage class or a type

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(151) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(152) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(152) : error C2371: 'nua_dialog_usage_t' : redefinition; different
basic types

38>
c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.
h(44) : see declaration of 'nua_dialog_usage_t'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(152) : error C2143: syntax error : missing ';' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(152) : warning C4218: nonstandard extension used : must specify at
least a storage class or a type

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(152) : error C2059: syntax error : ')'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(152) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(152) : warning C4218: nonstandard extension used : must specify at
least a storage class or a type

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(156) : error C2037: left of 'ds_reporting' specifies undefined
struct/union 'nua_dialog_state'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(156) : warning C4033: 'nua_dialog_is_reporting' must return a value

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(161) : error C2143: syntax error : missing ')' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(161) : error C2143: syntax error : missing '{' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(161) : error C2059: syntax error : ','

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(164) : error C2059: syntax error : ')'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(167) : error C2143: syntax error : missing ')' before 'const'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(167) : error C2081: 'nua_usage_class' : name in formal parameter list
illegal

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(167) : error C2143: syntax error : missing '{' before 'const'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(167) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(168) : error C2373: 'sip_event_t' : redefinition; different type
modifiers

38>
c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\sip\sofia-sip/
sip.h(204) : see declaration of 'sip_event_t'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(168) : error C2143: syntax error : missing ';' before 'const'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(168) : error C2059: syntax error : ')'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(170) : error C2143: syntax error : missing ')' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(170) : error C2143: syntax error : missing '{' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(170) : error C2059: syntax error : ','

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(174) : error C2059: syntax error : ')'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(186) : error C2143: syntax error : missing ')' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(186) : error C2143: syntax error : missing '{' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(186) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(187) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(187) : error C2371: 'nua_dialog_state_t' : redefinition; different
basic types

38>
c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.
h(43) : see declaration of 'nua_dialog_state_t'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(187) : error C2143: syntax error : missing ';' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(187) : warning C4218: nonstandard extension used : must specify at
least a storage class or a type

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(187) : error C2059: syntax error : ')'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(187) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(187) : warning C4218: nonstandard extension used : must specify at
least a storage class or a type

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(189) : error C2143: syntax error : missing ')' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(189) : error C2143: syntax error : missing '{' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(189) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(189) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(189) : error C2371: 'nua_dialog_state_t' : redefinition; different
basic types

38>
c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.
h(43) : see declaration of 'nua_dialog_state_t'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(189) : error C2143: syntax error : missing ';' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(189) : warning C4218: nonstandard extension used : must specify at
least a storage class or a type

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(189) : error C2059: syntax error : ')'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(189) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(189) : warning C4218: nonstandard extension used : must specify at
least a storage class or a type

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(191) : error C2143: syntax error : missing ')' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(191) : error C2143: syntax error : missing '{' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(191) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(192) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(192) : error C2371: 'nua_dialog_state_t' : redefinition; different
basic types

38>
c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.
h(43) : see declaration of 'nua_dialog_state_t'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(192) : error C2143: syntax error : missing ';' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(192) : warning C4218: nonstandard extension used : must specify at
least a storage class or a type

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(192) : error C2059: syntax error : ')'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(192) : warning C4431: missing type specifier - int assumed. Note: C no
longer supports default-int

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(192) : warning C4218: nonstandard extension used : must specify at
least a storage class or a type

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(204) : error C2143: syntax error : missing ')' before '*'

38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dia
log.h(204) : fatal error C1003: error count exceeds 100; stopping
compilation

38>nua_event_server.c

42>cst_reflpc.c

42>cst_regex.c

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(86) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(86) : warning
C4013: 'SIPTAG_EVENT_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(87) : warning
C4013: 'SIPTAG_EVENT_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(88) : warning
C4013: 'SIPTAG_CONTENT_TYPE_STR_REF' undefined; assuming extern returning
int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(89) : warning
C4013: 'SIPTAG_PAYLOAD_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(90) : warning
C4013: 'SIPTAG_PAYLOAD_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(108) : warning
C4013: 'sip_event_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(108) : warning
C4047: '=' : 'const sip_event_t *' differs in levels of indirection from
'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(122) : warning
C4013: 'SIPTAG_EVENT' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(122) : warning
C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(122) : warning
C4024: 'nua_stack_tevent' : different types for formal and actual parameter
7

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(123) : warning
C4013: 'SIPTAG_CONTENT_TYPE' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(158) : warning
C4013: 'SIPTAG_ACCEPT_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(158) : warning
C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(158) : warning
C4024: 'tl_gets' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(159) : warning
C4013: 'SIPTAG_ACCEPT_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(160) : warning
C4013: 'SIPTAG_CONTENT_TYPE_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(169) : warning
C4013: 'sip_header_as_string' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(169) : warning
C4047: '=' : 'char *' differs in levels of indirection from 'int'

42>cst_rel_io.c

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(334) : warning
C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(334) : warning
C4024: 'tl_gets' : different types for formal and actual parameter 2

38>nua_extension.c

42>cst_relation.c

38>nua_message.c

42>cst_sigpr.c

38>nua_notifier.c

42>cst_socket.c

42>cst_ss.c

42>cst_string.c

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_notifier.c(47) : fatal error
C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or
directory

38>nua_options.c

42>cst_sts.c

38>nua_params.c

42>cst_synth.c

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(166) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(166) : warning C4013:
'sip_allow_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(166) : warning C4047:
'=' : 'sip_allow_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(167) : warning C4013:
'sip_supported_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(167) : warning C4047:
'=' : 'sip_supported_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(175) : warning C4047:
'=' : 'sip_allow_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(196) : warning C4013:
'SIPTAG_FROM_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(196) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(196) : warning C4024:
'tl_gets' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(197) : warning C4013:
'SIPTAG_FROM_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(203) : warning C4013:
'sip_from_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(208) : warning C4013:
'sip_from_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(208) : warning C4047:
'=' : 'sip_from_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(212) : warning C4013:
'sip_from_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(212) : warning C4047:
'=' : 'sip_from_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(225) : warning C4047:
'=' : 'sip_from_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(833) : error C2065:
'siptag_supported' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(833) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(834) : error C2065:
'siptag_supported_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(834) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(839) : error C2065:
'sip_supported_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(839) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(839) : warning C4024:
'nhp_merge_lists' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(841) : error C2065:
'siptag_supported' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(841) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(853) : error C2065:
'siptag_allow_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(853) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(854) : error C2065:
'siptag_allow' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(854) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(859) : error C2065:
'sip_allow_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(859) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(859) : warning C4024:
'nhp_merge_lists' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(863) : error C2065:
'siptag_allow' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(863) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(875) : error C2065:
'siptag_allow_events_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(875) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(876) : error C2065:
'siptag_allow_events' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(876) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(881) : error C2065:
'sip_allow_events_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(881) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(881) : warning C4024:
'nhp_merge_lists' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(885) : error C2065:
'siptag_allow_events' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(885) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(903) : error C2065:
'sip_allow_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(903) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(903) : warning C4024:
'nhp_merge_lists' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(923) : warning C4013:
'sip_route_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(923) : warning C4013:
'sip_route_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(923) : warning C4047:
'=' : 'sip_route_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(927) : error C2065:
'siptag_user_agent' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(927) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(928) : warning C4013:
'sip_header_as_string' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(928) : warning C4047:
'=' : 'char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(931) : error C2065:
'siptag_user_agent_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(931) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(951) : error C2065:
'siptag_organization' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(951) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(952) : warning C4047:
'=' : 'char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(955) : error C2065:
'siptag_organization_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(955) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1201) : error C2065:
'siptag_from' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1201) : warning C4047:
'==' : 'const tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1205) : error C2065:
'siptag_from_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1205) : warning C4047:
'==' : 'const tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1209) : error C2065:
'siptag_to' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1209) : warning C4047:
'==' : 'const tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1213) : error C2065:
'siptag_to_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1213) : warning C4047:
'==' : 'const tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1235) : warning C4047:
'=' : 'const sip_from_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1242) : warning C4013:
'sip_to_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1242) : warning C4047:
'=' : 'const sip_to_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1244) : warning C4013:
'sip_to_create' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1244) : warning C4047:
'=' : 'const sip_to_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1256) : warning C4013:
'SIPTAG_FROM' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1256) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1257) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from 'const
tag_type_s *'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1257) : warning C4024:
'tl_filtered_tlist' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1258) : warning C4013:
'SIPTAG_TO' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1258) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1269) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1269) : warning C4024:
'tl_gets' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1270) : warning C4013:
'SIPTAG_TO_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1319) : error C2065:
'siptag_from' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1319) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1319) : error C2065:
'siptag_to' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1319) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1327) : error C2065:
'siptag_from_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1327) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1329) : error C2065:
'siptag_to_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1329) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1333) : error C2065:
'siptag_cseq' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1333) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1333) : error C2065:
'siptag_cseq_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1333) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1335) : error C2065:
'siptag_rseq' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1335) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1335) : error C2065:
'siptag_rseq_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1335) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1337) : error C2065:
'siptag_rack' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1337) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1337) : error C2065:
'siptag_rack_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1337) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1339) : error C2065:
'siptag_timestamp' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1339) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1339) : error C2065:
'siptag_timestamp_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1339) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1341) : error C2065:
'siptag_content_length' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1341) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1341) : error C2065:
'siptag_content_length_str' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1341) : warning C4047:
'==' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1575) : warning C4013:
'siptag_contact_vr' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1621) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1625) : warning C4013:
'SIPTAG_FROM_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1625) : warning C4047:
':' : 'int' differs in levels of indirection from 'void *'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1625) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1625) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from 'const
tag_type_s *'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1625) : warning C4024:
'tl_filtered_tlist' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1662) : warning C4013:
'SIPTAG_SUPPORTED' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1662) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1663) : warning C4013:
'SIPTAG_SUPPORTED_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1663) : warning C4047:
':' : 'int' differs in levels of indirection from 'void *'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1663) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1664) : warning C4013:
'SIPTAG_ALLOW' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1664) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1665) : warning C4013:
'SIPTAG_ALLOW_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1665) : warning C4047:
':' : 'int' differs in levels of indirection from 'void *'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1665) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1666) : warning C4047:
':' : 'int' differs in levels of indirection from 'void *'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1667) : warning C4013:
'SIPTAG_ALLOW_EVENTS' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1667) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1668) : warning C4013:
'SIPTAG_ALLOW_EVENTS_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1668) : warning C4047:
':' : 'int' differs in levels of indirection from 'void *'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1668) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1669) : warning C4013:
'SIPTAG_USER_AGENT' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1669) : warning C4013:
'sip_user_agent_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1669) : warning C4047:
':' : 'int' differs in levels of indirection from 'void *'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1669) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1670) : warning C4013:
'SIPTAG_USER_AGENT_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1670) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1673) : warning C4013:
'SIPTAG_ORGANIZATION' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1673) : warning C4013:
'sip_organization_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1673) : warning C4047:
':' : 'int' differs in levels of indirection from 'void *'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1673) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1674) : warning C4013:
'SIPTAG_ORGANIZATION_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1674) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1676) : warning C4013:
'siptag_route_v' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1677) : warning C4047:
':' : 'int' differs in levels of indirection from 'void *'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1705) : warning C4013:
'siptag_contact_v' undefined; assuming extern returning int

38>nua_publish.c

42>cst_tokenstream.c

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(292) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(292) : warning C4013:
'msg_copy' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(292) : warning C4047:
'=' : 'msg_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(314) : warning C4013:
'sip_etag_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(314) : warning C4047:
'=' : 'sip_etag_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(317) : warning C4013:
'sip_header_remove' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(355) : warning C4013:
'SIPTAG_IF_MATCH' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(355) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(355) : warning C4024:
'nua_base_client_trequest' : different types for formal and actual parameter
4

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(356) : warning C4013:
'SIPTAG_PAYLOAD' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(356) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(356) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from 'const
tag_type_s *'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(356) : warning C4024:
'nua_base_client_trequest' : different types for formal and actual parameter
5

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(357) : warning C4013:
'SIPTAG_CONTENT_TYPE' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(357) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(358) : warning C4013:
'SIPTAG_EXPIRES_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(358) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(408) : warning C4047:
'=' : 'sip_etag_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(534) : warning C4013:
'msg_header_find_param' undefined; assuming extern returning int

38>nua_register.c

42>cst_track.c

42>cst_track_io.c

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(756) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(756) : warning C4013:
'sip_add_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(770) : warning C4013:
'sip_header_remove' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(793) : warning C4013:
'SIPTAG_EXPIRES_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(793) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(798) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_typedef_t'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(798) : warning C4024:
'nua_base_client_trequest' : different types for formal and actual parameter
5

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(812) : warning C4013:
'sip_object' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(812) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(855) : warning C4013:
'sip_now' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(860) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(905) : warning C4013:
'sip_contact_expires' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(939) : warning C4013:
'sip_route_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(939) : warning C4047:
'=' : 'sip_route_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(952) : warning C4013:
'sip_path_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(952) : warning C4047:
'=' : 'sip_path_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1319) : warning C4013:
'sip_via_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1319) : error C2223:
left of '->v_next' must point to struct/union

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1320) : error C2065:
'sip_transport_udp' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1320) : warning C4047:
'=' : 'const char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1323) : error C2065:
'sip_transport_tcp' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1323) : warning C4047:
'=' : 'const char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1360) : warning C4013:
'sip_via_copy' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1360) : warning C4047:
'=' : 'sip_via_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1374) : error C2065:
'sip_transport_tcp' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1374) : warning C4047:
'==' : 'const char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1375) : error C2065:
'sip_transport_udp' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1375) : warning C4047:
'=' : 'const char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1376) : error C2065:
'sip_transport_udp' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1376) : warning C4047:
'==' : 'const char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1377) : error C2065:
'sip_transport_tcp' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1377) : warning C4047:
'=' : 'const char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1422) : warning C4013:
'sip_via_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1422) : warning C4047:
'=' : 'sip_via_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1647) : warning C4047:
'=' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1674) : warning C4047:
'=' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1743) : warning C4013:
'sip_contact_format' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1755) : warning C4047:
'=' : 'const sip_contact_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1806) : warning C4013:
'sip_from_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1806) : warning C4047:
'=' : 'sip_from_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1834) : warning C4013:
'sip_contact_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1834) : warning C4047:
'=' : 'sip_contact_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(2006) : warning C4013:
'sip_via_port' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(2006) : warning C4047:
'=' : 'const char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(2122) : warning C4013:
'sip_contact_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(2122) : warning C4047:
'=' : 'sip_contact_t *' differs in levels of indirection from 'int'

38>nua_registrar.c

42>cst_units.c

42>cst_utt_utils.c

38>nua_server.c

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(118) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(118) : warning C4013:
'SIPTAG_SUPPORTED' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(118) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(118) : warning C4024:
'nta_check_method' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(119) : warning C4013:
'SIPTAG_USER_AGENT_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(132) : warning C4013:
'SIPTAG_ALLOW' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(132) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(132) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(140) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(140) : warning C4024:
'nta_check_required' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(178) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(178) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

42>cst_utterance.c

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(207) : warning C4013:
'sip_object' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(207) : warning C4047:
'=' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(236) : warning C4013:
'sip_is_allowed' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(266) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(266) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(267) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(267) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 5

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(497) : warning C4013:
'SIPTAG_END' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(497) : warning C4047:
'initializing' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(519) : warning C4047:
'=' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(530) : warning C4013:
'sip_add_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(533) : warning C4013:
'sip_add_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(533) : error C2065:
'sip_user_agent_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(537) : error C2065:
'sip_organization_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(570) : warning C4013:
'sip_contact_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(570) : warning C4047:
'=' : 'sip_contact_t *' differs in levels of indirection from 'int'

38>nua_session.c

42>cst_val.c

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(796) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(796) : warning C4013:
'sip_has_feature' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(818) : warning C4013:
'sip_accept_contact_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(825) : warning C4013:
'sip_add_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1075) : warning C4013:
'SIP_IS_ALLOWED' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1078) : warning C4013:
'sip_rack_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1085) : warning C4013:
'SIPTAG_RACK' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1085) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1085) : warning C4024:
'nua_client_tcreate' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1086) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1086) : warning C4024:
'nua_client_tcreate' : different types for formal and actual parameter 5

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1260) : warning C4013:
'sip_object' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1260) : warning C4047:
'=' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1265) : warning C4013:
'sip_authorization' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1265) : warning C4047:
'=' : 'sip_authorization_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1266) : warning C4013:
'sip_proxy_authorization' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1266) : warning C4047:
'=' : 'sip_proxy_authorization_t *' differs in levels of indirection from
'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1271) : warning C4047:
'=' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1275) : warning C4013:
'sip_cseq_create' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1275) : warning C4047:
'=' : 'sip_cseq_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1279) : warning C4013:
'sip_add_tl' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1287) : warning C4013:
'sip_header_insert' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1294) : warning C4013:
'sip_header_remove' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1361) : warning C4013:
'SIPTAG_END' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2139) : warning C4013:
'sip_add_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2139) : error C2065:
'sip_accept_encoding_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2145) : error C2065:
'sip_accept_encoding_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2333) : warning C4013:
'sip_warning_format' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2334) : warning C4047:
'=' : 'sip_warning_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2710) : warning C4013:
'SIPTAG_REASON_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2710) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2710) : warning C4024:
'nua_server_trespond' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2711) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2711) : warning C4024:
'nua_server_trespond' : different types for formal and actual parameter 3

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2807) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2807) : warning C4024:
'nua_server_trespond' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2808) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2808) : warning C4024:
'nua_server_trespond' : different types for formal and actual parameter 3

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2967) : warning C4013:
'siptag_event_vr' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2984) : warning C4013:
'siptag_event_v' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2988) : warning C4013:
'sip_event_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2988) : warning C4047:
'=' : 'sip_event_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3048) : warning C4013:
'SIPTAG_EVENT' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3048) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3048) : warning C4024:
'nua_stack_post_signal' : different types for formal and actual parameter 3

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3049) : warning C4013:
'SIPTAG_SUBSCRIPTION_STATE_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3050) : warning C4013:
'SIPTAG_CONTENT_TYPE_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3051) : warning C4013:
'SIPTAG_PAYLOAD_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3752) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3752) : warning C4024:
'nua_base_client_trequest' : different types for formal and actual parameter
4

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3753) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_typedef_t'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3753) : warning C4024:
'nua_base_client_trequest' : different types for formal and actual parameter
5

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4214) : warning C4013:
'sip_retry_after_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4305) : warning C4013:
'sip_min_se_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4305) : error C2223:
left of '->min_delta' must point to struct/union

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4411) : error C2223:
left of '->min_delta' must point to struct/union

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4413) : warning C4013:
'sip_session_expires_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4413) : error C2223:
left of '->x_delta' must point to struct/union

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4424) : warning C4013:
'SIPTAG_SESSION_EXPIRES' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4424) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4428) : warning C4013:
'SIPTAG_MIN_SE' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4428) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4430) : warning C4013:
'SIPTAG_REQUIRE_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4430) : warning C4047:
':' : 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4624) : warning C4013:
'sip_payload_create' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4624) : warning C4047:
'=' : 'sip_payload_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4625) : warning C4013:
'sip_content_type_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4625) : warning C4047:
'=' : 'sip_content_type_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4627) : warning C4013:
'sip_content_disposition_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4627) : warning C4047:
'=' : 'sip_content_disposition_t *' differs in levels of indirection from
'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4702) : warning C4013:
'SIPTAG_ACCEPT' undefined; assuming extern returning int

42>cst_val_const.c

38>nua_stack.c

42>Generating Code...

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(152) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(152) : warning C4013:
'sip_from_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(174) : warning C4013:
'sip_accept_make' undefined; assuming extern returning int

42>Compiling...

42>cst_val_user.c

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(174) : warning C4047: '='
: 'sip_accept_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(396) : warning C4013:
'sip_object' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(396) : warning C4047: ':'
: 'int' differs in levels of indirection from 'void *'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(396) : warning C4047:
'function' : 'const sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(396) : warning C4024:
'function through pointer' : different types for formal and actual parameter
8

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(993) : error C2223: left
of '->a_display' must point to struct/union

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(996) : warning C4013:
'sip_to_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(996) : error C2223: left
of '->a_display' must point to struct/union

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(1001) : warning C4013:
'SIPTAG_TO' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(1002) : warning C4013:
'SIPTAG_FROM' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(1014) : warning C4013:
'SIPTAG_CALL_ID' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(1014) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(1014) : warning C4024:
'nta_leg_tcreate' : different types for formal and actual parameter 4

38>nua_subnotref.c

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_subnotref.c(50) : fatal error
C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or
directory

38>nua_tag.c

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_tag.c(2389) : error C2065:
'sip_event_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_tag.c(2389) : error C2099:
initializer is not a constant

38>Generating Code...

42>cst_vc.c

38>Compiling...

42>cst_viterbi.c

42>cst_voice.c

38>nua_tag_ref.c

38>outbound.c

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(405) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(405) : warning C4013:
'sip_contact_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(405) : warning C4047: '='
: 'sip_contact_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(523) : warning C4013:
'sip_via_port' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(523) : warning C4047: '='
: 'const char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(623) : warning C4013:
'sip_contact_format' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(623) : warning C4047: ':'
: 'int' differs in levels of indirection from 'void *'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(623) : warning C4047: '='
: 'sip_contact_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(667) : error C2065:
'sip_transport_udp' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(667) : warning C4047: '=='
: 'const char *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(709) : warning C4013:
'sip_object' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(709) : warning C4047:
'initializing' : 'const sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(739) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(754) : warning C4013:
'sip_accept_contact_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(754) : warning C4047: '='
: 'sip_accept_contact_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(773) : warning C4013:
'sip_add_tl' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(774) : warning C4013:
'SIPTAG_TO' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(775) : warning C4013:
'SIPTAG_FROM' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(780) : warning C4013:
'SIPTAG_ROUTE' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(782) : warning C4013:
'SIPTAG_MAX_FORWARDS_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(782) : warning C4047: ':'
: 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(783) : warning C4013:
'SIPTAG_SUBJECT_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(783) : warning C4047: ':'
: 'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(784) : warning C4013:
'SIPTAG_CALL_ID_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(785) : warning C4013:
'SIPTAG_ACCEPT_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(817) : warning C4047: '='
: 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(878) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(892) : error C2065:
'sip_authorization_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(892) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(892) : warning C4024:
'auc_challenge' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(897) : error C2065:
'sip_proxy_authorization_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(897) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(897) : warning C4024:
'auc_challenge' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1000) : warning C4047: '='
: 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1022) : warning C4013:
'SIPTAG_MAX_FORWARDS' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1061) : warning C4013:
'SIPTAG_CONTENT_TYPE_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1061) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1061) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1062) : warning C4013:
'SIPTAG_PAYLOAD_STR' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1101) : warning C4013:
'sip_via_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1101) : warning C4047: '='
: 'sip_via_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1187) : warning C4047: '='
: 'sip_contact_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1268) : warning C4013:
'sip_has_feature' undefined; assuming extern returning int

38>nta.c

42>cst_wave.c

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(909) : error C2220: warning
treated as error - no 'object' file generated

42>cst_wave_io.c

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(909) : warning C4013:
'sip_max_forwards_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(1494) : warning C4013:
'siptag_contact_vr' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(1564) : warning C4013:
'sip_contact_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(1564) : warning C4047: '=' :
'sip_contact_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(1776) : warning C4013:
'siptag_contact_v' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2379) : warning C4013:
'sip_via_format' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2384) : warning C4047: '=' :
'sip_via_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2395) : warning C4013:
'sip_via_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2395) : warning C4047: '=' :
'sip_via_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2397) : warning C4047: '=' :
'sip_via_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2534) : warning C4013:
'sip_object' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2534) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2548) : warning C4013:
'sip_via_copy' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2548) : warning C4047: '=' :
'sip_via_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2745) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2839) : error C2065:
'sip_error_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2839) : warning C4047: '==' :
'msg_hclass_t *const ' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3033) : warning C4047: '=' :
'sip_via_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3389) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3563) : warning C4047: '=' :
'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3647) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3648) : warning C4047:
'initializing' : 'const sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3658) : warning C4047: '=' :
'sip_via_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3660) : warning C4013:
'sip_from_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3660) : warning C4047: '=' :
'sip_from_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3662) : warning C4013:
'sip_to_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3662) : warning C4047: '=' :
'sip_to_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3665) : warning C4013:
'sip_call_id_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3665) : warning C4047: '=' :
'sip_call_id_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3667) : warning C4013:
'sip_cseq_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3667) : warning C4047: '=' :
'sip_cseq_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3705) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3707) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3721) : warning C4013:
'SIPTAG_TO' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3721) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3721) : warning C4024:
'sip_add_tl' : different types for formal and actual parameter 3

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3722) : warning C4013:
'SIPTAG_FROM' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3723) : warning C4013:
'SIPTAG_CALL_ID' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3740) : warning C4013:
'sip_route_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3740) : error C2223: left of
'->r_url' must point to struct/union

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3741) : warning C4013:
'sip_route_dup' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3741) : warning C4047: '=' :
'sip_route_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3750) : warning C4047: '=' :
'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3825) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3839) : warning C4047: '=' :
'sip_route_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3845) : warning C4047: '=' :
'sip_route_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3851) : warning C4047: '=' :
'sip_route_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3921) : warning C4047: '=' :
'sip_from_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3938) : warning C4047: '=' :
'sip_to_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3966) : warning C4047: '=' :
'sip_call_id_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4104) : warning C4013:
'SIPTAG_CALL_ID_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4105) : warning C4013:
'SIPTAG_CALL_ID_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4106) : warning C4013:
'SIPTAG_FROM_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4107) : warning C4013:
'SIPTAG_FROM_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4108) : warning C4013:
'SIPTAG_TO_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4109) : warning C4013:
'SIPTAG_TO_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4110) : warning C4013:
'SIPTAG_ROUTE_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4113) : warning C4013:
'SIPTAG_CSEQ_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4114) : warning C4013:
'SIPTAG_CSEQ_STR_REF' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4136) : warning C4013:
'sip_is_to' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4137) : warning C4047: '=' :
'sip_to_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4140) : warning C4013:
'sip_to_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4140) : warning C4047: '=' :
'sip_to_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4145) : warning C4047: '=' :
'sip_from_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4148) : warning C4047: '=' :
'sip_from_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4150) : warning C4013:
'sip_from_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4150) : warning C4047: '=' :
'sip_from_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4153) : warning C4047: '=' :
'sip_route_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4157) : warning C4013:
'sip_contact_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4160) : warning C4047: '=' :
'sip_contact_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4212) : warning C4047: '=' :
'sip_call_id_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4214) : warning C4013:
'sip_call_id_make' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4214) : warning C4047: '=' :
'sip_call_id_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4577) : warning C4013:
'sip_replaces_format' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4579) : warning C4047: 'return'
: 'sip_replaces_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4594) : warning C4013:
'sip_call_id_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5066) : warning C4047: '=' :
'sip_contact_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5217) : warning C4047: '=' :
'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5267) : warning C4013:
'sip_request_copy' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5267) : warning C4047: '=' :
'sip_request_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5268) : warning C4013:
'sip_from_copy' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5268) : warning C4047: '=' :
'sip_from_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5269) : warning C4013:
'sip_to_copy' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5269) : warning C4047: '=' :
'sip_to_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5270) : warning C4013:
'sip_call_id_copy' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5270) : warning C4047: '=' :
'sip_call_id_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5271) : warning C4013:
'sip_cseq_copy' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5271) : warning C4047: '=' :
'sip_cseq_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5272) : warning C4047: '=' :
'sip_via_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5285) : warning C4013:
'sip_record_route_copy' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5285) : warning C4047: '=' :
'sip_record_route_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5292) : warning C4013:
'sip_timestamp_copy' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5292) : warning C4047: '=' :
'sip_timestamp_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6307) : warning C4047: '=' :
'sip_from_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6309) : warning C4047: '=' :
'sip_to_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6311) : warning C4047: '=' :
'sip_call_id_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6313) : warning C4047: '=' :
'sip_cseq_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6315) : warning C4047: '=' :
'sip_via_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6346) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6404) : warning C4047: '=' :
'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6482) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(7221) : warning C4047: '=' :
'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(7312) : warning C4047:
'function' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(7312) : warning C4024:
'sip_add_tl' : different types for formal and actual parameter 2

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(7682) : warning C4047: '=' :
'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(7858) : warning C4047:
'initializing' : 'const sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8052) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8054) : warning C4013:
'sip_timestamp_format' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8055) : warning C4047:
'initializing' : 'sip_timestamp_t *' differs in levels of indirection from
'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8194) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8240) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8358) : warning C4013:
'sip_supported_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9443) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9443) : warning C4024:
'outgoing_ackmsg' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9443) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9443) : warning C4024:
'outgoing_ackmsg' : different types for formal and actual parameter 5

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9463) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9464) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9482) : warning C4013:
'sip_header_remove' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9605) : warning C4047: '=' :
'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9618) : warning C4013:
'sip_retry_after_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9618) : error C2223: left of
'->af_delta' must point to struct/union

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10794) : warning C4047: '=' :
'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10828) : warning C4047:
'initializing' : 'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10907) : warning C4013:
'sip_rseq_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10912) : error C2065:
'sip_require_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10912) : warning C4047:
'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10912) : warning C4024:
'sip_add_make' : different types for formal and actual parameter 3

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11075) : warning C4047: '=' :
'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11335) : warning C4047: '=' :
'sip_to_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11421) : warning C4013:
'sip_rack_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11476) : warning C4047: '=' :
'sip_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11484) : error C2223: left of
'->r_url' must point to struct/union

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11485) : warning C4047: '=' :
'sip_route_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11491) : warning C4047: '=' :
'sip_route_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11515) : warning C4013:
'SIPTAG_RACK' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11515) : warning C4047: ':' :
'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11516) : warning C4047: ':' :
'const tag_type_s *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11516) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from 'const
tag_type_s *'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11516) : warning C4024:
'sip_add_tl' : different types for formal and actual parameter 4

38>nta_check.c

42>cst_wave_utils.c

42>flite.c

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(83) : error C2220:
warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(83) : warning C4013:
'SIPTAG_UNSUPPORTED' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(83) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(83) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(84) : warning C4013:
'SIPTAG_SUPPORTED' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(124) : warning C4013:
'SIPTAG_REQUIRE' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(124) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(124) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(125) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(125) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 5

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(169) : warning C4013:
'SIPTAG_ALLOW' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(169) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(169) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(170) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(170) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 5

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(175) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(175) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(176) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(176) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 5

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(262) : warning C4013:
'SIPTAG_ACCEPT' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(262) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(262) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(263) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(263) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 5

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(335) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(335) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(336) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(336) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 5

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(376) : warning C4013:
'sip_min_se_init' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(376) : error C2223: left
of '->min_delta' must point to struct/union

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(380) : warning C4013:
'SIPTAG_MIN_SE' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(380) : warning C4047:
'function' : 'tag_type_t' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(380) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 4

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(381) : warning C4047:
'function' : 'tag_value_t' differs in levels of indirection from
'tag_type_t'

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(381) : warning C4024:
'nta_incoming_treply' : different types for formal and actual parameter 5

38>nta_tag.c

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(178) : error C2065:
'sip_contact_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(178) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(184) : error C2065:
'sip_contact_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(184) : error C2099:
initializer is not a constant

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(191) : error C2065:
'sip_contact_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(191) : error C2099:
initializer is not a constant

38>nta_tag_ref.c

42>rateconv.c

38>sl_read_payload.c

38>sl_utils_log.c

42>regexp.c

38>sl_utils_print.c

38>tport.c

42>regsub.c

42>us_aswd.c

38>tport_logging.c

38>tport_stub_sigcomp.c

42>us_dur_stats.c

38>tport_stub_stun.c

42>us_durz_cart.c

38>tport_tag.c

42>us_expand.c

38>tport_tag_ref.c

38>tport_type_connect.c

42>us_f0_model.c

38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(173) : error
C2220: warning treated as error - no 'object' file generated

38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(173) : warning
C4013: 'http_request_format' undefined; assuming extern returning int

38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(173) : warning
C4047: '=' : 'http_request_t *' differs in levels of indirection from 'int'

38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(179) : error
C2065: 'http_host_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(179) : warning
C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from
'int'

38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(179) : warning
C4024: 'msg_header_add_make' : different types for formal and actual
parameter 3

38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(180) : error
C2065: 'http_separator_class' : undeclared identifier

38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(180) : warning
C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from
'int'

38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(180) : warning
C4024: 'msg_header_add_make' : different types for formal and actual
parameter 3

38>tport_type_tcp.c

42>us_f0lr.c

38>tport_type_udp.c

38>sdp.c

42>us_ffeatures.c

38>sdp_parse.c

42>us_gpos.c

38>Generating Code...

42>us_int_accent_cart.c

38>Compiling...

38>sdp_print.c

38>sdp_tag.c

38>sdp_tag_ref.c

38>soa.c

42>Generating Code...

42>Compiling...

42>us_int_tone_cart.c

38>soa_static.c

42>us_nums_cart.c

38>soa_tag.c

42>us_phoneset.c

38>soa_tag_ref.c

38>inet_ntop.c

42>us_phrasing_cart.c

38>Generating Code...

38>Creating browse information file...

42>us_text.c

38>Microsoft Browse Information Maintenance Utility Version 9.00.30729

38>Copyright (C) Microsoft Corporation. All rights reserved.

38>BSCMAKE: error BK1506 : cannot open file '.\Debug\sip_basic.sbr': No such
file or directory

38>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\sofia\Debug\BuildLog.htm"

38>libsofia_sip_ua_static - 899 error(s), 1733 warning(s)

42>usenglish.c

43>------ Build started: Project: portaudio, Configuration: Debug Win32
------

43>Compiling...

43>pa_converters.c

43>pa_cpuload.c

43>pa_debugprint.c

43>pa_dither.c

43>pa_front.c

42>cmu_us_slt.c

43>pa_process.c

43>pa_skeleton.c

43>pa_stream.c

43>pa_trace.c

43>pa_win_wmme.c

42>cmu_us_slt_cg.c

43>pa_win_hostapis.c

42>cmu_us_slt_cg_durmodel.c

43>pa_win_util.c

43>pa_win_waveformat.c

42>cmu_us_slt_cg_f0_trees.c

43>pa_win_wdmks_utils.c

42>cmu_us_slt_cg_mcep_trees.c

43>Generating Code...

43>Compiling...

43>pa_x86_plain_converters.c

43>pa_allocation.c

43>Generating Code...

42>cmu_us_slt_cg_params.c

43>Creating library...

43>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\portaudio\build\msvc\Win32\Debug
\BuildLog.htm"

43>portaudio - 0 error(s), 0 warning(s)

44>------ Skipped Build: Project: Download 32khzsound, Configuration: Debug
Win32 ------

44>Project not selected to build for this solution configuration 

45>------ Build started: Project: mod_spidermonkey, Configuration: Debug
Win32 ------

45>Compiling...

45>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

45>mod_spidermonkey.c

42>cmu_us_slt_cg_phonestate.c

42>cmu_us_awb.c

45>Linking...

45>   Creating library Win32\Debug\mod_spidermonkey.lib and object
Win32\Debug\mod_spidermonkey.exp

45>Embedding manifest...

42>cmu_us_awb_cg.c

42>cmu_us_awb_cg_durmodel.c

45>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_spidermonkey\Wi
n32\Debug\BuildLog.htm"

45>mod_spidermonkey - 0 error(s), 1 warning(s)

46>------ Build started: Project: libspeex, Configuration: Debug Win32
------

46>Compiling...

46>cb_search.c

42>cmu_us_awb_cg_f0_trees.c

46>exc_10_16_table.c

46>exc_10_32_table.c

46>exc_20_32_table.c

46>exc_5_256_table.c

46>exc_5_64_table.c

46>exc_8_128_table.c

46>filters.c

46>gain_table.c

46>gain_table_lbr.c

46>hexc_10_32_table.c

46>hexc_table.c

46>high_lsp_tables.c

46>lpc.c

46>lsp.c

46>lsp_tables_nb.c

46>ltp.c

46>modes.c

42>cmu_us_awb_cg_mcep_trees.c

46>modes_wb.c

46>nb_celp.c

46>Generating Code...

46>Compiling...

46>quant_lsp.c

46>sb_celp.c

46>speex.c

46>speex_callbacks.c

46>speex_header.c

46>stereo.c

46>vbr.c

46>vq.c

46>window.c

46>bits.c

42>cmu_us_awb_cg_params.c

46>Generating Code...

46>Creating library...

46>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\speex\win32\VS2008\libspeex\Debu
g\BuildLog.htm"

46>libspeex - 0 error(s), 0 warning(s)

47>------ Skipped Build: Project: Download 16khzsound, Configuration: Debug
Win32 ------

47>Project not selected to build for this solution configuration 

48>------ Build started: Project: Download 8khzsound, Configuration: Debug
Win32 ------

48>Downloading 8khzsound.

48>Sound name: en-us-callie Version 1.0.12

48>URL:
http://files.freeswitch.org/freeswitch-sounds-en-us-callie-8000-1.0.12.tar.g
z

48>Downloading:
http://files.freeswitch.org/freeswitch-sounds-en-us-callie-8000-1.0.12.tar.g
z

42>cmu_us_awb_cg_phonestate.c

42>Generating Code...

42>Compiling...

42>cmu_us_rms.c

48>Extracting:
C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\freeswitch-sounds-en-us-callie-80
00-1.0.12.tar.gz

48>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

48>Processing archive:
C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\freeswitch-sounds-en-us-callie-80
00-1.0.12.tar.gz

48>Extracting  freeswitch-sounds-en-us-callie-8000-1.0.12.tar

48>Everything is Ok

42>cmu_us_rms_cg.c

48>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09

48>Processing archive:
C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\freeswitch-sounds-en-us-callie-80
00-1.0.12.tar

48>Extracting  en

48>Extracting  en\us

48>Extracting  en\us\callie

48>Extracting  en\us\callie\time

48>Extracting  en\us\callie\time\8000

48>Extracting  en\us\callie\time\8000\day-5.wav

48>Extracting  en\us\callie\time\8000\seconds.wav

48>Extracting  en\us\callie\time\8000\day-1.wav

48>Extracting  en\us\callie\time\8000\mon-5.wav

48>Extracting  en\us\callie\time\8000\today.wav

48>Extracting  en\us\callie\time\8000\mon-11.wav

48>Extracting  en\us\callie\time\8000\minutes.wav

48>Extracting  en\us\callie\time\8000\mon-10.wav

48>Extracting  en\us\callie\time\8000\yesterday.wav

48>Extracting  en\us\callie\time\8000\mon-1.wav

48>Extracting  en\us\callie\time\8000\day-3.wav

48>Extracting  en\us\callie\time\8000\oh.wav

48>Extracting  en\us\callie\time\8000\second.wav

48>Extracting  en\us\callie\time\8000\minute.wav

48>Extracting  en\us\callie\time\8000\day-4.wav

48>Extracting  en\us\callie\time\8000\a-m.wav

48>Extracting  en\us\callie\time\8000\mon-3.wav

48>Extracting  en\us\callie\time\8000\mon-8.wav

48>Extracting  en\us\callie\time\8000\at.wav

48>Extracting  en\us\callie\time\8000\tomorrow.wav

48>Extracting  en\us\callie\time\8000\mon-9.wav

48>Extracting  en\us\callie\time\8000\day-2.wav

48>Extracting  en\us\callie\time\8000\mon-4.wav

48>Extracting  en\us\callie\time\8000\p-m.wav

48>Extracting  en\us\callie\time\8000\hours.wav

48>Extracting  en\us\callie\time\8000\day-0.wav

48>Extracting  en\us\callie\time\8000\mon-7.wav

48>Extracting  en\us\callie\time\8000\mon-2.wav

48>Extracting  en\us\callie\time\8000\hour.wav

48>Extracting  en\us\callie\time\8000\oclock.wav

48>Extracting  en\us\callie\time\8000\day-6.wav

48>Extracting  en\us\callie\time\8000\mon-0.wav

48>Extracting  en\us\callie\time\8000\mon-6.wav

48>Extracting  en\us\callie\currency

48>Extracting  en\us\callie\currency\8000

48>Extracting  en\us\callie\currency\8000\dollar.wav

48>Extracting  en\us\callie\currency\8000\dollars.wav

48>Extracting  en\us\callie\currency\8000\minus.wav

48>Extracting  en\us\callie\currency\8000\cents-per-minute.wav

48>Extracting  en\us\callie\currency\8000\central.wav

48>Extracting  en\us\callie\currency\8000\cents.wav

48>Extracting  en\us\callie\currency\8000\cent.wav

48>Extracting  en\us\callie\currency\8000\negative.wav

48>Extracting  en\us\callie\currency\8000\and.wav

48>Extracting  en\us\callie\digits

48>Extracting  en\us\callie\digits\8000

48>Extracting  en\us\callie\digits\8000\h-10.wav

48>Extracting  en\us\callie\digits\8000\h-12.wav

48>Extracting  en\us\callie\digits\8000\h-7.wav

48>Extracting  en\us\callie\digits\8000\19.wav

48>Extracting  en\us\callie\digits\8000\dot.wav

48>Extracting  en\us\callie\digits\8000\40.wav

48>Extracting  en\us\callie\digits\8000\7.wav

48>Extracting  en\us\callie\digits\8000\12.wav

48>Extracting  en\us\callie\digits\8000\h-17.wav

48>Extracting  en\us\callie\digits\8000\h-13.wav

48>Extracting  en\us\callie\digits\8000\30.wav

48>Extracting  en\us\callie\digits\8000\70.wav

48>Extracting  en\us\callie\digits\8000\60.wav

48>Extracting  en\us\callie\digits\8000\h-14.wav

48>Extracting  en\us\callie\digits\8000\h-18.wav

48>Extracting  en\us\callie\digits\8000\9.wav

48>Extracting  en\us\callie\digits\8000\20.wav

48>Extracting  en\us\callie\digits\8000\6.wav

48>Extracting  en\us\callie\digits\8000\h-4.wav

48>Extracting  en\us\callie\digits\8000\1.wav

48>Extracting  en\us\callie\digits\8000\h-3.wav

48>Extracting  en\us\callie\digits\8000\h-6.wav

48>Extracting  en\us\callie\digits\8000\period.wav

48>Extracting  en\us\callie\digits\8000\50.wav

48>Extracting  en\us\callie\digits\8000\h-9.wav

48>Extracting  en\us\callie\digits\8000\2.wav

48>Extracting  en\us\callie\digits\8000\point.wav

48>Extracting  en\us\callie\digits\8000\h-2.wav

48>Extracting  en\us\callie\digits\8000\18.wav

48>Extracting  en\us\callie\digits\8000\h-5.wav

48>Extracting  en\us\callie\digits\8000\3.wav

48>Extracting  en\us\callie\digits\8000\80.wav

48>Extracting  en\us\callie\digits\8000\h-16.wav

48>Extracting  en\us\callie\digits\8000\15.wav

48>Extracting  en\us\callie\digits\8000\hundred.wav

48>Extracting  en\us\callie\digits\8000\10.wav

48>Extracting  en\us\callie\digits\8000\h-8.wav

48>Extracting  en\us\callie\digits\8000\8.wav

42>cmu_us_rms_cg_durmodel.c

48>Extracting  en\us\callie\digits\8000\star.wav

48>Extracting  en\us\callie\digits\8000\17.wav

48>Extracting  en\us\callie\digits\8000\h-15.wav

48>Extracting  en\us\callie\digits\8000\h-11.wav

48>Extracting  en\us\callie\digits\8000\h-19.wav

48>Extracting  en\us\callie\digits\8000\16.wav

48>Extracting  en\us\callie\digits\8000\million.wav

48>Extracting  en\us\callie\digits\8000\4.wav

48>Extracting  en\us\callie\digits\8000\13.wav

48>Extracting  en\us\callie\digits\8000\pound.wav

48>Extracting  en\us\callie\digits\8000\11.wav

48>Extracting  en\us\callie\digits\8000\h-1.wav

48>Extracting  en\us\callie\digits\8000\90.wav

48>Extracting  en\us\callie\digits\8000\h-30.wav

48>Extracting  en\us\callie\digits\8000\thousand.wav

48>Extracting  en\us\callie\digits\8000\0.wav

48>Extracting  en\us\callie\digits\8000\14.wav

48>Extracting  en\us\callie\digits\8000\5.wav

48>Extracting  en\us\callie\digits\8000\h-20.wav

48>Extracting  en\us\callie\base256

48>Extracting  en\us\callie\base256\8000

48>Extracting  en\us\callie\base256\8000\adviser.wav

48>Extracting  en\us\callie\base256\8000\python.wav

48>Extracting  en\us\callie\base256\8000\bookshelf.wav

48>Extracting  en\us\callie\base256\8000\consensus.wav

48>Extracting  en\us\callie\base256\8000\fallout.wav

48>Extracting  en\us\callie\base256\8000\detector.wav

48>Extracting  en\us\callie\base256\8000\crumpled.wav

48>Extracting  en\us\callie\base256\8000\dreadful.wav

48>Extracting  en\us\callie\base256\8000\paragraph.wav

48>Extracting  en\us\callie\base256\8000\skydive.wav

48>Extracting  en\us\callie\base256\8000\clergyman.wav

48>Extracting  en\us\callie\base256\8000\detergent.wav

48>Extracting  en\us\callie\base256\8000\prowler.wav

48>Extracting  en\us\callie\base256\8000\klaxon.wav

48>Extracting  en\us\callie\base256\8000\penetrate.wav

48>Extracting  en\us\callie\base256\8000\Apollo.wav

48>Extracting  en\us\callie\base256\8000\topmost.wav

48>Extracting  en\us\callie\base256\8000\conformist.wav

48>Extracting  en\us\callie\base256\8000\crossover.wav

48>Extracting  en\us\callie\base256\8000\repay.wav

48>Extracting  en\us\callie\base256\8000\fracture.wav

48>Extracting  en\us\callie\base256\8000\willow.wav

48>Extracting  en\us\callie\base256\8000\pedigree.wav

48>Extracting  en\us\callie\base256\8000\microwave.wav

48>Extracting  en\us\callie\base256\8000\inception.wav

48>Extracting  en\us\callie\base256\8000\Jamaica.wav

48>Extracting  en\us\callie\base256\8000\puppy.wav

48>Extracting  en\us\callie\base256\8000\showgirl.wav

48>Extracting  en\us\callie\base256\8000\dictator.wav

48>Extracting  en\us\callie\base256\8000\gadgetry.wav

48>Extracting  en\us\callie\base256\8000\Yucatan.wav

48>Extracting  en\us\callie\base256\8000\orca.wav

48>Extracting  en\us\callie\base256\8000\getaway.wav

48>Extracting  en\us\callie\base256\8000\dropper.wav

48>Extracting  en\us\callie\base256\8000\jawbone.wav

48>Extracting  en\us\callie\base256\8000\snapshot.wav

48>Extracting  en\us\callie\base256\8000\unwind.wav

48>Extracting  en\us\callie\base256\8000\optic.wav

48>Extracting  en\us\callie\base256\8000\designing.wav

48>Extracting  en\us\callie\base256\8000\payday.wav

48>Extracting  en\us\callie\base256\8000\truncated.wav

48>Extracting  en\us\callie\base256\8000\antenna.wav

48>Extracting  en\us\callie\base256\8000\briefcase.wav

48>Extracting  en\us\callie\base256\8000\Aztec.wav

48>Extracting  en\us\callie\base256\8000\coherence.wav

48>Extracting  en\us\callie\base256\8000\company.wav

48>Extracting  en\us\callie\base256\8000\therapist.wav

48>Extracting  en\us\callie\base256\8000\sensation.wav

48>Extracting  en\us\callie\base256\8000\megaton.wav

48>Extracting  en\us\callie\base256\8000\Eskimo.wav

48>Extracting  en\us\callie\base256\8000\Capricorn.wav

48>Extracting  en\us\callie\base256\8000\artist.wav

48>Extracting  en\us\callie\base256\8000\intention.wav

48>Extracting  en\us\callie\base256\8000\tissue.wav

48>Extracting  en\us\callie\base256\8000\quota.wav

48>Extracting  en\us\callie\base256\8000\vagabond.wav

48>Extracting  en\us\callie\base256\8000\phonetic.wav

48>Extracting  en\us\callie\base256\8000\village.wav

48>Extracting  en\us\callie\base256\8000\revenge.wav

48>Extracting  en\us\callie\base256\8000\celebrate.wav

48>Extracting  en\us\callie\base256\8000\exodus.wav

48>Extracting  en\us\callie\base256\8000\kiwi.wav

42>cmu_us_rms_cg_f0_trees.c

48>Extracting  en\us\callie\base256\8000\ancient.wav

48>Extracting  en\us\callie\base256\8000\supportive.wav

48>Extracting  en\us\callie\base256\8000\politeness.wav

48>Extracting  en\us\callie\base256\8000\prefer.wav

48>Extracting  en\us\callie\base256\8000\aimless.wav

48>Extracting  en\us\callie\base256\8000\infancy.wav

48>Extracting  en\us\callie\base256\8000\hurricane.wav

48>Extracting  en\us\callie\base256\8000\universe.wav

48>Extracting  en\us\callie\base256\8000\upset.wav

48>Extracting  en\us\callie\base256\8000\councilman.wav

48>Extracting  en\us\callie\base256\8000\component.wav

48>Extracting  en\us\callie\base256\8000\Waterloo.wav

48>Extracting  en\us\callie\base256\8000\molecule.wav

48>Extracting  en\us\callie\base256\8000\sandalwood.wav

48>Extracting  en\us\callie\base256\8000\cannonball.wav

48>Extracting  en\us\callie\base256\8000\undaunted.wav

48>Extracting  en\us\callie\base256\8000\waffle.wav

48>Extracting  en\us\callie\base256\8000\mosquito.wav

48>Extracting  en\us\callie\base256\8000\tolerance.wav

48>Extracting  en\us\callie\base256\8000\retraction.wav

48>Extracting  en\us\callie\base256\8000\drumbeat.wav

48>Extracting  en\us\callie\base256\8000\sweatband.wav

48>Extracting  en\us\callie\base256\8000\tracker.wav

48>Extracting  en\us\callie\base256\8000\nightbird.wav

48>Extracting  en\us\callie\base256\8000\music.wav

48>Extracting  en\us\callie\base256\8000\direction.wav

48>Extracting  en\us\callie\base256\8000\glossary.wav

48>Extracting  en\us\callie\base256\8000\cumbersome.wav

48>Extracting  en\us\callie\base256\8000\flatfoot.wav

48>Extracting  en\us\callie\base256\8000\travesty.wav

48>Extracting  en\us\callie\base256\8000\minnow.wav

48>Extracting  en\us\callie\base256\8000\cement.wav

48>Extracting  en\us\callie\base256\8000\cubic.wav

48>Extracting  en\us\callie\base256\8000\egghead.wav

48>Extracting  en\us\callie\base256\8000\autopsy.wav

48>Extracting  en\us\callie\base256\8000\souvenir.wav

48>Extracting  en\us\callie\base256\8000\sawdust.wav

48>Extracting  en\us\callie\base256\8000\vapor.wav

48>Extracting  en\us\callie\base256\8000\concurrent.wav

48>Extracting  en\us\callie\base256\8000\revolver.wav

48>Extracting  en\us\callie\base256\8000\reproduce.wav

48>Extracting  en\us\callie\base256\8000\enchanting.wav

48>Extracting  en\us\callie\base256\8000\Chicago.wav

48>Extracting  en\us\callie\base256\8000\offload.wav

48>Extracting  en\us\callie\base256\8000\stormy.wav

48>Extracting  en\us\callie\base256\8000\locale.wav

48>Extracting  en\us\callie\base256\8000\holiness.wav

48>Extracting  en\us\callie\base256\8000\recipe.wav

48>Extracting  en\us\callie\base256\8000\spheroid.wav

48>Extracting  en\us\callie\base256\8000\breakaway.wav

48>Extracting  en\us\callie\base256\8000\Dakota.wav

48>Extracting  en\us\callie\base256\8000\specialist.wav

48>Extracting  en\us\callie\base256\8000\whimsical.wav

48>Extracting  en\us\callie\base256\8000\backwater.wav

48>Extracting  en\us\callie\base256\8000\Zulu.wav

48>Extracting  en\us\callie\base256\8000\spellbind.wav

48>Extracting  en\us\callie\base256\8000\Athens.wav

48>Extracting  en\us\callie\base256\8000\vocalist.wav

48>Extracting  en\us\callie\base256\8000\standard.wav

48>Extracting  en\us\callie\base256\8000\insincere.wav

48>Extracting  en\us\callie\base256\8000\talon.wav

48>Extracting  en\us\callie\base256\8000\cleanup.wav

48>Extracting  en\us\callie\base256\8000\atlas.wav

48>Extracting  en\us\callie\base256\8000\provincial.wav

48>Extracting  en\us\callie\base256\8000\chambermaid.wav

48>Extracting  en\us\callie\base256\8000\revival.wav

48>Extracting  en\us\callie\base256\8000\stagehand.wav

48>Extracting  en\us\callie\base256\8000\millionaire.wav

48>Extracting  en\us\callie\base256\8000\paragon.wav

48>Extracting  en\us\callie\base256\8000\midsummer.wav

48>Extracting  en\us\callie\base256\8000\hamlet.wav

48>Extracting  en\us\callie\base256\8000\brickyard.wav

48>Extracting  en\us\callie\base256\8000\highchair.wav

48>Extracting  en\us\callie\base256\8000\retrieval.wav

48>Extracting  en\us\callie\base256\8000\bodyguard.wav

48>Extracting  en\us\callie\base256\8000\banjo.wav

48>Extracting  en\us\callie\base256\8000\positive.wav

48>Extracting  en\us\callie\base256\8000\keyboard.wav

48>Extracting  en\us\callie\base256\8000\flytrap.wav

48>Extracting  en\us\callie\base256\8000\enlist.wav

48>Extracting  en\us\callie\base256\8000\savagery.wav

48>Extracting  en\us\callie\base256\8000\slowdown.wav

48>Extracting  en\us\callie\base256\8000\Oakland.wav

48>Extracting  en\us\callie\base256\8000\amulet.wav

48>Extracting  en\us\callie\base256\8000\beeswax.wav

48>Extracting  en\us\callie\base256\8000\impetus.wav

48>Extracting  en\us\callie\base256\8000\candidate.wav

48>Extracting  en\us\callie\base256\8000\steamship.wav

48>Extracting  en\us\callie\base256\8000\befriend.wav

48>Extracting  en\us\callie\base256\8000\aggregate.wav

48>Extracting  en\us\callie\base256\8000\dwelling.wav

48>Extracting  en\us\callie\base256\8000\customer.wav

48>Extracting  en\us\callie\base256\8000\Neptune.wav

48>Extracting  en\us\callie\base256\8000\transit.wav

48>Extracting  en\us\callie\base256\8000\rebirth.wav

48>Extracting  en\us\callie\base256\8000\swelter.wav

48>Extracting  en\us\callie\base256\8000\hazardous.wav

48>Extracting  en\us\callie\base256\8000\Wyoming.wav

48>Extracting  en\us\callie\base256\8000\warranty.wav

48>Extracting  en\us\callie\base256\8000\performance.wav

48>Extracting  en\us\callie\base256\8000\trouble.wav

48>Extracting  en\us\callie\base256\8000\torpedo.wav

48>Extracting  en\us\callie\base256\8000\tambourine.wav

48>Extracting  en\us\callie\base256\8000\indulge.wav

48>Extracting  en\us\callie\base256\8000\eightball.wav

48>Extracting  en\us\callie\base256\8000\bedlamp.wav

48>Extracting  en\us\callie\base256\8000\accrue.wav

48>Extracting  en\us\callie\base256\8000\apple.wav

48>Extracting  en\us\callie\base256\8000\Virginia.wav

48>Extracting  en\us\callie\base256\8000\preclude.wav

48>Extracting  en\us\callie\base256\8000\southward.wav

48>Extracting  en\us\callie\base256\8000\backfield.wav

48>Extracting  en\us\callie\base256\8000\ahead.wav

48>Extracting  en\us\callie\base256\8000\tycoon.wav

48>Extracting  en\us\callie\base256\8000\ultimate.wav

48>Extracting  en\us\callie\base256\8000\scavenger.wav

48>Extracting  en\us\callie\base256\8000\maritime.wav

48>Extracting  en\us\callie\base256\8000\surrender.wav

48>Extracting  en\us\callie\base256\8000\chopper.wav

48>Extracting  en\us\callie\base256\8000\escapade.wav

48>Extracting  en\us\callie\base256\8000\pocketful.wav

48>Extracting  en\us\callie\base256\8000\sociable.wav

48>Extracting  en\us\callie\base256\8000\Scotland.wav

48>Extracting  en\us\callie\base256\8000\Mohawk.wav

48>Extracting  en\us\callie\base256\8000\classroom.wav

48>Extracting  en\us\callie\base256\8000\yesteryear.wav

48>Extracting  en\us\callie\base256\8000\eating.wav

48>Extracting  en\us\callie\base256\8000\stopwatch.wav

48>Extracting  en\us\callie\base256\8000\physique.wav

48>Extracting  en\us\callie\base256\8000\indoors.wav

48>Extracting  en\us\callie\base256\8000\select.wav

48>Extracting  en\us\callie\base256\8000\merit.wav

48>Extracting  en\us\callie\base256\8000\rhythm.wav

48>Extracting  en\us\callie\base256\8000\lockup.wav

48>Extracting  en\us\callie\base256\8000\embezzle.wav

48>Extracting  en\us\callie\base256\8000\tactics.wav

48>Extracting  en\us\callie\base256\8000\forever.wav

48>Extracting  en\us\callie\base256\8000\tunnel.wav

48>Extracting  en\us\callie\base256\8000\congregate.wav

48>Extracting  en\us\callie\base256\8000\commence.wav

48>Extracting  en\us\callie\base256\8000\crowfoot.wav

48>Extracting  en\us\callie\base256\8000\quadrant.wav

48>Extracting  en\us\callie\base256\8000\examine.wav

48>Extracting  en\us\callie\base256\8000\retouch.wav

48>Extracting  en\us\callie\base256\8000\matchmaker.wav

48>Extracting  en\us\callie\base256\8000\recover.wav

48>Extracting  en\us\callie\base256\8000\deadbolt.wav

48>Extracting  en\us\callie\base256\8000\virus.wav

48>Extracting  en\us\callie\base256\8000\brackish.wav

48>Extracting  en\us\callie\base256\8000\wayside.wav

48>Extracting  en\us\callie\base256\8000\watchword.wav

48>Extracting  en\us\callie\base256\8000\chairlift.wav

48>Extracting  en\us\callie\base256\8000\unearth.wav

48>Extracting  en\us\callie\base256\8000\Orlando.wav

48>Extracting  en\us\callie\base256\8000\hamburger.wav

48>Extracting  en\us\callie\base256\8000\beaming.wav

48>Extracting  en\us\callie\base256\8000\dinosaur.wav

48>Extracting  en\us\callie\base256\8000\frighten.wav

48>Extracting  en\us\callie\base256\8000\trombonist.wav

48>Extracting  en\us\callie\base256\8000\eyeglass.wav

48>Extracting  en\us\callie\base256\8000\baboon.wav

48>Extracting  en\us\callie\base256\8000\belowground.wav

42>cmu_us_rms_cg_mcep_trees.c

48>Extracting  en\us\callie\base256\8000\berserk.wav

48>Extracting  en\us\callie\base256\8000\upshot.wav

48>Extracting  en\us\callie\base256\8000\perceptive.wav

48>Extracting  en\us\callie\base256\8000\hesitate.wav

48>Extracting  en\us\callie\base256\8000\applicant.wav

48>Extracting  en\us\callie\base256\8000\aardvark.wav

48>Extracting  en\us\callie\base256\8000\publisher.wav

48>Extracting  en\us\callie\base256\8000\opulent.wav

48>Extracting  en\us\callie\base256\8000\ringbolt.wav

48>Extracting  en\us\callie\base256\8000\Brazilian.wav

48>Extracting  en\us\callie\base256\8000\onlooker.wav

48>Extracting  en\us\callie\base256\8000\Wichita.wav

48>Extracting  en\us\callie\base256\8000\breakup.wav

48>Extracting  en\us\callie\base256\8000\crusade.wav

48>Extracting  en\us\callie\base256\8000\hemisphere.wav

48>Extracting  en\us\callie\base256\8000\nebula.wav

48>Extracting  en\us\callie\base256\8000\tempest.wav

48>Extracting  en\us\callie\base256\8000\decadence.wav

48>Extracting  en\us\callie\base256\8000\glucose.wav

48>Extracting  en\us\callie\base256\8000\stairway.wav

48>Extracting  en\us\callie\base256\8000\printer.wav

48>Extracting  en\us\callie\base256\8000\typewriter.wav

48>Extracting  en\us\callie\base256\8000\bison.wav

48>Extracting  en\us\callie\base256\8000\adult.wav

48>Extracting  en\us\callie\base256\8000\potato.wav

48>Extracting  en\us\callie\base256\8000\gossamer.wav

48>Extracting  en\us\callie\base256\8000\sailboat.wav

48>Extracting  en\us\callie\base256\8000\assume.wav

48>Extracting  en\us\callie\base256\8000\pupil.wav

48>Extracting  en\us\callie\base256\8000\shadow.wav

48>Extracting  en\us\callie\base256\8000\Medusa.wav

48>Extracting  en\us\callie\base256\8000\businessman.wav

48>Extracting  en\us\callie\base256\8000\Trojan.wav

48>Extracting  en\us\callie\base256\8000\surmount.wav

48>Extracting  en\us\callie\base256\8000\exceed.wav

48>Extracting  en\us\callie\base256\8000\Vulcan.wav

48>Extracting  en\us\callie\base256\8000\newsletter.wav

48>Extracting  en\us\callie\base256\8000\filament.wav

48>Extracting  en\us\callie\base256\8000\informant.wav

48>Extracting  en\us\callie\base256\8000\afflict.wav

48>Extracting  en\us\callie\base256\8000\monument.wav

48>Extracting  en\us\callie\base256\8000\enterprise.wav

48>Extracting  en\us\callie\base256\8000\miser.wav

48>Extracting  en\us\callie\base256\8000\guitarist.wav

48>Extracting  en\us\callie\base256\8000\suspense.wav

48>Extracting  en\us\callie\base256\8000\chatter.wav

48>Extracting  en\us\callie\base256\8000\indigo.wav

48>Extracting  en\us\callie\base256\8000\ammo.wav

48>Extracting  en\us\callie\base256\8000\Bradbury.wav

48>Extracting  en\us\callie\base256\8000\commando.wav

48>Extracting  en\us\callie\base256\8000\certify.wav

48>Extracting  en\us\callie\base256\8000\hockey.wav

48>Extracting  en\us\callie\base256\8000\headwaters.wav

48>Extracting  en\us\callie\base256\8000\unravel.wav

48>Extracting  en\us\callie\base256\8000\bravado.wav

48>Extracting  en\us\callie\base256\8000\armistice.wav

48>Extracting  en\us\callie\base256\8000\liberty.wav

48>Extracting  en\us\callie\base256\8000\treadmill.wav

48>Extracting  en\us\callie\base256\8000\alone.wav

48>Extracting  en\us\callie\base256\8000\Pluto.wav

48>Extracting  en\us\callie\base256\8000\cowbell.wav

48>Extracting  en\us\callie\base256\8000\corrosion.wav

48>Extracting  en\us\callie\base256\8000\newborn.wav

48>Extracting  en\us\callie\base256\8000\Galveston.wav

48>Extracting  en\us\callie\base256\8000\aftermath.wav

48>Extracting  en\us\callie\base256\8000\preshrunk.wav

48>Extracting  en\us\callie\base256\8000\racketeer.wav

48>Extracting  en\us\callie\base256\8000\almighty.wav

48>Extracting  en\us\callie\base256\8000\stupendous.wav

48>Extracting  en\us\callie\base256\8000\equipment.wav

48>Extracting  en\us\callie\base256\8000\sympathy.wav

48>Extracting  en\us\callie\base256\8000\deckhand.wav

48>Extracting  en\us\callie\base256\8000\barbecue.wav

48>Extracting  en\us\callie\base256\8000\Belfast.wav

48>Extracting  en\us\callie\base256\8000\document.wav

48>Extracting  en\us\callie\base256\8000\regain.wav

48>Extracting  en\us\callie\base256\8000\kickoff.wav

48>Extracting  en\us\callie\base256\8000\woodlark.wav

48>Extracting  en\us\callie\base256\8000\checkup.wav

48>Extracting  en\us\callie\base256\8000\acme.wav

48>Extracting  en\us\callie\base256\8000\graduate.wav

48>Extracting  en\us\callie\base256\8000\mural.wav

48>Extracting  en\us\callie\base256\8000\concert.wav

48>Extracting  en\us\callie\base256\8000\caretaker.wav

48>Extracting  en\us\callie\base256\8000\spaniel.wav

48>Extracting  en\us\callie\base256\8000\Atlantic.wav

48>Extracting  en\us\callie\base256\8000\Montana.wav

48>Extracting  en\us\callie\base256\8000\seabird.wav

48>Extracting  en\us\callie\base256\8000\existence.wav

48>Extracting  en\us\callie\base256\8000\clockwork.wav

48>Extracting  en\us\callie\base256\8000\Saturday.wav

48>Extracting  en\us\callie\base256\8000\clamshell.wav

48>Extracting  en\us\callie\base256\8000\quantity.wav

48>Extracting  en\us\callie\base256\8000\blackjack.wav

48>Extracting  en\us\callie\base256\8000\Norwegian.wav

48>Extracting  en\us\callie\base256\8000\handiwork.wav

48>Extracting  en\us\callie\base256\8000\tradition.wav

48>Extracting  en\us\callie\base256\8000\tiger.wav

48>Extracting  en\us\callie\base256\8000\sardonic.wav

48>Extracting  en\us\callie\base256\8000\rebellion.wav

48>Extracting  en\us\callie\base256\8000\slingshot.wav

48>Extracting  en\us\callie\base256\8000\bookseller.wav

48>Extracting  en\us\callie\base256\8000\equation.wav

48>Extracting  en\us\callie\base256\8000\robust.wav

48>Extracting  en\us\callie\base256\8000\Wilmington.wav

48>Extracting  en\us\callie\base256\8000\voyager.wav

48>Extracting  en\us\callie\base256\8000\pharmacy.wav

48>Extracting  en\us\callie\base256\8000\ragtime.wav

48>Extracting  en\us\callie\base256\8000\spyglass.wav

48>Extracting  en\us\callie\base256\8000\edict.wav

48>Extracting  en\us\callie\base256\8000\Jupiter.wav

48>Extracting  en\us\callie\base256\8000\butterfat.wav

48>Extracting  en\us\callie\base256\8000\combustion.wav

48>Extracting  en\us\callie\base256\8000\erase.wav

48>Extracting  en\us\callie\base256\8000\Hamilton.wav

48>Extracting  en\us\callie\base256\8000\atmosphere.wav

48>Extracting  en\us\callie\base256\8000\repellent.wav

48>Extracting  en\us\callie\base256\8000\responsive.wav

48>Extracting  en\us\callie\base256\8000\skullcap.wav

48>Extracting  en\us\callie\base256\8000\involve.wav

48>Extracting  en\us\callie\base256\8000\frequency.wav

48>Extracting  en\us\callie\base256\8000\Pandora.wav

48>Extracting  en\us\callie\base256\8000\tumor.wav

48>Extracting  en\us\callie\base256\8000\Cherokee.wav

48>Extracting  en\us\callie\base256\8000\maverick.wav

48>Extracting  en\us\callie\base256\8000\classic.wav

48>Extracting  en\us\callie\base256\8000\speculate.wav

48>Extracting  en\us\callie\base256\8000\drifter.wav

48>Extracting  en\us\callie\base256\8000\everyday.wav

48>Extracting  en\us\callie\base256\8000\puberty.wav

48>Extracting  en\us\callie\base256\8000\necklace.wav

48>Extracting  en\us\callie\base256\8000\cobra.wav

48>Extracting  en\us\callie\base256\8000\endorse.wav

48>Extracting  en\us\callie\base256\8000\breadline.wav

48>Extracting  en\us\callie\base256\8000\blockade.wav

48>Extracting  en\us\callie\base256\8000\tomorrow.wav

48>Extracting  en\us\callie\base256\8000\pyramid.wav

48>Extracting  en\us\callie\base256\8000\blowtorch.wav

48>Extracting  en\us\callie\base256\8000\button.wav

48>Extracting  en\us\callie\base256\8000\impartial.wav

48>Extracting  en\us\callie\base256\8000\spindle.wav

48>Extracting  en\us\callie\base256\8000\insurgent.wav

48>Extracting  en\us\callie\base256\8000\microscope.wav

48>Extracting  en\us\callie\base256\8000\allow.wav

48>Extracting  en\us\callie\base256\8000\underfoot.wav

48>Extracting  en\us\callie\base256\8000\bottomless.wav

48>Extracting  en\us\callie\base256\8000\vertigo.wav

48>Extracting  en\us\callie\base256\8000\paramount.wav

48>Extracting  en\us\callie\base256\8000\scenic.wav

48>Extracting  en\us\callie\base256\8000\passenger.wav

48>Extracting  en\us\callie\base256\8000\billiard.wav

48>Extracting  en\us\callie\base256\8000\snowcap.wav

48>Extracting  en\us\callie\base256\8000\Burlington.wav

48>Extracting  en\us\callie\base256\8000\alkali.wav

48>Extracting  en\us\callie\base256\8000\quiver.wav

48>Extracting  en\us\callie\base256\8000\borderline.wav

48>Extracting  en\us\callie\base256\8000\integrate.wav

48>Extracting  en\us\callie\base256\8000\stagnate.wav

48>Extracting  en\us\callie\base256\8000\molasses.wav

48>Extracting  en\us\callie\base256\8000\sugar.wav

48>Extracting  en\us\callie\base256\8000\determine.wav

42>cmu_us_rms_cg_params.c

48>Extracting  en\us\callie\base256\8000\goldfish.wav

48>Extracting  en\us\callie\base256\8000\beehive.wav

48>Extracting  en\us\callie\base256\8000\snapline.wav

48>Extracting  en\us\callie\base256\8000\scallion.wav

48>Extracting  en\us\callie\base256\8000\consulting.wav

48>Extracting  en\us\callie\base256\8000\facial.wav

48>Extracting  en\us\callie\base256\8000\Ohio.wav

48>Extracting  en\us\callie\base256\8000\cellulose.wav

48>Extracting  en\us\callie\base256\8000\gazelle.wav

48>Extracting  en\us\callie\base256\8000\flagpole.wav

48>Extracting  en\us\callie\base256\8000\stethoscope.wav

48>Extracting  en\us\callie\base256\8000\scorecard.wav

48>Extracting  en\us\callie\base256\8000\eyetooth.wav

48>Extracting  en\us\callie\base256\8000\resistor.wav

48>Extracting  en\us\callie\base256\8000\absurd.wav

48>Extracting  en\us\callie\base256\8000\bifocals.wav

48>Extracting  en\us\callie\base256\8000\fortitude.wav

48>Extracting  en\us\callie\base256\8000\choking.wav

48>Extracting  en\us\callie\base256\8000\sentence.wav

48>Extracting  en\us\callie\base256\8000\paperweight.wav

48>Extracting  en\us\callie\base256\8000\article.wav

48>Extracting  en\us\callie\base256\8000\Pacific.wav

48>Extracting  en\us\callie\base256\8000\playhouse.wav

48>Extracting  en\us\callie\base256\8000\tonic.wav

48>Extracting  en\us\callie\base256\8000\confidence.wav

48>Extracting  en\us\callie\base256\8000\bombast.wav

48>Extracting  en\us\callie\base256\8000\guidance.wav

48>Extracting  en\us\callie\base256\8000\telephone.wav

48>Extracting  en\us\callie\base256\8000\revenue.wav

48>Extracting  en\us\callie\base256\8000\pandemic.wav

48>Extracting  en\us\callie\base256\8000\October.wav

48>Extracting  en\us\callie\base256\8000\tobacco.wav

48>Extracting  en\us\callie\base256\8000\uproot.wav

48>Extracting  en\us\callie\base256\8000\ribcage.wav

48>Extracting  en\us\callie\base256\8000\spigot.wav

48>Extracting  en\us\callie\base256\8000\cranky.wav

48>Extracting  en\us\callie\base256\8000\inventive.wav

48>Extracting  en\us\callie\base256\8000\Babylon.wav

48>Extracting  en\us\callie\base256\8000\caravan.wav

48>Extracting  en\us\callie\base256\8000\Camelot.wav

48>Extracting  en\us\callie\base256\8000\snowslide.wav

48>Extracting  en\us\callie\base256\8000\adroitness.wav

48>Extracting  en\us\callie\base256\8000\soybean.wav

48>Extracting  en\us\callie\base256\8000\amusement.wav

48>Extracting  en\us\callie\base256\8000\Dupont.wav

48>Extracting  en\us\callie\base256\8000\gravity.wav

48>Extracting  en\us\callie\base256\8000\solo.wav

48>Extracting  en\us\callie\base256\8000\crucial.wav

48>Extracting  en\us\callie\base256\8000\finicky.wav

48>Extracting  en\us\callie\base256\8000\dogsled.wav

48>Extracting  en\us\callie\base256\8000\outfielder.wav

48>Extracting  en\us\callie\base256\8000\adrift.wav

48>Extracting  en\us\callie\base256\8000\narrative.wav

48>Extracting  en\us\callie\base256\8000\Geiger.wav

48>Extracting  en\us\callie\base256\8000\December.wav

48>Extracting  en\us\callie\base256\8000\glitter.wav

48>Extracting  en\us\callie\base256\8000\visitor.wav

48>Extracting  en\us\callie\base256\8000\unify.wav

48>Extracting  en\us\callie\base256\8000\dashboard.wav

48>Extracting  en\us\callie\base256\8000\upcoming.wav

48>Extracting  en\us\callie\base256\8000\bluebird.wav

48>Extracting  en\us\callie\base256\8000\peachy.wav

48>Extracting  en\us\callie\base256\8000\misnomer.wav

48>Extracting  en\us\callie\base256\8000\hydraulic.wav

48>Extracting  en\us\callie\base256\8000\decimal.wav

48>Extracting  en\us\callie\base256\8000\stapler.wav

48>Extracting  en\us\callie\base256\8000\uncut.wav

48>Extracting  en\us\callie\base256\8000\replica.wav

48>Extracting  en\us\callie\base256\8000\crackdown.wav

48>Extracting  en\us\callie\base256\8000\pioneer.wav

48>Extracting  en\us\callie\base256\8000\escape.wav

48>Extracting  en\us\callie\base256\8000\stockman.wav

48>Extracting  en\us\callie\base256\8000\disruptive.wav

48>Extracting  en\us\callie\base256\8000\reindeer.wav

48>Extracting  en\us\callie\base256\8000\ratchet.wav

48>Extracting  en\us\callie\base256\8000\miracle.wav

48>Extracting  en\us\callie\base256\8000\framework.wav

48>Extracting  en\us\callie\base256\8000\rematch.wav

48>Extracting  en\us\callie\base256\8000\pheasant.wav

48>Extracting  en\us\callie\base256\8000\unicorn.wav

48>Extracting  en\us\callie\base256\8000\chisel.wav

48>Extracting  en\us\callie\base256\8000\Pegasus.wav

48>Extracting  en\us\callie\base256\8000\corporate.wav

48>Extracting  en\us\callie\base256\8000\shamrock.wav

48>Extracting  en\us\callie\base256\8000\leprosy.wav

48>Extracting  en\us\callie\base256\8000\drunken.wav

48>Extracting  en\us\callie\base256\8000\reward.wav

48>Extracting  en\us\callie\base256\8000\enrollment.wav

48>Extracting  en\us\callie\base256\8000\drainage.wav

48>Extracting  en\us\callie\base256\8000\processor.wav

48>Extracting  en\us\callie\base256\8000\fascinate.wav

48>Extracting  en\us\callie\base256\8000\obtuse.wav

48>Extracting  en\us\callie\base256\8000\buzzard.wav

48>Extracting  en\us\callie\base256\8000\hideaway.wav

48>Extracting  en\us\callie\base256\8000\vacancy.wav

48>Extracting  en\us\callie\base256\8000\photograph.wav

48>Extracting  en\us\callie\base256\8000\sterling.wav

48>Extracting  en\us\callie\base256\8000\Christmas.wav

48>Extracting  en\us\callie\base256\8000\inertia.wav

48>Extracting  en\us\callie\base256\8000\dragnet.wav

48>Extracting  en\us\callie\base256\8000\proximate.wav

48>Extracting  en\us\callie\base256\8000\goggles.wav

48>Extracting  en\us\callie\base256\8000\distortion.wav

48>Extracting  en\us\callie\base256\8000\trauma.wav

48>Extracting  en\us\callie\base256\8000\asteroid.wav

48>Extracting  en\us\callie\base256\8000\rocker.wav

48>Extracting  en\us\callie\base256\8000\Istanbul.wav

48>Extracting  en\us\callie\base256\8000\tapeworm.wav

48>Extracting  en\us\callie\base256\8000\spearhead.wav

48>Extracting  en\us\callie\base256\8000\backward.wav

48>Extracting  en\us\callie\base256\8000\reform.wav

48>Extracting  en\us\callie\base256\8000\inverse.wav

48>Extracting  en\us\callie\base256\8000\retrospect.wav

48>Extracting  en\us\callie\base256\8000\disable.wav

48>Extracting  en\us\callie\base256\8000\endow.wav

48>Extracting  en\us\callie\base256\8000\wallet.wav

48>Extracting  en\us\callie\base256\8000\Algol.wav

48>Extracting  en\us\callie\base256\8000\crucifix.wav

48>Extracting  en\us\callie\base256\8000\letterhead.wav

48>Extracting  en\us\callie\base256\8000\ruffled.wav

48>Extracting  en\us\callie\base256\8000\Burbank.wav

48>Extracting  en\us\callie\base256\8000\freedom.wav

48>Extracting  en\us\callie\base256\8000\island.wav

48>Extracting  en\us\callie\base256\8000\gremlin.wav

48>Extracting  en\us\callie\base256\8000\disbelief.wav

48>Extracting  en\us\callie\base256\8000\inferno.wav

48>Extracting  en\us\callie\base256\8000\suspicious.wav

48>Extracting  en\us\callie\voicemail

48>Extracting  en\us\callie\voicemail\8000

48>Extracting  en\us\callie\voicemail\8000\vm-record_message.wav

48>Extracting  en\us\callie\voicemail\8000\vm-new.wav

48>Extracting  en\us\callie\voicemail\8000\vm-delete_recording.wav

48>Extracting  en\us\callie\voicemail\8000\vm-abort.wav

48>Extracting  en\us\callie\voicemail\8000\vm-record_name1.wav

48>Extracting  en\us\callie\voicemail\8000\vm-you_have.wav

48>Extracting  en\us\callie\voicemail\8000\vm-marked-urgent.wav

48>Extracting  en\us\callie\voicemail\8000\vm-urgent-new.wav

48>Extracting  en\us\callie\voicemail\8000\vm-to_forward.wav

48>Extracting  en\us\callie\voicemail\8000\vm-urgent.wav

48>Extracting  en\us\callie\voicemail\8000\vm-greeting.wav

48>Extracting  en\us\callie\voicemail\8000\vm-record_greeting.wav

48>Extracting  en\us\callie\voicemail\8000\vm-send_message_now.wav

48>Extracting  en\us\callie\voicemail\8000\vm-enter_pass.wav

48>Extracting  en\us\callie\voicemail\8000\vm-followed_by.wav

48>Extracting  en\us\callie\voicemail\8000\vm-play_previous_message.wav

48>Extracting  en\us\callie\voicemail\8000\vm-not_available.wav

48>Extracting  en\us\callie\voicemail\8000\vm-rerecord.wav

48>Extracting  en\us\callie\voicemail\8000\vm-mailbox_full.wav

48>Extracting  en\us\callie\voicemail\8000\vm-from.wav

48>Extracting  en\us\callie\voicemail\8000\vm-message_envelope.wav

48>Extracting  en\us\callie\voicemail\8000\vm-record_name2.wav

48>Extracting  en\us\callie\voicemail\8000\vm-enter_id.wav

48>Extracting  en\us\callie\voicemail\8000\vm-deleted.wav

48>Extracting  en\us\callie\voicemail\8000\vm-listen_to_recording_again.wav

48>Extracting  en\us\callie\voicemail\8000\vm-fail_auth.wav

48>Extracting  en\us\callie\voicemail\8000\vm-marked_new.wav

48>Extracting  en\us\callie\voicemail\8000\vm-listen_saved.wav

48>Extracting  en\us\callie\voicemail\8000\vm-return_call.wav

42>cmu_us_rms_cg_phonestate.c

42>au_command.c

48>Extracting  en\us\callie\voicemail\8000\vm-person.wav

48>Extracting  en\us\callie\voicemail\8000\vm-to_exit_alt.wav

48>Extracting  en\us\callie\voicemail\8000\vm-listen_to_recording.wav

48>Extracting  en\us\callie\voicemail\8000\vm-in_folder.wav

48>Extracting  en\us\callie\voicemail\8000\vm-hello.wav

48>Extracting  en\us\callie\voicemail\8000\vm-next.wav

48>Extracting  en\us\callie\voicemail\8000\vm-emailed.wav

48>Extracting  en\us\callie\voicemail\8000\vm-forward_to_email.wav

48>Extracting  en\us\callie\voicemail\8000\vm-choose_greeting_fail.wav

48>Extracting  en\us\callie\voicemail\8000\vm-continue.wav

48>Extracting  en\us\callie\voicemail\8000\vm-main_menu_alt.wav

48>Extracting  en\us\callie\voicemail\8000\vm-save_recording.wav

48>Extracting  en\us\callie\voicemail\8000\vm-undeleted.wav

48>Extracting  en\us\callie\voicemail\8000\vm-urgent-saved.wav

48>Extracting  en\us\callie\voicemail\8000\vm-advanced.wav

48>Extracting  en\us\callie\voicemail\8000\vm-advanced_alt.wav

48>Extracting  en\us\callie\voicemail\8000\vm-goodbye.wav

48>Extracting  en\us\callie\voicemail\8000\vm-messages_alt.wav

48>Extracting  en\us\callie\voicemail\8000\vm-forward_enter_ext.wav

48>Extracting  en\us\callie\voicemail\8000\vm-listen_new.wav

48>Extracting  en\us\callie\voicemail\8000\vm-press.wav

48>Extracting  en\us\callie\voicemail\8000\vm-received.wav

48>Extracting  en\us\callie\voicemail\8000\vm-message_number.wav

48>Extracting  en\us\callie\voicemail\8000\vm-last.wav

48>Extracting  en\us\callie\voicemail\8000\vm-choose_greeting.wav

48>Extracting  en\us\callie\voicemail\8000\vm-has_been_changed_to.wav

48>Extracting  en\us\callie\voicemail\8000\vm-to_exit.wav

48>Extracting  en\us\callie\voicemail\8000\vm-too-small.wav

48>Extracting  en\us\callie\voicemail\8000\vm-delete_message.wav

48>Extracting  en\us\callie\voicemail\8000\vm-save_message.wav

48>Extracting  en\us\callie\voicemail\8000\vm-mark-urgent.wav

48>Extracting  en\us\callie\voicemail\8000\vm-repeat_message.wav

48>Extracting  en\us\callie\voicemail\8000\vm-mark_message_new.wav

48>Extracting  en\us\callie\voicemail\8000\vm-that_was_an_invalid_ext.wav

48>Extracting  en\us\callie\voicemail\8000\vm-message.wav

48>Extracting  en\us\callie\voicemail\8000\vm-followed_by_pound.wav

48>Extracting  en\us\callie\voicemail\8000\vm-saved.wav

48>Extracting  en\us\callie\voicemail\8000\vm-no_more_messages.wav

48>Extracting  en\us\callie\voicemail\8000\vm-choose_greeting_choose.wav

48>Extracting  en\us\callie\voicemail\8000\vm-change_password.wav

48>Extracting  en\us\callie\voicemail\8000\vm-messages.wav

48>Extracting  en\us\callie\voicemail\8000\vm-message_alt.wav

48>Extracting  en\us\callie\voicemail\8000\vm-to_record_greeting.wav

48>Extracting  en\us\callie\voicemail\8000\vm-play_greeting.wav

48>Extracting  en\us\callie\voicemail\8000\vm-play_next_message.wav

48>Extracting  en\us\callie\voicemail\8000\vm-main_menu.wav

48>Extracting  en\us\callie\voicemail\8000\vm-undelete_message.wav

48>Extracting  en\us\callie\voicemail\8000\vm-selected.wav

48>Extracting  en\us\callie\voicemail\8000\vm-forward_add_intro.wav

48>Extracting  en\us\callie\ascii

48>Extracting  en\us\callie\ascii\8000

48>Extracting  en\us\callie\ascii\8000\107.wav

48>Extracting  en\us\callie\ascii\8000\98.wav

48>Extracting  en\us\callie\ascii\8000\35.wav

48>Extracting  en\us\callie\ascii\8000\97.wav

48>Extracting  en\us\callie\ascii\8000\118.wav

48>Extracting  en\us\callie\ascii\8000\102.wav

48>Extracting  en\us\callie\ascii\8000\122.wav

48>Extracting  en\us\callie\ascii\8000\99.wav

48>Extracting  en\us\callie\ascii\8000\103.wav

48>Extracting  en\us\callie\ascii\8000\115.wav

48>Extracting  en\us\callie\ascii\8000\113.wav

48>Extracting  en\us\callie\ascii\8000\105.wav

48>Extracting  en\us\callie\ascii\8000\32.wav

48>Extracting  en\us\callie\ascii\8000\104.wav

48>Extracting  en\us\callie\ascii\8000\117.wav

48>Extracting  en\us\callie\ascii\8000\106.wav

48>Extracting  en\us\callie\ascii\8000\112.wav

48>Extracting  en\us\callie\ascii\8000\108.wav

48>Extracting  en\us\callie\ascii\8000\109.wav

48>Extracting  en\us\callie\ascii\8000\120.wav

48>Extracting  en\us\callie\ascii\8000\42.wav

48>Extracting  en\us\callie\ascii\8000\101.wav

48>Extracting  en\us\callie\ascii\8000\110.wav

48>Extracting  en\us\callie\ascii\8000\46.wav

48>Extracting  en\us\callie\ascii\8000\114.wav

42>Generating Code...

48>Extracting  en\us\callie\ascii\8000\111.wav

48>Extracting  en\us\callie\ascii\8000\100.wav

48>Extracting  en\us\callie\ascii\8000\119.wav

48>Extracting  en\us\callie\ascii\8000\121.wav

48>Extracting  en\us\callie\ascii\8000\116.wav

48>Extracting  en\us\callie\misc

48>Extracting  en\us\callie\misc\8000

48>Extracting  en\us\callie\misc\8000\phone_not_auth.wav

48>Extracting  en\us\callie\misc\8000\en.wav

48>Extracting  en\us\callie\misc\8000\sorry.wav

48>Extracting  en\us\callie\misc\8000\transfer2.wav

48>Extracting  en\us\callie\misc\8000\followed.wav

48>Extracting  en\us\callie\misc\8000\invalid_extension.wav

48>Extracting  en\us\callie\misc\8000\if_you_would_like_to.wav

48>Extracting  en\us\callie\misc\8000\error.wav

48>Extracting  en\us\callie\misc\8000\misc-your_call_has_been_terminated.wav

48>Extracting  en\us\callie\misc\8000\call_secured.wav

48>Extracting  en\us\callie\misc\8000\provide_reference_number.wav

48>Extracting  en\us\callie\misc\8000\es.wav

48>Extracting  en\us\callie\misc\8000\transfer1.wav

48>Extracting
en\us\callie\misc\8000\misc-your_call_will_be_terminated_in.wav

48>Extracting  en\us\callie\misc\8000\if_you_are_this_person.wav

48>Extracting  en\us\callie\misc\8000\we_are_trying_to_reach.wav

48>Extracting  en\us\callie\misc\8000\call_monitoring_blurb.wav

48>Extracting  en\us\callie\zrtp

48>Extracting  en\us\callie\zrtp\8000

48>Extracting  en\us\callie\zrtp\8000\zrtp-is_unverified.wav

48>Extracting  en\us\callie\zrtp\8000\zrtp-enroll_welcome.wav

48>Extracting  en\us\callie\zrtp\8000\zrtp-status_notsecure.wav

48>Extracting  en\us\callie\zrtp\8000\zrtp-enroll_already_enrolled.wav

48>Extracting  en\us\callie\zrtp\8000\zrtp-status_error.wav

48>Extracting  en\us\callie\zrtp\8000\zrtp-is_verified.wav

48>Extracting  en\us\callie\zrtp\8000\zrtp-enroll_notzrtp.wav

48>Extracting  en\us\callie\zrtp\8000\zrtp-check_sas.wav

48>Extracting  en\us\callie\zrtp\8000\zrtp-enroll_confirmed.wav

48>Extracting  en\us\callie\zrtp\8000\zrtp-is_secure.wav

48>Extracting  en\us\callie\zrtp\8000\zrtp-enroll_not_sip_registered.wav

48>Extracting  en\us\callie\zrtp\8000\zrtp-thankyou_goodbye.wav

48>Extracting  en\us\callie\zrtp\8000\zrtp-status_securing.wav

48>Extracting  en\us\callie\zrtp\8000\zrtp-status_secure.wav

48>Extracting  en\us\callie\zrtp\8000\zrtp-somethings_wrong.wav

48>Extracting  en\us\callie\conference

48>Extracting  en\us\callie\conference\8000

48>Extracting  en\us\callie\conference\8000\conf-kicked.wav

48>Extracting  en\us\callie\conference\8000\conf-locked.wav

48>Extracting  en\us\callie\conference\8000\conf-is-unlocked.wav

48>Extracting  en\us\callie\conference\8000\conf-welcome.wav

48>Extracting  en\us\callie\conference\8000\conf-unmuted.wav

48>Extracting  en\us\callie\conference\8000\conf-is-locked.wav

48>Extracting  en\us\callie\conference\8000\conf-muted.wav

48>Extracting  en\us\callie\conference\8000\conf-alone.wav

48>Extracting  en\us\callie\conference\8000\conf-bad-pin.wav

48>Extracting  en\us\callie\conference\8000\conf-pin.wav

48>Extracting  en\us\callie\conference\8000\conf-goodbye.wav

48>Extracting  en\us\callie\phonetic-ascii

48>Extracting  en\us\callie\phonetic-ascii\8000

48>Extracting  en\us\callie\phonetic-ascii\8000\107.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\98.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\35.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\97.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\118.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\102.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\122.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\99.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\103.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\115.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\113.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\105.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\32.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\104.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\117.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\106.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\112.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\108.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\109.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\120.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\42.wav

42>Compiling...

48>Extracting  en\us\callie\phonetic-ascii\8000\101.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\110.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\46.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\114.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\111.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\100.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\119.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\121.wav

48>Extracting  en\us\callie\phonetic-ascii\8000\116.wav

48>Extracting  en\us\callie\ivr

48>Extracting  en\us\callie\ivr\8000

48>Extracting  en\us\callie\ivr\8000\ivr-enter_ext_pound.wav

48>Extracting  en\us\callie\ivr\8000\ivr-account_number.wav

48>Extracting  en\us\callie\ivr\8000\ivr-to_repeat_these_options.wav

48>Extracting  en\us\callie\ivr\8000\ivr-press_one_q_or_z.wav

48>Extracting  en\us\callie\ivr\8000\ivr-to_do_a_fwd_echo_test.wav

48>Extracting  en\us\callie\ivr\8000\ivr-take_a_message.wav

48>Extracting  en\us\callie\ivr\8000\ivr-please_reenter_your_pin.wav

48>Extracting  en\us\callie\ivr\8000\ivr-this_is_a_call_from.wav

48>Extracting  en\us\callie\ivr\8000\ivr-to_do_a_freeswitch_echo_test.wav

48>Extracting  en\us\callie\ivr\8000\ivr-enter_ext.wav

48>Extracting  en\us\callie\ivr\8000\ivr-last_name_first.wav

48>Extracting  en\us\callie\ivr\8000\ivr-to_log_in.wav

48>Extracting  en\us\callie\ivr\8000\ivr-that_was_an_invalid_entry.wav

48>Extracting  en\us\callie\ivr\8000\ivr-please_try_again.wav

48>Extracting  en\us\callie\ivr\8000\ivr-connect_to_caller.wav

48>Extracting  en\us\callie\ivr\8000\ivr-first_name_first.wav

48>Extracting  en\us\callie\ivr\8000\ivr-send_to_voicemail.wav

48>Extracting
en\us\callie\ivr\8000\ivr-you_have_dialed_an_invalid_extension.wav

48>Extracting  en\us\callie\ivr\8000\ivr-to_log_out.wav

48>Extracting  en\us\callie\ivr\8000\ivr-pin_or_extension_is-invalid.wav

48>Extracting  en\us\callie\ivr\8000\ivr-hold_connect_call.wav

48>Extracting
en\us\callie\ivr\8000\ivr-please_enter_pin_followed_by_pound.wav

48>Extracting  en\us\callie\ivr\8000\ivr-sales.wav

48>Extracting  en\us\callie\ivr\8000\ivr-hello.wav

48>Extracting
en\us\callie\ivr\8000\ivr-speak_to_a_customer_service_representative.wav

48>Extracting  en\us\callie\ivr\8000\ivr-this_phone_will_now_reboot.wav

48>Extracting  en\us\callie\ivr\8000\ivr-you_are_now_logged_in.wav

48>Extracting  en\us\callie\ivr\8000\ivr-spell_name.wav

48>Extracting  en\us\callie\ivr\8000\ivr-you_may.wav

48>Extracting  en\us\callie\ivr\8000\ivr-save_review_record.wav

48>Extracting  en\us\callie\ivr\8000\ivr-im_sorry.wav

48>Extracting  en\us\callie\ivr\8000\ivr-you_may_exit_by_hanging_up.wav

48>Extracting  en\us\callie\ivr\8000\ivr-welcome_to_freeswitch.wav

48>Extracting  en\us\callie\ivr\8000\ivr-to_listen_to_moh.wav

48>Extracting  en\us\callie\ivr\8000\ivr-please_return_our_call_at.wav

48>Extracting  en\us\callie\ivr\8000\ivr-use_telephone_keypad.wav

48>Extracting  en\us\callie\ivr\8000\ivr-to_hear_screaming_monkeys.wav

48>Extracting  en\us\callie\ivr\8000\ivr-not.wav

48>Extracting  en\us\callie\ivr\8000\ivr-call.wav

48>Extracting  en\us\callie\ivr\8000\ivr-sample_submenu.wav

48>Extracting
en\us\callie\ivr\8000\ivr-to_call_the_freeswitch_conference.wav

48>Extracting  en\us\callie\ivr\8000\ivr-operator.wav

48>Extracting
en\us\callie\ivr\8000\ivr-extension_to_provision_this_phone.wav

48>Extracting  en\us\callie\ivr\8000\ivr-to_speak_with_an_operator.wav

48>Extracting  en\us\callie\ivr\8000\ivr-for_this_person.wav

48>Extracting
en\us\callie\ivr\8000\ivr-this_ivr_will_let_you_test_features.wav

48>Extracting  en\us\callie\ivr\8000\ivr-please.wav

48>Extracting  en\us\callie\ivr\8000\ivr-register_for_cluecon.wav

48>Extracting  en\us\callie\ivr\8000\ivr-unable_save.wav

48>Extracting  en\us\callie\ivr\8000\ivr-regarding_reference_number.wav

48>Extracting
en\us\callie\ivr\8000\ivr-provision_phone_permanently_to_extension.wav

48>Extracting  en\us\callie\ivr\8000\ivr-to_return_to_previous_menu.wav

48>Extracting  en\us\callie\ivr\8000\ivr-technical_support.wav

48>Extracting  en\us\callie\ivr\8000\ivr-you_are_now_logged_out.wav

48>Extracting  en\us\callie\ivr\8000\ivr-customer_service.wav

48>Extracting  en\us\callie\ivr\8000\ivr-say_name.wav

48>Extracting
en\us\callie\ivr\8000\ivr-please_enter_extension_followed_by_pound.wav

48>Extracting
en\us\callie\ivr\8000\ivr-you_are_about_to_provision_this_phone.wav

48>Extracting  en\us\callie\ivr\8000\ivr-recording_saved.wav

48>Extracting  en\us\callie\ivr\8000\ivr-to_hear_sample_submenu.wav

48>Extracting  en\us\callie\ivr\8000\ivr-or.wav

48>Extracting  en\us\callie\ivr\8000\ivr-thank_you.wav

48>Everything is Ok

48>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download
8khzsound.htm"

48>Download 8khzsound - 0 error(s), 0 warning(s)

49>------ Build started: Project: abyss, Configuration: Debug Win32 ------

49>Compiling...

42>cmu_time_awb.c

49>chanswitch.c

42>c:\freeswitch\freeswitch-1.0.6\libs\flite-1.3.99\lang\cmu_time_awb\cmu_ti
me_awb.c(81) : warning C4090: '=' : different 'const' qualifiers

42>c:\freeswitch\freeswitch-1.0.6\libs\flite-1.3.99\lang\cmu_time_awb\cmu_ti
me_awb.c(82) : warning C4090: '=' : different 'const' qualifiers

49>conf.c

42>Creating library...

49>conn.c

42>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\flite\Debug\BuildLog.htm"

49>data.c

42>flite - 0 error(s), 3 warning(s)

49>date.c

50>------ Build started: Project: pocketsphinx, Configuration: Debug Win32
------

49>file.c

50>Compiling...

50>acmod.c

49>handler.c

50>bin_mdef.c

49>http.c

50>blkarray_list.c

49>init.c

50>cmu6_lts_rules.c

50>dict2pid.c

49>response.c

50>fillpen.c

49>server.c

50>fsg_history.c

49>session.c

49>socket.c

50>fsg_lextree.c

49>socket_win.c

50>fsg_search.c

49>thread_windows.c

50>hmm.c

50>kdtree.c

49>token.c

50>lextree.c

49>trace.c

49>channel.c

50>mdef.c

49>Generating Code...

50>ms_gauden.c

49>Creating library...

49>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\xmlrpc-c\Windows\Debug\abyss\Bui
ldLog.htm"

49>abyss - 0 error(s), 0 warning(s)

50>ms_mgau.c

51>------ Build started: Project: xmlrpc, Configuration: Debug Win32 ------

51>Compiling...

51>double.c

51>error.c

50>ms_senone.c

51>make_printable.c

51>memblock.c

51>method.c

50>ngram_search.c

51>parse_value.c

51>pthreadx_win32.c

51>registry.c

50>ngram_search_fwdflat.c

51>resource.c

51>select.c

50>ngram_search_fwdtree.c

51>sleep.c

51>system_method.c

51>time.c

50>phone_loop_search.c

51>trace.c

50>Generating Code...

51>utf8.c

51>version.c

51>xmlrpc_array.c

51>xmlrpc_authcookie.c

50>Compiling...

50>pocketsphinx.c

51>xmlrpc_base64.c

51>xmlrpc_build.c

51>Generating Code...

51>Compiling...

51>xmlrpc_client.c

51>xmlrpc_client_global.c

51>xmlrpc_data.c

50>ps_lattice.c

51>xmlrpc_datetime.c

51>xmlrpc_decompose.c

50>ps_mllr.c

51>xmlrpc_expat.c

51>xmlrpc_parse.c

51>xmlrpc_serialize.c

51>xmlrpc_server_abyss.c

50>s2_semi_mgau.c

50>s3dict.c

51>xmlrpc_server_info.c

51>xmlrpc_string.c

50>tmat.c

51>xmlrpc_struct.c

51>xmlrpc_wininet_transport.c

50>tst_search.c

51>asprintf.c

51>Generating Code...

51>Creating library...

50>vector.c

51>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\xmlrpc-c\Windows\Debug\xmlrpc\Bu
ildLog.htm"

51>xmlrpc - 0 error(s), 0 warning(s)

50>vithist.c

50>Generating Code...

52>------ Build started: Project: xmlparse, Configuration: Debug Win32
------

52>Compiling...

52>xmlparse.c

50>Compiling manifest to resources...

50>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

50>Copyright (C) Microsoft Corporation.  All rights reserved.

50>Linking...

50>   Creating library C:\FreeSWITCH\freeswitch-1.0.6\Debug\pocketsphinx.lib
and object C:\FreeSWITCH\freeswitch-1.0.6\Debug\pocketsphinx.exp

52>Creating library...

52>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\xmlrpc-c\Windows\Debug\xmlparse\
BuildLog.htm"

52>xmlparse - 0 error(s), 0 warning(s)

50>Embedding manifest...

53>------ Build started: Project: libspandsp, Configuration: Debug Win32
------

53>Copying c:\FreeSWITCH\freeswitch-1.0.6\libs\spandsp\src\msvc\spandsp.h to
c:\FreeSWITCH\freeswitch-1.0.6\libs\spandsp\src\spandsp.h

53>        1 file(s) copied.

53>Compiling...

50>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

50>Copyright (C) Microsoft Corporation.  All rights reserved.

53>adsi.c

50>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\pocketsphinx\Debug\BuildLo
g.htm"

50>pocketsphinx - 0 error(s), 0 warning(s)

53>async.c

53>at_interpreter.c

53>awgn.c

53>bell_r2_mf.c

53>bert.c

53>bit_operations.c

54>------ Build started: Project: xmltok, Configuration: Debug Win32 ------

54>Compiling...

54>xmltok.c

53>bitstream.c

54>xmlrole.c

53>complex_filters.c

53>complex_vector_float.c

54>Generating Code...

53>complex_vector_int.c

54>Creating library...

53>crc.c

54>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\xmlrpc-c\Windows\Debug\xmltok\Bu
ildLog.htm"

54>xmltok - 0 error(s), 0 warning(s)

53>dds_float.c

53>dds_int.c

53>dtmf.c

53>echo.c

53>fax.c

53>fax_modems.c

55>------ Build started: Project: mod_lcr, Configuration: Debug Win32 ------

55>Compiling...

55>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

55>mod_lcr.c

53>fsk.c

53>g711.c

53>Generating Code...

53>Compiling...

53>g722.c

55>Linking...

55>   Creating library Win32\Debug/mod_lcr.2008.lib and object
Win32\Debug/mod_lcr.2008.exp

55>Embedding manifest...

53>g726.c

53>gsm0610_decode.c

53>gsm0610_encode.c

53>gsm0610_long_term.c

55>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_lcr\Win32\De
bug\BuildLog.htm"

55>mod_lcr - 0 error(s), 1 warning(s)

53>gsm0610_lpc.c

53>gsm0610_preprocess.c

56>------ Build started: Project: mod_easyroute, Configuration: Debug Win32
------

56>Compiling...

56>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

53>gsm0610_rpe.c

53>gsm0610_short_term.c

53>hdlc.c

56>mod_easyroute.c

53>ima_adpcm.c

53>logging.c

53>lpc10_analyse.c

56>Linking...

56>   Creating library Win32\Debug/mod_easyroute.2008.lib and object
Win32\Debug/mod_easyroute.2008.exp

53>lpc10_decode.c

56>Embedding manifest...

53>lpc10_encode.c

53>lpc10_placev.c

56>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_easyroute\Wi
n32\Debug\BuildLog.htm"

56>mod_easyroute - 0 error(s), 1 warning(s)

53>lpc10_voicing.c

57>------ Build started: Project: mod_pocketsphinx, Configuration: Debug
Win32 ------

57>Compiling...

57>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

57>mod_pocketsphinx.c

53>modem_echo.c

53>modem_connect_tones.c

53>noise.c

53>Generating Code...

57>Compiling manifest to resources...

57>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

57>Copyright (C) Microsoft Corporation.  All rights reserved.

57>Linking...

57>   Creating library Win32\Debug/mod_pocketsphinx.2008.lib and object
Win32\Debug/mod_pocketsphinx.2008.exp

57>Embedding manifest...

53>Compiling...

53>oki_adpcm.c

57>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

57>Copyright (C) Microsoft Corporation.  All rights reserved.

53>playout.c

57>Performing Post-Build Event...

53>plc.c

53>power_meter.c

53>queue.c

53>schedule.c

53>sig_tone.c

57>C:\FreeSWITCH\freeswitch-1.0.6\libs\Communicator_semi_40.cd_semi_6000\COP
YING

57>C:\FreeSWITCH\freeswitch-1.0.6\libs\Communicator_semi_40.cd_semi_6000\fea
t.params

57>C:\FreeSWITCH\freeswitch-1.0.6\libs\Communicator_semi_40.cd_semi_6000\mde
f

57>C:\FreeSWITCH\freeswitch-1.0.6\libs\Communicator_semi_40.cd_semi_6000\mea
ns

57>C:\FreeSWITCH\freeswitch-1.0.6\libs\Communicator_semi_40.cd_semi_6000\noi
sedict

57>C:\FreeSWITCH\freeswitch-1.0.6\libs\Communicator_semi_40.cd_semi_6000\sen
dump

57>C:\FreeSWITCH\freeswitch-1.0.6\libs\Communicator_semi_40.cd_semi_6000\tra
nsition_matrices

57>C:\FreeSWITCH\freeswitch-1.0.6\libs\Communicator_semi_40.cd_semi_6000\var
iances

57>8 File(s) copied

53>silence_gen.c

57>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\asr_tts\mod_pocketsphinx\Win3
2\Debug\BuildLog.htm"

57>mod_pocketsphinx - 0 error(s), 1 warning(s)

53>super_tone_rx.c

53>super_tone_tx.c

53>swept_tone.c

53>t4_rx.c

58>------ Build started: Project: mod_lua, Configuration: Debug Win32 ------

58>Compiling...

53>t4_tx.c

58>mod_lua.cpp

53>t30.c

53>t30_api.c

53>t30_logging.c

53>t31.c

53>t35.c

58>freeswitch_lua.cpp

53>t38_core.c

53>t38_gateway.c

53>Generating Code...

58>Generating Code...

53>Compiling...

53>t38_non_ecm_buffer.c

58>Compiling...

58>mod_lua_wrap.cpp

53>t38_terminal.c

53>testcpuid.c

53>time_scale.c

53>tone_detect.c

53>tone_generate.c

53>v17rx.c

53>v17tx.c

53>v18.c

58>Linking...

53>v22bis_rx.c

58>   Creating library Win32\Debug/mod_lua.2008.lib and object
Win32\Debug/mod_lua.2008.exp

53>v22bis_tx.c

53>v27ter_rx.c

58>Embedding manifest...

53>v27ter_tx.c

53>v29rx.c

58>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_lua\Win32\Debug
\BuildLog.htm"

58>mod_lua - 0 error(s), 0 warning(s)

59>------ Build started: Project: 8khz, Configuration: Debug Win32 ------

59>Performing Post-Build Event...

53>v29tx.c

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-abort.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-advanced.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-advanced_alt.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-change_password.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-choose_greeting.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-choose_greeting_choose.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-choose_greeting_fail.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-continue.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-deleted.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-delete_message.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-delete_recording.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-emailed.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-enter_id.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-enter_pass.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-fail_auth.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-followed_by.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-followed_by_pound.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-forward_add_intro.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-forward_enter_ext.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-forward_to_email.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-from.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-goodbye.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-greeting.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-has_been_changed_to.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-hello.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-in_folder.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-last.wav

53>v42.c

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-listen_new.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-listen_saved.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-listen_to_recording.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-listen_to_recording_again.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-mailbox_full.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-main_menu.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-main_menu_alt.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-mark-urgent.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-marked-urgent.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-marked_new.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-mark_message_new.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-message.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-messages.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-messages_alt.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-message_alt.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-message_envelope.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-message_number.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-new.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-next.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-not_available.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-no_more_messages.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-person.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-play_greeting.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-play_next_message.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-play_previous_message.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-press.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-received.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-record_greeting.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-record_message.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-record_name1.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-record_name2.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-repeat_message.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-rerecord.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-return_call.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-saved.wav

53>v42bis.c

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-save_message.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-save_recording.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-selected.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-send_message_now.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-that_was_an_invalid_ext.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-too-small.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-to_exit.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-to_exit_alt.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-to_forward.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-to_record_greeting.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-undeleted.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-undelete_message.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-urgent-new.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-urgent-saved.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-urgent.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm
-you_have.wav

59>78 File(s) copied

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-acco
unt_number.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-call
.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-conn
ect_to_caller.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-cust
omer_service.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-ente
r_ext.wav

53>v8.c

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-ente
r_ext_pound.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-exte
nsion_to_provision_this_phone.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-firs
t_name_first.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-for_
this_person.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-hell
o.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-hold
_connect_call.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-im_s
orry.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-last
_name_first.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-not.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-oper
ator.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-or.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-pin_
or_extension_is-invalid.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-plea
se.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-plea
se_enter_extension_followed_by_pound.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-plea
se_enter_pin_followed_by_pound.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-plea
se_reenter_your_pin.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-plea
se_return_our_call_at.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-plea
se_try_again.wav

53>vector_float.c

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-pres
s_one_q_or_z.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-prov
ision_phone_permanently_to_extension.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-reco
rding_saved.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-rega
rding_reference_number.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-regi
ster_for_cluecon.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-sale
s.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-samp
le_submenu.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-save
_review_record.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-say_
name.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-send
_to_voicemail.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-spea
k_to_a_customer_service_representative.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-spel
l_name.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-take
_a_message.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-tech
nical_support.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-than
k_you.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-that
_was_an_invalid_entry.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-this
_is_a_call_from.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-this
_ivr_will_let_you_test_features.wav

53>vector_int.c

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-this
_phone_will_now_reboot.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_c
all_the_freeswitch_conference.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_d
o_a_freeswitch_echo_test.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_d
o_a_fwd_echo_test.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_h
ear_sample_submenu.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_h
ear_screaming_monkeys.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_l
isten_to_moh.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_l
og_in.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_l
og_out.wav

53>Generating Code...

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_r
epeat_these_options.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_r
eturn_to_previous_menu.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_s
peak_with_an_operator.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-unab
le_save.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-use_
telephone_keypad.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-welc
ome_to_freeswitch.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-you_
are_about_to_provision_this_phone.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-you_
are_now_logged_in.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-you_
are_now_logged_out.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-you_
have_dialed_an_invalid_extension.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-you_
may.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-you_
may_exit_by_hanging_up.wav

59>62 File(s) copied

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\c
onf-alone.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\c
onf-bad-pin.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\c
onf-goodbye.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\c
onf-is-locked.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\c
onf-is-unlocked.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\c
onf-kicked.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\c
onf-locked.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\c
onf-muted.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\c
onf-pin.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\c
onf-unmuted.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\c
onf-welcome.wav

59>11 File(s) copied

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\a-m.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\at.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\day-0.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\day-1.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\day-2.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\day-3.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\day-4.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\day-5.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\day-6.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\hour.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\hours.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\minute.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\minutes
.wav

53>Compiling...

53>gettimeofday.c

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-0.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-1.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-10.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-11.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-2.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-3.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-4.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-5.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-6.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-7.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-8.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-9.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\oclock.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\oh.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\p-m.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\second.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\seconds
.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\today.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\tomorro
w.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\yesterd
ay.wav

59>33 File(s) copied

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\0.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\1.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\10.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\11.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\12.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\13.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\14.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\15.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\16.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\17.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\18.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\19.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\2.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\20.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\3.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\30.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\4.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\40.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\5.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\50.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\6.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\60.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\7.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\70.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\8.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\80.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\9.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\90.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\dot.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-1.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-10.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-11.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-12.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-13.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-14.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-15.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-16.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-17.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-18.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-19.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-2.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-20.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-3.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-30.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-4.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-5.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-6.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-7.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-8.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-9.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\hundr
ed.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\milli
on.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\perio
d.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\point
.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\pound
.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\star.
wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\thous
and.wav

53>Generating Code...

59>57 File(s) copied

53>Compiling manifest to resources...

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\100.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\101.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\102.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\103.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\104.wa
v

53>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

53>Copyright (C) Microsoft Corporation.  All rights reserved.

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\105.wa
v

53>Linking...

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\106.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\107.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\108.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\109.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\110.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\111.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\112.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\113.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\114.wa
v

53>   Creating library ./Debug\spandsp.lib and object ./Debug\spandsp.exp

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\115.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\116.wa
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\117.wa
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\118.wa
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\119.wa
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\120.wa
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\121.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\122.wa
v

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\32.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\35.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\42.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\46.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\97.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\98.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\99.wav

59>30 File(s) copied

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\call_mo
nitoring_blurb.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\call_se
cured.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\en.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\error.w
av

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\es.wav

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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\if_you_
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\if_you_
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\invalid
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\misc-yo
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\misc-yo
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\phone_n
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\provide
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\sorry.w
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\transfe
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\transfe
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\we_are_
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59>17 File(s) copied

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\and
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\cen
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\cen
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\cen
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\cen
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\dol
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\dol
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\min
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\neg
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59>9 File(s) copied

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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53>Embedding manifest...

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
00\116.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
00\118.wav

53>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

53>Copyright (C) Microsoft Corporation.  All rights reserved.

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
00\119.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
00\120.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
00\121.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
00\122.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
00\32.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
00\35.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
00\42.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
00\46.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
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59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
00\98.wav

59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\80
00\99.wav

59>30 File(s) copied

59>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Sound_Files\Debug\BuildLog
.htm"

59>8khz - 0 error(s), 0 warning(s)

60>------ Build started: Project: mod_fsv, Configuration: Debug Win32 ------

53>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\spandsp\src\Debug\BuildLog
libspandsp.htm"

53>libspandsp - 0 error(s), 0 warning(s)

60>Compiling...

60>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

60>mod_fsv.c

61>------ Build started: Project: mod_fax, Configuration: Debug Win32 ------

60>Linking...

61>Compiling...

61>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

61>mod_fax.c

60>   Creating library Win32\Debug/mod_fsv.2008.lib and object
Win32\Debug/mod_fsv.2008.exp

60>Embedding manifest...

60>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_fsv\Win32\De
bug\BuildLog.htm"

60>mod_fsv - 0 error(s), 1 warning(s)

62>------ Build started: Project: mod_voipcodecs, Configuration: Debug Win32
------

62>Compiling...

62>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

62>mod_voipcodecs.c

62>Linking...

62>   Creating library Win32\Debug/mod_voipcodecs.2008.lib and object
Win32\Debug/mod_voipcodecs.2008.exp

62>Embedding manifest...

61>Linking...

61>   Creating library Win32\Debug/mod_fax.2008.lib and object
Win32\Debug/mod_fax.2008.exp

61>Embedding manifest...

62>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\codecs\mod_voipcodecs\Win32\D
ebug\BuildLog.htm"

62>mod_voipcodecs - 0 error(s), 1 warning(s)

61>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_fax\Win32\De
bug\BuildLog.htm"

61>mod_fax - 0 error(s), 1 warning(s)

63>------ Build started: Project: mod_tone_stream, Configuration: Debug
Win32 ------

63>Compiling...

63>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

63>mod_tone_stream.c

64>------ Build started: Project: mod_cdr_csv, Configuration: Debug Win32
------

64>Compiling...

64>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

64>mod_cdr_csv.c

63>Linking...

63>   Creating library Win32\Debug/mod_tone_stream.2008.lib and object
Win32\Debug/mod_tone_stream.2008.exp

64>Linking...

64>   Creating library Win32\Debug/mod_cdr_csv.2008.lib and object
Win32\Debug/mod_cdr_csv.2008.exp

63>Embedding manifest...

64>Embedding manifest...

63>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\formats\mod_tone_stream\Win32
\Debug\BuildLog.htm"

63>mod_tone_stream - 0 error(s), 1 warning(s)

64>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\event_handlers\mod_cdr_csv\Wi
n32\Debug\BuildLog.htm"

64>mod_cdr_csv - 0 error(s), 1 warning(s)

65>------ Build started: Project: mod_logfile, Configuration: Debug Win32
------

65>Compiling...

66>------ Build started: Project: mod_dialplan_asterisk, Configuration:
Debug Win32 ------

65>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

65>mod_logfile.c

66>Compiling...

66>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

66>mod_dialplan_asterisk.c

65>Linking...

65>   Creating library Win32\Debug/mod_logfile.2008.lib and object
Win32\Debug/mod_logfile.2008.exp

66>Linking...

66>   Creating library Win32\Debug/mod_dialplan_asterisk.2008.lib and object
Win32\Debug/mod_dialplan_asterisk.2008.exp

65>Embedding manifest...

66>Embedding manifest...

65>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\loggers\mod_logfile\Win32\Deb
ug\BuildLog.htm"

65>mod_logfile - 0 error(s), 1 warning(s)

66>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\dialplans\mod_dialplan_asteri
sk\Win32\Debug\BuildLog.htm"

66>mod_dialplan_asterisk - 0 error(s), 1 warning(s)

67>------ Build started: Project: mod_expr, Configuration: Debug Win32
------

67>Compiling...

67>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

67>mod_expr.c

68>------ Build started: Project: mod_limit, Configuration: Debug Win32
------

68>Compiling...

68>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

68>mod_limit.c

67>exprval.c

67>exprutil.c

67>exprpars.c

67>exprobj.c

67>exprmem.c

67>exprinit.c

67>exprfunc.c

68>Linking...

68>   Creating library Win32\Debug/mod_limit.2008.lib and object
Win32\Debug/mod_limit.2008.exp

67>expreval.c

67>Generating Code...

68>Embedding manifest...

67>Linking...

67>   Creating library Win32\Debug/mod_expr.2008.lib and object
Win32\Debug/mod_expr.2008.exp

68>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_limit\Win32\
Debug\BuildLog.htm"

68>mod_limit - 0 error(s), 1 warning(s)

69>------ Build started: Project: mod_fifo, Configuration: Debug Win32
------

69>Compiling...

69>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

69>mod_fifo.c

67>Embedding manifest...

67>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_expr\Win32\D
ebug\BuildLog.htm"

67>mod_expr - 0 error(s), 1 warning(s)

70>------ Build started: Project: mod_say_nl, Configuration: Debug Win32
------

70>Compiling...

70>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

70>mod_say_nl.c

69>Linking...

69>   Creating library Win32\Debug/mod_fifo.2008.lib and object
Win32\Debug/mod_fifo.2008.exp

69>Embedding manifest...

70>Linking...

69>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_fifo\Win32\D
ebug\BuildLog.htm"

69>mod_fifo - 0 error(s), 1 warning(s)

70>   Creating library Win32\Debug/mod_say_nl.2008.lib and object
Win32\Debug/mod_say_nl.2008.exp

71>------ Build started: Project: mod_say_it, Configuration: Debug Win32
------

71>Compiling...

70>Embedding manifest...

71>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

71>mod_say_it.c

70>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\say\mod_say_nl\Win32\Debug\Bu
ildLog.htm"

70>mod_say_nl - 0 error(s), 1 warning(s)

72>------ Build started: Project: mod_say_fr, Configuration: Debug Win32
------

72>Compiling...

72>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

72>mod_say_fr.c

71>Linking...

71>   Creating library Win32\Debug/mod_say_it.2008.lib and object
Win32\Debug/mod_say_it.2008.exp

72>Linking...

71>Embedding manifest...

72>   Creating library Win32\Debug/mod_say_fr.2008.lib and object
Win32\Debug/mod_say_fr.2008.exp

72>Embedding manifest...

71>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\say\mod_say_it\Win32\Debug\Bu
ildLog.htm"

71>mod_say_it - 0 error(s), 1 warning(s)

72>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\say\mod_say_fr\Win32\Debug\Bu
ildLog.htm"

72>mod_say_fr - 0 error(s), 1 warning(s)

73>------ Skipped Build: Project: 16khz, Configuration: Debug Win32 ------

73>Project not selected to build for this solution configuration 

74>------ Build started: Project: mod_say_de, Configuration: Debug Win32
------

75>------ Build started: Project: mod_voicemail, Configuration: Debug Win32
------

74>Compiling...

75>Compiling...

74>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

74>mod_say_de.c

75>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

75>mod_voicemail.c

74>Linking...

74>   Creating library Win32\Debug/mod_say_de.2008.lib and object
Win32\Debug/mod_say_de.2008.exp

75>Linking...

75>   Creating library Win32\Debug/mod_voicemail.2008.lib and object
Win32\Debug/mod_voicemail.2008.exp

74>Embedding manifest...

75>Embedding manifest...

74>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\say\mod_say_de\Win32\Debug\Bu
ildLog.htm"

74>mod_say_de - 0 error(s), 1 warning(s)

75>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_voicemail\Wi
n32\Debug\BuildLog.htm"

75>mod_voicemail - 0 error(s), 1 warning(s)

76>------ Build started: Project: mod_spidermonkey_socket, Configuration:
Debug Win32 ------

76>Compiling...

76>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

76>mod_spidermonkey_socket.c

77>------ Build started: Project: mod_local_stream, Configuration: Debug
Win32 ------

77>Compiling...

77>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

77>mod_local_stream.c

77>Linking...

77>   Creating library Win32\Debug/mod_local_stream.2008.lib and object
Win32\Debug/mod_local_stream.2008.exp

76>Linking...

76>   Creating library Win32\Debug/mod_spidermonkey_socket.2008.lib and
object Win32\Debug/mod_spidermonkey_socket.2008.exp

77>Embedding manifest...

76>Embedding manifest...

77>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\formats\mod_local_stream\Win3
2\Debug\BuildLog.htm"

77>mod_local_stream - 0 error(s), 1 warning(s)

76>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_spidermonkey\Wi
n32\Debug\BuildLog.htm"

76>mod_spidermonkey_socket - 0 error(s), 1 warning(s)

78>------ Build started: Project: mod_esf, Configuration: Debug Win32 ------

79>------ Build started: Project: mod_h26x, Configuration: Debug Win32
------

78>Compiling...

79>Compiling...

78>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

78>mod_esf.c

79>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

79>mod_h26x.c

79>Linking...

78>Linking...

79>   Creating library Win32\Debug/mod_h26x.2008.lib and object
Win32\Debug/mod_h26x.2008.exp

78>   Creating library Win32\Debug/mod_esf.2008.lib and object
Win32\Debug/mod_esf.2008.exp

79>Embedding manifest...

78>Embedding manifest...

78>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_esf\Win32\De
bug\BuildLog.htm"

78>mod_esf - 0 error(s), 1 warning(s)

79>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\codecs\mod_h26x\Win32\Debug\B
uildLog.htm"

79>mod_h26x - 0 error(s), 1 warning(s)

80>------ Build started: Project: mod_amr, Configuration: Debug Passthrough
Win32 ------

80>Compiling...

80>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

80>mod_amr.c

81>------ Build started: Project: mod_xml_cdr, Configuration: Debug Win32
------

81>Compiling...

81>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

81>mod_xml_cdr.c

80>Linking...

80>   Creating library Win32\Debug Passthrough/mod_amr.2008.lib and object
Win32\Debug Passthrough/mod_amr.2008.exp

80>Embedding manifest...

80>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\codecs\mod_amr\Win32\Debug
Passthrough\BuildLog.htm"

80>mod_amr - 0 error(s), 1 warning(s)

81>Linking...

81>   Creating library Win32\Debug/mod_xml_cdr.2008.lib and object
Win32\Debug/mod_xml_cdr.2008.exp

82>------ Skipped Build: Project: 32khz, Configuration: Debug Win32 ------

82>Project not selected to build for this solution configuration 

83>------ Build started: Project: mod_say_en, Configuration: Debug Win32
------

83>Compiling...

81>Embedding manifest...

83>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

83>mod_say_en.c

81>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\xml_int\mod_xml_cdr\Win32\Deb
ug\BuildLog.htm"

81>mod_xml_cdr - 0 error(s), 1 warning(s)

84>------ Build started: Project: mod_xml_curl, Configuration: Debug Win32
------

84>Compiling...

84>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

84>mod_xml_curl.c

83>Linking...

83>   Creating library Win32\Debug/mod_say_en.2008.lib and object
Win32\Debug/mod_say_en.2008.exp

83>Embedding manifest...

84>Linking...

84>   Creating library Win32\Debug/mod_xml_curl.2008.lib and object
Win32\Debug/mod_xml_curl.2008.exp

83>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\say\mod_say_en\Win32\Debug\Bu
ildLog.htm"

83>mod_say_en - 0 error(s), 1 warning(s)

84>Embedding manifest...

84>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\xml_int\mod_xml_curl\Win32\De
bug\BuildLog.htm"

84>mod_xml_curl - 0 error(s), 1 warning(s)

85>------ Build started: Project: mod_spidermonkey_odbc, Configuration:
Debug Win32 ------

85>Compiling...

85>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

85>mod_spidermonkey_odbc.c

86>------ Build started: Project: mod_enum, Configuration: Debug Win32
------

86>Compiling...

86>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

86>mod_enum.c

86>Linking...

86>   Creating library Win32\Debug/mod_enum.2008.lib and object
Win32\Debug/mod_enum.2008.exp

85>Linking...

85>   Creating library Win32\Debug/mod_spidermonkey_odbc.2008.lib and object
Win32\Debug/mod_spidermonkey_odbc.2008.exp

86>Embedding manifest...

85>Embedding manifest...

86>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_enum\Win32\D
ebug\BuildLog.htm"

86>mod_enum - 0 error(s), 1 warning(s)

85>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_spidermonkey\Wi
n32\Debug\BuildLog.htm"

85>mod_spidermonkey_odbc - 0 error(s), 1 warning(s)

87>------ Build started: Project: mod_flite, Configuration: Debug Win32
------

87>Compiling...

87>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

87>mod_flite.c

88>------ Build started: Project: mod_spidermonkey_teletone, Configuration:
Debug Win32 ------

88>Compiling...

88>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

88>mod_spidermonkey_teletone.c

87>Compiling manifest to resources...

87>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

87>Copyright (C) Microsoft Corporation.  All rights reserved.

87>Linking...

87>   Creating library Win32\Debug/mod_flite.2008.lib and object
Win32\Debug/mod_flite.2008.exp

88>Linking...

88>   Creating library Win32\Debug/mod_spidermonkey_teletone.2008.lib and
object Win32\Debug/mod_spidermonkey_teletone.2008.exp

88>Embedding manifest...

88>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_spidermonkey\Wi
n32\Debug\BuildLog.htm"

88>mod_spidermonkey_teletone - 0 error(s), 1 warning(s)

87>Embedding manifest...

87>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1

87>Copyright (C) Microsoft Corporation.  All rights reserved.

89>------ Build started: Project: mod_spidermonkey_core_db, Configuration:
Debug Win32 ------

89>Compiling...

89>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

89>mod_spidermonkey_core_db.c

87>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\asr_tts\mod_flite\Win32\Debug
\BuildLog.htm"

87>mod_flite - 0 error(s), 1 warning(s)

90>------ Build started: Project: mod_native_file, Configuration: Debug
Win32 ------

90>Compiling...

90>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

90>mod_native_file.c

89>Linking...

90>Linking...

90>   Creating library Win32\Debug/mod_native_file.2008.lib and object
Win32\Debug/mod_native_file.2008.exp

89>   Creating library Win32\Debug/mod_spidermonkey_core_db.2008.lib and
object Win32\Debug/mod_spidermonkey_core_db.2008.exp

90>Embedding manifest...

89>Embedding manifest...

90>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\formats\mod_native_file\Win32
\Debug\BuildLog.htm"

90>mod_native_file - 0 error(s), 1 warning(s)

91>------ Build started: Project: mod_g723_1, Configuration: Debug
Passthrough Win32 ------

91>Compiling...

89>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_spidermonkey\Wi
n32\Debug\BuildLog.htm"

89>mod_spidermonkey_core_db - 0 error(s), 1 warning(s)

91>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

91>mod_g723_1.c

92>------ Build started: Project: fs_cli, Configuration: Debug Win32 ------

92>Compiling...

92>fs_cli.c

92>getopt_long.c

91>Linking...

91>   Creating library Win32\Debug Passthrough/mod_g723_1.2008.lib and
object Win32\Debug Passthrough/mod_g723_1.2008.exp

92>Generating Code...

92>Linking...

91>Embedding manifest...

92>Embedding manifest...

92>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\esl\Debug\BuildLog fs_cli.htm"

92>fs_cli - 0 error(s), 0 warning(s)

91>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\codecs\mod_g723_1\Win32\Debug
Passthrough\BuildLog.htm"

91>mod_g723_1 - 0 error(s), 1 warning(s)

93>------ Skipped Build: Project: mod_opal, Configuration: Debug Win32
------

93>Project not selected to build for this solution configuration 

94>------ Build started: Project: mod_siren, Configuration: Debug Win32
------

95>------ Build started: Project: mod_sofia, Configuration: Debug Win32
------

94>Compiling...

94>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

94>mod_siren.c

95>Compiling...

95>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

95>sofia_sla.c

95>c:\freeswitch\freeswitch-1.0.6\src\mod\endpoints\mod_sofia\mod_sofia.h(11
5) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h':
No such file or directory

95>sofia_reg.c

94>Linking...

94>   Creating library Win32\Debug/mod_siren.2008.lib and object
Win32\Debug/mod_siren.2008.exp

95>c:\freeswitch\freeswitch-1.0.6\src\mod\endpoints\mod_sofia\mod_sofia.h(11
5) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h':
No such file or directory

94>Embedding manifest...

95>sofia_presence.c

94>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\codecs\mod_siren\Win32\Debug\
BuildLog.htm"

94>mod_siren - 0 error(s), 1 warning(s)

96>------ Build started: Project: mod_shout, Configuration: Debug Win32
------

96>Compiling...

96>mod_shout.c

95>c:\freeswitch\freeswitch-1.0.6\src\mod\endpoints\mod_sofia\mod_sofia.h(11
5) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h':
No such file or directory

95>sofia_glue.c

95>c:\freeswitch\freeswitch-1.0.6\src\mod\endpoints\mod_sofia\mod_sofia.h(11
5) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h':
No such file or directory

95>sofia.c

96>Linking...

96>   Creating library Win32\Debug/mod_shout.lib and object
Win32\Debug/mod_shout.exp

95>c:\freeswitch\freeswitch-1.0.6\src\mod\endpoints\mod_sofia\mod_sofia.h(11
5) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h':
No such file or directory

95>sip-dig.c

95>.\sip-dig.c(813) : warning C4244: 'function' : conversion from 'int' to
'uint16_t', possible loss of data

95>.\sip-dig.c(822) : warning C4244: 'function' : conversion from 'int' to
'uint16_t', possible loss of data

96>Embedding manifest...

95>mod_sofia.c

96>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\formats\mod_shout\Win32\Debug
\BuildLog.htm"

96>mod_shout - 0 error(s), 0 warning(s)

97>------ Build started: Project: xml, Configuration: Debug Win32 ------

97>Creating config.h from winconfig.h

97>Creating expat.h from expat.h.in

95>c:\freeswitch\freeswitch-1.0.6\src\mod\endpoints\mod_sofia\mod_sofia.h(11
5) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h':
No such file or directory

95>Generating Code...

97>Compiling...

95>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\endpoints\mod_sofia\Win32\Deb
ug\BuildLog.htm"

95>mod_sofia - 6 error(s), 3 warning(s)

98>------ Build started: Project: mod_vmd, Configuration: Debug Win32 ------

97>xmlparse.c

98>Compiling...

98>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

98>mod_vmd.c

97>xmltok.c

97>xmlrole.c

97>Generating Code...

97>Creating library...

97>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\Debug\BuildLog.ht
m"

97>xml - 0 error(s), 0 warning(s)

98>Linking...

98>   Creating library Win32\Debug/mod_vmd.2008.lib and object
Win32\Debug/mod_vmd.2008.exp

99>------ Build started: Project: mod_snom, Configuration: Debug Win32
------

98>Embedding manifest...

99>Compiling...

99>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

99>mod_snom.c

98>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_vmd\Win32\De
bug\BuildLog.htm"

98>mod_vmd - 0 error(s), 1 warning(s)

100>------ Build started: Project: mod_say_zh, Configuration: Debug Win32
------

100>Compiling...

100>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

100>mod_say_zh.c

99>Linking...

99>   Creating library Win32\Debug/mod_snom.2008.lib and object
Win32\Debug/mod_snom.2008.exp

99>Embedding manifest...

100>Linking...

99>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_snom\Win32\D
ebug\BuildLog.htm"

99>mod_snom - 0 error(s), 1 warning(s)

100>   Creating library Win32\Debug/mod_say_zh.2008.lib and object
Win32\Debug/mod_say_zh.2008.exp

100>Embedding manifest...

101>------ Build started: Project: mod_managed, Configuration: Debug_CLR
Win32 ------

101>Compiling...

101>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

101>mod_managed.cpp

100>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\say\mod_say_zh\Win32\Debug\Bu
ildLog.htm"

100>mod_say_zh - 0 error(s), 1 warning(s)

102>------ Build started: Project: 8khz music, Configuration: Debug Win32
------

102>Performing Post-Build Event...

102>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\music\8000\danza-espanola-op-
37-h-142-xii-arabesca.wav

102>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\music\8000\partita-no-3-in-e-
major-bwv-1006-1-preludio.wav

102>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\music\8000\ponce-preludio-in-
e-major.wav

102>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\music\8000\suite-espanola-op-
47-leyenda.wav

102>4 File(s) copied

102>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Sound_Files\Debug\BuildLog
.htm"

102>8khz music - 0 error(s), 0 warning(s)

103>------ Skipped Build: Project: 16khz music, Configuration: Debug Win32
------

103>Project not selected to build for this solution configuration 

104>------ Skipped Build: Project: 32khz music, Configuration: Debug Win32
------

104>Project not selected to build for this solution configuration 

105>------ Build started: Project: mod_say_ru, Configuration: Debug Win32
------

105>Compiling...

105>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

105>mod_say_ru.c

105>Linking...

105>   Creating library Win32\Debug/mod_say_ru.2008.lib and object
Win32\Debug/mod_say_ru.2008.exp

105>Embedding manifest...

101>freeswitch_wrap.cxx

105>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\say\mod_say_ru\Win32\Debug\Bu
ildLog.htm"

105>mod_say_ru - 0 error(s), 1 warning(s)

106>------ Build started: Project: mod_skypopen, Configuration: Debug Win32
------

106>Compiling...

106>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

106>skypopen_protocol.c

106>mod_skypopen.c

106>Generating Code...

106>Linking...

106>   Creating library Win32\Debug/mod_skypopen.2008.lib and object
Win32\Debug/mod_skypopen.2008.exp

106>Embedding manifest...

106>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\endpoints\mod_skypopen\Win32\
Debug\BuildLog.htm"

106>mod_skypopen - 0 error(s), 1 warning(s)

101>freeswitch_managed.cpp

107>------ Build started: Project: mod_loopback, Configuration: Debug Win32
------

107>Compiling...

107>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

107>mod_loopback.c

107>Linking...

107>   Creating library Win32\Debug/mod_loopback.2008.lib and object
Win32\Debug/mod_loopback.2008.exp

107>Embedding manifest...

107>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\endpoints\mod_loopback\Win32\
Debug\BuildLog.htm"

107>mod_loopback - 0 error(s), 1 warning(s)

101>Generating Code...

108>------ Build started: Project: mod_event_socket, Configuration: Debug
Win32 ------

108>Compiling...

108>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

108>mod_event_socket.c

101>Linking...

108>Linking...

108>   Creating library Win32\Debug/mod_event_socket.2008.lib and object
Win32\Debug/mod_event_socket.2008.exp

108>Embedding manifest...

108>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\event_handlers\mod_event_sock
et\Win32\Debug\BuildLog.htm"

108>mod_event_socket - 0 error(s), 1 warning(s)

101>   Creating library Debug_CLR/mod_managed.lib and object
Debug_CLR/mod_managed.exp

109>------ Build started: Project: mod_dptools, Configuration: Debug Win32
------

109>Compiling...

109>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

109>mod_dptools.c

101>freeswitch_wrap.obj : warning LNK4248: unresolved typeref token
(0100001F) for 'switch_odbc_handle'; image may not run

109>Linking...

109>   Creating library Win32\Debug/mod_dptools.2008.lib and object
Win32\Debug/mod_dptools.2008.exp

109>Embedding manifest...

109>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_dptools\Win3
2\Debug\BuildLog.htm"

109>mod_dptools - 0 error(s), 1 warning(s)

101>Embedding manifest...

110>------ Build started: Project: mod_conference, Configuration: Debug
Win32 ------

110>Compiling...

110>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

110>mod_conference.c

101>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_managed\Debug_C
LR\BuildLog.htm"

101>mod_managed - 0 error(s), 2 warning(s)

111>------ Build started: Project: mod_rss, Configuration: Debug Win32
------

111>Compiling...

111>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

111>mod_rss.c

110>.\mod_conference.c(1167) : warning C4018: '<' : signed/unsigned mismatch

110>.\mod_conference.c(1168) : warning C4018: '<=' : signed/unsigned
mismatch

110>.\mod_conference.c(1182) : warning C4018: '<' : signed/unsigned mismatch

110>.\mod_conference.c(1205) : warning C4018: '<' : signed/unsigned mismatch

110>.\mod_conference.c(1208) : warning C4018: '<=' : signed/unsigned
mismatch

110>.\mod_conference.c(2098) : warning C4018: '<' : signed/unsigned mismatch

110>.\mod_conference.c(2101) : warning C4244: '=' : conversion from
'int32_t' to 'int16_t', possible loss of data

110>Linking...

110>   Creating library Win32\Debug/mod_conference.2008.lib and object
Win32\Debug/mod_conference.2008.exp

111>Linking...

111>   Creating library Win32\Debug/mod_rss.2008.lib and object
Win32\Debug/mod_rss.2008.exp

110>Embedding manifest...

110>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_conference\W
in32\Debug\BuildLog.htm"

110>mod_conference - 0 error(s), 8 warning(s)

111>Embedding manifest...

111>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_rss\Win32\De
bug\BuildLog.htm"

111>mod_rss - 0 error(s), 1 warning(s)

112>------ Build started: Project: mod_xml_rpc, Configuration: Debug Win32
------

112>Compiling...

112>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

112>mod_xml_rpc.c

113>------ Build started: Project: mod_console, Configuration: Debug Win32
------

113>Compiling...

113>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

113>mod_console.c

112>.\mod_xml_rpc.c(307) : warning C4090: 'function' : different 'const'
qualifiers

112>.\mod_xml_rpc.c(307) : warning C4244: 'function' : conversion from 'int'
to 'const xmlrpc_uint16_t', possible loss of data

112>.\mod_xml_rpc.c(309) : warning C4090: 'function' : different 'const'
qualifiers

112>.\mod_xml_rpc.c(311) : warning C4090: 'function' : different 'const'
qualifiers

112>.\mod_xml_rpc.c(391) : warning C4090: 'function' : different 'const'
qualifiers

112>.\mod_xml_rpc.c(471) : warning C4090: 'function' : different 'const'
qualifiers

112>.\mod_xml_rpc.c(472) : warning C4090: 'function' : different 'const'
qualifiers

112>.\mod_xml_rpc.c(473) : warning C4090: 'function' : different 'const'
qualifiers

112>Linking...

112>   Creating library Win32\Debug/mod_xml_rpc.2008.lib and object
Win32\Debug/mod_xml_rpc.2008.exp

113>Linking...

113>   Creating library Win32\Debug/mod_console.2008.lib and object
Win32\Debug/mod_console.2008.exp

113>Embedding manifest...

112>Embedding manifest...

113>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\loggers\mod_console\Win32\Deb
ug\BuildLog.htm"

113>mod_console - 0 error(s), 1 warning(s)

112>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\xml_int\mod_xml_rpc\Win32\Deb
ug\BuildLog.htm"

112>mod_xml_rpc - 0 error(s), 9 warning(s)

114>------ Build started: Project: mod_commands, Configuration: Debug Win32
------

114>Compiling...

114>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

114>mod_commands.c

115>------ Build started: Project: mod_dingaling, Configuration: Debug Win32
------

115>Compiling...

115>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

115>mod_dingaling.c

114>Linking...

114>   Creating library Win32\Debug/mod_commands.2008.lib and object
Win32\Debug/mod_commands.2008.exp

115>Linking...

115>   Creating library Win32\Debug/mod_dingaling.2008.lib and object
Win32\Debug/mod_dingaling.2008.exp

114>Embedding manifest...

115>Embedding manifest...

115>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\endpoints\mod_dingaling\Win32
\Debug\BuildLog.htm"

115>mod_dingaling - 0 error(s), 1 warning(s)

114>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_commands\Win
32\Debug\BuildLog.htm"

114>mod_commands - 0 error(s), 1 warning(s)

116>------ Build started: Project: mod_ilbc, Configuration: Debug Win32
------

116>Compiling...

116>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

116>mod_ilbc.c

117>------ Skipped Build: Project: mod_cepstral, Configuration: Debug Win32
------

117>Project not selected to build for this solution configuration 

118>------ Build started: Project: mod_say_es, Configuration: Debug Win32
------

118>Compiling...

118>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

118>mod_say_es.c

116>Linking...

116>   Creating library Win32\Debug/mod_ilbc.2008.lib and object
Win32\Debug/mod_ilbc.2008.exp

118>Linking...

118>   Creating library Win32\Debug/mod_say_es.2008.lib and object
Win32\Debug/mod_say_es.2008.exp

116>Embedding manifest...

118>Embedding manifest...

116>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\codecs\mod_ilbc\Win32\Debug\B
uildLog.htm"

116>mod_ilbc - 0 error(s), 1 warning(s)

118>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\say\mod_say_es\Win32\Debug\Bu
ildLog.htm"

118>mod_say_es - 0 error(s), 1 warning(s)

119>------ Build started: Project: mod_spidermonkey_curl, Configuration:
Debug Win32 ------

119>Compiling...

119>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

119>mod_spidermonkey_curl.c

120>------ Build started: Project: mod_event_multicast, Configuration: Debug
Win32 ------

120>Compiling...

120>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

120>mod_event_multicast.c

120>Linking...

120>   Creating library Win32\Debug/mod_event_multicast.2008.lib and object
Win32\Debug/mod_event_multicast.2008.exp

119>Linking...

119>   Creating library Win32\Debug/mod_spidermonkey_curl.2008.lib and
object Win32\Debug/mod_spidermonkey_curl.2008.exp

120>Embedding manifest...

119>Embedding manifest...

119>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_spidermonkey\Wi
n32\Debug\BuildLog.htm"

119>mod_spidermonkey_curl - 0 error(s), 1 warning(s)

120>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\event_handlers\mod_event_mult
icast\Win32\Debug\BuildLog.htm"

120>mod_event_multicast - 0 error(s), 1 warning(s)

121>------ Build started: Project: mod_dialplan_directory, Configuration:
Debug Win32 ------

121>Compiling...

122>------ Build started: Project: mod_ldap, Configuration: Debug MS-LDAP
Win32 ------

121>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

121>mod_dialplan_directory.c

122>Compiling...

122>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

122>mod_ldap.c

122>Linking...

121>Linking...

121>   Creating library Win32\Debug/mod_dialplan_directory.2008.lib and
object Win32\Debug/mod_dialplan_directory.2008.exp

122>   Creating library Win32\Debug MS-LDAP/mod_ldap.2008.lib and object
Win32\Debug MS-LDAP/mod_ldap.2008.exp

121>Embedding manifest...

122>Embedding manifest...

121>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\dialplans\mod_dialplan_direct
ory\Win32\Debug\BuildLog.htm"

121>mod_dialplan_directory - 0 error(s), 1 warning(s)

122>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\directories\mod_ldap\Win32\De
bug MS-LDAP\BuildLog.htm"

122>mod_ldap - 0 error(s), 1 warning(s)

123>------ Build started: Project: mod_dialplan_xml, Configuration: Debug
Win32 ------

123>Compiling...

123>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

123>mod_dialplan_xml.c

124>------ Skipped Build: Project: docs, Configuration: Debug Win32 ------

124>Project not selected to build for this solution configuration 

125>------ Build started: Project: mod_speex, Configuration: Debug Win32
------

125>Compiling...

125>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

125>mod_speex.c

123>Linking...

123>   Creating library Win32\Debug/mod_dialplan_xml.2008.lib and object
Win32\Debug/mod_dialplan_xml.2008.exp

125>Linking...

125>libspeexdsp.lib(preprocess.obj) : warning LNK4075: ignoring
'/EDITANDCONTINUE' due to '/INCREMENTAL:NO' specification

125>   Creating library Win32\Debug/mod_speex.2008.lib and object
Win32\Debug/mod_speex.2008.exp

123>Embedding manifest...

123>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\dialplans\mod_dialplan_xml\Wi
n32\Debug\BuildLog.htm"

123>mod_dialplan_xml - 0 error(s), 1 warning(s)

125>Embedding manifest...

126>------ Build started: Project: mod_PortAudio, Configuration: Debug Win32
------

126>Compiling...

126>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

126>pablio.c

125>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\codecs\mod_speex\Win32\Debug\
BuildLog.htm"

125>mod_speex - 0 error(s), 2 warning(s)

127>------ Build started: Project: mod_sndfile, Configuration: Debug Win32
------

127>Compiling...

127>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

127>mod_sndfile.c

126>pa_ringbuffer.c

126>mod_PortAudio.c

127>Linking...

126>Generating Code...

127>   Creating library Win32\Debug/mod_sndfile.2008.lib and object
Win32\Debug/mod_sndfile.2008.exp

126>Linking...

126>   Creating library Win32\Debug/mod_PortAudio.2008.lib and object
Win32\Debug/mod_PortAudio.2008.exp

127>Embedding manifest...

126>Embedding manifest...

127>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\formats\mod_sndfile\Win32\Deb
ug\BuildLog.htm"

127>mod_sndfile - 0 error(s), 1 warning(s)

126>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\endpoints\mod_portaudio\Win32
\Debug\BuildLog.htm"

126>mod_PortAudio - 0 error(s), 1 warning(s)

128>------ Build started: Project: mod_g729, Configuration: Debug
Passthrough Win32 ------

128>Compiling...

128>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

128>mod_g729.c

129>------ Build started: Project: mod_valet_parking, Configuration: Debug
Win32 ------

129>Compiling...

129>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

129>mod_valet_parking.c

128>Linking...

128>   Creating library Win32\Debug Passthrough/mod_g729.2008.lib and object
Win32\Debug Passthrough/mod_g729.2008.exp

128>Embedding manifest...

128>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\codecs\mod_g729\Win32\Debug
Passthrough\BuildLog.htm"

128>mod_g729 - 0 error(s), 1 warning(s)

129>Linking...

129>   Creating library Win32\Debug/mod_valet_parking.2008.lib and object
Win32\Debug/mod_valet_parking.2008.exp

129>Embedding manifest...

129>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_valet_parkin
g\Win32\Debug\BuildLog.htm"

129>mod_valet_parking - 0 error(s), 1 warning(s)

130>------ Build started: Project: FreeSwitchConsole, Configuration: Debug
Win32 ------

130>Compiling...

130>cl : Command line warning D9040 : ignoring option '/analyze'; Code
Analysis warnings are not available in this edition of the compiler

130>switch.c

130>Linking...

130>Embedding manifest...

130>Build log was saved at
"file://c:\FreeSWITCH\freeswitch-1.0.6\w32\Console\Debug\BuildLog
FreeSwitchConsole.htm"

130>FreeSwitchConsole - 0 error(s), 1 warning(s)

========== Build: 117 succeeded, 2 failed, 0 up-to-date, 11 skipped
==========

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From mlistsdm at gmail.com  Mon Dec 20 01:16:06 2010
From: mlistsdm at gmail.com (Dmitry M)
Date: Sun, 19 Dec 2010 17:16:06 -0500
Subject: [Freeswitch-users] attack, still no ip address logging. Fail2Ban
In-Reply-To: <71F57ED7-90C3-4181-8615-58E3F9A6CE66@freeswitch.org>
References: <18577.1290350714@ccs.covici.com>
	<71F57ED7-90C3-4181-8615-58E3F9A6CE66@freeswitch.org>
Message-ID: <4D0E8426.7090702@gmail.com>

Brian West wrote:
> Also if you go enable the profile debug you'll get this
> 
>             if (profile->debug) {                                                                                                                               
>                 switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Send %s for [%s@%s]\n", forbidden ? "forbidden" : "challenge", to_user, to_host);     
>             } 
> 
> for every single register also so you can use fail2ban.
> 
> /b
> 

Unfortunately some logging is missing which don't allow to use Fail2Ban
effectively.

I've added a patch http://jira.freeswitch.org/browse/FS-2943
It allows Fail2ban to monitor all authentication attempts and react
properly.

It'll be great if some of developers check it asap and apply to the main
tree.

Dmitry

> On Nov 21, 2010, at 8:45 AM, covici at ccs.covici.com wrote:
> 
>> I don't mean to bother people, but it was a pain and I couldn't make any
>> calls for some time till I fixed things manually.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From peter.olsson at visionutveckling.se  Mon Dec 20 13:50:52 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 20 Dec 2010 11:50:52 +0100
Subject: [Freeswitch-users] Error in compiling tar
	ball	freeswitch-1.0.6.tar.gz on windows
In-Reply-To: <SNT109-DS5FB277D37D16C78A3FF61C1190@phx.gbl>
References: <SNT109-DS5FB277D37D16C78A3FF61C1190@phx.gbl>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57ECCA34B4@cooper>

Please follow the instructions on wiki. 1.0.6 is quite old these days, you should use latest git HEAD.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Norman Lam
Skickat: den 20 december 2010 11:20
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Error in compiling tar ball freeswitch-1.0.6.tar.gz on windows

Hi , I am newbie , can't compile on AMD Athon X2, winxp sp3, vc++2008 express, follow the packt book , and the freeswitch website

Please advise, thanks advance

Norman


1>------ Build started: Project: libpcre Generate pcre_chartables.c, Configuration: Debug Win32 ------
2>------ Build started: Project: libapr, Configuration: Debug Win32 ------
1>Compiling...
2>Performing Pre-Build Event...
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_allocator.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_atomic.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_dso.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_env.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_errno.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_file_info.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_file_io.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_fnmatch.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_general.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_getopt.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_global_mutex.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_hash.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_inherit.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_lib.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_mmap.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_network_io.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_poll.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_pools.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_portable.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_proc_mutex.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_random.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_ring.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_shm.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_signal.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_strings.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_support.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_tables.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_thread_cond.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_thread_mutex.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_thread_proc.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_thread_rwlock.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_time.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_user.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_version.h
2>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\..\..\apr\include\apr_want.h
2>36 File(s) copied
2>Compiling...
1>dftables.c
2>userinfo.c
1>Compiling manifest to resources...
1>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
1>Copyright (C) Microsoft Corporation.  All rights reserved.
1>Linking...
1>   Creating library Debug\dftables.lib and object Debug\dftables.exp
2>groupinfo.c
1>Embedding manifest...
2>timestr.c
1>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
1>Copyright (C) Microsoft Corporation.  All rights reserved.
2>time.c
1>Performing Post-Build Event...
2>access.c
1>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\pcre\Debug\BuildLog.htm"
1>libpcre Generate pcre_chartables.c - 0 error(s), 0 warning(s)
2>threadpriv.c
3>------ Build started: Project: libpcre, Configuration: Debug Win32 ------
3>libpcre : warning PRJ0009 : Build log could not be opened for writing.
3>Make sure that the file is not open by another process and is not write-protected.
3>Compiling...
3>pcre_compile.c
2>thread.c
3>pcre_config.c
3>pcre_dfa_exec.c
2>signals.c
3>pcre_exec.c
2>proc.c
3>pcre_fullinfo.c
3>pcre_get.c
2>apr_tables.c
3>pcre_globals.c
2>apr_hash.c
3>pcre_info.c
2>apr_strtok.c
3>pcre_maketables.c
3>pcre_ord2utf8.c
2>apr_strnatcmp.c
3>pcre_refcount.c
2>apr_strings.c
3>pcre_study.c
3>pcre_tables.c
2>apr_snprintf.c
3>pcre_try_flipped.c
3>pcre_newline.c
2>apr_fnmatch.c
3>pcre_ucd.c
3>pcre_ucp_searchfuncs.c
2>apr_cpystrn.c
3>..\..\pcre\pcre_ucp_searchfuncs.c(158) : warning C4018: '<' : signed/unsigned mismatch
3>..\..\pcre\pcre_ucp_searchfuncs.c(163) : warning C4018: '<=' : signed/unsigned mismatch
3>pcre_valid_utf8.c
2>shm.c
3>pcre_version.c
3>pcre_xclass.c
2>sha2_glue.c
3>Generating Code...
2>sha2.c
2>Generating Code...
2>Compiling...
2>apr_random.c
3>Compiling...
3>pcre_chartables.c
3>Generating Code...
3>Creating library...
2>apr_getpass.c
3>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\pcre\Debug\BuildLog.htm"
3>libpcre - 0 error(s), 3 warning(s)
4>------ Build started: Project: libsqlite, Configuration: Debug Win32 ------
4>Performing Pre-Build Event...
2>sockopt.c
4>Compiling...
4>analyze.c
4>attach.c
2>sockets.c
4>auth.c
4>btree.c
2>sockaddr.c
4>build.c
4>callback.c
4>complete.c
2>sendrecv.c
4>date.c
4>delete.c
2>select.c
4>expr.c
4>func.c
2>multicast.c
4>hash.c
4>insert.c
4>legacy.c
2>inet_pton.c
4>loadext.c
2>inet_ntop.c
2>mmap.c
4>main.c
4>opcodes.c
4>os.c
2>common.c
4>os_win.c
2>version.c
2>utf8.c
4>pager.c
2>start.c
4>Generating Code...
4>Compiling...
4>parse.c
2>rand.c
4>pragma.c
2>otherchild.c
4>prepare.c
2>misc.c
4>printf.c
4>random.c
2>internal.c
4>select.c
4>shell.c
4>table.c
4>tokenize.c
2>getopt.c
2>Generating Code...
4>trigger.c
2>Compiling...
2>errorcodes.c
4>update.c
4>utf.c
4>util.c
2>env.c
4>vacuum.c
2>charset.c
4>vdbe.c
4>vdbeapi.c
2>apr_pools.c
4>vdbeaux.c
2>thread_rwlock.c
4>vdbefifo.c
4>vdbemem.c
2>thread_mutex.c
4>vtab.c
2>thread_cond.c
2>proc_mutex.c
4>Generating Code...
4>Compiling...
4>where.c
4>alter.c
2>tempdir.c
4>Generating Code...
4>Creating library...
4>Performing Post-Build Event...
4>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\sqlite\Debug\BuildLog.htm"
4>libsqlite - 0 error(s), 0 warning(s)
2>seek.c
5>------ Build started: Project: libsrtp, Configuration: Debug Win32 ------
2>readwrite.c
5>Creating config.h from config.hw
5>Compiling...
5>alloc.c
2>pipe.c
2>open.c
5>crypto_kernel.c
2>mktemp.c
2>fullrw.c
5>ctr_prng.c
2>flock.c
5>err.c
2>filesys.c
2>filestat.c
5>key.c
5>prng.c
2>filepath_util.c
2>filepath.c
5>rand_source.c
2>Generating Code...
5>aes.c
2>Compiling...
2>filedup.c
5>aes_cbc.c
2>fileacc.c
5>aes_icm.c
2>dir.c
2>copy.c
5>cipher.c
2>dso.c
5>null_cipher.c
2>apr_atomic.c
2>Generating Code...
5>auth.c
2>Compiling resources...
2>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
2>Copyright (C) Microsoft Corporation.  All rights reserved.
2>Linking...
2>   Creating library .\Debug/libapr-1.lib and object .\Debug/libapr-1.exp
5>hmac.c
5>null_auth.c
2>Embedding manifest...
2>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr\Debug\BuildLog.htm"
2>libapr - 0 error(s), 0 warning(s)
5>sha1.c
6>------ Build started: Project: libteletone, Configuration: Debug Win32 ------
5>rdb.c
5>rdbx.c
6>Compiling...
6>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
6>libteletone_detect.c
6>libteletone_generate.c
6>Generating Code...
6>Linking...
6>   Creating library Debug/libteletone.lib and object Debug/libteletone.exp
6>Embedding manifest...
5>ut_sim.c
5>datatypes.c
6>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\libteletone\Debug\BuildLog.htm"
6>libteletone - 0 error(s), 1 warning(s)
7>------ Build started: Project: libaprutil, Configuration: Debug Win32 ------
7>Performing Pre-Build Event...
7>The system cannot find the file specified.
7>The system cannot find the file specified.
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_anylock.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_base64.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_buckets.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_date.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_dbd.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_dbm.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_hooks.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_ldap.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_ldap_init.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_ldap_option.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_ldap_url.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_md4.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_md5.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_optional.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_optional_hooks.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_queue.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_reslist.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_rmm.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_sdbm.h
5>Generating Code...
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_sha1.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_strmatch.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_uri.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_uuid.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_xlate.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apr_xml.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apu.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apu_config.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apu_select_dbm.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apu_version.h
7>C:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\..\..\apr-util\include\apu_want.h
7>30 File(s) copied
7>Creating apu_want.h from apu_want.hw
7>Creating apu_select_dbm.h from apu_select_dbm.hw
7>Creating apu_config.h from apu_config.hw
7>Creating apu.h from apu.hw
5>Compiling...
5>stat.c
7>Creating apr_ldap.h from apr_ldap.hw
7>Compiling...
7>xlate.c
7>apr_queue.c
7>apr_base64.c
5>srtp.c
7>uuid.c
7>getuuid.c
7>apr_sha1.c
7>apr_md5.c
5>Generating Code...
5>Creating library...
7>apr_md4.c
5>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\srtp\Debug\BuildLog.htm"
5>libsrtp - 0 error(s), 0 warning(s)
7>Generating Code...
8>------ Build started: Project: Download sphinxmodel, Configuration: Debug Win32 ------
7>Compiling resources...
8>Downloading sphinxmodel.
7>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
7>Copyright (C) Microsoft Corporation.  All rights reserved.
8>Downloading: http://files.freeswitch.org/downloads/win32/7za.exe
8>Downloading: http://files.freeswitch.org/downloads/libs/communicator_semi_6000_20080321.tar.gz
8>Extracting: C:\FreeSWITCH\freeswitch-1.0.6\libs\communicator_semi_6000_20080321.tar.gz
8>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
8>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\communicator_semi_6000_20080321.tar.gz
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8>Everything is Ok
8>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
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9>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
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11>Extracting  pthreads-w32-2-7-0-release\tests\semaphore3.c
11>Extracting  pthreads-w32-2-7-0-release\tests\semaphore4.c
11>Extracting  pthreads-w32-2-7-0-release\tests\semaphore4t.c
11>Extracting  pthreads-w32-2-7-0-release\tests\sizes.c
11>Extracting  pthreads-w32-2-7-0-release\tests\SIZES.GC
11>Extracting  pthreads-w32-2-7-0-release\tests\SIZES.GCE
11>Extracting  pthreads-w32-2-7-0-release\tests\SIZES.VC
11>Extracting  pthreads-w32-2-7-0-release\tests\SIZES.VCE
11>Extracting  pthreads-w32-2-7-0-release\tests\SIZES.VSE
11>Extracting  pthreads-w32-2-7-0-release\tests\spin1.c
11>Extracting  pthreads-w32-2-7-0-release\tests\spin2.c
11>Extracting  pthreads-w32-2-7-0-release\tests\spin3.c
11>Extracting  pthreads-w32-2-7-0-release\tests\spin4.c
11>Extracting  pthreads-w32-2-7-0-release\tests\stress1.c
11>Extracting  pthreads-w32-2-7-0-release\tests\test.h
11>Extracting  pthreads-w32-2-7-0-release\tests\tryentercs.c
11>Extracting  pthreads-w32-2-7-0-release\tests\tryentercs2.c
11>Extracting  pthreads-w32-2-7-0-release\tests\tsd1.c
11>Extracting  pthreads-w32-2-7-0-release\tests\tsd2.c
11>Extracting  pthreads-w32-2-7-0-release\tests\valid1.c
11>Extracting  pthreads-w32-2-7-0-release\tests\valid2.c
11>Extracting  pthreads-w32-2-7-0-release\tests\Wmakefile
11>Extracting  pthreads-w32-2-7-0-release\TODO
11>Extracting  pthreads-w32-2-7-0-release\tsd.c
11>Extracting  pthreads-w32-2-7-0-release\version.rc
11>Extracting  pthreads-w32-2-7-0-release\w32_CancelableWait.c
11>Extracting  pthreads-w32-2-7-0-release\WinCE-PORT
11>Everything is Ok
9>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download sphinxbase.htm"
9>Download sphinxbase - 0 error(s), 0 warning(s)
12>------ Build started: Project: Download OGG, Configuration: Debug Win32 ------
12>Downloading Lame.
11>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download PTHREAD.htm"
11>Download PTHREAD - 0 error(s), 0 warning(s)
13>------ Build started: Project: make_at_dictionary, Configuration: All Win32 ------
13>Compiling...
12>Downloading: http://downloads.xiph.org/releases/ogg/libogg-1.1.3.tar.gz
13>make_at_dictionary.c
13>Linking...
13>Embedding manifest...
13>Performing Post-Build Event...
13>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\spandsp\src\msvc\All\BuildLog make_at_dictionary.htm"
13>make_at_dictionary - 0 error(s), 0 warning(s)
12>Extracting: C:\FreeSWITCH\freeswitch-1.0.6\libs\libogg-1.1.3.tar.gz
12>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
12>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\libogg-1.1.3.tar.gz
12>Extracting  libogg-1.1.3.tar
12>Everything is Ok
14>------ Build started: Project: pthread, Configuration: Debug DLL Win32 ------
14>Compiling...
14>pthread.c
12>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
12>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\libogg-1.1.3.tar
12>Extracting  libogg-1.1.3
12>Extracting  libogg-1.1.3\doc
12>Extracting  libogg-1.1.3\doc\white-ogg.png
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12>Extracting  libogg-1.1.3\doc\index.html
12>Extracting  libogg-1.1.3\doc\ogg-multiplex.html
12>Extracting  libogg-1.1.3\doc\Makefile.am
12>Extracting  libogg-1.1.3\doc\Makefile.in
12>Extracting  libogg-1.1.3\doc\white-xifish.png
12>Extracting  libogg-1.1.3\doc\libogg
12>Extracting  libogg-1.1.3\doc\libogg\ogg_page_continued.html
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12>Extracting  libogg-1.1.3\doc\libogg\style.css
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12>Extracting  libogg-1.1.3\doc\libogg\oggpack_adv1.html
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12>Extracting  libogg-1.1.3\doc\libogg\oggpack_readinit.html
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12>Extracting  libogg-1.1.3\doc\libogg\ogg_page_checksum_set.html
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12>Extracting  libogg-1.1.3\doc\libogg\overview.html
12>Extracting  libogg-1.1.3\doc\libogg\oggpack_adv.html
12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_clear.html
12>Extracting  libogg-1.1.3\doc\libogg\ogg_sync_wrote.html
12>Extracting  libogg-1.1.3\doc\libogg\oggpack_bytes.html
12>Extracting  libogg-1.1.3\doc\libogg\Makefile.am
12>Extracting  libogg-1.1.3\doc\libogg\Makefile.in
12>Extracting  libogg-1.1.3\doc\libogg\oggpack_reset.html
12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_packetpeek.html
12>Extracting  libogg-1.1.3\doc\libogg\reference.html
12>Extracting  libogg-1.1.3\doc\libogg\ogg_sync_pageseek.html
12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_destroy.html
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12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_pageout.html
12>Extracting  libogg-1.1.3\doc\libogg\ogg_page_granulepos.html
12>Extracting  libogg-1.1.3\doc\libogg\oggpack_look.html
12>Extracting  libogg-1.1.3\doc\libogg\bitpacking.html
12>Extracting  libogg-1.1.3\doc\libogg\ogg_page.html
12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_packetout.html
12>Extracting  libogg-1.1.3\doc\libogg\decoding.html
12>Extracting  libogg-1.1.3\doc\libogg\ogg_sync_init.html
12>Extracting  libogg-1.1.3\doc\libogg\oggpack_writealign.html
12>Extracting  libogg-1.1.3\doc\libogg\vorbis_comment.html
12>Extracting  libogg-1.1.3\doc\libogg\oggpack_get_buffer.html
12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_pagein.html
12>Extracting  libogg-1.1.3\doc\libogg\ogg_page_version.html
12>Extracting  libogg-1.1.3\doc\libogg\ogg_sync_destroy.html
14>Compiling resources...
14>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
14>Copyright (C) Microsoft Corporation.  All rights reserved.
14>Compiling manifest to resources...
14>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
14>Copyright (C) Microsoft Corporation.  All rights reserved.
14>Linking...
12>Extracting  libogg-1.1.3\doc\libogg\oggpack_buffer.html
12>Extracting  libogg-1.1.3\doc\libogg\ogg_stream_init.html
12>Extracting  libogg-1.1.3\doc\vorbisword2.png
12>Extracting  libogg-1.1.3\doc\stream.png
12>Extracting  libogg-1.1.3\doc\oggstream.html
12>Extracting  libogg-1.1.3\doc\rfc3533.txt
12>Extracting  libogg-1.1.3\doc\rfc3534.txt
12>Extracting  libogg-1.1.3\src
12>Extracting  libogg-1.1.3\src\framing.c
12>Extracting  libogg-1.1.3\src\bitwise.c
12>Extracting  libogg-1.1.3\src\Makefile.am
12>Extracting  libogg-1.1.3\src\Makefile.in
12>Extracting  libogg-1.1.3\compile
12>Extracting  libogg-1.1.3\depcomp
12>Extracting  libogg-1.1.3\aclocal.m4
12>Extracting  libogg-1.1.3\macos
12>Extracting  libogg-1.1.3\macos\libogg.mcp
12>Extracting  libogg-1.1.3\macos\libogg.mcp.exp
12>Extracting  libogg-1.1.3\macos\compat
12>Extracting  libogg-1.1.3\macos\compat\sys
12>Extracting  libogg-1.1.3\macos\compat\sys\types.h
12>Extracting  libogg-1.1.3\macos\compat\strdup.c
12>Extracting  libogg-1.1.3\win32
12>Extracting  libogg-1.1.3\win32\ogg_dynamic.dsp
12>Extracting  libogg-1.1.3\win32\build_ogg_dynamic_debug.bat
12>Extracting  libogg-1.1.3\win32\build_ogg_dynamic.bat
12>Extracting  libogg-1.1.3\win32\Makefile.am
12>Extracting  libogg-1.1.3\win32\Makefile.in
12>Extracting  libogg-1.1.3\win32\build_ogg_static_debug.bat
12>Extracting  libogg-1.1.3\win32\ogg_static.dsp
12>Extracting  libogg-1.1.3\win32\ogg.def
12>Extracting  libogg-1.1.3\win32\ogg.dsw
12>Extracting  libogg-1.1.3\win32\build_ogg_static.bat
12>Extracting  libogg-1.1.3\README
12>Extracting  libogg-1.1.3\ltmain.sh
12>Extracting  libogg-1.1.3\configure
12>Extracting  libogg-1.1.3\configure.in
12>Extracting  libogg-1.1.3\config.guess
12>Extracting  libogg-1.1.3\install-sh
12>Extracting  libogg-1.1.3\config.sub
12>Extracting  libogg-1.1.3\missing
12>Extracting  libogg-1.1.3\debian
12>Extracting  libogg-1.1.3\debian\control
12>Extracting  libogg-1.1.3\debian\libogg-dev.docs
12>Extracting  libogg-1.1.3\debian\rules
12>Extracting  libogg-1.1.3\debian\watch
12>Extracting  libogg-1.1.3\debian\changelog
12>Extracting  libogg-1.1.3\debian\libogg0.README.Debian
12>Extracting  libogg-1.1.3\debian\libogg-dev.install
12>Extracting  libogg-1.1.3\debian\copyright
12>Extracting  libogg-1.1.3\debian\libogg0.install
12>Extracting  libogg-1.1.3\debian\.cvsignore
12>Extracting  libogg-1.1.3\libogg.spec.in
12>Extracting  libogg-1.1.3\ogg.pc.in
12>Extracting  libogg-1.1.3\Makefile.am
12>Extracting  libogg-1.1.3\Makefile.in
12>Extracting  libogg-1.1.3\macosx
12>Extracting  libogg-1.1.3\macosx\Ogg.xcodeproj
12>Extracting  libogg-1.1.3\macosx\Ogg.xcodeproj\project.pbxproj
12>Extracting  libogg-1.1.3\macosx\Ogg_Prefix.pch
12>Extracting  libogg-1.1.3\macosx\English.lproj
12>Extracting  libogg-1.1.3\macosx\English.lproj\InfoPlist.strings
12>Extracting  libogg-1.1.3\macosx\Info.plist
12>Extracting  libogg-1.1.3\config.h.in
12>Extracting  libogg-1.1.3\ogg-uninstalled.pc.in
12>Extracting  libogg-1.1.3\ogg.m4
12>Extracting  libogg-1.1.3\AUTHORS
12>Extracting  libogg-1.1.3\CHANGES
12>Extracting  libogg-1.1.3\include
12>Extracting  libogg-1.1.3\include\ogg
12>Extracting  libogg-1.1.3\include\ogg\ogg.h
12>Extracting  libogg-1.1.3\include\ogg\Makefile.am
12>Extracting  libogg-1.1.3\include\ogg\Makefile.in
12>Extracting  libogg-1.1.3\include\ogg\config_types.h.in
12>Extracting  libogg-1.1.3\include\ogg\os_types.h
12>Extracting  libogg-1.1.3\include\Makefile.am
12>Extracting  libogg-1.1.3\include\Makefile.in
12>Extracting  libogg-1.1.3\COPYING
12>Extracting  libogg-1.1.3\libogg.spec
12>Everything is Ok
14>   Creating library .\./pthreadVC2.lib and object .\./pthreadVC2.exp
14>Embedding manifest...
14>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
14>Copyright (C) Microsoft Corporation.  All rights reserved.
14>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\pthread\Debug DLL\BuildLog.htm"
14>pthread - 0 error(s), 0 warning(s)
15>------ Build started: Project: make_modem_filter, Configuration: All Win32 ------
15>Compiling...
15>filter_tools.c
15>getopt.c
15>make_modem_filter.c
12>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download OGG.htm"
12>Download OGG - 0 error(s), 0 warning(s)
15>Generating Code...
15>Linking...
16>------ Build started: Project: FreeSwitchCoreLib, Configuration: Debug Win32 ------
15>Embedding manifest...
16>Generating switch_version.h
15>Performing Post-Build Event...
16>Downloading: http://files.freeswitch.org/downloads/win32/fs_svnversion.exe
15>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\spandsp\src\msvc\All\BuildLog make_modem_filter.htm"
15>make_modem_filter - 0 error(s), 0 warning(s)
17>------ Build started: Project: curllib, Configuration: Debug Win32 ------
17>Compiling...
16>Downloading: http://files.freeswitch.org/downloads/win32/libdb44.dll
17>connect.c
17>content_encoding.c
16>Downloading: http://files.freeswitch.org/downloads/win32/libsvn_diff-1.dll
17>cookie.c
17>dict.c
16>Downloading: http://files.freeswitch.org/downloads/win32/libsvn_subr-1.dll
17>easy.c
16>Downloading: http://files.freeswitch.org/downloads/win32/libsvn_wc-1.dll
17>escape.c
16>Downloading: http://files.freeswitch.org/downloads/win32/intl3_svn.dll
17>file.c
16>Downloading: http://files.freeswitch.org/downloads/win32/libapr-1.dll
17>formdata.c
16>Downloading: http://files.freeswitch.org/downloads/win32/libaprutil-1.dll
17>ftp.c
16>Downloading: http://files.freeswitch.org/downloads/win32/libapriconv-1.dll
17>getenv.c
16>Downloading: http://files.freeswitch.org/downloads/win32/libsvn_delta-1.dll
17>getinfo.c
17>gtls.c
17>hash.c
17>hostares.c
17>hostasyn.c
16>Compiling...
16>switch_buffer.c
17>hostip.c
17>hostip4.c
17>hostip6.c
17>hostsyn.c
17>hostthre.c
17>Generating Code...
17>Compiling...
17>http.c
17>http_chunks.c
17>http_digest.c
17>http_negotiate.c
17>http_ntlm.c
17>if2ip.c
17>inet_ntop.c
17>inet_pton.c
17>krb4.c
17>ldap.c
17>llist.c
17>md5.c
17>memdebug.c
17>mprintf.c
17>multi.c
17>netrc.c
17>parsedate.c
17>progress.c
17>security.c
17>select.c
16>Compiling...
16>switch_apr.c
17>Generating Code...
17>Compiling...
17>sendf.c
16>switch_caller.c
17>share.c
16>switch_channel.c
17>socks.c
16>switch_config.c
17>speedcheck.c
16>switch_console.c
17>splay.c
16>switch_core.c
17>sslgen.c
17>ssluse.c
16>switch_core_asr.c
17>strequal.c
16>switch_core_codec.c
17>strerror.c
16>switch_core_db.c
17>strtok.c
16>switch_core_directory.c
17>strtoofft.c
16>switch_core_event_hook.c
17>telnet.c
16>switch_core_file.c
17>tftp.c
17>timeval.c
16>switch_core_hash.c
17>transfer.c
17>url.c
16>switch_core_io.c
17>version.c
16>switch_core_media_bug.c
17>base64.c
17>Generating Code...
16>switch_core_memory.c
17>Creating library...
16>switch_core_port_allocator.c
17>Creating browse information file...
16>switch_core_rwlock.c
16>switch_core_speech.c
16>switch_core_sqldb.c
16>Generating Code...
16>Compiling...
16>switch_core_state_machine.c
16>switch_core_timer.c
16>switch_dso.c
16>switch_event.c
16>switch_ivr.c
16>switch_ivr_async.c
16>switch_ivr_bridge.c
16>switch_ivr_menu.c
16>switch_ivr_play_say.c
16>switch_ivr_say.c
16>switch_loadable_module.c
16>switch_log.c
16>switch_mprintf.c
16>switch_odbc.c
16>switch_pcm.c
16>switch_profile.c
16>switch_regex.c
16>switch_resample.c
16>switch_rtp.c
16>switch_scheduler.c
16>Generating Code...
16>Compiling...
16>switch_stun.c
16>switch_time.c
16>switch_utils.c
16>switch_xml.c
16>switch_xml_config.c
16>Generating Code...
16>Compiling...
16>getgateway.c
16>natpmp.c
16>Generating Code...
16>Compiling...
16>igd_desc_parse.c
16>minisoap.c
16>minissdpc.c
16>miniwget.c
17>Microsoft Browse Information Maintenance Utility Version 9.00.30729
17>Copyright (C) Microsoft Corporation. All rights reserved.
17>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\curl\Debug\BuildLog.htm"
17>curllib - 0 error(s), 0 warning(s)
16>minixml.c
16>upnpcommands.c
16>upnperrors.c
16>upnpreplyparse.c
18>------ Build started: Project: libtiff, Configuration: Debug Win32 ------
18>Performing Pre-Build Event...
16>Generating Code...
18>        1 file(s) copied.
18>        1 file(s) copied.
18>Compiling...
16>Compiling...
18>tif_close.c
16>miniupnpc.c
18>tif_codec.c
18>tif_color.c
18>tif_compress.c
18>tif_dir.c
16>Compiling...
18>tif_dirinfo.c
16>switch_nat.c
18>tif_dirread.c
18>tif_dirwrite.c
18>tif_dumpmode.c
16>Compiling...
18>tif_error.c
16>switch_ivr_originate.c
18>tif_extension.c
18>tif_fax3.c
18>tif_fax3sm.c
18>tif_flush.c
18>tif_getimage.c
16>Compiling...
16>switch_cpp.cpp
18>tif_jpeg.c
18>tif_luv.c
18>tif_lzw.c
18>tif_next.c
18>tif_ojpeg.c
18>Generating Code...
18>Compiling...
18>tif_open.c
18>tif_packbits.c
18>tif_pixarlog.c
18>tif_predict.c
18>tif_print.c
18>tif_read.c
18>tif_strip.c
18>tif_swab.c
18>tif_thunder.c
18>tif_tile.c
18>tif_unix.c
18>tif_version.c
18>tif_warning.c
18>tif_write.c
18>tif_zip.c
18>tif_aux.c
18>Generating Code...
18>Creating library...
18>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\spandsp\src\Debug\BuildLog libtiff.htm"
18>libtiff - 0 error(s), 0 warning(s)
19>------ Build started: Project: iksemel, Configuration: Debug Win32 ------
19>Compiling...
19>dom.c
19>filter.c
19>iks.c
19>ikstack.c
19>io-posix.c
16>Compiling...
16>switch_core_session.c
19>jabber.c
16>Compiling...
19>md5.c
16>stfu.c
19>sax.c
19>sha.c
16>Compiling...
16>g711.c
19>stream.c
16>inet_pton.c
19>utility.c
19>base64.c
19>Generating Code...
19>Creating library...
19>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\iksemel\Debug\BuildLog.htm"
19>iksemel - 0 error(s), 0 warning(s)
16>Generating Code...
16>Compiling manifest to resources...
16>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
16>Copyright (C) Microsoft Corporation.  All rights reserved.
20>------ Build started: Project: Download pocketsphinx, Configuration: Debug Win32 ------
16>Linking...
20>Downloading pocketsphinx.
20>Downloading: http://files.freeswitch.org/downloads/libs/pocketsphinx-0.5.99-20091212.tar.gz
16>   Creating library Debug/FreeSwitchCore.lib and object Debug/FreeSwitchCore.exp
16>Embedding manifest...
16>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
16>Copyright (C) Microsoft Corporation.  All rights reserved.
16>Performing Post-Build Event...
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\extensions.conf
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\freeswitch.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\fur_elise.ttml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\mime.types
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\notify-voicemail.tpl
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\tetris.ttml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\vars.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\voicemail.tpl
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\web-vm.tpl
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\acl.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\alsa.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\cdr_csv.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\cdr_pg_csv.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\cidlookup.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\conference.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\console.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\dialplan_directory.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\dingaling.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\directory.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\distributor.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\easyroute.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\enum.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\erlang_event.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\event_multicast.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\event_socket.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\fax.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\fifo.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\ivr.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\java.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\lcr.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\limit.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\local_stream.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\logfile.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\lua.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\memcache.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\modules.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\nibblebill.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\opal.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\perl.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\pocketsphinx.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\portaudio.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\post_load_modules.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\python.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\rss.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\sangoma_codec.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\shout.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\skinny.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\sofia.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\spidermonkey.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\switch.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\syslog.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\timezones.conf.xml
20>Extracting: C:\FreeSWITCH\freeswitch-1.0.6\libs\pocketsphinx-0.5.99-20091212.tar.gz
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\tts_commandline.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\unicall.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\unimrcp.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\voicemail.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\xml_cdr.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\xml_curl.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\xml_rpc.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\autoload_configs\zeroconf.conf.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\features.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\public.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default\00_pizza_demo.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default\01_example.com.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default\99999_enum.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default\ideasip.com.noload
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default\pulver.com.noload
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default\sipbroker.com.noload
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default\sipphone.com.noload
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\default\tollfreegateway.com.noload
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\dialplan\public\00_inbound_did.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1000.xml
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16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1007.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1008.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1009.xml
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16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1017.xml
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16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\1019.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\brian.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\default.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\directory\default\example.com.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\ivr_menus\demo_ivr.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\jingle_profiles\client.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\jingle_profiles\server.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\de\de.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\de\demo\demo.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\de\vm\tts.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\en\en.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\en\demo\demo-ivr.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\en\demo\demo.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\en\dir\sounds.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\en\dir\tts.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\en\vm\sounds.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\en\vm\tts.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\fr\fr.xml
20>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
20>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\pocketsphinx-0.5.99-20091212.tar.gz
20>Extracting  pocketsphinx-0.5.99-20091212.tar
20>Everything is Ok
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\fr\demo\demo.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\fr\dir\sounds.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\fr\dir\tts.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\fr\vm\sounds.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\ru\ru.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\ru\demo\demo-ivr.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\ru\demo\demo.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\ru\dir\sounds.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\ru\dir\tts.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\ru\vm\sounds.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\lang\ru\vm\tts.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\mrcp_profiles\loquendo-7-mrcp-v2.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\mrcp_profiles\nuance-1.0.0-mrcp-v1.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\mrcp_profiles\nuance-5.0-mrcp-v1.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\mrcp_profiles\nuance-5.0-mrcp-v2.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\mrcp_profiles\unimrcpserver-mrcp-v1.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\mrcp_profiles\voxeo-prophecy-8.0-mrcp-v1.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\sip_profiles\external.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\sip_profiles\internal-ipv6.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\sip_profiles\internal.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\sip_profiles\external\example.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\sip_profiles\internal\example.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\skinny_profiles\internal.xml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\yaml\extensions.yaml
16>C:\FreeSWITCH\freeswitch-1.0.6\conf\yaml\mod_yaml.yaml
16>135 File(s) copied
16>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\w32\Library\Debug\BuildLog FreeSwitchCoreLib.htm"
16>FreeSwitchCoreLib - 0 error(s), 0 warning(s)
20>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
20>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\pocketsphinx-0.5.99-20091212.tar
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21>------ Build started: Project: sphinxbase, Configuration: Debug Win32 ------
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20>Everything is Ok
21>dtoa.c
21>err.c
21>f2c_lite.c
21>fe_interface.c
21>fe_sigproc.c
21>fe_warp.c
21>fe_warp_affine.c
21>fe_warp_inverse_linear.c
20>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download pocketsphinx.htm"
20>Download pocketsphinx - 0 error(s), 0 warning(s)
21>fe_warp_piecewise_linear.c
21>feat.c
22>------ Build started: Project: Download FLITE, Configuration: Debug Win32 ------
22>Downloading Flite.
21>Generating Code...
22>Downloading: http://files.freeswitch.org/downloads/libs/flite-1.3.99-latest.tar.gz
21>Compiling...
21>filename.c
21>fixlog.c
21>fsg_model.c
21>genrand.c
21>glist.c
21>hash_table.c
21>heap.c
21>info.c
21>jsgf.c
21>jsgf_parser.c
21>jsgf_scanner.c
21>lda.c
21>listelem_alloc.c
21>lm3g_model.c
21>logmath.c
21>matrix.c
21>mmio.c
21>ngram_model.c
21>ngram_model_arpa.c
21>ngram_model_dmp.c
21>Generating Code...
21>Compiling...
21>ngram_model_dmp32.c
21>ngram_model_set.c
21>pio.c
21>play_win32.c
21>profile.c
21>rec_win32.c
21>sbthread.c
21>slamch.c
21>slapack_lite.c
21>strfuncs.c
21>unlimit.c
21>yin.c
21>Generating Code...
21>Compiling manifest to resources...
21>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
21>Copyright (C) Microsoft Corporation.  All rights reserved.
21>Linking...
21>   Creating library .\../../lib/Debug/sphinxbase.lib and object .\../../lib/Debug/sphinxbase.exp
21>Embedding manifest...
21>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
21>Copyright (C) Microsoft Corporation.  All rights reserved.
22>Extracting: C:\FreeSWITCH\freeswitch-1.0.6\libs\flite-1.3.99-latest.tar.gz
21>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\sphinxbase\Debug\BuildLog.htm"
21>sphinxbase - 0 error(s), 0 warning(s)
23>------ Build started: Project: Download LAME, Configuration: Debug Win32 ------
23>Downloading Lame.
23>Downloading: http://files.freeswitch.org/downloads/libs/lame-3.97.tar.gz
23>Extracting: C:\FreeSWITCH\freeswitch-1.0.6\libs\lame-3.97.tar.gz
22>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
22>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\flite-1.3.99-latest.tar.gz
22>Extracting  flite-1.3.99-latest.tar
22>Everything is Ok
23>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
23>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\lame-3.97.tar.gz
23>Extracting  lame-3.97.tar
23>Everything is Ok
23>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
23>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\lame-3.97.tar
23>Extracting  lame-3.97
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22>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
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22>Extracting  flite-1.3.99\missing
22>Extracting  flite-1.3.99\configure.in
22>Extracting  flite-1.3.99\palm
22>Extracting  flite-1.3.99\palm\include
22>Extracting  flite-1.3.99\palm\include\elf_common.h
22>Extracting  flite-1.3.99\palm\include\elf.h
22>Extracting  flite-1.3.99\palm\include\pocore.h
22>Extracting  flite-1.3.99\palm\include\pealstub.h
22>Extracting  flite-1.3.99\palm\include\elf32.h
22>Extracting  flite-1.3.99\palm\include\peal.h
22>Extracting  flite-1.3.99\palm\include\palm_flite.h
22>Extracting  flite-1.3.99\palm\include\Makefile
22>Extracting  flite-1.3.99\palm\arm_flite
22>Extracting  flite-1.3.99\palm\arm_flite\make_seg_ro
22>Extracting  flite-1.3.99\palm\arm_flite\pealstub.c
22>Extracting  flite-1.3.99\palm\arm_flite\arm_flite.c
22>Extracting  flite-1.3.99\palm\arm_flite\Makefile
22>Extracting  flite-1.3.99\palm\pocore
22>Extracting  flite-1.3.99\palm\pocore\po_alloc.c
22>Extracting  flite-1.3.99\palm\pocore\po_StrVPrintF.c
22>Extracting  flite-1.3.99\palm\pocore\po_FileClose.c
22>Extracting  flite-1.3.99\palm\pocore\po_MemChunkFree.c
22>Extracting  flite-1.3.99\palm\pocore\po_sio.c
22>Extracting  flite-1.3.99\palm\pocore\po_FileSeek.c
22>Extracting  flite-1.3.99\palm\pocore\po_FileWrite.c
22>Extracting  flite-1.3.99\palm\pocore\po_setjmp.c
22>Extracting  flite-1.3.99\palm\pocore\po_atof.c
22>Extracting  flite-1.3.99\palm\pocore\po_MemPtrNew.c
22>Extracting  flite-1.3.99\palm\pocore\po_FileOpen.c
22>Extracting  flite-1.3.99\palm\pocore\po_FileTell.c
22>Extracting  flite-1.3.99\palm\pocore\po_core.c
22>Extracting  flite-1.3.99\palm\pocore\po_FileReadLow.c
22>Extracting  flite-1.3.99\palm\pocore\po_StrPrintF.c
22>Extracting  flite-1.3.99\palm\pocore\Makefile
22>Extracting  flite-1.3.99\palm\flop
22>Extracting  flite-1.3.99\palm\flop\flop.def
22>Extracting  flite-1.3.99\palm\flop\flop.rcp
22>Extracting  flite-1.3.99\palm\flop\flop.bmp
22>Extracting  flite-1.3.99\palm\flop\flop.c
22>Extracting  flite-1.3.99\palm\flop\flop.h
22>Extracting  flite-1.3.99\palm\flop\flopsmall.bmp
22>Extracting  flite-1.3.99\palm\flop\Makefile
22>Extracting  flite-1.3.99\palm\m68k_flite
22>Extracting  flite-1.3.99\palm\m68k_flite\m68k_flite.c
22>Extracting  flite-1.3.99\palm\m68k_flite\peal.c
22>Extracting  flite-1.3.99\palm\m68k_flite\fms.c
22>Extracting  flite-1.3.99\palm\m68k_flite\Makefile
22>Extracting  flite-1.3.99\palm\Makefile
22>Extracting  flite-1.3.99\README
22>Extracting  flite-1.3.99\doc
22>Extracting  flite-1.3.99\doc\stuff.ed
22>Extracting  flite-1.3.99\doc\flite.texi
22>Extracting  flite-1.3.99\doc\intro.txt
22>Extracting  flite-1.3.99\doc\alice
22>Extracting  flite-1.3.99\doc\Makefile
22>Extracting  flite-1.3.99\config.guess
22>Extracting  flite-1.3.99\tools
22>Extracting  flite-1.3.99\tools\play_sync.scm
22>Extracting  flite-1.3.99\tools\make_clunits.scm
22>Extracting  flite-1.3.99\tools\make_lts_rewrite.scm
22>Extracting  flite-1.3.99\tools\make_cg.scm
22>Extracting  flite-1.3.99\tools\make_cart.scm
22>Extracting  flite-1.3.99\tools\make_vallist.scm
22>Extracting  flite-1.3.99\tools\make_f0lr.scm
22>Extracting  flite-1.3.99\tools\build_flite
22>Extracting  flite-1.3.99\tools\flite_sort_main.c
22>Extracting  flite-1.3.99\tools\make_didb2.scm
22>Extracting  flite-1.3.99\tools\VOICE_diphone.c
22>Extracting  flite-1.3.99\tools\VOICE_cg.c
22>Extracting  flite-1.3.99\tools\make_voice_list
22>Extracting  flite-1.3.99\tools\find_sts_main.c
22>Extracting  flite-1.3.99\tools\make_lex.scm
22>Extracting  flite-1.3.99\tools\make_lts_wfst.scm
22>Extracting  flite-1.3.99\tools\VOICE_clunits.c
22>Extracting  flite-1.3.99\tools\flite_test
22>Extracting  flite-1.3.99\tools\VOICE_ldom.c
22>Extracting  flite-1.3.99\tools\make_didb.scm
22>Extracting  flite-1.3.99\tools\huff_table
22>Extracting  flite-1.3.99\tools\find_cmimax
22>Extracting  flite-1.3.99\tools\Makefile.flite
22>Extracting  flite-1.3.99\tools\make_lts.scm
22>Extracting  flite-1.3.99\tools\make_phoneset.scm
22>Extracting  flite-1.3.99\tools\setup_flite
22>Extracting  flite-1.3.99\tools\Makefile
22>Extracting  flite-1.3.99\Makefile
22>Everything is Ok
23>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download LAME.htm"
23>Download LAME - 0 error(s), 0 warning(s)
22>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download FLITE.htm"
22>Download FLITE - 0 error(s), 0 warning(s)
24>------ Build started: Project: Download LIBSHOUT, Configuration: Debug Win32 ------
24>Downloading Flite.
24>Downloading: http://files.freeswitch.org/downloads/libs/libshout-2.2.2.tar.gz
25>------ Build started: Project: js, Configuration: Debug Win32 ------
25>Performing Pre-Build Event...
25>Compiling...
25>e_acos.c
25>e_acosh.c
25>e_asin.c
25>e_atan2.c
25>e_atanh.c
25>e_cosh.c
25>e_exp.c
25>e_fmod.c
25>e_gamma.c
25>e_gamma_r.c
25>e_hypot.c
25>e_j0.c
25>e_j1.c
25>e_jn.c
24>Extracting: C:\FreeSWITCH\freeswitch-1.0.6\libs\libshout-2.2.2.tar.gz
25>e_lgamma.c
24>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
24>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\libshout-2.2.2.tar.gz
24>Extracting  libshout-2.2.2.tar
24>Everything is Ok
25>e_lgamma_r.c
25>e_log.c
25>e_log10.c
25>e_pow.c
25>e_rem_pio2.c
25>Generating Code...
25>Compiling...
25>e_remainder.c
25>e_scalb.c
25>e_sinh.c
25>e_sqrt.c
25>jsapi.c
24>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
24>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\libshout-2.2.2.tar
24>Extracting  libshout-2.2.2
24>Extracting  libshout-2.2.2\m4
24>Extracting  libshout-2.2.2\m4\vorbis.m4
24>Extracting  libshout-2.2.2\m4\ac_config_libconfig_in.m4
24>Extracting  libshout-2.2.2\m4\xiph_compiler.m4
24>Extracting  libshout-2.2.2\m4\acx_pthread.m4
24>Extracting  libshout-2.2.2\m4\ogg.m4
24>Extracting  libshout-2.2.2\m4\xiph_net.m4
24>Extracting  libshout-2.2.2\m4\speex.m4
24>Extracting  libshout-2.2.2\m4\shout.m4
24>Extracting  libshout-2.2.2\m4\xiph_types.m4
24>Extracting  libshout-2.2.2\m4\theora.m4
24>Extracting  libshout-2.2.2\ltmain.sh
24>Extracting  libshout-2.2.2\src
24>Extracting  libshout-2.2.2\src\Makefile.in
24>Extracting  libshout-2.2.2\src\shout_private.h
24>Extracting  libshout-2.2.2\src\Makefile.am
24>Extracting  libshout-2.2.2\src\shout_ogg.h
24>Extracting  libshout-2.2.2\src\mp3.c
24>Extracting  libshout-2.2.2\src\vorbis.c
24>Extracting  libshout-2.2.2\src\ogg.c
24>Extracting  libshout-2.2.2\src\theora.c
24>Extracting  libshout-2.2.2\src\speex.c
24>Extracting  libshout-2.2.2\src\thread
24>Extracting  libshout-2.2.2\src\thread\Makefile.in
24>Extracting  libshout-2.2.2\src\thread\Makefile.am
24>Extracting  libshout-2.2.2\src\thread\TODO
24>Extracting  libshout-2.2.2\src\thread\thread.c
24>Extracting  libshout-2.2.2\src\thread\BUILDING
24>Extracting  libshout-2.2.2\src\thread\COPYING
24>Extracting  libshout-2.2.2\src\thread\README
24>Extracting  libshout-2.2.2\src\thread\thread.h
24>Extracting  libshout-2.2.2\src\util.c
24>Extracting  libshout-2.2.2\src\shout.c
24>Extracting  libshout-2.2.2\src\net
24>Extracting  libshout-2.2.2\src\net\Makefile.in
24>Extracting  libshout-2.2.2\src\net\resolver.c
24>Extracting  libshout-2.2.2\src\net\Makefile.am
24>Extracting  libshout-2.2.2\src\net\TODO
24>Extracting  libshout-2.2.2\src\net\sock.c
24>Extracting  libshout-2.2.2\src\net\test_resolver.c
24>Extracting  libshout-2.2.2\src\net\resolver.h
24>Extracting  libshout-2.2.2\src\net\BUILDING
24>Extracting  libshout-2.2.2\src\net\COPYING
24>Extracting  libshout-2.2.2\src\net\README
24>Extracting  libshout-2.2.2\src\net\sock.h
24>Extracting  libshout-2.2.2\src\timing
24>Extracting  libshout-2.2.2\src\timing\timing.h
24>Extracting  libshout-2.2.2\src\timing\Makefile.in
24>Extracting  libshout-2.2.2\src\timing\Makefile.am
24>Extracting  libshout-2.2.2\src\timing\TODO
24>Extracting  libshout-2.2.2\src\timing\timing.c
24>Extracting  libshout-2.2.2\src\timing\BUILDING
24>Extracting  libshout-2.2.2\src\timing\COPYING
24>Extracting  libshout-2.2.2\src\timing\README
24>Extracting  libshout-2.2.2\src\util.h
24>Extracting  libshout-2.2.2\src\avl
24>Extracting  libshout-2.2.2\src\avl\Makefile.in
24>Extracting  libshout-2.2.2\src\avl\test.c
24>Extracting  libshout-2.2.2\src\avl\Makefile.am
24>Extracting  libshout-2.2.2\src\avl\TODO
24>Extracting  libshout-2.2.2\src\avl\avl.dsp
24>Extracting  libshout-2.2.2\src\avl\avl.c
24>Extracting  libshout-2.2.2\src\avl\avl.h
24>Extracting  libshout-2.2.2\src\avl\BUILDING
24>Extracting  libshout-2.2.2\src\avl\COPYING
24>Extracting  libshout-2.2.2\src\avl\README
24>Extracting  libshout-2.2.2\src\httpp
24>Extracting  libshout-2.2.2\src\httpp\Makefile.in
24>Extracting  libshout-2.2.2\src\httpp\Makefile.am
24>Extracting  libshout-2.2.2\src\httpp\httpp.h
24>Extracting  libshout-2.2.2\src\httpp\TODO
24>Extracting  libshout-2.2.2\src\httpp\httpp.c
24>Extracting  libshout-2.2.2\src\httpp\COPYING
24>Extracting  libshout-2.2.2\src\httpp\README
24>Extracting  libshout-2.2.2\examples
24>Extracting  libshout-2.2.2\examples\Makefile.in
24>Extracting  libshout-2.2.2\examples\Makefile.am
24>Extracting  libshout-2.2.2\examples\nonblocking.c
24>Extracting  libshout-2.2.2\examples\example.c
24>Extracting  libshout-2.2.2\Makefile.in
24>Extracting  libshout-2.2.2\compile
24>Extracting  libshout-2.2.2\debian
24>Extracting  libshout-2.2.2\debian\rules
24>Extracting  libshout-2.2.2\debian\Makefile.in
24>Extracting  libshout-2.2.2\debian\watch
24>Extracting  libshout-2.2.2\debian\libshout3-dev.examples
24>Extracting  libshout-2.2.2\debian\Makefile.am
24>Extracting  libshout-2.2.2\debian\libshout3.install
24>Extracting  libshout-2.2.2\debian\copyright
24>Extracting  libshout-2.2.2\debian\compat
24>Extracting  libshout-2.2.2\debian\control
24>Extracting  libshout-2.2.2\debian\changelog
24>Extracting  libshout-2.2.2\debian\libshout3-dev.install
24>Extracting  libshout-2.2.2\configure
24>Extracting  libshout-2.2.2\configure.ac
24>Extracting  libshout-2.2.2\Makefile.am
24>Extracting  libshout-2.2.2\aclocal.m4
24>Extracting  libshout-2.2.2\shout-config.in
24>Extracting  libshout-2.2.2\install-sh
24>Extracting  libshout-2.2.2\missing
24>Extracting  libshout-2.2.2\config.h.in
24>Extracting  libshout-2.2.2\NEWS
24>Extracting  libshout-2.2.2\config.guess
24>Extracting  libshout-2.2.2\config.sub
24>Extracting  libshout-2.2.2\doc
24>Extracting  libshout-2.2.2\doc\Makefile.in
24>Extracting  libshout-2.2.2\doc\Makefile.am
24>Extracting  libshout-2.2.2\doc\libshout.xml
24>Extracting  libshout-2.2.2\doc\spec-html.xsl
24>Extracting  libshout-2.2.2\shout.pc.in
24>Extracting  libshout-2.2.2\COPYING
24>Extracting  libshout-2.2.2\include
24>Extracting  libshout-2.2.2\include\Makefile.in
24>Extracting  libshout-2.2.2\include\os.h
24>Extracting  libshout-2.2.2\include\Makefile.am
24>Extracting  libshout-2.2.2\include\shout
24>Extracting  libshout-2.2.2\include\shout\Makefile.in
24>Extracting  libshout-2.2.2\include\shout\Makefile.am
24>Extracting  libshout-2.2.2\include\shout\shout.h.in
24>Extracting  libshout-2.2.2\README
24>Extracting  libshout-2.2.2\INSTALL
24>Extracting  libshout-2.2.2\win32
24>Extracting  libshout-2.2.2\win32\Makefile.in
24>Extracting  libshout-2.2.2\win32\Makefile.am
24>Extracting  libshout-2.2.2\win32\libshout.dsw
24>Extracting  libshout-2.2.2\win32\libshout.dsp
24>Extracting  libshout-2.2.2\depcomp
24>Everything is Ok
25>jsarena.c
25>jsarray.c
25>jsatom.c
25>jsbool.c
25>jscntxt.c
25>jsdate.c
25>jsdbgapi.c
25>jsdhash.c
24>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download LIBSHOUT.htm"
24>Download LIBSHOUT - 0 error(s), 0 warning(s)
25>jsdso.c
25>jsdtoa.c
26>------ Build started: Project: libogg, Configuration: Debug Win32 ------
26>Compiling...
26>framing.c
25>jsemit.c
26>bitwise.c
26>Generating Code...
26>Creating library...
26>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\libogg\Debug\BuildLog.htm"
26>libogg - 0 error(s), 0 warning(s)
25>jsexn.c
27>------ Build started: Project: Download mpg123, Configuration: Debug Win32 ------
27>Downloading Flite.
25>jsfile.c
27>Downloading: http://files.freeswitch.org/downloads/libs/mpg123.tar.gz
25>jsfun.c
25>jsgc.c
25>Generating Code...
27>Extracting: C:\FreeSWITCH\freeswitch-1.0.6\libs\mpg123.tar.gz
27>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
27>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\mpg123.tar.gz
27>Extracting  mpg123.tar
27>Everything is Ok
25>Compiling...
25>jshash.c
25>jsinterp.c
25>jslock.c
27>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
27>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\mpg123.tar
27>Extracting  mpg123
27>Extracting  mpg123\build
27>Extracting  mpg123\build\compile
27>Extracting  mpg123\build\config.guess
27>Extracting  mpg123\build\config.sub
27>Extracting  mpg123\build\depcomp
27>Extracting  mpg123\build\install-sh
27>Extracting  mpg123\build\ltmain.sh
27>Extracting  mpg123\build\missing
27>Extracting  mpg123\man1
27>Extracting  mpg123\man1\mpg123.1
27>Extracting  mpg123\ports
27>Extracting  mpg123\ports\MSVC++
27>Extracting  mpg123\ports\MSVC++\INCLUDE
27>Extracting  mpg123\ports\MSVC++\INCLUDE\CORE
27>Extracting  mpg123\ports\MSVC++\INCLUDE\CORE\CORE_FileIn.H
27>Extracting  mpg123\ports\MSVC++\INCLUDE\CORE\SourceFilter_MP3.H
27>Extracting  mpg123\ports\MSVC++\INCLUDE\IIEP_FileIn.H
27>Extracting  mpg123\ports\MSVC++\INCLUDE\IIEP_Def.H
27>Extracting  mpg123\ports\MSVC++\SOURCE
27>Extracting  mpg123\ports\MSVC++\SOURCE\CORE_Log.CPP
27>Extracting  mpg123\ports\MSVC++\SOURCE\CORE_FileIn.CPP
27>Extracting  mpg123\ports\MSVC++\SOURCE\SourceFilter_MP3Stream.CPP
27>Extracting  mpg123\ports\MSVC++\SOURCE\CORE_Mutex.CPP
27>Extracting  mpg123\ports\MSVC++\libMPG123
27>Extracting  mpg123\ports\MSVC++\libMPG123\libMPG123.vcproj
27>Extracting  mpg123\ports\MSVC++\libMPG123\PLACE_LIBMPG123_SOURCES_HERE
27>Extracting  mpg123\ports\MSVC++\README
27>Extracting  mpg123\ports\Sony_PSP
27>Extracting  mpg123\ports\Sony_PSP\config.h
27>Extracting  mpg123\ports\Sony_PSP\README
27>Extracting  mpg123\ports\Sony_PSP\Makefile.psp
27>Extracting  mpg123\ports\Sony_PSP\readers.c.patch
27>Extracting  mpg123\ports\README
27>Extracting  mpg123\ports\mpg123_.pas
27>Extracting  mpg123\src
27>Extracting  mpg123\src\libmpg123
27>Extracting  mpg123\src\libmpg123\compat.c
27>Extracting  mpg123\src\libmpg123\compat.h
27>Extracting  mpg123\src\libmpg123\Makefile.am
27>Extracting  mpg123\src\libmpg123\Makefile.in
27>Extracting  mpg123\src\libmpg123\mpg123.h.in
27>Extracting  mpg123\src\libmpg123\parse.c
27>Extracting  mpg123\src\libmpg123\parse.h
27>Extracting  mpg123\src\libmpg123\frame.c
27>Extracting  mpg123\src\libmpg123\format.c
27>Extracting  mpg123\src\libmpg123\frame.h
27>Extracting  mpg123\src\libmpg123\reader.h
27>Extracting  mpg123\src\libmpg123\debug.h
27>Extracting  mpg123\src\libmpg123\decode.h
27>Extracting  mpg123\src\libmpg123\decode_2to1.c
27>Extracting  mpg123\src\libmpg123\decode_4to1.c
27>Extracting  mpg123\src\libmpg123\decode_ntom.c
27>Extracting  mpg123\src\libmpg123\equalizer.c
27>Extracting  mpg123\src\libmpg123\huffman.h
27>Extracting  mpg123\src\libmpg123\icy.c
27>Extracting  mpg123\src\libmpg123\icy.h
27>Extracting  mpg123\src\libmpg123\icy2utf8.c
27>Extracting  mpg123\src\libmpg123\icy2utf8.h
27>Extracting  mpg123\src\libmpg123\id3.c
27>Extracting  mpg123\src\libmpg123\id3.h
27>Extracting  mpg123\src\libmpg123\true.h
27>Extracting  mpg123\src\libmpg123\l2tables.h
27>Extracting  mpg123\src\libmpg123\layer1.c
27>Extracting  mpg123\src\libmpg123\layer2.c
27>Extracting  mpg123\src\libmpg123\layer3.c
27>Extracting  mpg123\src\libmpg123\getbits.h
27>Extracting  mpg123\src\libmpg123\optimize.h
27>Extracting  mpg123\src\libmpg123\optimize.c
27>Extracting  mpg123\src\libmpg123\readers.c
27>Extracting  mpg123\src\libmpg123\tabinit.c
27>Extracting  mpg123\src\libmpg123\stringbuf.c
27>Extracting  mpg123\src\libmpg123\libmpg123.c
27>Extracting  mpg123\src\libmpg123\mpg123lib_intern.h
27>Extracting  mpg123\src\libmpg123\mangle.h
27>Extracting  mpg123\src\libmpg123\getcpuflags.h
27>Extracting  mpg123\src\libmpg123\libmpg123.sym
27>Extracting  mpg123\src\libmpg123\dct36_3dnowext.S
27>Extracting  mpg123\src\libmpg123\dct36_3dnow.S
27>Extracting  mpg123\src\libmpg123\dct64_3dnowext.S
27>Extracting  mpg123\src\libmpg123\dct64_3dnow.S
27>Extracting  mpg123\src\libmpg123\dct64_altivec.c
27>Extracting  mpg123\src\libmpg123\dct64.c
27>Extracting  mpg123\src\libmpg123\dct64_i386.c
27>Extracting  mpg123\src\libmpg123\dct64_i486.c
27>Extracting  mpg123\src\libmpg123\dct64_mmx.S
27>Extracting  mpg123\src\libmpg123\dct64_sse.S
27>Extracting  mpg123\src\libmpg123\decode_3dnowext.S
27>Extracting  mpg123\src\libmpg123\decode_3dnow.S
27>Extracting  mpg123\src\libmpg123\decode_altivec.c
27>Extracting  mpg123\src\libmpg123\decode.c
25>jslog2.c
25>jslong.c
25>jsmath.c
25>jsnum.c
25>jsobj.c
27>Extracting  mpg123\src\libmpg123\decode_i386.c
27>Extracting  mpg123\src\libmpg123\decode_i486.c
27>Extracting  mpg123\src\libmpg123\decode_i586_dither.S
27>Extracting  mpg123\src\libmpg123\decode_i586.S
27>Extracting  mpg123\src\libmpg123\decode_mmx.S
27>Extracting  mpg123\src\libmpg123\decode_sse3d.h
27>Extracting  mpg123\src\libmpg123\decode_sse.S
27>Extracting  mpg123\src\libmpg123\equalizer_3dnow.S
27>Extracting  mpg123\src\libmpg123\tabinit_mmx.S
27>Extracting  mpg123\src\libmpg123\getcpuflags.S
27>Extracting  mpg123\src\libmpg123\testcpu.c
27>Extracting  mpg123\src\libmpg123\dnoise.sh
27>Extracting  mpg123\src\libmpg123\dnoise.dat
27>Extracting  mpg123\src\Makefile.am
27>Extracting  mpg123\src\Makefile.in
27>Extracting  mpg123\src\config.h.in
27>Extracting  mpg123\src\audio.c
27>Extracting  mpg123\src\audio.h
27>Extracting  mpg123\src\buffer.c
27>Extracting  mpg123\src\buffer.h
27>Extracting  mpg123\src\common.c
27>Extracting  mpg123\src\common.h
27>Extracting  mpg123\src\control_generic.c
27>Extracting  mpg123\src\getlopt.c
27>Extracting  mpg123\src\getlopt.h
27>Extracting  mpg123\src\httpget.c
27>Extracting  mpg123\src\httpget.h
27>Extracting  mpg123\src\resolver.c
27>Extracting  mpg123\src\resolver.h
27>Extracting  mpg123\src\genre.h
27>Extracting  mpg123\src\genre.c
27>Extracting  mpg123\src\module.h
27>Extracting  mpg123\src\mpg123.c
27>Extracting  mpg123\src\mpg123app.h
27>Extracting  mpg123\src\metaprint.c
27>Extracting  mpg123\src\metaprint.h
27>Extracting  mpg123\src\playlist.c
27>Extracting  mpg123\src\playlist.h
27>Extracting  mpg123\src\sfifo.c
27>Extracting  mpg123\src\sfifo.h
27>Extracting  mpg123\src\term.c
27>Extracting  mpg123\src\term.h
27>Extracting  mpg123\src\wav.c
27>Extracting  mpg123\src\xfermem.c
27>Extracting  mpg123\src\xfermem.h
27>Extracting  mpg123\src\Makefile.legacy
27>Extracting  mpg123\src\config.h.legacy
27>Extracting  mpg123\src\legacy_module.c
27>Extracting  mpg123\src\module.c
27>Extracting  mpg123\src\output
27>Extracting  mpg123\src\output\Makefile.am
27>Extracting  mpg123\src\output\Makefile.in
27>Extracting  mpg123\src\output\aix.c
27>Extracting  mpg123\src\output\alib.c
27>Extracting  mpg123\src\output\alsa.c
27>Extracting  mpg123\src\output\arts.c
27>Extracting  mpg123\src\output\coreaudio.c
27>Extracting  mpg123\src\output\dummy.c
27>Extracting  mpg123\src\output\esd.c
27>Extracting  mpg123\src\output\hp.c
27>Extracting  mpg123\src\output\jack.c
27>Extracting  mpg123\src\output\mint.c
27>Extracting  mpg123\src\output\nas.c
27>Extracting  mpg123\src\output\os2.c
27>Extracting  mpg123\src\output\oss.c
27>Extracting  mpg123\src\output\portaudio.c
27>Extracting  mpg123\src\output\pulse.c
27>Extracting  mpg123\src\output\sdl.c
27>Extracting  mpg123\src\output\sgi.c
27>Extracting  mpg123\src\output\sun.c
27>Extracting  mpg123\src\output\win32.c
27>Extracting  mpg123\test
27>Extracting  mpg123\test\forkfaint.c
27>Extracting  mpg123\test\rms16.c
27>Extracting  mpg123\xmms2-plugin
27>Extracting  mpg123\xmms2-plugin\mpg123
27>Extracting  mpg123\xmms2-plugin\mpg123\mpg123.c
27>Extracting  mpg123\xmms2-plugin\mpg123\wscript
27>Extracting  mpg123\xmms2-plugin\README
27>Extracting  mpg123\README
27>Extracting  mpg123\configure.ac
27>Extracting  mpg123\aclocal.m4
27>Extracting  mpg123\Makefile.am
27>Extracting  mpg123\Makefile.in
27>Extracting  mpg123\libmpg123.pc.in
27>Extracting  mpg123\configure
27>Extracting  mpg123\AUTHORS
27>Extracting  mpg123\COPYING
27>Extracting  mpg123\ChangeLog
27>Extracting  mpg123\INSTALL
27>Extracting  mpg123\NEWS
27>Extracting  mpg123\TODO
27>Extracting  mpg123\MakeLegacy.sh
27>Extracting  mpg123\mpg123.spec.in
27>Extracting  mpg123\mpg123.spec
27>Extracting  mpg123\makedll.sh
27>Extracting  mpg123\NEWS.libmpg123
27>Extracting  mpg123\autogen.sh
27>Extracting  mpg123\libltdl
27>Extracting  mpg123\libltdl\README
27>Extracting  mpg123\libltdl\acinclude.m4
27>Extracting  mpg123\libltdl\configure.ac
27>Extracting  mpg123\libltdl\aclocal.m4
27>Extracting  mpg123\libltdl\ltdl.h
27>Extracting  mpg123\libltdl\Makefile.am
27>Extracting  mpg123\libltdl\Makefile.in
27>Extracting  mpg123\libltdl\config-h.in
27>Extracting  mpg123\libltdl\configure
27>Extracting  mpg123\libltdl\COPYING.LIB
27>Extracting  mpg123\libltdl\config.guess
27>Extracting  mpg123\libltdl\config.sub
27>Extracting  mpg123\libltdl\install-sh
27>Extracting  mpg123\libltdl\ltmain.sh
27>Extracting  mpg123\libltdl\missing
27>Extracting  mpg123\libltdl\ltdl.c
27>Extracting  mpg123\doc
27>Extracting  mpg123\doc\examples
27>Extracting  mpg123\doc\examples\mpg123_to_wav.c
27>Extracting  mpg123\doc\examples\scan.c
27>Extracting  mpg123\doc\examples\mpglib.c
27>Extracting  mpg123\doc\examples\id3dump.c
27>Extracting  mpg123\doc\examples\Makefile
27>Extracting  mpg123\doc\Makefile.am
27>Extracting  mpg123\doc\Makefile.in
27>Extracting  mpg123\doc\THANKS
27>Extracting  mpg123\doc\TODO
27>Extracting  mpg123\doc\BENCHMARKING
27>Extracting  mpg123\doc\BUGS
27>Extracting  mpg123\doc\CONTACT
27>Extracting  mpg123\doc\PATENTS
27>Extracting  mpg123\doc\README.3DNOW
27>Extracting  mpg123\doc\README.WIN32
27>Extracting  mpg123\doc\README.gain
27>Extracting  mpg123\doc\README.remote
27>Extracting  mpg123\doc\ROAD_TO_LGPL
27>Extracting  mpg123\doc\LICENSE
27>Extracting  mpg123\doc\ACCURACY
27>Extracting  mpg123\doc\libmpg123_speed.txt
27>Extracting  mpg123\doc\doxyhead.xhtml
27>Extracting  mpg123\doc\doxy_examples.c
27>Extracting  mpg123\doc\doxygen.conf
27>Everything is Ok
25>jsopcode.c
25>jsparse.c
25>jsprf.c
25>jsregexp.c
25>jsscan.c
25>jsscope.c
25>jsscript.c
25>jsstr.c
25>jsutil.c
27>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download mpg123.htm"
27>Download mpg123 - 0 error(s), 0 warning(s)
28>------ Build started: Project: libmpg123, Configuration: Debug Win32 ------
28>Compiling...
28>dct64.c
25>jsxdrapi.c
28>decode.c
25>jsxml.c
25>k_cos.c
28>decode_2to1.c
25>Generating Code...
28>decode_4to1.c
28>decode_ntom.c
28>equalizer.c
28>format.c
28>frame.c
25>Compiling...
25>k_rem_pio2.c
25>k_sin.c
25>k_standard.c
28>icy.c
25>k_tan.c
28>icy2utf8.c
25>prmjtime.c
28>id3.c
28>layer1.c
25>s_asinh.c
25>s_atan.c
25>s_cbrt.c
28>layer2.c
25>s_ceil.c
25>s_copysign.c
25>s_cos.c
28>layer3.c
25>s_erf.c
25>s_expm1.c
25>s_fabs.c
25>s_finite.c
28>libmpg123.c
25>s_floor.c
25>s_frexp.c
25>s_ilogb.c
25>s_isnan.c
25>s_ldexp.c
28>optimize.c
25>Generating Code...
28>parse.c
25>Compiling...
25>s_lib_version.c
25>s_log1p.c
25>s_logb.c
25>s_matherr.c
25>s_modf.c
25>s_nextafter.c
25>s_rint.c
25>s_scalbn.c
25>s_signgam.c
25>s_significand.c
25>s_sin.c
28>readers.c
25>s_tan.c
25>s_tanh.c
25>w_acos.c
28>stringbuf.c
25>w_acosh.c
25>w_asin.c
25>w_atan2.c
28>tabinit.c
25>w_atanh.c
25>w_cosh.c
25>w_exp.c
28>Generating Code...
25>Generating Code...
28>Compiling...
28>compat.c
25>Compiling...
25>w_fmod.c
28>Generating Code...
25>w_gamma.c
25>w_gamma_r.c
28>Creating library...
25>w_hypot.c
25>w_j0.c
25>w_j1.c
25>w_jn.c
25>w_lgamma.c
25>w_lgamma_r.c
25>w_log.c
25>w_log10.c
25>w_pow.c
25>w_remainder.c
25>w_scalb.c
25>w_sinh.c
25>w_sqrt.c
25>ntinrval.c
28>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\mpg123\Debug\BuildLog.htm"
28>libmpg123 - 0 error(s), 0 warning(s)
29>------ Build started: Project: libdingaling, Configuration: Debug Win32 ------
29>Compiling...
29>sha1.c
25>ntio.c
29>libdingaling.c
25>ntmisc.c
29>Generating Code...
29>Creating library...
29>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\libdingaling\Debug\BuildLog.htm"
29>libdingaling - 0 error(s), 0 warning(s)
30>------ Skipped Build: Project: Download 16khz music, Configuration: Debug Win32 ------
30>Project not selected to build for this solution configuration
31>------ Build started: Project: Download 8khz music, Configuration: Debug Win32 ------
31>Downloading 8khzsound.
25>ntsec.c
31>Downloading: http://files.freeswitch.org/freeswitch-sounds-music-8000-1.0.8.tar.gz
25>Generating Code...
25>Compiling...
25>ntthread.c
25>pratom.c
25>prcthr.c
25>prdir.c
25>prerror.c
25>prfdcach.c
25>prfile.c
25>prinit.c
25>prinrval.c
31>Extracting: C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\freeswitch-sounds-music-8000-1.0.8.tar.gz
25>prio.c
25>priometh.c
31>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
31>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\freeswitch-sounds-music-8000-1.0.8.tar.gz
31>Extracting  freeswitch-sounds-music-8000-1.0.8.tar
31>Everything is Ok
31>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
31>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\freeswitch-sounds-music-8000-1.0.8.tar
31>Extracting  music\8000
31>Extracting  music\8000\partita-no-3-in-e-major-bwv-1006-1-preludio.wav
31>Extracting  music\8000\ponce-preludio-in-e-major.wav
31>Extracting  music\8000\suite-espanola-op-47-leyenda.wav
31>Extracting  music\8000\danza-espanola-op-37-h-142-xii-arabesca.wav
31>Everything is Ok
25>prlayer.c
25>prlog.c
25>prmem.c
31>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download 8khz music.htm"
31>Download 8khz music - 0 error(s), 0 warning(s)
32>------ Build started: Project: libsndfile, Configuration: Debug Win32 ------
32>Compiling...
25>prmmap.c
32>alaw.c
32>au.c
25>prmwait.c
32>audio_detect.c
32>avr.c
32>broadcast.c
25>prolock.c
25>prosdep.c
32>caf.c
32>chunk.c
25>prprf.c
32>command.c
32>common.c
32>dither.c
25>prseg.c
32>double64.c
32>dwd.c
25>Generating Code...
32>dwvw.c
25>Compiling...
25>prtime.c
32>file_io.c
25>prtpd.c
32>flac.c
25>prucpu.c
32>float32.c
25>prucv.c
32>gsm610.c
32>htk.c
32>ima_adpcm.c
25>prulock.c
32>ima_oki_adpcm.c
32>Generating Code...
25>prustack.c
32>Compiling...
32>ircam.c
25>pruthr.c
32>mat4.c
32>mat5.c
25>w32poll.c
32>mpc2k.c
32>ms_adpcm.c
32>nist.c
25>win32_errors.c
32>ogg.c
25>Generating Code...
32>paf.c
25>Linking...
32>pcm.c
25>   Creating library .\Debug/js32.lib and object .\Debug/js32.exp
32>pvf.c
32>raw.c
32>rf64.c
32>rx2.c
32>sd2.c
25>Embedding manifest...
32>sds.c
32>sndfile.c
32>strings.c
32>svx.c
25>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\js\Debug\BuildLog.htm"
25>js - 0 error(s), 0 warning(s)
33>------ Skipped Build: Project: Download 32khz music, Configuration: Debug Win32 ------
33>Project not selected to build for this solution configuration
34>------ Build started: Project: lua51, Configuration: Debug Win32 ------
32>txw.c
34>Compiling...
34>lzio.c
32>ulaw.c
32>Generating Code...
34>lvm.c
32>Compiling...
32>voc.c
34>lundump.c
34>ltm.c
32>vox_adpcm.c
34>ltablib.c
32>w64.c
34>ltable.c
32>wav.c
34>lstrlib.c
34>lstring.c
32>wav_w64.c
34>lstate.c
32>wve.c
34>lparser.c
32>xi.c
32>add.c
34>loslib.c
32>code.c
34>lopcodes.c
32>decode.c
34>lobject.c
32>gsm_create.c
32>gsm_decode.c
34>loadlib.c
32>gsm_destroy.c
32>gsm_encode.c
32>gsm_option.c
32>long_term.c
32>lpc.c
32>preprocess.c
32>rpe.c
32>short_term.c
34>lmem.c
32>Generating Code...
34>lmathlib.c
34>llex.c
32>Compiling...
34>liolib.c
32>table.c
32>g721.c
34>linit.c
34>lgc.c
32>g723_16.c
32>g723_24.c
32>g723_40.c
32>g72x.c
32>aiff.c
34>Generating Code...
32>Generating Code...
34>Compiling...
34>lfunc.c
32>Compiling...
34>ldump.c
32>g72x.c
34>ldo.c
32>Creating library...
34>ldebug.c
32>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\libsndfile\Debug\BuildLog.htm"
32>libsndfile - 0 error(s), 0 warning(s)
34>ldblib.c
34>lcode.c
34>lbaselib.c
34>lauxlib.c
34>lapi.c
35>------ Build started: Project: libshout, Configuration: Debug Win32 ------
34>Generating Code...
35>Compiling...
35>httpp.c
34>Linking...
35>mp3.c
34>   Creating library .\Debug/lua5.1.lib and object .\Debug/lua5.1.exp
35>ogg.c
34>Embedding manifest...
35>resolver.c
34>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_lua\lua\Debug\BuildLog.htm"
34>lua51 - 0 error(s), 0 warning(s)
36>------ Build started: Project: libmp3lame, Configuration: Debug Win32 ------
36>Compiling...
36>bitstream.c
35>shout.c
36>version.c
35>sock.c
36>VbrTag.c
36>vbrquantize.c
35>thread.c
36>util.c
35>timing.c
36>takehiro.c
35>util.c
36>set_get.c
35>avl.c
36>reservoir.c
35>Generating Code...
35>Creating library...
36>quantize_pvt.c
35>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\libshout\Debug\BuildLog.htm"
35>libshout - 0 error(s), 0 warning(s)
36>quantize.c
36>psymodel.c
36>presets.c
37>------ Build started: Project: libg722_1, Configuration: Debug Win32 ------
37>Compiling...
37>bitstream.c
36>newmdct.c
36>mpglib_interface.c
36>lame.c
36>id3tag.c
37>coef2sam.c
36>gain_analysis.c
36>fft.c
37>common.c
36>encoder.c
36>Generating Code...
37>commonf.c
36>Compiling...
36>tables.c
37>dct4.c
36>Creating library...
36>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\libmp3lame\Debug\BuildLog.htm"
36>libmp3lame - 0 error(s), 0 warning(s)
38>------ Build started: Project: libsofia_sip_ua_static, Configuration: Debug Win32 ------
38>Performing Pre-Build Event...
37>dct4_a.c
37>dct4_s.c
38>Downloading: http://files.freeswitch.org/downloads/win32/gawk.exe
37>decoder.c
37>decoderf.c
38>multipart mismatch with Recursive multipart ()
37>encoder.c
37>encoderf.c
38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
38>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
37>huff_tab.c
38>NOTE:
38>NOTE: Remember to install pthreadVC2.dll to your path, too!
38>NOTE:
38>Compiling...
38>inet_pton.c
37>sam2coef.c
38>smoothsort.c
38>string0.c
38>su.c
37>tables.c
38>su_addrinfo.c
37>basop32.c
37>Generating Code...
38>su_alloc.c
37>Creating library...
38>su_alloc_lock.c
37>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\libg722_1\Debug\BuildLog.htm"
37>libg722_1 - 0 error(s), 0 warning(s)
39>------ Build started: Project: esl, Configuration: Debug Win32 ------
38>su_base_port.c
39>Compiling...
39>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
39>esl_config.c
39>esl_event.c
38>su_bm.c
39>esl_threadmutex.c
38>su_default_log.c
38>su_errno.c
39>esl.c
39>Generating Code...
39>Creating library...
38>su_global_log.c
39>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\esl\src\Debug\BuildLog.htm"
38>su_localinfo.c
39>esl - 0 error(s), 1 warning(s)
38>su_log.c
38>su_md5.c
38>su_os_nw.c
40>------ Build started: Project: libilbc, Configuration: Debug Win32 ------
40>Compiling...
40>constants.c
38>su_port.c
38>su_pthread_port.c
40>createCB.c
38>su_root.c
40>doCPLC.c
38>su_socket_port.c
40>enhancer.c
38>Generating Code...
40>filter.c
38>Compiling...
38>su_sprintf.c
38>su_strdup.c
38>su_string.c
40>FrameClassify.c
38>su_strlst.c
38>su_tag.c
38>su_tag_io.c
40>gainquant.c
38>su_taglist.c
40>getCBvec.c
38>su_time.c
40>helpfun.c
38>su_time0.c
40>hpInput.c
38>su_timer.c
40>hpOutput.c
38>su_uniqueid.c
40>iCBConstruct.c
38>su_vector.c
38>su_wait.c
40>iCBSearch.c
38>su_win32_port.c
40>iLBC_decode.c
38>base64.c
38>rc4.c
38>token64.c
38>url.c
40>iLBC_encode.c
38>url_tag.c
40>LPCdecode.c
38>url_tag_ref.c
38>Generating Code...
40>LPCencode.c
38>Compiling...
40>lsf.c
38>features.c
40>packing.c
38>bnf.c
38>msg.c
40>StateConstructW.c
38>msg_auth.c
38>msg_basic.c
40>Generating Code...
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065: 'msg_error_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065: 'msg_error_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065: 'msg_error_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2065: 'msg_unknown_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2065: 'msg_unknown_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2065: 'msg_unknown_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : error C2065: 'msg_payload_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : error C2065: 'msg_separator_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>msg_date.c
40>Compiling...
40>StateSearchW.c
38>msg_generic.c
38>msg_header_copy.c
40>syntFilter.c
38>msg_header_make.c
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_header_make.c(71) : error C2065: 'msg_payload_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_header_make.c(72) : error C2065: 'msg_separator_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_header_make.c(73) : error C2065: 'msg_error_hash' : undeclared identifier
40>anaFilter.c
38>msg_mclass.c
40>Generating Code...
40>Creating library...
38>msg_mime.c
40>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\ilbc\Debug\BuildLog.htm"
40>libilbc - 0 error(s), 0 warning(s)
41>------ Build started: Project: libudns, Configuration: Debug Win32 ------
41>Compiling...
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(246) : error C2065: 'msg_multipart_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(246) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(246) : warning C4024: 'msg_header_alloc' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(250) : warning C4013: 'msg_content_type_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(250) : warning C4047: '=' : 'msg_content_type_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(387) : error C2065: 'msg_multipart_mclass' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(387) : warning C4047: '=' : 'const msg_mclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(435) : error C2065: 'msg_multipart_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(435) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(435) : warning C4024: 'msg_header_alloc' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(454) : error C2065: 'msg_payload_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(454) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
41>udns_bl.c
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(454) : warning C4024: 'msg_header_alloc' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(635) : warning C4013: 'msg_payload_format' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(636) : warning C4047: '=' : 'msg_payload_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(655) : warning C4013: 'msg_separator_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(655) : warning C4047: '=' : 'msg_separator_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(830) : warning C4013: 'msg_is_multipart' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(832) : warning C4013: 'msg_payload_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2065: 'msg_accept_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2065: 'msg_accept_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2065: 'msg_accept_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2065: 'msg_accept_dup_xtra' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2065: 'msg_accept_dup_one' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2065: 'msg_accept_update' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'char [7]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'size_t'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1062) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1095) : warning C4013: 'msg_is_accept' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : error C2065: 'msg_accept_charset_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : error C2065: 'msg_accept_charset_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : error C2065: 'msg_accept_charset_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [15]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1278) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1287) : warning C4013: 'msg_is_accept_charset' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : error C2065: 'msg_accept_encoding_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : error C2065: 'msg_accept_encoding_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : error C2065: 'msg_accept_encoding_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [16]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1329) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : error C2065: 'msg_accept_language_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : error C2065: 'msg_accept_language_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : error C2065: 'msg_accept_language_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [16]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1380) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1389) : warning C4013: 'msg_is_accept_language' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2065: 'msg_content_disposition_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2065: 'msg_content_disposition_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2065: 'msg_content_disposition_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2065: 'msg_content_disposition_dup_xtra' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2065: 'msg_content_disposition_dup_one' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2065: 'msg_content_disposition_update' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'char [20]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'size_t'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1456) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1477) : warning C4013: 'msg_is_content_disposition' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : error C2065: 'msg_content_encoding_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : error C2065: 'msg_content_encoding_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : error C2065: 'msg_content_encoding_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [17]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1567) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1577) : warning C4013: 'msg_is_content_encoding' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : error C2065: 'msg_content_language_hash' : undeclared identifier
41>udns_codes.c
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : error C2099: initializer is not a constant
41>udns_dn.c
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : error C2065: 'msg_content_language_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : error C2065: 'msg_content_language_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [17]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1622) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1632) : warning C4013: 'msg_is_content_language' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : error C2065: 'msg_content_length_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [15]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1672) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [2]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : error C2065: 'msg_content_md5_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [12]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1736) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : error C2065: 'msg_content_id_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [11]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1773) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2065: 'msg_content_type_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2065: 'msg_content_type_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2065: 'msg_content_type_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2065: 'msg_content_type_dup_xtra' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2065: 'msg_content_type_dup_one' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'void *'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char [13]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [2]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'size_t'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1817) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1844) : warning C4013: 'msg_is_content_type' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : error C2065: 'msg_mime_version_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : error C2065: 'msg_mime_version_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : error C2065: 'msg_mime_version_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [13]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1915) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1924) : warning C4013: 'msg_is_mime_version' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : error C2065: 'msg_content_location_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [17]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1956) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : error C2065: 'msg_content_transfer_encoding_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [26]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2023) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : error C2065: 'msg_warning_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [8]'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2147) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>msg_mime_table.c
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime_table.c(6) : error C2143: syntax error : missing '{' before '<cv-qualifer>'
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime_table.c(6) : error C2059: syntax error : '<Unknown>'
41>udns_dntosp.c
38>msg_parser.c
41>udns_misc.c
41>udns_parse.c
41>udns_resolver.c
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_parser.c(127) : error C2065: 'msg_request_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_parser.c(132) : error C2065: 'msg_status_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_parser.c(1051) : error C2065: 'msg_unknown_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\msg\msg_parser.c(2151) : warning C4013: 'msg_is_payload' undefined; assuming extern returning int
38>msg_parser_util.c
41>udns_rr_a.c
41>udns_rr_mx.c
41>udns_rr_naptr.c
41>udns_rr_ptr.c
41>udns_rr_srv.c
41>udns_rr_txt.c
38>msg_tag.c
41>inet_pton.c
41>Generating Code...
41>Creating library...
38>memcspn.c
38>memmem.c
41>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\udns\Debug\BuildLog.htm"
38>memspn.c
41>libudns - 0 error(s), 0 warning(s)
38>strcasestr.c
38>strtoull.c
38>Generating Code...
42>------ Build started: Project: flite, Configuration: Debug Win32 ------
42>Compiling...
42>au_none.c
38>Compiling...
38>sip_basic.c
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2065: 'sip_request_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2065: 'sip_request_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2065: 'sip_request_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2065: 'sip_status_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2065: 'sip_status_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2065: 'sip_status_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(321) : warning C4013: 'sip_is_status' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : error C2065: 'sip_payload_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(459) : warning C4013: 'sip_header_data' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(459) : warning C4047: 'initializing' : 'char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : error C2065: 'sip_separator_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2065: 'sip_unknown_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2065: 'sip_unknown_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2065: 'sip_unknown_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(564) : warning C4013: 'msg_unknown_d' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(569) : warning C4013: 'msg_unknown_e' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2065: 'sip_error_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2065: 'sip_error_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2065: 'sip_error_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(962) : warning C4047: 'initializing' : 'char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2065: 'sip_call_id_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2065: 'sip_call_id_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2065: 'sip_call_id_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [8]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2065: 'sip_cseq_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2065: 'sip_cseq_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2065: 'sip_cseq_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [5]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1224) : warning C4013: 'sip_is_cseq' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1374) : error C2065: 'sip_contact_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1374) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1375) : error C2065: 'sip_contact_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1375) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1376) : error C2065: 'sip_contact_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1376) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1377) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1378) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1380) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [8]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1381) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1382) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1383) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1414) : warning C4013: 'sip_is_contact' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2065: 'sip_content_length_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2065: 'sip_content_length_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2065: 'sip_content_length_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [15]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1546) : warning C4013: 'sip_is_content_length' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2065: 'sip_date_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2065: 'sip_date_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2065: 'sip_date_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [5]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2065: 'sip_expires_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2065: 'sip_expires_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2065: 'sip_expires_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [8]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2065: 'sip_from_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2065: 'sip_from_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2065: 'sip_from_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [5]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1798) : warning C4013: 'sip_is_from' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2065: 'sip_max_forwards_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2065: 'sip_max_forwards_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2065: 'sip_max_forwards_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [13]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1909) : warning C4013: 'sip_is_max_forwards' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2065: 'sip_min_expires_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2065: 'sip_min_expires_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2065: 'sip_min_expires_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [12]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1955) : warning C4013: 'sip_is_min_expires' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2065: 'sip_retry_after_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2065: 'sip_retry_after_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2065: 'sip_retry_after_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [12]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2196) : warning C4047: '=' : 'char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2065: 'sip_route_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2065: 'sip_route_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2065: 'sip_route_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [6]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2264) : warning C4013: 'sip_is_route' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2065: 'sip_record_route_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2065: 'sip_record_route_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2065: 'sip_record_route_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [13]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2338) : warning C4013: 'sip_is_record_route' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2401) : error C2065: 'sip_to_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2401) : fatal error C1003: error count exceeds 100; stopping compilation
38>sip_caller_prefs.c
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error C2065: 'sip_request_disposition_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error C2065: 'sip_request_disposition_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error C2065: 'sip_request_disposition_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [20]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(116) : warning C4013: 'sip_is_request_disposition' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error C2065: 'sip_accept_contact_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error C2065: 'sip_accept_contact_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error C2065: 'sip_accept_contact_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [15]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error C2065: 'sip_reject_contact_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error C2065: 'sip_reject_contact_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error C2065: 'sip_reject_contact_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [15]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>sip_event.c
42>au_streaming.c
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2065: 'sip_event_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2065: 'sip_event_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2065: 'sip_event_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [6]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(120) : warning C4013: 'sip_is_event' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2065: 'sip_allow_events_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2065: 'sip_allow_events_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2065: 'sip_allow_events_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [13]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(219) : warning C4013: 'sip_is_allow_events' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2065: 'sip_subscription_state_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2065: 'sip_subscription_state_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2065: 'sip_subscription_state_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [19]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(331) : warning C4013: 'sip_is_subscription_state' undefined; assuming extern returning int
38>sip_extra.c
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_extra.c(44) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory
38>sip_feature.c
42>au_wince.c
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2065: 'sip_allow_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2065: 'sip_allow_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2065: 'sip_allow_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [6]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(105) : warning C4013: 'sip_is_allow' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2065: 'sip_proxy_require_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2065: 'sip_proxy_require_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2065: 'sip_proxy_require_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [14]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(192) : warning C4013: 'sip_is_proxy_require' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2065: 'sip_require_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2065: 'sip_require_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2065: 'sip_require_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [8]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(240) : warning C4013: 'sip_is_require' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2065: 'sip_supported_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2065: 'sip_supported_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2065: 'sip_supported_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [10]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(289) : warning C4013: 'sip_is_supported' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2065: 'sip_unsupported_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2065: 'sip_unsupported_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2065: 'sip_unsupported_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [12]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(336) : warning C4013: 'sip_is_unsupported' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(450) : warning C4013: 'sip_unsupported_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(450) : warning C4047: '=' : 'sip_unsupported_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2065: 'sip_path_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2065: 'sip_path_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2065: 'sip_path_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [5]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(538) : warning C4013: 'sip_is_path' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2065: 'sip_service_route_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2065: 'sip_service_route_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2065: 'sip_service_route_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [14]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(594) : warning C4013: 'sip_is_service_route' undefined; assuming extern returning int
38>sip_header.c
38>sip_mime.c
42>audio.c
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065: 'sip_accept_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065: 'sip_accept_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065: 'sip_accept_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065: 'msg_accept_dup_xtra' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065: 'msg_accept_dup_one' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065: 'msg_accept_update' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'char [7]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'size_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(105) : warning C4013: 'msg_accept_d' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(110) : warning C4013: 'msg_accept_e' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2065: 'sip_accept_encoding_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2065: 'sip_accept_encoding_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2065: 'sip_accept_encoding_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2065: 'msg_accept_any_update' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'char [16]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'size_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(220) : warning C4013: 'msg_accept_encoding_d' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(234) : warning C4013: 'msg_accept_encoding_e' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2065: 'sip_accept_language_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2065: 'sip_accept_language_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2065: 'sip_accept_language_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2065: 'msg_accept_any_update' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'char [16]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'size_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(286) : warning C4013: 'msg_accept_language_d' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(299) : warning C4013: 'msg_accept_language_e' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2065: 'sip_content_disposition_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2065: 'sip_content_disposition_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2065: 'sip_content_disposition_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2065: 'msg_content_disposition_update' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'char [20]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'size_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(376) : warning C4013: 'msg_content_disposition_d' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(382) : warning C4013: 'msg_content_disposition_e' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(388) : warning C4013: 'msg_content_disposition_dup_xtra' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(397) : warning C4013: 'msg_content_disposition_dup_one' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(397) : warning C4047: 'return' : 'char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2065: 'sip_content_encoding_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2065: 'sip_content_encoding_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2065: 'sip_content_encoding_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [17]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2065: 'sip_content_language_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2065: 'sip_content_language_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2065: 'sip_content_language_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [17]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2065: 'sip_content_type_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2065: 'sip_content_type_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2065: 'sip_content_type_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [13]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(557) : warning C4013: 'msg_content_type_d' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(562) : warning C4013: 'msg_content_type_e' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(568) : warning C4013: 'msg_content_type_dup_xtra' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(576) : warning C4013: 'msg_content_type_dup_one' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(578) : warning C4047: 'return' : 'char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2065: 'sip_mime_version_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2065: 'sip_mime_version_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2065: 'sip_mime_version_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [13]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2065: 'sip_warning_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2065: 'sip_warning_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2065: 'sip_warning_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2065: 'msg_warning_dup_xtra' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2065: 'msg_warning_dup_one' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'void *'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char [8]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(677) : warning C4013: 'msg_warning_d' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(681) : warning C4013: 'msg_warning_e' undefined; assuming extern returning int
38>sip_parser.c
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser.c(598) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser.c(598) : warning C4013: 'sip_object' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser.c(598) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
42>auserver.c
38>sip_parser_table.c
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(5) : error C2143: syntax error : missing '{' before 'const'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(5) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(5) : error C2065: 'MC_SHORT_SIZE' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(5) : error C2057: expected constant expression
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(5) : error C2466: cannot allocate an array of constant size 0
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : error C2065: 'sip_accept_contact_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : warning C4013: 'offsetof' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : error C2065: 'sip_t' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : error C2065: 'sip_accept_contact' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(8) : error C2065: 'sip_mask_pref' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(8) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(9) : error C2065: 'sip_referred_by_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(9) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(9) : error C2065: 'sip_t' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(9) : error C2065: 'sip_referred_by' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(9) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(10) : error C2065: 'sip_content_type_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(10) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(10) : error C2065: 'sip_t' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(10) : error C2065: 'sip_content_type' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(10) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(11) : error C2065: 'sip_mask_ua' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(11) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(12) : error C2065: 'sip_request_disposition_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(12) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(12) : error C2065: 'sip_t' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(12) : error C2065: 'sip_request_disposition' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(12) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(13) : error C2065: 'sip_mask_pref' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(13) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(14) : error C2065: 'sip_content_encoding_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(14) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(14) : error C2065: 'sip_t' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(14) : error C2065: 'sip_content_encoding' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(14) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(15) : error C2065: 'sip_mask_ua' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(15) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(16) : error C2065: 'sip_from_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(16) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(16) : error C2065: 'sip_t' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(16) : error C2065: 'sip_from' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(16) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(17) : error C2065: 'sip_mask_request' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(17) : error C2065: 'sip_mask_response' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(17) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(18) : error C2065: 'NULL' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(18) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(19) : error C2065: 'NULL' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(19) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(20) : error C2065: 'sip_call_id_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(20) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(20) : error C2065: 'sip_t' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(20) : error C2065: 'sip_call_id' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(20) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(21) : error C2065: 'sip_mask_request' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(21) : error C2065: 'sip_mask_response' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(21) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(22) : error C2065: 'sip_reject_contact_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(22) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(22) : error C2065: 'sip_t' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(22) : error C2065: 'sip_reject_contact' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(22) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(23) : error C2065: 'sip_mask_pref' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(23) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(24) : error C2065: 'sip_supported_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(24) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(24) : error C2065: 'sip_t' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(24) : error C2065: 'sip_supported' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(24) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(25) : error C2065: 'sip_content_length_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(25) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(25) : error C2065: 'sip_t' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(25) : error C2065: 'sip_content_length' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(25) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(26) : error C2065: 'sip_mask_request' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(26) : error C2065: 'sip_mask_response' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(26) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(27) : error C2065: 'sip_contact_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(27) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(27) : error C2065: 'sip_t' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(27) : error C2065: 'sip_contact' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(27) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(28) : error C2065: 'sip_mask_ua' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(28) : error C2065: 'sip_mask_proxy' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(28) : error C2065: 'sip_mask_registrar' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(28) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(29) : error C2065: 'NULL' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(29) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(30) : error C2065: 'sip_event_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(30) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(30) : error C2065: 'sip_t' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(30) : error C2065: 'sip_event' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(30) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(31) : error C2065: 'sip_mask_events' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(31) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(32) : error C2065: 'NULL' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(32) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(33) : error C2065: 'NULL' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(33) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(34) : error C2065: 'sip_refer_to_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(34) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(34) : fatal error C1003: error count exceeds 100; stopping compilation
38>sip_prack.c
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2065: 'sip_rack_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2065: 'sip_rack_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2065: 'sip_rack_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [5]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(122) : warning C4013: 'sip_is_rack' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2065: 'sip_rseq_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2065: 'sip_rseq_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2065: 'sip_rseq_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [5]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(202) : warning C4013: 'sip_is_rseq' undefined; assuming extern returning int
38>sip_pref_util.c
42>cmu_lex.c
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_pref_util.c(529) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_pref_util.c(529) : warning C4013: 'sip_contact_copy' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_pref_util.c(529) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
38>sip_reason.c
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2065: 'sip_reason_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2065: 'sip_reason_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2065: 'sip_reason_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [7]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(120) : warning C4013: 'sip_is_reason' undefined; assuming extern returning int
38>sip_refer.c
42>c:\freeswitch\freeswitch-1.0.6\libs\flite-1.3.99\lang\cmulex\cmu_lex.c(356) : warning C4090: '=' : different 'const' qualifiers
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_refer.c(43) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory
38>sip_security.c
42>cmu_lex_data.c
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2065: 'sip_authorization_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2065: 'sip_authorization_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2065: 'sip_authorization_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [14]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(140) : warning C4013: 'sip_is_authorization' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2065: 'sip_proxy_authenticate_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2065: 'sip_proxy_authenticate_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2065: 'sip_proxy_authenticate_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [19]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(211) : warning C4013: 'sip_is_proxy_authenticate' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2065: 'sip_proxy_authorization_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2065: 'sip_proxy_authorization_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2065: 'sip_proxy_authorization_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [20]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(267) : warning C4013: 'sip_is_proxy_authorization' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2065: 'sip_www_authenticate_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2065: 'sip_www_authenticate_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2065: 'sip_www_authenticate_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [17]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(321) : warning C4013: 'sip_is_www_authenticate' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2065: 'sip_authentication_info_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2065: 'sip_authentication_info_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2065: 'sip_authentication_info_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [20]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(380) : warning C4013: 'sip_is_authentication_info' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2065: 'sip_proxy_authentication_info_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2065: 'sip_proxy_authentication_info_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2065: 'sip_proxy_authentication_info_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [26]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(447) : warning C4013: 'sip_is_proxy_authentication_info' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2065: 'sip_security_client_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2065: 'sip_security_client_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2065: 'sip_security_client_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [16]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2065: 'sip_security_server_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2065: 'sip_security_server_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2065: 'sip_security_server_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [16]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2065: 'sip_security_verify_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2065: 'sip_security_verify_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2065: 'sip_security_verify_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [16]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2065: 'sip_privacy_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2065: 'sip_privacy_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2065: 'sip_privacy_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [8]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>sip_session.c
42>cmu_lex_entries.c
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2065: 'sip_session_expires_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2065: 'sip_session_expires_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2065: 'sip_session_expires_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [16]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2065: 'sip_min_se_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2065: 'sip_min_se_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2065: 'sip_min_se_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [7]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
38>sip_status.c
38>sip_tag.c
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag.c(2) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag.c(2) : warning C4206: nonstandard extension used : translation unit is empty
38>sip_tag_class.c
42>cmu_lts_model.c
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(219) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(219) : warning C4013: 'sip_object' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(219) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(224) : error C2065: 'siptag_end' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(224) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(232) : warning C4013: 'SIPTAG_P' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(247) : error C2065: 'siptag_header' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(247) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(253) : warning C4013: 'SIPTAG_STR_P' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(259) : error C2065: 'siptag_header_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(259) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(364) : error C2065: 'sip_payload_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(454) : error C2065: 'siptag_payload' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(454) : warning C4047: '=' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(454) : error C2065: 'sip_payload_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(454) : warning C4047: '=' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(456) : error C2065: 'sip_tag_list' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(456) : error C2109: subscript requires array or pointer type
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(457) : error C2065: 'sip_tag_list' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(457) : error C2109: subscript requires array or pointer type
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(483) : error C2065: 'siptag_header_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(483) : warning C4047: '=' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(493) : error C2065: 'siptag_header_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(493) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>sip_tag_ref.c
38>sip_time.c
42>cmu_lts_rules.c
38>Generating Code...
38>Compiling...
38>sip_util.c
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(130) : error C2220: warning treated as error - no 'object' file generated
42>cmu_postlex.c
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(130) : warning C4013: 'sip_contact_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(130) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(196) : warning C4047: 'initializing' : 'sip_contact_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(575) : warning C4013: 'sip_route_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(615) : error C2065: 'sip_route_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(615) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(615) : warning C4024: 'sip_route_reverse_as' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(668) : error C2065: 'sip_route_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(668) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(668) : warning C4024: 'sip_route_fixdup_as' : different types for formal and actual parameter 2
38>http_basic.c
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(161) : error C2065: 'http_request_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(161) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(161) : warning C4024: 'msg_header_alloc' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : error C2065: 'http_request_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4113: 'isize_t (__cdecl *)(const http_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const http_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4113: 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(271) : error C2065: 'http_status_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(271) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(271) : warning C4024: 'msg_header_alloc' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : error C2065: 'http_status_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4113: 'isize_t (__cdecl *)(const http_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const http_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4113: 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : error C2065: 'http_accept_ranges_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [14]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : error C2065: 'http_age_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [4]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : error C2065: 'http_allow_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [6]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : error C2065: 'http_authentication_info_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [20]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : error C2065: 'http_authorization_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [14]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : error C2065: 'http_cache_control_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [14]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : error C2065: 'http_connection_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [11]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : error C2065: 'http_content_range_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [14]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(584) : error C2065: 'http_date_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(584) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(584) : warning C4024: 'msg_header_alloc' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : error C2065: 'http_date_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [5]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : error C2065: 'http_etag_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [5]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : error C2065: 'http_expect_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [7]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : error C2065: 'http_expires_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [8]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : error C2065: 'http_from_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [5]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(735) : warning C4013: 'http_host_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(740) : warning C4013: 'http_host_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(740) : warning C4047: 'return' : 'http_host_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : error C2065: 'http_host_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4113: 'isize_t (__cdecl *)(const http_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const http_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4113: 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [5]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : error C2065: 'http_if_match_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [9]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : error C2065: 'http_if_modified_since_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [18]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : error C2065: 'http_if_none_match_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [14]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : error C2065: 'http_if_range_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4113: 'isize_t (__cdecl *)(const http_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const http_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4113: 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [9]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : error C2065: 'http_if_unmodified_since_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [20]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : error C2065: 'http_last_modified_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [14]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : error C2065: 'http_location_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [9]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : error C2065: 'http_max_forwards_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
42>cmu_time_awb_cart.c
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [13]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : error C2065: 'http_pragma_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [7]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : error C2065: 'http_proxy_authenticate_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [19]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : error C2065: 'http_proxy_authorization_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [20]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : error C2065: 'http_range_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [6]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : error C2065: 'http_referer_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [8]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : error C2065: 'http_retry_after_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [12]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : error C2065: 'http_server_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [7]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1402) : warning C4013: 'http_is_te' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : error C2065: 'http_te_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [3]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : error C2065: 'http_trailer_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [8]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : error C2065: 'http_transfer_encoding_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [18]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : error C2065: 'http_upgrade_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [8]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : error C2065: 'http_user_agent_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [11]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : error C2065: 'http_vary_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [5]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : error C2065: 'http_via_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4113: 'isize_t (__cdecl *)(const http_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const http_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4113: 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [4]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2065: 'http_warning_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2065: 'msg_warning_d' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2065: 'msg_warning_e' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2065: 'msg_warning_dup_xtra' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2065: 'msg_warning_dup_one' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'void *'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char [8]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : error C2065: 'http_www_authenticate_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [17]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>http_extra.c
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : error C2065: 'http_proxy_connection_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [17]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : error C2065: 'http_cookie_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [7]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : error C2065: 'http_set_cookie_hash' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [11]'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
38>http_header.c
42>cmu_time_awb_clunits.c
38>..\..\sofia-sip\libsofia-sip-ua\http\http_header.c(223) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\http\http_header.c(223) : warning C4013: 'http_header_vformat' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\http\http_header.c(223) : warning C4047: '=' : 'http_header_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_header.c(274) : warning C4047: '=' : 'http_header_t *' differs in levels of indirection from 'int'
38>http_parser.c
42>cmu_time_awb_lex_entry.c
38>http_parser_table.c
38>..\..\sofia-sip\libsofia-sip-ua\http\http_parser_table.c(6) : error C2143: syntax error : missing '{' before '<cv-qualifer>'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_parser_table.c(6) : error C2059: syntax error : '<Unknown>'
38>http_status.c
38>http_tag.c
38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag.c(2) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag.c(2) : warning C4206: nonstandard extension used : translation unit is empty
38>http_tag_class.c
42>cmu_time_awb_lpc.c
38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(177) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(177) : warning C4013: 'HTTPTAG_P' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(196) : error C2065: 'httptag_header' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(196) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(203) : warning C4013: 'HTTPTAG_STR_P' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(209) : error C2065: 'httptag_header_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(209) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>http_tag_ref.c
38>nth_client.c
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_client.c(652) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_client.c(652) : warning C4013: 'HTTPTAG_VERSION_REF' undefined; assuming extern returning int
38>nth_server.c
42>cmu_time_awb_mcep.c
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(718) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(718) : warning C4013: 'HTTPTAG_SERVER_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(719) : warning C4013: 'HTTPTAG_SERVER_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(751) : warning C4013: 'http_server_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(751) : warning C4047: '=' : 'http_server_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(753) : warning C4013: 'http_server_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(753) : warning C4047: '=' : 'http_server_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(892) : warning C4013: 'http_location_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(966) : warning C4013: 'http_payload_format' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(970) : warning C4047: '=' : 'http_payload_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(975) : warning C4013: 'http_status_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(981) : warning C4013: 'HTTPTAG_STATUS' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(981) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(981) : warning C4024: 'http_add_tl' : different types for formal and actual parameter 3
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(982) : warning C4013: 'HTTPTAG_SERVER' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(983) : warning C4013: 'HTTPTAG_CONTENT_TYPE_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(984) : warning C4013: 'HTTPTAG_SEPARATOR_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(985) : warning C4013: 'HTTPTAG_CONNECTION_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(985) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1077) : error C2065: 'http_www_authenticate_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1077) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1109) : warning C4013: 'HTTPTAG_HEADER' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1109) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1109) : warning C4024: 'nth_request_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1147) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1147) : warning C4024: 'nth_request_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1148) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1148) : warning C4024: 'nth_request_treply' : different types for formal and actual parameter 5
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1263) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1263) : warning C4024: 'http_add_tl' : different types for formal and actual parameter 3
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1295) : warning C4013: 'http_date_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1295) : error C2223: left of '->d_time' must point to struct/union
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1311) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1311) : warning C4024: 'http_add_tl' : different types for formal and actual parameter 3
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1311) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1311) : warning C4024: 'http_add_tl' : different types for formal and actual parameter 4
38>nth_tag.c
38>nth_tag_ref.c
42>cmu_us_kal.c
38>sres.c
42>cmu_us_kal_diphone.c
42>cmu_us_kal_lpc.c
38>sres_blocking.c
42>cmu_us_kal_res.c
38>sres_cache.c
38>sres_sip.c
38>sresolv.c
38>nea.c
42>Generating Code...
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : error C2065: 'siptag_to' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : warning C4024: 'tl_find' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : error C2065: 'siptag_to_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : warning C4024: 'tl_find' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : error C2065: 'siptag_from' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : warning C4024: 'tl_find' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : error C2065: 'siptag_from_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : warning C4024: 'tl_find' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(135) : error C2065: 'siptag_contact' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(135) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(135) : warning C4024: 'tl_find' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(136) : error C2065: 'siptag_contact_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(136) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(136) : warning C4024: 'tl_find' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(155) : warning C4013: 'SIPTAG_CONTACT' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(155) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(156) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(156) : warning C4024: 'tl_tlist' : different types for formal and actual parameter 3
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(160) : warning C4013: 'SIPTAG_EXPIRES_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(160) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(160) : warning C4024: 'tl_gets' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(161) : warning C4013: 'SIPTAG_EXPIRES_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(162) : warning C4013: 'SIPTAG_TO_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(170) : warning C4013: 'sip_to_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(170) : warning C4047: '=' : 'sip_to_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(173) : warning C4013: 'sip_expires_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(173) : warning C4047: '=' : 'sip_expires_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(175) : warning C4013: 'sip_expires_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(175) : warning C4047: '=' : 'sip_expires_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(180) : warning C4013: 'SIPTAG_EXPIRES' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(180) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(180) : warning C4024: 'tl_tremove' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(181) : warning C4013: 'SIPTAG_EXPIRES_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(195) : warning C4013: 'SIPTAG_FROM' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(195) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(196) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_typedef_t'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(196) : warning C4024: 'nta_leg_tcreate' : different types for formal and actual parameter 5
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(205) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(205) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 8
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(206) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_typedef_t'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(206) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 9
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(242) : warning C4013: 'SIPTAG_CONTENT_TYPE_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(242) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(242) : warning C4024: 'tl_gets' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(243) : warning C4013: 'SIPTAG_CONTENT_TYPE_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(244) : warning C4013: 'SIPTAG_PAYLOAD_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(254) : warning C4013: 'SIPTAG_CONTENT_TYPE' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(254) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(254) : warning C4024: 'tl_tremove' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(255) : warning C4013: 'SIPTAG_CONTENT_TYPE_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(256) : warning C4013: 'SIPTAG_PAYLOAD' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(257) : warning C4013: 'SIPTAG_PAYLOAD_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(263) : warning C4047: '=' : 'sip_expires_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(273) : warning C4013: 'SIPTAG_TO' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(273) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(273) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 8
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(275) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(276) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(307) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(307) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 8
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(308) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_typedef_t'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(308) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 9
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(346) : warning C4013: 'SIPTAG_ALLOW_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(346) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(346) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(346) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(346) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(410) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(410) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 8
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(411) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_typedef_t'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(411) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 9
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(416) : warning C4013: 'sip_expires_format' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(416) : warning C4047: '=' : 'sip_expires_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(425) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(425) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 8
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(426) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_typedef_t'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(426) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 9
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(496) : warning C4013: 'sip_subscription_state_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(594) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(594) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 8
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(595) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_typedef_t'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(595) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 9
38>Generating Code...
38>Compiling...
38>nea_debug.c
38>nea_event.c
42>Compiling...
42>cmu_us_kal_residx.c
42>cst_alloc.c
38>nea_server.c
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(401) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(401) : warning C4013: 'SIPTAG_CONTACT_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(401) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(401) : warning C4024: 'tl_gets' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(402) : warning C4013: 'SIPTAG_CONTACT_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(403) : warning C4013: 'SIPTAG_ALLOW_EVENTS_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(404) : warning C4013: 'SIPTAG_SERVER_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(405) : warning C4013: 'SIPTAG_REQUIRE_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(406) : warning C4013: 'SIPTAG_REQUIRE_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(450) : warning C4013: 'sip_allow_events_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(450) : warning C4047: '=' : 'sip_allow_events_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(452) : warning C4013: 'sip_allow_events_make' undefined; assuming extern returning int
42>cst_args.c
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(452) : warning C4047: '=' : 'sip_allow_events_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(454) : warning C4013: 'sip_allow_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(454) : warning C4047: '=' : 'sip_allow_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(463) : warning C4013: 'sip_contact_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(463) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(465) : warning C4013: 'sip_contact_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(465) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(488) : warning C4013: 'sip_require_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(488) : warning C4047: '=' : 'sip_require_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(490) : warning C4013: 'sip_require_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(490) : warning C4047: '=' : 'sip_require_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(726) : warning C4013: 'SIPTAG_CONTENT_TYPE_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(726) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(726) : warning C4024: 'tl_gets' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(727) : warning C4013: 'SIPTAG_CONTENT_TYPE_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(728) : warning C4013: 'SIPTAG_PAYLOAD_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(729) : warning C4013: 'SIPTAG_PAYLOAD_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(785) : warning C4013: 'sip_payload_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(786) : warning C4013: 'sip_payload_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(786) : warning C4047: '=' : 'sip_payload_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(788) : warning C4013: 'sip_content_type_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(789) : warning C4013: 'sip_content_type_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(789) : warning C4047: '=' : 'sip_content_type_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(825) : warning C4047: '=' : 'sip_payload_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1208) : warning C4013: 'SIPTAG_CONTENT_TYPE_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1208) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1208) : warning C4024: 'nea_event_tcreate' : different types for formal and actual parameter 6
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1209) : warning C4013: 'SIPTAG_ACCEPT_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1270) : warning C4013: 'SIPTAG_ACCEPT_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1271) : warning C4013: 'SIPTAG_ACCEPT_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1272) : warning C4013: 'SIPTAG_SUPPORTED_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1273) : warning C4013: 'SIPTAG_SUPPORTED_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1280) : warning C4013: 'sip_event_format' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1283) : warning C4047: '=' : 'sip_event_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1294) : warning C4047: '=' : 'sip_require_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1296) : warning C4047: '=' : 'sip_require_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1307) : warning C4013: 'sip_accept_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1307) : warning C4047: '=' : 'const sip_accept_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1309) : warning C4047: '=' : 'const sip_accept_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1315) : warning C4013: 'sip_accept_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1315) : warning C4047: '=' : 'const sip_accept_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1317) : warning C4047: '=' : 'const sip_accept_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1320) : warning C4013: 'sip_supported_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1320) : warning C4047: '=' : 'sip_supported_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1322) : warning C4013: 'sip_supported_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1322) : warning C4047: '=' : 'sip_supported_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1503) : warning C4013: 'sip_from_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1503) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1508) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1516) : warning C4013: 'SIPTAG_FROM' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1516) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1516) : warning C4024: 'nta_leg_tcreate' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1517) : warning C4013: 'SIPTAG_TO' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1518) : warning C4013: 'SIPTAG_CALL_ID' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1576) : warning C4013: 'SIPTAG_ALLOW_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1576) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1576) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1577) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1577) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1619) : warning C4013: 'SIP_EXPIRES_INIT' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1619) : warning C4047: 'initializing' : 'msg_header_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1631) : warning C4013: 'SIPTAG_SERVER_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1631) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1631) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1632) : warning C4013: 'SIPTAG_ALLOW_EVENTS' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1633) : warning C4013: 'SIPTAG_ALLOW' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1643) : warning C4013: 'sip_min_expires_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1648) : warning C4013: 'SIPTAG_ACCEPT' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1648) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1648) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1649) : warning C4013: 'SIPTAG_MIN_EXPIRES' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1666) : warning C4013: 'SIPTAG_REQUIRE' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1666) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1666) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1667) : warning C4013: 'SIPTAG_UNSUPPORTED' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1688) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1688) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1689) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1689) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1734) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1734) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1746) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1746) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1748) : warning C4013: 'SIPTAG_SUPPORTED' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1763) : warning C4013: 'sip_accept_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1777) : warning C4047: '=' : 'sip_accept_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1853) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1853) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1879) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1884) : warning C4047: '=' : 'sip_content_type_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1886) : warning C4047: '=' : 'sip_payload_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1901) : warning C4013: 'sip_event_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1901) : warning C4047: '=' : 'sip_event_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1909) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1909) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1914) : warning C4013: 'SIPTAG_EXPIRES' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1915) : warning C4013: 'SIPTAG_CONTACT' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1942) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1942) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1943) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1943) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1946) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1946) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1956) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1956) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2038) : warning C4013: 'sip_subscription_state_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2092) : warning C4013: 'SIPTAG_SUBSCRIPTION_STATE' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2092) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2092) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 8
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2095) : warning C4013: 'SIPTAG_USER_AGENT_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2097) : warning C4013: 'SIPTAG_EVENT' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2099) : warning C4013: 'SIPTAG_CONTENT_TYPE' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2099) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2101) : warning C4013: 'SIPTAG_PAYLOAD' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2101) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>nea_tag.c
42>cst_cart.c
38>nea_tag_ref.c
38>auth_client.c
42>cst_cg.c
38>auth_common.c
42>cst_clunits.c
38>auth_digest.c
38>auth_module.c
42>cst_diphone.c
38>auth_module_http.c
38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_http.c(47) : error C2065: 'http_www_authenticate_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_http.c(47) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_http.c(50) : error C2065: 'http_proxy_authenticate_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_http.c(50) : error C2099: initializer is not a constant
42>cst_endian.c
38>auth_module_sip.c
42>cst_error.c
38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(49) : error C2065: 'sip_www_authenticate_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(49) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(51) : error C2065: 'sip_authentication_info_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(51) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(54) : error C2065: 'sip_proxy_authenticate_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(54) : error C2099: initializer is not a constant
42>cst_features.c
38>auth_plugin.c
42>cst_ffeature.c
38>auth_plugin_delayed.c
38>auth_tag.c
42>cst_ffeatures.c
38>auth_tag_ref.c
38>iptsec_debug.c
42>cst_file_stdio.c
38>stun.c
42>cst_item.c
38>stun_common.c
42>cst_lexicon.c
38>stun_dns.c
42>cst_lpcres.c
38>stun_mini.c
38>Generating Code...
42>cst_lts.c
38>Compiling...
38>stun_tag.c
42>cst_lts_rewrites.c
38>stun_tag_ref.c
38>nua.c
42>cst_mlpg.c
38>nua_client.c
42>cst_mlsa.c
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(496) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(496) : warning C4013: 'sip_object' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(496) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(502) : error C2065: 'siptag_contact' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(502) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(503) : error C2065: 'siptag_contact_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(503) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(529) : error C2065: 'siptag_contact' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(529) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(530) : error C2065: 'siptag_contact_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(530) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(549) : warning C4013: 'sip_add_tagis' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(569) : warning C4013: 'sip_to_tag' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(573) : warning C4013: 'sip_add_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(577) : warning C4013: 'sip_add_dup_as' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(577) : error C2065: 'sip_to_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(615) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(625) : error C2065: 'siptag_from' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(625) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(743) : warning C4013: 'SIPTAG_CALL_ID' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(743) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(743) : warning C4024: 'nta_leg_tcreate' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(744) : warning C4013: 'SIPTAG_FROM' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(745) : warning C4013: 'SIPTAG_TO' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(746) : warning C4013: 'SIPTAG_CSEQ' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(754) : warning C4013: 'sip_from_tag' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(768) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(781) : warning C4013: 'sip_route_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(781) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(813) : warning C4013: 'sip_to_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(813) : warning C4047: '=' : 'const sip_to_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(815) : warning C4013: 'sip_from_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(815) : warning C4047: '=' : 'const sip_from_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(830) : warning C4013: 'sip_has_feature' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(832) : warning C4013: 'sip_add_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(832) : error C2065: 'sip_supported_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(835) : error C2065: 'sip_organization_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(838) : error C2065: 'sip_user_agent_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(869) : warning C4013: 'sip_contact_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(869) : warning C4047: 'initializing' : 'sip_contact_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1164) : warning C4013: 'sip_expires_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1181) : error C2065: 'sip_authorization_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1181) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1181) : warning C4024: 'auc_challenge' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1186) : error C2065: 'sip_proxy_authorization_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1186) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1186) : warning C4024: 'auc_challenge' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1477) : error C2065: 'sip_authorization_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1477) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1477) : warning C4024: 'auc_info' : different types for formal and actual parameter 3
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1481) : error C2065: 'sip_proxy_authorization_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1481) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1481) : warning C4024: 'auc_info' : different types for formal and actual parameter 3
38>nua_common.c
42>Generating Code...
42>Compiling...
42>cst_mmap_win32.c
38>nua_dialog.c
42>cst_phoneset.c
38>c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.h(41) : error C2061: syntax error : identifier 'nua_owner_t'
38>c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.h(41) : error C2059: syntax error : ';'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(47) : error C2016: C requires that a struct or union has at least one member
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(47) : error C2061: syntax error : identifier 'nua_owner_t'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(58) : error C2143: syntax error : missing '{' before ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(58) : error C2059: syntax error : ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(60) : error C2143: syntax error : missing '{' before ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(60) : error C2059: syntax error : ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(61) : error C2143: syntax error : missing '{' before ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(61) : error C2059: syntax error : ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(63) : error C2143: syntax error : missing '{' before ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(63) : error C2059: syntax error : ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(64) : error C2143: syntax error : missing '{' before ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(64) : error C2059: syntax error : ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(65) : error C2143: syntax error : missing '{' before ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(65) : error C2059: syntax error : ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(67) : error C2143: syntax error : missing '{' before ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(67) : error C2059: syntax error : ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(68) : error C2143: syntax error : missing '{' before ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(68) : error C2059: syntax error : ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(70) : error C2059: syntax error : ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(92) : error C2059: syntax error : '}'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(97) : error C2143: syntax error : missing ')' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(97) : error C2143: syntax error : missing ';' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(97) : error C2059: syntax error : '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(99) : error C2059: syntax error : ')'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(100) : error C2143: syntax error : missing ')' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(100) : error C2143: syntax error : missing '{' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(100) : error C2059: syntax error : ','
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(104) : error C2059: syntax error : ')'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(113) : error C2143: syntax error : missing ')' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(113) : error C2143: syntax error : missing '{' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(113) : error C2059: syntax error : ','
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(114) : error C2059: syntax error : ')'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(115) : error C2143: syntax error : missing ')' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(115) : error C2143: syntax error : missing '{' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(115) : error C2059: syntax error : ','
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(116) : error C2059: syntax error : ')'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(117) : error C2059: syntax error : '}'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(123) : error C2061: syntax error : identifier 'nua_usage_class'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(129) : error C2143: syntax error : missing '{' before ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(129) : error C2059: syntax error : ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(130) : error C2143: syntax error : missing '{' before ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(130) : error C2059: syntax error : ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(131) : error C2059: syntax error : ':'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(140) : error C2059: syntax error : '}'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(142) : error C2143: syntax error : missing ')' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(142) : error C2143: syntax error : missing '{' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(142) : error C2059: syntax error : ','
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(143) : error C2059: syntax error : ')'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(144) : error C2143: syntax error : missing ')' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(144) : error C2143: syntax error : missing '{' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(144) : error C2059: syntax error : ','
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(145) : error C2059: syntax error : ')'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(146) : error C2143: syntax error : missing ')' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(146) : error C2143: syntax error : missing '{' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(146) : error C2059: syntax error : ','
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(147) : error C2059: syntax error : ')'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(148) : error C2143: syntax error : missing ')' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(148) : error C2143: syntax error : missing '{' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(148) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(149) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(149) : error C2371: 'nua_dialog_state_t' : redefinition; different basic types
38>        c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.h(43) : see declaration of 'nua_dialog_state_t'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(149) : error C2143: syntax error : missing ';' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(149) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(149) : error C2059: syntax error : ')'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(149) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(149) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(150) : error C2143: syntax error : missing ')' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(150) : error C2143: syntax error : missing '{' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(150) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(151) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(151) : error C2371: 'nua_dialog_state_t' : redefinition; different basic types
38>        c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.h(43) : see declaration of 'nua_dialog_state_t'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(151) : error C2143: syntax error : missing ';' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(151) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(151) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(152) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(152) : error C2371: 'nua_dialog_usage_t' : redefinition; different basic types
38>        c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.h(44) : see declaration of 'nua_dialog_usage_t'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(152) : error C2143: syntax error : missing ';' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(152) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(152) : error C2059: syntax error : ')'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(152) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(152) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(156) : error C2037: left of 'ds_reporting' specifies undefined struct/union 'nua_dialog_state'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(156) : warning C4033: 'nua_dialog_is_reporting' must return a value
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(161) : error C2143: syntax error : missing ')' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(161) : error C2143: syntax error : missing '{' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(161) : error C2059: syntax error : ','
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(164) : error C2059: syntax error : ')'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(167) : error C2143: syntax error : missing ')' before 'const'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(167) : error C2081: 'nua_usage_class' : name in formal parameter list illegal
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(167) : error C2143: syntax error : missing '{' before 'const'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(167) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(168) : error C2373: 'sip_event_t' : redefinition; different type modifiers
38>        c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\sip\sofia-sip/sip.h(204) : see declaration of 'sip_event_t'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(168) : error C2143: syntax error : missing ';' before 'const'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(168) : error C2059: syntax error : ')'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(170) : error C2143: syntax error : missing ')' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(170) : error C2143: syntax error : missing '{' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(170) : error C2059: syntax error : ','
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(174) : error C2059: syntax error : ')'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(186) : error C2143: syntax error : missing ')' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(186) : error C2143: syntax error : missing '{' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(186) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(187) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(187) : error C2371: 'nua_dialog_state_t' : redefinition; different basic types
38>        c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.h(43) : see declaration of 'nua_dialog_state_t'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(187) : error C2143: syntax error : missing ';' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(187) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(187) : error C2059: syntax error : ')'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(187) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(187) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : error C2143: syntax error : missing ')' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : error C2143: syntax error : missing '{' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : error C2371: 'nua_dialog_state_t' : redefinition; different basic types
38>        c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.h(43) : see declaration of 'nua_dialog_state_t'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : error C2143: syntax error : missing ';' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : error C2059: syntax error : ')'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(191) : error C2143: syntax error : missing ')' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(191) : error C2143: syntax error : missing '{' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(191) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(192) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(192) : error C2371: 'nua_dialog_state_t' : redefinition; different basic types
38>        c:\FreeSWITCH\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.h(43) : see declaration of 'nua_dialog_state_t'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(192) : error C2143: syntax error : missing ';' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(192) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(192) : error C2059: syntax error : ')'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(192) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(192) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(204) : error C2143: syntax error : missing ')' before '*'
38>c:\freeswitch\freeswitch-1.0.6\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(204) : fatal error C1003: error count exceeds 100; stopping compilation
38>nua_event_server.c
42>cst_reflpc.c
42>cst_regex.c
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(86) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(86) : warning C4013: 'SIPTAG_EVENT_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(87) : warning C4013: 'SIPTAG_EVENT_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(88) : warning C4013: 'SIPTAG_CONTENT_TYPE_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(89) : warning C4013: 'SIPTAG_PAYLOAD_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(90) : warning C4013: 'SIPTAG_PAYLOAD_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(108) : warning C4013: 'sip_event_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(108) : warning C4047: '=' : 'const sip_event_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(122) : warning C4013: 'SIPTAG_EVENT' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(122) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(122) : warning C4024: 'nua_stack_tevent' : different types for formal and actual parameter 7
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(123) : warning C4013: 'SIPTAG_CONTENT_TYPE' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(158) : warning C4013: 'SIPTAG_ACCEPT_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(158) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(158) : warning C4024: 'tl_gets' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(159) : warning C4013: 'SIPTAG_ACCEPT_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(160) : warning C4013: 'SIPTAG_CONTENT_TYPE_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(169) : warning C4013: 'sip_header_as_string' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(169) : warning C4047: '=' : 'char *' differs in levels of indirection from 'int'
42>cst_rel_io.c
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(334) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(334) : warning C4024: 'tl_gets' : different types for formal and actual parameter 2
38>nua_extension.c
42>cst_relation.c
38>nua_message.c
42>cst_sigpr.c
38>nua_notifier.c
42>cst_socket.c
42>cst_ss.c
42>cst_string.c
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_notifier.c(47) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory
38>nua_options.c
42>cst_sts.c
38>nua_params.c
42>cst_synth.c
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(166) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(166) : warning C4013: 'sip_allow_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(166) : warning C4047: '=' : 'sip_allow_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(167) : warning C4013: 'sip_supported_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(167) : warning C4047: '=' : 'sip_supported_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(175) : warning C4047: '=' : 'sip_allow_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(196) : warning C4013: 'SIPTAG_FROM_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(196) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(196) : warning C4024: 'tl_gets' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(197) : warning C4013: 'SIPTAG_FROM_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(203) : warning C4013: 'sip_from_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(208) : warning C4013: 'sip_from_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(208) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(212) : warning C4013: 'sip_from_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(212) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(225) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(833) : error C2065: 'siptag_supported' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(833) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(834) : error C2065: 'siptag_supported_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(834) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(839) : error C2065: 'sip_supported_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(839) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(839) : warning C4024: 'nhp_merge_lists' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(841) : error C2065: 'siptag_supported' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(841) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(853) : error C2065: 'siptag_allow_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(853) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(854) : error C2065: 'siptag_allow' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(854) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(859) : error C2065: 'sip_allow_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(859) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(859) : warning C4024: 'nhp_merge_lists' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(863) : error C2065: 'siptag_allow' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(863) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(875) : error C2065: 'siptag_allow_events_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(875) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(876) : error C2065: 'siptag_allow_events' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(876) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(881) : error C2065: 'sip_allow_events_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(881) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(881) : warning C4024: 'nhp_merge_lists' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(885) : error C2065: 'siptag_allow_events' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(885) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(903) : error C2065: 'sip_allow_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(903) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(903) : warning C4024: 'nhp_merge_lists' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(923) : warning C4013: 'sip_route_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(923) : warning C4013: 'sip_route_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(923) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(927) : error C2065: 'siptag_user_agent' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(927) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(928) : warning C4013: 'sip_header_as_string' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(928) : warning C4047: '=' : 'char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(931) : error C2065: 'siptag_user_agent_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(931) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(951) : error C2065: 'siptag_organization' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(951) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(952) : warning C4047: '=' : 'char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(955) : error C2065: 'siptag_organization_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(955) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1201) : error C2065: 'siptag_from' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1201) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1205) : error C2065: 'siptag_from_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1205) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1209) : error C2065: 'siptag_to' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1209) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1213) : error C2065: 'siptag_to_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1213) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1235) : warning C4047: '=' : 'const sip_from_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1242) : warning C4013: 'sip_to_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1242) : warning C4047: '=' : 'const sip_to_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1244) : warning C4013: 'sip_to_create' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1244) : warning C4047: '=' : 'const sip_to_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1256) : warning C4013: 'SIPTAG_FROM' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1256) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1257) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'const tag_type_s *'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1257) : warning C4024: 'tl_filtered_tlist' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1258) : warning C4013: 'SIPTAG_TO' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1258) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1269) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1269) : warning C4024: 'tl_gets' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1270) : warning C4013: 'SIPTAG_TO_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1319) : error C2065: 'siptag_from' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1319) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1319) : error C2065: 'siptag_to' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1319) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1327) : error C2065: 'siptag_from_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1327) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1329) : error C2065: 'siptag_to_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1329) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1333) : error C2065: 'siptag_cseq' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1333) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1333) : error C2065: 'siptag_cseq_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1333) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1335) : error C2065: 'siptag_rseq' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1335) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1335) : error C2065: 'siptag_rseq_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1335) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1337) : error C2065: 'siptag_rack' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1337) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1337) : error C2065: 'siptag_rack_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1337) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1339) : error C2065: 'siptag_timestamp' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1339) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1339) : error C2065: 'siptag_timestamp_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1339) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1341) : error C2065: 'siptag_content_length' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1341) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1341) : error C2065: 'siptag_content_length_str' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1341) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1575) : warning C4013: 'siptag_contact_vr' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1621) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1625) : warning C4013: 'SIPTAG_FROM_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1625) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1625) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1625) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'const tag_type_s *'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1625) : warning C4024: 'tl_filtered_tlist' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1662) : warning C4013: 'SIPTAG_SUPPORTED' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1662) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1663) : warning C4013: 'SIPTAG_SUPPORTED_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1663) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1663) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1664) : warning C4013: 'SIPTAG_ALLOW' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1664) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1665) : warning C4013: 'SIPTAG_ALLOW_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1665) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1665) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1666) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1667) : warning C4013: 'SIPTAG_ALLOW_EVENTS' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1667) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1668) : warning C4013: 'SIPTAG_ALLOW_EVENTS_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1668) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1668) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1669) : warning C4013: 'SIPTAG_USER_AGENT' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1669) : warning C4013: 'sip_user_agent_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1669) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1669) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1670) : warning C4013: 'SIPTAG_USER_AGENT_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1670) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1673) : warning C4013: 'SIPTAG_ORGANIZATION' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1673) : warning C4013: 'sip_organization_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1673) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1673) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1674) : warning C4013: 'SIPTAG_ORGANIZATION_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1674) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1676) : warning C4013: 'siptag_route_v' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1677) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1705) : warning C4013: 'siptag_contact_v' undefined; assuming extern returning int
38>nua_publish.c
42>cst_tokenstream.c
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(292) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(292) : warning C4013: 'msg_copy' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(292) : warning C4047: '=' : 'msg_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(314) : warning C4013: 'sip_etag_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(314) : warning C4047: '=' : 'sip_etag_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(317) : warning C4013: 'sip_header_remove' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(355) : warning C4013: 'SIPTAG_IF_MATCH' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(355) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(355) : warning C4024: 'nua_base_client_trequest' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(356) : warning C4013: 'SIPTAG_PAYLOAD' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(356) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(356) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'const tag_type_s *'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(356) : warning C4024: 'nua_base_client_trequest' : different types for formal and actual parameter 5
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(357) : warning C4013: 'SIPTAG_CONTENT_TYPE' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(357) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(358) : warning C4013: 'SIPTAG_EXPIRES_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(358) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(408) : warning C4047: '=' : 'sip_etag_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(534) : warning C4013: 'msg_header_find_param' undefined; assuming extern returning int
38>nua_register.c
42>cst_track.c
42>cst_track_io.c
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(756) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(756) : warning C4013: 'sip_add_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(770) : warning C4013: 'sip_header_remove' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(793) : warning C4013: 'SIPTAG_EXPIRES_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(793) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(798) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_typedef_t'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(798) : warning C4024: 'nua_base_client_trequest' : different types for formal and actual parameter 5
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(812) : warning C4013: 'sip_object' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(812) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(855) : warning C4013: 'sip_now' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(860) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(905) : warning C4013: 'sip_contact_expires' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(939) : warning C4013: 'sip_route_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(939) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(952) : warning C4013: 'sip_path_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(952) : warning C4047: '=' : 'sip_path_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1319) : warning C4013: 'sip_via_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1319) : error C2223: left of '->v_next' must point to struct/union
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1320) : error C2065: 'sip_transport_udp' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1320) : warning C4047: '=' : 'const char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1323) : error C2065: 'sip_transport_tcp' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1323) : warning C4047: '=' : 'const char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1360) : warning C4013: 'sip_via_copy' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1360) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1374) : error C2065: 'sip_transport_tcp' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1374) : warning C4047: '==' : 'const char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1375) : error C2065: 'sip_transport_udp' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1375) : warning C4047: '=' : 'const char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1376) : error C2065: 'sip_transport_udp' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1376) : warning C4047: '==' : 'const char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1377) : error C2065: 'sip_transport_tcp' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1377) : warning C4047: '=' : 'const char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1422) : warning C4013: 'sip_via_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1422) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1647) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1674) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1743) : warning C4013: 'sip_contact_format' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1755) : warning C4047: '=' : 'const sip_contact_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1806) : warning C4013: 'sip_from_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1806) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1834) : warning C4013: 'sip_contact_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1834) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(2006) : warning C4013: 'sip_via_port' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(2006) : warning C4047: '=' : 'const char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(2122) : warning C4013: 'sip_contact_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(2122) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
38>nua_registrar.c
42>cst_units.c
42>cst_utt_utils.c
38>nua_server.c
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(118) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(118) : warning C4013: 'SIPTAG_SUPPORTED' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(118) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(118) : warning C4024: 'nta_check_method' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(119) : warning C4013: 'SIPTAG_USER_AGENT_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(132) : warning C4013: 'SIPTAG_ALLOW' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(132) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(132) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(140) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(140) : warning C4024: 'nta_check_required' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(178) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(178) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
42>cst_utterance.c
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(207) : warning C4013: 'sip_object' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(207) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(236) : warning C4013: 'sip_is_allowed' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(266) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(266) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(267) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(267) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(497) : warning C4013: 'SIPTAG_END' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(497) : warning C4047: 'initializing' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(519) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(530) : warning C4013: 'sip_add_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(533) : warning C4013: 'sip_add_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(533) : error C2065: 'sip_user_agent_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(537) : error C2065: 'sip_organization_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(570) : warning C4013: 'sip_contact_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(570) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
38>nua_session.c
42>cst_val.c
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(796) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(796) : warning C4013: 'sip_has_feature' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(818) : warning C4013: 'sip_accept_contact_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(825) : warning C4013: 'sip_add_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1075) : warning C4013: 'SIP_IS_ALLOWED' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1078) : warning C4013: 'sip_rack_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1085) : warning C4013: 'SIPTAG_RACK' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1085) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1085) : warning C4024: 'nua_client_tcreate' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1086) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1086) : warning C4024: 'nua_client_tcreate' : different types for formal and actual parameter 5
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1260) : warning C4013: 'sip_object' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1260) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1265) : warning C4013: 'sip_authorization' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1265) : warning C4047: '=' : 'sip_authorization_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1266) : warning C4013: 'sip_proxy_authorization' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1266) : warning C4047: '=' : 'sip_proxy_authorization_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1271) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1275) : warning C4013: 'sip_cseq_create' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1275) : warning C4047: '=' : 'sip_cseq_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1279) : warning C4013: 'sip_add_tl' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1287) : warning C4013: 'sip_header_insert' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1294) : warning C4013: 'sip_header_remove' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1361) : warning C4013: 'SIPTAG_END' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2139) : warning C4013: 'sip_add_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2139) : error C2065: 'sip_accept_encoding_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2145) : error C2065: 'sip_accept_encoding_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2333) : warning C4013: 'sip_warning_format' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2334) : warning C4047: '=' : 'sip_warning_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2710) : warning C4013: 'SIPTAG_REASON_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2710) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2710) : warning C4024: 'nua_server_trespond' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2711) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2711) : warning C4024: 'nua_server_trespond' : different types for formal and actual parameter 3
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2807) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2807) : warning C4024: 'nua_server_trespond' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2808) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2808) : warning C4024: 'nua_server_trespond' : different types for formal and actual parameter 3
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2967) : warning C4013: 'siptag_event_vr' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2984) : warning C4013: 'siptag_event_v' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2988) : warning C4013: 'sip_event_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2988) : warning C4047: '=' : 'sip_event_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3048) : warning C4013: 'SIPTAG_EVENT' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3048) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3048) : warning C4024: 'nua_stack_post_signal' : different types for formal and actual parameter 3
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3049) : warning C4013: 'SIPTAG_SUBSCRIPTION_STATE_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3050) : warning C4013: 'SIPTAG_CONTENT_TYPE_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3051) : warning C4013: 'SIPTAG_PAYLOAD_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3752) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3752) : warning C4024: 'nua_base_client_trequest' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3753) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_typedef_t'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3753) : warning C4024: 'nua_base_client_trequest' : different types for formal and actual parameter 5
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4214) : warning C4013: 'sip_retry_after_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4305) : warning C4013: 'sip_min_se_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4305) : error C2223: left of '->min_delta' must point to struct/union
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4411) : error C2223: left of '->min_delta' must point to struct/union
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4413) : warning C4013: 'sip_session_expires_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4413) : error C2223: left of '->x_delta' must point to struct/union
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4424) : warning C4013: 'SIPTAG_SESSION_EXPIRES' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4424) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4428) : warning C4013: 'SIPTAG_MIN_SE' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4428) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4430) : warning C4013: 'SIPTAG_REQUIRE_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4430) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4624) : warning C4013: 'sip_payload_create' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4624) : warning C4047: '=' : 'sip_payload_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4625) : warning C4013: 'sip_content_type_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4625) : warning C4047: '=' : 'sip_content_type_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4627) : warning C4013: 'sip_content_disposition_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4627) : warning C4047: '=' : 'sip_content_disposition_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4702) : warning C4013: 'SIPTAG_ACCEPT' undefined; assuming extern returning int
42>cst_val_const.c
38>nua_stack.c
42>Generating Code...
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(152) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(152) : warning C4013: 'sip_from_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(174) : warning C4013: 'sip_accept_make' undefined; assuming extern returning int
42>Compiling...
42>cst_val_user.c
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(174) : warning C4047: '=' : 'sip_accept_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(396) : warning C4013: 'sip_object' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(396) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(396) : warning C4047: 'function' : 'const sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(396) : warning C4024: 'function through pointer' : different types for formal and actual parameter 8
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(993) : error C2223: left of '->a_display' must point to struct/union
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(996) : warning C4013: 'sip_to_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(996) : error C2223: left of '->a_display' must point to struct/union
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(1001) : warning C4013: 'SIPTAG_TO' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(1002) : warning C4013: 'SIPTAG_FROM' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(1014) : warning C4013: 'SIPTAG_CALL_ID' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(1014) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(1014) : warning C4024: 'nta_leg_tcreate' : different types for formal and actual parameter 4
38>nua_subnotref.c
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_subnotref.c(50) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory
38>nua_tag.c
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_tag.c(2389) : error C2065: 'sip_event_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\nua_tag.c(2389) : error C2099: initializer is not a constant
38>Generating Code...
42>cst_vc.c
38>Compiling...
42>cst_viterbi.c
42>cst_voice.c
38>nua_tag_ref.c
38>outbound.c
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(405) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(405) : warning C4013: 'sip_contact_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(405) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(523) : warning C4013: 'sip_via_port' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(523) : warning C4047: '=' : 'const char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(623) : warning C4013: 'sip_contact_format' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(623) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(623) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(667) : error C2065: 'sip_transport_udp' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(667) : warning C4047: '==' : 'const char *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(709) : warning C4013: 'sip_object' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(709) : warning C4047: 'initializing' : 'const sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(739) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(754) : warning C4013: 'sip_accept_contact_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(754) : warning C4047: '=' : 'sip_accept_contact_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(773) : warning C4013: 'sip_add_tl' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(774) : warning C4013: 'SIPTAG_TO' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(775) : warning C4013: 'SIPTAG_FROM' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(780) : warning C4013: 'SIPTAG_ROUTE' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(782) : warning C4013: 'SIPTAG_MAX_FORWARDS_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(782) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(783) : warning C4013: 'SIPTAG_SUBJECT_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(783) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(784) : warning C4013: 'SIPTAG_CALL_ID_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(785) : warning C4013: 'SIPTAG_ACCEPT_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(817) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(878) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(892) : error C2065: 'sip_authorization_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(892) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(892) : warning C4024: 'auc_challenge' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(897) : error C2065: 'sip_proxy_authorization_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(897) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(897) : warning C4024: 'auc_challenge' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1000) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1022) : warning C4013: 'SIPTAG_MAX_FORWARDS' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1061) : warning C4013: 'SIPTAG_CONTENT_TYPE_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1061) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1061) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1062) : warning C4013: 'SIPTAG_PAYLOAD_STR' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1101) : warning C4013: 'sip_via_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1101) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1187) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1268) : warning C4013: 'sip_has_feature' undefined; assuming extern returning int
38>nta.c
42>cst_wave.c
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(909) : error C2220: warning treated as error - no 'object' file generated
42>cst_wave_io.c
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(909) : warning C4013: 'sip_max_forwards_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(1494) : warning C4013: 'siptag_contact_vr' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(1564) : warning C4013: 'sip_contact_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(1564) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(1776) : warning C4013: 'siptag_contact_v' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2379) : warning C4013: 'sip_via_format' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2384) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2395) : warning C4013: 'sip_via_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2395) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2397) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2534) : warning C4013: 'sip_object' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2534) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2548) : warning C4013: 'sip_via_copy' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2548) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2745) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2839) : error C2065: 'sip_error_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2839) : warning C4047: '==' : 'msg_hclass_t *const ' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3033) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3389) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3563) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3647) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3648) : warning C4047: 'initializing' : 'const sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3658) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3660) : warning C4013: 'sip_from_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3660) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3662) : warning C4013: 'sip_to_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3662) : warning C4047: '=' : 'sip_to_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3665) : warning C4013: 'sip_call_id_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3665) : warning C4047: '=' : 'sip_call_id_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3667) : warning C4013: 'sip_cseq_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3667) : warning C4047: '=' : 'sip_cseq_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3705) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3707) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3721) : warning C4013: 'SIPTAG_TO' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3721) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3721) : warning C4024: 'sip_add_tl' : different types for formal and actual parameter 3
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3722) : warning C4013: 'SIPTAG_FROM' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3723) : warning C4013: 'SIPTAG_CALL_ID' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3740) : warning C4013: 'sip_route_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3740) : error C2223: left of '->r_url' must point to struct/union
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3741) : warning C4013: 'sip_route_dup' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3741) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3750) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3825) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3839) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3845) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3851) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3921) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3938) : warning C4047: '=' : 'sip_to_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3966) : warning C4047: '=' : 'sip_call_id_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4104) : warning C4013: 'SIPTAG_CALL_ID_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4105) : warning C4013: 'SIPTAG_CALL_ID_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4106) : warning C4013: 'SIPTAG_FROM_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4107) : warning C4013: 'SIPTAG_FROM_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4108) : warning C4013: 'SIPTAG_TO_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4109) : warning C4013: 'SIPTAG_TO_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4110) : warning C4013: 'SIPTAG_ROUTE_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4113) : warning C4013: 'SIPTAG_CSEQ_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4114) : warning C4013: 'SIPTAG_CSEQ_STR_REF' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4136) : warning C4013: 'sip_is_to' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4137) : warning C4047: '=' : 'sip_to_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4140) : warning C4013: 'sip_to_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4140) : warning C4047: '=' : 'sip_to_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4145) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4148) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4150) : warning C4013: 'sip_from_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4150) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4153) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4157) : warning C4013: 'sip_contact_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4160) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4212) : warning C4047: '=' : 'sip_call_id_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4214) : warning C4013: 'sip_call_id_make' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4214) : warning C4047: '=' : 'sip_call_id_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4577) : warning C4013: 'sip_replaces_format' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4579) : warning C4047: 'return' : 'sip_replaces_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4594) : warning C4013: 'sip_call_id_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5066) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5217) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5267) : warning C4013: 'sip_request_copy' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5267) : warning C4047: '=' : 'sip_request_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5268) : warning C4013: 'sip_from_copy' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5268) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5269) : warning C4013: 'sip_to_copy' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5269) : warning C4047: '=' : 'sip_to_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5270) : warning C4013: 'sip_call_id_copy' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5270) : warning C4047: '=' : 'sip_call_id_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5271) : warning C4013: 'sip_cseq_copy' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5271) : warning C4047: '=' : 'sip_cseq_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5272) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5285) : warning C4013: 'sip_record_route_copy' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5285) : warning C4047: '=' : 'sip_record_route_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5292) : warning C4013: 'sip_timestamp_copy' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5292) : warning C4047: '=' : 'sip_timestamp_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6307) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6309) : warning C4047: '=' : 'sip_to_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6311) : warning C4047: '=' : 'sip_call_id_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6313) : warning C4047: '=' : 'sip_cseq_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6315) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6346) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6404) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6482) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(7221) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(7312) : warning C4047: 'function' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(7312) : warning C4024: 'sip_add_tl' : different types for formal and actual parameter 2
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(7682) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(7858) : warning C4047: 'initializing' : 'const sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8052) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8054) : warning C4013: 'sip_timestamp_format' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8055) : warning C4047: 'initializing' : 'sip_timestamp_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8194) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8240) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8358) : warning C4013: 'sip_supported_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9443) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9443) : warning C4024: 'outgoing_ackmsg' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9443) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9443) : warning C4024: 'outgoing_ackmsg' : different types for formal and actual parameter 5
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9463) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9464) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9482) : warning C4013: 'sip_header_remove' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9605) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9618) : warning C4013: 'sip_retry_after_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9618) : error C2223: left of '->af_delta' must point to struct/union
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10794) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10828) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10907) : warning C4013: 'sip_rseq_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10912) : error C2065: 'sip_require_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10912) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10912) : warning C4024: 'sip_add_make' : different types for formal and actual parameter 3
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11075) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11335) : warning C4047: '=' : 'sip_to_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11421) : warning C4013: 'sip_rack_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11476) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11484) : error C2223: left of '->r_url' must point to struct/union
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11485) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11491) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11515) : warning C4013: 'SIPTAG_RACK' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11515) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11516) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11516) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'const tag_type_s *'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11516) : warning C4024: 'sip_add_tl' : different types for formal and actual parameter 4
38>nta_check.c
42>cst_wave_utils.c
42>flite.c
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(83) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(83) : warning C4013: 'SIPTAG_UNSUPPORTED' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(83) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(83) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(84) : warning C4013: 'SIPTAG_SUPPORTED' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(124) : warning C4013: 'SIPTAG_REQUIRE' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(124) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(124) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(125) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(125) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(169) : warning C4013: 'SIPTAG_ALLOW' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(169) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(169) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(170) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(170) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(175) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(175) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(176) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(176) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(262) : warning C4013: 'SIPTAG_ACCEPT' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(262) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(262) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(263) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(263) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(335) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(335) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(336) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(336) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(376) : warning C4013: 'sip_min_se_init' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(376) : error C2223: left of '->min_delta' must point to struct/union
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(380) : warning C4013: 'SIPTAG_MIN_SE' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(380) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(380) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(381) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(381) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
38>nta_tag.c
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(178) : error C2065: 'sip_contact_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(178) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(184) : error C2065: 'sip_contact_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(184) : error C2099: initializer is not a constant
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(191) : error C2065: 'sip_contact_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(191) : error C2099: initializer is not a constant
38>nta_tag_ref.c
42>rateconv.c
38>sl_read_payload.c
38>sl_utils_log.c
42>regexp.c
38>sl_utils_print.c
38>tport.c
42>regsub.c
42>us_aswd.c
38>tport_logging.c
38>tport_stub_sigcomp.c
42>us_dur_stats.c
38>tport_stub_stun.c
42>us_durz_cart.c
38>tport_tag.c
42>us_expand.c
38>tport_tag_ref.c
38>tport_type_connect.c
42>us_f0_model.c
38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(173) : error C2220: warning treated as error - no 'object' file generated
38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(173) : warning C4013: 'http_request_format' undefined; assuming extern returning int
38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(173) : warning C4047: '=' : 'http_request_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(179) : error C2065: 'http_host_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(179) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(179) : warning C4024: 'msg_header_add_make' : different types for formal and actual parameter 3
38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(180) : error C2065: 'http_separator_class' : undeclared identifier
38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(180) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
38>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(180) : warning C4024: 'msg_header_add_make' : different types for formal and actual parameter 3
38>tport_type_tcp.c
42>us_f0lr.c
38>tport_type_udp.c
38>sdp.c
42>us_ffeatures.c
38>sdp_parse.c
42>us_gpos.c
38>Generating Code...
42>us_int_accent_cart.c
38>Compiling...
38>sdp_print.c
38>sdp_tag.c
38>sdp_tag_ref.c
38>soa.c
42>Generating Code...
42>Compiling...
42>us_int_tone_cart.c
38>soa_static.c
42>us_nums_cart.c
38>soa_tag.c
42>us_phoneset.c
38>soa_tag_ref.c
38>inet_ntop.c
42>us_phrasing_cart.c
38>Generating Code...
38>Creating browse information file...
42>us_text.c
38>Microsoft Browse Information Maintenance Utility Version 9.00.30729
38>Copyright (C) Microsoft Corporation. All rights reserved.
38>BSCMAKE: error BK1506 : cannot open file '.\Debug\sip_basic.sbr': No such file or directory
38>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\sofia\Debug\BuildLog.htm"
38>libsofia_sip_ua_static - 899 error(s), 1733 warning(s)
42>usenglish.c
43>------ Build started: Project: portaudio, Configuration: Debug Win32 ------
43>Compiling...
43>pa_converters.c
43>pa_cpuload.c
43>pa_debugprint.c
43>pa_dither.c
43>pa_front.c
42>cmu_us_slt.c
43>pa_process.c
43>pa_skeleton.c
43>pa_stream.c
43>pa_trace.c
43>pa_win_wmme.c
42>cmu_us_slt_cg.c
43>pa_win_hostapis.c
42>cmu_us_slt_cg_durmodel.c
43>pa_win_util.c
43>pa_win_waveformat.c
42>cmu_us_slt_cg_f0_trees.c
43>pa_win_wdmks_utils.c
42>cmu_us_slt_cg_mcep_trees.c
43>Generating Code...
43>Compiling...
43>pa_x86_plain_converters.c
43>pa_allocation.c
43>Generating Code...
42>cmu_us_slt_cg_params.c
43>Creating library...
43>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\portaudio\build\msvc\Win32\Debug\BuildLog.htm"
43>portaudio - 0 error(s), 0 warning(s)
44>------ Skipped Build: Project: Download 32khzsound, Configuration: Debug Win32 ------
44>Project not selected to build for this solution configuration
45>------ Build started: Project: mod_spidermonkey, Configuration: Debug Win32 ------
45>Compiling...
45>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
45>mod_spidermonkey.c
42>cmu_us_slt_cg_phonestate.c
42>cmu_us_awb.c
45>Linking...
45>   Creating library Win32\Debug\mod_spidermonkey.lib and object Win32\Debug\mod_spidermonkey.exp
45>Embedding manifest...
42>cmu_us_awb_cg.c
42>cmu_us_awb_cg_durmodel.c
45>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_spidermonkey\Win32\Debug\BuildLog.htm"
45>mod_spidermonkey - 0 error(s), 1 warning(s)
46>------ Build started: Project: libspeex, Configuration: Debug Win32 ------
46>Compiling...
46>cb_search.c
42>cmu_us_awb_cg_f0_trees.c
46>exc_10_16_table.c
46>exc_10_32_table.c
46>exc_20_32_table.c
46>exc_5_256_table.c
46>exc_5_64_table.c
46>exc_8_128_table.c
46>filters.c
46>gain_table.c
46>gain_table_lbr.c
46>hexc_10_32_table.c
46>hexc_table.c
46>high_lsp_tables.c
46>lpc.c
46>lsp.c
46>lsp_tables_nb.c
46>ltp.c
46>modes.c
42>cmu_us_awb_cg_mcep_trees.c
46>modes_wb.c
46>nb_celp.c
46>Generating Code...
46>Compiling...
46>quant_lsp.c
46>sb_celp.c
46>speex.c
46>speex_callbacks.c
46>speex_header.c
46>stereo.c
46>vbr.c
46>vq.c
46>window.c
46>bits.c
42>cmu_us_awb_cg_params.c
46>Generating Code...
46>Creating library...
46>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\speex\win32\VS2008\libspeex\Debug\BuildLog.htm"
46>libspeex - 0 error(s), 0 warning(s)
47>------ Skipped Build: Project: Download 16khzsound, Configuration: Debug Win32 ------
47>Project not selected to build for this solution configuration
48>------ Build started: Project: Download 8khzsound, Configuration: Debug Win32 ------
48>Downloading 8khzsound.
48>Sound name: en-us-callie Version 1.0.12
48>URL: http://files.freeswitch.org/freeswitch-sounds-en-us-callie-8000-1.0.12.tar.gz
48>Downloading: http://files.freeswitch.org/freeswitch-sounds-en-us-callie-8000-1.0.12.tar.gz
42>cmu_us_awb_cg_phonestate.c
42>Generating Code...
42>Compiling...
42>cmu_us_rms.c
48>Extracting: C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\freeswitch-sounds-en-us-callie-8000-1.0.12.tar.gz
48>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
48>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\freeswitch-sounds-en-us-callie-8000-1.0.12.tar.gz
48>Extracting  freeswitch-sounds-en-us-callie-8000-1.0.12.tar
48>Everything is Ok
42>cmu_us_rms_cg.c
48>7-Zip (A) 4.32  Copyright (c) 1999-2005 Igor Pavlov  2005-12-09
48>Processing archive: C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\freeswitch-sounds-en-us-callie-8000-1.0.12.tar
48>Extracting  en
48>Extracting  en\us
48>Extracting  en\us\callie
48>Extracting  en\us\callie\time
48>Extracting  en\us\callie\time\8000
48>Extracting  en\us\callie\time\8000\day-5.wav
48>Extracting  en\us\callie\time\8000\seconds.wav
48>Extracting  en\us\callie\time\8000\day-1.wav
48>Extracting  en\us\callie\time\8000\mon-5.wav
48>Extracting  en\us\callie\time\8000\today.wav
48>Extracting  en\us\callie\time\8000\mon-11.wav
48>Extracting  en\us\callie\time\8000\minutes.wav
48>Extracting  en\us\callie\time\8000\mon-10.wav
48>Extracting  en\us\callie\time\8000\yesterday.wav
48>Extracting  en\us\callie\time\8000\mon-1.wav
48>Extracting  en\us\callie\time\8000\day-3.wav
48>Extracting  en\us\callie\time\8000\oh.wav
48>Extracting  en\us\callie\time\8000\second.wav
48>Extracting  en\us\callie\time\8000\minute.wav
48>Extracting  en\us\callie\time\8000\day-4.wav
48>Extracting  en\us\callie\time\8000\a-m.wav
48>Extracting  en\us\callie\time\8000\mon-3.wav
48>Extracting  en\us\callie\time\8000\mon-8.wav
48>Extracting  en\us\callie\time\8000\at.wav
48>Extracting  en\us\callie\time\8000\tomorrow.wav
48>Extracting  en\us\callie\time\8000\mon-9.wav
48>Extracting  en\us\callie\time\8000\day-2.wav
48>Extracting  en\us\callie\time\8000\mon-4.wav
48>Extracting  en\us\callie\time\8000\p-m.wav
48>Extracting  en\us\callie\time\8000\hours.wav
48>Extracting  en\us\callie\time\8000\day-0.wav
48>Extracting  en\us\callie\time\8000\mon-7.wav
48>Extracting  en\us\callie\time\8000\mon-2.wav
48>Extracting  en\us\callie\time\8000\hour.wav
48>Extracting  en\us\callie\time\8000\oclock.wav
48>Extracting  en\us\callie\time\8000\day-6.wav
48>Extracting  en\us\callie\time\8000\mon-0.wav
48>Extracting  en\us\callie\time\8000\mon-6.wav
48>Extracting  en\us\callie\currency
48>Extracting  en\us\callie\currency\8000
48>Extracting  en\us\callie\currency\8000\dollar.wav
48>Extracting  en\us\callie\currency\8000\dollars.wav
48>Extracting  en\us\callie\currency\8000\minus.wav
48>Extracting  en\us\callie\currency\8000\cents-per-minute.wav
48>Extracting  en\us\callie\currency\8000\central.wav
48>Extracting  en\us\callie\currency\8000\cents.wav
48>Extracting  en\us\callie\currency\8000\cent.wav
48>Extracting  en\us\callie\currency\8000\negative.wav
48>Extracting  en\us\callie\currency\8000\and.wav
48>Extracting  en\us\callie\digits
48>Extracting  en\us\callie\digits\8000
48>Extracting  en\us\callie\digits\8000\h-10.wav
48>Extracting  en\us\callie\digits\8000\h-12.wav
48>Extracting  en\us\callie\digits\8000\h-7.wav
48>Extracting  en\us\callie\digits\8000\19.wav
48>Extracting  en\us\callie\digits\8000\dot.wav
48>Extracting  en\us\callie\digits\8000\40.wav
48>Extracting  en\us\callie\digits\8000\7.wav
48>Extracting  en\us\callie\digits\8000\12.wav
48>Extracting  en\us\callie\digits\8000\h-17.wav
48>Extracting  en\us\callie\digits\8000\h-13.wav
48>Extracting  en\us\callie\digits\8000\30.wav
48>Extracting  en\us\callie\digits\8000\70.wav
48>Extracting  en\us\callie\digits\8000\60.wav
48>Extracting  en\us\callie\digits\8000\h-14.wav
48>Extracting  en\us\callie\digits\8000\h-18.wav
48>Extracting  en\us\callie\digits\8000\9.wav
48>Extracting  en\us\callie\digits\8000\20.wav
48>Extracting  en\us\callie\digits\8000\6.wav
48>Extracting  en\us\callie\digits\8000\h-4.wav
48>Extracting  en\us\callie\digits\8000\1.wav
48>Extracting  en\us\callie\digits\8000\h-3.wav
48>Extracting  en\us\callie\digits\8000\h-6.wav
48>Extracting  en\us\callie\digits\8000\period.wav
48>Extracting  en\us\callie\digits\8000\50.wav
48>Extracting  en\us\callie\digits\8000\h-9.wav
48>Extracting  en\us\callie\digits\8000\2.wav
48>Extracting  en\us\callie\digits\8000\point.wav
48>Extracting  en\us\callie\digits\8000\h-2.wav
48>Extracting  en\us\callie\digits\8000\18.wav
48>Extracting  en\us\callie\digits\8000\h-5.wav
48>Extracting  en\us\callie\digits\8000\3.wav
48>Extracting  en\us\callie\digits\8000\80.wav
48>Extracting  en\us\callie\digits\8000\h-16.wav
48>Extracting  en\us\callie\digits\8000\15.wav
48>Extracting  en\us\callie\digits\8000\hundred.wav
48>Extracting  en\us\callie\digits\8000\10.wav
48>Extracting  en\us\callie\digits\8000\h-8.wav
48>Extracting  en\us\callie\digits\8000\8.wav
42>cmu_us_rms_cg_durmodel.c
48>Extracting  en\us\callie\digits\8000\star.wav
48>Extracting  en\us\callie\digits\8000\17.wav
48>Extracting  en\us\callie\digits\8000\h-15.wav
48>Extracting  en\us\callie\digits\8000\h-11.wav
48>Extracting  en\us\callie\digits\8000\h-19.wav
48>Extracting  en\us\callie\digits\8000\16.wav
48>Extracting  en\us\callie\digits\8000\million.wav
48>Extracting  en\us\callie\digits\8000\4.wav
48>Extracting  en\us\callie\digits\8000\13.wav
48>Extracting  en\us\callie\digits\8000\pound.wav
48>Extracting  en\us\callie\digits\8000\11.wav
48>Extracting  en\us\callie\digits\8000\h-1.wav
48>Extracting  en\us\callie\digits\8000\90.wav
48>Extracting  en\us\callie\digits\8000\h-30.wav
48>Extracting  en\us\callie\digits\8000\thousand.wav
48>Extracting  en\us\callie\digits\8000\0.wav
48>Extracting  en\us\callie\digits\8000\14.wav
48>Extracting  en\us\callie\digits\8000\5.wav
48>Extracting  en\us\callie\digits\8000\h-20.wav
48>Extracting  en\us\callie\base256
48>Extracting  en\us\callie\base256\8000
48>Extracting  en\us\callie\base256\8000\adviser.wav
48>Extracting  en\us\callie\base256\8000\python.wav
48>Extracting  en\us\callie\base256\8000\bookshelf.wav
48>Extracting  en\us\callie\base256\8000\consensus.wav
48>Extracting  en\us\callie\base256\8000\fallout.wav
48>Extracting  en\us\callie\base256\8000\detector.wav
48>Extracting  en\us\callie\base256\8000\crumpled.wav
48>Extracting  en\us\callie\base256\8000\dreadful.wav
48>Extracting  en\us\callie\base256\8000\paragraph.wav
48>Extracting  en\us\callie\base256\8000\skydive.wav
48>Extracting  en\us\callie\base256\8000\clergyman.wav
48>Extracting  en\us\callie\base256\8000\detergent.wav
48>Extracting  en\us\callie\base256\8000\prowler.wav
48>Extracting  en\us\callie\base256\8000\klaxon.wav
48>Extracting  en\us\callie\base256\8000\penetrate.wav
48>Extracting  en\us\callie\base256\8000\Apollo.wav
48>Extracting  en\us\callie\base256\8000\topmost.wav
48>Extracting  en\us\callie\base256\8000\conformist.wav
48>Extracting  en\us\callie\base256\8000\crossover.wav
48>Extracting  en\us\callie\base256\8000\repay.wav
48>Extracting  en\us\callie\base256\8000\fracture.wav
48>Extracting  en\us\callie\base256\8000\willow.wav
48>Extracting  en\us\callie\base256\8000\pedigree.wav
48>Extracting  en\us\callie\base256\8000\microwave.wav
48>Extracting  en\us\callie\base256\8000\inception.wav
48>Extracting  en\us\callie\base256\8000\Jamaica.wav
48>Extracting  en\us\callie\base256\8000\puppy.wav
48>Extracting  en\us\callie\base256\8000\showgirl.wav
48>Extracting  en\us\callie\base256\8000\dictator.wav
48>Extracting  en\us\callie\base256\8000\gadgetry.wav
48>Extracting  en\us\callie\base256\8000\Yucatan.wav
48>Extracting  en\us\callie\base256\8000\orca.wav
48>Extracting  en\us\callie\base256\8000\getaway.wav
48>Extracting  en\us\callie\base256\8000\dropper.wav
48>Extracting  en\us\callie\base256\8000\jawbone.wav
48>Extracting  en\us\callie\base256\8000\snapshot.wav
48>Extracting  en\us\callie\base256\8000\unwind.wav
48>Extracting  en\us\callie\base256\8000\optic.wav
48>Extracting  en\us\callie\base256\8000\designing.wav
48>Extracting  en\us\callie\base256\8000\payday.wav
48>Extracting  en\us\callie\base256\8000\truncated.wav
48>Extracting  en\us\callie\base256\8000\antenna.wav
48>Extracting  en\us\callie\base256\8000\briefcase.wav
48>Extracting  en\us\callie\base256\8000\Aztec.wav
48>Extracting  en\us\callie\base256\8000\coherence.wav
48>Extracting  en\us\callie\base256\8000\company.wav
48>Extracting  en\us\callie\base256\8000\therapist.wav
48>Extracting  en\us\callie\base256\8000\sensation.wav
48>Extracting  en\us\callie\base256\8000\megaton.wav
48>Extracting  en\us\callie\base256\8000\Eskimo.wav
48>Extracting  en\us\callie\base256\8000\Capricorn.wav
48>Extracting  en\us\callie\base256\8000\artist.wav
48>Extracting  en\us\callie\base256\8000\intention.wav
48>Extracting  en\us\callie\base256\8000\tissue.wav
48>Extracting  en\us\callie\base256\8000\quota.wav
48>Extracting  en\us\callie\base256\8000\vagabond.wav
48>Extracting  en\us\callie\base256\8000\phonetic.wav
48>Extracting  en\us\callie\base256\8000\village.wav
48>Extracting  en\us\callie\base256\8000\revenge.wav
48>Extracting  en\us\callie\base256\8000\celebrate.wav
48>Extracting  en\us\callie\base256\8000\exodus.wav
48>Extracting  en\us\callie\base256\8000\kiwi.wav
42>cmu_us_rms_cg_f0_trees.c
48>Extracting  en\us\callie\base256\8000\ancient.wav
48>Extracting  en\us\callie\base256\8000\supportive.wav
48>Extracting  en\us\callie\base256\8000\politeness.wav
48>Extracting  en\us\callie\base256\8000\prefer.wav
48>Extracting  en\us\callie\base256\8000\aimless.wav
48>Extracting  en\us\callie\base256\8000\infancy.wav
48>Extracting  en\us\callie\base256\8000\hurricane.wav
48>Extracting  en\us\callie\base256\8000\universe.wav
48>Extracting  en\us\callie\base256\8000\upset.wav
48>Extracting  en\us\callie\base256\8000\councilman.wav
48>Extracting  en\us\callie\base256\8000\component.wav
48>Extracting  en\us\callie\base256\8000\Waterloo.wav
48>Extracting  en\us\callie\base256\8000\molecule.wav
48>Extracting  en\us\callie\base256\8000\sandalwood.wav
48>Extracting  en\us\callie\base256\8000\cannonball.wav
48>Extracting  en\us\callie\base256\8000\undaunted.wav
48>Extracting  en\us\callie\base256\8000\waffle.wav
48>Extracting  en\us\callie\base256\8000\mosquito.wav
48>Extracting  en\us\callie\base256\8000\tolerance.wav
48>Extracting  en\us\callie\base256\8000\retraction.wav
48>Extracting  en\us\callie\base256\8000\drumbeat.wav
48>Extracting  en\us\callie\base256\8000\sweatband.wav
48>Extracting  en\us\callie\base256\8000\tracker.wav
48>Extracting  en\us\callie\base256\8000\nightbird.wav
48>Extracting  en\us\callie\base256\8000\music.wav
48>Extracting  en\us\callie\base256\8000\direction.wav
48>Extracting  en\us\callie\base256\8000\glossary.wav
48>Extracting  en\us\callie\base256\8000\cumbersome.wav
48>Extracting  en\us\callie\base256\8000\flatfoot.wav
48>Extracting  en\us\callie\base256\8000\travesty.wav
48>Extracting  en\us\callie\base256\8000\minnow.wav
48>Extracting  en\us\callie\base256\8000\cement.wav
48>Extracting  en\us\callie\base256\8000\cubic.wav
48>Extracting  en\us\callie\base256\8000\egghead.wav
48>Extracting  en\us\callie\base256\8000\autopsy.wav
48>Extracting  en\us\callie\base256\8000\souvenir.wav
48>Extracting  en\us\callie\base256\8000\sawdust.wav
48>Extracting  en\us\callie\base256\8000\vapor.wav
48>Extracting  en\us\callie\base256\8000\concurrent.wav
48>Extracting  en\us\callie\base256\8000\revolver.wav
48>Extracting  en\us\callie\base256\8000\reproduce.wav
48>Extracting  en\us\callie\base256\8000\enchanting.wav
48>Extracting  en\us\callie\base256\8000\Chicago.wav
48>Extracting  en\us\callie\base256\8000\offload.wav
48>Extracting  en\us\callie\base256\8000\stormy.wav
48>Extracting  en\us\callie\base256\8000\locale.wav
48>Extracting  en\us\callie\base256\8000\holiness.wav
48>Extracting  en\us\callie\base256\8000\recipe.wav
48>Extracting  en\us\callie\base256\8000\spheroid.wav
48>Extracting  en\us\callie\base256\8000\breakaway.wav
48>Extracting  en\us\callie\base256\8000\Dakota.wav
48>Extracting  en\us\callie\base256\8000\specialist.wav
48>Extracting  en\us\callie\base256\8000\whimsical.wav
48>Extracting  en\us\callie\base256\8000\backwater.wav
48>Extracting  en\us\callie\base256\8000\Zulu.wav
48>Extracting  en\us\callie\base256\8000\spellbind.wav
48>Extracting  en\us\callie\base256\8000\Athens.wav
48>Extracting  en\us\callie\base256\8000\vocalist.wav
48>Extracting  en\us\callie\base256\8000\standard.wav
48>Extracting  en\us\callie\base256\8000\insincere.wav
48>Extracting  en\us\callie\base256\8000\talon.wav
48>Extracting  en\us\callie\base256\8000\cleanup.wav
48>Extracting  en\us\callie\base256\8000\atlas.wav
48>Extracting  en\us\callie\base256\8000\provincial.wav
48>Extracting  en\us\callie\base256\8000\chambermaid.wav
48>Extracting  en\us\callie\base256\8000\revival.wav
48>Extracting  en\us\callie\base256\8000\stagehand.wav
48>Extracting  en\us\callie\base256\8000\millionaire.wav
48>Extracting  en\us\callie\base256\8000\paragon.wav
48>Extracting  en\us\callie\base256\8000\midsummer.wav
48>Extracting  en\us\callie\base256\8000\hamlet.wav
48>Extracting  en\us\callie\base256\8000\brickyard.wav
48>Extracting  en\us\callie\base256\8000\highchair.wav
48>Extracting  en\us\callie\base256\8000\retrieval.wav
48>Extracting  en\us\callie\base256\8000\bodyguard.wav
48>Extracting  en\us\callie\base256\8000\banjo.wav
48>Extracting  en\us\callie\base256\8000\positive.wav
48>Extracting  en\us\callie\base256\8000\keyboard.wav
48>Extracting  en\us\callie\base256\8000\flytrap.wav
48>Extracting  en\us\callie\base256\8000\enlist.wav
48>Extracting  en\us\callie\base256\8000\savagery.wav
48>Extracting  en\us\callie\base256\8000\slowdown.wav
48>Extracting  en\us\callie\base256\8000\Oakland.wav
48>Extracting  en\us\callie\base256\8000\amulet.wav
48>Extracting  en\us\callie\base256\8000\beeswax.wav
48>Extracting  en\us\callie\base256\8000\impetus.wav
48>Extracting  en\us\callie\base256\8000\candidate.wav
48>Extracting  en\us\callie\base256\8000\steamship.wav
48>Extracting  en\us\callie\base256\8000\befriend.wav
48>Extracting  en\us\callie\base256\8000\aggregate.wav
48>Extracting  en\us\callie\base256\8000\dwelling.wav
48>Extracting  en\us\callie\base256\8000\customer.wav
48>Extracting  en\us\callie\base256\8000\Neptune.wav
48>Extracting  en\us\callie\base256\8000\transit.wav
48>Extracting  en\us\callie\base256\8000\rebirth.wav
48>Extracting  en\us\callie\base256\8000\swelter.wav
48>Extracting  en\us\callie\base256\8000\hazardous.wav
48>Extracting  en\us\callie\base256\8000\Wyoming.wav
48>Extracting  en\us\callie\base256\8000\warranty.wav
48>Extracting  en\us\callie\base256\8000\performance.wav
48>Extracting  en\us\callie\base256\8000\trouble.wav
48>Extracting  en\us\callie\base256\8000\torpedo.wav
48>Extracting  en\us\callie\base256\8000\tambourine.wav
48>Extracting  en\us\callie\base256\8000\indulge.wav
48>Extracting  en\us\callie\base256\8000\eightball.wav
48>Extracting  en\us\callie\base256\8000\bedlamp.wav
48>Extracting  en\us\callie\base256\8000\accrue.wav
48>Extracting  en\us\callie\base256\8000\apple.wav
48>Extracting  en\us\callie\base256\8000\Virginia.wav
48>Extracting  en\us\callie\base256\8000\preclude.wav
48>Extracting  en\us\callie\base256\8000\southward.wav
48>Extracting  en\us\callie\base256\8000\backfield.wav
48>Extracting  en\us\callie\base256\8000\ahead.wav
48>Extracting  en\us\callie\base256\8000\tycoon.wav
48>Extracting  en\us\callie\base256\8000\ultimate.wav
48>Extracting  en\us\callie\base256\8000\scavenger.wav
48>Extracting  en\us\callie\base256\8000\maritime.wav
48>Extracting  en\us\callie\base256\8000\surrender.wav
48>Extracting  en\us\callie\base256\8000\chopper.wav
48>Extracting  en\us\callie\base256\8000\escapade.wav
48>Extracting  en\us\callie\base256\8000\pocketful.wav
48>Extracting  en\us\callie\base256\8000\sociable.wav
48>Extracting  en\us\callie\base256\8000\Scotland.wav
48>Extracting  en\us\callie\base256\8000\Mohawk.wav
48>Extracting  en\us\callie\base256\8000\classroom.wav
48>Extracting  en\us\callie\base256\8000\yesteryear.wav
48>Extracting  en\us\callie\base256\8000\eating.wav
48>Extracting  en\us\callie\base256\8000\stopwatch.wav
48>Extracting  en\us\callie\base256\8000\physique.wav
48>Extracting  en\us\callie\base256\8000\indoors.wav
48>Extracting  en\us\callie\base256\8000\select.wav
48>Extracting  en\us\callie\base256\8000\merit.wav
48>Extracting  en\us\callie\base256\8000\rhythm.wav
48>Extracting  en\us\callie\base256\8000\lockup.wav
48>Extracting  en\us\callie\base256\8000\embezzle.wav
48>Extracting  en\us\callie\base256\8000\tactics.wav
48>Extracting  en\us\callie\base256\8000\forever.wav
48>Extracting  en\us\callie\base256\8000\tunnel.wav
48>Extracting  en\us\callie\base256\8000\congregate.wav
48>Extracting  en\us\callie\base256\8000\commence.wav
48>Extracting  en\us\callie\base256\8000\crowfoot.wav
48>Extracting  en\us\callie\base256\8000\quadrant.wav
48>Extracting  en\us\callie\base256\8000\examine.wav
48>Extracting  en\us\callie\base256\8000\retouch.wav
48>Extracting  en\us\callie\base256\8000\matchmaker.wav
48>Extracting  en\us\callie\base256\8000\recover.wav
48>Extracting  en\us\callie\base256\8000\deadbolt.wav
48>Extracting  en\us\callie\base256\8000\virus.wav
48>Extracting  en\us\callie\base256\8000\brackish.wav
48>Extracting  en\us\callie\base256\8000\wayside.wav
48>Extracting  en\us\callie\base256\8000\watchword.wav
48>Extracting  en\us\callie\base256\8000\chairlift.wav
48>Extracting  en\us\callie\base256\8000\unearth.wav
48>Extracting  en\us\callie\base256\8000\Orlando.wav
48>Extracting  en\us\callie\base256\8000\hamburger.wav
48>Extracting  en\us\callie\base256\8000\beaming.wav
48>Extracting  en\us\callie\base256\8000\dinosaur.wav
48>Extracting  en\us\callie\base256\8000\frighten.wav
48>Extracting  en\us\callie\base256\8000\trombonist.wav
48>Extracting  en\us\callie\base256\8000\eyeglass.wav
48>Extracting  en\us\callie\base256\8000\baboon.wav
48>Extracting  en\us\callie\base256\8000\belowground.wav
42>cmu_us_rms_cg_mcep_trees.c
48>Extracting  en\us\callie\base256\8000\berserk.wav
48>Extracting  en\us\callie\base256\8000\upshot.wav
48>Extracting  en\us\callie\base256\8000\perceptive.wav
48>Extracting  en\us\callie\base256\8000\hesitate.wav
48>Extracting  en\us\callie\base256\8000\applicant.wav
48>Extracting  en\us\callie\base256\8000\aardvark.wav
48>Extracting  en\us\callie\base256\8000\publisher.wav
48>Extracting  en\us\callie\base256\8000\opulent.wav
48>Extracting  en\us\callie\base256\8000\ringbolt.wav
48>Extracting  en\us\callie\base256\8000\Brazilian.wav
48>Extracting  en\us\callie\base256\8000\onlooker.wav
48>Extracting  en\us\callie\base256\8000\Wichita.wav
48>Extracting  en\us\callie\base256\8000\breakup.wav
48>Extracting  en\us\callie\base256\8000\crusade.wav
48>Extracting  en\us\callie\base256\8000\hemisphere.wav
48>Extracting  en\us\callie\base256\8000\nebula.wav
48>Extracting  en\us\callie\base256\8000\tempest.wav
48>Extracting  en\us\callie\base256\8000\decadence.wav
48>Extracting  en\us\callie\base256\8000\glucose.wav
48>Extracting  en\us\callie\base256\8000\stairway.wav
48>Extracting  en\us\callie\base256\8000\printer.wav
48>Extracting  en\us\callie\base256\8000\typewriter.wav
48>Extracting  en\us\callie\base256\8000\bison.wav
48>Extracting  en\us\callie\base256\8000\adult.wav
48>Extracting  en\us\callie\base256\8000\potato.wav
48>Extracting  en\us\callie\base256\8000\gossamer.wav
48>Extracting  en\us\callie\base256\8000\sailboat.wav
48>Extracting  en\us\callie\base256\8000\assume.wav
48>Extracting  en\us\callie\base256\8000\pupil.wav
48>Extracting  en\us\callie\base256\8000\shadow.wav
48>Extracting  en\us\callie\base256\8000\Medusa.wav
48>Extracting  en\us\callie\base256\8000\businessman.wav
48>Extracting  en\us\callie\base256\8000\Trojan.wav
48>Extracting  en\us\callie\base256\8000\surmount.wav
48>Extracting  en\us\callie\base256\8000\exceed.wav
48>Extracting  en\us\callie\base256\8000\Vulcan.wav
48>Extracting  en\us\callie\base256\8000\newsletter.wav
48>Extracting  en\us\callie\base256\8000\filament.wav
48>Extracting  en\us\callie\base256\8000\informant.wav
48>Extracting  en\us\callie\base256\8000\afflict.wav
48>Extracting  en\us\callie\base256\8000\monument.wav
48>Extracting  en\us\callie\base256\8000\enterprise.wav
48>Extracting  en\us\callie\base256\8000\miser.wav
48>Extracting  en\us\callie\base256\8000\guitarist.wav
48>Extracting  en\us\callie\base256\8000\suspense.wav
48>Extracting  en\us\callie\base256\8000\chatter.wav
48>Extracting  en\us\callie\base256\8000\indigo.wav
48>Extracting  en\us\callie\base256\8000\ammo.wav
48>Extracting  en\us\callie\base256\8000\Bradbury.wav
48>Extracting  en\us\callie\base256\8000\commando.wav
48>Extracting  en\us\callie\base256\8000\certify.wav
48>Extracting  en\us\callie\base256\8000\hockey.wav
48>Extracting  en\us\callie\base256\8000\headwaters.wav
48>Extracting  en\us\callie\base256\8000\unravel.wav
48>Extracting  en\us\callie\base256\8000\bravado.wav
48>Extracting  en\us\callie\base256\8000\armistice.wav
48>Extracting  en\us\callie\base256\8000\liberty.wav
48>Extracting  en\us\callie\base256\8000\treadmill.wav
48>Extracting  en\us\callie\base256\8000\alone.wav
48>Extracting  en\us\callie\base256\8000\Pluto.wav
48>Extracting  en\us\callie\base256\8000\cowbell.wav
48>Extracting  en\us\callie\base256\8000\corrosion.wav
48>Extracting  en\us\callie\base256\8000\newborn.wav
48>Extracting  en\us\callie\base256\8000\Galveston.wav
48>Extracting  en\us\callie\base256\8000\aftermath.wav
48>Extracting  en\us\callie\base256\8000\preshrunk.wav
48>Extracting  en\us\callie\base256\8000\racketeer.wav
48>Extracting  en\us\callie\base256\8000\almighty.wav
48>Extracting  en\us\callie\base256\8000\stupendous.wav
48>Extracting  en\us\callie\base256\8000\equipment.wav
48>Extracting  en\us\callie\base256\8000\sympathy.wav
48>Extracting  en\us\callie\base256\8000\deckhand.wav
48>Extracting  en\us\callie\base256\8000\barbecue.wav
48>Extracting  en\us\callie\base256\8000\Belfast.wav
48>Extracting  en\us\callie\base256\8000\document.wav
48>Extracting  en\us\callie\base256\8000\regain.wav
48>Extracting  en\us\callie\base256\8000\kickoff.wav
48>Extracting  en\us\callie\base256\8000\woodlark.wav
48>Extracting  en\us\callie\base256\8000\checkup.wav
48>Extracting  en\us\callie\base256\8000\acme.wav
48>Extracting  en\us\callie\base256\8000\graduate.wav
48>Extracting  en\us\callie\base256\8000\mural.wav
48>Extracting  en\us\callie\base256\8000\concert.wav
48>Extracting  en\us\callie\base256\8000\caretaker.wav
48>Extracting  en\us\callie\base256\8000\spaniel.wav
48>Extracting  en\us\callie\base256\8000\Atlantic.wav
48>Extracting  en\us\callie\base256\8000\Montana.wav
48>Extracting  en\us\callie\base256\8000\seabird.wav
48>Extracting  en\us\callie\base256\8000\existence.wav
48>Extracting  en\us\callie\base256\8000\clockwork.wav
48>Extracting  en\us\callie\base256\8000\Saturday.wav
48>Extracting  en\us\callie\base256\8000\clamshell.wav
48>Extracting  en\us\callie\base256\8000\quantity.wav
48>Extracting  en\us\callie\base256\8000\blackjack.wav
48>Extracting  en\us\callie\base256\8000\Norwegian.wav
48>Extracting  en\us\callie\base256\8000\handiwork.wav
48>Extracting  en\us\callie\base256\8000\tradition.wav
48>Extracting  en\us\callie\base256\8000\tiger.wav
48>Extracting  en\us\callie\base256\8000\sardonic.wav
48>Extracting  en\us\callie\base256\8000\rebellion.wav
48>Extracting  en\us\callie\base256\8000\slingshot.wav
48>Extracting  en\us\callie\base256\8000\bookseller.wav
48>Extracting  en\us\callie\base256\8000\equation.wav
48>Extracting  en\us\callie\base256\8000\robust.wav
48>Extracting  en\us\callie\base256\8000\Wilmington.wav
48>Extracting  en\us\callie\base256\8000\voyager.wav
48>Extracting  en\us\callie\base256\8000\pharmacy.wav
48>Extracting  en\us\callie\base256\8000\ragtime.wav
48>Extracting  en\us\callie\base256\8000\spyglass.wav
48>Extracting  en\us\callie\base256\8000\edict.wav
48>Extracting  en\us\callie\base256\8000\Jupiter.wav
48>Extracting  en\us\callie\base256\8000\butterfat.wav
48>Extracting  en\us\callie\base256\8000\combustion.wav
48>Extracting  en\us\callie\base256\8000\erase.wav
48>Extracting  en\us\callie\base256\8000\Hamilton.wav
48>Extracting  en\us\callie\base256\8000\atmosphere.wav
48>Extracting  en\us\callie\base256\8000\repellent.wav
48>Extracting  en\us\callie\base256\8000\responsive.wav
48>Extracting  en\us\callie\base256\8000\skullcap.wav
48>Extracting  en\us\callie\base256\8000\involve.wav
48>Extracting  en\us\callie\base256\8000\frequency.wav
48>Extracting  en\us\callie\base256\8000\Pandora.wav
48>Extracting  en\us\callie\base256\8000\tumor.wav
48>Extracting  en\us\callie\base256\8000\Cherokee.wav
48>Extracting  en\us\callie\base256\8000\maverick.wav
48>Extracting  en\us\callie\base256\8000\classic.wav
48>Extracting  en\us\callie\base256\8000\speculate.wav
48>Extracting  en\us\callie\base256\8000\drifter.wav
48>Extracting  en\us\callie\base256\8000\everyday.wav
48>Extracting  en\us\callie\base256\8000\puberty.wav
48>Extracting  en\us\callie\base256\8000\necklace.wav
48>Extracting  en\us\callie\base256\8000\cobra.wav
48>Extracting  en\us\callie\base256\8000\endorse.wav
48>Extracting  en\us\callie\base256\8000\breadline.wav
48>Extracting  en\us\callie\base256\8000\blockade.wav
48>Extracting  en\us\callie\base256\8000\tomorrow.wav
48>Extracting  en\us\callie\base256\8000\pyramid.wav
48>Extracting  en\us\callie\base256\8000\blowtorch.wav
48>Extracting  en\us\callie\base256\8000\button.wav
48>Extracting  en\us\callie\base256\8000\impartial.wav
48>Extracting  en\us\callie\base256\8000\spindle.wav
48>Extracting  en\us\callie\base256\8000\insurgent.wav
48>Extracting  en\us\callie\base256\8000\microscope.wav
48>Extracting  en\us\callie\base256\8000\allow.wav
48>Extracting  en\us\callie\base256\8000\underfoot.wav
48>Extracting  en\us\callie\base256\8000\bottomless.wav
48>Extracting  en\us\callie\base256\8000\vertigo.wav
48>Extracting  en\us\callie\base256\8000\paramount.wav
48>Extracting  en\us\callie\base256\8000\scenic.wav
48>Extracting  en\us\callie\base256\8000\passenger.wav
48>Extracting  en\us\callie\base256\8000\billiard.wav
48>Extracting  en\us\callie\base256\8000\snowcap.wav
48>Extracting  en\us\callie\base256\8000\Burlington.wav
48>Extracting  en\us\callie\base256\8000\alkali.wav
48>Extracting  en\us\callie\base256\8000\quiver.wav
48>Extracting  en\us\callie\base256\8000\borderline.wav
48>Extracting  en\us\callie\base256\8000\integrate.wav
48>Extracting  en\us\callie\base256\8000\stagnate.wav
48>Extracting  en\us\callie\base256\8000\molasses.wav
48>Extracting  en\us\callie\base256\8000\sugar.wav
48>Extracting  en\us\callie\base256\8000\determine.wav
42>cmu_us_rms_cg_params.c
48>Extracting  en\us\callie\base256\8000\goldfish.wav
48>Extracting  en\us\callie\base256\8000\beehive.wav
48>Extracting  en\us\callie\base256\8000\snapline.wav
48>Extracting  en\us\callie\base256\8000\scallion.wav
48>Extracting  en\us\callie\base256\8000\consulting.wav
48>Extracting  en\us\callie\base256\8000\facial.wav
48>Extracting  en\us\callie\base256\8000\Ohio.wav
48>Extracting  en\us\callie\base256\8000\cellulose.wav
48>Extracting  en\us\callie\base256\8000\gazelle.wav
48>Extracting  en\us\callie\base256\8000\flagpole.wav
48>Extracting  en\us\callie\base256\8000\stethoscope.wav
48>Extracting  en\us\callie\base256\8000\scorecard.wav
48>Extracting  en\us\callie\base256\8000\eyetooth.wav
48>Extracting  en\us\callie\base256\8000\resistor.wav
48>Extracting  en\us\callie\base256\8000\absurd.wav
48>Extracting  en\us\callie\base256\8000\bifocals.wav
48>Extracting  en\us\callie\base256\8000\fortitude.wav
48>Extracting  en\us\callie\base256\8000\choking.wav
48>Extracting  en\us\callie\base256\8000\sentence.wav
48>Extracting  en\us\callie\base256\8000\paperweight.wav
48>Extracting  en\us\callie\base256\8000\article.wav
48>Extracting  en\us\callie\base256\8000\Pacific.wav
48>Extracting  en\us\callie\base256\8000\playhouse.wav
48>Extracting  en\us\callie\base256\8000\tonic.wav
48>Extracting  en\us\callie\base256\8000\confidence.wav
48>Extracting  en\us\callie\base256\8000\bombast.wav
48>Extracting  en\us\callie\base256\8000\guidance.wav
48>Extracting  en\us\callie\base256\8000\telephone.wav
48>Extracting  en\us\callie\base256\8000\revenue.wav
48>Extracting  en\us\callie\base256\8000\pandemic.wav
48>Extracting  en\us\callie\base256\8000\October.wav
48>Extracting  en\us\callie\base256\8000\tobacco.wav
48>Extracting  en\us\callie\base256\8000\uproot.wav
48>Extracting  en\us\callie\base256\8000\ribcage.wav
48>Extracting  en\us\callie\base256\8000\spigot.wav
48>Extracting  en\us\callie\base256\8000\cranky.wav
48>Extracting  en\us\callie\base256\8000\inventive.wav
48>Extracting  en\us\callie\base256\8000\Babylon.wav
48>Extracting  en\us\callie\base256\8000\caravan.wav
48>Extracting  en\us\callie\base256\8000\Camelot.wav
48>Extracting  en\us\callie\base256\8000\snowslide.wav
48>Extracting  en\us\callie\base256\8000\adroitness.wav
48>Extracting  en\us\callie\base256\8000\soybean.wav
48>Extracting  en\us\callie\base256\8000\amusement.wav
48>Extracting  en\us\callie\base256\8000\Dupont.wav
48>Extracting  en\us\callie\base256\8000\gravity.wav
48>Extracting  en\us\callie\base256\8000\solo.wav
48>Extracting  en\us\callie\base256\8000\crucial.wav
48>Extracting  en\us\callie\base256\8000\finicky.wav
48>Extracting  en\us\callie\base256\8000\dogsled.wav
48>Extracting  en\us\callie\base256\8000\outfielder.wav
48>Extracting  en\us\callie\base256\8000\adrift.wav
48>Extracting  en\us\callie\base256\8000\narrative.wav
48>Extracting  en\us\callie\base256\8000\Geiger.wav
48>Extracting  en\us\callie\base256\8000\December.wav
48>Extracting  en\us\callie\base256\8000\glitter.wav
48>Extracting  en\us\callie\base256\8000\visitor.wav
48>Extracting  en\us\callie\base256\8000\unify.wav
48>Extracting  en\us\callie\base256\8000\dashboard.wav
48>Extracting  en\us\callie\base256\8000\upcoming.wav
48>Extracting  en\us\callie\base256\8000\bluebird.wav
48>Extracting  en\us\callie\base256\8000\peachy.wav
48>Extracting  en\us\callie\base256\8000\misnomer.wav
48>Extracting  en\us\callie\base256\8000\hydraulic.wav
48>Extracting  en\us\callie\base256\8000\decimal.wav
48>Extracting  en\us\callie\base256\8000\stapler.wav
48>Extracting  en\us\callie\base256\8000\uncut.wav
48>Extracting  en\us\callie\base256\8000\replica.wav
48>Extracting  en\us\callie\base256\8000\crackdown.wav
48>Extracting  en\us\callie\base256\8000\pioneer.wav
48>Extracting  en\us\callie\base256\8000\escape.wav
48>Extracting  en\us\callie\base256\8000\stockman.wav
48>Extracting  en\us\callie\base256\8000\disruptive.wav
48>Extracting  en\us\callie\base256\8000\reindeer.wav
48>Extracting  en\us\callie\base256\8000\ratchet.wav
48>Extracting  en\us\callie\base256\8000\miracle.wav
48>Extracting  en\us\callie\base256\8000\framework.wav
48>Extracting  en\us\callie\base256\8000\rematch.wav
48>Extracting  en\us\callie\base256\8000\pheasant.wav
48>Extracting  en\us\callie\base256\8000\unicorn.wav
48>Extracting  en\us\callie\base256\8000\chisel.wav
48>Extracting  en\us\callie\base256\8000\Pegasus.wav
48>Extracting  en\us\callie\base256\8000\corporate.wav
48>Extracting  en\us\callie\base256\8000\shamrock.wav
48>Extracting  en\us\callie\base256\8000\leprosy.wav
48>Extracting  en\us\callie\base256\8000\drunken.wav
48>Extracting  en\us\callie\base256\8000\reward.wav
48>Extracting  en\us\callie\base256\8000\enrollment.wav
48>Extracting  en\us\callie\base256\8000\drainage.wav
48>Extracting  en\us\callie\base256\8000\processor.wav
48>Extracting  en\us\callie\base256\8000\fascinate.wav
48>Extracting  en\us\callie\base256\8000\obtuse.wav
48>Extracting  en\us\callie\base256\8000\buzzard.wav
48>Extracting  en\us\callie\base256\8000\hideaway.wav
48>Extracting  en\us\callie\base256\8000\vacancy.wav
48>Extracting  en\us\callie\base256\8000\photograph.wav
48>Extracting  en\us\callie\base256\8000\sterling.wav
48>Extracting  en\us\callie\base256\8000\Christmas.wav
48>Extracting  en\us\callie\base256\8000\inertia.wav
48>Extracting  en\us\callie\base256\8000\dragnet.wav
48>Extracting  en\us\callie\base256\8000\proximate.wav
48>Extracting  en\us\callie\base256\8000\goggles.wav
48>Extracting  en\us\callie\base256\8000\distortion.wav
48>Extracting  en\us\callie\base256\8000\trauma.wav
48>Extracting  en\us\callie\base256\8000\asteroid.wav
48>Extracting  en\us\callie\base256\8000\rocker.wav
48>Extracting  en\us\callie\base256\8000\Istanbul.wav
48>Extracting  en\us\callie\base256\8000\tapeworm.wav
48>Extracting  en\us\callie\base256\8000\spearhead.wav
48>Extracting  en\us\callie\base256\8000\backward.wav
48>Extracting  en\us\callie\base256\8000\reform.wav
48>Extracting  en\us\callie\base256\8000\inverse.wav
48>Extracting  en\us\callie\base256\8000\retrospect.wav
48>Extracting  en\us\callie\base256\8000\disable.wav
48>Extracting  en\us\callie\base256\8000\endow.wav
48>Extracting  en\us\callie\base256\8000\wallet.wav
48>Extracting  en\us\callie\base256\8000\Algol.wav
48>Extracting  en\us\callie\base256\8000\crucifix.wav
48>Extracting  en\us\callie\base256\8000\letterhead.wav
48>Extracting  en\us\callie\base256\8000\ruffled.wav
48>Extracting  en\us\callie\base256\8000\Burbank.wav
48>Extracting  en\us\callie\base256\8000\freedom.wav
48>Extracting  en\us\callie\base256\8000\island.wav
48>Extracting  en\us\callie\base256\8000\gremlin.wav
48>Extracting  en\us\callie\base256\8000\disbelief.wav
48>Extracting  en\us\callie\base256\8000\inferno.wav
48>Extracting  en\us\callie\base256\8000\suspicious.wav
48>Extracting  en\us\callie\voicemail
48>Extracting  en\us\callie\voicemail\8000
48>Extracting  en\us\callie\voicemail\8000\vm-record_message.wav
48>Extracting  en\us\callie\voicemail\8000\vm-new.wav
48>Extracting  en\us\callie\voicemail\8000\vm-delete_recording.wav
48>Extracting  en\us\callie\voicemail\8000\vm-abort.wav
48>Extracting  en\us\callie\voicemail\8000\vm-record_name1.wav
48>Extracting  en\us\callie\voicemail\8000\vm-you_have.wav
48>Extracting  en\us\callie\voicemail\8000\vm-marked-urgent.wav
48>Extracting  en\us\callie\voicemail\8000\vm-urgent-new.wav
48>Extracting  en\us\callie\voicemail\8000\vm-to_forward.wav
48>Extracting  en\us\callie\voicemail\8000\vm-urgent.wav
48>Extracting  en\us\callie\voicemail\8000\vm-greeting.wav
48>Extracting  en\us\callie\voicemail\8000\vm-record_greeting.wav
48>Extracting  en\us\callie\voicemail\8000\vm-send_message_now.wav
48>Extracting  en\us\callie\voicemail\8000\vm-enter_pass.wav
48>Extracting  en\us\callie\voicemail\8000\vm-followed_by.wav
48>Extracting  en\us\callie\voicemail\8000\vm-play_previous_message.wav
48>Extracting  en\us\callie\voicemail\8000\vm-not_available.wav
48>Extracting  en\us\callie\voicemail\8000\vm-rerecord.wav
48>Extracting  en\us\callie\voicemail\8000\vm-mailbox_full.wav
48>Extracting  en\us\callie\voicemail\8000\vm-from.wav
48>Extracting  en\us\callie\voicemail\8000\vm-message_envelope.wav
48>Extracting  en\us\callie\voicemail\8000\vm-record_name2.wav
48>Extracting  en\us\callie\voicemail\8000\vm-enter_id.wav
48>Extracting  en\us\callie\voicemail\8000\vm-deleted.wav
48>Extracting  en\us\callie\voicemail\8000\vm-listen_to_recording_again.wav
48>Extracting  en\us\callie\voicemail\8000\vm-fail_auth.wav
48>Extracting  en\us\callie\voicemail\8000\vm-marked_new.wav
48>Extracting  en\us\callie\voicemail\8000\vm-listen_saved.wav
48>Extracting  en\us\callie\voicemail\8000\vm-return_call.wav
42>cmu_us_rms_cg_phonestate.c
42>au_command.c
48>Extracting  en\us\callie\voicemail\8000\vm-person.wav
48>Extracting  en\us\callie\voicemail\8000\vm-to_exit_alt.wav
48>Extracting  en\us\callie\voicemail\8000\vm-listen_to_recording.wav
48>Extracting  en\us\callie\voicemail\8000\vm-in_folder.wav
48>Extracting  en\us\callie\voicemail\8000\vm-hello.wav
48>Extracting  en\us\callie\voicemail\8000\vm-next.wav
48>Extracting  en\us\callie\voicemail\8000\vm-emailed.wav
48>Extracting  en\us\callie\voicemail\8000\vm-forward_to_email.wav
48>Extracting  en\us\callie\voicemail\8000\vm-choose_greeting_fail.wav
48>Extracting  en\us\callie\voicemail\8000\vm-continue.wav
48>Extracting  en\us\callie\voicemail\8000\vm-main_menu_alt.wav
48>Extracting  en\us\callie\voicemail\8000\vm-save_recording.wav
48>Extracting  en\us\callie\voicemail\8000\vm-undeleted.wav
48>Extracting  en\us\callie\voicemail\8000\vm-urgent-saved.wav
48>Extracting  en\us\callie\voicemail\8000\vm-advanced.wav
48>Extracting  en\us\callie\voicemail\8000\vm-advanced_alt.wav
48>Extracting  en\us\callie\voicemail\8000\vm-goodbye.wav
48>Extracting  en\us\callie\voicemail\8000\vm-messages_alt.wav
48>Extracting  en\us\callie\voicemail\8000\vm-forward_enter_ext.wav
48>Extracting  en\us\callie\voicemail\8000\vm-listen_new.wav
48>Extracting  en\us\callie\voicemail\8000\vm-press.wav
48>Extracting  en\us\callie\voicemail\8000\vm-received.wav
48>Extracting  en\us\callie\voicemail\8000\vm-message_number.wav
48>Extracting  en\us\callie\voicemail\8000\vm-last.wav
48>Extracting  en\us\callie\voicemail\8000\vm-choose_greeting.wav
48>Extracting  en\us\callie\voicemail\8000\vm-has_been_changed_to.wav
48>Extracting  en\us\callie\voicemail\8000\vm-to_exit.wav
48>Extracting  en\us\callie\voicemail\8000\vm-too-small.wav
48>Extracting  en\us\callie\voicemail\8000\vm-delete_message.wav
48>Extracting  en\us\callie\voicemail\8000\vm-save_message.wav
48>Extracting  en\us\callie\voicemail\8000\vm-mark-urgent.wav
48>Extracting  en\us\callie\voicemail\8000\vm-repeat_message.wav
48>Extracting  en\us\callie\voicemail\8000\vm-mark_message_new.wav
48>Extracting  en\us\callie\voicemail\8000\vm-that_was_an_invalid_ext.wav
48>Extracting  en\us\callie\voicemail\8000\vm-message.wav
48>Extracting  en\us\callie\voicemail\8000\vm-followed_by_pound.wav
48>Extracting  en\us\callie\voicemail\8000\vm-saved.wav
48>Extracting  en\us\callie\voicemail\8000\vm-no_more_messages.wav
48>Extracting  en\us\callie\voicemail\8000\vm-choose_greeting_choose.wav
48>Extracting  en\us\callie\voicemail\8000\vm-change_password.wav
48>Extracting  en\us\callie\voicemail\8000\vm-messages.wav
48>Extracting  en\us\callie\voicemail\8000\vm-message_alt.wav
48>Extracting  en\us\callie\voicemail\8000\vm-to_record_greeting.wav
48>Extracting  en\us\callie\voicemail\8000\vm-play_greeting.wav
48>Extracting  en\us\callie\voicemail\8000\vm-play_next_message.wav
48>Extracting  en\us\callie\voicemail\8000\vm-main_menu.wav
48>Extracting  en\us\callie\voicemail\8000\vm-undelete_message.wav
48>Extracting  en\us\callie\voicemail\8000\vm-selected.wav
48>Extracting  en\us\callie\voicemail\8000\vm-forward_add_intro.wav
48>Extracting  en\us\callie\ascii
48>Extracting  en\us\callie\ascii\8000
48>Extracting  en\us\callie\ascii\8000\107.wav
48>Extracting  en\us\callie\ascii\8000\98.wav
48>Extracting  en\us\callie\ascii\8000\35.wav
48>Extracting  en\us\callie\ascii\8000\97.wav
48>Extracting  en\us\callie\ascii\8000\118.wav
48>Extracting  en\us\callie\ascii\8000\102.wav
48>Extracting  en\us\callie\ascii\8000\122.wav
48>Extracting  en\us\callie\ascii\8000\99.wav
48>Extracting  en\us\callie\ascii\8000\103.wav
48>Extracting  en\us\callie\ascii\8000\115.wav
48>Extracting  en\us\callie\ascii\8000\113.wav
48>Extracting  en\us\callie\ascii\8000\105.wav
48>Extracting  en\us\callie\ascii\8000\32.wav
48>Extracting  en\us\callie\ascii\8000\104.wav
48>Extracting  en\us\callie\ascii\8000\117.wav
48>Extracting  en\us\callie\ascii\8000\106.wav
48>Extracting  en\us\callie\ascii\8000\112.wav
48>Extracting  en\us\callie\ascii\8000\108.wav
48>Extracting  en\us\callie\ascii\8000\109.wav
48>Extracting  en\us\callie\ascii\8000\120.wav
48>Extracting  en\us\callie\ascii\8000\42.wav
48>Extracting  en\us\callie\ascii\8000\101.wav
48>Extracting  en\us\callie\ascii\8000\110.wav
48>Extracting  en\us\callie\ascii\8000\46.wav
48>Extracting  en\us\callie\ascii\8000\114.wav
42>Generating Code...
48>Extracting  en\us\callie\ascii\8000\111.wav
48>Extracting  en\us\callie\ascii\8000\100.wav
48>Extracting  en\us\callie\ascii\8000\119.wav
48>Extracting  en\us\callie\ascii\8000\121.wav
48>Extracting  en\us\callie\ascii\8000\116.wav
48>Extracting  en\us\callie\misc
48>Extracting  en\us\callie\misc\8000
48>Extracting  en\us\callie\misc\8000\phone_not_auth.wav
48>Extracting  en\us\callie\misc\8000\en.wav
48>Extracting  en\us\callie\misc\8000\sorry.wav
48>Extracting  en\us\callie\misc\8000\transfer2.wav
48>Extracting  en\us\callie\misc\8000\followed.wav
48>Extracting  en\us\callie\misc\8000\invalid_extension.wav
48>Extracting  en\us\callie\misc\8000\if_you_would_like_to.wav
48>Extracting  en\us\callie\misc\8000\error.wav
48>Extracting  en\us\callie\misc\8000\misc-your_call_has_been_terminated.wav
48>Extracting  en\us\callie\misc\8000\call_secured.wav
48>Extracting  en\us\callie\misc\8000\provide_reference_number.wav
48>Extracting  en\us\callie\misc\8000\es.wav
48>Extracting  en\us\callie\misc\8000\transfer1.wav
48>Extracting  en\us\callie\misc\8000\misc-your_call_will_be_terminated_in.wav
48>Extracting  en\us\callie\misc\8000\if_you_are_this_person.wav
48>Extracting  en\us\callie\misc\8000\we_are_trying_to_reach.wav
48>Extracting  en\us\callie\misc\8000\call_monitoring_blurb.wav
48>Extracting  en\us\callie\zrtp
48>Extracting  en\us\callie\zrtp\8000
48>Extracting  en\us\callie\zrtp\8000\zrtp-is_unverified.wav
48>Extracting  en\us\callie\zrtp\8000\zrtp-enroll_welcome.wav
48>Extracting  en\us\callie\zrtp\8000\zrtp-status_notsecure.wav
48>Extracting  en\us\callie\zrtp\8000\zrtp-enroll_already_enrolled.wav
48>Extracting  en\us\callie\zrtp\8000\zrtp-status_error.wav
48>Extracting  en\us\callie\zrtp\8000\zrtp-is_verified.wav
48>Extracting  en\us\callie\zrtp\8000\zrtp-enroll_notzrtp.wav
48>Extracting  en\us\callie\zrtp\8000\zrtp-check_sas.wav
48>Extracting  en\us\callie\zrtp\8000\zrtp-enroll_confirmed.wav
48>Extracting  en\us\callie\zrtp\8000\zrtp-is_secure.wav
48>Extracting  en\us\callie\zrtp\8000\zrtp-enroll_not_sip_registered.wav
48>Extracting  en\us\callie\zrtp\8000\zrtp-thankyou_goodbye.wav
48>Extracting  en\us\callie\zrtp\8000\zrtp-status_securing.wav
48>Extracting  en\us\callie\zrtp\8000\zrtp-status_secure.wav
48>Extracting  en\us\callie\zrtp\8000\zrtp-somethings_wrong.wav
48>Extracting  en\us\callie\conference
48>Extracting  en\us\callie\conference\8000
48>Extracting  en\us\callie\conference\8000\conf-kicked.wav
48>Extracting  en\us\callie\conference\8000\conf-locked.wav
48>Extracting  en\us\callie\conference\8000\conf-is-unlocked.wav
48>Extracting  en\us\callie\conference\8000\conf-welcome.wav
48>Extracting  en\us\callie\conference\8000\conf-unmuted.wav
48>Extracting  en\us\callie\conference\8000\conf-is-locked.wav
48>Extracting  en\us\callie\conference\8000\conf-muted.wav
48>Extracting  en\us\callie\conference\8000\conf-alone.wav
48>Extracting  en\us\callie\conference\8000\conf-bad-pin.wav
48>Extracting  en\us\callie\conference\8000\conf-pin.wav
48>Extracting  en\us\callie\conference\8000\conf-goodbye.wav
48>Extracting  en\us\callie\phonetic-ascii
48>Extracting  en\us\callie\phonetic-ascii\8000
48>Extracting  en\us\callie\phonetic-ascii\8000\107.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\98.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\35.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\97.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\118.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\102.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\122.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\99.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\103.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\115.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\113.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\105.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\32.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\104.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\117.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\106.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\112.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\108.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\109.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\120.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\42.wav
42>Compiling...
48>Extracting  en\us\callie\phonetic-ascii\8000\101.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\110.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\46.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\114.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\111.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\100.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\119.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\121.wav
48>Extracting  en\us\callie\phonetic-ascii\8000\116.wav
48>Extracting  en\us\callie\ivr
48>Extracting  en\us\callie\ivr\8000
48>Extracting  en\us\callie\ivr\8000\ivr-enter_ext_pound.wav
48>Extracting  en\us\callie\ivr\8000\ivr-account_number.wav
48>Extracting  en\us\callie\ivr\8000\ivr-to_repeat_these_options.wav
48>Extracting  en\us\callie\ivr\8000\ivr-press_one_q_or_z.wav
48>Extracting  en\us\callie\ivr\8000\ivr-to_do_a_fwd_echo_test.wav
48>Extracting  en\us\callie\ivr\8000\ivr-take_a_message.wav
48>Extracting  en\us\callie\ivr\8000\ivr-please_reenter_your_pin.wav
48>Extracting  en\us\callie\ivr\8000\ivr-this_is_a_call_from.wav
48>Extracting  en\us\callie\ivr\8000\ivr-to_do_a_freeswitch_echo_test.wav
48>Extracting  en\us\callie\ivr\8000\ivr-enter_ext.wav
48>Extracting  en\us\callie\ivr\8000\ivr-last_name_first.wav
48>Extracting  en\us\callie\ivr\8000\ivr-to_log_in.wav
48>Extracting  en\us\callie\ivr\8000\ivr-that_was_an_invalid_entry.wav
48>Extracting  en\us\callie\ivr\8000\ivr-please_try_again.wav
48>Extracting  en\us\callie\ivr\8000\ivr-connect_to_caller.wav
48>Extracting  en\us\callie\ivr\8000\ivr-first_name_first.wav
48>Extracting  en\us\callie\ivr\8000\ivr-send_to_voicemail.wav
48>Extracting  en\us\callie\ivr\8000\ivr-you_have_dialed_an_invalid_extension.wav
48>Extracting  en\us\callie\ivr\8000\ivr-to_log_out.wav
48>Extracting  en\us\callie\ivr\8000\ivr-pin_or_extension_is-invalid.wav
48>Extracting  en\us\callie\ivr\8000\ivr-hold_connect_call.wav
48>Extracting  en\us\callie\ivr\8000\ivr-please_enter_pin_followed_by_pound.wav
48>Extracting  en\us\callie\ivr\8000\ivr-sales.wav
48>Extracting  en\us\callie\ivr\8000\ivr-hello.wav
48>Extracting  en\us\callie\ivr\8000\ivr-speak_to_a_customer_service_representative.wav
48>Extracting  en\us\callie\ivr\8000\ivr-this_phone_will_now_reboot.wav
48>Extracting  en\us\callie\ivr\8000\ivr-you_are_now_logged_in.wav
48>Extracting  en\us\callie\ivr\8000\ivr-spell_name.wav
48>Extracting  en\us\callie\ivr\8000\ivr-you_may.wav
48>Extracting  en\us\callie\ivr\8000\ivr-save_review_record.wav
48>Extracting  en\us\callie\ivr\8000\ivr-im_sorry.wav
48>Extracting  en\us\callie\ivr\8000\ivr-you_may_exit_by_hanging_up.wav
48>Extracting  en\us\callie\ivr\8000\ivr-welcome_to_freeswitch.wav
48>Extracting  en\us\callie\ivr\8000\ivr-to_listen_to_moh.wav
48>Extracting  en\us\callie\ivr\8000\ivr-please_return_our_call_at.wav
48>Extracting  en\us\callie\ivr\8000\ivr-use_telephone_keypad.wav
48>Extracting  en\us\callie\ivr\8000\ivr-to_hear_screaming_monkeys.wav
48>Extracting  en\us\callie\ivr\8000\ivr-not.wav
48>Extracting  en\us\callie\ivr\8000\ivr-call.wav
48>Extracting  en\us\callie\ivr\8000\ivr-sample_submenu.wav
48>Extracting  en\us\callie\ivr\8000\ivr-to_call_the_freeswitch_conference.wav
48>Extracting  en\us\callie\ivr\8000\ivr-operator.wav
48>Extracting  en\us\callie\ivr\8000\ivr-extension_to_provision_this_phone.wav
48>Extracting  en\us\callie\ivr\8000\ivr-to_speak_with_an_operator.wav
48>Extracting  en\us\callie\ivr\8000\ivr-for_this_person.wav
48>Extracting  en\us\callie\ivr\8000\ivr-this_ivr_will_let_you_test_features.wav
48>Extracting  en\us\callie\ivr\8000\ivr-please.wav
48>Extracting  en\us\callie\ivr\8000\ivr-register_for_cluecon.wav
48>Extracting  en\us\callie\ivr\8000\ivr-unable_save.wav
48>Extracting  en\us\callie\ivr\8000\ivr-regarding_reference_number.wav
48>Extracting  en\us\callie\ivr\8000\ivr-provision_phone_permanently_to_extension.wav
48>Extracting  en\us\callie\ivr\8000\ivr-to_return_to_previous_menu.wav
48>Extracting  en\us\callie\ivr\8000\ivr-technical_support.wav
48>Extracting  en\us\callie\ivr\8000\ivr-you_are_now_logged_out.wav
48>Extracting  en\us\callie\ivr\8000\ivr-customer_service.wav
48>Extracting  en\us\callie\ivr\8000\ivr-say_name.wav
48>Extracting  en\us\callie\ivr\8000\ivr-please_enter_extension_followed_by_pound.wav
48>Extracting  en\us\callie\ivr\8000\ivr-you_are_about_to_provision_this_phone.wav
48>Extracting  en\us\callie\ivr\8000\ivr-recording_saved.wav
48>Extracting  en\us\callie\ivr\8000\ivr-to_hear_sample_submenu.wav
48>Extracting  en\us\callie\ivr\8000\ivr-or.wav
48>Extracting  en\us\callie\ivr\8000\ivr-thank_you.wav
48>Everything is Ok
48>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Debug\BuildLog Download 8khzsound.htm"
48>Download 8khzsound - 0 error(s), 0 warning(s)
49>------ Build started: Project: abyss, Configuration: Debug Win32 ------
49>Compiling...
42>cmu_time_awb.c
49>chanswitch.c
42>c:\freeswitch\freeswitch-1.0.6\libs\flite-1.3.99\lang\cmu_time_awb\cmu_time_awb.c(81) : warning C4090: '=' : different 'const' qualifiers
42>c:\freeswitch\freeswitch-1.0.6\libs\flite-1.3.99\lang\cmu_time_awb\cmu_time_awb.c(82) : warning C4090: '=' : different 'const' qualifiers
49>conf.c
42>Creating library...
49>conn.c
42>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\flite\Debug\BuildLog.htm"
49>data.c
42>flite - 0 error(s), 3 warning(s)
49>date.c
50>------ Build started: Project: pocketsphinx, Configuration: Debug Win32 ------
49>file.c
50>Compiling...
50>acmod.c
49>handler.c
50>bin_mdef.c
49>http.c
50>blkarray_list.c
49>init.c
50>cmu6_lts_rules.c
50>dict2pid.c
49>response.c
50>fillpen.c
49>server.c
50>fsg_history.c
49>session.c
49>socket.c
50>fsg_lextree.c
49>socket_win.c
50>fsg_search.c
49>thread_windows.c
50>hmm.c
50>kdtree.c
49>token.c
50>lextree.c
49>trace.c
49>channel.c
50>mdef.c
49>Generating Code...
50>ms_gauden.c
49>Creating library...
49>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\xmlrpc-c\Windows\Debug\abyss\BuildLog.htm"
49>abyss - 0 error(s), 0 warning(s)
50>ms_mgau.c
51>------ Build started: Project: xmlrpc, Configuration: Debug Win32 ------
51>Compiling...
51>double.c
51>error.c
50>ms_senone.c
51>make_printable.c
51>memblock.c
51>method.c
50>ngram_search.c
51>parse_value.c
51>pthreadx_win32.c
51>registry.c
50>ngram_search_fwdflat.c
51>resource.c
51>select.c
50>ngram_search_fwdtree.c
51>sleep.c
51>system_method.c
51>time.c
50>phone_loop_search.c
51>trace.c
50>Generating Code...
51>utf8.c
51>version.c
51>xmlrpc_array.c
51>xmlrpc_authcookie.c
50>Compiling...
50>pocketsphinx.c
51>xmlrpc_base64.c
51>xmlrpc_build.c
51>Generating Code...
51>Compiling...
51>xmlrpc_client.c
51>xmlrpc_client_global.c
51>xmlrpc_data.c
50>ps_lattice.c
51>xmlrpc_datetime.c
51>xmlrpc_decompose.c
50>ps_mllr.c
51>xmlrpc_expat.c
51>xmlrpc_parse.c
51>xmlrpc_serialize.c
51>xmlrpc_server_abyss.c
50>s2_semi_mgau.c
50>s3dict.c
51>xmlrpc_server_info.c
51>xmlrpc_string.c
50>tmat.c
51>xmlrpc_struct.c
51>xmlrpc_wininet_transport.c
50>tst_search.c
51>asprintf.c
51>Generating Code...
51>Creating library...
50>vector.c
51>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\xmlrpc-c\Windows\Debug\xmlrpc\BuildLog.htm"
51>xmlrpc - 0 error(s), 0 warning(s)
50>vithist.c
50>Generating Code...
52>------ Build started: Project: xmlparse, Configuration: Debug Win32 ------
52>Compiling...
52>xmlparse.c
50>Compiling manifest to resources...
50>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
50>Copyright (C) Microsoft Corporation.  All rights reserved.
50>Linking...
50>   Creating library C:\FreeSWITCH\freeswitch-1.0.6\Debug\pocketsphinx.lib and object C:\FreeSWITCH\freeswitch-1.0.6\Debug\pocketsphinx.exp
52>Creating library...
52>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\xmlrpc-c\Windows\Debug\xmlparse\BuildLog.htm"
52>xmlparse - 0 error(s), 0 warning(s)
50>Embedding manifest...
53>------ Build started: Project: libspandsp, Configuration: Debug Win32 ------
53>Copying c:\FreeSWITCH\freeswitch-1.0.6\libs\spandsp\src\msvc\spandsp.h to c:\FreeSWITCH\freeswitch-1.0.6\libs\spandsp\src\spandsp.h
53>        1 file(s) copied.
53>Compiling...
50>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
50>Copyright (C) Microsoft Corporation.  All rights reserved.
53>adsi.c
50>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\pocketsphinx\Debug\BuildLog.htm"
50>pocketsphinx - 0 error(s), 0 warning(s)
53>async.c
53>at_interpreter.c
53>awgn.c
53>bell_r2_mf.c
53>bert.c
53>bit_operations.c
54>------ Build started: Project: xmltok, Configuration: Debug Win32 ------
54>Compiling...
54>xmltok.c
53>bitstream.c
54>xmlrole.c
53>complex_filters.c
53>complex_vector_float.c
54>Generating Code...
53>complex_vector_int.c
54>Creating library...
53>crc.c
54>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\xmlrpc-c\Windows\Debug\xmltok\BuildLog.htm"
54>xmltok - 0 error(s), 0 warning(s)
53>dds_float.c
53>dds_int.c
53>dtmf.c
53>echo.c
53>fax.c
53>fax_modems.c
55>------ Build started: Project: mod_lcr, Configuration: Debug Win32 ------
55>Compiling...
55>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
55>mod_lcr.c
53>fsk.c
53>g711.c
53>Generating Code...
53>Compiling...
53>g722.c
55>Linking...
55>   Creating library Win32\Debug/mod_lcr.2008.lib and object Win32\Debug/mod_lcr.2008.exp
55>Embedding manifest...
53>g726.c
53>gsm0610_decode.c
53>gsm0610_encode.c
53>gsm0610_long_term.c
55>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_lcr\Win32\Debug\BuildLog.htm"
55>mod_lcr - 0 error(s), 1 warning(s)
53>gsm0610_lpc.c
53>gsm0610_preprocess.c
56>------ Build started: Project: mod_easyroute, Configuration: Debug Win32 ------
56>Compiling...
56>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
53>gsm0610_rpe.c
53>gsm0610_short_term.c
53>hdlc.c
56>mod_easyroute.c
53>ima_adpcm.c
53>logging.c
53>lpc10_analyse.c
56>Linking...
56>   Creating library Win32\Debug/mod_easyroute.2008.lib and object Win32\Debug/mod_easyroute.2008.exp
53>lpc10_decode.c
56>Embedding manifest...
53>lpc10_encode.c
53>lpc10_placev.c
56>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_easyroute\Win32\Debug\BuildLog.htm"
56>mod_easyroute - 0 error(s), 1 warning(s)
53>lpc10_voicing.c
57>------ Build started: Project: mod_pocketsphinx, Configuration: Debug Win32 ------
57>Compiling...
57>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
57>mod_pocketsphinx.c
53>modem_echo.c
53>modem_connect_tones.c
53>noise.c
53>Generating Code...
57>Compiling manifest to resources...
57>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
57>Copyright (C) Microsoft Corporation.  All rights reserved.
57>Linking...
57>   Creating library Win32\Debug/mod_pocketsphinx.2008.lib and object Win32\Debug/mod_pocketsphinx.2008.exp
57>Embedding manifest...
53>Compiling...
53>oki_adpcm.c
57>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
57>Copyright (C) Microsoft Corporation.  All rights reserved.
53>playout.c
57>Performing Post-Build Event...
53>plc.c
53>power_meter.c
53>queue.c
53>schedule.c
53>sig_tone.c
57>C:\FreeSWITCH\freeswitch-1.0.6\libs\Communicator_semi_40.cd_semi_6000\COPYING
57>C:\FreeSWITCH\freeswitch-1.0.6\libs\Communicator_semi_40.cd_semi_6000\feat.params
57>C:\FreeSWITCH\freeswitch-1.0.6\libs\Communicator_semi_40.cd_semi_6000\mdef
57>C:\FreeSWITCH\freeswitch-1.0.6\libs\Communicator_semi_40.cd_semi_6000\means
57>C:\FreeSWITCH\freeswitch-1.0.6\libs\Communicator_semi_40.cd_semi_6000\noisedict
57>C:\FreeSWITCH\freeswitch-1.0.6\libs\Communicator_semi_40.cd_semi_6000\sendump
57>C:\FreeSWITCH\freeswitch-1.0.6\libs\Communicator_semi_40.cd_semi_6000\transition_matrices
57>C:\FreeSWITCH\freeswitch-1.0.6\libs\Communicator_semi_40.cd_semi_6000\variances
57>8 File(s) copied
53>silence_gen.c
57>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\asr_tts\mod_pocketsphinx\Win32\Debug\BuildLog.htm"
57>mod_pocketsphinx - 0 error(s), 1 warning(s)
53>super_tone_rx.c
53>super_tone_tx.c
53>swept_tone.c
53>t4_rx.c
58>------ Build started: Project: mod_lua, Configuration: Debug Win32 ------
58>Compiling...
53>t4_tx.c
58>mod_lua.cpp
53>t30.c
53>t30_api.c
53>t30_logging.c
53>t31.c
53>t35.c
58>freeswitch_lua.cpp
53>t38_core.c
53>t38_gateway.c
53>Generating Code...
58>Generating Code...
53>Compiling...
53>t38_non_ecm_buffer.c
58>Compiling...
58>mod_lua_wrap.cpp
53>t38_terminal.c
53>testcpuid.c
53>time_scale.c
53>tone_detect.c
53>tone_generate.c
53>v17rx.c
53>v17tx.c
53>v18.c
58>Linking...
53>v22bis_rx.c
58>   Creating library Win32\Debug/mod_lua.2008.lib and object Win32\Debug/mod_lua.2008.exp
53>v22bis_tx.c
53>v27ter_rx.c
58>Embedding manifest...
53>v27ter_tx.c
53>v29rx.c
58>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_lua\Win32\Debug\BuildLog.htm"
58>mod_lua - 0 error(s), 0 warning(s)
59>------ Build started: Project: 8khz, Configuration: Debug Win32 ------
59>Performing Post-Build Event...
53>v29tx.c
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-abort.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-advanced.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-advanced_alt.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-change_password.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-choose_greeting.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-choose_greeting_choose.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-choose_greeting_fail.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-continue.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-deleted.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-delete_message.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-delete_recording.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-emailed.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-enter_id.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-enter_pass.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-fail_auth.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-followed_by.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-followed_by_pound.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-forward_add_intro.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-forward_enter_ext.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-forward_to_email.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-from.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-goodbye.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-greeting.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-has_been_changed_to.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-hello.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-in_folder.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-last.wav
53>v42.c
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-listen_new.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-listen_saved.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-listen_to_recording.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-listen_to_recording_again.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-mailbox_full.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-main_menu.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-main_menu_alt.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-mark-urgent.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-marked-urgent.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-marked_new.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-mark_message_new.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-message.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-messages.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-messages_alt.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-message_alt.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-message_envelope.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-message_number.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-new.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-next.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-not_available.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-no_more_messages.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-person.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-play_greeting.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-play_next_message.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-play_previous_message.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-press.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-received.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-record_greeting.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-record_message.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-record_name1.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-record_name2.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-repeat_message.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-rerecord.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-return_call.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-saved.wav
53>v42bis.c
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-save_message.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-save_recording.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-selected.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-send_message_now.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-that_was_an_invalid_ext.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-too-small.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-to_exit.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-to_exit_alt.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-to_forward.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-to_record_greeting.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-undeleted.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-undelete_message.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-urgent-new.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-urgent-saved.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-urgent.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\voicemail\8000\vm-you_have.wav
59>78 File(s) copied
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-account_number.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-call.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-connect_to_caller.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-customer_service.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-enter_ext.wav
53>v8.c
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-enter_ext_pound.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-extension_to_provision_this_phone.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-first_name_first.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-for_this_person.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-hello.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-hold_connect_call.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-im_sorry.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-last_name_first.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-not.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-operator.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-or.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-pin_or_extension_is-invalid.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-please.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-please_enter_extension_followed_by_pound.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-please_enter_pin_followed_by_pound.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-please_reenter_your_pin.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-please_return_our_call_at.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-please_try_again.wav
53>vector_float.c
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-press_one_q_or_z.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-provision_phone_permanently_to_extension.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-recording_saved.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-regarding_reference_number.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-register_for_cluecon.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-sales.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-sample_submenu.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-save_review_record.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-say_name.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-send_to_voicemail.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-speak_to_a_customer_service_representative.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-spell_name.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-take_a_message.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-technical_support.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-thank_you.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-that_was_an_invalid_entry.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-this_is_a_call_from.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-this_ivr_will_let_you_test_features.wav
53>vector_int.c
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-this_phone_will_now_reboot.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_call_the_freeswitch_conference.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_do_a_freeswitch_echo_test.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_do_a_fwd_echo_test.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_hear_sample_submenu.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_hear_screaming_monkeys.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_listen_to_moh.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_log_in.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_log_out.wav
53>Generating Code...
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_repeat_these_options.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_return_to_previous_menu.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-to_speak_with_an_operator.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-unable_save.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-use_telephone_keypad.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-welcome_to_freeswitch.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-you_are_about_to_provision_this_phone.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-you_are_now_logged_in.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-you_are_now_logged_out.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-you_have_dialed_an_invalid_extension.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-you_may.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ivr\8000\ivr-you_may_exit_by_hanging_up.wav
59>62 File(s) copied
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\conf-alone.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\conf-bad-pin.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\conf-goodbye.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\conf-is-locked.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\conf-is-unlocked.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\conf-kicked.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\conf-locked.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\conf-muted.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\conf-pin.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\conf-unmuted.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\conference\8000\conf-welcome.wav
59>11 File(s) copied
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\a-m.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\at.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\day-0.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\day-1.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\day-2.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\day-3.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\day-4.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\day-5.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\day-6.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\hour.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\hours.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\minute.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\minutes.wav
53>Compiling...
53>gettimeofday.c
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-0.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-1.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-10.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-11.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-2.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-3.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-4.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-5.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-6.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-7.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-8.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\mon-9.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\oclock.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\oh.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\p-m.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\second.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\seconds.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\today.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\tomorrow.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\time\8000\yesterday.wav
59>33 File(s) copied
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\0.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\1.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\10.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\11.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\12.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\13.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\14.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\15.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\16.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\17.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\18.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\19.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\2.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\20.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\3.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\30.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\4.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\40.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\5.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\50.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\6.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\60.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\7.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\70.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\8.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\80.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\9.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\90.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\dot.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-1.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-10.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-11.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-12.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-13.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-14.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-15.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-16.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-17.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-18.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-19.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-2.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-20.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-3.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-30.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-4.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-5.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-6.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-7.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-8.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\h-9.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\hundred.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\million.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\period.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\point.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\pound.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\star.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\digits\8000\thousand.wav
53>Generating Code...
59>57 File(s) copied
53>Compiling manifest to resources...
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\100.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\101.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\102.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\103.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\104.wav
53>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
53>Copyright (C) Microsoft Corporation.  All rights reserved.
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\105.wav
53>Linking...
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\106.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\107.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\108.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\109.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\110.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\111.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\112.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\113.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\114.wav
53>   Creating library ./Debug\spandsp.lib and object ./Debug\spandsp.exp
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\115.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\116.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\117.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\118.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\119.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\120.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\121.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\122.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\32.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\35.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\42.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\46.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\97.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\98.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\ascii\8000\99.wav
59>30 File(s) copied
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\call_monitoring_blurb.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\call_secured.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\en.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\error.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\es.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\followed.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\if_you_are_this_person.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\if_you_would_like_to.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\invalid_extension.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\misc-your_call_has_been_terminated.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\misc-your_call_will_be_terminated_in.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\phone_not_auth.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\provide_reference_number.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\sorry.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\transfer1.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\transfer2.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\misc\8000\we_are_trying_to_reach.wav
59>17 File(s) copied
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\and.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\cent.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\central.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\cents-per-minute.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\cents.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\dollar.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\dollars.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\minus.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\currency\8000\negative.wav
59>9 File(s) copied
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\100.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\101.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\102.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\103.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\104.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\105.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\106.wav
53>Embedding manifest...
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\107.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\108.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\109.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\110.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\111.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\112.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\113.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\114.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\115.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\116.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\117.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\118.wav
53>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
53>Copyright (C) Microsoft Corporation.  All rights reserved.
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\119.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\120.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\121.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\122.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\32.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\35.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\42.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\46.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\97.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\98.wav
59>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\en\us\callie\phonetic-ascii\8000\99.wav
59>30 File(s) copied
59>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Sound_Files\Debug\BuildLog.htm"
59>8khz - 0 error(s), 0 warning(s)
60>------ Build started: Project: mod_fsv, Configuration: Debug Win32 ------
53>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\spandsp\src\Debug\BuildLog libspandsp.htm"
53>libspandsp - 0 error(s), 0 warning(s)
60>Compiling...
60>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
60>mod_fsv.c
61>------ Build started: Project: mod_fax, Configuration: Debug Win32 ------
60>Linking...
61>Compiling...
61>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
61>mod_fax.c
60>   Creating library Win32\Debug/mod_fsv.2008.lib and object Win32\Debug/mod_fsv.2008.exp
60>Embedding manifest...
60>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_fsv\Win32\Debug\BuildLog.htm"
60>mod_fsv - 0 error(s), 1 warning(s)
62>------ Build started: Project: mod_voipcodecs, Configuration: Debug Win32 ------
62>Compiling...
62>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
62>mod_voipcodecs.c
62>Linking...
62>   Creating library Win32\Debug/mod_voipcodecs.2008.lib and object Win32\Debug/mod_voipcodecs.2008.exp
62>Embedding manifest...
61>Linking...
61>   Creating library Win32\Debug/mod_fax.2008.lib and object Win32\Debug/mod_fax.2008.exp
61>Embedding manifest...
62>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\codecs\mod_voipcodecs\Win32\Debug\BuildLog.htm"
62>mod_voipcodecs - 0 error(s), 1 warning(s)
61>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_fax\Win32\Debug\BuildLog.htm"
61>mod_fax - 0 error(s), 1 warning(s)
63>------ Build started: Project: mod_tone_stream, Configuration: Debug Win32 ------
63>Compiling...
63>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
63>mod_tone_stream.c
64>------ Build started: Project: mod_cdr_csv, Configuration: Debug Win32 ------
64>Compiling...
64>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
64>mod_cdr_csv.c
63>Linking...
63>   Creating library Win32\Debug/mod_tone_stream.2008.lib and object Win32\Debug/mod_tone_stream.2008.exp
64>Linking...
64>   Creating library Win32\Debug/mod_cdr_csv.2008.lib and object Win32\Debug/mod_cdr_csv.2008.exp
63>Embedding manifest...
64>Embedding manifest...
63>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\formats\mod_tone_stream\Win32\Debug\BuildLog.htm"
63>mod_tone_stream - 0 error(s), 1 warning(s)
64>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\event_handlers\mod_cdr_csv\Win32\Debug\BuildLog.htm"
64>mod_cdr_csv - 0 error(s), 1 warning(s)
65>------ Build started: Project: mod_logfile, Configuration: Debug Win32 ------
65>Compiling...
66>------ Build started: Project: mod_dialplan_asterisk, Configuration: Debug Win32 ------
65>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
65>mod_logfile.c
66>Compiling...
66>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
66>mod_dialplan_asterisk.c
65>Linking...
65>   Creating library Win32\Debug/mod_logfile.2008.lib and object Win32\Debug/mod_logfile.2008.exp
66>Linking...
66>   Creating library Win32\Debug/mod_dialplan_asterisk.2008.lib and object Win32\Debug/mod_dialplan_asterisk.2008.exp
65>Embedding manifest...
66>Embedding manifest...
65>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\loggers\mod_logfile\Win32\Debug\BuildLog.htm"
65>mod_logfile - 0 error(s), 1 warning(s)
66>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\dialplans\mod_dialplan_asterisk\Win32\Debug\BuildLog.htm"
66>mod_dialplan_asterisk - 0 error(s), 1 warning(s)
67>------ Build started: Project: mod_expr, Configuration: Debug Win32 ------
67>Compiling...
67>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
67>mod_expr.c
68>------ Build started: Project: mod_limit, Configuration: Debug Win32 ------
68>Compiling...
68>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
68>mod_limit.c
67>exprval.c
67>exprutil.c
67>exprpars.c
67>exprobj.c
67>exprmem.c
67>exprinit.c
67>exprfunc.c
68>Linking...
68>   Creating library Win32\Debug/mod_limit.2008.lib and object Win32\Debug/mod_limit.2008.exp
67>expreval.c
67>Generating Code...
68>Embedding manifest...
67>Linking...
67>   Creating library Win32\Debug/mod_expr.2008.lib and object Win32\Debug/mod_expr.2008.exp
68>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_limit\Win32\Debug\BuildLog.htm"
68>mod_limit - 0 error(s), 1 warning(s)
69>------ Build started: Project: mod_fifo, Configuration: Debug Win32 ------
69>Compiling...
69>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
69>mod_fifo.c
67>Embedding manifest...
67>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_expr\Win32\Debug\BuildLog.htm"
67>mod_expr - 0 error(s), 1 warning(s)
70>------ Build started: Project: mod_say_nl, Configuration: Debug Win32 ------
70>Compiling...
70>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
70>mod_say_nl.c
69>Linking...
69>   Creating library Win32\Debug/mod_fifo.2008.lib and object Win32\Debug/mod_fifo.2008.exp
69>Embedding manifest...
70>Linking...
69>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_fifo\Win32\Debug\BuildLog.htm"
69>mod_fifo - 0 error(s), 1 warning(s)
70>   Creating library Win32\Debug/mod_say_nl.2008.lib and object Win32\Debug/mod_say_nl.2008.exp
71>------ Build started: Project: mod_say_it, Configuration: Debug Win32 ------
71>Compiling...
70>Embedding manifest...
71>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
71>mod_say_it.c
70>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\say\mod_say_nl\Win32\Debug\BuildLog.htm"
70>mod_say_nl - 0 error(s), 1 warning(s)
72>------ Build started: Project: mod_say_fr, Configuration: Debug Win32 ------
72>Compiling...
72>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
72>mod_say_fr.c
71>Linking...
71>   Creating library Win32\Debug/mod_say_it.2008.lib and object Win32\Debug/mod_say_it.2008.exp
72>Linking...
71>Embedding manifest...
72>   Creating library Win32\Debug/mod_say_fr.2008.lib and object Win32\Debug/mod_say_fr.2008.exp
72>Embedding manifest...
71>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\say\mod_say_it\Win32\Debug\BuildLog.htm"
71>mod_say_it - 0 error(s), 1 warning(s)
72>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\say\mod_say_fr\Win32\Debug\BuildLog.htm"
72>mod_say_fr - 0 error(s), 1 warning(s)
73>------ Skipped Build: Project: 16khz, Configuration: Debug Win32 ------
73>Project not selected to build for this solution configuration
74>------ Build started: Project: mod_say_de, Configuration: Debug Win32 ------
75>------ Build started: Project: mod_voicemail, Configuration: Debug Win32 ------
74>Compiling...
75>Compiling...
74>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
74>mod_say_de.c
75>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
75>mod_voicemail.c
74>Linking...
74>   Creating library Win32\Debug/mod_say_de.2008.lib and object Win32\Debug/mod_say_de.2008.exp
75>Linking...
75>   Creating library Win32\Debug/mod_voicemail.2008.lib and object Win32\Debug/mod_voicemail.2008.exp
74>Embedding manifest...
75>Embedding manifest...
74>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\say\mod_say_de\Win32\Debug\BuildLog.htm"
74>mod_say_de - 0 error(s), 1 warning(s)
75>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_voicemail\Win32\Debug\BuildLog.htm"
75>mod_voicemail - 0 error(s), 1 warning(s)
76>------ Build started: Project: mod_spidermonkey_socket, Configuration: Debug Win32 ------
76>Compiling...
76>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
76>mod_spidermonkey_socket.c
77>------ Build started: Project: mod_local_stream, Configuration: Debug Win32 ------
77>Compiling...
77>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
77>mod_local_stream.c
77>Linking...
77>   Creating library Win32\Debug/mod_local_stream.2008.lib and object Win32\Debug/mod_local_stream.2008.exp
76>Linking...
76>   Creating library Win32\Debug/mod_spidermonkey_socket.2008.lib and object Win32\Debug/mod_spidermonkey_socket.2008.exp
77>Embedding manifest...
76>Embedding manifest...
77>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\formats\mod_local_stream\Win32\Debug\BuildLog.htm"
77>mod_local_stream - 0 error(s), 1 warning(s)
76>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_spidermonkey\Win32\Debug\BuildLog.htm"
76>mod_spidermonkey_socket - 0 error(s), 1 warning(s)
78>------ Build started: Project: mod_esf, Configuration: Debug Win32 ------
79>------ Build started: Project: mod_h26x, Configuration: Debug Win32 ------
78>Compiling...
79>Compiling...
78>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
78>mod_esf.c
79>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
79>mod_h26x.c
79>Linking...
78>Linking...
79>   Creating library Win32\Debug/mod_h26x.2008.lib and object Win32\Debug/mod_h26x.2008.exp
78>   Creating library Win32\Debug/mod_esf.2008.lib and object Win32\Debug/mod_esf.2008.exp
79>Embedding manifest...
78>Embedding manifest...
78>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_esf\Win32\Debug\BuildLog.htm"
78>mod_esf - 0 error(s), 1 warning(s)
79>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\codecs\mod_h26x\Win32\Debug\BuildLog.htm"
79>mod_h26x - 0 error(s), 1 warning(s)
80>------ Build started: Project: mod_amr, Configuration: Debug Passthrough Win32 ------
80>Compiling...
80>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
80>mod_amr.c
81>------ Build started: Project: mod_xml_cdr, Configuration: Debug Win32 ------
81>Compiling...
81>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
81>mod_xml_cdr.c
80>Linking...
80>   Creating library Win32\Debug Passthrough/mod_amr.2008.lib and object Win32\Debug Passthrough/mod_amr.2008.exp
80>Embedding manifest...
80>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\codecs\mod_amr\Win32\Debug Passthrough\BuildLog.htm"
80>mod_amr - 0 error(s), 1 warning(s)
81>Linking...
81>   Creating library Win32\Debug/mod_xml_cdr.2008.lib and object Win32\Debug/mod_xml_cdr.2008.exp
82>------ Skipped Build: Project: 32khz, Configuration: Debug Win32 ------
82>Project not selected to build for this solution configuration
83>------ Build started: Project: mod_say_en, Configuration: Debug Win32 ------
83>Compiling...
81>Embedding manifest...
83>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
83>mod_say_en.c
81>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\xml_int\mod_xml_cdr\Win32\Debug\BuildLog.htm"
81>mod_xml_cdr - 0 error(s), 1 warning(s)
84>------ Build started: Project: mod_xml_curl, Configuration: Debug Win32 ------
84>Compiling...
84>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
84>mod_xml_curl.c
83>Linking...
83>   Creating library Win32\Debug/mod_say_en.2008.lib and object Win32\Debug/mod_say_en.2008.exp
83>Embedding manifest...
84>Linking...
84>   Creating library Win32\Debug/mod_xml_curl.2008.lib and object Win32\Debug/mod_xml_curl.2008.exp
83>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\say\mod_say_en\Win32\Debug\BuildLog.htm"
83>mod_say_en - 0 error(s), 1 warning(s)
84>Embedding manifest...
84>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\xml_int\mod_xml_curl\Win32\Debug\BuildLog.htm"
84>mod_xml_curl - 0 error(s), 1 warning(s)
85>------ Build started: Project: mod_spidermonkey_odbc, Configuration: Debug Win32 ------
85>Compiling...
85>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
85>mod_spidermonkey_odbc.c
86>------ Build started: Project: mod_enum, Configuration: Debug Win32 ------
86>Compiling...
86>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
86>mod_enum.c
86>Linking...
86>   Creating library Win32\Debug/mod_enum.2008.lib and object Win32\Debug/mod_enum.2008.exp
85>Linking...
85>   Creating library Win32\Debug/mod_spidermonkey_odbc.2008.lib and object Win32\Debug/mod_spidermonkey_odbc.2008.exp
86>Embedding manifest...
85>Embedding manifest...
86>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_enum\Win32\Debug\BuildLog.htm"
86>mod_enum - 0 error(s), 1 warning(s)
85>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_spidermonkey\Win32\Debug\BuildLog.htm"
85>mod_spidermonkey_odbc - 0 error(s), 1 warning(s)
87>------ Build started: Project: mod_flite, Configuration: Debug Win32 ------
87>Compiling...
87>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
87>mod_flite.c
88>------ Build started: Project: mod_spidermonkey_teletone, Configuration: Debug Win32 ------
88>Compiling...
88>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
88>mod_spidermonkey_teletone.c
87>Compiling manifest to resources...
87>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
87>Copyright (C) Microsoft Corporation.  All rights reserved.
87>Linking...
87>   Creating library Win32\Debug/mod_flite.2008.lib and object Win32\Debug/mod_flite.2008.exp
88>Linking...
88>   Creating library Win32\Debug/mod_spidermonkey_teletone.2008.lib and object Win32\Debug/mod_spidermonkey_teletone.2008.exp
88>Embedding manifest...
88>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_spidermonkey\Win32\Debug\BuildLog.htm"
88>mod_spidermonkey_teletone - 0 error(s), 1 warning(s)
87>Embedding manifest...
87>Microsoft (R) Windows (R) Resource Compiler Version 6.1.6723.1
87>Copyright (C) Microsoft Corporation.  All rights reserved.
89>------ Build started: Project: mod_spidermonkey_core_db, Configuration: Debug Win32 ------
89>Compiling...
89>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
89>mod_spidermonkey_core_db.c
87>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\asr_tts\mod_flite\Win32\Debug\BuildLog.htm"
87>mod_flite - 0 error(s), 1 warning(s)
90>------ Build started: Project: mod_native_file, Configuration: Debug Win32 ------
90>Compiling...
90>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
90>mod_native_file.c
89>Linking...
90>Linking...
90>   Creating library Win32\Debug/mod_native_file.2008.lib and object Win32\Debug/mod_native_file.2008.exp
89>   Creating library Win32\Debug/mod_spidermonkey_core_db.2008.lib and object Win32\Debug/mod_spidermonkey_core_db.2008.exp
90>Embedding manifest...
89>Embedding manifest...
90>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\formats\mod_native_file\Win32\Debug\BuildLog.htm"
90>mod_native_file - 0 error(s), 1 warning(s)
91>------ Build started: Project: mod_g723_1, Configuration: Debug Passthrough Win32 ------
91>Compiling...
89>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_spidermonkey\Win32\Debug\BuildLog.htm"
89>mod_spidermonkey_core_db - 0 error(s), 1 warning(s)
91>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
91>mod_g723_1.c
92>------ Build started: Project: fs_cli, Configuration: Debug Win32 ------
92>Compiling...
92>fs_cli.c
92>getopt_long.c
91>Linking...
91>   Creating library Win32\Debug Passthrough/mod_g723_1.2008.lib and object Win32\Debug Passthrough/mod_g723_1.2008.exp
92>Generating Code...
92>Linking...
91>Embedding manifest...
92>Embedding manifest...
92>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\esl\Debug\BuildLog fs_cli.htm"
92>fs_cli - 0 error(s), 0 warning(s)
91>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\codecs\mod_g723_1\Win32\Debug Passthrough\BuildLog.htm"
91>mod_g723_1 - 0 error(s), 1 warning(s)
93>------ Skipped Build: Project: mod_opal, Configuration: Debug Win32 ------
93>Project not selected to build for this solution configuration
94>------ Build started: Project: mod_siren, Configuration: Debug Win32 ------
95>------ Build started: Project: mod_sofia, Configuration: Debug Win32 ------
94>Compiling...
94>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
94>mod_siren.c
95>Compiling...
95>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
95>sofia_sla.c
95>c:\freeswitch\freeswitch-1.0.6\src\mod\endpoints\mod_sofia\mod_sofia.h(115) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory
95>sofia_reg.c
94>Linking...
94>   Creating library Win32\Debug/mod_siren.2008.lib and object Win32\Debug/mod_siren.2008.exp
95>c:\freeswitch\freeswitch-1.0.6\src\mod\endpoints\mod_sofia\mod_sofia.h(115) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory
94>Embedding manifest...
95>sofia_presence.c
94>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\codecs\mod_siren\Win32\Debug\BuildLog.htm"
94>mod_siren - 0 error(s), 1 warning(s)
96>------ Build started: Project: mod_shout, Configuration: Debug Win32 ------
96>Compiling...
96>mod_shout.c
95>c:\freeswitch\freeswitch-1.0.6\src\mod\endpoints\mod_sofia\mod_sofia.h(115) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory
95>sofia_glue.c
95>c:\freeswitch\freeswitch-1.0.6\src\mod\endpoints\mod_sofia\mod_sofia.h(115) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory
95>sofia.c
96>Linking...
96>   Creating library Win32\Debug/mod_shout.lib and object Win32\Debug/mod_shout.exp
95>c:\freeswitch\freeswitch-1.0.6\src\mod\endpoints\mod_sofia\mod_sofia.h(115) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory
95>sip-dig.c
95>.\sip-dig.c(813) : warning C4244: 'function' : conversion from 'int' to 'uint16_t', possible loss of data
95>.\sip-dig.c(822) : warning C4244: 'function' : conversion from 'int' to 'uint16_t', possible loss of data
96>Embedding manifest...
95>mod_sofia.c
96>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\formats\mod_shout\Win32\Debug\BuildLog.htm"
96>mod_shout - 0 error(s), 0 warning(s)
97>------ Build started: Project: xml, Configuration: Debug Win32 ------
97>Creating config.h from winconfig.h
97>Creating expat.h from expat.h.in
95>c:\freeswitch\freeswitch-1.0.6\src\mod\endpoints\mod_sofia\mod_sofia.h(115) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory
95>Generating Code...
97>Compiling...
95>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\endpoints\mod_sofia\Win32\Debug\BuildLog.htm"
95>mod_sofia - 6 error(s), 3 warning(s)
98>------ Build started: Project: mod_vmd, Configuration: Debug Win32 ------
97>xmlparse.c
98>Compiling...
98>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
98>mod_vmd.c
97>xmltok.c
97>xmlrole.c
97>Generating Code...
97>Creating library...
97>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\apr-util\Debug\BuildLog.htm"
97>xml - 0 error(s), 0 warning(s)
98>Linking...
98>   Creating library Win32\Debug/mod_vmd.2008.lib and object Win32\Debug/mod_vmd.2008.exp
99>------ Build started: Project: mod_snom, Configuration: Debug Win32 ------
98>Embedding manifest...
99>Compiling...
99>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
99>mod_snom.c
98>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_vmd\Win32\Debug\BuildLog.htm"
98>mod_vmd - 0 error(s), 1 warning(s)
100>------ Build started: Project: mod_say_zh, Configuration: Debug Win32 ------
100>Compiling...
100>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
100>mod_say_zh.c
99>Linking...
99>   Creating library Win32\Debug/mod_snom.2008.lib and object Win32\Debug/mod_snom.2008.exp
99>Embedding manifest...
100>Linking...
99>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_snom\Win32\Debug\BuildLog.htm"
99>mod_snom - 0 error(s), 1 warning(s)
100>   Creating library Win32\Debug/mod_say_zh.2008.lib and object Win32\Debug/mod_say_zh.2008.exp
100>Embedding manifest...
101>------ Build started: Project: mod_managed, Configuration: Debug_CLR Win32 ------
101>Compiling...
101>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
101>mod_managed.cpp
100>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\say\mod_say_zh\Win32\Debug\BuildLog.htm"
100>mod_say_zh - 0 error(s), 1 warning(s)
102>------ Build started: Project: 8khz music, Configuration: Debug Win32 ------
102>Performing Post-Build Event...
102>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\music\8000\danza-espanola-op-37-h-142-xii-arabesca.wav
102>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\music\8000\partita-no-3-in-e-major-bwv-1006-1-preludio.wav
102>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\music\8000\ponce-preludio-in-e-major.wav
102>C:\FreeSWITCH\freeswitch-1.0.6\libs\sounds\music\8000\suite-espanola-op-47-leyenda.wav
102>4 File(s) copied
102>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\libs\win32\Sound_Files\Debug\BuildLog.htm"
102>8khz music - 0 error(s), 0 warning(s)
103>------ Skipped Build: Project: 16khz music, Configuration: Debug Win32 ------
103>Project not selected to build for this solution configuration
104>------ Skipped Build: Project: 32khz music, Configuration: Debug Win32 ------
104>Project not selected to build for this solution configuration
105>------ Build started: Project: mod_say_ru, Configuration: Debug Win32 ------
105>Compiling...
105>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
105>mod_say_ru.c
105>Linking...
105>   Creating library Win32\Debug/mod_say_ru.2008.lib and object Win32\Debug/mod_say_ru.2008.exp
105>Embedding manifest...
101>freeswitch_wrap.cxx
105>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\say\mod_say_ru\Win32\Debug\BuildLog.htm"
105>mod_say_ru - 0 error(s), 1 warning(s)
106>------ Build started: Project: mod_skypopen, Configuration: Debug Win32 ------
106>Compiling...
106>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
106>skypopen_protocol.c
106>mod_skypopen.c
106>Generating Code...
106>Linking...
106>   Creating library Win32\Debug/mod_skypopen.2008.lib and object Win32\Debug/mod_skypopen.2008.exp
106>Embedding manifest...
106>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\endpoints\mod_skypopen\Win32\Debug\BuildLog.htm"
106>mod_skypopen - 0 error(s), 1 warning(s)
101>freeswitch_managed.cpp
107>------ Build started: Project: mod_loopback, Configuration: Debug Win32 ------
107>Compiling...
107>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
107>mod_loopback.c
107>Linking...
107>   Creating library Win32\Debug/mod_loopback.2008.lib and object Win32\Debug/mod_loopback.2008.exp
107>Embedding manifest...
107>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\endpoints\mod_loopback\Win32\Debug\BuildLog.htm"
107>mod_loopback - 0 error(s), 1 warning(s)
101>Generating Code...
108>------ Build started: Project: mod_event_socket, Configuration: Debug Win32 ------
108>Compiling...
108>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
108>mod_event_socket.c
101>Linking...
108>Linking...
108>   Creating library Win32\Debug/mod_event_socket.2008.lib and object Win32\Debug/mod_event_socket.2008.exp
108>Embedding manifest...
108>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\event_handlers\mod_event_socket\Win32\Debug\BuildLog.htm"
108>mod_event_socket - 0 error(s), 1 warning(s)
101>   Creating library Debug_CLR/mod_managed.lib and object Debug_CLR/mod_managed.exp
109>------ Build started: Project: mod_dptools, Configuration: Debug Win32 ------
109>Compiling...
109>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
109>mod_dptools.c
101>freeswitch_wrap.obj : warning LNK4248: unresolved typeref token (0100001F) for 'switch_odbc_handle'; image may not run
109>Linking...
109>   Creating library Win32\Debug/mod_dptools.2008.lib and object Win32\Debug/mod_dptools.2008.exp
109>Embedding manifest...
109>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_dptools\Win32\Debug\BuildLog.htm"
109>mod_dptools - 0 error(s), 1 warning(s)
101>Embedding manifest...
110>------ Build started: Project: mod_conference, Configuration: Debug Win32 ------
110>Compiling...
110>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
110>mod_conference.c
101>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_managed\Debug_CLR\BuildLog.htm"
101>mod_managed - 0 error(s), 2 warning(s)
111>------ Build started: Project: mod_rss, Configuration: Debug Win32 ------
111>Compiling...
111>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
111>mod_rss.c
110>.\mod_conference.c(1167) : warning C4018: '<' : signed/unsigned mismatch
110>.\mod_conference.c(1168) : warning C4018: '<=' : signed/unsigned mismatch
110>.\mod_conference.c(1182) : warning C4018: '<' : signed/unsigned mismatch
110>.\mod_conference.c(1205) : warning C4018: '<' : signed/unsigned mismatch
110>.\mod_conference.c(1208) : warning C4018: '<=' : signed/unsigned mismatch
110>.\mod_conference.c(2098) : warning C4018: '<' : signed/unsigned mismatch
110>.\mod_conference.c(2101) : warning C4244: '=' : conversion from 'int32_t' to 'int16_t', possible loss of data
110>Linking...
110>   Creating library Win32\Debug/mod_conference.2008.lib and object Win32\Debug/mod_conference.2008.exp
111>Linking...
111>   Creating library Win32\Debug/mod_rss.2008.lib and object Win32\Debug/mod_rss.2008.exp
110>Embedding manifest...
110>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_conference\Win32\Debug\BuildLog.htm"
110>mod_conference - 0 error(s), 8 warning(s)
111>Embedding manifest...
111>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_rss\Win32\Debug\BuildLog.htm"
111>mod_rss - 0 error(s), 1 warning(s)
112>------ Build started: Project: mod_xml_rpc, Configuration: Debug Win32 ------
112>Compiling...
112>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
112>mod_xml_rpc.c
113>------ Build started: Project: mod_console, Configuration: Debug Win32 ------
113>Compiling...
113>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
113>mod_console.c
112>.\mod_xml_rpc.c(307) : warning C4090: 'function' : different 'const' qualifiers
112>.\mod_xml_rpc.c(307) : warning C4244: 'function' : conversion from 'int' to 'const xmlrpc_uint16_t', possible loss of data
112>.\mod_xml_rpc.c(309) : warning C4090: 'function' : different 'const' qualifiers
112>.\mod_xml_rpc.c(311) : warning C4090: 'function' : different 'const' qualifiers
112>.\mod_xml_rpc.c(391) : warning C4090: 'function' : different 'const' qualifiers
112>.\mod_xml_rpc.c(471) : warning C4090: 'function' : different 'const' qualifiers
112>.\mod_xml_rpc.c(472) : warning C4090: 'function' : different 'const' qualifiers
112>.\mod_xml_rpc.c(473) : warning C4090: 'function' : different 'const' qualifiers
112>Linking...
112>   Creating library Win32\Debug/mod_xml_rpc.2008.lib and object Win32\Debug/mod_xml_rpc.2008.exp
113>Linking...
113>   Creating library Win32\Debug/mod_console.2008.lib and object Win32\Debug/mod_console.2008.exp
113>Embedding manifest...
112>Embedding manifest...
113>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\loggers\mod_console\Win32\Debug\BuildLog.htm"
113>mod_console - 0 error(s), 1 warning(s)
112>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\xml_int\mod_xml_rpc\Win32\Debug\BuildLog.htm"
112>mod_xml_rpc - 0 error(s), 9 warning(s)
114>------ Build started: Project: mod_commands, Configuration: Debug Win32 ------
114>Compiling...
114>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
114>mod_commands.c
115>------ Build started: Project: mod_dingaling, Configuration: Debug Win32 ------
115>Compiling...
115>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
115>mod_dingaling.c
114>Linking...
114>   Creating library Win32\Debug/mod_commands.2008.lib and object Win32\Debug/mod_commands.2008.exp
115>Linking...
115>   Creating library Win32\Debug/mod_dingaling.2008.lib and object Win32\Debug/mod_dingaling.2008.exp
114>Embedding manifest...
115>Embedding manifest...
115>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\endpoints\mod_dingaling\Win32\Debug\BuildLog.htm"
115>mod_dingaling - 0 error(s), 1 warning(s)
114>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_commands\Win32\Debug\BuildLog.htm"
114>mod_commands - 0 error(s), 1 warning(s)
116>------ Build started: Project: mod_ilbc, Configuration: Debug Win32 ------
116>Compiling...
116>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
116>mod_ilbc.c
117>------ Skipped Build: Project: mod_cepstral, Configuration: Debug Win32 ------
117>Project not selected to build for this solution configuration
118>------ Build started: Project: mod_say_es, Configuration: Debug Win32 ------
118>Compiling...
118>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
118>mod_say_es.c
116>Linking...
116>   Creating library Win32\Debug/mod_ilbc.2008.lib and object Win32\Debug/mod_ilbc.2008.exp
118>Linking...
118>   Creating library Win32\Debug/mod_say_es.2008.lib and object Win32\Debug/mod_say_es.2008.exp
116>Embedding manifest...
118>Embedding manifest...
116>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\codecs\mod_ilbc\Win32\Debug\BuildLog.htm"
116>mod_ilbc - 0 error(s), 1 warning(s)
118>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\say\mod_say_es\Win32\Debug\BuildLog.htm"
118>mod_say_es - 0 error(s), 1 warning(s)
119>------ Build started: Project: mod_spidermonkey_curl, Configuration: Debug Win32 ------
119>Compiling...
119>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
119>mod_spidermonkey_curl.c
120>------ Build started: Project: mod_event_multicast, Configuration: Debug Win32 ------
120>Compiling...
120>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
120>mod_event_multicast.c
120>Linking...
120>   Creating library Win32\Debug/mod_event_multicast.2008.lib and object Win32\Debug/mod_event_multicast.2008.exp
119>Linking...
119>   Creating library Win32\Debug/mod_spidermonkey_curl.2008.lib and object Win32\Debug/mod_spidermonkey_curl.2008.exp
120>Embedding manifest...
119>Embedding manifest...
119>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\languages\mod_spidermonkey\Win32\Debug\BuildLog.htm"
119>mod_spidermonkey_curl - 0 error(s), 1 warning(s)
120>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\event_handlers\mod_event_multicast\Win32\Debug\BuildLog.htm"
120>mod_event_multicast - 0 error(s), 1 warning(s)
121>------ Build started: Project: mod_dialplan_directory, Configuration: Debug Win32 ------
121>Compiling...
122>------ Build started: Project: mod_ldap, Configuration: Debug MS-LDAP Win32 ------
121>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
121>mod_dialplan_directory.c
122>Compiling...
122>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
122>mod_ldap.c
122>Linking...
121>Linking...
121>   Creating library Win32\Debug/mod_dialplan_directory.2008.lib and object Win32\Debug/mod_dialplan_directory.2008.exp
122>   Creating library Win32\Debug MS-LDAP/mod_ldap.2008.lib and object Win32\Debug MS-LDAP/mod_ldap.2008.exp
121>Embedding manifest...
122>Embedding manifest...
121>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\dialplans\mod_dialplan_directory\Win32\Debug\BuildLog.htm"
121>mod_dialplan_directory - 0 error(s), 1 warning(s)
122>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\directories\mod_ldap\Win32\Debug MS-LDAP\BuildLog.htm"
122>mod_ldap - 0 error(s), 1 warning(s)
123>------ Build started: Project: mod_dialplan_xml, Configuration: Debug Win32 ------
123>Compiling...
123>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
123>mod_dialplan_xml.c
124>------ Skipped Build: Project: docs, Configuration: Debug Win32 ------
124>Project not selected to build for this solution configuration
125>------ Build started: Project: mod_speex, Configuration: Debug Win32 ------
125>Compiling...
125>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
125>mod_speex.c
123>Linking...
123>   Creating library Win32\Debug/mod_dialplan_xml.2008.lib and object Win32\Debug/mod_dialplan_xml.2008.exp
125>Linking...
125>libspeexdsp.lib(preprocess.obj) : warning LNK4075: ignoring '/EDITANDCONTINUE' due to '/INCREMENTAL:NO' specification
125>   Creating library Win32\Debug/mod_speex.2008.lib and object Win32\Debug/mod_speex.2008.exp
123>Embedding manifest...
123>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\dialplans\mod_dialplan_xml\Win32\Debug\BuildLog.htm"
123>mod_dialplan_xml - 0 error(s), 1 warning(s)
125>Embedding manifest...
126>------ Build started: Project: mod_PortAudio, Configuration: Debug Win32 ------
126>Compiling...
126>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
126>pablio.c
125>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\codecs\mod_speex\Win32\Debug\BuildLog.htm"
125>mod_speex - 0 error(s), 2 warning(s)
127>------ Build started: Project: mod_sndfile, Configuration: Debug Win32 ------
127>Compiling...
127>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
127>mod_sndfile.c
126>pa_ringbuffer.c
126>mod_PortAudio.c
127>Linking...
126>Generating Code...
127>   Creating library Win32\Debug/mod_sndfile.2008.lib and object Win32\Debug/mod_sndfile.2008.exp
126>Linking...
126>   Creating library Win32\Debug/mod_PortAudio.2008.lib and object Win32\Debug/mod_PortAudio.2008.exp
127>Embedding manifest...
126>Embedding manifest...
127>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\formats\mod_sndfile\Win32\Debug\BuildLog.htm"
127>mod_sndfile - 0 error(s), 1 warning(s)
126>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\endpoints\mod_portaudio\Win32\Debug\BuildLog.htm"
126>mod_PortAudio - 0 error(s), 1 warning(s)
128>------ Build started: Project: mod_g729, Configuration: Debug Passthrough Win32 ------
128>Compiling...
128>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
128>mod_g729.c
129>------ Build started: Project: mod_valet_parking, Configuration: Debug Win32 ------
129>Compiling...
129>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
129>mod_valet_parking.c
128>Linking...
128>   Creating library Win32\Debug Passthrough/mod_g729.2008.lib and object Win32\Debug Passthrough/mod_g729.2008.exp
128>Embedding manifest...
128>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\codecs\mod_g729\Win32\Debug Passthrough\BuildLog.htm"
128>mod_g729 - 0 error(s), 1 warning(s)
129>Linking...
129>   Creating library Win32\Debug/mod_valet_parking.2008.lib and object Win32\Debug/mod_valet_parking.2008.exp
129>Embedding manifest...
129>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\src\mod\applications\mod_valet_parking\Win32\Debug\BuildLog.htm"
129>mod_valet_parking - 0 error(s), 1 warning(s)
130>------ Build started: Project: FreeSwitchConsole, Configuration: Debug Win32 ------
130>Compiling...
130>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler
130>switch.c
130>Linking...
130>Embedding manifest...
130>Build log was saved at "file://c:\FreeSWITCH\freeswitch-1.0.6\w32\Console\Debug\BuildLog FreeSwitchConsole.htm"
130>FreeSwitchConsole - 0 error(s), 1 warning(s)
========== Build: 117 succeeded, 2 failed, 0 up-to-date, 11 skipped ==========
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From h.dogger at telecats.nl  Mon Dec 20 18:52:32 2010
From: h.dogger at telecats.nl (Henry Dogger)
Date: Mon, 20 Dec 2010 16:52:32 +0100
Subject: [Freeswitch-users] SBC on kamailio
Message-ID: <0A1FDB5DAA23564F8758BA05D26DCD74C6C15E@exchange.telecats.nl>

Hi all,

 

We have a kamailio installation for our voip system and would like to
add a SBC.

We found the freeswitch possibility:
http://wiki.freeswitch.org/wiki/SBC_Setup

But in this case, 302 Redirect SIP is used, we would like the SBC to act
as a pass-through controller, just to be a gateway for SIP clients from
outside.

Is something like this possible? And will registrations be handled
correctly since all SIP messages will be on port 5060.

Thanks in advance.

 

Kind regards,

 

Henry Dogger

Telecats BV

 

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From lucian at e-utile.ro  Mon Dec 20 19:18:16 2010
From: lucian at e-utile.ro (Lucian Marginean)
Date: Mon, 20 Dec 2010 11:18:16 -0500
Subject: [Freeswitch-users] On what condition P-Asserted-Identity is
	replaced by P-Preferred-Identity?
Message-ID: <AANLkTinZ3Vr_N=pB8QuvDVHUy2rCscz1yPLHfJFVQoQL@mail.gmail.com>

I try to setup P-Asserted-Identity on setup but I get P-Preferred-Identity
and I don't understand why

Did I do something wrong or some other flags affect the PID feature?

<action application="set" data="effective_caller_id_name=647476xxxx"/>
<action application="set" data="effective_caller_id_number=647788xxxx"/>
<action application="set" data="origination_privacy=screen"/>
<action application="bridge"
data="[sip_cid_type=pid,origination_caller_id_name=647476xxxx,origination_caller_id_number=647788xxxx]sofia/private/
905780xxxx at 10.10.67.82:5060"/>

and invite generated look like this:

U 10.10.67.83:5060 -> 10.10.67.82:5060
INVITE sip:905780xxxx at 10.10.67.82:5060 SIP/2.0.
Via: SIP/2.0/UDP 10.10.67.83;rport;branch=z9hG4bKH8KNm98v4Qv1N.
Max-Forwards: 68.
From: "647476xxxx" <sip:647788xxxx at 10.10.67.83<sip%3A647788xxxx at 10.10.67.83>
>;tag=F1QF1Z26cBQZm.
To: <sip:905780xxxx at 10.10.67.82:5060>.
Call-ID: ff01d92d-86ec-122e-cba2-001143e7e50c.
CSeq: 6092608 INVITE.
Contact: <sip:mod_sofia at 10.10.67.83:5060>.
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-79ced28 2010-08-22 20-58-52
-0400.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, refer.
Privacy: none.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 242.
X-FS-Support: update_display.
*P-Preferred-Identity*: "647476xxxx"
<sip:647788xxxx at 10.10.67.83<sip%3A647788xxxx at 10.10.67.83>
>

Lucian
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From mel0torme at gmail.com  Mon Dec 20 20:19:43 2010
From: mel0torme at gmail.com (Tom C)
Date: Mon, 20 Dec 2010 09:19:43 -0800
Subject: [Freeswitch-users] Disable UPnP for "internal" port only?
Message-ID: <AANLkTikm_Z6-eEQX_LjdOf6omYEsyvoRYF5cxh5pU6fM@mail.gmail.com>

I see that I can run ./freeswitch -nonat to completely disable the use of
UPnP.

But what if I only want FS to stop opening a path to its internal
registration port (5060)?  All the devices that I want to register are on my
home network, behind the NAT, and I'd prefer not to expose the internal
registration port to hackers until I need to.  (And then, only after I have
learned more about security measures that I can implement).
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From curriegrad2004 at gmail.com  Mon Dec 20 20:39:45 2010
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 20 Dec 2010 09:39:45 -0800
Subject: [Freeswitch-users] Disable UPnP for "internal" port only?
In-Reply-To: <AANLkTikm_Z6-eEQX_LjdOf6omYEsyvoRYF5cxh5pU6fM@mail.gmail.com>
References: <AANLkTikm_Z6-eEQX_LjdOf6omYEsyvoRYF5cxh5pU6fM@mail.gmail.com>
Message-ID: <AANLkTinb6VhecA+W6fEUa+9=B03yD5OWqvQTqn7g_QNm@mail.gmail.com>

nonat and ip_nat_sip module from netfilter would be a better choice in
this situation

On Mon, Dec 20, 2010 at 9:19 AM, Tom C <mel0torme at gmail.com> wrote:
> I see that I can run ./freeswitch -nonat to completely disable the use of
> UPnP.
>
> But what if I only want FS to stop opening a path to its internal
> registration port (5060)?? All the devices that I want to register are on my
> home network, behind the NAT, and I'd prefer not to expose the internal
> registration port to hackers until I need to.? (And then, only after I have
> learned more about security measures that I can implement).
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Mon Dec 20 20:59:13 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 20 Dec 2010 09:59:13 -0800
Subject: [Freeswitch-users] members audio conference
In-Reply-To: <532EC039F378476A9515E88F30BBB432@e1705>
References: <223AD13D879E4B19B15AEEDFC387386A@e1705>
	<AANLkTikydvXoJOk5kkWPYpaDqCGYTJgXbzj2=ffTB07p@mail.gmail.com>
	<532EC039F378476A9515E88F30BBB432@e1705>
Message-ID: <AANLkTi=_DDyiOtBUDkJ9ET8FATyu3qntDK3PFA1xXi=7@mail.gmail.com>

The conference itself doesn't do any codec stuff - FreeSWITCH core does. All
I can say is compare the working versus non-working logs and look for clues.
-MC

On Fri, Dec 17, 2010 at 6:49 PM, Madovsky <infos at madovsky.org> wrote:

>  nohgint strange on logs.
> but I guess it's a codec and rate problem.
> tried with different SIP phones and it works.
> how a conference manage the codecs  ?
> I know the rate can be set in profile,
> but how conference codec is managed
>  if all members have different codec and rate ?
>
>
>
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Friday, December 17, 2010 7:54 PM
> *Subject:* Re: [Freeswitch-users] members audio conference
>
> What do you see in the debug logs? Did you compare the logs for a working
> vs. non-working call? Anything different?
>
> -MC
>
> On Fri, Dec 17, 2010 at 7:35 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  when the first member creates and enters in a new conference
>> everything is ok. but if a new memeber enters there is no audio
>> in the conference, unless the ivr.
>>
>> I have a very simple conference dialplan like this
>>
>>         <extension name="create_conference">
>>                 <condition field="${sip_to_uri}"
>> expression="^000(\d{10,15})@$${domain}$">
>>                         <action application="set"
>> data="instant_ringback=true"/>
>>                         <action application="ring_ready"/>
>>                         <action application="sleep" data="3000"/>
>>                         <action application="answer"/>
>>                         <action application="conference" data="
>> $1-${domain_name}@wideband"/ <$1-$%7Bdomain_name%7D at wideband%22/>>
>>                 </condition>
>>         </extension>
>>
>> I tried to add
>> <action application="export" data="absolute_codec_string=speex at 16000k"/<absolute_codec_string=speex at 16000k%22/>
>> >
>>
>> but no success
>>
>> Thanks
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Dec 20 20:59:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 20 Dec 2010 11:59:30 -0600
Subject: [Freeswitch-users] On what condition P-Asserted-Identity is
 replaced by P-Preferred-Identity?
In-Reply-To: <AANLkTinZ3Vr_N=pB8QuvDVHUy2rCscz1yPLHfJFVQoQL@mail.gmail.com>
References: <AANLkTinZ3Vr_N=pB8QuvDVHUy2rCscz1yPLHfJFVQoQL@mail.gmail.com>
Message-ID: <AANLkTi=tU2ceXzoOjqPNzJJJVLafmB7EAjP8Gqz3NgF_@mail.gmail.com>

use the privacy app to set the proper flags and it should do asserted

On Mon, Dec 20, 2010 at 10:18 AM, Lucian Marginean <lucian at e-utile.ro> wrote:
> I try to setup P-Asserted-Identity on setup but I get P-Preferred-Identity
> and I don't understand why
>
> Did I do something wrong or some other flags affect the PID feature?
>
> <action application="set" data="effective_caller_id_name=647476xxxx"/>
> <action application="set" data="effective_caller_id_number=647788xxxx"/>
> <action application="set" data="origination_privacy=screen"/>
> <action application="bridge"
> data="[sip_cid_type=pid,origination_caller_id_name=647476xxxx,origination_caller_id_number=647788xxxx]sofia/private/905780xxxx at 10.10.67.82:5060"/>
>
> and invite generated look like this:
>
> U 10.10.67.83:5060 -> 10.10.67.82:5060
> INVITE sip:905780xxxx at 10.10.67.82:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.10.67.83;rport;branch=z9hG4bKH8KNm98v4Qv1N.
> Max-Forwards: 68.
> From: "647476xxxx" <sip:647788xxxx at 10.10.67.83>;tag=F1QF1Z26cBQZm.
> To: <sip:905780xxxx at 10.10.67.82:5060>.
> Call-ID: ff01d92d-86ec-122e-cba2-001143e7e50c.
> CSeq: 6092608 INVITE.
> Contact: <sip:mod_sofia at 10.10.67.83:5060>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-79ced28 2010-08-22 20-58-52
> -0400.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, hold, refer.
> Privacy: none.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 242.
> X-FS-Support: update_display.
> P-Preferred-Identity: "647476xxxx" <sip:647788xxxx at 10.10.67.83>
>
> Lucian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From msc at freeswitch.org  Mon Dec 20 21:07:11 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 20 Dec 2010 10:07:11 -0800
Subject: [Freeswitch-users] gateway without static IP
In-Reply-To: <AANLkTi=KTyAGuqQMRTNcqxkyN1fqxP-z9NXikzeepKPM@mail.gmail.com>
References: <AANLkTi=kDe23KtpwxMd9S66JjrXhga1RK7+BWE5WAqV5@mail.gmail.com>
	<AANLkTi=KTyAGuqQMRTNcqxkyN1fqxP-z9NXikzeepKPM@mail.gmail.com>
Message-ID: <AANLkTi=hsRskXoJtRN4B+WZe4_2SkiM6_AQeEpBjdcxb@mail.gmail.com>

Just a note, the syntax for using a gateway is:

sofia/gateway/<gateway name>/<target>

If you do this:

sofia/<gateway name>/<target>

...then FS will look for a *SIP profile* named <gateway name>. Hope this
helps. :)

-MC

On Sun, Dec 19, 2010 at 2:00 PM, Steven Ayre <steveayre at gmail.com> wrote:

> First of all, I would  recommend you configure the gateway within the
> <gateways> section of the SIP profile. The dialstring will then be
> sofia/gwname/$1. That format has an advantage because FS will monitor
> the status of the gateway and automatically mark it as offline and
> stop sending traffic to it if it stops responding.
>
> When configuring a gateway you can specify a domain name instead of an
> IP. If you combine that with a service such as www.dyndns.org you can
> run the gateway on a dynamic IP.
>
> Also, have you tried:
> <action application="bridge" data="sofia/internal/$1@<domain>"/>
> that might work, but I've never tried it with anything but an IP so
> can't say for sure without trying it.
>
> This does not do what you think it does:
> <action application="bridge"
> data="sofia/internal/userGateway%{domain}^$1"/>
> The % syntax is for dialing an extension on the *local* server, on the
> {domain} handled by the current server. That won't send a call out to
> a gateway.
>
> -Steve
>
>
> On 19 December 2010 20:17, Octavio Duarte <xduvox at gmail.com> wrote:
> > Hello everyone
> >
> > i need some help with a gateway, i can receive calls from it because i
> can
> > use it as a user so i can see it by sofia status profile internal
> > but when i try to make a call to the PSTN like this
> >
> >
> > <action application="bridge"
> data="sofia/internal/userGateway%{domain}^$1"/>
> >
> > the gateway doesn't dial the number on variable $1
> >
> > so when to do like this
> >
> > <action application="bridge" data="sofia/internal/$1@<gateway_IP>"/>
> >
> >
> > the gateway connect the calls to the PSTN
> >
> > my problem its that the gateway does not have an static IP and i want to
> > know
> > if it is possible to make calls through this gateway if
> > it has a variable IP, and how to do it if so;
> >
> > the gateway uses SIP accounts to register to FS so i think it might be
> > possible to know the IP's gateway by using one of the registered
> extension
> > or tell fs to use the ip of one user for the gateway set up?
> >
> >
> > Anyone has ideas or knows how to do this?, help is greatly appreciated!
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Dec 20 21:09:38 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 20 Dec 2010 10:09:38 -0800
Subject: [Freeswitch-users] rebinding meta_app digit does not work on
	b-leg
In-Reply-To: <10356.1292765310@ccs.covici.com>
References: <10356.1292765310@ccs.covici.com>
Message-ID: <AANLkTikA9NoBfwOUEZHx_SNNrNYAukYuq==zU3bcS4L8@mail.gmail.com>

John,

I don't know if this is a bug or not, however I know for a fact that the new
bind_digit_action will do what you want. I highly recommend checking it out.
It is a bit more complicated at first, but once you start using it you will
never go back to bind_meta_app. :)

-MC

On Sun, Dec 19, 2010 at 5:28 AM, <covici at ccs.covici.com> wrote:

> Hi.  I have some dialplan code which allows me to press *2 to record and
> *2 to end the recording.  The way I do this is to rebind the digit by
> executing an extension.  Now if I call, it works fine, but if soneone
> calls me, it does not work -- looking at the logs I found out that even
> though it says it rebound the digit 2 on the b-leg, it actually still
> executes the previous bind of meta-app.
>
> Here is the pastebin to show the log:
> http://pastebin.freeswitch.org/14825
>
> Thanks in advance for any help.
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mel0torme at gmail.com  Mon Dec 20 21:13:06 2010
From: mel0torme at gmail.com (Tom C)
Date: Mon, 20 Dec 2010 10:13:06 -0800
Subject: [Freeswitch-users] Disable UPnP for "internal" port only?
In-Reply-To: <AANLkTinb6VhecA+W6fEUa+9=B03yD5OWqvQTqn7g_QNm@mail.gmail.com>
References: <AANLkTikm_Z6-eEQX_LjdOf6omYEsyvoRYF5cxh5pU6fM@mail.gmail.com>
	<AANLkTinb6VhecA+W6fEUa+9=B03yD5OWqvQTqn7g_QNm@mail.gmail.com>
Message-ID: <AANLkTi=pLcZUdr24-v=wb=qrD0S-Bnt+YmwyG+B6m9ey@mail.gmail.com>

Thanks for the reply.  I actually just routed the appropriate ports via port
forwarding in my router, and now -nonat does exactly what I want it to do.
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From msc at freeswitch.org  Mon Dec 20 21:36:04 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 20 Dec 2010 10:36:04 -0800
Subject: [Freeswitch-users] questions about VAD and echo cancellation
In-Reply-To: <AANLkTin0Q_NNJXdoFX4d-PYpkBKj2ct84Er5GAWCfoPf@mail.gmail.com>
References: <AANLkTi=G-gm4V9zV1hLtFVJ4B6-tyQY1Rw=+LEPdnaXa@mail.gmail.com>
	<AANLkTin0Q_NNJXdoFX4d-PYpkBKj2ct84Er5GAWCfoPf@mail.gmail.com>
Message-ID: <AANLkTi=xHUEy5zS+arr_Ud_aWL6rw+GkkzDi6SrtVGTV@mail.gmail.com>

What is NB?

On Sun, Dec 19, 2010 at 11:00 AM, joy this <thisjoy0528 at gmail.com> wrote:

> Could you give me any suggestions?
>
> Sincerely yours,
> Thisjoy.
> 2010/12/17 joy this <thisjoy0528 at gmail.com>
>
>> Dear all:
>>
>>
>>
>>         I have questions about VAD and echo cancellation. My FS is
>> Version 1.0.head (git-) under Windows XP. My soft-phone is X-Lite. I use
>> earphones and microphones for sip 1 which means the talking and hearing are
>> separated; on the other hand, I use NB for sip 2 which makes the talking and
>> hearing in the same place. Sip 1 (1000) calls sip 2 (1001) via FS. When I
>> say something via sip 1, the echo will occur, and the echo will only occur
>> on sip 1.
>>
>>         How do I cancel the echo? Buy a hardware card or something else?
>> I have two gateways which are Wellgate2644 and ata171m. Please give me some
>> suggestions.
>>
>>         When I enabled VAD, I found something strange. In the start of
>> the session, I can hear the noise and echo on sip 1. If I say something via
>> sip 2, the echo and the noise will disappear. Then the echo will not occur
>> on sip 1 only in a short time, about several seconds. The echo and the noise
>> will occur gradually if I say something via sip1. Is it a normal situation?
>>
>>
>>
>> Sincerely yours,
>> Thisjoy.
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From covici at ccs.covici.com  Mon Dec 20 22:41:17 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 20 Dec 2010 14:41:17 -0500
Subject: [Freeswitch-users] rebinding meta_app digit does not work on
	b-leg
In-Reply-To: <AANLkTikA9NoBfwOUEZHx_SNNrNYAukYuq==zU3bcS4L8@mail.gmail.com>
References: <10356.1292765310@ccs.covici.com>
	<AANLkTikA9NoBfwOUEZHx_SNNrNYAukYuq==zU3bcS4L8@mail.gmail.com>
Message-ID: <25987.1292874077@ccs.covici.com>

I looked up bind_digit_action, but the documentation was very sparse and
I could not figure out how to specify which leg, nor did it give the
meaning of the parameters.  If someone could explain this more, maybe it
would be useful.

Thanks.


Michael Collins <msc at freeswitch.org> wrote:

> John,
> 
> I don't know if this is a bug or not, however I know for a fact that the new
> bind_digit_action will do what you want. I highly recommend checking it out.
> It is a bit more complicated at first, but once you start using it you will
> never go back to bind_meta_app. :)
> 
> -MC
> 
> On Sun, Dec 19, 2010 at 5:28 AM, <covici at ccs.covici.com> wrote:
> 
> > Hi.  I have some dialplan code which allows me to press *2 to record and
> > *2 to end the recording.  The way I do this is to rebind the digit by
> > executing an extension.  Now if I call, it works fine, but if soneone
> > calls me, it does not work -- looking at the logs I found out that even
> > though it says it rebound the digit 2 on the b-leg, it actually still
> > executes the previous bind of meta-app.
> >
> > Here is the pastebin to show the log:
> > http://pastebin.freeswitch.org/14825
> >
> > Thanks in advance for any help.
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From steveayre at gmail.com  Mon Dec 20 22:48:43 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 20 Dec 2010 19:48:43 +0000
Subject: [Freeswitch-users] members audio conference
In-Reply-To: <AANLkTi=_DDyiOtBUDkJ9ET8FATyu3qntDK3PFA1xXi=7@mail.gmail.com>
References: <223AD13D879E4B19B15AEEDFC387386A@e1705>
	<AANLkTikydvXoJOk5kkWPYpaDqCGYTJgXbzj2=ffTB07p@mail.gmail.com>
	<532EC039F378476A9515E88F30BBB432@e1705>
	<AANLkTi=_DDyiOtBUDkJ9ET8FATyu3qntDK3PFA1xXi=7@mail.gmail.com>
Message-ID: <D43C6389-2A38-4DF7-9428-7D84D5D5EC9D@gmail.com>

More specifically FS core converts every member from their codec to L16 which is given to the conference. The conference combines all speaking channels and the resulting L16 is given back to the core to send to the members, the core converting to the correct codec for each member.

Steve on iPhone

On 20 Dec 2010, at 17:59, Michael Collins <msc at freeswitch.org> wrote:

> The conference itself doesn't do any codec stuff - FreeSWITCH core does. All I can say is compare the working versus non-working logs and look for clues.
> -MC
> 
> On Fri, Dec 17, 2010 at 6:49 PM, Madovsky <infos at madovsky.org> wrote:
> nohgint strange on logs.
> but I guess it's a codec and rate problem.
> tried with different SIP phones and it works.
> how a conference manage the codecs  ?
> I know the rate can be set in profile,
> but how conference codec is managed
>  if all members have different codec and rate ?
>  
>  
>  
> ----- Original Message -----
> From: Michael Collins
> To: FreeSWITCH Users Help
> Sent: Friday, December 17, 2010 7:54 PM
> Subject: Re: [Freeswitch-users] members audio conference
> 
> What do you see in the debug logs? Did you compare the logs for a working vs. non-working call? Anything different?
> 
> -MC
> 
> On Fri, Dec 17, 2010 at 7:35 AM, Madovsky <infos at madovsky.org> wrote:
> when the first member creates and enters in a new conference
> everything is ok. but if a new memeber enters there is no audio
> in the conference, unless the ivr.
>  
> I have a very simple conference dialplan like this
>  
>         <extension name="create_conference">
>                 <condition field="${sip_to_uri}"      expression="^000(\d{10,15})@$${domain}$">
>                         <action application="set"      data="instant_ringback=true"/>
>                         <action application="ring_ready"/>
>                         <action application="sleep" data="3000"/>
>                         <action application="answer"/>
>                         <action application="conference" data="$1-${domain_name}@wideband"/>
>                 </condition>
>         </extension>
>  
> I tried to add
> <action application="export" data="absolute_codec_string=speex at 16000k"/>
>  
> but no success
>  
> Thanks
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From infos at madovsky.org  Mon Dec 20 23:02:49 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 20 Dec 2010 15:02:49 -0500
Subject: [Freeswitch-users] members audio conference
References: <223AD13D879E4B19B15AEEDFC387386A@e1705><AANLkTikydvXoJOk5kkWPYpaDqCGYTJgXbzj2=ffTB07p@mail.gmail.com><532EC039F378476A9515E88F30BBB432@e1705><AANLkTi=_DDyiOtBUDkJ9ET8FATyu3qntDK3PFA1xXi=7@mail.gmail.com>
	<D43C6389-2A38-4DF7-9428-7D84D5D5EC9D@gmail.com>
Message-ID: <6AC2F3E2B53F494FB9A47BC9C5E314C3@e1705>

> FS core converts every member from their codec to L16 
ok I understand, so is it possible to change from L16 to another codec in a conference dialplan ?
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Monday, December 20, 2010 2:48 PM
  Subject: Re: [Freeswitch-users] members audio conference


  More specifically FS core converts every member from their codec to L16 which is given to the conference. The conference combines all speaking channels and the resulting L16 is given back to the core to send to the members, the core converting to the correct codec for each member.

  Steve on iPhone

  On 20 Dec 2010, at 17:59, Michael Collins <msc at freeswitch.org> wrote:


    The conference itself doesn't do any codec stuff - FreeSWITCH core does. All I can say is compare the working versus non-working logs and look for clues.
    -MC


    On Fri, Dec 17, 2010 at 6:49 PM, Madovsky <infos at madovsky.org> wrote:

      nohgint strange on logs.
      but I guess it's a codec and rate problem.
      tried with different SIP phones and it works.
      how a conference manage the codecs  ?
      I know the rate can be set in profile,
      but how conference codec is managed
       if all members have different codec and rate ?



        ----- Original Message ----- 
        From: Michael Collins 
        To: FreeSWITCH Users Help 
        Sent: Friday, December 17, 2010 7:54 PM
        Subject: Re: [Freeswitch-users] members audio conference


        What do you see in the debug logs? Did you compare the logs for a working vs. non-working call? Anything different? 


        -MC


        On Fri, Dec 17, 2010 at 7:35 AM, Madovsky <infos at madovsky.org> wrote:

          when the first member creates and enters in a new conference
          everything is ok. but if a new memeber enters there is no audio
          in the conference, unless the ivr.

          I have a very simple conference dialplan like this

                  <extension name="create_conference">
                          <condition field="${sip_to_uri}" expression="^000(\d{10,15})@$${domain}$">
                                  <action application="set" data="instant_ringback=true"/>
                                  <action application="ring_ready"/>
                                  <action application="sleep" data="3000"/>
                                  <action application="answer"/>
                                  <action application="conference" data="$1-${domain_name}@wideband"/>
                          </condition>
                  </extension>


          I tried to add
          <action application="export" data="absolute_codec_string=speex at 16000k"/>

          but no success

          Thanks

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






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


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



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




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



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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From steveayre at gmail.com  Tue Dec 21 00:09:31 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 20 Dec 2010 21:09:31 +0000
Subject: [Freeswitch-users] members audio conference
In-Reply-To: <6AC2F3E2B53F494FB9A47BC9C5E314C3@e1705>
References: <223AD13D879E4B19B15AEEDFC387386A@e1705>
	<AANLkTikydvXoJOk5kkWPYpaDqCGYTJgXbzj2=ffTB07p@mail.gmail.com>
	<532EC039F378476A9515E88F30BBB432@e1705>
	<AANLkTi=_DDyiOtBUDkJ9ET8FATyu3qntDK3PFA1xXi=7@mail.gmail.com>
	<D43C6389-2A38-4DF7-9428-7D84D5D5EC9D@gmail.com>
	<6AC2F3E2B53F494FB9A47BC9C5E314C3@e1705>
Message-ID: <E90F6461-422D-4119-B1E8-34EBA33EB972@gmail.com>

L16 is raw audio, not a codec.

It can't be changed. The only way to merge audio from multiple channels is to merge the raw audio samples data.

Steve on iPhone

On 20 Dec 2010, at 20:02, "Madovsky" <infos at madovsky.org> wrote:

> > FS core converts every member from their codec to L16
> ok I understand, so is it possible to change from L16 to another codec in a conference dialplan ?
> ----- Original Message -----
> From: Steven Ayre
> To: FreeSWITCH Users Help
> Sent: Monday, December 20, 2010 2:48 PM
> Subject: Re: [Freeswitch-users] members audio conference
> 
> More specifically FS core converts every member from their codec to L16 which is given to the conference. The conference combines all speaking channels and the resulting L16 is given back to the core to send to the members, the core converting to the correct codec for each member.
> 
> Steve on iPhone
> 
> On 20 Dec 2010, at 17:59, Michael Collins <msc at freeswitch.org> wrote:
> 
>> The conference itself doesn't do any codec stuff - FreeSWITCH core does. All I can say is compare the working versus non-working logs and look for clues.
>> -MC
>> 
>> On Fri, Dec 17, 2010 at 6:49 PM, Madovsky <infos at madovsky.org>      wrote:
>> nohgint strange on logs.
>> but I guess it's a codec and rate problem.
>> tried with different SIP phones and it works.
>> how a conference manage the codecs  ?
>> I know the rate can be set in profile,
>> but how conference codec is managed
>>  if all members have different codec and rate ?
>>  
>>  
>>  
>> ----- Original Message -----
>> From: Michael Collins
>> To: FreeSWITCH Users Help
>> Sent: Friday, December 17, 2010 7:54 PM
>> Subject: Re: [Freeswitch-users] members audio conference
>> 
>> What do you see in the debug logs? Did you compare the logs for a working vs. non-working call? Anything different?
>> 
>> -MC
>> 
>> On Fri, Dec 17, 2010 at 7:35 AM, Madovsky <infos at madovsky.org> wrote:
>> when the first member creates and enters in a new conference
>> everything is ok. but if a new memeber enters there is no audio
>> in the conference, unless the ivr.
>>  
>> I have a very simple conference dialplan like this
>>  
>>         <extension name="create_conference">
>>                 <condition field="${sip_to_uri}" expression="^000(\d{10,15})@$${domain}$">
>>                         <action application="set" data="instant_ringback=true"/>
>>                         <action application="ring_ready"/>
>>                         <action application="sleep" data="3000"/>
>>                         <action application="answer"/>
>>                         <action application="conference" data="$1-${domain_name}@wideband"/>
>>                 </condition>
>>         </extension>
>>  
>> I tried to add
>> <action application="export" data="absolute_codec_string=speex at 16000k"/>
>>  
>> but no success
>>  
>> Thanks
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
>> _______________________________________________
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>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
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From steveayre at gmail.com  Tue Dec 21 00:12:14 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 20 Dec 2010 21:12:14 +0000
Subject: [Freeswitch-users] members audio conference
In-Reply-To: <6AC2F3E2B53F494FB9A47BC9C5E314C3@e1705>
References: <223AD13D879E4B19B15AEEDFC387386A@e1705>
	<AANLkTikydvXoJOk5kkWPYpaDqCGYTJgXbzj2=ffTB07p@mail.gmail.com>
	<532EC039F378476A9515E88F30BBB432@e1705>
	<AANLkTi=_DDyiOtBUDkJ9ET8FATyu3qntDK3PFA1xXi=7@mail.gmail.com>
	<D43C6389-2A38-4DF7-9428-7D84D5D5EC9D@gmail.com>
	<6AC2F3E2B53F494FB9A47BC9C5E314C3@e1705>
Message-ID: <666DE35E-29AE-4593-89A7-C12AC2D2AF81@gmail.com>

Each member will connect to the codec with whichever codec they want to use. Conference always operates at raw audio level (L16). Core handles transcoding to convert between codecs. You can have members in the same conference using different codecs. You can control which codec a caller connects to the server with by adjusting the codec preferences on the sip profile.

Steve on iPhone

On 20 Dec 2010, at 20:02, "Madovsky" <infos at madovsky.org> wrote:

> > FS core converts every member from their codec to L16
> ok I understand, so is it possible to change from L16 to another codec in a conference dialplan ?
> ----- Original Message -----
> From: Steven Ayre
> To: FreeSWITCH Users Help
> Sent: Monday, December 20, 2010 2:48 PM
> Subject: Re: [Freeswitch-users] members audio conference
> 
> More specifically FS core converts every member from their codec to L16 which is given to the conference. The conference combines all speaking channels and the resulting L16 is given back to the core to send to the members, the core converting to the correct codec for each member.
> 
> Steve on iPhone
> 
> On 20 Dec 2010, at 17:59, Michael Collins <msc at freeswitch.org> wrote:
> 
>> The conference itself doesn't do any codec stuff - FreeSWITCH core does. All I can say is compare the working versus non-working logs and look for clues.
>> -MC
>> 
>> On Fri, Dec 17, 2010 at 6:49 PM, Madovsky <infos at madovsky.org>      wrote:
>> nohgint strange on logs.
>> but I guess it's a codec and rate problem.
>> tried with different SIP phones and it works.
>> how a conference manage the codecs  ?
>> I know the rate can be set in profile,
>> but how conference codec is managed
>>  if all members have different codec and rate ?
>>  
>>  
>>  
>> ----- Original Message -----
>> From: Michael Collins
>> To: FreeSWITCH Users Help
>> Sent: Friday, December 17, 2010 7:54 PM
>> Subject: Re: [Freeswitch-users] members audio conference
>> 
>> What do you see in the debug logs? Did you compare the logs for a working vs. non-working call? Anything different?
>> 
>> -MC
>> 
>> On Fri, Dec 17, 2010 at 7:35 AM, Madovsky <infos at madovsky.org> wrote:
>> when the first member creates and enters in a new conference
>> everything is ok. but if a new memeber enters there is no audio
>> in the conference, unless the ivr.
>>  
>> I have a very simple conference dialplan like this
>>  
>>         <extension name="create_conference">
>>                 <condition field="${sip_to_uri}" expression="^000(\d{10,15})@$${domain}$">
>>                         <action application="set" data="instant_ringback=true"/>
>>                         <action application="ring_ready"/>
>>                         <action application="sleep" data="3000"/>
>>                         <action application="answer"/>
>>                         <action application="conference" data="$1-${domain_name}@wideband"/>
>>                 </condition>
>>         </extension>
>>  
>> I tried to add
>> <action application="export" data="absolute_codec_string=speex at 16000k"/>
>>  
>> but no success
>>  
>> Thanks
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
>> _______________________________________________
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>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From steveayre at gmail.com  Tue Dec 21 00:12:51 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 20 Dec 2010 21:12:51 +0000
Subject: [Freeswitch-users] gateway without static IP
In-Reply-To: <AANLkTi=hsRskXoJtRN4B+WZe4_2SkiM6_AQeEpBjdcxb@mail.gmail.com>
References: <AANLkTi=kDe23KtpwxMd9S66JjrXhga1RK7+BWE5WAqV5@mail.gmail.com>
	<AANLkTi=KTyAGuqQMRTNcqxkyN1fqxP-z9NXikzeepKPM@mail.gmail.com>
	<AANLkTi=hsRskXoJtRN4B+WZe4_2SkiM6_AQeEpBjdcxb@mail.gmail.com>
Message-ID: <8DF4DE18-3B01-45CA-A89F-E511DBD2BA92@gmail.com>

Yes... Sorry :)

Steve on iPhone

On 20 Dec 2010, at 18:07, Michael Collins <msc at freeswitch.org> wrote:

> Just a note, the syntax for using a gateway is:
> 
> sofia/gateway/<gateway name>/<target>
> 
> If you do this:
> 
> sofia/<gateway name>/<target>
> 
> ...then FS will look for a *SIP profile* named <gateway name>. Hope this helps. :)
> 
> -MC
> 
> On Sun, Dec 19, 2010 at 2:00 PM, Steven Ayre <steveayre at gmail.com> wrote:
> First of all, I would  recommend you configure the gateway within the
> <gateways> section of the SIP profile. The dialstring will then be
> sofia/gwname/$1. That format has an advantage because FS will monitor
> the status of the gateway and automatically mark it as offline and
> stop sending traffic to it if it stops responding.
> 
> When configuring a gateway you can specify a domain name instead of an
> IP. If you combine that with a service such as www.dyndns.org you can
> run the gateway on a dynamic IP.
> 
> Also, have you tried:
> <action application="bridge" data="sofia/internal/$1@<domain>"/>
> that might work, but I've never tried it with anything but an IP so
> can't say for sure without trying it.
> 
> This does not do what you think it does:
> <action application="bridge" data="sofia/internal/userGateway%{domain}^$1"/>
> The % syntax is for dialing an extension on the *local* server, on the
> {domain} handled by the current server. That won't send a call out to
> a gateway.
> 
> -Steve
> 
> 
> On 19 December 2010 20:17, Octavio Duarte <xduvox at gmail.com> wrote:
> > Hello everyone
> >
> > i need some help with a gateway, i can receive calls from it because i can
> > use it as a user so i can see it by sofia status profile internal
> > but when i try to make a call to the PSTN like this
> >
> >
> > <action application="bridge" data="sofia/internal/userGateway%{domain}^$1"/>
> >
> > the gateway doesn't dial the number on variable $1
> >
> > so when to do like this
> >
> > <action application="bridge" data="sofia/internal/$1@<gateway_IP>"/>
> >
> >
> > the gateway connect the calls to the PSTN
> >
> > my problem its that the gateway does not have an static IP and i want to
> > know
> > if it is possible to make calls through this gateway if
> > it has a variable IP, and how to do it if so;
> >
> > the gateway uses SIP accounts to register to FS so i think it might be
> > possible to know the IP's gateway by using one of the registered extension
> > or tell fs to use the ip of one user for the gateway set up?
> >
> >
> > Anyone has ideas or knows how to do this?, help is greatly appreciated!
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From lucian at e-utile.ro  Tue Dec 21 00:19:40 2010
From: lucian at e-utile.ro (Lucian Marginean)
Date: Mon, 20 Dec 2010 16:19:40 -0500
Subject: [Freeswitch-users] On what condition P-Asserted-Identity is
 replaced by P-Preferred-Identity?
In-Reply-To: <AANLkTi=tU2ceXzoOjqPNzJJJVLafmB7EAjP8Gqz3NgF_@mail.gmail.com>
References: <AANLkTinZ3Vr_N=pB8QuvDVHUy2rCscz1yPLHfJFVQoQL@mail.gmail.com>
	<AANLkTi=tU2ceXzoOjqPNzJJJVLafmB7EAjP8Gqz3NgF_@mail.gmail.com>
Message-ID: <AANLkTikOvQj_R8zqO0hyZ9WWf38aDBrBfM-SDxehkddL@mail.gmail.com>

It is ok to put a note on
http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type like this:

Send P-Asserted-Identity

{sip_cid_type=pid}sofia/default/user at example.com


*Note: you must set privacy flag, otherwise will be inserted
P-Preferred-Identity instead of P-Asserted-Identity



*Lucian


On Mon, Dec 20, 2010 at 12:59 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> use the privacy app to set the proper flags and it should do asserted
>
> On Mon, Dec 20, 2010 at 10:18 AM, Lucian Marginean <lucian at e-utile.ro>
> wrote:
> > I try to setup P-Asserted-Identity on setup but I get
> P-Preferred-Identity
> > and I don't understand why
> >
> > Did I do something wrong or some other flags affect the PID feature?
> >
> > <action application="set" data="effective_caller_id_name=647476xxxx"/>
> > <action application="set" data="effective_caller_id_number=647788xxxx"/>
> > <action application="set" data="origination_privacy=screen"/>
> > <action application="bridge"
> >
> data="[sip_cid_type=pid,origination_caller_id_name=647476xxxx,origination_caller_id_number=647788xxxx]sofia/private/
> 905780xxxx at 10.10.67.82:5060"/>
> >
> > and invite generated look like this:
> >
> > U 10.10.67.83:5060 -> 10.10.67.82:5060
> > INVITE sip:905780xxxx at 10.10.67.82:5060 SIP/2.0.
> > Via: SIP/2.0/UDP 10.10.67.83;rport;branch=z9hG4bKH8KNm98v4Qv1N.
> > Max-Forwards: 68.
> > From: "647476xxxx" <sip:647788xxxx at 10.10.67.83<sip%3A647788xxxx at 10.10.67.83>
> >;tag=F1QF1Z26cBQZm.
> > To: <sip:905780xxxx at 10.10.67.82:5060>.
> > Call-ID: ff01d92d-86ec-122e-cba2-001143e7e50c.
> > CSeq: 6092608 INVITE.
> > Contact: <sip:mod_sofia at 10.10.67.83:5060>.
> > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-79ced28 2010-08-22 20-58-52
> > -0400.
> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER,
> > REFER, NOTIFY.
> > Supported: timer, precondition, path, replaces.
> > Allow-Events: talk, hold, refer.
> > Privacy: none.
> > Content-Type: application/sdp.
> > Content-Disposition: session.
> > Content-Length: 242.
> > X-FS-Support: update_display.
> > P-Preferred-Identity: "647476xxxx" <sip:647788xxxx at 10.10.67.83<sip%3A647788xxxx at 10.10.67.83>
> >
> >
> > Lucian
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Dec 21 00:29:09 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 20 Dec 2010 13:29:09 -0800
Subject: [Freeswitch-users] members audio conference
In-Reply-To: <6AC2F3E2B53F494FB9A47BC9C5E314C3@e1705>
References: <223AD13D879E4B19B15AEEDFC387386A@e1705>
	<AANLkTikydvXoJOk5kkWPYpaDqCGYTJgXbzj2=ffTB07p@mail.gmail.com>
	<532EC039F378476A9515E88F30BBB432@e1705>
	<AANLkTi=_DDyiOtBUDkJ9ET8FATyu3qntDK3PFA1xXi=7@mail.gmail.com>
	<D43C6389-2A38-4DF7-9428-7D84D5D5EC9D@gmail.com>
	<6AC2F3E2B53F494FB9A47BC9C5E314C3@e1705>
Message-ID: <AANLkTimuVtkLkOVYUk-5qqzj9bQZ-u0dj9BgaQj61SaT@mail.gmail.com>

On Mon, Dec 20, 2010 at 12:02 PM, Madovsky <infos at madovsky.org> wrote:

>  > FS core converts every member from their codec to L16
> ok I understand, so is it possible to change from L16 to another codec in a
> conference dialplan ?
>
I don't think you understood what Steven was saying. You don't need to worry
about the transcoding. As long as there is a codec that FS supports then the
transcoding happens without you having to do anything. Unless you've done
something silly like proxy media then you shouldn't need to worry about
codecs.

-MC

 ----- Original Message -----
> *From:* Steven Ayre <steveayre at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, December 20, 2010 2:48 PM
> *Subject:* Re: [Freeswitch-users] members audio conference
>
> More specifically FS core converts every member from their codec to L16
> which is given to the conference. The conference combines all speaking
> channels and the resulting L16 is given back to the core to send to the
> members, the core converting to the correct codec for each member.
>
> Steve on iPhone
>
> On 20 Dec 2010, at 17:59, Michael Collins <msc at freeswitch.org> wrote:
>
>  The conference itself doesn't do any codec stuff - FreeSWITCH core does.
> All I can say is compare the working versus non-working logs and look for
> clues.
> -MC
>
> On Fri, Dec 17, 2010 at 6:49 PM, Madovsky < <infos at madovsky.org>
> infos at madovsky.org> wrote:
>
>>  nohgint strange on logs.
>> but I guess it's a codec and rate problem.
>> tried with different SIP phones and it works.
>> how a conference manage the codecs  ?
>> I know the rate can be set in profile,
>> but how conference codec is managed
>>  if all members have different codec and rate ?
>>
>>
>>
>>
>>   ----- Original Message -----
>> *From:* Michael Collins <msc at freeswitch.org>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Friday, December 17, 2010 7:54 PM
>> *Subject:* Re: [Freeswitch-users] members audio conference
>>
>> What do you see in the debug logs? Did you compare the logs for a working
>> vs. non-working call? Anything different?
>>
>> -MC
>>
>> On Fri, Dec 17, 2010 at 7:35 AM, Madovsky < <infos at madovsky.org>
>> infos at madovsky.org> wrote:
>>
>>>  when the first member creates and enters in a new conference
>>> everything is ok. but if a new memeber enters there is no audio
>>> in the conference, unless the ivr.
>>>
>>> I have a very simple conference dialplan like this
>>>
>>>         <extension name="create_conference">
>>>                 <condition field="${sip_to_uri}"
>>> expression="^000(\d{10,15})@$${domain}$">
>>>                         <action application="set"
>>> data="instant_ringback=true"/>
>>>                         <action application="ring_ready"/>
>>>                         <action application="sleep" data="3000"/>
>>>                         <action application="answer"/>
>>>                         <action application="conference" data="
>>> $1-${domain_name}@wideband"/ <$1-$%7Bdomain_name%7D at wideband%22/>>
>>>                 </condition>
>>>         </extension>
>>>
>>> I tried to add
>>> <action application="export" data="absolute_codec_string=speex at 16000k"/<absolute_codec_string=speex at 16000k%22/>
>>> >
>>>
>>> but no success
>>>
>>> Thanks
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> <http://www.freeswitch.org>http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> <http://www.freeswitch.org>http://www.freeswitch.org
>>
>>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From shamun.toha at gmail.com  Tue Dec 21 01:47:06 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Mon, 20 Dec 2010 23:47:06 +0100
Subject: [Freeswitch-users] My sip client program gets fail to login
Message-ID: <AANLkTimUhPDO8xu8LSgf134HH=oAaXOasbmZK2eC3ZAT@mail.gmail.com>

I was writing a small application i never get it working with FreeSwitch sip
login. This is my packets switching i get always auth fail. What is there
missing ?


recv 407 bytes from udp/[78.23.93.41]:5091 at 22:42:15.817398:
   ------------------------------------------------------------------------
   REGISTER sip:78.23.93.41 SIP/2.0
   Via: SIP/2.0/UDP localhost:5091;rport;branch=z9hG4bK2049230244
   Max-forwards: 70
   From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
>;tag=2049230244
   To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>
   Call-ID: @localhost
   CSeq: 400 REGISTER
   Contact: <sip:1002 at localhost:5091>;expires=3600
   Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
   User-agent: test
   Content-Length: 0

   ------------------------------------------------------------------------
send 634 bytes to udp/[78.23.93.41]:5091 at 22:42:16.072075:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP
localhost:5091;rport=5091;branch=z9hG4bK2049230244;received=78.23.93.41
   From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
>;tag=2049230244
   To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>;tag=XD2H1ja8U4jyr
   Call-ID: @localhost
   CSeq: 400 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-828960a 2010-09-25 12-51-42
-0500
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   WWW-Authenticate: Digest realm="78.23.93.41",
nonce="64da83a6-0c8a-11e0-a943-13f4f379a0fc", algorithm=MD5, qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 540 bytes from udp/[78.23.93.41]:5091 at 22:42:16.091547:
   ------------------------------------------------------------------------
   REGISTER sip:78.23.93.41 SIP/2.0
   Via: SIP/2.0/UDP localhost:5091;rport;branch=z9hG4bK2049230244
   To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>
   From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
>;tag=2049230244
   Call-ID: @localhost
   CSeq: 402 REGISTER
   Contact: <sip:1002 at localhost:5091>;expires=3600
   Authorization: Digest username="1002", realm="78.23.93.41",
nonce="64da83a6-0c8a-11e0-a943-13f4f379a0fc", uri="sip:78.23.93.41",
response="e747a46dbb1169661e9435773921cff5", qop="auth", algorithm="MD5",
cnonce="", nc="000000001"
   User-agent: test
   Content-Length: 0

   ------------------------------------------------------------------------
2010-12-20 23:42:16.328313 [WARNING] sofia_reg.c:1088 SIP auth failure
(REGISTER) on sofia profile 'internal' for [1002 at 78.23.93.41] from ip
78.23.93.41
send 512 bytes to udp/[78.23.93.41]:5091 at 22:42:16.330038:
   ------------------------------------------------------------------------
   SIP/2.0 403 Forbidden
   Via: SIP/2.0/UDP
localhost:5091;rport=5091;branch=z9hG4bK2049230244;received=78.23.93.41
   From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
>;tag=2049230244
   To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>;tag=ypUa3DUBSD9gm
   Call-ID: @localhost
   CSeq: 402 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-828960a 2010-09-25 12-51-42
-0500
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
recv 268 bytes from udp/[78.23.93.41]:5091 at 22:42:16.334712:
   ------------------------------------------------------------------------
   ACK sip:1002 at 78.23.93.41:5060 SIP/2.0
   Via: SIP/2.0/UDP null:5091;rport;branch=z9hG4bK2049258514
   From: 1002 <sip:1002 at 78.23.93.41:5060>;tag=2049258514
   To: <sip:1002 at 78.23.93.41:5060>;tag=tag=ypUa3DUBSD9gm
   Call-ID: 58514 at localhost
   CSeq: 0 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
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From rajil.s at gmail.com  Tue Dec 21 01:48:19 2010
From: rajil.s at gmail.com (Rajil Saraswat)
Date: Mon, 20 Dec 2010 22:48:19 +0000
Subject: [Freeswitch-users] Nokia N900 and poor Voicemail IVR audio
Message-ID: <4d0fdd1f.017b0e0a.437e.ffffa766@mx.google.com>

Hi all,
For some strange reason my Nokia N900 doesn't play the audio for the digits of the IVR. For example it will play "For advanced options Press" but will not play number 5. The IVR audio works fine if i call using ekiga.
Is there explanation for this? Initially i thought this was my FS server issue but i have now tested it against somebody elses FS with the same response.  
Thanks



From jerre at j-cope.com  Tue Dec 21 01:48:21 2010
From: jerre at j-cope.com (Jerre Cope)
Date: Mon, 20 Dec 2010 16:48:21 -0600
Subject: [Freeswitch-users] Incomplete offer/answer debugging
Message-ID: <4D0FDD35.5030208@j-cope.com>

SOLVED.

I modified global_codec_prefs=G772,PCMU,PCMA,GSM

and now the phones don't barf trying to negotiate codecs they don't 
understand.

I made an note on the interop wiki with an entry for that phone in case 
anyone else in the world has one.




From brian at freeswitch.org  Tue Dec 21 01:54:54 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 20 Dec 2010 16:54:54 -0600
Subject: [Freeswitch-users] Incomplete offer/answer debugging
In-Reply-To: <4D0FDD35.5030208@j-cope.com>
References: <4D0FDD35.5030208@j-cope.com>
Message-ID: <120D34CE-B309-4A42-8FBD-7622A52D9FC7@freeswitch.org>

Do you mean G722?  G772 is?  :P

/b

On Dec 20, 2010, at 4:48 PM, Jerre Cope wrote:

> I modified global_codec_prefs=G772,PCMU,PCMA,GSM



From msc at freeswitch.org  Tue Dec 21 01:55:12 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 20 Dec 2010 14:55:12 -0800
Subject: [Freeswitch-users] My sip client program gets fail to login
In-Reply-To: <AANLkTimUhPDO8xu8LSgf134HH=oAaXOasbmZK2eC3ZAT@mail.gmail.com>
References: <AANLkTimUhPDO8xu8LSgf134HH=oAaXOasbmZK2eC3ZAT@mail.gmail.com>
Message-ID: <AANLkTinsZAr4USY5E46F3qr+EePt+p6ryJJMR_KmaArK@mail.gmail.com>

Well, I don't claim to be a genius in these sorts of things but I don't know
if your nonce count field is correct. IIRC it needs to be 8 characters long
and yours appears to be 9 characters long. Check your app to make sure it
uses a proper nc="00000001" and see what happens.

-MC

On Mon, Dec 20, 2010 at 2:47 PM, Shamun toha md <shamun.toha at gmail.com>wrote:

> I was writing a small application i never get it working with FreeSwitch
> sip login. This is my packets switching i get always auth fail. What is
> there missing ?
>
>
> recv 407 bytes from udp/[78.23.93.41]:5091 at 22:42:15.817398:
>    ------------------------------------------------------------------------
>    REGISTER sip:78.23.93.41 SIP/2.0
>    Via: SIP/2.0/UDP localhost:5091;rport;branch=z9hG4bK2049230244
>    Max-forwards: 70
>    From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
> >;tag=2049230244
>    To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>
>    Call-ID: @localhost
>    CSeq: 400 REGISTER
>    Contact: <sip:1002 at localhost:5091>;expires=3600
>    Allow:
> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
>    User-agent: test
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 634 bytes to udp/[78.23.93.41]:5091 at 22:42:16.072075:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP
> localhost:5091;rport=5091;branch=z9hG4bK2049230244;received=78.23.93.41
>    From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
> >;tag=2049230244
>    To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>;tag=XD2H1ja8U4jyr
>    Call-ID: @localhost
>    CSeq: 400 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-828960a 2010-09-25
> 12-51-42 -0500
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    WWW-Authenticate: Digest realm="78.23.93.41",
> nonce="64da83a6-0c8a-11e0-a943-13f4f379a0fc", algorithm=MD5, qop="auth"
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 540 bytes from udp/[78.23.93.41]:5091 at 22:42:16.091547:
>    ------------------------------------------------------------------------
>    REGISTER sip:78.23.93.41 SIP/2.0
>    Via: SIP/2.0/UDP localhost:5091;rport;branch=z9hG4bK2049230244
>    To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>
>    From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
> >;tag=2049230244
>    Call-ID: @localhost
>    CSeq: 402 REGISTER
>    Contact: <sip:1002 at localhost:5091>;expires=3600
>    Authorization: Digest username="1002", realm="78.23.93.41",
> nonce="64da83a6-0c8a-11e0-a943-13f4f379a0fc", uri="sip:78.23.93.41",
> response="e747a46dbb1169661e9435773921cff5", qop="auth", algorithm="MD5",
> cnonce="", nc="000000001"
>    User-agent: test
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2010-12-20 23:42:16.328313 [WARNING] sofia_reg.c:1088 SIP auth failure
> (REGISTER) on sofia profile 'internal' for [1002 at 78.23.93.41] from ip
> 78.23.93.41
> send 512 bytes to udp/[78.23.93.41]:5091 at 22:42:16.330038:
>    ------------------------------------------------------------------------
>    SIP/2.0 403 Forbidden
>    Via: SIP/2.0/UDP
> localhost:5091;rport=5091;branch=z9hG4bK2049230244;received=78.23.93.41
>    From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
> >;tag=2049230244
>    To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>;tag=ypUa3DUBSD9gm
>    Call-ID: @localhost
>    CSeq: 402 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-828960a 2010-09-25
> 12-51-42 -0500
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 268 bytes from udp/[78.23.93.41]:5091 at 22:42:16.334712:
>    ------------------------------------------------------------------------
>    ACK sip:1002 at 78.23.93.41:5060 SIP/2.0
>    Via: SIP/2.0/UDP null:5091;rport;branch=z9hG4bK2049258514
>    From: 1002 <sip:1002 at 78.23.93.41:5060>;tag=2049258514
>    To: <sip:1002 at 78.23.93.41:5060>;tag=tag=ypUa3DUBSD9gm
>    Call-ID: 58514 at localhost
>    CSeq: 0 ACK
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Dec 21 01:56:45 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 20 Dec 2010 14:56:45 -0800
Subject: [Freeswitch-users] Nokia N900 and poor Voicemail IVR audio
In-Reply-To: <4d0fdd1f.017b0e0a.437e.ffffa766@mx.google.com>
References: <4d0fdd1f.017b0e0a.437e.ffffa766@mx.google.com>
Message-ID: <AANLkTi=2p_Lf15g2wWQWSSbn_5VV4MLijN4z-E3S-MiW@mail.gmail.com>

This is an interesting symptom. Please get a console debug of the call. Use
this page for guidance on how to collect the information and put in into
pastebin:

http://wiki.freeswitch.org/wiki/Reporting_Bugs

-MC

On Mon, Dec 20, 2010 at 2:48 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:

> Hi all,
> For some strange reason my Nokia N900 doesn't play the audio for the digits
> of the IVR. For example it will play "For advanced options Press" but will
> not play number 5. The IVR audio works fine if i call using ekiga.
> Is there explanation for this? Initially i thought this was my FS server
> issue but i have now tested it against somebody elses FS with the same
> response.
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From shamun.toha at gmail.com  Tue Dec 21 02:08:47 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Tue, 21 Dec 2010 00:08:47 +0100
Subject: [Freeswitch-users] My sip client program gets fail to login
In-Reply-To: <AANLkTinsZAr4USY5E46F3qr+EePt+p6ryJJMR_KmaArK@mail.gmail.com>
References: <AANLkTimUhPDO8xu8LSgf134HH=oAaXOasbmZK2eC3ZAT@mail.gmail.com>
	<AANLkTinsZAr4USY5E46F3qr+EePt+p6ryJJMR_KmaArK@mail.gmail.com>
Message-ID: <AANLkTi=r=YBB5U3oeW5x6S5kcTkJ_uOLP4J-4afdoqzU@mail.gmail.com>

I changed it to 8 now, but here is the same result.

   ------------------------------------------------------------------------
send 617 bytes to udp/[78.23.93.41]:5091 at 23:07:22.171723:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP 78.23.93.41:5091;rport=5091;branch=z9hG4bK2049216478
   From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
>;tag=2049216478
   To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>;tag=33S6BNFXatree
   Call-ID: @78.23.93.41
   CSeq: 400 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-828960a 2010-09-25 12-51-42
-0500
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   WWW-Authenticate: Digest realm="78.23.93.41",
nonce="e6942c46-0c8d-11e0-a946-13f4f379a0fc", algorithm=MD5, qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 545 bytes from udp/[78.23.93.41]:5091 at 23:07:22.184403:
   ------------------------------------------------------------------------
   REGISTER sip:78.23.93.41 SIP/2.0
   Via: SIP/2.0/UDP 78.23.93.41:5091;rport;branch=z9hG4bK2049216478
   To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>
   From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
>;tag=2049216478
   Call-ID: @78.23.93.41
   CSeq: 402 REGISTER
   Contact: <sip:1002 at 78.23.93.41:5091>;expires=3600
   Authorization: Digest username="1002", realm="78.23.93.41",
nonce="e6942c46-0c8d-11e0-a946-13f4f379a0fc", uri="sip:78.23.93.41",
response="294b6b91485aaa66e917c1378736267b", qop="auth", algorithm="MD5",
cnonce="", nc="00000001"
   User-agent: portsip
   Content-Length: 0

   ------------------------------------------------------------------------
2010-12-21 00:07:22.394599 [WARNING] sofia_reg.c:1088 SIP auth failure
(REGISTER) on sofia profile 'internal' for [1002 at 78.23.93.41] from ip
78.23.93.41


send 495 bytes to udp/[78.23.93.41]:5091 at 23:07:22.396247:
   ------------------------------------------------------------------------
   SIP/2.0 403 Forbidden
   Via: SIP/2.0/UDP 78.23.93.41:5091;rport=5091;branch=z9hG4bK2049216478
   From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
>;tag=2049216478
   To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>;tag=4cKZDg0072e1S
   Call-ID: @78.23.93.41
   CSeq: 402 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-828960a 2010-09-25 12-51-42
-0500
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Content-Length: 0





On Mon, Dec 20, 2010 at 11:55 PM, Michael Collins <msc at freeswitch.org>wrote:

> Well, I don't claim to be a genius in these sorts of things but I don't
> know if your nonce count field is correct. IIRC it needs to be 8 characters
> long and yours appears to be 9 characters long. Check your app to make sure
> it uses a proper nc="00000001" and see what happens.
>
> -MC
>
> On Mon, Dec 20, 2010 at 2:47 PM, Shamun toha md <shamun.toha at gmail.com>wrote:
>
>> I was writing a small application i never get it working with FreeSwitch
>> sip login. This is my packets switching i get always auth fail. What is
>> there missing ?
>>
>>
>> recv 407 bytes from udp/[78.23.93.41]:5091 at 22:42:15.817398:
>>
>> ------------------------------------------------------------------------
>>    REGISTER sip:78.23.93.41 SIP/2.0
>>    Via: SIP/2.0/UDP localhost:5091;rport;branch=z9hG4bK2049230244
>>    Max-forwards: 70
>>    From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
>> >;tag=2049230244
>>    To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>
>>    Call-ID: @localhost
>>    CSeq: 400 REGISTER
>>    Contact: <sip:1002 at localhost:5091>;expires=3600
>>    Allow:
>> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
>>    User-agent: test
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> send 634 bytes to udp/[78.23.93.41]:5091 at 22:42:16.072075:
>>
>> ------------------------------------------------------------------------
>>    SIP/2.0 401 Unauthorized
>>    Via: SIP/2.0/UDP
>> localhost:5091;rport=5091;branch=z9hG4bK2049230244;received=78.23.93.41
>>    From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
>> >;tag=2049230244
>>    To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>;tag=XD2H1ja8U4jyr
>>    Call-ID: @localhost
>>    CSeq: 400 REGISTER
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-828960a 2010-09-25
>> 12-51-42 -0500
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>    Supported: timer, precondition, path, replaces
>>    WWW-Authenticate: Digest realm="78.23.93.41",
>> nonce="64da83a6-0c8a-11e0-a943-13f4f379a0fc", algorithm=MD5, qop="auth"
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> recv 540 bytes from udp/[78.23.93.41]:5091 at 22:42:16.091547:
>>
>> ------------------------------------------------------------------------
>>    REGISTER sip:78.23.93.41 SIP/2.0
>>    Via: SIP/2.0/UDP localhost:5091;rport;branch=z9hG4bK2049230244
>>    To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>
>>    From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
>> >;tag=2049230244
>>    Call-ID: @localhost
>>    CSeq: 402 REGISTER
>>    Contact: <sip:1002 at localhost:5091>;expires=3600
>>    Authorization: Digest username="1002", realm="78.23.93.41",
>> nonce="64da83a6-0c8a-11e0-a943-13f4f379a0fc", uri="sip:78.23.93.41",
>> response="e747a46dbb1169661e9435773921cff5", qop="auth", algorithm="MD5",
>> cnonce="", nc="000000001"
>>    User-agent: test
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> 2010-12-20 23:42:16.328313 [WARNING] sofia_reg.c:1088 SIP auth failure
>> (REGISTER) on sofia profile 'internal' for [1002 at 78.23.93.41] from ip
>> 78.23.93.41
>> send 512 bytes to udp/[78.23.93.41]:5091 at 22:42:16.330038:
>>
>> ------------------------------------------------------------------------
>>    SIP/2.0 403 Forbidden
>>    Via: SIP/2.0/UDP
>> localhost:5091;rport=5091;branch=z9hG4bK2049230244;received=78.23.93.41
>>    From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
>> >;tag=2049230244
>>    To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>;tag=ypUa3DUBSD9gm
>>    Call-ID: @localhost
>>    CSeq: 402 REGISTER
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-828960a 2010-09-25
>> 12-51-42 -0500
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>    Supported: timer, precondition, path, replaces
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> recv 268 bytes from udp/[78.23.93.41]:5091 at 22:42:16.334712:
>>
>> ------------------------------------------------------------------------
>>    ACK sip:1002 at 78.23.93.41:5060 SIP/2.0
>>    Via: SIP/2.0/UDP null:5091;rport;branch=z9hG4bK2049258514
>>    From: 1002 <sip:1002 at 78.23.93.41:5060>;tag=2049258514
>>    To: <sip:1002 at 78.23.93.41:5060>;tag=tag=ypUa3DUBSD9gm
>>    Call-ID: 58514 at localhost
>>    CSeq: 0 ACK
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Dec 21 02:10:22 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 20 Dec 2010 15:10:22 -0800
Subject: [Freeswitch-users] rebinding meta_app digit does not work on
	b-leg
In-Reply-To: <25987.1292874077@ccs.covici.com>
References: <10356.1292765310@ccs.covici.com>
	<AANLkTikA9NoBfwOUEZHx_SNNrNYAukYuq==zU3bcS4L8@mail.gmail.com>
	<25987.1292874077@ccs.covici.com>
Message-ID: <AANLkTincRa0cMNYc=GuTv7X7vTWHMd6eKxgW0PcaoYdi@mail.gmail.com>

If someone has a few minutes to write up a nice bind_digit_action how-to,
including strategies for handling the b-leg, before Wednesday then we would
appreciate that. If not then let's have this be a subject of discussion on
Wednesday's conf call. This is definitely something that every FS admin
should become familiar with, especially if you are doing any kind of DTMF
input from your callers.

-MC

On Mon, Dec 20, 2010 at 11:41 AM, <covici at ccs.covici.com> wrote:

> I looked up bind_digit_action, but the documentation was very sparse and
> I could not figure out how to specify which leg, nor did it give the
> meaning of the parameters.  If someone could explain this more, maybe it
> would be useful.
>
> Thanks.
>
>
> Michael Collins <msc at freeswitch.org> wrote:
>
> > John,
> >
> > I don't know if this is a bug or not, however I know for a fact that the
> new
> > bind_digit_action will do what you want. I highly recommend checking it
> out.
> > It is a bit more complicated at first, but once you start using it you
> will
> > never go back to bind_meta_app. :)
> >
> > -MC
> >
> > On Sun, Dec 19, 2010 at 5:28 AM, <covici at ccs.covici.com> wrote:
> >
> > > Hi.  I have some dialplan code which allows me to press *2 to record
> and
> > > *2 to end the recording.  The way I do this is to rebind the digit by
> > > executing an extension.  Now if I call, it works fine, but if soneone
> > > calls me, it does not work -- looking at the logs I found out that even
> > > though it says it rebound the digit 2 on the b-leg, it actually still
> > > executes the previous bind of meta-app.
> > >
> > > Here is the pastebin to show the log:
> > > http://pastebin.freeswitch.org/14825
> > >
> > > Thanks in advance for any help.
> > > --
> > > Your life is like a penny.  You're going to lose it.  The question is:
> > > How do
> > > you spend it?
> > >
> > >         John Covici
> > >         covici at ccs.covici.com
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> >
> > ----------------------------------------------------
> > Alternatives:
> >
> > ----------------------------------------------------
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From shamun.toha at gmail.com  Tue Dec 21 02:14:50 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Tue, 21 Dec 2010 00:14:50 +0100
Subject: [Freeswitch-users] My sip client program gets fail to login
In-Reply-To: <AANLkTinsZAr4USY5E46F3qr+EePt+p6ryJJMR_KmaArK@mail.gmail.com>
References: <AANLkTimUhPDO8xu8LSgf134HH=oAaXOasbmZK2eC3ZAT@mail.gmail.com>
	<AANLkTinsZAr4USY5E46F3qr+EePt+p6ryJJMR_KmaArK@mail.gmail.com>
Message-ID: <AANLkTik-Nbz0NNzrKtAJFw3R7kXi5XsPq4-Nij33mnZO@mail.gmail.com>

My nonce is correct i guess here is the formula i am using:
nonce="e6942c46-0c8d-11e0-a946-13f4f379a0fc",
uri="sip:78.23.93.41",
response="294b6b91485aaa66e917c1378736267b"

   /** Calculates the digest-response.
     * <p> If the "qop" value is "auth" or "auth-int":
     * <br>   KD ( H(A1), unq(nonce) ":" nc ":" unq(cnonce) ":" unq(qop) ":"
H(A2) )
     *
     * <p> If the "qop" directive is not present:
     * <br>   KD ( H(A1), unq(nonce) ":" H(A2) )
     */
   public String getResponse()
   {  String secret=HEX(MD5(A1()));
      StringBuffer sb=new StringBuffer();
      if (nonce!=null) sb.append(nonce);
      sb.append(":");
      if (qop!=null)
      {  if (nc!=null) sb.append(nc);
         sb.append(":");
         if (cnonce!=null) sb.append(cnonce);
         sb.append(":");
         sb.append(qop);
         sb.append(":");
      }
      sb.append(HEX(MD5(A2())));
      String data=sb.toString();
      return HEX(KD(secret,data));
   }




On Mon, Dec 20, 2010 at 11:55 PM, Michael Collins <msc at freeswitch.org>wrote:

> Well, I don't claim to be a genius in these sorts of things but I don't
> know if your nonce count field is correct. IIRC it needs to be 8 characters
> long and yours appears to be 9 characters long. Check your app to make sure
> it uses a proper nc="00000001" and see what happens.
>
> -MC
>
> On Mon, Dec 20, 2010 at 2:47 PM, Shamun toha md <shamun.toha at gmail.com>wrote:
>
>> I was writing a small application i never get it working with FreeSwitch
>> sip login. This is my packets switching i get always auth fail. What is
>> there missing ?
>>
>>
>> recv 407 bytes from udp/[78.23.93.41]:5091 at 22:42:15.817398:
>>
>> ------------------------------------------------------------------------
>>    REGISTER sip:78.23.93.41 SIP/2.0
>>    Via: SIP/2.0/UDP localhost:5091;rport;branch=z9hG4bK2049230244
>>    Max-forwards: 70
>>    From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
>> >;tag=2049230244
>>    To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>
>>    Call-ID: @localhost
>>    CSeq: 400 REGISTER
>>    Contact: <sip:1002 at localhost:5091>;expires=3600
>>    Allow:
>> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
>>    User-agent: test
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> send 634 bytes to udp/[78.23.93.41]:5091 at 22:42:16.072075:
>>
>> ------------------------------------------------------------------------
>>    SIP/2.0 401 Unauthorized
>>    Via: SIP/2.0/UDP
>> localhost:5091;rport=5091;branch=z9hG4bK2049230244;received=78.23.93.41
>>    From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
>> >;tag=2049230244
>>    To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>;tag=XD2H1ja8U4jyr
>>    Call-ID: @localhost
>>    CSeq: 400 REGISTER
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-828960a 2010-09-25
>> 12-51-42 -0500
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>    Supported: timer, precondition, path, replaces
>>    WWW-Authenticate: Digest realm="78.23.93.41",
>> nonce="64da83a6-0c8a-11e0-a943-13f4f379a0fc", algorithm=MD5, qop="auth"
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> recv 540 bytes from udp/[78.23.93.41]:5091 at 22:42:16.091547:
>>
>> ------------------------------------------------------------------------
>>    REGISTER sip:78.23.93.41 SIP/2.0
>>    Via: SIP/2.0/UDP localhost:5091;rport;branch=z9hG4bK2049230244
>>    To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>
>>    From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
>> >;tag=2049230244
>>    Call-ID: @localhost
>>    CSeq: 402 REGISTER
>>    Contact: <sip:1002 at localhost:5091>;expires=3600
>>    Authorization: Digest username="1002", realm="78.23.93.41",
>> nonce="64da83a6-0c8a-11e0-a943-13f4f379a0fc", uri="sip:78.23.93.41",
>> response="e747a46dbb1169661e9435773921cff5", qop="auth", algorithm="MD5",
>> cnonce="", nc="000000001"
>>    User-agent: test
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> 2010-12-20 23:42:16.328313 [WARNING] sofia_reg.c:1088 SIP auth failure
>> (REGISTER) on sofia profile 'internal' for [1002 at 78.23.93.41] from ip
>> 78.23.93.41
>> send 512 bytes to udp/[78.23.93.41]:5091 at 22:42:16.330038:
>>
>> ------------------------------------------------------------------------
>>    SIP/2.0 403 Forbidden
>>    Via: SIP/2.0/UDP
>> localhost:5091;rport=5091;branch=z9hG4bK2049230244;received=78.23.93.41
>>    From: "1002" <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>
>> >;tag=2049230244
>>    To: <sip:1002 at 78.23.93.41 <sip%3A1002 at 78.23.93.41>>;tag=ypUa3DUBSD9gm
>>    Call-ID: @localhost
>>    CSeq: 402 REGISTER
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-828960a 2010-09-25
>> 12-51-42 -0500
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>    Supported: timer, precondition, path, replaces
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> recv 268 bytes from udp/[78.23.93.41]:5091 at 22:42:16.334712:
>>
>> ------------------------------------------------------------------------
>>    ACK sip:1002 at 78.23.93.41:5060 SIP/2.0
>>    Via: SIP/2.0/UDP null:5091;rport;branch=z9hG4bK2049258514
>>    From: 1002 <sip:1002 at 78.23.93.41:5060>;tag=2049258514
>>    To: <sip:1002 at 78.23.93.41:5060>;tag=tag=ypUa3DUBSD9gm
>>    Call-ID: 58514 at localhost
>>    CSeq: 0 ACK
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From jerre at j-cope.com  Tue Dec 21 02:15:18 2010
From: jerre at j-cope.com (Jerre Cope)
Date: Mon, 20 Dec 2010 17:15:18 -0600
Subject: [Freeswitch-users] Incomplete offer/answer debugging
In-Reply-To: <mailman.820.1292885744.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.820.1292885744.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <4D0FE386.5010406@j-cope.com>

Dang!  At least I got the wiki entry right.  I hate it when I miss-spell 
numbers!


From infos at madovsky.org  Tue Dec 21 03:48:36 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 20 Dec 2010 19:48:36 -0500
Subject: [Freeswitch-users] members audio conference
References: <223AD13D879E4B19B15AEEDFC387386A@e1705><AANLkTikydvXoJOk5kkWPYpaDqCGYTJgXbzj2=ffTB07p@mail.gmail.com><532EC039F378476A9515E88F30BBB432@e1705><AANLkTi=_DDyiOtBUDkJ9ET8FATyu3qntDK3PFA1xXi=7@mail.gmail.com><D43C6389-2A38-4DF7-9428-7D84D5D5EC9D@gmail.com><6AC2F3E2B53F494FB9A47BC9C5E314C3@e1705>
	<AANLkTimuVtkLkOVYUk-5qqzj9bQZ-u0dj9BgaQj61SaT@mail.gmail.com>
Message-ID: <2CF4BD5149DA49AC96278241CEDC6EE4@e1705>

I did a lot of silly things since I learn FS   :D
but not to try to transcode on FS if I use a proxy media....
I would just to minimize the transcoding....
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, December 20, 2010 4:29 PM
  Subject: Re: [Freeswitch-users] members audio conference





  On Mon, Dec 20, 2010 at 12:02 PM, Madovsky <infos at madovsky.org> wrote:

    > FS core converts every member from their codec to L16 
    ok I understand, so is it possible to change from L16 to another codec in a conference dialplan ?
  I don't think you understood what Steven was saying. You don't need to worry about the transcoding. As long as there is a codec that FS supports then the transcoding happens without you having to do anything. Unless you've done something silly like proxy media then you shouldn't need to worry about codecs.


  -MC


      ----- Original Message ----- 
      From: Steven Ayre 
      To: FreeSWITCH Users Help 
      Sent: Monday, December 20, 2010 2:48 PM
      Subject: Re: [Freeswitch-users] members audio conference


      More specifically FS core converts every member from their codec to L16 which is given to the conference. The conference combines all speaking channels and the resulting L16 is given back to the core to send to the members, the core converting to the correct codec for each member.

      Steve on iPhone

      On 20 Dec 2010, at 17:59, Michael Collins <msc at freeswitch.org> wrote:


        The conference itself doesn't do any codec stuff - FreeSWITCH core does. All I can say is compare the working versus non-working logs and look for clues. 
        -MC


        On Fri, Dec 17, 2010 at 6:49 PM, Madovsky <infos at madovsky.org> wrote:

          nohgint strange on logs.
          but I guess it's a codec and rate problem.
          tried with different SIP phones and it works.
          how a conference manage the codecs  ?
          I know the rate can be set in profile,
          but how conference codec is managed
           if all members have different codec and rate ?



            ----- Original Message ----- 
            From: Michael Collins 
            To: FreeSWITCH Users Help 
            Sent: Friday, December 17, 2010 7:54 PM
            Subject: Re: [Freeswitch-users] members audio conference


            What do you see in the debug logs? Did you compare the logs for a working vs. non-working call? Anything different? 


            -MC


            On Fri, Dec 17, 2010 at 7:35 AM, Madovsky <infos at madovsky.org> wrote:

              when the first member creates and enters in a new conference
              everything is ok. but if a new memeber enters there is no audio
              in the conference, unless the ivr.

              I have a very simple conference dialplan like this

                      <extension name="create_conference">
                              <condition field="${sip_to_uri}" expression="^000(\d{10,15})@$${domain}$">
                                      <action application="set" data="instant_ringback=true"/>
                                      <action application="ring_ready"/>
                                      <action application="sleep" data="3000"/>
                                      <action application="answer"/>
                                      <action application="conference" data="$1-${domain_name}@wideband"/>
                              </condition>
                      </extension>


              I tried to add
              <action application="export" data="absolute_codec_string=speex at 16000k"/>

              but no success

              Thanks

              _______________________________________________
              FreeSWITCH-users mailing list
              FreeSWITCH-users at lists.freeswitch.org
              http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
              UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From covici at ccs.covici.com  Tue Dec 21 04:13:04 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 20 Dec 2010 20:13:04 -0500
Subject: [Freeswitch-users] rebinding meta_app digit does not work on
	b-leg
In-Reply-To: <AANLkTincRa0cMNYc=GuTv7X7vTWHMd6eKxgW0PcaoYdi@mail.gmail.com>
References: <10356.1292765310@ccs.covici.com>
	<AANLkTikA9NoBfwOUEZHx_SNNrNYAukYuq==zU3bcS4L8@mail.gmail.com>
	<25987.1292874077@ccs.covici.com>
	<AANLkTincRa0cMNYc=GuTv7X7vTWHMd6eKxgW0PcaoYdi@mail.gmail.com>
Message-ID: <14664.1292893984@ccs.covici.com>

Thanks much -- I will definitely be on that call!

Michael Collins <msc at freeswitch.org> wrote:

> If someone has a few minutes to write up a nice bind_digit_action how-to,
> including strategies for handling the b-leg, before Wednesday then we would
> appreciate that. If not then let's have this be a subject of discussion on
> Wednesday's conf call. This is definitely something that every FS admin
> should become familiar with, especially if you are doing any kind of DTMF
> input from your callers.
> 
> -MC
> 
> On Mon, Dec 20, 2010 at 11:41 AM, <covici at ccs.covici.com> wrote:
> 
> > I looked up bind_digit_action, but the documentation was very sparse and
> > I could not figure out how to specify which leg, nor did it give the
> > meaning of the parameters.  If someone could explain this more, maybe it
> > would be useful.
> >
> > Thanks.
> >
> >
> > Michael Collins <msc at freeswitch.org> wrote:
> >
> > > John,
> > >
> > > I don't know if this is a bug or not, however I know for a fact that the
> > new
> > > bind_digit_action will do what you want. I highly recommend checking it
> > out.
> > > It is a bit more complicated at first, but once you start using it you
> > will
> > > never go back to bind_meta_app. :)
> > >
> > > -MC
> > >
> > > On Sun, Dec 19, 2010 at 5:28 AM, <covici at ccs.covici.com> wrote:
> > >
> > > > Hi.  I have some dialplan code which allows me to press *2 to record
> > and
> > > > *2 to end the recording.  The way I do this is to rebind the digit by
> > > > executing an extension.  Now if I call, it works fine, but if soneone
> > > > calls me, it does not work -- looking at the logs I found out that even
> > > > though it says it rebound the digit 2 on the b-leg, it actually still
> > > > executes the previous bind of meta-app.
> > > >
> > > > Here is the pastebin to show the log:
> > > > http://pastebin.freeswitch.org/14825
> > > >
> > > > Thanks in advance for any help.
> > > > --
> > > > Your life is like a penny.  You're going to lose it.  The question is:
> > > > How do
> > > > you spend it?
> > > >
> > > >         John Covici
> > > >         covici at ccs.covici.com
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org
> > > >
> > >
> > > ----------------------------------------------------
> > > Alternatives:
> > >
> > > ----------------------------------------------------
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From xduvox at gmail.com  Tue Dec 21 05:05:34 2010
From: xduvox at gmail.com (xduvox)
Date: Mon, 20 Dec 2010 18:05:34 -0800 (PST)
Subject: [Freeswitch-users] gateway without static IP
In-Reply-To: <AANLkTi=KTyAGuqQMRTNcqxkyN1fqxP-z9NXikzeepKPM@mail.gmail.com>
References: <AANLkTi=kDe23KtpwxMd9S66JjrXhga1RK7+BWE5WAqV5@mail.gmail.com>
	<AANLkTi=KTyAGuqQMRTNcqxkyN1fqxP-z9NXikzeepKPM@mail.gmail.com>
Message-ID: <1292897134970-5854523.post@n2.nabble.com>


Hello Steven for reply
I have done the <gateways> configuration and it works perfectly, my problem
is that the public ip of mi gateway changes almost every two weeks, and i
have to replace the ip of the gateway's configuration file.

i have and user or extension in my gateway that registers to FS so
i can know the new ip by sofia status profile internal and see the ip of the
user that is registered to FS

what i am wondering to know is if is there a way to use the ip of the
registered user to configure the gateway and not have to rewrite the ip in
the configuration file of the gateway each time ip changes? 

Another problem is that i can not use www.dyndns.org because the gateway
does not have that ability


thanks in advance!




-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/gateway-without-static-IP-tp5851640p5854523.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Tue Dec 21 05:25:35 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 20 Dec 2010 18:25:35 -0800
Subject: [Freeswitch-users] gateway without static IP
In-Reply-To: <1292897134970-5854523.post@n2.nabble.com>
References: <AANLkTi=kDe23KtpwxMd9S66JjrXhga1RK7+BWE5WAqV5@mail.gmail.com>
	<AANLkTi=KTyAGuqQMRTNcqxkyN1fqxP-z9NXikzeepKPM@mail.gmail.com>
	<1292897134970-5854523.post@n2.nabble.com>
Message-ID: <AANLkTikZG5JWUeaQ1vNvUYgKFYRo23h78D9FZkR0kJ1z@mail.gmail.com>

Well, in a roundabout way you can, by using "sofia_contact" API and regex at
the fs_cli. Here's a quick session I just did:

freeswitch at internal> sofia_contact 1002 at 10.15.0.94
sofia/internal/sip:1002 at 10.15.0.136:61524
;rinstance=f832369f26387993;transport=udp;fs_nat=yes
freeswitch at internal> regex sofia/internal/sip:1002 at 10.15.0.136:61524
;rinstance=f832369f26387993;transport=udp;fs_nat=yes|(\d+\.\d+\.\d+\.\d+)|$1
10.15.0.136
freeswitch at internal>

My user's IP addr is 10.15.0.136. Your challenge, I assume, is knowing when
to update the gateway. If you know that then you can launch the above
commands via a shell script using fs_cli -x. Just be sure to use all the
proper escape sequences for passing all those parens, pipes and backslashes.
:)

-MC



On Mon, Dec 20, 2010 at 6:05 PM, xduvox <xduvox at gmail.com> wrote:

>
> Hello Steven for reply
> I have done the <gateways> configuration and it works perfectly, my problem
> is that the public ip of mi gateway changes almost every two weeks, and i
> have to replace the ip of the gateway's configuration file.
>
> i have and user or extension in my gateway that registers to FS so
> i can know the new ip by sofia status profile internal and see the ip of
> the
> user that is registered to FS
>
> what i am wondering to know is if is there a way to use the ip of the
> registered user to configure the gateway and not have to rewrite the ip in
> the configuration file of the gateway each time ip changes?
>
> Another problem is that i can not use www.dyndns.org because the gateway
> does not have that ability
>
>
> thanks in advance!
>
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/gateway-without-static-IP-tp5851640p5854523.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From thisjoy0528 at gmail.com  Tue Dec 21 05:33:04 2010
From: thisjoy0528 at gmail.com (joy this)
Date: Tue, 21 Dec 2010 10:33:04 +0800
Subject: [Freeswitch-users] questions about VAD and echo cancellation
In-Reply-To: <AANLkTi=xHUEy5zS+arr_Ud_aWL6rw+GkkzDi6SrtVGTV@mail.gmail.com>
References: <AANLkTi=G-gm4V9zV1hLtFVJ4B6-tyQY1Rw=+LEPdnaXa@mail.gmail.com>
	<AANLkTin0Q_NNJXdoFX4d-PYpkBKj2ct84Er5GAWCfoPf@mail.gmail.com>
	<AANLkTi=xHUEy5zS+arr_Ud_aWL6rw+GkkzDi6SrtVGTV@mail.gmail.com>
Message-ID: <AANLkTinjrY1vmh4udy_GBi_fGFp_Cx5qysEyg9eHc8Wd@mail.gmail.com>

NB is the notebook computer..

2010/12/21 Michael Collins <msc at freeswitch.org>

> What is NB?
>
> On Sun, Dec 19, 2010 at 11:00 AM, joy this <thisjoy0528 at gmail.com> wrote:
>
>> Could you give me any suggestions?
>>
>> Sincerely yours,
>> Thisjoy.
>> 2010/12/17 joy this <thisjoy0528 at gmail.com>
>>
>>> Dear all:
>>>
>>>
>>>
>>>         I have questions about VAD and echo cancellation. My FS is
>>> Version 1.0.head (git-) under Windows XP. My soft-phone is X-Lite. I use
>>> earphones and microphones for sip 1 which means the talking and hearing are
>>> separated; on the other hand, I use NB for sip 2 which makes the talking and
>>> hearing in the same place. Sip 1 (1000) calls sip 2 (1001) via FS. When I
>>> say something via sip 1, the echo will occur, and the echo will only occur
>>> on sip 1.
>>>
>>>         How do I cancel the echo? Buy a hardware card or something else?
>>> I have two gateways which are Wellgate2644 and ata171m. Please give me some
>>> suggestions.
>>>
>>>         When I enabled VAD, I found something strange. In the start of
>>> the session, I can hear the noise and echo on sip 1. If I say something via
>>> sip 2, the echo and the noise will disappear. Then the echo will not occur
>>> on sip 1 only in a short time, about several seconds. The echo and the noise
>>> will occur gradually if I say something via sip1. Is it a normal situation?
>>>
>>>
>>>
>>> Sincerely yours,
>>> Thisjoy.
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue Dec 21 05:33:39 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 20 Dec 2010 20:33:39 -0600
Subject: [Freeswitch-users] On what condition P-Asserted-Identity is
 replaced by P-Preferred-Identity?
In-Reply-To: <AANLkTikOvQj_R8zqO0hyZ9WWf38aDBrBfM-SDxehkddL@mail.gmail.com>
References: <AANLkTinZ3Vr_N=pB8QuvDVHUy2rCscz1yPLHfJFVQoQL@mail.gmail.com>
	<AANLkTi=tU2ceXzoOjqPNzJJJVLafmB7EAjP8Gqz3NgF_@mail.gmail.com>
	<AANLkTikOvQj_R8zqO0hyZ9WWf38aDBrBfM-SDxehkddL@mail.gmail.com>
Message-ID: <AANLkTi=zf-_JAJWQrn5VjO2UPrJ9BkOSUBUUBLGiSNxi@mail.gmail.com>

yes it's always safe to make wiki edits, someone always audits the
changes to make sure they are correct.


On Mon, Dec 20, 2010 at 3:19 PM, Lucian Marginean <lucian at e-utile.ro> wrote:
> It is ok to put a note on
> http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type like this:
>
> Send P-Asserted-Identity
>
> {sip_cid_type=pid}sofia/default/user at example.com
>
> Note: you must set privacy flag, otherwise will be inserted
> P-Preferred-Identity instead of P-Asserted-Identity
>
>
>
> Lucian
>
>
> On Mon, Dec 20, 2010 at 12:59 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> use the privacy app to set the proper flags and it should do asserted
>>
>> On Mon, Dec 20, 2010 at 10:18 AM, Lucian Marginean <lucian at e-utile.ro>
>> wrote:
>> > I try to setup P-Asserted-Identity on setup but I get
>> > P-Preferred-Identity
>> > and I don't understand why
>> >
>> > Did I do something wrong or some other flags affect the PID feature?
>> >
>> > <action application="set" data="effective_caller_id_name=647476xxxx"/>
>> > <action application="set" data="effective_caller_id_number=647788xxxx"/>
>> > <action application="set" data="origination_privacy=screen"/>
>> > <action application="bridge"
>> >
>> > data="[sip_cid_type=pid,origination_caller_id_name=647476xxxx,origination_caller_id_number=647788xxxx]sofia/private/905780xxxx at 10.10.67.82:5060"/>
>> >
>> > and invite generated look like this:
>> >
>> > U 10.10.67.83:5060 -> 10.10.67.82:5060
>> > INVITE sip:905780xxxx at 10.10.67.82:5060 SIP/2.0.
>> > Via: SIP/2.0/UDP 10.10.67.83;rport;branch=z9hG4bKH8KNm98v4Qv1N.
>> > Max-Forwards: 68.
>> > From: "647476xxxx" <sip:647788xxxx at 10.10.67.83>;tag=F1QF1Z26cBQZm.
>> > To: <sip:905780xxxx at 10.10.67.82:5060>.
>> > Call-ID: ff01d92d-86ec-122e-cba2-001143e7e50c.
>> > CSeq: 6092608 INVITE.
>> > Contact: <sip:mod_sofia at 10.10.67.83:5060>.
>> > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-79ced28 2010-08-22
>> > 20-58-52
>> > -0400.
>> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> > REGISTER,
>> > REFER, NOTIFY.
>> > Supported: timer, precondition, path, replaces.
>> > Allow-Events: talk, hold, refer.
>> > Privacy: none.
>> > Content-Type: application/sdp.
>> > Content-Disposition: session.
>> > Content-Length: 242.
>> > X-FS-Support: update_display.
>> > P-Preferred-Identity: "647476xxxx" <sip:647788xxxx at 10.10.67.83>
>> >
>> > Lucian
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From eagle.antonio at gmail.com  Tue Dec 21 10:44:11 2010
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Tue, 21 Dec 2010 07:44:11 +0000
Subject: [Freeswitch-users] CallCenter Agent UUID-Standby
In-Reply-To: <AANLkTin88ZknYRanO4sY2S56tPM2w7ZRTFKxfz0V4fE6@mail.gmail.com>
References: <AANLkTikNKi7Rzo8tKdMq+=iG7MaO304Q2CD15e6CJTs9@mail.gmail.com>
	<E9D43200-E2B4-4272-86AD-4D642E3165ED@tlainvestments.com>
	<AANLkTin88ZknYRanO4sY2S56tPM2w7ZRTFKxfz0V4fE6@mail.gmail.com>
Message-ID: <AANLkTi=7aCKRR0m3UWiYdF17kOA+wqNUtdEY3FXxm2Ng@mail.gmail.com>

Hello guys

Sorry to bug you again , but I'm still unable to find out a solution for
this , should i send a bug report or is the bug sited on the chair :)

P.S Have a Merry Christmas to all :)

Ant?nio Teixeira


2010/12/17 Antonio Teixeira <eagle.antonio at gmail.com>

> Hello Troy ,
>
> But that part is present in the dial plan and as soon as a caller hangups
> an hangup is also send to the agent and it doesn't seem to recover from
> that.
>
> Full Debug Log And Dialplan in paste bin
> http://pastebin.com/ADFfLPtR
>
> Thank you all for all your time
> Antonio
>
>
> 2010/12/17 Troy Anderson <freeswitch at tlainvestments.com>
>
> Hi Ant?nio,
>>
>> Take a look at this part of the wiki:
>> http://wiki.freeswitch.org/wiki/Mod_callcenter#Callback
>>
>> Note the last line of the sample dial plan:
>>
>>     <action application="transfer" data="4099"/> <!-- Remove this if you just want to get a single call -->
>>
>> If you want to throw them back in the loop, you have to explicitly do it
>> in your dial plan.
>>
>> -Troy
>>
>> On Dec 16, 2010, at 10:30 AM, Antonio Teixeira wrote:
>>
>> Good Afternoon.
>>
>> I'm leaving Asterisk ( yayyyyyyyyyyyyy) and was curious in trying out the
>> Call Center feature,  uuid-standby , now i created a dialplan and I'm able
>> to connect a caller to an agent.
>> Now by my assumption as soon as the caller disconnects the agent should be
>> trow back ( transfer) into the loop (4099 ext.) but instead i get an hangup
>> and the sip client terminates the call.
>>
>> Any ideas ?
>>
>> Sample Dialplan
>>
>> <extension>
>>   <condition field="destination_number" expression="^(4099)$">
>>     <action application="set" data="res=${callcenter_config(agent set uuid ${caller_id_number}@${domain_name} '${uuid}')}" />
>>
>>
>>
>>
>>     <action application="set" data="res=${callcenter_config(agent set type ${caller_id_number}@${domain_name} 'uuid-standby')}" />
>>     <action application="set" data="res=${callcenter_config(agent set status ${caller_id_number}@${domain_name} 'Available (On Demand)')}" />
>>
>>
>>
>>
>>     <action application="set" data="res=${callcenter_config(agent set state ${caller_id_number}@${domain_name} 'Waiting')}" />
>>     <action application="set" data="cc_warning_tone=tone_stream://%(200,0,500,600,700)"/>
>>
>>
>>
>>
>>     <action application="answer" />
>>     <action application="playback" data="$${hold_music}"/>
>>     <action application="transfer" data="4099"/> <!-- Remove this if you just want to get a single call -->
>>
>>
>>
>>
>>   </condition>
>> </extension>
>>
>>
>> Thanks For Your Time
>> Ant?nio Teixeira
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From steveayre at gmail.com  Tue Dec 21 10:53:48 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 21 Dec 2010 07:53:48 +0000
Subject: [Freeswitch-users] members audio conference
In-Reply-To: <2CF4BD5149DA49AC96278241CEDC6EE4@e1705>
References: <223AD13D879E4B19B15AEEDFC387386A@e1705>
	<AANLkTikydvXoJOk5kkWPYpaDqCGYTJgXbzj2=ffTB07p@mail.gmail.com>
	<532EC039F378476A9515E88F30BBB432@e1705>
	<AANLkTi=_DDyiOtBUDkJ9ET8FATyu3qntDK3PFA1xXi=7@mail.gmail.com>
	<D43C6389-2A38-4DF7-9428-7D84D5D5EC9D@gmail.com>
	<6AC2F3E2B53F494FB9A47BC9C5E314C3@e1705>
	<AANLkTimuVtkLkOVYUk-5qqzj9bQZ-u0dj9BgaQj61SaT@mail.gmail.com>
	<2CF4BD5149DA49AC96278241CEDC6EE4@e1705>
Message-ID: <AANLkTik6WcT8UhgCMKUjd34FJNv=C7rK_RwRYxocHeqU@mail.gmail.com>

Conferencing is one of the cases where every member will be doing
transcoding, because even if they all use the same codec it still
needs to uncompress every member's media stream so that it can mix and
recompress them.

If you're worried about CPU usage, the try to get all members to be
using one of the codecs that's a little nicer on the CPU.

-Steve


On 21 December 2010 00:48, Madovsky <infos at madovsky.org> wrote:
> I did a lot of silly things since I learn FS?? :D
> but not to try to transcode on FS if I use a proxy media....
> I would just to minimize the transcoding....
>
> ----- Original Message -----
> From: Michael Collins
> To: FreeSWITCH Users Help
> Sent: Monday, December 20, 2010 4:29 PM
> Subject: Re: [Freeswitch-users] members audio conference
>
>
> On Mon, Dec 20, 2010 at 12:02 PM, Madovsky <infos at madovsky.org> wrote:
>>
>> > FS core converts every member from their codec to L16
>> ok I understand, so is it possible to change from L16 to another codec?in
>> a conference dialplan ?
>
> I don't think you understood what Steven was saying. You don't need to worry
> about the transcoding. As long as there is a codec that FS supports then the
> transcoding happens without you having to do anything. Unless you've done
> something silly like proxy media then you shouldn't need to worry about
> codecs.
> -MC
>>
>> ----- Original Message -----
>> From: Steven Ayre
>> To: FreeSWITCH Users Help
>> Sent: Monday, December 20, 2010 2:48 PM
>> Subject: Re: [Freeswitch-users] members audio conference
>> More specifically FS core converts every member from their codec to L16
>> which is given to the conference. The conference combines all speaking
>> channels and the resulting L16 is given back to the core to send to the
>> members, the core converting to the correct codec for each member.
>>
>> Steve on iPhone
>> On 20 Dec 2010, at 17:59, Michael Collins <msc at freeswitch.org> wrote:
>>
>> The conference itself doesn't do any codec stuff - FreeSWITCH core does.
>> All I can say is compare the working versus non-working logs and look for
>> clues.
>> -MC
>>
>> On Fri, Dec 17, 2010 at 6:49 PM, Madovsky <infos at madovsky.org> wrote:
>>>
>>> nohgint strange on logs.
>>> but I guess it's a codec and rate problem.
>>> tried with different SIP phones and it works.
>>> how a conference manage the codecs? ?
>>> I know the rate can be set in profile,
>>> but how conference codec is managed
>>> ?if all members have different codec and rate ?
>>>
>>>
>>>
>>>
>>> ----- Original Message -----
>>> From: Michael Collins
>>> To: FreeSWITCH Users Help
>>> Sent: Friday, December 17, 2010 7:54 PM
>>> Subject: Re: [Freeswitch-users] members audio conference
>>> What do you see in the debug logs? Did you compare the logs for a working
>>> vs. non-working call? Anything different?
>>> -MC
>>>
>>> On Fri, Dec 17, 2010 at 7:35 AM, Madovsky <infos at madovsky.org> wrote:
>>>>
>>>> when the first member creates and enters in a new conference
>>>> everything is ok. but if a new memeber enters there is no audio
>>>> in the conference, unless the ivr.
>>>>
>>>> I have a very simple conference dialplan like this
>>>>
>>>> ??????? <extension name="create_conference">
>>>> ??????????????? <condition field="${sip_to_uri}"
>>>> expression="^000(\d{10,15})@$${domain}$">
>>>> ??????????????????????? <action application="set"
>>>> data="instant_ringback=true"/>
>>>> ??????????????????????? <action application="ring_ready"/>
>>>> ??????????????????????? <action application="sleep" data="3000"/>
>>>> ??????????????????????? <action application="answer"/>
>>>> ??????????????????????? <action application="conference"
>>>> data="$1-${domain_name}@wideband"/>
>>>> ??????????????? </condition>
>>>> ??????? </extension>
>>>>
>>>> I tried to add
>>>> <action application="export" data="absolute_codec_string=speex at 16000k"/>
>>>>
>>>> but no success
>>>>
>>>> Thanks
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> ________________________________
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> ________________________________
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> ________________________________
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Tue Dec 21 11:01:41 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 21 Dec 2010 03:01:41 -0500
Subject: [Freeswitch-users] members audio conference
References: <223AD13D879E4B19B15AEEDFC387386A@e1705><AANLkTikydvXoJOk5kkWPYpaDqCGYTJgXbzj2=ffTB07p@mail.gmail.com><532EC039F378476A9515E88F30BBB432@e1705><AANLkTi=_DDyiOtBUDkJ9ET8FATyu3qntDK3PFA1xXi=7@mail.gmail.com><D43C6389-2A38-4DF7-9428-7D84D5D5EC9D@gmail.com><6AC2F3E2B53F494FB9A47BC9C5E314C3@e1705><AANLkTimuVtkLkOVYUk-5qqzj9bQZ-u0dj9BgaQj61SaT@mail.gmail.com><2CF4BD5149DA49AC96278241CEDC6EE4@e1705>
	<AANLkTik6WcT8UhgCMKUjd34FJNv=C7rK_RwRYxocHeqU@mail.gmail.com>
Message-ID: <9319261E4F6C44ACBF197CF94B65C1BA@e1705>

Thanks to make me understand the audio concept of conference
I was almost sure...


----- Original Message ----- 
From: "Steven Ayre" <steveayre at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, December 21, 2010 2:53 AM
Subject: Re: [Freeswitch-users] members audio conference


Conferencing is one of the cases where every member will be doing
transcoding, because even if they all use the same codec it still
needs to uncompress every member's media stream so that it can mix and
recompress them.

If you're worried about CPU usage, the try to get all members to be
using one of the codecs that's a little nicer on the CPU.

-Steve


On 21 December 2010 00:48, Madovsky <infos at madovsky.org> wrote:
> I did a lot of silly things since I learn FS :D
> but not to try to transcode on FS if I use a proxy media....
> I would just to minimize the transcoding....
>
> ----- Original Message -----
> From: Michael Collins
> To: FreeSWITCH Users Help
> Sent: Monday, December 20, 2010 4:29 PM
> Subject: Re: [Freeswitch-users] members audio conference
>
>
> On Mon, Dec 20, 2010 at 12:02 PM, Madovsky <infos at madovsky.org> wrote:
>>
>> > FS core converts every member from their codec to L16
>> ok I understand, so is it possible to change from L16 to another codec in
>> a conference dialplan ?
>
> I don't think you understood what Steven was saying. You don't need to 
> worry
> about the transcoding. As long as there is a codec that FS supports then 
> the
> transcoding happens without you having to do anything. Unless you've done
> something silly like proxy media then you shouldn't need to worry about
> codecs.
> -MC
>>
>> ----- Original Message -----
>> From: Steven Ayre
>> To: FreeSWITCH Users Help
>> Sent: Monday, December 20, 2010 2:48 PM
>> Subject: Re: [Freeswitch-users] members audio conference
>> More specifically FS core converts every member from their codec to L16
>> which is given to the conference. The conference combines all speaking
>> channels and the resulting L16 is given back to the core to send to the
>> members, the core converting to the correct codec for each member.
>>
>> Steve on iPhone
>> On 20 Dec 2010, at 17:59, Michael Collins <msc at freeswitch.org> wrote:
>>
>> The conference itself doesn't do any codec stuff - FreeSWITCH core does.
>> All I can say is compare the working versus non-working logs and look for
>> clues.
>> -MC
>>
>> On Fri, Dec 17, 2010 at 6:49 PM, Madovsky <infos at madovsky.org> wrote:
>>>
>>> nohgint strange on logs.
>>> but I guess it's a codec and rate problem.
>>> tried with different SIP phones and it works.
>>> how a conference manage the codecs ?
>>> I know the rate can be set in profile,
>>> but how conference codec is managed
>>> if all members have different codec and rate ?
>>>
>>>
>>>
>>>
>>> ----- Original Message -----
>>> From: Michael Collins
>>> To: FreeSWITCH Users Help
>>> Sent: Friday, December 17, 2010 7:54 PM
>>> Subject: Re: [Freeswitch-users] members audio conference
>>> What do you see in the debug logs? Did you compare the logs for a 
>>> working
>>> vs. non-working call? Anything different?
>>> -MC
>>>
>>> On Fri, Dec 17, 2010 at 7:35 AM, Madovsky <infos at madovsky.org> wrote:
>>>>
>>>> when the first member creates and enters in a new conference
>>>> everything is ok. but if a new memeber enters there is no audio
>>>> in the conference, unless the ivr.
>>>>
>>>> I have a very simple conference dialplan like this
>>>>
>>>> <extension name="create_conference">
>>>> <condition field="${sip_to_uri}"
>>>> expression="^000(\d{10,15})@$${domain}$">
>>>> <action application="set"
>>>> data="instant_ringback=true"/>
>>>> <action application="ring_ready"/>
>>>> <action application="sleep" data="3000"/>
>>>> <action application="answer"/>
>>>> <action application="conference"
>>>> data="$1-${domain_name}@wideband"/>
>>>> </condition>
>>>> </extension>
>>>>
>>>> I tried to add
>>>> <action application="export" 
>>>> data="absolute_codec_string=speex at 16000k"/>
>>>>
>>>> but no success
>>>>
>>>> Thanks
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> ________________________________
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> ________________________________
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> ________________________________
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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



From sameer2k3t at gmail.com  Tue Dec 21 11:18:02 2010
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Tue, 21 Dec 2010 13:18:02 +0500
Subject: [Freeswitch-users] forcing codec on leg b
Message-ID: <AANLkTi=V4e5jeZEB4VsafKtNU3BbV7p1qGQDPQPOkbbc@mail.gmail.com>

Hello guyz,
Need your help

I want my fs to behave like this

when call arrives for a number 1234656789 at fs i call a php file in which i
check in the database that what should be the forwarding address of this
number. for example the forwarding address is 123456789 at 1.1.1.1 then i
return that forwarding address in bridge application in the end
i want to define some setting in sip profiles, user directory or wherever so
when ever ip is 1.1.1.1 it set the codec on leg b to pcma
like it is defined in asterisk as
[1.1.1.1]
host=1.1.1.1
type=peer
disallow=all
allow=alaw


i want my sip profile to check if network address is 1.1.1.1, it uses a
particular codec lets say pcma but if network address is 2.2.2.2 it uses the
defult
and this i want to define for around 40 ip addresses
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From david.ponzone at ipeva.fr  Tue Dec 21 11:22:39 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 21 Dec 2010 09:22:39 +0100
Subject: [Freeswitch-users] forcing codec on leg b
In-Reply-To: <AANLkTi=V4e5jeZEB4VsafKtNU3BbV7p1qGQDPQPOkbbc@mail.gmail.com>
References: <AANLkTi=V4e5jeZEB4VsafKtNU3BbV7p1qGQDPQPOkbbc@mail.gmail.com>
Message-ID: <55EEC03D-A5F5-4669-9452-888A6BAEBAD9@ipeva.fr>

in the dialplan, use a condition to match network_addr and then set absolute_codec_string to PCMA.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 21/12/2010 ? 09:18, Sameer Khan a ?crit :

> Hello guyz, 
> Need your help
> 
> I want my fs to behave like this
> 
> when call arrives for a number 1234656789 at fs i call a php file in which i check in the database that what should be the forwarding address of this number. for example the forwarding address is 123456789 at 1.1.1.1 then i return that forwarding address in bridge application in the end
> i want to define some setting in sip profiles, user directory or wherever so when ever ip is 1.1.1.1 it set the codec on leg b to pcma
> like it is defined in asterisk as 
> [1.1.1.1] 
> host=1.1.1.1 
> type=peer
> disallow=all 
> allow=alaw
> 
> 
> i want my sip profile to check if network address is 1.1.1.1, it uses a particular codec lets say pcma but if network address is 2.2.2.2 it uses the defult 
> and this i want to define for around 40 ip addresses
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sameer2k3t at gmail.com  Tue Dec 21 11:32:35 2010
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Tue, 21 Dec 2010 13:32:35 +0500
Subject: [Freeswitch-users] forcing codec on leg b
In-Reply-To: <55EEC03D-A5F5-4669-9452-888A6BAEBAD9@ipeva.fr>
References: <AANLkTi=V4e5jeZEB4VsafKtNU3BbV7p1qGQDPQPOkbbc@mail.gmail.com>
	<55EEC03D-A5F5-4669-9452-888A6BAEBAD9@ipeva.fr>
Message-ID: <AANLkTinuDSy=iR+ccWAcz1im7jbYM+qzq77kSCYub0Y4@mail.gmail.com>

thanks for your reply david,

but for that i will need to call absolute_Codec_string value from datbase
which i dont want to do now.

as i have one data base and running two asterisk on them with this fs

cant i define in sip users or profile ,

i saw an option there in brian.xml for cidr but all examples are username
dependent

i dont want user authentication as i am just forwarding the calls to my user

i just want to check if ip is 1.1.1.1 it uses pcma for all numbers/extension
going to the ip like this XXX at 1.1.1.1


On Tue, Dec 21, 2010 at 1:22 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> in the dialplan, use a condition to match network_addr and then set
> absolute_codec_string to PCMA.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 21/12/2010 ? 09:18, Sameer Khan a ?crit :
>
> Hello guyz,
> Need your help
>
> I want my fs to behave like this
>
> when call arrives for a number 1234656789 at fs i call a php file in which
> i check in the database that what should be the forwarding address of this
> number. for example the forwarding address is 123456789 at 1.1.1.1 then i
> return that forwarding address in bridge application in the end
> i want to define some setting in sip profiles, user directory or wherever
> so when ever ip is 1.1.1.1 it set the codec on leg b to pcma
> like it is defined in asterisk as
> [1.1.1.1]
> host=1.1.1.1
> type=peer
> disallow=all
> allow=alaw
>
>
> i want my sip profile to check if network address is 1.1.1.1, it uses a
> particular codec lets say pcma but if network address is 2.2.2.2 it uses the
> defult
> and this i want to define for around 40 ip addresses
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Tue Dec 21 11:40:37 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 21 Dec 2010 09:40:37 +0100
Subject: [Freeswitch-users] forcing codec on leg b
In-Reply-To: <AANLkTinuDSy=iR+ccWAcz1im7jbYM+qzq77kSCYub0Y4@mail.gmail.com>
References: <AANLkTi=V4e5jeZEB4VsafKtNU3BbV7p1qGQDPQPOkbbc@mail.gmail.com>
	<55EEC03D-A5F5-4669-9452-888A6BAEBAD9@ipeva.fr>
	<AANLkTinuDSy=iR+ccWAcz1im7jbYM+qzq77kSCYub0Y4@mail.gmail.com>
Message-ID: <5B3F4C11-7AED-4F34-BE27-E4A2EFEE5EC4@ipeva.fr>

sorry, I dont understand what you want to achieve

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 21/12/2010 ? 09:32, Sameer Khan a ?crit :

> thanks for your reply david, 
> 
> but for that i will need to call absolute_Codec_string value from datbase which i dont want to do now.
> 
> as i have one data base and running two asterisk on them with this fs
> 
> cant i define in sip users or profile , 
> 
> i saw an option there in brian.xml for cidr but all examples are username dependent 
> 
> i dont want user authentication as i am just forwarding the calls to my user
> 
> i just want to check if ip is 1.1.1.1 it uses pcma for all numbers/extension going to the ip like this XXX at 1.1.1.1
> 
> 
> On Tue, Dec 21, 2010 at 1:22 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> in the dialplan, use a condition to match network_addr and then set absolute_codec_string to PCMA.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 21/12/2010 ? 09:18, Sameer Khan a ?crit :
> 
>> Hello guyz, 
>> Need your help
>> 
>> I want my fs to behave like this
>> 
>> when call arrives for a number 1234656789 at fs i call a php file in which i check in the database that what should be the forwarding address of this number. for example the forwarding address is 123456789 at 1.1.1.1 then i return that forwarding address in bridge application in the end
>> i want to define some setting in sip profiles, user directory or wherever so when ever ip is 1.1.1.1 it set the codec on leg b to pcma
>> like it is defined in asterisk as 
>> [1.1.1.1] 
>> host=1.1.1.1 
>> type=peer
>> disallow=all 
>> allow=alaw
>> 
>> 
>> i want my sip profile to check if network address is 1.1.1.1, it uses a particular codec lets say pcma but if network address is 2.2.2.2 it uses the defult 
>> and this i want to define for around 40 ip addresses
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sameer2k3t at gmail.com  Tue Dec 21 11:58:49 2010
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Tue, 21 Dec 2010 13:58:49 +0500
Subject: [Freeswitch-users] forcing codec on leg b
In-Reply-To: <5B3F4C11-7AED-4F34-BE27-E4A2EFEE5EC4@ipeva.fr>
References: <AANLkTi=V4e5jeZEB4VsafKtNU3BbV7p1qGQDPQPOkbbc@mail.gmail.com>
	<55EEC03D-A5F5-4669-9452-888A6BAEBAD9@ipeva.fr>
	<AANLkTinuDSy=iR+ccWAcz1im7jbYM+qzq77kSCYub0Y4@mail.gmail.com>
	<5B3F4C11-7AED-4F34-BE27-E4A2EFEE5EC4@ipeva.fr>
Message-ID: <AANLkTinsojjgQHsTUSYVx7O_oM=-YSaEtUmPuZrDS2nx@mail.gmail.com>

in asterisk i m doing it like

i call perl file via agi and in that perl file i check in  the database what
should be the forwarding address for the incoming number
lets say call arrives at my box for 12121212 then a perl file is called
which checks in the database for the destination address and lets say the
destination address is xyz at 1.1.1.1

this all is achieved by xml curl

now in asterisk i add these lines in sip.conf
[1.1.1.1]
host=1.1.1.1
type=peer
disallow=all
allow=alaw

so when asterisk find that ip in sip.conf it force the codec on leg b to
alaw.


this i want to achieve in fs


On Tue, Dec 21, 2010 at 1:40 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> sorry, I dont understand what you want to achieve
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 21/12/2010 ? 09:32, Sameer Khan a ?crit :
>
> thanks for your reply david,
>
> but for that i will need to call absolute_Codec_string value from datbase
> which i dont want to do now.
>
> as i have one data base and running two asterisk on them with this fs
>
> cant i define in sip users or profile ,
>
> i saw an option there in brian.xml for cidr but all examples are username
> dependent
>
> i dont want user authentication as i am just forwarding the calls to my
> user
>
> i just want to check if ip is 1.1.1.1 it uses pcma for all
> numbers/extension going to the ip like this XXX at 1.1.1.1
>
>
> On Tue, Dec 21, 2010 at 1:22 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> in the dialplan, use a condition to match network_addr and then set
>> absolute_codec_string to PCMA.
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 21/12/2010 ? 09:18, Sameer Khan a ?crit :
>>
>> Hello guyz,
>> Need your help
>>
>> I want my fs to behave like this
>>
>> when call arrives for a number 1234656789 at fs i call a php file in which
>> i check in the database that what should be the forwarding address of this
>> number. for example the forwarding address is 123456789 at 1.1.1.1 then i
>> return that forwarding address in bridge application in the end
>> i want to define some setting in sip profiles, user directory or wherever
>> so when ever ip is 1.1.1.1 it set the codec on leg b to pcma
>> like it is defined in asterisk as
>> [1.1.1.1]
>> host=1.1.1.1
>> type=peer
>> disallow=all
>> allow=alaw
>>
>>
>> i want my sip profile to check if network address is 1.1.1.1, it uses a
>> particular codec lets say pcma but if network address is 2.2.2.2 it uses the
>> defult
>> and this i want to define for around 40 ip addresses
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Avi at aMarcus.com  Tue Dec 21 12:03:13 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Tue, 21 Dec 2010 11:03:13 +0200
Subject: [Freeswitch-users] VPS Host suggestions?
In-Reply-To: <AANLkTinawP5qWB4447QveMqAq0fvvtOtsen74FYdF30-@mail.gmail.com>
References: <AANLkTinawP5qWB4447QveMqAq0fvvtOtsen74FYdF30-@mail.gmail.com>
Message-ID: <AANLkTi=haM9NkLrQgsyUvnf431L2bYpA+w=z6DDVowNz@mail.gmail.com>

I'm running less than 5 concurrent calls for the next month at least
and slowly gaining volume, so I don't need anything fancy.
The wiki seems to say that Xen virtualization works with FS.

Any recommendations for VPS hosting? I know you get what you pay for,
but I'm sure there's inexpensive that's not bad.

Linode has a very slick website and feature list, 64bit hosting, and
512mb/200gb bandwidth for $20/month.
Any VPS for voip suggestions?
Thanks!

-Avi Marcus


From hwnorman at hotmail.com  Tue Dec 21 11:42:47 2010
From: hwnorman at hotmail.com (Norman Lam)
Date: Tue, 21 Dec 2010 16:42:47 +0800
Subject: [Freeswitch-users] compile error on libmp3lame and mod_shout
Message-ID: <SNT109-DS118A0D1D7049B8F876EE9C11A0@phx.gbl>

Hi there 

 

I am using AMD athlon X2, windows xp sp3, VC++2008 express (only installed
this software) using the Git source downloaded today the first time 

 

I am getting these 2 error ,  is there any ways to resolve this or  work
around.

 

Please advise 

 

Norman Lam 

 

 

1st error 

:

:

47>------ Build started: Project: libmp3lame, Configuration: Debug Win32
------

47>Compiling...

44>s2_srvr.c

47>bitstream.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\bitstream.c': No such file or directory

47>version.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\version.c': No such file or directory

47>VbrTag.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\VbrTag.c': No such file or directory

47>vbrquantize.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\vbrquantize.c': No such file or directory

47>util.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\util.c': No such file or directory

47>takehiro.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\takehiro.c': No such file or directory

47>set_get.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\set_get.c': No such file or directory

47>reservoir.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\reservoir.c': No such file or directory

47>quantize_pvt.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\quantize_pvt.c': No such file or directory

47>quantize.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\quantize.c': No such file or directory

47>psymodel.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\psymodel.c': No such file or directory

47>presets.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\presets.c': No such file or directory

47>newmdct.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\newmdct.c': No such file or directory

47>mpglib_interface.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\mpglib_interface.c': No such file or directory

47>lame.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\lame.c': No such file or directory

47>id3tag.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\id3tag.c': No such file or directory

47>gain_analysis.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\gain_analysis.c': No such file or directory

47>fft.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\fft.c': No such file or directory

47>encoder.c

47>c1 : fatal error C1083: Cannot open source file:
'..\..\lame-3.97\libmp3lame\encoder.c': No such file or directory

47>Generating Code...

47>Build log was saved at
"file://c:\FS_GIT\libs\win32\libmp3lame\Debug\BuildLog.htm"

47>libmp3lame - 19 error(s), 0 warning(s)

44>s2_pkt.c

:

:

2nd error 

:

:

65>.\mod_shout.c(38) : fatal error C1083: Cannot open include file:
'lame.h': No such file or directory

65>Build log was saved at
"file://c:\FS_GIT\src\mod\formats\mod_shout\Win32\Debug\BuildLog.htm"

65>mod_shout - 1 error(s), 0 warning(s)

:

:

:

140>Linking...

140>Embedding manifest...

140>Build log was saved at "file://c:\FS_GIT\w32\Console\Debug\BuildLog
FreeSwitchConsole.htm"

140>FreeSwitchConsole - 0 error(s), 1 warning(s)

========== Build: 124 succeeded, 2 failed, 0 up-to-date, 14 skipped
==========

 

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From nagalenoj at gmail.com  Tue Dec 21 12:30:27 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Tue, 21 Dec 2010 15:00:27 +0530
Subject: [Freeswitch-users] No media when bind_meta_app
In-Reply-To: <AANLkTimn03z=+Apzg-o5m1euiEeBdd6pNS-huVs6BA3A@mail.gmail.com>
References: <AANLkTimn03z=+Apzg-o5m1euiEeBdd6pNS-huVs6BA3A@mail.gmail.com>
Message-ID: <AANLkTim8YNWyK_8oJ_QPtX_9=eqQBi7EAwQ+FmwS1dRU@mail.gmail.com>

Does anyone else face this issue?!

On Wed, Dec 8, 2010 at 5:15 PM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Dear friends,
>      I've tried using bind_meta_app for recording the call in my ESL
> script. It works fine as intended, But I face a serious media breakage(a and
> b leg doesn't hear the opposite leg's voice) for about 5 seconds once the
> person enters the DTMF to record.
>
> ESL script:
> http://pastebin.freeswitch.org/14734
>
> Freeswitch log:
> http://pastebin.freeswitch.org/14735
>
> I tried applications other than record_session and I face the same with all
> applications. Have I done something?!
> Help me to solve this.
>
> --
> Thanks.
>



-- 
Regards,
Nagalenoj H.
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From u2nsam at gmail.com  Tue Dec 21 12:36:18 2010
From: u2nsam at gmail.com (Sam)
Date: Tue, 21 Dec 2010 15:06:18 +0530
Subject: [Freeswitch-users] loops
Message-ID: <AANLkTim0iPTrZL64MGJzrW0XT2z6+XFXzfYKW_MeW-FA@mail.gmail.com>

Hello,

I am trying to get a loop done using if ... then .. return function,
using below syntax but getting error on it.


if (session:ready()) then return ack()
end


function ack()

session:answer();
session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/zrtp/8000/zrtp-somethings_wrong.wav")

end



error:---
EXECUTE sofia/internal/7001 at 192.168.2.190 lua(hello.lua)
2010-12-21 14:55:34.724790 [ERR] mod_lua.cpp:182
/usr/local/freeswitch/scripts/hello.lua:1: attempt to call global 'ack' (a
nil value)
stack traceback:
        /usr/local/freeswitch/scripts/hello.lua:1: in main chunk


Any ideas to resolve this.

Regards
Sam
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From u2nsam at gmail.com  Tue Dec 21 12:54:32 2010
From: u2nsam at gmail.com (Sam)
Date: Tue, 21 Dec 2010 15:24:32 +0530
Subject: [Freeswitch-users] forcing codec on leg b
In-Reply-To: <AANLkTinsojjgQHsTUSYVx7O_oM=-YSaEtUmPuZrDS2nx@mail.gmail.com>
References: <AANLkTi=V4e5jeZEB4VsafKtNU3BbV7p1qGQDPQPOkbbc@mail.gmail.com>
	<55EEC03D-A5F5-4669-9452-888A6BAEBAD9@ipeva.fr>
	<AANLkTinuDSy=iR+ccWAcz1im7jbYM+qzq77kSCYub0Y4@mail.gmail.com>
	<5B3F4C11-7AED-4F34-BE27-E4A2EFEE5EC4@ipeva.fr>
	<AANLkTinsojjgQHsTUSYVx7O_oM=-YSaEtUmPuZrDS2nx@mail.gmail.com>
Message-ID: <AANLkTimrUPoucPZsaSBvgq6s2_-vnyKHYCE+qWVE=x_E@mail.gmail.com>

Hi,

You can use this,

absolute_codec_string=PCMA

http://wiki.freeswitch.org/wiki/Codec_negotiation#Early_Negotiation_parameters

Regds
Sam



On Tue, Dec 21, 2010 at 2:28 PM, Sameer Khan <sameer2k3t at gmail.com> wrote:

> in asterisk i m doing it like
>
> i call perl file via agi and in that perl file i check in  the database
> what should be the forwarding address for the incoming number
> lets say call arrives at my box for 12121212 then a perl file is called
> which checks in the database for the destination address and lets say the
> destination address is xyz at 1.1.1.1
>
> this all is achieved by xml curl
>
> now in asterisk i add these lines in sip.conf
> [1.1.1.1]
> host=1.1.1.1
> type=peer
> disallow=all
> allow=alaw
>
> so when asterisk find that ip in sip.conf it force the codec on leg b to
> alaw.
>
>
> this i want to achieve in fs
>
>
> On Tue, Dec 21, 2010 at 1:40 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> sorry, I dont understand what you want to achieve
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 21/12/2010 ? 09:32, Sameer Khan a ?crit :
>>
>> thanks for your reply david,
>>
>> but for that i will need to call absolute_Codec_string value from datbase
>> which i dont want to do now.
>>
>> as i have one data base and running two asterisk on them with this fs
>>
>> cant i define in sip users or profile ,
>>
>> i saw an option there in brian.xml for cidr but all examples are username
>> dependent
>>
>> i dont want user authentication as i am just forwarding the calls to my
>> user
>>
>> i just want to check if ip is 1.1.1.1 it uses pcma for all
>> numbers/extension going to the ip like this XXX at 1.1.1.1
>>
>>
>> On Tue, Dec 21, 2010 at 1:22 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>
>>> in the dialplan, use a condition to match network_addr and then set
>>> absolute_codec_string to PCMA.
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 21/12/2010 ? 09:18, Sameer Khan a ?crit :
>>>
>>> Hello guyz,
>>> Need your help
>>>
>>> I want my fs to behave like this
>>>
>>> when call arrives for a number 1234656789 at fs i call a php file in
>>> which i check in the database that what should be the forwarding address of
>>> this number. for example the forwarding address is 123456789 at 1.1.1.1then i return that forwarding address in bridge application in the end
>>> i want to define some setting in sip profiles, user directory or wherever
>>> so when ever ip is 1.1.1.1 it set the codec on leg b to pcma
>>> like it is defined in asterisk as
>>> [1.1.1.1]
>>> host=1.1.1.1
>>> type=peer
>>> disallow=all
>>> allow=alaw
>>>
>>>
>>> i want my sip profile to check if network address is 1.1.1.1, it uses a
>>> particular codec lets say pcma but if network address is 2.2.2.2 it uses the
>>> defult
>>> and this i want to define for around 40 ip addresses
>>>  _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From marty at maui-systems.co.uk  Tue Dec 21 13:50:41 2010
From: marty at maui-systems.co.uk (Marty Lee)
Date: Tue, 21 Dec 2010 10:50:41 +0000
Subject: [Freeswitch-users] Group calls & voicemail
Message-ID: <752EEB96-74F9-40D1-AEEB-296B3B7F016C@maui-systems.co.uk>


Trying to get my head around the dialplan in FreeSwitch and can't
find a good example of how to do call a group of extensions, but
if there is no answer, leave it in a common voicemail.

Think of an incoming line to a small business; it rings all the
extensions and if nobody answers, the message is left in an 'office'
voicemail, rather than an individual.

I've got an entry in the public dialplan that maps the incoming
number to the extension 'office'; what I currently have for the
'office' extension is below and is a blatant frankenstein job on
the Local-extensions entry, but doesn't work.

I'll keep looking, but if anyone has any ideas, then feel free to
point me in the right direction.

m



<extension name="office">
    <condition field="destination_number" expression="^office$">
      <action application="set" data="dialed_extension=$1"/>
      <action application="export" data="dialed_extension=$1"/>
      <action application="set" data="ringback=${uk-ring}"/>
      <action application="set" data="call_timeout=10"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="set" data="ignore_early_media=true"/>
      <action application="set" data="continue_on_fail=true"/>
      <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>

      <action application="bridge" data="${group_call(maui_group@${domain_name}+F)}"/>
      <action application="answer"/>
      <action application="sleep" data="10000"/>
      <action application="bridge" data="loopback/app=voicemail:default ${domain_name} 1000"/>
    </condition>
  </extension>

Notes:

'maui_group' is defined as two test extensions; '1000' is the destination
voicemail account I want it to go to if there is no answer or the
handsets are offline.




-----
Marty Lee                         e: marty at maui-systems.co.uk
Technical Director                v: +44 845 869 2661
Maui Systems Ltd                  f: +44 871 433 8922
Scotland, UK                      w: http://www.maui-systems.co.uk



From sameer2k3t at gmail.com  Tue Dec 21 13:54:35 2010
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Tue, 21 Dec 2010 15:54:35 +0500
Subject: [Freeswitch-users] forcing codec on leg b
In-Reply-To: <AANLkTimrUPoucPZsaSBvgq6s2_-vnyKHYCE+qWVE=x_E@mail.gmail.com>
References: <AANLkTi=V4e5jeZEB4VsafKtNU3BbV7p1qGQDPQPOkbbc@mail.gmail.com>
	<55EEC03D-A5F5-4669-9452-888A6BAEBAD9@ipeva.fr>
	<AANLkTinuDSy=iR+ccWAcz1im7jbYM+qzq77kSCYub0Y4@mail.gmail.com>
	<5B3F4C11-7AED-4F34-BE27-E4A2EFEE5EC4@ipeva.fr>
	<AANLkTinsojjgQHsTUSYVx7O_oM=-YSaEtUmPuZrDS2nx@mail.gmail.com>
	<AANLkTimrUPoucPZsaSBvgq6s2_-vnyKHYCE+qWVE=x_E@mail.gmail.com>
Message-ID: <AANLkTi=qBM6qdQgktyP01+y9=JZxoWe1ELWxLU55oPqw@mail.gmail.com>

i know that but i want such things from sip profiles,

i dont want to pass this value from dialplan as my dialplan is binded in xml
curl

can i set this in sip settings? different codecs for different ips

On Tue, Dec 21, 2010 at 2:54 PM, Sam <u2nsam at gmail.com> wrote:

> Hi,
>
> You can use this,
>
> absolute_codec_string=PCMA
>
> http://wiki.freeswitch.org/wiki/Codec_negotiation#Early_Negotiation_parameters
>
> Regds
> Sam
>
>
>
>
> On Tue, Dec 21, 2010 at 2:28 PM, Sameer Khan <sameer2k3t at gmail.com> wrote:
>
>> in asterisk i m doing it like
>>
>> i call perl file via agi and in that perl file i check in  the database
>> what should be the forwarding address for the incoming number
>> lets say call arrives at my box for 12121212 then a perl file is called
>> which checks in the database for the destination address and lets say the
>> destination address is xyz at 1.1.1.1
>>
>> this all is achieved by xml curl
>>
>> now in asterisk i add these lines in sip.conf
>> [1.1.1.1]
>> host=1.1.1.1
>> type=peer
>> disallow=all
>> allow=alaw
>>
>> so when asterisk find that ip in sip.conf it force the codec on leg b to
>> alaw.
>>
>>
>> this i want to achieve in fs
>>
>>
>> On Tue, Dec 21, 2010 at 1:40 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>
>>> sorry, I dont understand what you want to achieve
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 21/12/2010 ? 09:32, Sameer Khan a ?crit :
>>>
>>> thanks for your reply david,
>>>
>>> but for that i will need to call absolute_Codec_string value from datbase
>>> which i dont want to do now.
>>>
>>> as i have one data base and running two asterisk on them with this fs
>>>
>>> cant i define in sip users or profile ,
>>>
>>> i saw an option there in brian.xml for cidr but all examples are username
>>> dependent
>>>
>>> i dont want user authentication as i am just forwarding the calls to my
>>> user
>>>
>>> i just want to check if ip is 1.1.1.1 it uses pcma for all
>>> numbers/extension going to the ip like this XXX at 1.1.1.1
>>>
>>>
>>> On Tue, Dec 21, 2010 at 1:22 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>>
>>>> in the dialplan, use a condition to match network_addr and then set
>>>> absolute_codec_string to PCMA.
>>>>
>>>>  David Ponzone  Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>
>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>>> l'exp?diteur.*
>>>> *
>>>> *
>>>>
>>>>
>>>>
>>>> Le 21/12/2010 ? 09:18, Sameer Khan a ?crit :
>>>>
>>>> Hello guyz,
>>>> Need your help
>>>>
>>>> I want my fs to behave like this
>>>>
>>>> when call arrives for a number 1234656789 at fs i call a php file in
>>>> which i check in the database that what should be the forwarding address of
>>>> this number. for example the forwarding address is 123456789 at 1.1.1.1then i return that forwarding address in bridge application in the end
>>>> i want to define some setting in sip profiles, user directory or
>>>> wherever so when ever ip is 1.1.1.1 it set the codec on leg b to pcma
>>>> like it is defined in asterisk as
>>>> [1.1.1.1]
>>>> host=1.1.1.1
>>>> type=peer
>>>> disallow=all
>>>> allow=alaw
>>>>
>>>>
>>>> i want my sip profile to check if network address is 1.1.1.1, it uses a
>>>> particular codec lets say pcma but if network address is 2.2.2.2 it uses the
>>>> defult
>>>> and this i want to define for around 40 ip addresses
>>>>  _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nazim.aghabayov at gmail.com  Tue Dec 21 14:27:17 2010
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Tue, 21 Dec 2010 15:27:17 +0400
Subject: [Freeswitch-users] loops
In-Reply-To: <AANLkTim0iPTrZL64MGJzrW0XT2z6+XFXzfYKW_MeW-FA@mail.gmail.com>
References: <AANLkTim0iPTrZL64MGJzrW0XT2z6+XFXzfYKW_MeW-FA@mail.gmail.com>
Message-ID: <4D108F15.5070700@gmail.com>

You are trying to call the function which is not declared yet. First
define a function, then call it.
You should really read this: http://www.lua.org/manual/5.1/

Regards,
Nazim

On 12/21/2010 01:36 PM, Sam wrote:
> Hello,
>
> I am trying to get a loop done using if ... then .. return function,
> using below syntax but getting error on it.
>
>
> if (session:ready()) then return ack()
> end
>
>
> function ack()
>
> session:answer();
> session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/zrtp/8000/zrtp-somethings_wrong.wav")
>
> end
>
>
>
> error:---
> EXECUTE sofia/internal/7001 at 192.168.2.190 lua(hello.lua)
> 2010-12-21 14:55:34.724790 [ERR] mod_lua.cpp:182
> /usr/local/freeswitch/scripts/hello.lua:1: attempt to call global 'ack' (a
> nil value)
> stack traceback:
>         /usr/local/freeswitch/scripts/hello.lua:1: in main chunk
>
>
> Any ideas to resolve this.
>
> Regards
> Sam
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From marcdecorny at gmail.com  Tue Dec 21 17:28:44 2010
From: marcdecorny at gmail.com (Marc de Corny)
Date: Tue, 21 Dec 2010 14:28:44 +0000
Subject: [Freeswitch-users] Lua not playing any wav files
In-Reply-To: <AANLkTimKUWvBgo9M980xDJqCSHN8cCePGffPw-M2varx@mail.gmail.com>
References: <AANLkTimHrCzzzg8S=PVTVDmH1sd4iv4jQiMFbFHR7teo@mail.gmail.com>
	<AANLkTik3MPkku8-Zk-JKRezUs=tMmrrG=xmCvZPcQ3wt@mail.gmail.com>
	<C4D7BFCA-3104-4D35-8F47-E45064DA2D30@gmail.com>
	<AANLkTimKUWvBgo9M980xDJqCSHN8cCePGffPw-M2varx@mail.gmail.com>
Message-ID: <AANLkTimgR1uECs_q24uYbSSQZ2aiUV-DE4njZxpE2jNM@mail.gmail.com>

Hi Sam

I tried your method with the sleep(100) in front, but it made no difference.

I changed the lua script to :

local ddi = argv[1]
-- answer the call
session:answer();
freeswitch.consoleLog("info", "All Answered\n");
-- sleep a second
session:sleep(100);
ivr_invalid_msg = sound_file_folder .. "invalid_msg.wav"
main_msg = sound_file_folder .. "default_autoattendant.wav"
-- Play with Execute
*session:execute("playback","/tmp/main.wav");*
-- Play with StreamFile
session:streamFile(ivr_invalid_msg);
dialstr_main  = dialstr_main .. dialstr_prefix .. breakoutcode ..
"02031701665"

In the logs I get

2010-12-21 11:07:00.202821 [DEBUG] switch_core_session.c:1882 Application
playback Requires media! pre_answering channel
sofia/external/2031701665 at 194.0.147.16:5060
EXECUTE sofia/external/2031701665 at 194.0.147.16:5060 playback(/tmp/main.wav)
2010-12-21 11:07:00.202821 [DEBUG] switch_cpp.cpp:972
sofia/external/2031701665 at 194.0.147.16:5060 destroy/unlink session from
object
It is strange, because I can see that the command is being run, but as soon
as it gets run, I get in the same millisecond a desctry/unlink which maybe
exists the command.

Are there any additional logs I can take to understand?

thanks
Marc

On Sat, Dec 18, 2010 at 9:16 AM, Sam <u2nsam at gmail.com> wrote:

> you can try this ...
>
>
> session:answer();
>
> -- sleep a second
> session:sleep(100);
>
> -- play a file
>
> session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav");
>
> Regards
> Sam
>
>
>
> On Sat, Dec 18, 2010 at 2:33 PM, Marc De Corny <marcdecorny at gmail.com>wrote:
>
>>  Thanks Chris,
>> Stil got issue unfortunately
>> i tried with both answer and preAnswer and got same results. I will post
>> the logs of the two examples and see.
>> Thanks for following up on this.
>> Marc
>>
>>
>> On 18 Dec 2010, at 08:42, Chris Burns <chris at cloudtel.com> wrote:
>>
>>    Hmmm no help for you yet huh ... you may have solved it on your own
>> already, but ...
>>
>> You want to answer the call there, and not pre-answer. Pre-answer is for
>> early media, which is for exchanging media before committing to answer the
>> call. Admittedly you should hear something either way, but you definitely
>> want to answer the call in your case. If your XML dialplan works as you
>> said, you should compare the log output between these 2 extensions:
>>
>> <extension name="testXML">
>>   <condition field="destination_number" expression="^9001$">
>>     <action application="answer"/>
>>     <action application="playback" data="local_stream://moh"/>
>>   </condition>
>> </extension>
>>
>> <extension name="testLUA">
>>   <condition field="destination_number" expression="^9002$">
>>     <action application="lua" data="test.lua"/>
>>   </condition>
>> </extension>
>>
>> Contents of test.lua:
>> session:answer();
>> session:execute("playback","local_stream://moh");
>>
>>
>> On Wed, Dec 15, 2010 at 1:59 AM, Marc de Corny < <marcdecorny at gmail.com>
>> marcdecorny at gmail.com> wrote:
>>
>>> Hi all,
>>>
>>> I have run into an issue on something so basic that I must be as simple
>>> as enabling a feature somewhere.
>>>
>>> I have been trying to get lua to play a message from a WAV file. I have
>>> tried session:execute("playback", main_msg) and
>>> session:streamFile(ivr_invalid_msg) but neither of them play any music to
>>> the caller. I tried both to answer and preAnswer the call first but it made
>>> no difference. However if I put the same file into the XML dialplan and play
>>> it with the commands below I hear the music fine.
>>> <action application="set" data="playback_terminators=#"/>
>>> <action application="playback" data="${401_sound_file_path}"/>
>>>
>>> The issue only seems to be from lua when playing any type of wav file and
>>> those files are definitelly there as can be read by the XML
>>>
>>> The error message is below for the execute(playback) command, but nothing
>>> can be seen for the
>>> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application
>>> playback Requires media! pre_answering channel
>>> sofia/external/2031701665 at 194.0.147.16:5060 EXECUTE
>>> sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswitch/sounds/svc_sound_files/default_autoattendant.wav)
>>> But there is no mention of the streamFile command. I have had similar
>>> issue with the PlayAndGetDigits command.
>>> Is there something that I need to enable in lua so that is can playback
>>> messages to the caller.
>>>
>>> Many thanks to anyone who can help.
>>> Marc
>>>
>>>
>>> below is the XML dialplan and lua script as well as the log at the very
>>> end.
>>>
>>> XML DIALPLAN:
>>> <include>
>>> <extension name="IVR_FROM_MYSQL">
>>> <condition field="caller_id_number" expression="Anonymous" break="never">
>>> <action application="set" data="effective_caller_id_number=0000000000"/>
>>> </condition>
>>> <condition field="destination_number" expression="^(404)">
>>> <action inline="true" application="lua" data="ivr_mysql.lua
>>> ${destination_number:3}"/>
>>> <action application="set" data="effective_caller_id_name=${404_tag}"/>
>>> <action application="bridge" data="${404_dial}"/>
>>> </condition>
>>> </extension>
>>> </include>
>>>
>>> The LUA script ivr_mysql.lua is callsed and this is it.
>>> -- IVR : PLAY IVR WAV FILES
>>> -- Global Variables:
>>> local dialstr_prefix = "sofia/gateway/CS2k/"
>>> local dialstr_main = ""
>>> local breakoutcode = "184"
>>> local sound_file_folder = "/usr/local/freeswitch/sounds/svc_sound_files/"
>>> local ddi = argv[1]
>>> -- answer the call
>>> session:preAnswer();
>>> freeswitch.consoleLog("info", "All Answered\n");
>>> ivr_invalid_msg = sound_file_folder .. "invalid_msg.wav"
>>> main_msg = sound_file_folder .. "default_autoattendant.wav"
>>> -- Play with Execute
>>> session:execute("playback", main_msg)
>>> -- Play with StreamFile
>>> session:streamFile(ivr_invalid_msg);
>>> dialstr_main  = dialstr_main .. dialstr_prefix .. breakoutcode ..
>>> "02031701665"
>>> session:setVariable("404_dial",dialstr_main)
>>> session:setVariable("404_tag","IVR")
>>>
>>>
>>> RELEVANT LOGS :
>>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Regex (PASS)
>>> [IVR_FROM_MYS                          QL] destination_number(4042031956241)
>>> =~ /^(404)/ break=on-false
>>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
>>> lua(ivr_mysql.lua $                          {destination_number:3}) INLINE
>>> EXECUTE sofia/external/2031701665 at 194.0.147.16:5060 lua(ivr_mysql.lua
>>> 2031956241                          )
>>> 2010-12-01 11:56:01.525426 [INFO] switch_cpp.cpp:584 Sending early media
>>> 2010-12-01 11:56:01.525426 [DEBUG] sofia_glue.c:2972 AUDIO RTP
>>> [sofia/external/2                          031701665 at 194.0.147.16:5060]
>>> 10.5.2.105 port 29900 -> 194.0.147.164 port 50202 c
>>> odec: 8 ms: 20
>>> 2010-12-01 11:56:01.525426 [DEBUG] switch_rtp.c:1418 Starting timer
>>> [soft] 160 b                          ytes per 20ms
>>> 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3190 Set 2833 dtmf send
>>> payload                           to 101
>>> 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3195 Set 2833 dtmf
>>> receive paylo                          ad to 101
>>> 2010-12-01 11:56:01.532280 [DEBUG] mod_sofia.c:2172 Ring SDP:
>>> v=0
>>> o=FreeSWITCH 1291174661 1291174662 IN IP4 10.5.2.105
>>> s=FreeSWITCH
>>> c=IN IP4 10.5.2.105
>>> t=0 0
>>> m=audio 29900 RTP/AVP 8 101
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>> a=sendrecv
>>> 2010-12-01 11:56:01.532280 [NOTICE] mod_sofia.c:2175 Pre-Answer
>>> sofia/external/2                          031701665 at 194.0.147.16:5060!
>>> 2010-12-01 11:56:01.532280 [DEBUG] switch_channel.c:2544
>>> (sofia/external/2031701                          665 at 194.0.147.16:5060)
>>> Callstate Change RINGING -> EARLY
>>> 2010-12-01 11:56:01.534727 [DEBUG] sofia.c:4576 Channel
>>> sofia/external/203170166                          5 at 194.0.147.16:5060skipping state [early][183]
>>> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:676 Send signal
>>> sofia/e                          xternal/2031701665 at 194.0.147.16:5060[BREAK]
>>> 2010-12-01 11:56:01.534727 [INFO] switch_cpp.cpp:1181 All Answered
>>> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application
>>> playba                          ck Requires media! pre_answering channel
>>> <sofia/external/2031701665 at 194.0.147.16>
>>> sofia/external/2031701665 at 194.0.147.16:                          5060
>>> EXECUTE sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswit
>>> ch/sounds/svc_sound_files/default_autoattendant.wav)
>>> 2010-12-01 11:56:01.537644 [DEBUG] switch_cpp.cpp:972
>>> sofia/external/2031701665@                          194.0.147.16:5060destroy/unlink session from object
>>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
>>> set(effective_calle                          r_id_name=${404_tag})
>>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
>>> bridge(${404_dial})
>>> 2010-12-01 11:56:01.537644 [DEBUG] switch_core_state_machine.c:119
>>> (sofia/extern                          al/2031701665 at 194.0.147.16:5060)
>>> State Change CS_ROUTING -> CS_EXECUTE
>>> 2010-12-01 11:56:01.537644 [DEBUG] switch_core_session.c:1057 Send signal
>>> sofia/                          external/2031701665 at 194.0.147.16:5060[BREAK]
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
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>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> <http://www.freeswitch.org/>http://www.freeswitch.org
>>>
>>>
>>  _______________________________________________
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From helmut.kuper at ewetel.de  Tue Dec 21 17:34:48 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 21 Dec 2010 15:34:48 +0100
Subject: [Freeswitch-users] Support for RTCP report via SIP PUBLISH
Message-ID: <4D10BB08.1050205@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi,

I want to collect RTCP reports from snom devices via FS's event system.
Unfortunately it doesn't support it. Maybe because it is still a draft:

http://tools.ietf.org/html/draft-ietf-sipping-rtcp-summary-05

I wonder whether this feature is planed in near future.



Nevertheless I hacked sofia.c and sofia-presence.c to get it working
just for testing. Here is the "git diff":

-
---DIFF------------------------------------------------------------------------
diff --git a/src/mod/endpoints/mod_sofia/sofia.c
b/src/mod/endpoints/mod_sofia/sofia.c
index 2e13b54..3622506 100644
- --- a/src/mod/endpoints/mod_sofia/sofia.c
+++ b/src/mod/endpoints/mod_sofia/sofia.c
@@ -1530,6 +1530,7 @@ void *SWITCH_THREAD_FUNC
sofia_profile_thread_run(switch_thread_t *thread, void
                                   TAG_IF(profile->pres_type,
NUTAG_ALLOW_EVENTS("include-session-description")),
                                   TAG_IF(profile->pres_type,
NUTAG_ALLOW_EVENTS("presence.winfo")),
                                   TAG_IF(profile->pres_type,
NUTAG_ALLOW_EVENTS("message-summary")),
+                                       TAG_IF(profile->pres_type,
NUTAG_ALLOW_EVENTS("vq-rtcpxr")),
                                   NUTAG_ALLOW_EVENTS("refer"),
SIPTAG_SUPPORTED_STR(supported),
SIPTAG_USER_AGENT_STR(profile->user_agent), TAG_END(

        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Set
params for %s\n", profile->name);
diff --git a/src/mod/endpoints/mod_sofia/sofia_presence.c
b/src/mod/endpoints/mod_sofia/sofia_presence.c
index 143666c..c998280 100644
- --- a/src/mod/endpoints/mod_sofia/sofia_presence.c
+++ b/src/mod/endpoints/mod_sofia/sofia_presence.c
@@ -2679,7 +2679,17 @@ void sofia_presence_handle_sip_i_publish(nua_t
*nua, sofia_profile_t *profile, n

                        pd_dup = strdup(payload->pl_data);

- -                       if ((xml = switch_xml_parse_str(pd_dup,
strlen(pd_dup)))) {
+            //EWETEL RTCP hack
+            //event_type = sip_header_as_string(profile->home, (void *)
sip->sip_event);
+            if (strcmp(sip_header_as_string(profile->home, (void *)
sip->sip_event), "vq-rtcpxr")==0) {
+                if (switch_event_create(&event,
SWITCH_EVENT_PRESENCE_IN) == SWITCH_STATUS_SUCCESS) {
+                    switch_event_add_header_string(event,
SWITCH_STACK_BOTTOM, "event_type", "vq-rtcpxr");
+                    switch_event_add_body(event, "%s", pd_dup);
+                    switch_event_fire(&event);
+                }
+            }
+                       //DONE
+                       else if ((xml = switch_xml_parse_str(pd_dup,
strlen(pd_dup)))) {
                                char *open_closed = "", *note_txt = "";

                                if (sip->sip_user_agent) {
- ---DIFF END-----------------------------------------------------------

It simply allows SIP PUBLISH events of type "vq-rtcpxr" and adds the sip
payload to the internal event body.



The corresponding event is this:

Content-Length: 1178
Content-Type: text/event-plain

Event-Name: PRESENCE_IN
Core-UUID: f852daae-6da9-4979-8dc8-fa11651a7891
FreeSWITCH-Hostname: test
FreeSWITCH-IPv4: 1.2.3.4
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2010-12-21%2015%3A19%3A09
Event-Date-GMT: Tue,%2021%20Dec%202010%2014%3A19%3A09%20GMT
Event-Date-Timestamp: 1292941149981062
Event-Calling-File: sofia_presence.c
Event-Calling-Function: sofia_presence_handle_sip_i_publish
Event-Calling-Line-Number: 2685
event_type: vq-rtcpxr
Content-Length: 703

VQSessionReport
LocalMetrics:
Timestamps:START=2010-12-21T14:19:08Z STOP=2010-12-21T14:19:10Z
SessionDesc:PT=9 PD=G.722 PPS=50 SSUP=off
CallID:3c2f99c5dc8f-k1jiuwkd70m9
x-UserAgent:snom370/8.4.22
FromID:"Helmut Kuper" <sip:8000 at sipdomain.de>
ToID:<sip:***6 at sipdomain.de;user=phone>
LocalAddr:IP=85.16.245.234 PORT=10568 SSRC=0xF919B973
RemoteAddr:IP=85.16.246.16 PORT=19770 SSRC=0x
DialogID:3c2f99c5dc8f-k1jiuwkd70m9;to-tag=7HU6y795c8r8B;from-tag=rbxbm5jjme
x-SIPmetrics:SVA=RG SRD=890 SFC=0
x-SIPterm:SDC=OK SDR=OR
JitterBuffer:JBA=0 JBR=0 JBN=0 JBM=0 JBX=65535
PacketLoss:NLR=0.0 JDR=0.0
BurstGapLoss:BLD=0.0 BD=0 GLD=0.0 GD=2440 GMIN=16
Delay:RTD=0 ESD=0 IAJ=0
QualityEst:MOSLQ=4.2 MOSCQ=4.1



regards
helmut
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Version: GnuPG v1.4.10 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

iEYEARECAAYFAk0QuwgACgkQ4tZeNddg3dx6/gCgij7KZDD0qHLFkujDjzH/hYu+
7UEAn2SM1eFQsPA6ICNxtexEprdWyGCk
=kbkn
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From marcdecorny at gmail.com  Tue Dec 21 17:51:21 2010
From: marcdecorny at gmail.com (Marc de Corny)
Date: Tue, 21 Dec 2010 14:51:21 +0000
Subject: [Freeswitch-users] Lua not playing any wav files
In-Reply-To: <AANLkTik3MPkku8-Zk-JKRezUs=tMmrrG=xmCvZPcQ3wt@mail.gmail.com>
References: <AANLkTimHrCzzzg8S=PVTVDmH1sd4iv4jQiMFbFHR7teo@mail.gmail.com>
	<AANLkTik3MPkku8-Zk-JKRezUs=tMmrrG=xmCvZPcQ3wt@mail.gmail.com>
Message-ID: <AANLkTimXMWewnFnnMuNAeuZySy4Hcb_d2vopHpmPVo+9@mail.gmail.com>

Hi Chris

That seemed to work I got audio in both scnearios, in the XML and LUA
script.

so the only difference between my originial issue and the successful moh
test. is that I was trying to playback a file and here we are playing a
local stream://moh.

I have changed all the files too chmod 777, is there any thing else on the
rights point of view that I need to know in order to get it to work?

I am grateful for any ideas or anything else that could help me understand
what the issue is

thanks
Marc




On Sat, Dec 18, 2010 at 8:42 AM, Chris Burns <chris at cloudtel.com> wrote:

> Hmmm no help for you yet huh ... you may have solved it on your own
> already, but ...
>
> You want to answer the call there, and not pre-answer. Pre-answer is for
> early media, which is for exchanging media before committing to answer the
> call. Admittedly you should hear something either way, but you definitely
> want to answer the call in your case. If your XML dialplan works as you
> said, you should compare the log output between these 2 extensions:
>
> <extension name="testXML">
>   <condition field="destination_number" expression="^9001$">
>     <action application="answer"/>
>     <action application="playback" data="local_stream://moh"/>
>   </condition>
> </extension>
>
> <extension name="testLUA">
>   <condition field="destination_number" expression="^9002$">
>     <action application="lua" data="test.lua"/>
>   </condition>
> </extension>
>
> Contents of test.lua:
> session:answer();
> session:execute("playback","local_stream://moh");
>
>
>   On Wed, Dec 15, 2010 at 1:59 AM, Marc de Corny <marcdecorny at gmail.com>wrote:
>
>>  Hi all,
>>
>> I have run into an issue on something so basic that I must be as simple as
>> enabling a feature somewhere.
>>
>> I have been trying to get lua to play a message from a WAV file. I have
>> tried session:execute("playback", main_msg) and
>> session:streamFile(ivr_invalid_msg) but neither of them play any music to
>> the caller. I tried both to answer and preAnswer the call first but it made
>> no difference. However if I put the same file into the XML dialplan and play
>> it with the commands below I hear the music fine.
>> <action application="set" data="playback_terminators=#"/>
>> <action application="playback" data="${401_sound_file_path}"/>
>>
>> The issue only seems to be from lua when playing any type of wav file and
>> those files are definitelly there as can be read by the XML
>>
>> The error message is below for the execute(playback) command, but nothing
>> can be seen for the
>> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application
>> playback Requires media! pre_answering channel
>> sofia/external/2031701665 at 194.0.147.16:5060 EXECUTE
>> sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswitch/sounds/svc_sound_files/default_autoattendant.wav)
>> But there is no mention of the streamFile command. I have had similar
>> issue with the PlayAndGetDigits command.
>> Is there something that I need to enable in lua so that is can playback
>> messages to the caller.
>>
>> Many thanks to anyone who can help.
>> Marc
>>
>>
>> below is the XML dialplan and lua script as well as the log at the very
>> end.
>>
>> XML DIALPLAN:
>> <include>
>> <extension name="IVR_FROM_MYSQL">
>> <condition field="caller_id_number" expression="Anonymous" break="never">
>> <action application="set" data="effective_caller_id_number=0000000000"/>
>> </condition>
>> <condition field="destination_number" expression="^(404)">
>> <action inline="true" application="lua" data="ivr_mysql.lua
>> ${destination_number:3}"/>
>> <action application="set" data="effective_caller_id_name=${404_tag}"/>
>> <action application="bridge" data="${404_dial}"/>
>> </condition>
>> </extension>
>> </include>
>>
>> The LUA script ivr_mysql.lua is callsed and this is it.
>> -- IVR : PLAY IVR WAV FILES
>> -- Global Variables:
>> local dialstr_prefix = "sofia/gateway/CS2k/"
>> local dialstr_main = ""
>> local breakoutcode = "184"
>> local sound_file_folder = "/usr/local/freeswitch/sounds/svc_sound_files/"
>> local ddi = argv[1]
>> -- answer the call
>> session:preAnswer();
>> freeswitch.consoleLog("info", "All Answered\n");
>> ivr_invalid_msg = sound_file_folder .. "invalid_msg.wav"
>> main_msg = sound_file_folder .. "default_autoattendant.wav"
>> -- Play with Execute
>> session:execute("playback", main_msg)
>> -- Play with StreamFile
>> session:streamFile(ivr_invalid_msg);
>> dialstr_main  = dialstr_main .. dialstr_prefix .. breakoutcode ..
>> "02031701665"
>> session:setVariable("404_dial",dialstr_main)
>> session:setVariable("404_tag","IVR")
>>
>>
>> RELEVANT LOGS :
>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Regex (PASS)
>> [IVR_FROM_MYS                          QL] destination_number(4042031956241)
>> =~ /^(404)/ break=on-false
>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
>> lua(ivr_mysql.lua $                          {destination_number:3}) INLINE
>> EXECUTE sofia/external/2031701665 at 194.0.147.16:5060 lua(ivr_mysql.lua
>> 2031956241                          )
>> 2010-12-01 11:56:01.525426 [INFO] switch_cpp.cpp:584 Sending early media
>> 2010-12-01 11:56:01.525426 [DEBUG] sofia_glue.c:2972 AUDIO RTP
>> [sofia/external/2                          031701665 at 194.0.147.16:5060]
>> 10.5.2.105 port 29900 -> 194.0.147.164 port 50202 c
>> odec: 8 ms: 20
>> 2010-12-01 11:56:01.525426 [DEBUG] switch_rtp.c:1418 Starting timer [soft]
>> 160 b                          ytes per 20ms
>> 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3190 Set 2833 dtmf send
>> payload                           to 101
>> 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3195 Set 2833 dtmf receive
>> paylo                          ad to 101
>> 2010-12-01 11:56:01.532280 [DEBUG] mod_sofia.c:2172 Ring SDP:
>> v=0
>> o=FreeSWITCH 1291174661 1291174662 IN IP4 10.5.2.105
>> s=FreeSWITCH
>> c=IN IP4 10.5.2.105
>> t=0 0
>> m=audio 29900 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>> 2010-12-01 11:56:01.532280 [NOTICE] mod_sofia.c:2175 Pre-Answer
>> sofia/external/2                          031701665 at 194.0.147.16:5060!
>> 2010-12-01 11:56:01.532280 [DEBUG] switch_channel.c:2544
>> (sofia/external/2031701                          665 at 194.0.147.16:5060)
>> Callstate Change RINGING -> EARLY
>> 2010-12-01 11:56:01.534727 [DEBUG] sofia.c:4576 Channel
>> sofia/external/203170166                          5 at 194.0.147.16:5060skipping state [early][183]
>> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:676 Send signal
>> sofia/e                          xternal/2031701665 at 194.0.147.16:5060[BREAK]
>> 2010-12-01 11:56:01.534727 [INFO] switch_cpp.cpp:1181 All Answered
>> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application
>> playba                          ck Requires media! pre_answering channel
>> sofia/external/2031701665 at 194.0.147.16:                          5060
>> EXECUTE sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswit
>> ch/sounds/svc_sound_files/default_autoattendant.wav)
>> 2010-12-01 11:56:01.537644 [DEBUG] switch_cpp.cpp:972
>> sofia/external/2031701665@                          194.0.147.16:5060destroy/unlink session from object
>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
>> set(effective_calle                          r_id_name=${404_tag})
>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
>> bridge(${404_dial})
>> 2010-12-01 11:56:01.537644 [DEBUG] switch_core_state_machine.c:119
>> (sofia/extern                          al/2031701665 at 194.0.147.16:5060)
>> State Change CS_ROUTING -> CS_EXECUTE
>> 2010-12-01 11:56:01.537644 [DEBUG] switch_core_session.c:1057 Send signal
>> sofia/                          external/2031701665 at 194.0.147.16:5060[BREAK]
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From marcdecorny at gmail.com  Tue Dec 21 18:17:39 2010
From: marcdecorny at gmail.com (Marc de Corny)
Date: Tue, 21 Dec 2010 15:17:39 +0000
Subject: [Freeswitch-users] Lua not playing any wav files
In-Reply-To: <AANLkTimXMWewnFnnMuNAeuZySy4Hcb_d2vopHpmPVo+9@mail.gmail.com>
References: <AANLkTimHrCzzzg8S=PVTVDmH1sd4iv4jQiMFbFHR7teo@mail.gmail.com>
	<AANLkTik3MPkku8-Zk-JKRezUs=tMmrrG=xmCvZPcQ3wt@mail.gmail.com>
	<AANLkTimXMWewnFnnMuNAeuZySy4Hcb_d2vopHpmPVo+9@mail.gmail.com>
Message-ID: <AANLkTimZwk-R5VQDVQEpypKchrKBaQ5zr-ZtVcnrHwEG@mail.gmail.com>

RESOLVED :

for anyone with the same problem, I have just foudn the solution.

My action was being executed inline
<action inline="true" application="lua" data="ivr_mysql.lua
${destination_number:3}"/>

This does not allow you to playback anything.

Thanks
Marc

On Tue, Dec 21, 2010 at 2:51 PM, Marc de Corny <marcdecorny at gmail.com>wrote:

> Hi Chris
>
> That seemed to work I got audio in both scnearios, in the XML and LUA
> script.
>
> so the only difference between my originial issue and the successful moh
> test. is that I was trying to playback a file and here we are playing a
> local stream://moh.
>
> I have changed all the files too chmod 777, is there any thing else on the
> rights point of view that I need to know in order to get it to work?
>
> I am grateful for any ideas or anything else that could help me understand
> what the issue is
>
> thanks
> Marc
>
>
>
>
> On Sat, Dec 18, 2010 at 8:42 AM, Chris Burns <chris at cloudtel.com> wrote:
>
>> Hmmm no help for you yet huh ... you may have solved it on your own
>> already, but ...
>>
>> You want to answer the call there, and not pre-answer. Pre-answer is for
>> early media, which is for exchanging media before committing to answer the
>> call. Admittedly you should hear something either way, but you definitely
>> want to answer the call in your case. If your XML dialplan works as you
>> said, you should compare the log output between these 2 extensions:
>>
>> <extension name="testXML">
>>   <condition field="destination_number" expression="^9001$">
>>     <action application="answer"/>
>>     <action application="playback" data="local_stream://moh"/>
>>   </condition>
>> </extension>
>>
>> <extension name="testLUA">
>>   <condition field="destination_number" expression="^9002$">
>>     <action application="lua" data="test.lua"/>
>>   </condition>
>> </extension>
>>
>> Contents of test.lua:
>> session:answer();
>> session:execute("playback","local_stream://moh");
>>
>>
>>   On Wed, Dec 15, 2010 at 1:59 AM, Marc de Corny <marcdecorny at gmail.com>wrote:
>>
>>>  Hi all,
>>>
>>> I have run into an issue on something so basic that I must be as simple
>>> as enabling a feature somewhere.
>>>
>>> I have been trying to get lua to play a message from a WAV file. I have
>>> tried session:execute("playback", main_msg) and
>>> session:streamFile(ivr_invalid_msg) but neither of them play any music to
>>> the caller. I tried both to answer and preAnswer the call first but it made
>>> no difference. However if I put the same file into the XML dialplan and play
>>> it with the commands below I hear the music fine.
>>> <action application="set" data="playback_terminators=#"/>
>>> <action application="playback" data="${401_sound_file_path}"/>
>>>
>>> The issue only seems to be from lua when playing any type of wav file and
>>> those files are definitelly there as can be read by the XML
>>>
>>> The error message is below for the execute(playback) command, but nothing
>>> can be seen for the
>>> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application
>>> playback Requires media! pre_answering channel
>>> sofia/external/2031701665 at 194.0.147.16:5060 EXECUTE
>>> sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswitch/sounds/svc_sound_files/default_autoattendant.wav)
>>> But there is no mention of the streamFile command. I have had similar
>>> issue with the PlayAndGetDigits command.
>>> Is there something that I need to enable in lua so that is can playback
>>> messages to the caller.
>>>
>>> Many thanks to anyone who can help.
>>> Marc
>>>
>>>
>>> below is the XML dialplan and lua script as well as the log at the very
>>> end.
>>>
>>> XML DIALPLAN:
>>> <include>
>>> <extension name="IVR_FROM_MYSQL">
>>> <condition field="caller_id_number" expression="Anonymous" break="never">
>>> <action application="set" data="effective_caller_id_number=0000000000"/>
>>> </condition>
>>> <condition field="destination_number" expression="^(404)">
>>> <action inline="true" application="lua" data="ivr_mysql.lua
>>> ${destination_number:3}"/>
>>> <action application="set" data="effective_caller_id_name=${404_tag}"/>
>>> <action application="bridge" data="${404_dial}"/>
>>> </condition>
>>> </extension>
>>> </include>
>>>
>>> The LUA script ivr_mysql.lua is callsed and this is it.
>>> -- IVR : PLAY IVR WAV FILES
>>> -- Global Variables:
>>> local dialstr_prefix = "sofia/gateway/CS2k/"
>>> local dialstr_main = ""
>>> local breakoutcode = "184"
>>> local sound_file_folder = "/usr/local/freeswitch/sounds/svc_sound_files/"
>>> local ddi = argv[1]
>>> -- answer the call
>>> session:preAnswer();
>>> freeswitch.consoleLog("info", "All Answered\n");
>>> ivr_invalid_msg = sound_file_folder .. "invalid_msg.wav"
>>> main_msg = sound_file_folder .. "default_autoattendant.wav"
>>> -- Play with Execute
>>> session:execute("playback", main_msg)
>>> -- Play with StreamFile
>>> session:streamFile(ivr_invalid_msg);
>>> dialstr_main  = dialstr_main .. dialstr_prefix .. breakoutcode ..
>>> "02031701665"
>>> session:setVariable("404_dial",dialstr_main)
>>> session:setVariable("404_tag","IVR")
>>>
>>>
>>> RELEVANT LOGS :
>>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Regex (PASS)
>>> [IVR_FROM_MYS                          QL] destination_number(4042031956241)
>>> =~ /^(404)/ break=on-false
>>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
>>> lua(ivr_mysql.lua $                          {destination_number:3}) INLINE
>>> EXECUTE sofia/external/2031701665 at 194.0.147.16:5060 lua(ivr_mysql.lua
>>> 2031956241                          )
>>> 2010-12-01 11:56:01.525426 [INFO] switch_cpp.cpp:584 Sending early media
>>> 2010-12-01 11:56:01.525426 [DEBUG] sofia_glue.c:2972 AUDIO RTP
>>> [sofia/external/2                          031701665 at 194.0.147.16:5060]
>>> 10.5.2.105 port 29900 -> 194.0.147.164 port 50202 c
>>> odec: 8 ms: 20
>>> 2010-12-01 11:56:01.525426 [DEBUG] switch_rtp.c:1418 Starting timer
>>> [soft] 160 b                          ytes per 20ms
>>> 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3190 Set 2833 dtmf send
>>> payload                           to 101
>>> 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3195 Set 2833 dtmf
>>> receive paylo                          ad to 101
>>> 2010-12-01 11:56:01.532280 [DEBUG] mod_sofia.c:2172 Ring SDP:
>>> v=0
>>> o=FreeSWITCH 1291174661 1291174662 IN IP4 10.5.2.105
>>> s=FreeSWITCH
>>> c=IN IP4 10.5.2.105
>>> t=0 0
>>> m=audio 29900 RTP/AVP 8 101
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>> a=sendrecv
>>> 2010-12-01 11:56:01.532280 [NOTICE] mod_sofia.c:2175 Pre-Answer
>>> sofia/external/2                          031701665 at 194.0.147.16:5060!
>>> 2010-12-01 11:56:01.532280 [DEBUG] switch_channel.c:2544
>>> (sofia/external/2031701                          665 at 194.0.147.16:5060)
>>> Callstate Change RINGING -> EARLY
>>> 2010-12-01 11:56:01.534727 [DEBUG] sofia.c:4576 Channel
>>> sofia/external/203170166                          5 at 194.0.147.16:5060skipping state [early][183]
>>> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:676 Send signal
>>> sofia/e                          xternal/2031701665 at 194.0.147.16:5060[BREAK]
>>> 2010-12-01 11:56:01.534727 [INFO] switch_cpp.cpp:1181 All Answered
>>> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application
>>> playba                          ck Requires media! pre_answering channel
>>> sofia/external/2031701665 at 194.0.147.16:                          5060
>>> EXECUTE sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswit
>>> ch/sounds/svc_sound_files/default_autoattendant.wav)
>>> 2010-12-01 11:56:01.537644 [DEBUG] switch_cpp.cpp:972
>>> sofia/external/2031701665@                          194.0.147.16:5060destroy/unlink session from object
>>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
>>> set(effective_calle                          r_id_name=${404_tag})
>>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
>>> bridge(${404_dial})
>>> 2010-12-01 11:56:01.537644 [DEBUG] switch_core_state_machine.c:119
>>> (sofia/extern                          al/2031701665 at 194.0.147.16:5060)
>>> State Change CS_ROUTING -> CS_EXECUTE
>>> 2010-12-01 11:56:01.537644 [DEBUG] switch_core_session.c:1057 Send signal
>>> sofia/                          external/2031701665 at 194.0.147.16:5060[BREAK]
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From jeff at jefflenk.com  Tue Dec 21 18:29:27 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 21 Dec 2010 09:29:27 -0600
Subject: [Freeswitch-users] compile error on libmp3lame and mod_shout
In-Reply-To: <SNT109-DS118A0D1D7049B8F876EE9C11A0@phx.gbl>
References: <SNT109-DS118A0D1D7049B8F876EE9C11A0@phx.gbl>
Message-ID: <COL117-W475362B9E13477B3355945B91A0@phx.gbl>


Norman,
 
Look in the log when you are first building - do you have any download errors.
 
Also are these files present in fs folder\libs
 
lame-3.97.tar.gz 
mpg123.tar.gz
 
if not you had a download error when first building.
 


From: hwnorman at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 21 Dec 2010 16:42:47 +0800
Subject: [Freeswitch-users] compile error on libmp3lame and mod_shout






Hi there 
 
I am using AMD athlon X2, windows xp sp3, VC++2008 express (only installed this software) using the Git source downloaded today the first time 
 
I am getting these 2 error ,  is there any ways to resolve this or  work around.
 
Please advise 
 
Norman Lam 
 
 
---rest of message removed
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 		 	   		  
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From adminjew at gmail.com  Tue Dec 21 19:41:17 2010
From: adminjew at gmail.com (Yitzchok)
Date: Tue, 21 Dec 2010 11:41:17 -0500
Subject: [Freeswitch-users] VPS Host suggestions?
In-Reply-To: <AANLkTi=haM9NkLrQgsyUvnf431L2bYpA+w=z6DDVowNz@mail.gmail.com>
References: <AANLkTinawP5qWB4447QveMqAq0fvvtOtsen74FYdF30-@mail.gmail.com>
	<AANLkTi=haM9NkLrQgsyUvnf431L2bYpA+w=z6DDVowNz@mail.gmail.com>
Message-ID: <AANLkTimN7ss22AfXeXSW=Gug5-iQ6+atNrAa4njgGX+C@mail.gmail.com>

I have one of my servers running on Linode 1024 and and I was able to get a
little more then 30 concurrent calls with transcoding (CPU was the
bottleneck).

Linode 512 will probably work for what you need and you can upgrade at any
time.

There is also http://lylix.net/ that specialize in voip but i don't know if
you can get FS on there.


Yitzchok


On Tue, Dec 21, 2010 at 4:03 AM, Avi Marcus <Avi at amarcus.com> wrote:

> I'm running less than 5 concurrent calls for the next month at least
> and slowly gaining volume, so I don't need anything fancy.
> The wiki seems to say that Xen virtualization works with FS.
>
> Any recommendations for VPS hosting? I know you get what you pay for,
> but I'm sure there's inexpensive that's not bad.
>
> Linode has a very slick website and feature list, 64bit hosting, and
> 512mb/200gb bandwidth for $20/month.
> Any VPS for voip suggestions?
> Thanks!
>
> -Avi Marcus
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From norm at voicenetwork.ca  Tue Dec 21 19:53:38 2010
From: norm at voicenetwork.ca (Norman Tomlins)
Date: Tue, 21 Dec 2010 11:53:38 -0500
Subject: [Freeswitch-users] VPS Host suggestions?
In-Reply-To: <AANLkTi=haM9NkLrQgsyUvnf431L2bYpA+w=z6DDVowNz@mail.gmail.com>
References: <AANLkTinawP5qWB4447QveMqAq0fvvtOtsen74FYdF30-@mail.gmail.com>
	<AANLkTi=haM9NkLrQgsyUvnf431L2bYpA+w=z6DDVowNz@mail.gmail.com>
Message-ID: <AANLkTikDLDt1nvooNgYtwKsYQz-3xwsMV3zFXf41aeat@mail.gmail.com>

Avi,

You can also try www.VoiceNetwork.ca  we support FreeSwitch running on a 64
Bit OpenVZ system.    We will also offer 30% off our posted pricing for VPS
running FreeSwitch.

We can also offer Canada/USA/UK DID numbers at great prices starting at
$0.99 per DID.

Norman Tomlins
Voice Network Inc.



On Tue, Dec 21, 2010 at 4:03 AM, Avi Marcus <Avi at amarcus.com> wrote:

> I'm running less than 5 concurrent calls for the next month at least
> and slowly gaining volume, so I don't need anything fancy.
> The wiki seems to say that Xen virtualization works with FS.
>
> Any recommendations for VPS hosting? I know you get what you pay for,
> but I'm sure there's inexpensive that's not bad.
>
> Linode has a very slick website and feature list, 64bit hosting, and
> 512mb/200gb bandwidth for $20/month.
> Any VPS for voip suggestions?
> Thanks!
>
> -Avi Marcus
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Dec 21 20:33:34 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 21 Dec 2010 09:33:34 -0800
Subject: [Freeswitch-users] forcing codec on leg b
In-Reply-To: <AANLkTi=qBM6qdQgktyP01+y9=JZxoWe1ELWxLU55oPqw@mail.gmail.com>
References: <AANLkTi=V4e5jeZEB4VsafKtNU3BbV7p1qGQDPQPOkbbc@mail.gmail.com>
	<55EEC03D-A5F5-4669-9452-888A6BAEBAD9@ipeva.fr>
	<AANLkTinuDSy=iR+ccWAcz1im7jbYM+qzq77kSCYub0Y4@mail.gmail.com>
	<5B3F4C11-7AED-4F34-BE27-E4A2EFEE5EC4@ipeva.fr>
	<AANLkTinsojjgQHsTUSYVx7O_oM=-YSaEtUmPuZrDS2nx@mail.gmail.com>
	<AANLkTimrUPoucPZsaSBvgq6s2_-vnyKHYCE+qWVE=x_E@mail.gmail.com>
	<AANLkTi=qBM6qdQgktyP01+y9=JZxoWe1ELWxLU55oPqw@mail.gmail.com>
Message-ID: <AANLkTimdJpWZXVXqBw1gvWcwb3ULrBfU1KbSMZv30uUj@mail.gmail.com>

No, you cannot do this in SIP profiles. The whole point of xml_curl is to
allow you to make these kinds of decisions outside of FreeSWITCH using the
dialplan binding. When using xml_curl you are not *forced* into using a
dynamic dialplan. You can always fall back to the static XML. You would
still need to create a dialplan extension that matches based on the criteria
that are important to you. If I understand correctly you want to route the
call based upon the source IP address. If that's the case then you could
have a few extensions in the public context like this:

<extension name="sample route, PCMA">
  <condition field="network_address" expression="^10\.20\.30\.40$"/>
  <condition field="destination_number" expression="^(\d+)$">
    <action application="bridge" data="{absolute_codec_string=PCMA}
sofia/gateway/your_gateway/$1"/>
  </condition>
</extension>

<extension name="sample route, GSM">
  <condition field="network_address" expression="^11\.22\.33\.44$"/>
  <condition field="destination_number" expression="^(\d+)$">
    <action application="bridge" data="{absolute_codec_string=GSM}
sofia/gateway/your_other_gateway/$1"/>
  </condition>
</extension>

To keep the dialplan simple I would use one extension per IP address.

Hope this helps,
-MC

On Tue, Dec 21, 2010 at 2:54 AM, Sameer Khan <sameer2k3t at gmail.com> wrote:

> i know that but i want such things from sip profiles,
>
> i dont want to pass this value from dialplan as my dialplan is binded in
> xml curl
>
> can i set this in sip settings? different codecs for different ips
>
>
> On Tue, Dec 21, 2010 at 2:54 PM, Sam <u2nsam at gmail.com> wrote:
>
>> Hi,
>>
>> You can use this,
>>
>> absolute_codec_string=PCMA
>>
>> http://wiki.freeswitch.org/wiki/Codec_negotiation#Early_Negotiation_parameters
>>
>> Regds
>> Sam
>>
>>
>>
>>
>> On Tue, Dec 21, 2010 at 2:28 PM, Sameer Khan <sameer2k3t at gmail.com>wrote:
>>
>>> in asterisk i m doing it like
>>>
>>> i call perl file via agi and in that perl file i check in  the database
>>> what should be the forwarding address for the incoming number
>>> lets say call arrives at my box for 12121212 then a perl file is called
>>> which checks in the database for the destination address and lets say the
>>> destination address is xyz at 1.1.1.1
>>>
>>> this all is achieved by xml curl
>>>
>>> now in asterisk i add these lines in sip.conf
>>> [1.1.1.1]
>>> host=1.1.1.1
>>> type=peer
>>> disallow=all
>>> allow=alaw
>>>
>>> so when asterisk find that ip in sip.conf it force the codec on leg b to
>>> alaw.
>>>
>>>
>>> this i want to achieve in fs
>>>
>>>
>>> On Tue, Dec 21, 2010 at 1:40 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>>
>>>> sorry, I dont understand what you want to achieve
>>>>
>>>>  David Ponzone  Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>
>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>>> l'exp?diteur.*
>>>> *
>>>> *
>>>>
>>>>
>>>>
>>>> Le 21/12/2010 ? 09:32, Sameer Khan a ?crit :
>>>>
>>>> thanks for your reply david,
>>>>
>>>> but for that i will need to call absolute_Codec_string value from
>>>> datbase which i dont want to do now.
>>>>
>>>> as i have one data base and running two asterisk on them with this fs
>>>>
>>>> cant i define in sip users or profile ,
>>>>
>>>> i saw an option there in brian.xml for cidr but all examples are
>>>> username dependent
>>>>
>>>> i dont want user authentication as i am just forwarding the calls to my
>>>> user
>>>>
>>>> i just want to check if ip is 1.1.1.1 it uses pcma for all
>>>> numbers/extension going to the ip like this XXX at 1.1.1.1
>>>>
>>>>
>>>> On Tue, Dec 21, 2010 at 1:22 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>>>
>>>>> in the dialplan, use a condition to match network_addr and then set
>>>>> absolute_codec_string to PCMA.
>>>>>
>>>>>  David Ponzone  Direction Technique
>>>>> email: david.ponzone at ipeva.fr
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>>
>>>>> Service Client IPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>
>>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>>>> l'exp?diteur.*
>>>>> *
>>>>> *
>>>>>
>>>>>
>>>>>
>>>>> Le 21/12/2010 ? 09:18, Sameer Khan a ?crit :
>>>>>
>>>>> Hello guyz,
>>>>> Need your help
>>>>>
>>>>> I want my fs to behave like this
>>>>>
>>>>> when call arrives for a number 1234656789 at fs i call a php file in
>>>>> which i check in the database that what should be the forwarding address of
>>>>> this number. for example the forwarding address is 123456789 at 1.1.1.1then i return that forwarding address in bridge application in the end
>>>>> i want to define some setting in sip profiles, user directory or
>>>>> wherever so when ever ip is 1.1.1.1 it set the codec on leg b to pcma
>>>>> like it is defined in asterisk as
>>>>> [1.1.1.1]
>>>>> host=1.1.1.1
>>>>> type=peer
>>>>> disallow=all
>>>>> allow=alaw
>>>>>
>>>>>
>>>>> i want my sip profile to check if network address is 1.1.1.1, it uses a
>>>>> particular codec lets say pcma but if network address is 2.2.2.2 it uses the
>>>>> defult
>>>>> and this i want to define for around 40 ip addresses
>>>>>  _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Dec 21 20:35:42 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 21 Dec 2010 09:35:42 -0800
Subject: [Freeswitch-users] Lua not playing any wav files
In-Reply-To: <AANLkTimZwk-R5VQDVQEpypKchrKBaQ5zr-ZtVcnrHwEG@mail.gmail.com>
References: <AANLkTimHrCzzzg8S=PVTVDmH1sd4iv4jQiMFbFHR7teo@mail.gmail.com>
	<AANLkTik3MPkku8-Zk-JKRezUs=tMmrrG=xmCvZPcQ3wt@mail.gmail.com>
	<AANLkTimXMWewnFnnMuNAeuZySy4Hcb_d2vopHpmPVo+9@mail.gmail.com>
	<AANLkTimZwk-R5VQDVQEpypKchrKBaQ5zr-ZtVcnrHwEG@mail.gmail.com>
Message-ID: <AANLkTi=yTjLtoNcfXZPSEGO45qvppw_gOkEV46Af-B8w@mail.gmail.com>

Hahaha! Yes, that would make sense. Executing inline would definitely fail
spectacularly! Glad you figured it out.
-MC

On Tue, Dec 21, 2010 at 7:17 AM, Marc de Corny <marcdecorny at gmail.com>wrote:

> RESOLVED :
>
> for anyone with the same problem, I have just foudn the solution.
>
> My action was being executed inline
> <action inline="true" application="lua" data="ivr_mysql.lua
> ${destination_number:3}"/>
>
> This does not allow you to playback anything.
>
> Thanks
> Marc
>
> On Tue, Dec 21, 2010 at 2:51 PM, Marc de Corny <marcdecorny at gmail.com>wrote:
>
>> Hi Chris
>>
>> That seemed to work I got audio in both scnearios, in the XML and LUA
>> script.
>>
>> so the only difference between my originial issue and the successful moh
>> test. is that I was trying to playback a file and here we are playing a
>> local stream://moh.
>>
>> I have changed all the files too chmod 777, is there any thing else on the
>> rights point of view that I need to know in order to get it to work?
>>
>> I am grateful for any ideas or anything else that could help me understand
>> what the issue is
>>
>> thanks
>> Marc
>>
>>
>>
>>
>> On Sat, Dec 18, 2010 at 8:42 AM, Chris Burns <chris at cloudtel.com> wrote:
>>
>>> Hmmm no help for you yet huh ... you may have solved it on your own
>>> already, but ...
>>>
>>> You want to answer the call there, and not pre-answer. Pre-answer is for
>>> early media, which is for exchanging media before committing to answer the
>>> call. Admittedly you should hear something either way, but you definitely
>>> want to answer the call in your case. If your XML dialplan works as you
>>> said, you should compare the log output between these 2 extensions:
>>>
>>> <extension name="testXML">
>>>   <condition field="destination_number" expression="^9001$">
>>>     <action application="answer"/>
>>>     <action application="playback" data="local_stream://moh"/>
>>>   </condition>
>>> </extension>
>>>
>>> <extension name="testLUA">
>>>   <condition field="destination_number" expression="^9002$">
>>>     <action application="lua" data="test.lua"/>
>>>   </condition>
>>> </extension>
>>>
>>> Contents of test.lua:
>>> session:answer();
>>> session:execute("playback","local_stream://moh");
>>>
>>>
>>>   On Wed, Dec 15, 2010 at 1:59 AM, Marc de Corny <marcdecorny at gmail.com>wrote:
>>>
>>>>  Hi all,
>>>>
>>>> I have run into an issue on something so basic that I must be as simple
>>>> as enabling a feature somewhere.
>>>>
>>>> I have been trying to get lua to play a message from a WAV file. I have
>>>> tried session:execute("playback", main_msg) and
>>>> session:streamFile(ivr_invalid_msg) but neither of them play any music to
>>>> the caller. I tried both to answer and preAnswer the call first but it made
>>>> no difference. However if I put the same file into the XML dialplan and play
>>>> it with the commands below I hear the music fine.
>>>> <action application="set" data="playback_terminators=#"/>
>>>> <action application="playback" data="${401_sound_file_path}"/>
>>>>
>>>> The issue only seems to be from lua when playing any type of wav file
>>>> and those files are definitelly there as can be read by the XML
>>>>
>>>> The error message is below for the execute(playback) command, but
>>>> nothing can be seen for the
>>>> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827
>>>> Application playback Requires media! pre_answering channel
>>>> sofia/external/2031701665 at 194.0.147.16:5060 EXECUTE
>>>> sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswitch/sounds/svc_sound_files/default_autoattendant.wav)
>>>> But there is no mention of the streamFile command. I have had similar
>>>> issue with the PlayAndGetDigits command.
>>>> Is there something that I need to enable in lua so that is can playback
>>>> messages to the caller.
>>>>
>>>> Many thanks to anyone who can help.
>>>> Marc
>>>>
>>>>
>>>> below is the XML dialplan and lua script as well as the log at the very
>>>> end.
>>>>
>>>> XML DIALPLAN:
>>>> <include>
>>>> <extension name="IVR_FROM_MYSQL">
>>>> <condition field="caller_id_number" expression="Anonymous"
>>>> break="never">
>>>> <action application="set" data="effective_caller_id_number=0000000000"/>
>>>> </condition>
>>>> <condition field="destination_number" expression="^(404)">
>>>> <action inline="true" application="lua" data="ivr_mysql.lua
>>>> ${destination_number:3}"/>
>>>> <action application="set" data="effective_caller_id_name=${404_tag}"/>
>>>> <action application="bridge" data="${404_dial}"/>
>>>> </condition>
>>>> </extension>
>>>> </include>
>>>>
>>>> The LUA script ivr_mysql.lua is callsed and this is it.
>>>> -- IVR : PLAY IVR WAV FILES
>>>> -- Global Variables:
>>>> local dialstr_prefix = "sofia/gateway/CS2k/"
>>>> local dialstr_main = ""
>>>> local breakoutcode = "184"
>>>> local sound_file_folder =
>>>> "/usr/local/freeswitch/sounds/svc_sound_files/"
>>>> local ddi = argv[1]
>>>> -- answer the call
>>>> session:preAnswer();
>>>> freeswitch.consoleLog("info", "All Answered\n");
>>>> ivr_invalid_msg = sound_file_folder .. "invalid_msg.wav"
>>>> main_msg = sound_file_folder .. "default_autoattendant.wav"
>>>> -- Play with Execute
>>>> session:execute("playback", main_msg)
>>>> -- Play with StreamFile
>>>> session:streamFile(ivr_invalid_msg);
>>>> dialstr_main  = dialstr_main .. dialstr_prefix .. breakoutcode ..
>>>> "02031701665"
>>>> session:setVariable("404_dial",dialstr_main)
>>>> session:setVariable("404_tag","IVR")
>>>>
>>>>
>>>> RELEVANT LOGS :
>>>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Regex (PASS)
>>>> [IVR_FROM_MYS                          QL] destination_number(4042031956241)
>>>> =~ /^(404)/ break=on-false
>>>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
>>>> lua(ivr_mysql.lua $                          {destination_number:3}) INLINE
>>>> EXECUTE sofia/external/2031701665 at 194.0.147.16:5060 lua(ivr_mysql.lua
>>>> 2031956241                          )
>>>> 2010-12-01 11:56:01.525426 [INFO] switch_cpp.cpp:584 Sending early media
>>>> 2010-12-01 11:56:01.525426 [DEBUG] sofia_glue.c:2972 AUDIO RTP
>>>> [sofia/external/2                          031701665 at 194.0.147.16:5060]
>>>> 10.5.2.105 port 29900 -> 194.0.147.164 port 50202 c
>>>> odec: 8 ms: 20
>>>> 2010-12-01 11:56:01.525426 [DEBUG] switch_rtp.c:1418 Starting timer
>>>> [soft] 160 b                          ytes per 20ms
>>>> 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3190 Set 2833 dtmf send
>>>> payload                           to 101
>>>> 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3195 Set 2833 dtmf
>>>> receive paylo                          ad to 101
>>>> 2010-12-01 11:56:01.532280 [DEBUG] mod_sofia.c:2172 Ring SDP:
>>>> v=0
>>>> o=FreeSWITCH 1291174661 1291174662 IN IP4 10.5.2.105
>>>> s=FreeSWITCH
>>>> c=IN IP4 10.5.2.105
>>>> t=0 0
>>>> m=audio 29900 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=silenceSupp:off - - - -
>>>> a=ptime:20
>>>> a=sendrecv
>>>> 2010-12-01 11:56:01.532280 [NOTICE] mod_sofia.c:2175 Pre-Answer
>>>> sofia/external/2                          031701665 at 194.0.147.16:5060!
>>>> 2010-12-01 11:56:01.532280 [DEBUG] switch_channel.c:2544
>>>> (sofia/external/2031701                          665 at 194.0.147.16:5060)
>>>> Callstate Change RINGING -> EARLY
>>>> 2010-12-01 11:56:01.534727 [DEBUG] sofia.c:4576 Channel
>>>> sofia/external/203170166                          5 at 194.0.147.16:5060skipping state [early][183]
>>>> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:676 Send signal
>>>> sofia/e                          xternal/2031701665 at 194.0.147.16:5060[BREAK]
>>>> 2010-12-01 11:56:01.534727 [INFO] switch_cpp.cpp:1181 All Answered
>>>> 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827
>>>> Application playba                          ck Requires media! pre_answering
>>>> channel sofia/external/2031701665 at 194.0.147.16:
>>>> 5060
>>>> EXECUTE sofia/external/2031701665 at 194.0.147.16:5060playback(/usr/local/freeswit
>>>> ch/sounds/svc_sound_files/default_autoattendant.wav)
>>>> 2010-12-01 11:56:01.537644 [DEBUG] switch_cpp.cpp:972
>>>> sofia/external/2031701665@                          194.0.147.16:5060destroy/unlink session from object
>>>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
>>>> set(effective_calle                          r_id_name=${404_tag})
>>>> Dialplan: sofia/external/2031701665 at 194.0.147.16:5060 Action
>>>> bridge(${404_dial})
>>>> 2010-12-01 11:56:01.537644 [DEBUG] switch_core_state_machine.c:119
>>>> (sofia/extern                          al/2031701665 at 194.0.147.16:5060)
>>>> State Change CS_ROUTING -> CS_EXECUTE
>>>> 2010-12-01 11:56:01.537644 [DEBUG] switch_core_session.c:1057 Send
>>>> signal sofia/
>>>> external/2031701665 at 194.0.147.16:5060 [BREAK]
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Dec 21 20:46:22 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 21 Dec 2010 09:46:22 -0800
Subject: [Freeswitch-users] CallCenter Agent UUID-Standby
In-Reply-To: <AANLkTi=7aCKRR0m3UWiYdF17kOA+wqNUtdEY3FXxm2Ng@mail.gmail.com>
References: <AANLkTikNKi7Rzo8tKdMq+=iG7MaO304Q2CD15e6CJTs9@mail.gmail.com>
	<E9D43200-E2B4-4272-86AD-4D642E3165ED@tlainvestments.com>
	<AANLkTin88ZknYRanO4sY2S56tPM2w7ZRTFKxfz0V4fE6@mail.gmail.com>
	<AANLkTi=7aCKRR0m3UWiYdF17kOA+wqNUtdEY3FXxm2Ng@mail.gmail.com>
Message-ID: <AANLkTimjNi8dPtZHw6t0GH8TVa-xR2LmXxn32xEyX=ON@mail.gmail.com>

You should probably track down Moc and pick his brain since he actually
wrote mod_callcenter...

-MC

On Mon, Dec 20, 2010 at 11:44 PM, Antonio Teixeira
<eagle.antonio at gmail.com>wrote:

> Hello guys
>
> Sorry to bug you again , but I'm still unable to find out a solution for
> this , should i send a bug report or is the bug sited on the chair :)
>
> P.S Have a Merry Christmas to all :)
>
> Ant?nio Teixeira
>
>
> 2010/12/17 Antonio Teixeira <eagle.antonio at gmail.com>
>
>> Hello Troy ,
>>
>> But that part is present in the dial plan and as soon as a caller hangups
>> an hangup is also send to the agent and it doesn't seem to recover from
>> that.
>>
>> Full Debug Log And Dialplan in paste bin
>> http://pastebin.com/ADFfLPtR
>>
>> Thank you all for all your time
>> Antonio
>>
>>
>> 2010/12/17 Troy Anderson <freeswitch at tlainvestments.com>
>>
>> Hi Ant?nio,
>>>
>>> Take a look at this part of the wiki:
>>> http://wiki.freeswitch.org/wiki/Mod_callcenter#Callback
>>>
>>> Note the last line of the sample dial plan:
>>>
>>>     <action application="transfer" data="4099"/> <!-- Remove this if you just want to get a single call -->
>>>
>>> If you want to throw them back in the loop, you have to explicitly do it
>>> in your dial plan.
>>>
>>> -Troy
>>>
>>> On Dec 16, 2010, at 10:30 AM, Antonio Teixeira wrote:
>>>
>>> Good Afternoon.
>>>
>>> I'm leaving Asterisk ( yayyyyyyyyyyyyy) and was curious in trying out the
>>> Call Center feature,  uuid-standby , now i created a dialplan and I'm able
>>> to connect a caller to an agent.
>>> Now by my assumption as soon as the caller disconnects the agent should
>>> be trow back ( transfer) into the loop (4099 ext.) but instead i get an
>>> hangup and the sip client terminates the call.
>>>
>>> Any ideas ?
>>>
>>> Sample Dialplan
>>>
>>> <extension>
>>>   <condition field="destination_number" expression="^(4099)$">
>>>     <action application="set" data="res=${callcenter_config(agent set uuid ${caller_id_number}@${domain_name} '${uuid}')}" />
>>>
>>>
>>>
>>>
>>>
>>>     <action application="set" data="res=${callcenter_config(agent set type ${caller_id_number}@${domain_name} 'uuid-standby')}" />
>>>     <action application="set" data="res=${callcenter_config(agent set status ${caller_id_number}@${domain_name} 'Available (On Demand)')}" />
>>>
>>>
>>>
>>>
>>>
>>>     <action application="set" data="res=${callcenter_config(agent set state ${caller_id_number}@${domain_name} 'Waiting')}" />
>>>     <action application="set" data="cc_warning_tone=tone_stream://%(200,0,500,600,700)"/>
>>>
>>>
>>>
>>>
>>>
>>>     <action application="answer" />
>>>     <action application="playback" data="$${hold_music}"/>
>>>     <action application="transfer" data="4099"/> <!-- Remove this if you just want to get a single call -->
>>>
>>>
>>>
>>>
>>>
>>>   </condition>
>>> </extension>
>>>
>>>
>>> Thanks For Your Time
>>> Ant?nio Teixeira
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From freeswitch-list at puzzled.xs4all.nl  Tue Dec 21 20:49:25 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Tue, 21 Dec 2010 18:49:25 +0100
Subject: [Freeswitch-users] Support for RTCP report via SIP PUBLISH
In-Reply-To: <4D10BB08.1050205@ewetel.de>
References: <4D10BB08.1050205@ewetel.de>
Message-ID: <4D10E8A5.6060403@puzzled.xs4all.nl>

On 12/21/2010 03:34 PM, Helmut Kuper wrote:
> Hi,
> 
> I want to collect RTCP reports from snom devices via FS's event system.
> Unfortunately it doesn't support it. Maybe because it is still a draft:
> 
> http://tools.ietf.org/html/draft-ietf-sipping-rtcp-summary-05
> 
> I wonder whether this feature is planed in near future.
> 
> 
> 
> Nevertheless I hacked sofia.c and sofia-presence.c to get it working
> just for testing. Here is the "git diff":

Can you please open a ticket on jira.freeswitch.org and attach your
patch (diff against latest git) for review:

Create new issue -> select project FreeSWITCH -> select mod_sofia
Then provide information and your patch.

Thanks and regards,
Patrick


From msc at freeswitch.org  Tue Dec 21 20:49:31 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 21 Dec 2010 09:49:31 -0800
Subject: [Freeswitch-users] Group calls & voicemail
In-Reply-To: <752EEB96-74F9-40D1-AEEB-296B3B7F016C@maui-systems.co.uk>
References: <752EEB96-74F9-40D1-AEEB-296B3B7F016C@maui-systems.co.uk>
Message-ID: <AANLkTi=5D9-dDgmcUK61K5mwJvVSmbzgj_ih9LPBcFy9@mail.gmail.com>

Marty,

Try putting the ignore_early_media in the actual dialstring:

<action application="bridge"
data="{ignore_early_media=true}${group_call(maui_group@${domain_name}+F)}"/>

If that doesn't work then pastebin the debug output so we can have a look at
what's happening.
-MC

On Tue, Dec 21, 2010 at 2:50 AM, Marty Lee <marty at maui-systems.co.uk> wrote:

>
> Trying to get my head around the dialplan in FreeSwitch and can't
> find a good example of how to do call a group of extensions, but
> if there is no answer, leave it in a common voicemail.
>
> Think of an incoming line to a small business; it rings all the
> extensions and if nobody answers, the message is left in an 'office'
> voicemail, rather than an individual.
>
> I've got an entry in the public dialplan that maps the incoming
> number to the extension 'office'; what I currently have for the
> 'office' extension is below and is a blatant frankenstein job on
> the Local-extensions entry, but doesn't work.
>
> I'll keep looking, but if anyone has any ideas, then feel free to
> point me in the right direction.
>
> m
>
>
>
> <extension name="office">
>    <condition field="destination_number" expression="^office$">
>      <action application="set" data="dialed_extension=$1"/>
>      <action application="export" data="dialed_extension=$1"/>
>      <action application="set" data="ringback=${uk-ring}"/>
>      <action application="set" data="call_timeout=10"/>
>      <action application="set" data="hangup_after_bridge=true"/>
>      <action application="set" data="ignore_early_media=true"/>
>      <action application="set" data="continue_on_fail=true"/>
>      <action application="set"
> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
> var callgroup)}"/>
>
>      <action application="bridge" data="${group_call(maui_group@
> ${domain_name}+F)}"/>
>      <action application="answer"/>
>      <action application="sleep" data="10000"/>
>      <action application="bridge" data="loopback/app=voicemail:default
> ${domain_name} 1000"/>
>    </condition>
>  </extension>
>
> Notes:
>
> 'maui_group' is defined as two test extensions; '1000' is the destination
> voicemail account I want it to go to if there is no answer or the
> handsets are offline.
>
>
>
>
> -----
> Marty Lee                         e: marty at maui-systems.co.uk
> Technical Director                v: +44 845 869 2661
> Maui Systems Ltd                  f: +44 871 433 8922
> Scotland, UK                      w: http://www.maui-systems.co.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rupa at rupa.com  Tue Dec 21 21:03:54 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 21 Dec 2010 12:03:54 -0600
Subject: [Freeswitch-users] forcing codec on leg b
In-Reply-To: <AANLkTinsojjgQHsTUSYVx7O_oM=-YSaEtUmPuZrDS2nx@mail.gmail.com>
References: <AANLkTi=V4e5jeZEB4VsafKtNU3BbV7p1qGQDPQPOkbbc@mail.gmail.com>
	<55EEC03D-A5F5-4669-9452-888A6BAEBAD9@ipeva.fr>
	<AANLkTinuDSy=iR+ccWAcz1im7jbYM+qzq77kSCYub0Y4@mail.gmail.com>
	<5B3F4C11-7AED-4F34-BE27-E4A2EFEE5EC4@ipeva.fr>
	<AANLkTinsojjgQHsTUSYVx7O_oM=-YSaEtUmPuZrDS2nx@mail.gmail.com>
Message-ID: <AANLkTinWj-2NK+mhH35Dke71P=11RQz+qLR7J4fpAFe4@mail.gmail.com>

You can try looking at mod_easyroute.  It should do what you want.

On Tue, Dec 21, 2010 at 2:58 AM, Sameer Khan <sameer2k3t at gmail.com> wrote:

> in asterisk i m doing it like
>
> i call perl file via agi and in that perl file i check in  the database
> what should be the forwarding address for the incoming number
> lets say call arrives at my box for 12121212 then a perl file is called
> which checks in the database for the destination address and lets say the
> destination address is xyz at 1.1.1.1
>
> this all is achieved by xml curl
>
> now in asterisk i add these lines in sip.conf
> [1.1.1.1]
> host=1.1.1.1
> type=peer
> disallow=all
> allow=alaw
>
> so when asterisk find that ip in sip.conf it force the codec on leg b to
> alaw.
>
>
> this i want to achieve in fs
>
>
> On Tue, Dec 21, 2010 at 1:40 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> sorry, I dont understand what you want to achieve
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 21/12/2010 ? 09:32, Sameer Khan a ?crit :
>>
>> thanks for your reply david,
>>
>> but for that i will need to call absolute_Codec_string value from datbase
>> which i dont want to do now.
>>
>> as i have one data base and running two asterisk on them with this fs
>>
>> cant i define in sip users or profile ,
>>
>> i saw an option there in brian.xml for cidr but all examples are username
>> dependent
>>
>> i dont want user authentication as i am just forwarding the calls to my
>> user
>>
>> i just want to check if ip is 1.1.1.1 it uses pcma for all
>> numbers/extension going to the ip like this XXX at 1.1.1.1
>>
>>
>> On Tue, Dec 21, 2010 at 1:22 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>>
>>> in the dialplan, use a condition to match network_addr and then set
>>> absolute_codec_string to PCMA.
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 21/12/2010 ? 09:18, Sameer Khan a ?crit :
>>>
>>> Hello guyz,
>>> Need your help
>>>
>>> I want my fs to behave like this
>>>
>>> when call arrives for a number 1234656789 at fs i call a php file in
>>> which i check in the database that what should be the forwarding address of
>>> this number. for example the forwarding address is 123456789 at 1.1.1.1then i return that forwarding address in bridge application in the end
>>> i want to define some setting in sip profiles, user directory or wherever
>>> so when ever ip is 1.1.1.1 it set the codec on leg b to pcma
>>> like it is defined in asterisk as
>>> [1.1.1.1]
>>> host=1.1.1.1
>>> type=peer
>>> disallow=all
>>> allow=alaw
>>>
>>>
>>> i want my sip profile to check if network address is 1.1.1.1, it uses a
>>> particular codec lets say pcma but if network address is 2.2.2.2 it uses the
>>> defult
>>> and this i want to define for around 40 ip addresses
>>>  _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From bggoutham at gmail.com  Tue Dec 21 21:00:28 2010
From: bggoutham at gmail.com (Goutham BG)
Date: Tue, 21 Dec 2010 23:30:28 +0530
Subject: [Freeswitch-users] Query related to enabling SRTP in
	FreeSWITCH-1.0.7
In-Reply-To: <AANLkTikyGM3fN7Joi9VodrYiNf7TQ9gHWPz0iRDWBj9U@mail.gmail.com>
References: <AANLkTikyGM3fN7Joi9VodrYiNf7TQ9gHWPz0iRDWBj9U@mail.gmail.com>
Message-ID: <AANLkTi=JoWuUKrGM0qiBfVJ93nBzN7HQZOUxHEhkYi2P@mail.gmail.com>

Posting the below query to freeswitch-users list as well. Any hints will be
really helpful.

---------- Forwarded message ----------
From: Goutham BG <bggoutham at gmail.com>
Date: Mon, Dec 20, 2010 at 9:16 PM
Subject: Query related to enabling SRTP in FreeSWITCH-1.0.7
To: freeswitch-dev at lists.freeswitch.org


Hi,

I have been trying to enable SRTP in FreeSWITCH-1.0.7 and have been facing
some issues.
I have the following entry in my dialplan XML file:

*<extension name="IVR">
    <condition field="destination_number" expression="^IVR$">
      <action application="bridge" data="loopback/app=socket:
47.152.232.156:8084 async full"/>
    </condition
</extension>*

A SIP phone (Avaya 12XX) configured in "SRTP best effort" mode dials into
this extension and is connected to the IVR. But the media is established in
SRTP in one way and RTP in the other way.
The phone offers the following SDP in the INVITE message:

v=0
o=- 10170 10170 IN IP4 47.152.232.147
s=Sip Call
c=IN IP4 47.152.232.147
t=0 0
m=audio 5016 RTP/AVP 0 8 18 101 102
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:102 X-nt-inforeq/8000
a=sendrecv
m=audio 5016 RTP/SAVP 0 8 18 101 102
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:102 X-nt-inforeq/8000
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Tjivoci1I/mVkt/Fq/ZsiY+
+ornJoXjZ5tSadho4
a=sendrecv

As we can see, there are two "m=" lines in the SDP of the offer; one for RTP
and another for SRTP. FreeSWITCH-1.0.7 answers the call by sending 200OK
with the following SDP:

v=0
o=FreeSWITCH 1291628984 1291628985 IN IP4 47.152.232.156
s=FreeSWITCH
c=IN IP4 47.152.232.156
t=0 0
m=audio 11280 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
m=audio 0 RTP/SAVP 19

As you can see above, FreeSWITCH accepts the RTP stream and rejects the SRTP
stream (by sending port as 0) in the SDP. The SIP phone sends the media in
RTP(which is expected). But, FreeSWITCH sends the media in SRTP to the SIP
phone. I believe this is a bug in FreeSWITCH as it is supposed to send the
media in RTP since it accepted RTP in the answer (200OK).

*Query:
======*
In order to make FreeSWITCH select SRTP in the SDP of the answer(200OK), I
made the following change(i.e, setting sip_secure_media=true) in FS dial
plan:

*<extension name="IVR">
    <condition field="destination_number" expression="^IVR$">
      <action application="set" data="sip_secure_media=true"/>
      <action application="bridge" data="loopback/app=socket:
47.152.232.156:8084 async full"/>
    </condition
</extension>*

In FreeSWITCH-1.0.6(before updating to 1.0.7), this worked and FS accepted
the SRTP stream and rejected RTP in the answer(200 OK) as shown below:

m=audio 0 RTP/AVP 19
m=audio 12084 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:hgv7ClqDx1irTRrXq2NEm9Gbouw0969bBU3n+LcM

But after updating the FreeSWITCH-1.0.6 to 1.0.7, the above mentioned dial
plan change (i.e, setting sip_secure_media=true) is not working. It is still
behaving in the same way as it did without the XML change.

Can you please let me know if anything else needs to be added in dialplan
XML file for enabling SRTP in this case in FreeSWITCH-1.0.7 or am I missing
something here?

I have referred the following FS wiki pages for making the SRTP changes:
http://wiki.freeswitch.org/wiki/Secure_RTP
http://wiki.freeswitch.org/wiki/SRTP

Note: There is no issue when the SIP phone is configured in "SRTP only" mode
where only SRTP stream is offered in the SDP of the INVITE. In this case,
SIP phone and FreeSWITCH communicate properly using SRTP. This doesn't
require setting "sip_secure_media=true" in the dialplan XML file.
P.S: I am a newbie to FreeSWITCH. So, please forgive me if I am asking basic
questions.

Thanks
Goutham B G
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From brian at freeswitch.org  Tue Dec 21 21:17:31 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 21 Dec 2010 12:17:31 -0600
Subject: [Freeswitch-users] Query related to enabling SRTP in
	FreeSWITCH-1.0.7
In-Reply-To: <AANLkTi=JoWuUKrGM0qiBfVJ93nBzN7HQZOUxHEhkYi2P@mail.gmail.com>
References: <AANLkTikyGM3fN7Joi9VodrYiNf7TQ9gHWPz0iRDWBj9U@mail.gmail.com>
	<AANLkTi=JoWuUKrGM0qiBfVJ93nBzN7HQZOUxHEhkYi2P@mail.gmail.com>
Message-ID: <D95EC7E6-EA17-4B5B-B12A-83A19A1D18E5@freeswitch.org>

And clearly you overlooked my response... are you setting sip_secure_media=true after the call is answered in your dialplan?\

I need to see the full debug log of a call coming in to FreeSWITCH please on our pastebin.

/b

On Dec 21, 2010, at 12:00 PM, Goutham BG wrote:

> Posting the below query to freeswitch-users list as well. Any hints will be really helpful.
> 
> ---------- Forwarded message ----------
> From: Goutham BG <bggoutham at gmail.com>
> Date: Mon, Dec 20, 2010 at 9:16 PM
> Subject: Query related to enabling SRTP in FreeSWITCH-1.0.7
> To: freeswitch-dev at lists.freeswitch.org
> 
> 
> Hi,
> 
> I have been trying to enable SRTP in FreeSWITCH-1.0.7 and have been facing some issues.
> I have the following entry in my dialplan XML file:
> 
> <extension name="IVR">
>     <condition field="destination_number" expression="^IVR$">
>       <action application="bridge" data="loopback/app=socket:47.152.232.156:8084 async full"/>
>     </condition
> </extension>
> 
> A SIP phone (Avaya 12XX) configured in "SRTP best effort" mode dials into this extension and is connected to the IVR. But the media is established in SRTP in one way and RTP in the other way.
> The phone offers the following SDP in the INVITE message:
> 
> v=0
> o=- 10170 10170 IN IP4 47.152.232.147
> s=Sip Call
> c=IN IP4 47.152.232.147
> t=0 0
> m=audio 5016 RTP/AVP 0 8 18 101 102
> a=rtpmap:0 PCMU/8000
> a=ptime:20
> a=rtpmap:8 PCMA/8000
> a=ptime:20
> a=rtpmap:18 G729/8000
> a=ptime:20
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=rtpmap:102 X-nt-inforeq/8000
> a=sendrecv
> m=audio 5016 RTP/SAVP 0 8 18 101 102
> a=rtpmap:0 PCMU/8000
> a=ptime:20
> a=rtpmap:8 PCMA/8000
> a=ptime:20
> a=rtpmap:18 G729/8000
> a=ptime:20
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=rtpmap:102 X-nt-inforeq/8000
> a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Tjivoci1I/mVkt/Fq/ZsiY+
> +ornJoXjZ5tSadho4
> a=sendrecv
> 
> As we can see, there are two "m=" lines in the SDP of the offer; one for RTP and another for SRTP. FreeSWITCH-1.0.7 answers the call by sending 200OK with the following SDP:
> 
> v=0
> o=FreeSWITCH 1291628984 1291628985 IN IP4 47.152.232.156
> s=FreeSWITCH
> c=IN IP4 47.152.232.156
> t=0 0
> m=audio 11280 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> m=audio 0 RTP/SAVP 19
> 
> As you can see above, FreeSWITCH accepts the RTP stream and rejects the SRTP stream (by sending port as 0) in the SDP. The SIP phone sends the media in RTP(which is expected). But, FreeSWITCH sends the media in SRTP to the SIP phone. I believe this is a bug in FreeSWITCH as it is supposed to send the media in RTP since it accepted RTP in the answer (200OK).
> 
> Query:
> ======
> In order to make FreeSWITCH select SRTP in the SDP of the answer(200OK), I made the following change(i.e, setting sip_secure_media=true) in FS dial plan:
> 
> <extension name="IVR">
>     <condition field="destination_number" expression="^IVR$">
>       <action application="set" data="sip_secure_media=true"/>
>       <action application="bridge" data="loopback/app=socket:47.152.232.156:8084 async full"/>
>     </condition
> </extension>
> 
> In FreeSWITCH-1.0.6(before updating to 1.0.7), this worked and FS accepted the SRTP stream and rejected RTP in the answer(200 OK) as shown below:
> 
> m=audio 0 RTP/AVP 19
> m=audio 12084 RTP/SAVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:hgv7ClqDx1irTRrXq2NEm9Gbouw0969bBU3n+LcM
> 
> But after updating the FreeSWITCH-1.0.6 to 1.0.7, the above mentioned dial plan change (i.e, setting sip_secure_media=true) is not working. It is still behaving in the same way as it did without the XML change.
> 
> Can you please let me know if anything else needs to be added in dialplan XML file for enabling SRTP in this case in FreeSWITCH-1.0.7 or am I missing something here?
> 
> I have referred the following FS wiki pages for making the SRTP changes:
> http://wiki.freeswitch.org/wiki/Secure_RTP
> http://wiki.freeswitch.org/wiki/SRTP
> 
> Note: There is no issue when the SIP phone is configured in "SRTP only" mode where only SRTP stream is offered in the SDP of the INVITE. In this case, SIP phone and FreeSWITCH communicate properly using SRTP. This doesn't require setting "sip_secure_media=true" in the dialplan XML file.
> P.S: I am a newbie to FreeSWITCH. So, please forgive me if I am asking basic questions.
> 
> Thanks
> Goutham B G
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From rajil.s at gmail.com  Tue Dec 21 22:50:55 2010
From: rajil.s at gmail.com (Rajil Saraswat)
Date: Tue, 21 Dec 2010 19:50:55 +0000
Subject: [Freeswitch-users] Nokia N900 and poor Voicemail IVR audio
In-Reply-To: <AANLkTi=2p_Lf15g2wWQWSSbn_5VV4MLijN4z-E3S-MiW@mail.gmail.com>
References: <4d0fdd1f.017b0e0a.437e.ffffa766@mx.google.com>
	<AANLkTi=2p_Lf15g2wWQWSSbn_5VV4MLijN4z-E3S-MiW@mail.gmail.com>
Message-ID: <AANLkTinSVAJRovr+uW-57whqhr82FvUjBUJjguLJuGWi@mail.gmail.com>

I captured the voicemail call log which is here
http://pastebin.freeswitch.org/14839. Also, i recorded the ivr audio
which is at http://174.132.148.146/ivr.wav

Thanks

On 20 December 2010 22:56, Michael Collins <msc at freeswitch.org> wrote:
> This is an interesting symptom. Please get a console debug of the call. Use
> this page for guidance on how to collect the information and put in into
> pastebin:
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
> -MC
>
> On Mon, Dec 20, 2010 at 2:48 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:
>>
>> Hi all,
>> For some strange reason my Nokia N900 doesn't play the audio for the
>> digits of the IVR. For example it will play "For advanced options Press" but
>> will not play number 5. The IVR audio works fine if i call using ekiga.
>> Is there explanation for this? Initially i thought this was my FS server
>> issue but i have now tested it against somebody elses FS with the same
>> response.
>> Thanks
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From hwnorman at hotmail.com  Wed Dec 22 05:11:24 2010
From: hwnorman at hotmail.com (Norman Lam)
Date: Wed, 22 Dec 2010 10:11:24 +0800
Subject: [Freeswitch-users] compile error on libmp3lame and mod_shout
In-Reply-To: <COL117-W475362B9E13477B3355945B91A0@phx.gbl>
References: <SNT109-DS118A0D1D7049B8F876EE9C11A0@phx.gbl>
	<COL117-W475362B9E13477B3355945B91A0@phx.gbl>
Message-ID: <SNT109-DS1178EDDED9B267F1A7DC2BC11B0@phx.gbl>

Jeff 

 

There is no download error , there is no lame3.97.tar.gz, but there is
mpg123.tar.gz in /libs folder

 

Please advise 

 

Thanks in advance

 

Norman 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
Lenk
Sent: Tuesday, December 21, 2010 11:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] compile error on libmp3lame and mod_shout

 

Norman,
 
Look in the log when you are first building - do you have any download
errors.
 
Also are these files present in fs folder\libs
 
lame-3.97.tar.gz 
mpg123.tar.gz
 
if not you had a download error when first building.
 

  _____  

From: hwnorman at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 21 Dec 2010 16:42:47 +0800
Subject: [Freeswitch-users] compile error on libmp3lame and mod_shout

Hi there 

 

I am using AMD athlon X2, windows xp sp3, VC++2008 express (only installed
this software) using the Git source downloaded today the first time 

 

I am getting these 2 error ,  is there any ways to resolve this or  work
around.

 

Please advise 

 

Norman Lam 

 

 

---rest of message removed

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

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From curriegrad2004 at gmail.com  Wed Dec 22 09:18:48 2010
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 21 Dec 2010 22:18:48 -0800
Subject: [Freeswitch-users] compile error on libmp3lame and mod_shout
In-Reply-To: <SNT109-DS1178EDDED9B267F1A7DC2BC11B0@phx.gbl>
References: <SNT109-DS118A0D1D7049B8F876EE9C11A0@phx.gbl>
	<COL117-W475362B9E13477B3355945B91A0@phx.gbl>
	<SNT109-DS1178EDDED9B267F1A7DC2BC11B0@phx.gbl>
Message-ID: <AANLkTimh6HAGafh1iSOuAmBkuE7=fyFknraBcwO8HiJM@mail.gmail.com>

FreeSwitch can definitely function without mod_shout unless you rely
on this module to get your MOH from an icecast/shoutcast source or
need to stream some conference calls to a nicecast/shoutcast server

On Tue, Dec 21, 2010 at 6:11 PM, Norman Lam <hwnorman at hotmail.com> wrote:
> Jeff
>
>
>
> There is no download error , there is no lame3.97.tar.gz, but there is
> mpg123.tar.gz in /libs folder
>
>
>
> Please advise
>
>
>
> Thanks in advance
>
>
>
> Norman
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff
> Lenk
> Sent: Tuesday, December 21, 2010 11:29 PM
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] compile error on libmp3lame and mod_shout
>
>
>
> Norman,
>
> Look in the log when you are first building?- do you have any download
> errors.
>
> Also?are these files present in fs folder\libs
>
> lame-3.97.tar.gz
> mpg123.tar.gz
>
> if not you had a download error when first building.
>
>
> ________________________________
>
> From: hwnorman at hotmail.com
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 21 Dec 2010 16:42:47 +0800
> Subject: [Freeswitch-users] compile error on libmp3lame and mod_shout
>
> Hi there
>
>
>
> I am using AMD athlon X2, windows xp sp3, VC++2008 express (only installed
> this software) using the Git source downloaded today the first time
>
>
>
> I am getting these 2 error , ?is there any ways to resolve this or ?work
> around.
>
>
>
> Please advise
>
>
>
> Norman Lam
>
>
>
>
>
> ---rest of message removed
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From helmut.kuper at ewetel.de  Wed Dec 22 10:41:51 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 22 Dec 2010 08:41:51 +0100
Subject: [Freeswitch-users] Support for RTCP report via SIP PUBLISH
In-Reply-To: <4D10E8A5.6060403@puzzled.xs4all.nl>
References: <4D10BB08.1050205@ewetel.de> <4D10E8A5.6060403@puzzled.xs4all.nl>
Message-ID: <4D11ABBF.50400@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Patrick,

ok done.

Regards
helmut

Am 21.12.2010 18:49, schrieb Patrick Lists:

> 
> Can you please open a ticket on jira.freeswitch.org and attach your
> patch (diff against latest git) for review:
> 
> Create new issue -> select project FreeSWITCH -> select mod_sofia
> Then provide information and your patch.
> 
> Thanks and regards,
> Patrick
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.10 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

iEYEARECAAYFAk0Rq78ACgkQ4tZeNddg3dxSgQCeNu++2P7JdO8rpzUhwktR8YPV
UmEAnji9I33lMGlvLwiHbY2wtVAN4Lpg
=j7TB
-----END PGP SIGNATURE-----


From freeswitch-list at puzzled.xs4all.nl  Wed Dec 22 11:49:47 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Wed, 22 Dec 2010 09:49:47 +0100
Subject: [Freeswitch-users] Support for RTCP report via SIP PUBLISH
In-Reply-To: <4D11ABBF.50400@ewetel.de>
References: <4D10BB08.1050205@ewetel.de> <4D10E8A5.6060403@puzzled.xs4all.nl>
	<4D11ABBF.50400@ewetel.de>
Message-ID: <4D11BBAB.3030303@puzzled.xs4all.nl>

On 12/22/2010 08:41 AM, Helmut Kuper wrote:
> Hi Patrick,
> 
> ok done.
> 
> Regards
> helmut

Hi Helmut,

Thanks. I just noticed that there is another patch in Jira that also
relates to RTCP reports: http://jira.freeswitch.org/browse/FS-949
If it makes sense perhaps you and Guillaume can work together to merge
your patches and come up with one?

Also it's probably a good idea to ask the core developers what their
intentions are regarding RTCP and see if both your patches fit their ideas.

Regards,
Patrick


From helmut.kuper at ewetel.de  Wed Dec 22 13:17:38 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 22 Dec 2010 11:17:38 +0100
Subject: [Freeswitch-users] Support for RTCP report via SIP PUBLISH
In-Reply-To: <4D11BBAB.3030303@puzzled.xs4all.nl>
References: <4D10BB08.1050205@ewetel.de>
	<4D10E8A5.6060403@puzzled.xs4all.nl>	<4D11ABBF.50400@ewetel.de>
	<4D11BBAB.3030303@puzzled.xs4all.nl>
Message-ID: <4D11D042.5050507@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello Patrick,

> Thanks. I just noticed that there is another patch in Jira that also
> relates to RTCP reports: http://jira.freeswitch.org/browse/FS-949
> If it makes sense perhaps you and Guillaume can work together to merge
> your patches and come up with one?

I just emailed Guillaume. His patch seems to add some more informations
from periodically received RTCP reports to RECV_RTCP_MESSAGE while mine
is collecting the Vendor specific RTCP report after a call has terminated.

But basically I guess we both just want a detailed QOS report to keep
track of the network/voice quality.


> Also it's probably a good idea to ask the core developers what their
> intentions are regarding RTCP and see if both your patches fit their
ideas.

That was exactly my intention in my first email yesterday.
"I wonder whether this feature is planed in near future."

The biggest pain in my project is currently that I have nearly no tools
to debug RTP data for calls from the past. Mainly I have have rare cases
where the first 0.5 second of "Hello" is missed and also rare cases
where one RTP datastream is completely missed. Sure, this could be
caused by Network or SIP-Devices instead by FS, but back in reality I am
the one who has to deliver the first facts for starting debugging
outside of FS.


best regards
helmut
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From michal.bielicki at seventhsignal.de  Wed Dec 22 14:34:45 2010
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Wed, 22 Dec 2010 12:34:45 +0100
Subject: [Freeswitch-users] Fosdem 2011
Message-ID: <1E6DD4B8-7CD9-4676-9077-E6C7E8C5B046@seventhsignal.de>

Hi everybody,
I will be running a freeswich booth at fosdem 2011 in Brussels, Belgium (http://www.fosdem.org/). If anybody is interested to meet, participate etc, just drop me an email. I will see if we can create some flyers for the show and a nice poster for the back of the booth. Ideas, helping hands, human booth decoration or any other help are greatly appreciated.

See you in Brussels

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115
D-12059 Berlin
Voice: +49 30 60988730
Amtsgericht Berlin Charlottenburg HRA 44413 B, Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From yurazilot1 at list.ru  Wed Dec 22 11:08:30 2010
From: yurazilot1 at list.ru (ZILOT)
Date: Wed, 22 Dec 2010 11:08:30 +0300
Subject: [Freeswitch-users] 2600hz. Access to my web interface.
Message-ID: <E1PVJkM-0003gp-00.yurazilot1-list-ru@f142.mail.ru>


 Hello !! I have installed FS+GUI (2600hz) on FreeBSD
  http://www.2600hz.org/ (http://www.2600hz.org/) 

I have forgotten the password to my web interface (2600hz), but i have ssh root access. Where I can search login\password to my web interface or replace them? Please help.
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From bggoutham at gmail.com  Wed Dec 22 13:01:27 2010
From: bggoutham at gmail.com (Goutham BG)
Date: Wed, 22 Dec 2010 15:31:27 +0530
Subject: [Freeswitch-users] Query related to enabling SRTP in
	FreeSWITCH-1.0.7
In-Reply-To: <D95EC7E6-EA17-4B5B-B12A-83A19A1D18E5@freeswitch.org>
References: <AANLkTikyGM3fN7Joi9VodrYiNf7TQ9gHWPz0iRDWBj9U@mail.gmail.com>
	<AANLkTi=JoWuUKrGM0qiBfVJ93nBzN7HQZOUxHEhkYi2P@mail.gmail.com>
	<D95EC7E6-EA17-4B5B-B12A-83A19A1D18E5@freeswitch.org>
Message-ID: <AANLkTimU5PWDas2_m0Ykj_4bwC=ioUMaha22Tipri2rA@mail.gmail.com>

Thanks for the response.
I have pasted the freeswitch debug log of the call coming in to FreeSWITCH
in http://pastebin.freeswitch.org/14852 .

I think I have set sip_secure_media=true before answering the call in my
dialplan. The following is the entry for this extension in my dialplan:

  <extension name="IVR">
    <condition field="destination_number" expression="^IVR$">
*      <action application="set" data="sip_secure_media=true"/>*
      <action application="bridge" data="loopback/app=socket:
47.152.232.156:8084 async full"/>
    </condition>
  </extension>

Thanks
Goutham B G

On Tue, Dec 21, 2010 at 11:47 PM, Brian West <brian at freeswitch.org> wrote:

> And clearly you overlooked my response... are you setting
> sip_secure_media=true after the call is answered in your dialplan?\
>
> I need to see the full debug log of a call coming in to FreeSWITCH please
> on our pastebin.
>
> /b
>
> On Dec 21, 2010, at 12:00 PM, Goutham BG wrote:
>
> > Posting the below query to freeswitch-users list as well. Any hints will
> be really helpful.
> >
> > ---------- Forwarded message ----------
> > From: Goutham BG <bggoutham at gmail.com>
> > Date: Mon, Dec 20, 2010 at 9:16 PM
> > Subject: Query related to enabling SRTP in FreeSWITCH-1.0.7
> > To: freeswitch-dev at lists.freeswitch.org
> >
> >
> > Hi,
> >
> > I have been trying to enable SRTP in FreeSWITCH-1.0.7 and have been
> facing some issues.
> > I have the following entry in my dialplan XML file:
> >
> > <extension name="IVR">
> >     <condition field="destination_number" expression="^IVR$">
> >       <action application="bridge" data="loopback/app=socket:
> 47.152.232.156:8084 async full"/>
> >     </condition
> > </extension>
> >
> > A SIP phone (Avaya 12XX) configured in "SRTP best effort" mode dials into
> this extension and is connected to the IVR. But the media is established in
> SRTP in one way and RTP in the other way.
> > The phone offers the following SDP in the INVITE message:
> >
> > v=0
> > o=- 10170 10170 IN IP4 47.152.232.147
> > s=Sip Call
> > c=IN IP4 47.152.232.147
> > t=0 0
> > m=audio 5016 RTP/AVP 0 8 18 101 102
> > a=rtpmap:0 PCMU/8000
> > a=ptime:20
> > a=rtpmap:8 PCMA/8000
> > a=ptime:20
> > a=rtpmap:18 G729/8000
> > a=ptime:20
> > a=fmtp:18 annexb=no
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-15
> > a=rtpmap:102 X-nt-inforeq/8000
> > a=sendrecv
> > m=audio 5016 RTP/SAVP 0 8 18 101 102
> > a=rtpmap:0 PCMU/8000
> > a=ptime:20
> > a=rtpmap:8 PCMA/8000
> > a=ptime:20
> > a=rtpmap:18 G729/8000
> > a=ptime:20
> > a=fmtp:18 annexb=no
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-15
> > a=rtpmap:102 X-nt-inforeq/8000
> > a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Tjivoci1I/mVkt/Fq/ZsiY+
> > +ornJoXjZ5tSadho4
> > a=sendrecv
> >
> > As we can see, there are two "m=" lines in the SDP of the offer; one for
> RTP and another for SRTP. FreeSWITCH-1.0.7 answers the call by sending 200OK
> with the following SDP:
> >
> > v=0
> > o=FreeSWITCH 1291628984 1291628985 IN IP4 47.152.232.156
> > s=FreeSWITCH
> > c=IN IP4 47.152.232.156
> > t=0 0
> > m=audio 11280 RTP/AVP 0 101
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > m=audio 0 RTP/SAVP 19
> >
> > As you can see above, FreeSWITCH accepts the RTP stream and rejects the
> SRTP stream (by sending port as 0) in the SDP. The SIP phone sends the media
> in RTP(which is expected). But, FreeSWITCH sends the media in SRTP to the
> SIP phone. I believe this is a bug in FreeSWITCH as it is supposed to send
> the media in RTP since it accepted RTP in the answer (200OK).
> >
> > Query:
> > ======
> > In order to make FreeSWITCH select SRTP in the SDP of the answer(200OK),
> I made the following change(i.e, setting sip_secure_media=true) in FS dial
> plan:
> >
> > <extension name="IVR">
> >     <condition field="destination_number" expression="^IVR$">
> >       <action application="set" data="sip_secure_media=true"/>
> >       <action application="bridge" data="loopback/app=socket:
> 47.152.232.156:8084 async full"/>
> >     </condition
> > </extension>
> >
> > In FreeSWITCH-1.0.6(before updating to 1.0.7), this worked and FS
> accepted the SRTP stream and rejected RTP in the answer(200 OK) as shown
> below:
> >
> > m=audio 0 RTP/AVP 19
> > m=audio 12084 RTP/SAVP 0 101
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:hgv7ClqDx1irTRrXq2NEm9Gbouw0969bBU3n+LcM
> >
> > But after updating the FreeSWITCH-1.0.6 to 1.0.7, the above mentioned
> dial plan change (i.e, setting sip_secure_media=true) is not working. It is
> still behaving in the same way as it did without the XML change.
> >
> > Can you please let me know if anything else needs to be added in dialplan
> XML file for enabling SRTP in this case in FreeSWITCH-1.0.7 or am I missing
> something here?
> >
> > I have referred the following FS wiki pages for making the SRTP changes:
> > http://wiki.freeswitch.org/wiki/Secure_RTP
> > http://wiki.freeswitch.org/wiki/SRTP
> >
> > Note: There is no issue when the SIP phone is configured in "SRTP only"
> mode where only SRTP stream is offered in the SDP of the INVITE. In this
> case, SIP phone and FreeSWITCH communicate properly using SRTP. This doesn't
> require setting "sip_secure_media=true" in the dialplan XML file.
> > P.S: I am a newbie to FreeSWITCH. So, please forgive me if I am asking
> basic questions.
> >
> > Thanks
> > Goutham B G
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fbigliardi at siweb.it  Wed Dec 22 18:05:38 2010
From: fbigliardi at siweb.it (Fabio Bigliardi)
Date: Wed, 22 Dec 2010 16:05:38 +0100
Subject: [Freeswitch-users] How to select and play audio in a conference
Message-ID: <4D1213C2.5090709@siweb.it>

Hi all,

I would like to select audio files to play to all members of a 
conference from within the conference through the use of DTMF keypad inputs.
How can I do this?

Thank you very much.

Fabio Bigliardi





:: SYSNET TELEMATICA srl ::
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From brian at freeswitch.org  Wed Dec 22 18:09:53 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 22 Dec 2010 09:09:53 -0600
Subject: [Freeswitch-users] Query related to enabling SRTP in
	FreeSWITCH-1.0.7
In-Reply-To: <AANLkTimU5PWDas2_m0Ykj_4bwC=ioUMaha22Tipri2rA@mail.gmail.com>
References: <AANLkTikyGM3fN7Joi9VodrYiNf7TQ9gHWPz0iRDWBj9U@mail.gmail.com>
	<AANLkTi=JoWuUKrGM0qiBfVJ93nBzN7HQZOUxHEhkYi2P@mail.gmail.com>
	<D95EC7E6-EA17-4B5B-B12A-83A19A1D18E5@freeswitch.org>
	<AANLkTimU5PWDas2_m0Ykj_4bwC=ioUMaha22Tipri2rA@mail.gmail.com>
Message-ID: <419F90D2-8F89-4E71-B8AE-AEE8E5F83E37@freeswitch.org>

What device are you talking to ?

/b

On Dec 22, 2010, at 4:01 AM, Goutham BG wrote:

> Thanks for the response.
> I have pasted the freeswitch debug log of the call coming in to FreeSWITCH in http://pastebin.freeswitch.org/14852 .
> 
> I think I have set sip_secure_media=true before answering the call in my dialplan. The following is the entry for this extension in my dialplan:
> 
>   <extension name="IVR">
>     <condition field="destination_number" expression="^IVR$">
>       <action application="set" data="sip_secure_media=true"/>
>       <action application="bridge" data="loopback/app=socket:47.152.232.156:8084 async full"/>
>     </condition>
>   </extension>
> 
> Thanks
> Goutham B G
> 
> On Tue, Dec 21, 2010 at 11:47 PM, Brian West <brian at freeswitch.org> wrote:
> And clearly you overlooked my response... are you setting sip_secure_media=true after the call is answered in your dialplan?\
> 
> I need to see the full debug log of a call coming in to FreeSWITCH please on our pastebin.
> 
> /b

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From brian at freeswitch.org  Wed Dec 22 18:13:11 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 22 Dec 2010 09:13:11 -0600
Subject: [Freeswitch-users] Query related to enabling SRTP in
	FreeSWITCH-1.0.7
In-Reply-To: <AANLkTimU5PWDas2_m0Ykj_4bwC=ioUMaha22Tipri2rA@mail.gmail.com>
References: <AANLkTikyGM3fN7Joi9VodrYiNf7TQ9gHWPz0iRDWBj9U@mail.gmail.com>
	<AANLkTi=JoWuUKrGM0qiBfVJ93nBzN7HQZOUxHEhkYi2P@mail.gmail.com>
	<D95EC7E6-EA17-4B5B-B12A-83A19A1D18E5@freeswitch.org>
	<AANLkTimU5PWDas2_m0Ykj_4bwC=ioUMaha22Tipri2rA@mail.gmail.com>
Message-ID: <68416830-413A-4F58-95F2-AE8C8FAB82BC@freeswitch.org>

Just tested with my snom phone and it works fine.

2010-12-22 09:00:27.304273 [DEBUG] switch_rtp.c:1424 Starting timer [soft] 160 bytes per 20ms
2010-12-22 09:00:27.306260 [DEBUG] sofia_glue.c:3219 Set 2833 dtmf send payload to 101
2010-12-22 09:00:27.306260 [DEBUG] sofia_glue.c:3224 Set 2833 dtmf receive payload to 101
2010-12-22 09:00:27.306260 [INFO] switch_rtp.c:1253 Activating Secure RTP SEND
2010-12-22 09:00:27.306260 [DEBUG] switch_core_sqldb.c:1438 Secure Type: srtp:AES_CM_128_HMAC_SHA1_32
2010-12-22 09:00:27.306260 [INFO] switch_rtp.c:1233 Activating Secure RTP RECV
2010-12-22 09:00:27.306260 [DEBUG] switch_core_sqldb.c:1438 Secure Type: srtp:AES_CM_128_HMAC_SHA1_32
2010-12-22 09:00:27.306260 [DEBUG] mod_sofia.c:683 Local SDP sofia/internal/1000 at 192.168.1.113:5062:
v=0
o=FreeSWITCH 1293010327 1293010328 IN IP4 192.168.1.113
s=FreeSWITCH
c=IN IP4 192.168.1.113
t=0 0
m=audio 19700 RTP/SAVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:uJA327Ni4Srlz/MPcfvRc0H4gjeP3lpU9KEcPWcm



/b

On Dec 22, 2010, at 4:01 AM, Goutham BG wrote:

> Thanks for the response.
> I have pasted the freeswitch debug log of the call coming in to FreeSWITCH in http://pastebin.freeswitch.org/14852 .
> 
> I think I have set sip_secure_media=true before answering the call in my dialplan. The following is the entry for this extension in my dialplan:
> 
>   <extension name="IVR">
>     <condition field="destination_number" expression="^IVR$">
>       <action application="set" data="sip_secure_media=true"/>
>       <action application="bridge" data="loopback/app=socket:47.152.232.156:8084 async full"/>
>     </condition>
>   </extension>
> 
> Thanks
> Goutham B G
> 
> On Tue, Dec 21, 2010 at 11:47 PM, Brian West <brian at freeswitch.org> wrote:
> And clearly you overlooked my response... are you setting sip_secure_media=true after the call is answered in your dialplan?\
> 
> I need to see the full debug log of a call coming in to FreeSWITCH please on our pastebin.
> 
> /b
> 
> On Dec 21, 2010, at 12:00 PM, Goutham BG wrote:
> 
> > Posting the below query to freeswitch-users list as well. Any hints will be really helpful.
> >
> > ---------- Forwarded message ----------
> > From: Goutham BG <bggoutham at gmail.com>
> > Date: Mon, Dec 20, 2010 at 9:16 PM
> > Subject: Query related to enabling SRTP in FreeSWITCH-1.0.7
> > To: freeswitch-dev at lists.freeswitch.org
> >
> >
> > Hi,
> >
> > I have been trying to enable SRTP in FreeSWITCH-1.0.7 and have been facing some issues.
> > I have the following entry in my dialplan XML file:
> >
> > <extension name="IVR">
> >     <condition field="destination_number" expression="^IVR$">
> >       <action application="bridge" data="loopback/app=socket:47.152.232.156:8084 async full"/>
> >     </condition
> > </extension>
> >
> > A SIP phone (Avaya 12XX) configured in "SRTP best effort" mode dials into this extension and is connected to the IVR. But the media is established in SRTP in one way and RTP in the other way.
> > The phone offers the following SDP in the INVITE message:
> >
> > v=0
> > o=- 10170 10170 IN IP4 47.152.232.147
> > s=Sip Call
> > c=IN IP4 47.152.232.147
> > t=0 0
> > m=audio 5016 RTP/AVP 0 8 18 101 102
> > a=rtpmap:0 PCMU/8000
> > a=ptime:20
> > a=rtpmap:8 PCMA/8000
> > a=ptime:20
> > a=rtpmap:18 G729/8000
> > a=ptime:20
> > a=fmtp:18 annexb=no
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-15
> > a=rtpmap:102 X-nt-inforeq/8000
> > a=sendrecv
> > m=audio 5016 RTP/SAVP 0 8 18 101 102
> > a=rtpmap:0 PCMU/8000
> > a=ptime:20
> > a=rtpmap:8 PCMA/8000
> > a=ptime:20
> > a=rtpmap:18 G729/8000
> > a=ptime:20
> > a=fmtp:18 annexb=no
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-15
> > a=rtpmap:102 X-nt-inforeq/8000
> > a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Tjivoci1I/mVkt/Fq/ZsiY+
> > +ornJoXjZ5tSadho4
> > a=sendrecv
> >
> > As we can see, there are two "m=" lines in the SDP of the offer; one for RTP and another for SRTP. FreeSWITCH-1.0.7 answers the call by sending 200OK with the following SDP:
> >
> > v=0
> > o=FreeSWITCH 1291628984 1291628985 IN IP4 47.152.232.156
> > s=FreeSWITCH
> > c=IN IP4 47.152.232.156
> > t=0 0
> > m=audio 11280 RTP/AVP 0 101
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > m=audio 0 RTP/SAVP 19
> >
> > As you can see above, FreeSWITCH accepts the RTP stream and rejects the SRTP stream (by sending port as 0) in the SDP. The SIP phone sends the media in RTP(which is expected). But, FreeSWITCH sends the media in SRTP to the SIP phone. I believe this is a bug in FreeSWITCH as it is supposed to send the media in RTP since it accepted RTP in the answer (200OK).
> >
> > Query:
> > ======
> > In order to make FreeSWITCH select SRTP in the SDP of the answer(200OK), I made the following change(i.e, setting sip_secure_media=true) in FS dial plan:
> >
> > <extension name="IVR">
> >     <condition field="destination_number" expression="^IVR$">
> >       <action application="set" data="sip_secure_media=true"/>
> >       <action application="bridge" data="loopback/app=socket:47.152.232.156:8084 async full"/>
> >     </condition
> > </extension>
> >
> > In FreeSWITCH-1.0.6(before updating to 1.0.7), this worked and FS accepted the SRTP stream and rejected RTP in the answer(200 OK) as shown below:
> >
> > m=audio 0 RTP/AVP 19
> > m=audio 12084 RTP/SAVP 0 101
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:hgv7ClqDx1irTRrXq2NEm9Gbouw0969bBU3n+LcM
> >
> > But after updating the FreeSWITCH-1.0.6 to 1.0.7, the above mentioned dial plan change (i.e, setting sip_secure_media=true) is not working. It is still behaving in the same way as it did without the XML change.
> >
> > Can you please let me know if anything else needs to be added in dialplan XML file for enabling SRTP in this case in FreeSWITCH-1.0.7 or am I missing something here?
> >
> > I have referred the following FS wiki pages for making the SRTP changes:
> > http://wiki.freeswitch.org/wiki/Secure_RTP
> > http://wiki.freeswitch.org/wiki/SRTP
> >
> > Note: There is no issue when the SIP phone is configured in "SRTP only" mode where only SRTP stream is offered in the SDP of the INVITE. In this case, SIP phone and FreeSWITCH communicate properly using SRTP. This doesn't require setting "sip_secure_media=true" in the dialplan XML file.
> > P.S: I am a newbie to FreeSWITCH. So, please forgive me if I am asking basic questions.
> >
> > Thanks
> > Goutham B G
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From bggoutham at gmail.com  Wed Dec 22 18:21:44 2010
From: bggoutham at gmail.com (Goutham BG)
Date: Wed, 22 Dec 2010 20:51:44 +0530
Subject: [Freeswitch-users] Query related to enabling SRTP in
	FreeSWITCH-1.0.7
In-Reply-To: <68416830-413A-4F58-95F2-AE8C8FAB82BC@freeswitch.org>
References: <AANLkTikyGM3fN7Joi9VodrYiNf7TQ9gHWPz0iRDWBj9U@mail.gmail.com>
	<AANLkTi=JoWuUKrGM0qiBfVJ93nBzN7HQZOUxHEhkYi2P@mail.gmail.com>
	<D95EC7E6-EA17-4B5B-B12A-83A19A1D18E5@freeswitch.org>
	<AANLkTimU5PWDas2_m0Ykj_4bwC=ioUMaha22Tipri2rA@mail.gmail.com>
	<68416830-413A-4F58-95F2-AE8C8FAB82BC@freeswitch.org>
Message-ID: <AANLkTikoNa0_h40KNZtEUczE8rkv2B48JPoSfwyyyU=Q@mail.gmail.com>

I am using Avaya 1220 deskphone.

Yes, it works fine when we are using a phone configured to talk only in SRTP
(i.e, when there is a single "m=" line in the SDP of the INVITE sent to
FreeSWITCH ). This works for me as well.

But when the phone is configured in "SRTP best effort" mode, FreeSWITCH
accepts the RTP media line and rejects the SRTP media line. In the "SRTP
best effort" mode, phone sends two "m=" lines in the INVITE; one for RTP and
one for SRTP (the SDP is mentioned in my 1st mail).

Thanks
Goutham B G

On Wed, Dec 22, 2010 at 8:43 PM, Brian West <brian at freeswitch.org> wrote:

> Just tested with my snom phone and it works fine.
>
> 2010-12-22 09:00:27.304273 [DEBUG] switch_rtp.c:1424 Starting timer [soft]
> 160 bytes per 20ms
> 2010-12-22 09:00:27.306260 [DEBUG] sofia_glue.c:3219 Set 2833 dtmf send
> payload to 101
> 2010-12-22 09:00:27.306260 [DEBUG] sofia_glue.c:3224 Set 2833 dtmf receive
> payload to 101
> 2010-12-22 09:00:27.306260 [INFO] switch_rtp.c:1253 Activating Secure RTP
> SEND
> 2010-12-22 09:00:27.306260 [DEBUG] switch_core_sqldb.c:1438 Secure Type:
> srtp:AES_CM_128_HMAC_SHA1_32
> 2010-12-22 09:00:27.306260 [INFO] switch_rtp.c:1233 Activating Secure RTP
> RECV
> 2010-12-22 09:00:27.306260 [DEBUG] switch_core_sqldb.c:1438 Secure Type:
> srtp:AES_CM_128_HMAC_SHA1_32
> 2010-12-22 09:00:27.306260 [DEBUG] mod_sofia.c:683 Local SDP
> sofia/internal/1000 at 192.168.1.113:5062:
> v=0
> o=FreeSWITCH 1293010327 1293010328 IN IP4 192.168.1.113
> s=FreeSWITCH
> c=IN IP4 192.168.1.113
> t=0 0
> m=audio 19700 RTP/SAVP 0 101
> a=rtpmap:0 pcmu/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:uJA327Ni4Srlz/MPcfvRc0H4gjeP3lpU9KEcPWcm
>
>
>
> /b
>
> On Dec 22, 2010, at 4:01 AM, Goutham BG wrote:
>
> Thanks for the response.
> I have pasted the freeswitch debug log of the call coming in to FreeSWITCH
> in http://pastebin.freeswitch.org/14852 .
>
> I think I have set sip_secure_media=true before answering the call in my
> dialplan. The following is the entry for this extension in my dialplan:
>
>   <extension name="IVR">
>     <condition field="destination_number" expression="^IVR$">
> *      <action application="set" data="sip_secure_media=true"/>*
>       <action application="bridge" data="loopback/app=socket:
> 47.152.232.156:8084 async full"/>
>     </condition>
>   </extension>
>
> Thanks
> Goutham B G
>
> On Tue, Dec 21, 2010 at 11:47 PM, Brian West <brian at freeswitch.org> wrote:
>
>> And clearly you overlooked my response... are you setting
>> sip_secure_media=true after the call is answered in your dialplan?\
>>
>> I need to see the full debug log of a call coming in to FreeSWITCH please
>> on our pastebin.
>>
>> /b
>>
>> On Dec 21, 2010, at 12:00 PM, Goutham BG wrote:
>>
>> > Posting the below query to freeswitch-users list as well. Any hints will
>> be really helpful.
>> >
>> > ---------- Forwarded message ----------
>> > From: Goutham BG <bggoutham at gmail.com>
>> > Date: Mon, Dec 20, 2010 at 9:16 PM
>> > Subject: Query related to enabling SRTP in FreeSWITCH-1.0.7
>> > To: freeswitch-dev at lists.freeswitch.org
>> >
>> >
>> > Hi,
>> >
>> > I have been trying to enable SRTP in FreeSWITCH-1.0.7 and have been
>> facing some issues.
>> > I have the following entry in my dialplan XML file:
>> >
>> > <extension name="IVR">
>> >     <condition field="destination_number" expression="^IVR$">
>> >       <action application="bridge" data="loopback/app=socket:
>> 47.152.232.156:8084 async full"/>
>> >     </condition
>> > </extension>
>> >
>> > A SIP phone (Avaya 12XX) configured in "SRTP best effort" mode dials
>> into this extension and is connected to the IVR. But the media is
>> established in SRTP in one way and RTP in the other way.
>> > The phone offers the following SDP in the INVITE message:
>> >
>> > v=0
>> > o=- 10170 10170 IN IP4 47.152.232.147
>> > s=Sip Call
>> > c=IN IP4 47.152.232.147
>> > t=0 0
>> > m=audio 5016 RTP/AVP 0 8 18 101 102
>> > a=rtpmap:0 PCMU/8000
>> > a=ptime:20
>> > a=rtpmap:8 PCMA/8000
>> > a=ptime:20
>> > a=rtpmap:18 G729/8000
>> > a=ptime:20
>> > a=fmtp:18 annexb=no
>> > a=rtpmap:101 telephone-event/8000
>> > a=fmtp:101 0-15
>> > a=rtpmap:102 X-nt-inforeq/8000
>> > a=sendrecv
>> > m=audio 5016 RTP/SAVP 0 8 18 101 102
>> > a=rtpmap:0 PCMU/8000
>> > a=ptime:20
>> > a=rtpmap:8 PCMA/8000
>> > a=ptime:20
>> > a=rtpmap:18 G729/8000
>> > a=ptime:20
>> > a=fmtp:18 annexb=no
>> > a=rtpmap:101 telephone-event/8000
>> > a=fmtp:101 0-15
>> > a=rtpmap:102 X-nt-inforeq/8000
>> > a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Tjivoci1I/mVkt/Fq/ZsiY+
>> > +ornJoXjZ5tSadho4
>> > a=sendrecv
>> >
>> > As we can see, there are two "m=" lines in the SDP of the offer; one for
>> RTP and another for SRTP. FreeSWITCH-1.0.7 answers the call by sending 200OK
>> with the following SDP:
>> >
>> > v=0
>> > o=FreeSWITCH 1291628984 1291628985 IN IP4 47.152.232.156
>> > s=FreeSWITCH
>> > c=IN IP4 47.152.232.156
>> > t=0 0
>> > m=audio 11280 RTP/AVP 0 101
>> > a=rtpmap:0 PCMU/8000
>> > a=rtpmap:101 telephone-event/8000
>> > a=fmtp:101 0-16
>> > a=silenceSupp:off - - - -
>> > a=ptime:20
>> > m=audio 0 RTP/SAVP 19
>> >
>> > As you can see above, FreeSWITCH accepts the RTP stream and rejects the
>> SRTP stream (by sending port as 0) in the SDP. The SIP phone sends the media
>> in RTP(which is expected). But, FreeSWITCH sends the media in SRTP to the
>> SIP phone. I believe this is a bug in FreeSWITCH as it is supposed to send
>> the media in RTP since it accepted RTP in the answer (200OK).
>> >
>> > Query:
>> > ======
>> > In order to make FreeSWITCH select SRTP in the SDP of the answer(200OK),
>> I made the following change(i.e, setting sip_secure_media=true) in FS dial
>> plan:
>> >
>> > <extension name="IVR">
>> >     <condition field="destination_number" expression="^IVR$">
>> >       <action application="set" data="sip_secure_media=true"/>
>> >       <action application="bridge" data="loopback/app=socket:
>> 47.152.232.156:8084 async full"/>
>> >     </condition
>> > </extension>
>> >
>> > In FreeSWITCH-1.0.6(before updating to 1.0.7), this worked and FS
>> accepted the SRTP stream and rejected RTP in the answer(200 OK) as shown
>> below:
>> >
>> > m=audio 0 RTP/AVP 19
>> > m=audio 12084 RTP/SAVP 0 101
>> > a=rtpmap:0 PCMU/8000
>> > a=rtpmap:101 telephone-event/8000
>> > a=fmtp:101 0-16
>> > a=silenceSupp:off - - - -
>> > a=ptime:20
>> > a=crypto:1 AES_CM_128_HMAC_SHA1_80
>> inline:hgv7ClqDx1irTRrXq2NEm9Gbouw0969bBU3n+LcM
>> >
>> > But after updating the FreeSWITCH-1.0.6 to 1.0.7, the above mentioned
>> dial plan change (i.e, setting sip_secure_media=true) is not working. It is
>> still behaving in the same way as it did without the XML change.
>> >
>> > Can you please let me know if anything else needs to be added in
>> dialplan XML file for enabling SRTP in this case in FreeSWITCH-1.0.7 or am I
>> missing something here?
>> >
>> > I have referred the following FS wiki pages for making the SRTP changes:
>> > http://wiki.freeswitch.org/wiki/Secure_RTP
>> > http://wiki.freeswitch.org/wiki/SRTP
>> >
>> > Note: There is no issue when the SIP phone is configured in "SRTP only"
>> mode where only SRTP stream is offered in the SDP of the INVITE. In this
>> case, SIP phone and FreeSWITCH communicate properly using SRTP. This doesn't
>> require setting "sip_secure_media=true" in the dialplan XML file.
>> > P.S: I am a newbie to FreeSWITCH. So, please forgive me if I am asking
>> basic questions.
>> >
>> > Thanks
>> > Goutham B G
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Dec 22 18:24:50 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 22 Dec 2010 09:24:50 -0600
Subject: [Freeswitch-users] How to select and play audio in a conference
In-Reply-To: <4D1213C2.5090709@siweb.it>
References: <4D1213C2.5090709@siweb.it>
Message-ID: <98AD5415-C255-4CD5-B11E-688A4493C0B8@freeswitch.org>

Depends we have a new dtmf digit parser that you could use to bind keys to do any number of things..

        <action application="bind_digit_action" data="cool,500,exec:playback,ivr/ivr-welcome_to_freeswitch.wav"/>                                               
        <action application="bind_digit_action" data="cool,~7\d{3},exec:playback,ivr/ivr-welcome_to_freeswitch.wav"/>                                           
        <action application="bind_digit_action" data="cool,~1\d+,exec:digit_action_set_realm,rad"/>                                                             
        <action application="bind_digit_action" data="rad,~1\d+,exec:digit_action_set_realm,cool"/>                                                             
        <action application="digit_action_set_realm" data="cool"/> 


Is one example.

Thanks,
Brian

On Dec 22, 2010, at 9:05 AM, Fabio Bigliardi wrote:

> Hi all,
> 
> I would like to select audio files to play to all members of a conference from within the conference through the use of DTMF keypad inputs. 
> How can I do this?
> 
> Thank you very much.
> 
> Fabio Bigliardi
> 
> 
> 
> 
> :: SYSNET TELEMATICA srl ::
> CONFIDENZIALE:
> Questo messaggio e gli eventuali allegati sono confidenziali e riservati.
> Se vi ? stato recapitato per errore e non siete fra i destinatari elencati,
> siete pregati di darne immediatamente avviso al mittente e cancellare il messaggio
> di posta e gli eventuali file allegati. Le informazioni contenute non devono
> essere mostrate ad altri, n? utilizzate, memorizzate o copiate in qualsiasi forma.
> 
> CONFIDENTIALITY :
> This e-mail and any attachments are confidential and may be privileged.
> If you are not a named recipient, please notify the sender immediately and delete
> this e-mail and any attachment. Do not disclose the contents to another person,
> use it for any purpose or store or copy the information in any medium.

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From rafonline at hotmail.com  Wed Dec 22 18:29:16 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Wed, 22 Dec 2010 15:29:16 +0000
Subject: [Freeswitch-users] calling card  app design decision
Message-ID: <COL122-W40EEE9A9EEB0A4C5B52283B51B0@phx.gbl>


Hi,

I am wanting to develop a calling card service using FreeSWITCH.  It's seems FreeSWITCH is very flexible in terms of how one could go about implementing this service.    I am intending to use mod_nibblebill and mod_lcr to do the core stuff.  I would like to use Java for any programming (as it's the language I am most familiar with) however, I am unsure whether to take an event socket based approach to the application or to use something like mod_java.  

Any advice/suggestions will be most welcome.

Cheers

Raf
 		 	   		  
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From mkane02 at harris.com  Wed Dec 22 18:45:49 2010
From: mkane02 at harris.com (Kane, Michael (mkane02))
Date: Wed, 22 Dec 2010 10:45:49 -0500
Subject: [Freeswitch-users] Dropping the Subject: header
Message-ID: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net>

Hello all, we are developing a custom app that will require sending a
Subject: header (with a specific value) with the initial invite to
trigger additional calling features and behavior.  What we're finding
when we send the Invite,  is that FS is dropping the Subject: header and
continues normal call processing.  Is there a configuration parameter
that allows the header to pass through FS?

I submitted a SIP trace in pastbin and the reference number is 14853.

Mike
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From steveayre at gmail.com  Wed Dec 22 18:57:19 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 22 Dec 2010 15:57:19 +0000
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net>
Message-ID: <AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com>

I assume you're doing a bridge?

FreeSWITCH is a b2bua... meaning the call going into FreeSWITCH is a
separate call from the one going out (it just links the media on the 2
calls together to bridge them). That's why the variable's not being
copied across by default.

It is however possible to tell FreeSWITCH to copy a variable from the
A-leg to the B-leg, and any non-standard SIP headers can be read and
written using the sip_h_ variable prefix. Try this:

<extension name="...">
  <condition ...>
    ...
    <action application="export" data="sip_h_Subject" />
    <action application="bridge" data="..." />
    ...
  </condition>
</extension>

The subject should be in the sip_h_Subject variable on the A-leg, the
above will export (copy) that variable onto the B-leg and it should
then appear in the outgoing invite.

-Steve


On 22 December 2010 15:45, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
> Hello all, we are developing a custom app that will require sending a
> Subject: header (with a specific value) with the initial invite to trigger
> additional calling features and behavior.? What we?re finding when we send
> the Invite,? is that FS is dropping the Subject: header and continues normal
> call processing.? Is there a configuration parameter that allows the header
> to pass through FS?
>
> I submitted a SIP trace in pastbin and the reference number is 14853.
>
> Mike
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Wed Dec 22 19:02:13 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 22 Dec 2010 10:02:13 -0600
Subject: [Freeswitch-users] Query related to enabling SRTP in
	FreeSWITCH-1.0.7
In-Reply-To: <AANLkTikoNa0_h40KNZtEUczE8rkv2B48JPoSfwyyyU=Q@mail.gmail.com>
References: <AANLkTikyGM3fN7Joi9VodrYiNf7TQ9gHWPz0iRDWBj9U@mail.gmail.com>
	<AANLkTi=JoWuUKrGM0qiBfVJ93nBzN7HQZOUxHEhkYi2P@mail.gmail.com>
	<D95EC7E6-EA17-4B5B-B12A-83A19A1D18E5@freeswitch.org>
	<AANLkTimU5PWDas2_m0Ykj_4bwC=ioUMaha22Tipri2rA@mail.gmail.com>
	<68416830-413A-4F58-95F2-AE8C8FAB82BC@freeswitch.org>
	<AANLkTikoNa0_h40KNZtEUczE8rkv2B48JPoSfwyyyU=Q@mail.gmail.com>
Message-ID: <2AC422BD-321F-49D0-ADFA-EA6905BC09D0@freeswitch.org>

The SAVP line needs to come first in that case if you want to prefer it since you offered it first.

/b

On Dec 22, 2010, at 9:21 AM, Goutham BG wrote:

> I am using Avaya 1220 deskphone.
> 
> Yes, it works fine when we are using a phone configured to talk only in SRTP (i.e, when there is a single "m=" line in the SDP of the INVITE sent to FreeSWITCH ). This works for me as well.
> 
> But when the phone is configured in "SRTP best effort" mode, FreeSWITCH accepts the RTP media line and rejects the SRTP media line. In the "SRTP best effort" mode, phone sends two "m=" lines in the INVITE; one for RTP and one for SRTP (the SDP is mentioned in my 1st mail). 
> 
> Thanks
> Goutham B G



From mkane02 at harris.com  Wed Dec 22 19:03:58 2010
From: mkane02 at harris.com (Kane, Michael (mkane02))
Date: Wed, 22 Dec 2010 11:03:58 -0500
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net>
	<AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com>
Message-ID: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF6@mlbe2k9.cs.myharris.net>

Thanks Steve, I truly appreciate the tip.

Mike

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Wednesday, December 22, 2010 10:57 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropping the Subject: header

I assume you're doing a bridge?

FreeSWITCH is a b2bua... meaning the call going into FreeSWITCH is a
separate call from the one going out (it just links the media on the 2
calls together to bridge them). That's why the variable's not being
copied across by default.

It is however possible to tell FreeSWITCH to copy a variable from the
A-leg to the B-leg, and any non-standard SIP headers can be read and
written using the sip_h_ variable prefix. Try this:

<extension name="...">
  <condition ...>
    ...
    <action application="export" data="sip_h_Subject" />
    <action application="bridge" data="..." />
    ...
  </condition>
</extension>

The subject should be in the sip_h_Subject variable on the A-leg, the
above will export (copy) that variable onto the B-leg and it should
then appear in the outgoing invite.

-Steve


On 22 December 2010 15:45, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
> Hello all, we are developing a custom app that will require sending a
> Subject: header (with a specific value) with the initial invite to trigger
> additional calling features and behavior.? What we're finding when we send
> the Invite,? is that FS is dropping the Subject: header and continues normal
> call processing.? Is there a configuration parameter that allows the header
> to pass through FS?
>
> I submitted a SIP trace in pastbin and the reference number is 14853.
>
> Mike
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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


From mkane02 at harris.com  Wed Dec 22 19:29:04 2010
From: mkane02 at harris.com (Kane, Michael (mkane02))
Date: Wed, 22 Dec 2010 11:29:04 -0500
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net>
	<AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com>
Message-ID: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net>

Hey Steve, I tried to implement what you suggested and posted the console output to pastbin (14854).  I forgot to add my dialplan xml, so I've pasted it here, since it's only a few lines.  Sorry for pasting in the body of my email.

Mike

    <extension name="sipexten">
     <condition field="destination_number" expression="^(10[0-9][0-9])$">
       <action application="export" data="sip_h_Subject" />
       <action application="set" data="call_timeout=180"/>
       <action application="log" data="crit dialing extension $1" />
       <action application="bridge" data="user/$1@${domain_name}" />
     </condition>
    </extension>


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Wednesday, December 22, 2010 10:57 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropping the Subject: header

I assume you're doing a bridge?

FreeSWITCH is a b2bua... meaning the call going into FreeSWITCH is a
separate call from the one going out (it just links the media on the 2
calls together to bridge them). That's why the variable's not being
copied across by default.

It is however possible to tell FreeSWITCH to copy a variable from the
A-leg to the B-leg, and any non-standard SIP headers can be read and
written using the sip_h_ variable prefix. Try this:

<extension name="...">
  <condition ...>
    ...
    <action application="export" data="sip_h_Subject" />
    <action application="bridge" data="..." />
    ...
  </condition>
</extension>

The subject should be in the sip_h_Subject variable on the A-leg, the
above will export (copy) that variable onto the B-leg and it should
then appear in the outgoing invite.

-Steve


On 22 December 2010 15:45, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
> Hello all, we are developing a custom app that will require sending a
> Subject: header (with a specific value) with the initial invite to trigger
> additional calling features and behavior.? What we're finding when we send
> the Invite,? is that FS is dropping the Subject: header and continues normal
> call processing.? Is there a configuration parameter that allows the header
> to pass through FS?
>
> I submitted a SIP trace in pastbin and the reference number is 14853.
>
> Mike
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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


From helmut.kuper at ewetel.de  Wed Dec 22 19:38:20 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 22 Dec 2010 17:38:20 +0100
Subject: [Freeswitch-users] XMAS Greets
Message-ID: <4D12297C.3050504@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello dear FS Community,

because today is my last day at work for this year (hurray) I would like
to wish you all a peacefull christmas and a healthy new year 2011.

Thanks a lot for all the tips&tricks, bugfixes and new features this year.

Greets from verrrrrry cold and snowy germany
Helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.10 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

iEYEARECAAYFAk0SKXwACgkQ4tZeNddg3dy+BACeMt0qUo+s6Qq6KKT9zYI5dKys
7nkAn0coBygql4jAPCAfm60B6FfGC7/U
=UboS
-----END PGP SIGNATURE-----


From steveayre at gmail.com  Wed Dec 22 20:15:45 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 22 Dec 2010 17:15:45 +0000
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net>
	<AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net>
Message-ID: <AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com>

Hmm,

Can you see whether sip_h_Subject is on the A-leg (you can use <action
application="info"/> to log all the variables on the channel)?

If is is, also try:
<action application="export" data="sip_h_Subject=${sip_h_Subject}"/>
although I didn't think this was required if the variable was already
set with the same name.

-Steve


On 22 December 2010 16:29, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
> Hey Steve, I tried to implement what you suggested and posted the console output to pastbin (14854). ?I forgot to add my dialplan xml, so I've pasted it here, since it's only a few lines. ?Sorry for pasting in the body of my email.
>
> Mike
>
> ? ?<extension name="sipexten">
> ? ? <condition field="destination_number" expression="^(10[0-9][0-9])$">
> ? ? ? <action application="export" data="sip_h_Subject" />
> ? ? ? <action application="set" data="call_timeout=180"/>
> ? ? ? <action application="log" data="crit dialing extension $1" />
> ? ? ? <action application="bridge" data="user/$1@${domain_name}" />
> ? ? </condition>
> ? ?</extension>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
> Sent: Wednesday, December 22, 2010 10:57 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>
> I assume you're doing a bridge?
>
> FreeSWITCH is a b2bua... meaning the call going into FreeSWITCH is a
> separate call from the one going out (it just links the media on the 2
> calls together to bridge them). That's why the variable's not being
> copied across by default.
>
> It is however possible to tell FreeSWITCH to copy a variable from the
> A-leg to the B-leg, and any non-standard SIP headers can be read and
> written using the sip_h_ variable prefix. Try this:
>
> <extension name="...">
> ?<condition ...>
> ? ?...
> ? ?<action application="export" data="sip_h_Subject" />
> ? ?<action application="bridge" data="..." />
> ? ?...
> ?</condition>
> </extension>
>
> The subject should be in the sip_h_Subject variable on the A-leg, the
> above will export (copy) that variable onto the B-leg and it should
> then appear in the outgoing invite.
>
> -Steve
>
>
> On 22 December 2010 15:45, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
>> Hello all, we are developing a custom app that will require sending a
>> Subject: header (with a specific value) with the initial invite to trigger
>> additional calling features and behavior.? What we're finding when we send
>> the Invite,? is that FS is dropping the Subject: header and continues normal
>> call processing.? Is there a configuration parameter that allows the header
>> to pass through FS?
>>
>> I submitted a SIP trace in pastbin and the reference number is 14853.
>>
>> Mike
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From chris at cloudtel.com  Wed Dec 22 20:17:19 2010
From: chris at cloudtel.com (Chris Burns)
Date: Wed, 22 Dec 2010 12:17:19 -0500
Subject: [Freeswitch-users] SBC on kamailio
In-Reply-To: <0A1FDB5DAA23564F8758BA05D26DCD74C6C15E@exchange.telecats.nl>
References: <0A1FDB5DAA23564F8758BA05D26DCD74C6C15E@exchange.telecats.nl>
Message-ID: <AANLkTinE5XmMqUHeCQe=gAzcLC81Xjj1nHvVr=gnqcEE@mail.gmail.com>

FreeSWITCH is not a SIP proxy, so you won't be able to proxy registrations.
I use FS as a SBC in a few cases ... I mainly use it for billing / routing,
and I usually bypass media to one of a cluster of switches playing the role
of PBX. SIP regs get aimed at the PBXs using DNS SRV and carriers point
their trunks at our SBC ... hope that helps

On Mon, Dec 20, 2010 at 10:52 AM, Henry Dogger <h.dogger at telecats.nl> wrote:

> Hi all,
>
>
>
> We have a kamailio installation for our voip system and would like to add a
> SBC.
>
> We found the freeswitch possibility:
> http://wiki.freeswitch.org/wiki/SBC_Setup
>
> But in this case, 302 Redirect SIP is used, we would like the SBC to act as
> a pass-through controller, just to be a gateway for SIP clients from
> outside.
>
> Is something like this possible? And will registrations be handled
> correctly since all SIP messages will be on port 5060.
>
> Thanks in advance.
>
>
>
> Kind regards,
>
>
>
> Henry Dogger
>
> Telecats BV
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris at cloudtel.com  Wed Dec 22 17:50:37 2010
From: chris at cloudtel.com (Chris Burns)
Date: Wed, 22 Dec 2010 09:50:37 -0500
Subject: [Freeswitch-users] calling card  app design decision
In-Reply-To: <COL122-W40EEE9A9EEB0A4C5B52283B51B0@phx.gbl>
References: <COL122-W40EEE9A9EEB0A4C5B52283B51B0@phx.gbl>
Message-ID: <201012220950.37452.chris@cloudtel.com>

Keep your app server and call server separate is my opinion. I would recommend 
inline scripting the small stuff with lua instead of loading the JVM into 
your switch

Anything for billing should be done through events and not inline. Run the JVM 
outside the switch and connect to the event socket. If you design it right, 
you can scale out multiple switches and have your code in a single JVM 
monitor them all

On December 22, 2010 10:29:16 am Rafqat . wrote:
> Hi,
>
> I am wanting to develop a calling card service using FreeSWITCH.  It's
> seems FreeSWITCH is very flexible in terms of how one could go about
> implementing this service.    I am intending to use mod_nibblebill and
> mod_lcr to do the core stuff.  I would like to use Java for any programming
> (as it's the language I am most familiar with) however, I am unsure whether
> to take an event socket based approach to the application or to use
> something like mod_java.
>
> Any advice/suggestions will be most welcome.
>
> Cheers
>
> Raf




From rafonline at hotmail.com  Wed Dec 22 21:01:30 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Wed, 22 Dec 2010 18:01:30 +0000
Subject: [Freeswitch-users] calling card  app design decision
In-Reply-To: <201012220950.37452.chris@cloudtel.com>
References: <COL122-W40EEE9A9EEB0A4C5B52283B51B0@phx.gbl>,
	<201012220950.37452.chris@cloudtel.com>
Message-ID: <COL122-W65D10EB0D7FF85BF6A03E3B51B0@phx.gbl>



Thanks for your advice.  I will take it onboard.

Cheers

Raf

> From: chris at cloudtel.com
> To: freeswitch-users at lists.freeswitch.org
> Date: Wed, 22 Dec 2010 09:50:37 -0500
> Subject: Re: [Freeswitch-users] calling card  app design decision
> 
> Keep your app server and call server separate is my opinion. I would recommend 
> inline scripting the small stuff with lua instead of loading the JVM into 
> your switch
> 
> Anything for billing should be done through events and not inline. Run the JVM 
> outside the switch and connect to the event socket. If you design it right, 
> you can scale out multiple switches and have your code in a single JVM 
> monitor them all
> 
> On December 22, 2010 10:29:16 am Rafqat . wrote:
> > Hi,
> >
> > I am wanting to develop a calling card service using FreeSWITCH.  It's
> > seems FreeSWITCH is very flexible in terms of how one could go about
> > implementing this service.    I am intending to use mod_nibblebill and
> > mod_lcr to do the core stuff.  I would like to use Java for any programming
> > (as it's the language I am most familiar with) however, I am unsure whether
> > to take an event socket based approach to the application or to use
> > something like mod_java.
> >
> > Any advice/suggestions will be most welcome.
> >
> > Cheers
> >
> > Raf
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
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From kbdfck at gmail.com  Wed Dec 22 19:01:30 2010
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 22 Dec 2010 11:01:30 -0500
Subject: [Freeswitch-users] bind_meta_app and bind_digit_action with inband
	DTMF?
Message-ID: <AANLkTimiDuz8iHLkWL2Z1osyAz1pFQr3jV-hvXw=iGAR@mail.gmail.com>

Hi! We are trying to migrate our Asterisk system to Freeswitch, and
faced with some trouble:

We use inband at user side, and want Freeswitch to regenerate it to
RFC2833 before sending to media gateway.

When user endpoints use rfc2833 bind_meta_app and bind_digit_action
work as expected, but not with start_dtmf (as needed to parse inband
DTMF). We can't use rfc2833 DTMF on user endpoints as there seems to
be a problem in freeswitch or our media gateway leading to audible
DTMF duplicates.

When user endpoints are set to inband, I see messages on console about
DTMF detection:

2010-12-22 18:50:30.802253 [DEBUG] switch_ivr_async.c:2089 DTMF DETECTED: [*]
2010-12-22 18:50:30.803253 [DEBUG] switch_rtp.c:1985 Send start packet
for [*] ts=503795131 dur=160/160/2000 seq=43701
2010-12-22 18:50:30.823282 [DEBUG] switch_rtp.c:1921 Send middle
packet for [*] ts=503795131 dur=320/320/2000 seq=43702
...skip...
2010-12-22 18:50:30.923393 [DEBUG] switch_rtp.c:1921 Send middle
packet for [*] ts=503795131 dur=1920/1920/2000 seq=43712
2010-12-22 18:50:30.923393 [DEBUG] switch_rtp.c:1921 Send end packet
for [*] ts=503795131 dur=2080/2080/2000 seq=43713
...skip...
2010-12-22 18:50:31.242695 [DEBUG] switch_ivr_async.c:2089 DTMF DETECTED: [1]
2010-12-22 18:50:31.242695 [DEBUG] switch_rtp.c:1985 Send start packet
for [1] ts=503798651 dur=160/160/2000 seq=43732

So digits are received but no in-call action launched by
bind_meta_app. I tried start_dtmf before and after bind_meta_app but
with no success. Is there a way to make inband dtmf launch in-call
actions while transcoding it to RFC2833 on trunk side?
I can't make inband dtmf trigger my actions :(

-- 
Best regards,

Dmitry Sytchev,
IT Engineer


From george at ezuce.com  Wed Dec 22 19:35:07 2010
From: george at ezuce.com (George Niculae)
Date: Wed, 22 Dec 2010 18:35:07 +0200
Subject: [Freeswitch-users] call dropped while trying to transfer
Message-ID: <AANLkTiku8ZRDxFxQECys14BW94X2J28UpxKZu+p_BOXq@mail.gmail.com>

Hi All,

I am working on an IVR application based on FS (running FreeSWITCH
Version 1.0.head (git-43393f2 2010-12-15 20-59-42 -0600) where the
following scenario fails:
user 201 calls to 100 (autoattendant), hears menu then press # to
transfer to voicemail (101), but the call is dropped (transfer is made
using uuid_deflect api command)
Dialplan extension configured like:

  <extension name="IVR">
    <condition field="destination_number" expression="^IVR${dollar}">
      <action application="socket" data="192.168.0.114:8084 async full"/>
    </condition>
  </extension>

Actions taken are:
- when call arrives to extension 100 call is bridged (hangup_after_bridge=true)
- answer the call, autoattendant menu is played and DTMF collected
- when # pressed, call is transfered to 101 using uuid_deflect
- call arrives to voicemail extension and is again bridged
- call is answered - at this point in time the initial bridge hangs up
and the whole call is dropped
Please see console output http://pastebin.freeswitch.org/14855

When debugging the application, If I keep the first channel connected
transfer works just fine without dropping the call.
Pretty sure I'm missing something here, any suggestion highly appreciated

Thanks,
George


From william.nishio at voicetechnology.com.br  Wed Dec 22 21:03:57 2010
From: william.nishio at voicetechnology.com.br (William Kendi ...)
Date: Wed, 22 Dec 2010 16:03:57 -0200
Subject: [Freeswitch-users] mod_fsv, record and playback options
Message-ID: <AANLkTink1-sZBh9zO9p8_he8XfaH+NdLHRHwZi5n_BE5@mail.gmail.com>

Greetings,

Currently, I am trying to record and play videos through FreeSWITCH using
the "mod_fsv" module. Some options like timeouts and interrupts by dtmf
digits seems to be lacking in the "mod_fsv" module and I cant even stop the
video recording using the "break" application. The video recording stop when
the user disconnects.
Anyone has any ideas about how to stop the FSV video recording without
disconnecting the user?

Thanks in advance.
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From bggoutham at gmail.com  Wed Dec 22 21:26:31 2010
From: bggoutham at gmail.com (Goutham BG)
Date: Wed, 22 Dec 2010 23:56:31 +0530
Subject: [Freeswitch-users] Query related to enabling SRTP in
	FreeSWITCH-1.0.7
In-Reply-To: <2AC422BD-321F-49D0-ADFA-EA6905BC09D0@freeswitch.org>
References: <AANLkTikyGM3fN7Joi9VodrYiNf7TQ9gHWPz0iRDWBj9U@mail.gmail.com>
	<AANLkTi=JoWuUKrGM0qiBfVJ93nBzN7HQZOUxHEhkYi2P@mail.gmail.com>
	<D95EC7E6-EA17-4B5B-B12A-83A19A1D18E5@freeswitch.org>
	<AANLkTimU5PWDas2_m0Ykj_4bwC=ioUMaha22Tipri2rA@mail.gmail.com>
	<68416830-413A-4F58-95F2-AE8C8FAB82BC@freeswitch.org>
	<AANLkTikoNa0_h40KNZtEUczE8rkv2B48JPoSfwyyyU=Q@mail.gmail.com>
	<2AC422BD-321F-49D0-ADFA-EA6905BC09D0@freeswitch.org>
Message-ID: <AANLkTi=J=rKLeMikPFPJa60v7yx5N1cZVuaH71Xp-QW7@mail.gmail.com>

Ok. But, I was under the impression that by setting the sip_secure_media
=true, we would be able to force FreeSWITCH to choose SRTP for media. Is
this understanding wrong?

Thanks
Goutham B G

On Wed, Dec 22, 2010 at 9:32 PM, Brian West <brian at freeswitch.org> wrote:

> The SAVP line needs to come first in that case if you want to prefer it
> since you offered it first.
>
> /b
>
> On Dec 22, 2010, at 9:21 AM, Goutham BG wrote:
>
> > I am using Avaya 1220 deskphone.
> >
> > Yes, it works fine when we are using a phone configured to talk only in
> SRTP (i.e, when there is a single "m=" line in the SDP of the INVITE sent to
> FreeSWITCH ). This works for me as well.
> >
> > But when the phone is configured in "SRTP best effort" mode, FreeSWITCH
> accepts the RTP media line and rejects the SRTP media line. In the "SRTP
> best effort" mode, phone sends two "m=" lines in the INVITE; one for RTP and
> one for SRTP (the SDP is mentioned in my 1st mail).
> >
> > Thanks
> > Goutham B G
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Dec 22 21:45:13 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 22 Dec 2010 12:45:13 -0600
Subject: [Freeswitch-users] Query related to enabling SRTP in
	FreeSWITCH-1.0.7
In-Reply-To: <AANLkTi=J=rKLeMikPFPJa60v7yx5N1cZVuaH71Xp-QW7@mail.gmail.com>
References: <AANLkTikyGM3fN7Joi9VodrYiNf7TQ9gHWPz0iRDWBj9U@mail.gmail.com>
	<AANLkTi=JoWuUKrGM0qiBfVJ93nBzN7HQZOUxHEhkYi2P@mail.gmail.com>
	<D95EC7E6-EA17-4B5B-B12A-83A19A1D18E5@freeswitch.org>
	<AANLkTimU5PWDas2_m0Ykj_4bwC=ioUMaha22Tipri2rA@mail.gmail.com>
	<68416830-413A-4F58-95F2-AE8C8FAB82BC@freeswitch.org>
	<AANLkTikoNa0_h40KNZtEUczE8rkv2B48JPoSfwyyyU=Q@mail.gmail.com>
	<2AC422BD-321F-49D0-ADFA-EA6905BC09D0@freeswitch.org>
	<AANLkTi=J=rKLeMikPFPJa60v7yx5N1cZVuaH71Xp-QW7@mail.gmail.com>
Message-ID: <92112AAC-66D4-47F5-BF48-FD9FA3E36D02@freeswitch.org>

But go look up sip offer answer since you offered it first you obviously prefer it and we have no way to prefer it right now since sofia handles that if you set the srtp to option and its the first in the offier it will pick it and reject the non-srtp one.

Otherwise you can file a bounty on this and we can investigate what we can do about it.

/b

On Dec 22, 2010, at 12:26 PM, Goutham BG wrote:

> Ok. But, I was under the impression that by setting the sip_secure_media =true, we would be able to force FreeSWITCH to choose SRTP for media. Is this understanding wrong?
> 
> Thanks
> Goutham B G



From mkane02 at harris.com  Wed Dec 22 21:50:54 2010
From: mkane02 at harris.com (Kane, Michael (mkane02))
Date: Wed, 22 Dec 2010 13:50:54 -0500
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net><AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net>
	<AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com>
Message-ID: <AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net>

Hey Steve, the Subject is still not regenerated through the switch.  I posted a new pastbin.

14858

Mike

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Wednesday, December 22, 2010 12:16 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropping the Subject: header

Hmm,

Can you see whether sip_h_Subject is on the A-leg (you can use <action
application="info"/> to log all the variables on the channel)?

If is is, also try:
<action application="export" data="sip_h_Subject=${sip_h_Subject}"/>
although I didn't think this was required if the variable was already
set with the same name.

-Steve


On 22 December 2010 16:29, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
> Hey Steve, I tried to implement what you suggested and posted the console output to pastbin (14854). ?I forgot to add my dialplan xml, so I've pasted it here, since it's only a few lines. ?Sorry for pasting in the body of my email.
>
> Mike
>
> ? ?<extension name="sipexten">
> ? ? <condition field="destination_number" expression="^(10[0-9][0-9])$">
> ? ? ? <action application="export" data="sip_h_Subject" />
> ? ? ? <action application="set" data="call_timeout=180"/>
> ? ? ? <action application="log" data="crit dialing extension $1" />
> ? ? ? <action application="bridge" data="user/$1@${domain_name}" />
> ? ? </condition>
> ? ?</extension>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
> Sent: Wednesday, December 22, 2010 10:57 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>
> I assume you're doing a bridge?
>
> FreeSWITCH is a b2bua... meaning the call going into FreeSWITCH is a
> separate call from the one going out (it just links the media on the 2
> calls together to bridge them). That's why the variable's not being
> copied across by default.
>
> It is however possible to tell FreeSWITCH to copy a variable from the
> A-leg to the B-leg, and any non-standard SIP headers can be read and
> written using the sip_h_ variable prefix. Try this:
>
> <extension name="...">
> ?<condition ...>
> ? ?...
> ? ?<action application="export" data="sip_h_Subject" />
> ? ?<action application="bridge" data="..." />
> ? ?...
> ?</condition>
> </extension>
>
> The subject should be in the sip_h_Subject variable on the A-leg, the
> above will export (copy) that variable onto the B-leg and it should
> then appear in the outgoing invite.
>
> -Steve
>
>
> On 22 December 2010 15:45, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
>> Hello all, we are developing a custom app that will require sending a
>> Subject: header (with a specific value) with the initial invite to trigger
>> additional calling features and behavior.? What we're finding when we send
>> the Invite,? is that FS is dropping the Subject: header and continues normal
>> call processing.? Is there a configuration parameter that allows the header
>> to pass through FS?
>>
>> I submitted a SIP trace in pastbin and the reference number is 14853.
>>
>> Mike
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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


From Avi at aMarcus.com  Wed Dec 22 21:54:58 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Wed, 22 Dec 2010 20:54:58 +0200
Subject: [Freeswitch-users] 2600hz. Access to my web interface.
In-Reply-To: <E1PVJkM-0003gp-00.yurazilot1-list-ru@f142.mail.ru>
References: <E1PVJkM-0003gp-00.yurazilot1-list-ru@f142.mail.ru>
Message-ID: <AANLkTim8wdBju_HcHjbAHU_JzWNzbqtNwiTRcZWqpqjQ@mail.gmail.com>

Ask in #2600hz on freenode if you didn't get an answer yet.
-Avi

2010/12/22 ZILOT <yurazilot1 at list.ru>

>  Hello !! I have installed FS+GUI (2600hz) on FreeBSD
>  http://www.2600hz.org/
>
> I have forgotten the password to my web interface (2600hz), but i have ssh
> root access. Where I can search login\password to my web interface or
> replace them? Please help.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Dec 22 21:57:37 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 22 Dec 2010 12:57:37 -0600
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net><AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net>
	<AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net>
Message-ID: <018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org>

What is the sudden obsession with the subject header?  Looking thru the code it appears as if sip_h_X and sip_h_P are the only two things that you can do... a patch will have to be made if you wish to set the subject on an outbound invite... see sofia_glue.c  in do_invite.

/b

On Dec 22, 2010, at 12:50 PM, Kane, Michael (mkane02) wrote:

> Hey Steve, the Subject is still not regenerated through the switch.  I posted a new pastbin.
> 
> 14858
> 
> Mike



From mkane02 at harris.com  Wed Dec 22 22:09:56 2010
From: mkane02 at harris.com (Kane, Michael (mkane02))
Date: Wed, 22 Dec 2010 14:09:56 -0500
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net><AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net><AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net>
	<018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org>
Message-ID: <AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net>

We have a custom app that requires the Subject header with values so our
endpoints can route the audio accordingly.  I'm ok with using sip_h_X
and or sip_h_P, but can't seem to get FS to pass the Subject header.
Any direction would be greatly appreciated.

Mike

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West
Sent: Wednesday, December 22, 2010 1:58 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropping the Subject: header

What is the sudden obsession with the subject header?  Looking thru the
code it appears as if sip_h_X and sip_h_P are the only two things that
you can do... a patch will have to be made if you wish to set the
subject on an outbound invite... see sofia_glue.c  in do_invite.

/b

On Dec 22, 2010, at 12:50 PM, Kane, Michael (mkane02) wrote:

> Hey Steve, the Subject is still not regenerated through the switch.  I
posted a new pastbin.
> 
> 14858
> 
> Mike


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


From bggoutham at gmail.com  Wed Dec 22 22:16:32 2010
From: bggoutham at gmail.com (Goutham BG)
Date: Thu, 23 Dec 2010 00:46:32 +0530
Subject: [Freeswitch-users] Query related to enabling SRTP in
	FreeSWITCH-1.0.7
In-Reply-To: <92112AAC-66D4-47F5-BF48-FD9FA3E36D02@freeswitch.org>
References: <AANLkTikyGM3fN7Joi9VodrYiNf7TQ9gHWPz0iRDWBj9U@mail.gmail.com>
	<AANLkTi=JoWuUKrGM0qiBfVJ93nBzN7HQZOUxHEhkYi2P@mail.gmail.com>
	<D95EC7E6-EA17-4B5B-B12A-83A19A1D18E5@freeswitch.org>
	<AANLkTimU5PWDas2_m0Ykj_4bwC=ioUMaha22Tipri2rA@mail.gmail.com>
	<68416830-413A-4F58-95F2-AE8C8FAB82BC@freeswitch.org>
	<AANLkTikoNa0_h40KNZtEUczE8rkv2B48JPoSfwyyyU=Q@mail.gmail.com>
	<2AC422BD-321F-49D0-ADFA-EA6905BC09D0@freeswitch.org>
	<AANLkTi=J=rKLeMikPFPJa60v7yx5N1cZVuaH71Xp-QW7@mail.gmail.com>
	<92112AAC-66D4-47F5-BF48-FD9FA3E36D02@freeswitch.org>
Message-ID: <AANLkTi=11sXc28KssxmuQ+kRqp2mGnyfzYYHR0FFhMqD@mail.gmail.com>

Thanks a lot for clarifying this.

Thanks
Goutham B G

On Thu, Dec 23, 2010 at 12:15 AM, Brian West <brian at freeswitch.org> wrote:

> But go look up sip offer answer since you offered it first you obviously
> prefer it and we have no way to prefer it right now since sofia handles that
> if you set the srtp to option and its the first in the offier it will pick
> it and reject the non-srtp one.
>
> Otherwise you can file a bounty on this and we can investigate what we can
> do about it.
>
> /b
>
> On Dec 22, 2010, at 12:26 PM, Goutham BG wrote:
>
> > Ok. But, I was under the impression that by setting the sip_secure_media
> =true, we would be able to force FreeSWITCH to choose SRTP for media. Is
> this understanding wrong?
> >
> > Thanks
> > Goutham B G
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.ponzone at ipeva.fr  Wed Dec 22 22:30:49 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 22 Dec 2010 20:30:49 +0100
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net><AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net><AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net>
	<018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net>
Message-ID: <41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr>

Michael,

what Brian meant is that FreeSWICH can only passes the X-something and P-something headers, which are the only allowed ones.
Any system requiring custom headers should implement:
X-Subject
and not
Subject


David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 22/12/2010 ? 20:09, Kane, Michael (mkane02) a ?crit :

> We have a custom app that requires the Subject header with values so our
> endpoints can route the audio accordingly.  I'm ok with using sip_h_X
> and or sip_h_P, but can't seem to get FS to pass the Subject header.
> Any direction would be greatly appreciated.
> 
> Mike
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Brian West
> Sent: Wednesday, December 22, 2010 1:58 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Dropping the Subject: header
> 
> What is the sudden obsession with the subject header?  Looking thru the
> code it appears as if sip_h_X and sip_h_P are the only two things that
> you can do... a patch will have to be made if you wish to set the
> subject on an outbound invite... see sofia_glue.c  in do_invite.
> 
> /b
> 
> On Dec 22, 2010, at 12:50 PM, Kane, Michael (mkane02) wrote:
> 
>> Hey Steve, the Subject is still not regenerated through the switch.  I
> posted a new pastbin.
>> 
>> 14858
>> 
>> Mike
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mkane02 at harris.com  Wed Dec 22 22:40:26 2010
From: mkane02 at harris.com (Kane, Michael (mkane02))
Date: Wed, 22 Dec 2010 14:40:26 -0500
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net><AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net><AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net><018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org><AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net>
	<41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr>
Message-ID: <AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net>

Thanks David, I really appreciate your clarification.

 

 

Brian, please contact me regarding how we can get a patch incorporated into FS to pass the Subject header.  

 

Thanks Mike

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Wednesday, December 22, 2010 2:31 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropping the Subject: header

 

Michael,

 

what Brian meant is that FreeSWICH can only passes the X-something and P-something headers, which are the only allowed ones.

Any system requiring custom headers should implement:

X-Subject

and not

Subject

 

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/> 

 

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.

 





 

Le 22/12/2010 ? 20:09, Kane, Michael (mkane02) a ?crit :





We have a custom app that requires the Subject header with values so our
endpoints can route the audio accordingly.  I'm ok with using sip_h_X
and or sip_h_P, but can't seem to get FS to pass the Subject header.
Any direction would be greatly appreciated.

Mike

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West
Sent: Wednesday, December 22, 2010 1:58 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropping the Subject: header

What is the sudden obsession with the subject header?  Looking thru the
code it appears as if sip_h_X and sip_h_P are the only two things that
you can do... a patch will have to be made if you wish to set the
subject on an outbound invite... see sofia_glue.c  in do_invite.

/b

On Dec 22, 2010, at 12:50 PM, Kane, Michael (mkane02) wrote:




Hey Steve, the Subject is still not regenerated through the switch.  I

posted a new pastbin.



 

	14858

	 

	Mike



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

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

 

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From brian at freeswitch.org  Wed Dec 22 22:42:33 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 22 Dec 2010 13:42:33 -0600
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net><AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net><AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net><018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org><AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net>
	<41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net>
Message-ID: <9D7D01EA-48E7-46C0-80CE-166C460C115F@freeswitch.org>

Please visit jira.freeswitch.org and open a ticket.

/b

On Dec 22, 2010, at 1:40 PM, Kane, Michael (mkane02) wrote:

> Thanks David, I really appreciate your clarification.
>  
>  
> Brian, please contact me regarding how we can get a patch incorporated into FS to pass the Subject header. 
>  
> Thanks Mike
>  
>  

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From mkane02 at harris.com  Wed Dec 22 22:53:35 2010
From: mkane02 at harris.com (Kane, Michael (mkane02))
Date: Wed, 22 Dec 2010 14:53:35 -0500
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <9D7D01EA-48E7-46C0-80CE-166C460C115F@freeswitch.org>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net><AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net><AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net><018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org><AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net><41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr><AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net>
	<9D7D01EA-48E7-46C0-80CE-166C460C115F@freeswitch.org>
Message-ID: <AF824FE3F385E543ACA98ADFD670BA6A0184D005@mlbe2k9.cs.myharris.net>

Thank Brian.

 

Mike

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West
Sent: Wednesday, December 22, 2010 2:43 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropping the Subject: header

 

Please visit jira.freeswitch.org and open a ticket.

 

/b

 

On Dec 22, 2010, at 1:40 PM, Kane, Michael (mkane02) wrote:





Thanks David, I really appreciate your clarification.

 

 

Brian, please contact me regarding how we can get a patch incorporated
into FS to pass the Subject header. 

 

Thanks Mike

 

 

 

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From wagnerspi at gmail.com  Wed Dec 22 22:35:06 2010
From: wagnerspi at gmail.com (Wagner)
Date: Wed, 22 Dec 2010 17:35:06 -0200
Subject: [Freeswitch-users] Using auto-record
Message-ID: <AANLkTi=N_ng-aUgs+RJ2CecFhaRa-zUkPz4W+cfAkHU=@mail.gmail.com>

Hello,

I'm quite new to FreeSwitch and so far I'm liking it

I have 2 questions about FS

first: If I start as a daemon using the -nc command, how do I access the
console again?

Second: I have a conference that I want to record, I'm trying to use
auto-record but so far I'm not getting it to work

my public.xml looks like that:

<extension name="conferences">
      <condition field="destination_number" expression="^(7\d{4})$">
          <action application="answer"/>
          <action application="conference" data="$1 at wideband"/>
  <profiles>
    <profile name="default">
      <action applicatioon="conference" data="auto-record
/tmp/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
    </profile>
  </profiles>

          <!-- <action application="conference" data="$1 at wideband"/> -->
      </condition>
    </extension>


I got no error on freeswitch.log

but the file it's not created =/

I'm trying with more than one user, as I saw that it only start to record
with 2 or more users

Thanks for the help
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From brian at freeswitch.org  Thu Dec 23 00:14:04 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 22 Dec 2010 15:14:04 -0600
Subject: [Freeswitch-users] Using auto-record
In-Reply-To: <AANLkTi=N_ng-aUgs+RJ2CecFhaRa-zUkPz4W+cfAkHU=@mail.gmail.com>
References: <AANLkTi=N_ng-aUgs+RJ2CecFhaRa-zUkPz4W+cfAkHU=@mail.gmail.com>
Message-ID: <22FC7BC4-F2E6-40FC-AC18-48C59BA3CA94@freeswitch.org>

auto-record is a conference profile param that you set in conference.conf.xml and NOT in your dialplan... You seem to have confused the two options some how.

/b

On Dec 22, 2010, at 1:35 PM, Wagner wrote:

> Hello, 
> 
> I'm quite new to FreeSwitch and so far I'm liking it
> 
> I have 2 questions about FS
> 
> first: If I start as a daemon using the -nc command, how do I access the console again?
> 
> Second: I have a conference that I want to record, I'm trying to use auto-record but so far I'm not getting it to work
> 
> my public.xml looks like that:
> 
> <extension name="conferences">
>      <condition field="destination_number" expression="^(7\d{4})$">
>          <action application="answer"/>
>          <action application="conference" data="$1 at wideband"/>
>  <profiles>
>    <profile name="default">
>      <action applicatioon="conference" data="auto-record /tmp/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>    </profile>
>  </profiles>
> 
>          <!-- <action application="conference" data="$1 at wideband"/> -->
>      </condition>
>    </extension>
> 
> 
> I got no error on freeswitch.log
> 
> but the file it's not created =/
> 
> I'm trying with more than one user, as I saw that it only start to record with 2 or more users
> 
> Thanks for the help



From steveayre at gmail.com  Thu Dec 23 01:02:38 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 22 Dec 2010 22:02:38 +0000
Subject: [Freeswitch-users] Using auto-record
In-Reply-To: <AANLkTi=N_ng-aUgs+RJ2CecFhaRa-zUkPz4W+cfAkHU=@mail.gmail.com>
References: <AANLkTi=N_ng-aUgs+RJ2CecFhaRa-zUkPz4W+cfAkHU=@mail.gmail.com>
Message-ID: <AANLkTinmkKKjJVjh5-Y0JtLQ8Z63LYrKiE0GaJ3YUdUO@mail.gmail.com>

> first: If I start as a daemon using the -nc command, how do I access the
> console again?

fs_cli - it connects to the event socket (check mod_event_socket is
loaded and configure it in conf/autoload_config/event_socket.conf.xml,
where the listening ip and password is set)

> <extension name="conferences">
> ????? <condition field="destination_number" expression="^(7\d{4})$">
> ????????? <action application="answer"/>
> ????????? <action application="conference" data="$1 at wideband"/>
> ? <profiles>
> ??? <profile name="default">
> ??? ? <action applicatioon="conference" data="auto-record
> /tmp/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
> ??? </profile>
> ? </profiles>

Profiles are configured in conf/autoload_config/conference.conf.xml,
not the dialplan. Also, it will be a <param> not an <action>. For
example:

 <configuration name="conference.conf" description="Audio Conference">
  <profiles>
   <profile name="default">
    <param name="auto-record"
value="/tmp/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
   </profile>
  </profiles>
  ... (rest of the file) ...
 </configuration>

You won't have seen an error because the <profiles> tag is completely
unknown to the dialplan, so it would have been ignored.

-Steve



On 22 December 2010 19:35, Wagner <wagnerspi at gmail.com> wrote:
> Hello,
>
> I'm quite new to FreeSwitch and so far I'm liking it
>
> I have 2 questions about FS
>
> first: If I start as a daemon using the -nc command, how do I access the
> console again?
>
> Second: I have a conference that I want to record, I'm trying to use
> auto-record but so far I'm not getting it to work
>
> my public.xml looks like that:
>
> <extension name="conferences">
> ????? <condition field="destination_number" expression="^(7\d{4})$">
> ????????? <action application="answer"/>
> ????????? <action application="conference" data="$1 at wideband"/>
> ? <profiles>
> ??? <profile name="default">
> ??? ? <action applicatioon="conference" data="auto-record
> /tmp/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
> ??? </profile>
> ? </profiles>
>
> ????????? <!-- <action application="conference" data="$1 at wideband"/> -->
> ????? </condition>
> ??? </extension>
>
>
> I got no error on freeswitch.log
>
> but the file it's not created =/
>
> I'm trying with more than one user, as I saw that it only start to record
> with 2 or more users
>
> Thanks for the help
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From steveayre at gmail.com  Thu Dec 23 01:04:11 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 22 Dec 2010 22:04:11 +0000
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net>
	<AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net>
	<AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net>
	<018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net>
	<41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net>
Message-ID: <AANLkTikD8J_-EgtrheOfqNWOgDMpSYr+sGDkBw6-w_Pp@mail.gmail.com>

David,

Change your custom app to use X-Subject not Subject (AFAIK the RFC
says any custom headers should be prefixed X-). My previous example
will then work if you export sip_h_X-Subject.

-Steve


On 22 December 2010 19:40, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
> Thanks David, I really appreciate your clarification.
>
>
>
>
>
> Brian, please contact me regarding how we can get a patch incorporated into
> FS to pass the Subject header.
>
>
>
> Thanks Mike
>
>
>
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David
> Ponzone
> Sent: Wednesday, December 22, 2010 2:31 PM
>
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>
>
>
> Michael,
>
>
>
> what Brian meant is that FreeSWICH can only passes the X-something and
> P-something headers, which are the only allowed ones.
>
> Any system requiring custom headers should implement:
>
> X-Subject
>
> and not
>
> Subject
>
>
>
>
>
> David Ponzone ?Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel: ? ? ?01 74 03 18 97
>
> gsm: ? 06 66 98 76 34
>
>
>
> Service Client?IPeva
>
> tel: ? ? ?0811 46 26 26
>
> www.ipeva.fr? -? ?www.ipeva-studio.com
>
>
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
>
> Le 22/12/2010 ? 20:09, Kane, Michael (mkane02) a ?crit :
>
> We have a custom app that requires the Subject header with values so our
> endpoints can route the audio accordingly. ?I'm ok with using sip_h_X
> and or sip_h_P, but can't seem to get FS to pass the Subject header.
> Any direction would be greatly appreciated.
>
> Mike
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Brian West
> Sent: Wednesday, December 22, 2010 1:58 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>
> What is the sudden obsession with the subject header? ?Looking thru the
> code it appears as if sip_h_X and sip_h_P are the only two things that
> you can do... a patch will have to be made if you wish to set the
> subject on an outbound invite... see sofia_glue.c ?in do_invite.
>
> /b
>
> On Dec 22, 2010, at 12:50 PM, Kane, Michael (mkane02) wrote:
>
>
> Hey Steve, the Subject is still not regenerated through the switch. ?I
>
> posted a new pastbin.
>
>
>
> 14858
>
>
>
> Mike
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From me at nevian.org  Thu Dec 23 01:09:55 2010
From: me at nevian.org (Serge S. Yuriev)
Date: Thu, 23 Dec 2010 01:09:55 +0300
Subject: [Freeswitch-users] Asynchronous PTIME
Message-ID: <4D127733.8010409@nevian.org>

Hello,

2010-12-20 16:51:13.960502 [WARNING] mod_sofia.c:1036 Asynchronous PTIME 
not supported, changing our end from 20 to 60

I'm getting this warning and client hears chopped sound :(
That is "Our end"?

Call flow is Cisco (sip) FS (sip) YATE (h323) MeraTransitSoftSwitch
All but MVTS under my control.

I doesn't see any clue in logs and can't reproduce this with my testing 
via Blink/PortSIP/PhonerLite/Twinkle or Grandstream devices

Which debug/logs I should take? Any ideas?

Thanks a lot.

btw how I can save debug into log not only console?
-- 
wbr,
Serge


From jeff at jefflenk.com  Thu Dec 23 01:16:25 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 22 Dec 2010 16:16:25 -0600
Subject: [Freeswitch-users] compile error on libmp3lame and mod_shout
Message-ID: <snt0-eas108A42049A99CF0143FB85BB91B0@phx.gbl>

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From rex.alex345 at yahoo.com  Thu Dec 23 01:43:41 2010
From: rex.alex345 at yahoo.com (Rex Alex)
Date: Wed, 22 Dec 2010 14:43:41 -0800 (PST)
Subject: [Freeswitch-users] (no subject)
Message-ID: <731680.56494.qm@web59513.mail.ac4.yahoo.com>

http://rogycomo.110mb.com/qigalaso.html
Oper a teWithY ourComm ents Onl in e 


      


From wagnerspi at gmail.com  Thu Dec 23 00:30:54 2010
From: wagnerspi at gmail.com (Wagner)
Date: Wed, 22 Dec 2010 19:30:54 -0200
Subject: [Freeswitch-users] Using auto-record
In-Reply-To: <22FC7BC4-F2E6-40FC-AC18-48C59BA3CA94@freeswitch.org>
References: <AANLkTi=N_ng-aUgs+RJ2CecFhaRa-zUkPz4W+cfAkHU=@mail.gmail.com>
	<22FC7BC4-F2E6-40FC-AC18-48C59BA3CA94@freeswitch.org>
Message-ID: <AANLkTin7X-g5FuMVr7Zw=uo4CGtfxfcWNcspANTRM6-L@mail.gmail.com>

Thanks a lot Brian,

got it working

and what about accessing the console after it's on background?

Thanks

2010/12/22 Brian West <brian at freeswitch.org>

> auto-record is a conference profile param that you set in
> conference.conf.xml and NOT in your dialplan... You seem to have confused
> the two options some how.
>
> /b
>
> On Dec 22, 2010, at 1:35 PM, Wagner wrote:
>
> > Hello,
> >
> > I'm quite new to FreeSwitch and so far I'm liking it
> >
> > I have 2 questions about FS
> >
> > first: If I start as a daemon using the -nc command, how do I access the
> console again?
> >
> > Second: I have a conference that I want to record, I'm trying to use
> auto-record but so far I'm not getting it to work
> >
> > my public.xml looks like that:
> >
> > <extension name="conferences">
> >      <condition field="destination_number" expression="^(7\d{4})$">
> >          <action application="answer"/>
> >          <action application="conference" data="$1 at wideband"/>
> >  <profiles>
> >    <profile name="default">
> >      <action applicatioon="conference" data="auto-record
> /tmp/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
> >    </profile>
> >  </profiles>
> >
> >          <!-- <action application="conference" data="$1 at wideband"/> -->
> >      </condition>
> >    </extension>
> >
> >
> > I got no error on freeswitch.log
> >
> > but the file it's not created =/
> >
> > I'm trying with more than one user, as I saw that it only start to record
> with 2 or more users
> >
> > Thanks for the help
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Thu Dec 23 02:16:14 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 22 Dec 2010 17:16:14 -0600
Subject: [Freeswitch-users] Using auto-record
In-Reply-To: <AANLkTin7X-g5FuMVr7Zw=uo4CGtfxfcWNcspANTRM6-L@mail.gmail.com>
References: <AANLkTi=N_ng-aUgs+RJ2CecFhaRa-zUkPz4W+cfAkHU=@mail.gmail.com>
	<22FC7BC4-F2E6-40FC-AC18-48C59BA3CA94@freeswitch.org>
	<AANLkTin7X-g5FuMVr7Zw=uo4CGtfxfcWNcspANTRM6-L@mail.gmail.com>
Message-ID: <3F3F52EE-9BAE-4B95-816A-3B44E9519C5A@freeswitch.org>

install fs_cli from build dir

/b

On Dec 22, 2010, at 3:30 PM, Wagner wrote:

> and what about accessing the console after it's on background?



From msc at freeswitch.org  Thu Dec 23 03:05:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Dec 2010 16:05:16 -0800
Subject: [Freeswitch-users] bind_meta_app and bind_digit_action with
 inband DTMF?
In-Reply-To: <AANLkTimiDuz8iHLkWL2Z1osyAz1pFQr3jV-hvXw=iGAR@mail.gmail.com>
References: <AANLkTimiDuz8iHLkWL2Z1osyAz1pFQr3jV-hvXw=iGAR@mail.gmail.com>
Message-ID: <AANLkTik1Fuds_ephKqeOQfwi-pB=PpCR=wFnEvdEkiHo@mail.gmail.com>

FYI,

I just tested bind_digit_action with inband dtmf and it works splendidly. I
recommend using bind_digit_action. Also, in case you didn't see it, Brian
West added a magic extension that looks for the presence of
"telephone-event" in the SDP and then turns on inband dtmf detection if the
rtpmap is missing. Look in conf/dialplan/default.xml for the "global"
extension. The lines that do the auto dtmf thing are commented out by
default. I just uncommented them and tested with Eyebeam configured for
inband dtmf. (It also works when Eyebeam is set for 2833...)

Also, on the FS conf call today the community did a tag-team effort to test
and document the bind_digit_action app. It has a lot more information than
it did previously:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_digit_action

Hope this helps.
-MC

On Wed, Dec 22, 2010 at 8:01 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> Hi! We are trying to migrate our Asterisk system to Freeswitch, and
> faced with some trouble:
>
> We use inband at user side, and want Freeswitch to regenerate it to
> RFC2833 before sending to media gateway.
>
> When user endpoints use rfc2833 bind_meta_app and bind_digit_action
> work as expected, but not with start_dtmf (as needed to parse inband
> DTMF). We can't use rfc2833 DTMF on user endpoints as there seems to
> be a problem in freeswitch or our media gateway leading to audible
> DTMF duplicates.
>
> When user endpoints are set to inband, I see messages on console about
> DTMF detection:
>
> 2010-12-22 18:50:30.802253 [DEBUG] switch_ivr_async.c:2089 DTMF DETECTED:
> [*]
> 2010-12-22 18:50:30.803253 [DEBUG] switch_rtp.c:1985 Send start packet
> for [*] ts=503795131 dur=160/160/2000 seq=43701
> 2010-12-22 18:50:30.823282 [DEBUG] switch_rtp.c:1921 Send middle
> packet for [*] ts=503795131 dur=320/320/2000 seq=43702
> ...skip...
> 2010-12-22 18:50:30.923393 [DEBUG] switch_rtp.c:1921 Send middle
> packet for [*] ts=503795131 dur=1920/1920/2000 seq=43712
> 2010-12-22 18:50:30.923393 [DEBUG] switch_rtp.c:1921 Send end packet
> for [*] ts=503795131 dur=2080/2080/2000 seq=43713
> ...skip...
> 2010-12-22 18:50:31.242695 [DEBUG] switch_ivr_async.c:2089 DTMF DETECTED:
> [1]
> 2010-12-22 18:50:31.242695 [DEBUG] switch_rtp.c:1985 Send start packet
> for [1] ts=503798651 dur=160/160/2000 seq=43732
>
> So digits are received but no in-call action launched by
> bind_meta_app. I tried start_dtmf before and after bind_meta_app but
> with no success. Is there a way to make inband dtmf launch in-call
> actions while transcoding it to RFC2833 on trunk side?
> I can't make inband dtmf trigger my actions :(
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Dec 23 03:07:22 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Dec 2010 16:07:22 -0800
Subject: [Freeswitch-users] call dropped while trying to transfer
In-Reply-To: <AANLkTiku8ZRDxFxQECys14BW94X2J28UpxKZu+p_BOXq@mail.gmail.com>
References: <AANLkTiku8ZRDxFxQECys14BW94X2J28UpxKZu+p_BOXq@mail.gmail.com>
Message-ID: <AANLkTin0Lmz64n+chbcQ6UqYY704bcim+yOf5AOcV6sv@mail.gmail.com>

Please pastebin the code that performs the uuid_deflect so that we can see
what you are doing to produce this symptom.
-MC

On Wed, Dec 22, 2010 at 8:35 AM, George Niculae <george at ezuce.com> wrote:

> Hi All,
>
> I am working on an IVR application based on FS (running FreeSWITCH
> Version 1.0.head (git-43393f2 2010-12-15 20-59-42 -0600) where the
> following scenario fails:
> user 201 calls to 100 (autoattendant), hears menu then press # to
> transfer to voicemail (101), but the call is dropped (transfer is made
> using uuid_deflect api command)
> Dialplan extension configured like:
>
>  <extension name="IVR">
>    <condition field="destination_number" expression="^IVR${dollar}">
>      <action application="socket" data="192.168.0.114:8084 async full"/>
>    </condition>
>  </extension>
>
> Actions taken are:
> - when call arrives to extension 100 call is bridged
> (hangup_after_bridge=true)
> - answer the call, autoattendant menu is played and DTMF collected
> - when # pressed, call is transfered to 101 using uuid_deflect
> - call arrives to voicemail extension and is again bridged
> - call is answered - at this point in time the initial bridge hangs up
> and the whole call is dropped
> Please see console output http://pastebin.freeswitch.org/14855
>
> When debugging the application, If I keep the first channel connected
> transfer works just fine without dropping the call.
> Pretty sure I'm missing something here, any suggestion highly appreciated
>
> Thanks,
> George
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mel0torme at gmail.com  Thu Dec 23 09:47:32 2010
From: mel0torme at gmail.com (Tom C)
Date: Wed, 22 Dec 2010 22:47:32 -0800
Subject: [Freeswitch-users] Can't get demo_ivr to work - just goes to
	Music-on-hold
Message-ID: <AANLkTimTX_DMzJK97ACJ_9Xmf4BQu4-6f3f4QmVfiVwE@mail.gmail.com>

When I dial extension 6000 to use the demo_ivr, I hear the "please dial it
now" message.  But no matter what number I dial, it just sends me to Music
On Hold.

I've tried this on my Dockstar (debian squeeze) and my P4 (debian lenny)
with the same results.

I want my screaming monkeys!!!  :-)   What am I doing wrong?  Is there
another module I need to load?  I had this working a week ago.

module_exists mod_flite
true

Here's the debug log after it plays the initial greeting and I start to
dial:
2010-12-22 22:33:13.592915 [DEBUG] switch_rtp.c:3018 RTP RECV DTMF 5:480
2010-12-22 22:33:14.272565 [DEBUG] switch_rtp.c:3018 RTP RECV DTMF #:480
2010-12-22 22:33:14.272565 [DEBUG] mod_local_stream.c:421 Opening Stream
[moh/8000] 8000hz
2010-12-22 22:33:14.272565 [DEBUG] switch_ivr_play_say.c:1236 Codec
Activated L16 at 8000hz 1 channels 20ms

I've tried with internally registered clients as well as an external SIP
call.  I've re-copied the demo_ivr.xml from the recently git-pulled source
directory, no help.
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From u2nsam at gmail.com  Thu Dec 23 12:20:44 2010
From: u2nsam at gmail.com (Sam)
Date: Thu, 23 Dec 2010 14:50:44 +0530
Subject: [Freeswitch-users] no invite send
Message-ID: <AANLkTimBh+CqTz1O5qNmd7=z+Z6WZGeY1ua8kdZAwzH5@mail.gmail.com>

Hello friends,

I am using the dialplan,

<action application="bridge" data="sofia/external/12345 at 192.168.2.3"/>

its not sending INVITE to the server 192.168.2.3

i am getting the below logs:-

2010-12-23 14:03:02.509500 [NOTICE] switch_channel.c:784 New Channel
sofia/external/12345 at 192.168.2.3 [418e0dfa-0e6f-11e0-a5e1-33ac6c473a4e]
2010-12-23 14:03:02.509500 [DEBUG] mod_sofia.c:4052 (sofia/external/
12345 at 192.168.2.3) State Change CS_NEW -> CS_INIT
2010-12-23 14:03:02.509500 [DEBUG] switch_core_session.c:1083 Send signal
sofia/external/12345 at 192.168.2.3 [BREAK]
2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
(sofia/external/12345 at 192.168.2.3) Running State Change CS_INIT
2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
(sofia/external/12345 at 192.168.2.3) State INIT
2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:86 sofia/external/
12345 at 192.168.2.3 SOFIA INIT
2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:126 (sofia/external/
12345 at 192.168.2.3) State Change CS_INIT -> CS_ROUTING
2010-12-23 14:03:02.510864 [DEBUG] switch_core_session.c:1083 Send signal
sofia/external/12345 at 192.168.2.3 [BREAK]
2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
(sofia/external/12345 at 192.168.2.3) State INIT going to sleep
2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
(sofia/external/12345 at 192.168.2.3) Running State Change CS_ROUTING
2010-12-23 14:03:02.510864 [DEBUG] switch_channel.c:1615 (sofia/external/
12345 at 192.168.2.3) Callstate Change DOWN -> RINGING
2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
(sofia/external/12345 at 192.168.2.3) State ROUTING
2010-12-23 14:03:02.511888 [DEBUG] mod_sofia.c:149 sofia/external/
12345 at 192.168.2.3 SOFIA ROUTING
2010-12-23 14:03:02.511888 [DEBUG] switch_ivr_originate.c:66
(sofia/external/12345 at 192.168.2.3) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2010-12-23 14:03:02.511888 [DEBUG] switch_core_session.c:1083 Send signal
sofia/external/12345 at 192.168.2.3 [BREAK]
2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
(sofia/external/12345 at 192.168.2.3) State ROUTING going to sleep
2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:320
(sofia/external/12345 at 192.168.2.3) Running State Change CS_CONSUME_MEDIA
2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:378
(sofia/external/12345 at 192.168.2.3) State CONSUME_MEDIA
2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel sofia/external/
12345 at 192.168.2.3 entering state [calling][0]
2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel sofia/external/
12345 at 192.168.2.3 entering state [terminated][503]
2010-12-23 14:03:02.511888 [DEBUG] switch_channel.c:2493 (sofia/external/
12345 at 192.168.2.3) Callstate Change RINGING -> HANGUP
2010-12-23 14:03:02.511888 [NOTICE] sofia.c:5246 Hangup sofia/external/
12345 at 192.168.2.3 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]


Regards
Sam
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From steveayre at gmail.com  Thu Dec 23 13:19:20 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 23 Dec 2010 10:19:20 +0000
Subject: [Freeswitch-users] no invite send
In-Reply-To: <AANLkTimBh+CqTz1O5qNmd7=z+Z6WZGeY1ua8kdZAwzH5@mail.gmail.com>
References: <AANLkTimBh+CqTz1O5qNmd7=z+Z6WZGeY1ua8kdZAwzH5@mail.gmail.com>
Message-ID: <AANLkTikpL3pfFieW3Yv20iFxDo+yNdYA9zoyd3tgT-iK@mail.gmail.com>

Try enabling siptrace (sofia profile external siptrace on) - does it
show an INVITE being sent in the logs?

-Steve


On 23 December 2010 09:20, Sam <u2nsam at gmail.com> wrote:
> Hello friends,
>
> I am using the dialplan,
>
> <action application="bridge" data="sofia/external/12345 at 192.168.2.3"/>
>
> its not sending INVITE to the server 192.168.2.3
>
> i am getting the below logs:-
>
> 2010-12-23 14:03:02.509500 [NOTICE] switch_channel.c:784 New Channel
> sofia/external/12345 at 192.168.2.3 [418e0dfa-0e6f-11e0-a5e1-33ac6c473a4e]
> 2010-12-23 14:03:02.509500 [DEBUG] mod_sofia.c:4052
> (sofia/external/12345 at 192.168.2.3) State Change CS_NEW -> CS_INIT
> 2010-12-23 14:03:02.509500 [DEBUG] switch_core_session.c:1083 Send signal
> sofia/external/12345 at 192.168.2.3 [BREAK]
> 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/12345 at 192.168.2.3) Running State Change CS_INIT
> 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/12345 at 192.168.2.3) State INIT
> 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:86
> sofia/external/12345 at 192.168.2.3 SOFIA INIT
> 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:126
> (sofia/external/12345 at 192.168.2.3) State Change CS_INIT -> CS_ROUTING
> 2010-12-23 14:03:02.510864 [DEBUG] switch_core_session.c:1083 Send signal
> sofia/external/12345 at 192.168.2.3 [BREAK]
> 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/12345 at 192.168.2.3) State INIT going to sleep
> 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/12345 at 192.168.2.3) Running State Change CS_ROUTING
> 2010-12-23 14:03:02.510864 [DEBUG] switch_channel.c:1615
> (sofia/external/12345 at 192.168.2.3) Callstate Change DOWN -> RINGING
> 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
> (sofia/external/12345 at 192.168.2.3) State ROUTING
> 2010-12-23 14:03:02.511888 [DEBUG] mod_sofia.c:149
> sofia/external/12345 at 192.168.2.3 SOFIA ROUTING
> 2010-12-23 14:03:02.511888 [DEBUG] switch_ivr_originate.c:66
> (sofia/external/12345 at 192.168.2.3) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2010-12-23 14:03:02.511888 [DEBUG] switch_core_session.c:1083 Send signal
> sofia/external/12345 at 192.168.2.3 [BREAK]
> 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
> (sofia/external/12345 at 192.168.2.3) State ROUTING going to sleep
> 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/12345 at 192.168.2.3) Running State Change CS_CONSUME_MEDIA
> 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:378
> (sofia/external/12345 at 192.168.2.3) State CONSUME_MEDIA
> 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
> sofia/external/12345 at 192.168.2.3 entering state [calling][0]
> 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
> sofia/external/12345 at 192.168.2.3 entering state [terminated][503]
> 2010-12-23 14:03:02.511888 [DEBUG] switch_channel.c:2493
> (sofia/external/12345 at 192.168.2.3) Callstate Change RINGING -> HANGUP
> 2010-12-23 14:03:02.511888 [NOTICE] sofia.c:5246 Hangup
> sofia/external/12345 at 192.168.2.3 [CS_CONSUME_MEDIA]
> [NORMAL_TEMPORARY_FAILURE]
>
>
> Regards
> Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From u2nsam at gmail.com  Thu Dec 23 13:28:42 2010
From: u2nsam at gmail.com (Sam)
Date: Thu, 23 Dec 2010 15:58:42 +0530
Subject: [Freeswitch-users] no invite send
In-Reply-To: <AANLkTikpL3pfFieW3Yv20iFxDo+yNdYA9zoyd3tgT-iK@mail.gmail.com>
References: <AANLkTimBh+CqTz1O5qNmd7=z+Z6WZGeY1ua8kdZAwzH5@mail.gmail.com>
	<AANLkTikpL3pfFieW3Yv20iFxDo+yNdYA9zoyd3tgT-iK@mail.gmail.com>
Message-ID: <AANLkTim0hhJBaJSg9ghh8cwrbSY8_rC+4oOP79vT-Hw4@mail.gmail.com>

I check it in sip trace ... it do not sends invite, probably i might have
miss configured something,
on the another FS server it works fine, this happened on the new
installation where i was using prefix,

I was trying to reinstall FS on the new server with prefix it gave me below
error.

./configure: line 38199: 30565 Segmentation fault      /bin/sh
/usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat/configure
'--prefix=/usr/local/conference/' 'CONFIGURE_CFLAGS=-g -O2'
'CONFIGURE_CXXFLAGS=-g -O2' 'CONFIGURE_LDFLAGS=' '--cache-file=/dev/null'
'--srcdir=.' '--with-apr=../apr' '--disable-shared' '--with-pic'
'--without-sqlite2' '--without-sqlite3' '--with-expat=builtin'
--cache-file=/dev/null
--srcdir=/usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat
--prefix=/usr/local/conference/ --exec-prefix=${prefix}
--libdir=${exec_prefix}/lib
--includedir=${prefix}/include/apr-${APRUTIL_MAJOR_VERSION}
--bindir=${exec_prefix}/bin
configure failed for xml/expat
configure: error: /bin/sh './configure.gnu' failed for libs/apr-util



i was using ./configure --prefix=/usr/local/conference/



Regards
Sam



On Thu, Dec 23, 2010 at 3:49 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Try enabling siptrace (sofia profile external siptrace on) - does it
> show an INVITE being sent in the logs?
>
> -Steve
>
>
> On 23 December 2010 09:20, Sam <u2nsam at gmail.com> wrote:
> > Hello friends,
> >
> > I am using the dialplan,
> >
> > <action application="bridge" data="sofia/external/12345 at 192.168.2.3"/>
> >
> > its not sending INVITE to the server 192.168.2.3
> >
> > i am getting the below logs:-
> >
> > 2010-12-23 14:03:02.509500 [NOTICE] switch_channel.c:784 New Channel
> > sofia/external/12345 at 192.168.2.3 [418e0dfa-0e6f-11e0-a5e1-33ac6c473a4e]
> > 2010-12-23 14:03:02.509500 [DEBUG] mod_sofia.c:4052
> > (sofia/external/12345 at 192.168.2.3) State Change CS_NEW -> CS_INIT
> > 2010-12-23 14:03:02.509500 [DEBUG] switch_core_session.c:1083 Send signal
> > sofia/external/12345 at 192.168.2.3 [BREAK]
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
> > (sofia/external/12345 at 192.168.2.3) Running State Change CS_INIT
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
> > (sofia/external/12345 at 192.168.2.3) State INIT
> > 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:86
> > sofia/external/12345 at 192.168.2.3 SOFIA INIT
> > 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:126
> > (sofia/external/12345 at 192.168.2.3) State Change CS_INIT -> CS_ROUTING
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_session.c:1083 Send signal
> > sofia/external/12345 at 192.168.2.3 [BREAK]
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
> > (sofia/external/12345 at 192.168.2.3) State INIT going to sleep
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
> > (sofia/external/12345 at 192.168.2.3) Running State Change CS_ROUTING
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_channel.c:1615
> > (sofia/external/12345 at 192.168.2.3) Callstate Change DOWN -> RINGING
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
> > (sofia/external/12345 at 192.168.2.3) State ROUTING
> > 2010-12-23 14:03:02.511888 [DEBUG] mod_sofia.c:149
> > sofia/external/12345 at 192.168.2.3 SOFIA ROUTING
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_ivr_originate.c:66
> > (sofia/external/12345 at 192.168.2.3) State Change CS_ROUTING ->
> > CS_CONSUME_MEDIA
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_session.c:1083 Send signal
> > sofia/external/12345 at 192.168.2.3 [BREAK]
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
> > (sofia/external/12345 at 192.168.2.3) State ROUTING going to sleep
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:320
> > (sofia/external/12345 at 192.168.2.3) Running State Change CS_CONSUME_MEDIA
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:378
> > (sofia/external/12345 at 192.168.2.3) State CONSUME_MEDIA
> > 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
> > sofia/external/12345 at 192.168.2.3 entering state [calling][0]
> > 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
> > sofia/external/12345 at 192.168.2.3 entering state [terminated][503]
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_channel.c:2493
> > (sofia/external/12345 at 192.168.2.3) Callstate Change RINGING -> HANGUP
> > 2010-12-23 14:03:02.511888 [NOTICE] sofia.c:5246 Hangup
> > sofia/external/12345 at 192.168.2.3 [CS_CONSUME_MEDIA]
> > [NORMAL_TEMPORARY_FAILURE]
> >
> >
> > Regards
> > Sam
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Thu Dec 23 13:55:25 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 23 Dec 2010 10:55:25 +0000
Subject: [Freeswitch-users] no invite send
In-Reply-To: <AANLkTim0hhJBaJSg9ghh8cwrbSY8_rC+4oOP79vT-Hw4@mail.gmail.com>
References: <AANLkTimBh+CqTz1O5qNmd7=z+Z6WZGeY1ua8kdZAwzH5@mail.gmail.com>
	<AANLkTikpL3pfFieW3Yv20iFxDo+yNdYA9zoyd3tgT-iK@mail.gmail.com>
	<AANLkTim0hhJBaJSg9ghh8cwrbSY8_rC+4oOP79vT-Hw4@mail.gmail.com>
Message-ID: <AANLkTim=7KA-StOrjc3h7FsetycG38MQ5UvryCPJDe55@mail.gmail.com>

In that case the 503 is probably being generated due to an error
within the Sofia-SIP stack.

Run this at the console:
sofia loglevel all 9

You'll see some low level logging information from within the stack,
which should show what causes the 503.

-Steve


On 23 December 2010 10:28, Sam <u2nsam at gmail.com> wrote:
> I check it in sip trace ... it do not sends invite, probably i might have
> miss configured something,
> on the another FS server it works fine, this happened on the new
> installation where i was using prefix,
>
> I was trying to reinstall FS on the new server with prefix it gave me below
> error.
>
> ./configure: line 38199: 30565 Segmentation fault????? /bin/sh
> /usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat/configure
> '--prefix=/usr/local/conference/' 'CONFIGURE_CFLAGS=-g -O2'
> 'CONFIGURE_CXXFLAGS=-g -O2' 'CONFIGURE_LDFLAGS=' '--cache-file=/dev/null'
> '--srcdir=.' '--with-apr=../apr' '--disable-shared' '--with-pic'
> '--without-sqlite2' '--without-sqlite3' '--with-expat=builtin'
> --cache-file=/dev/null
> --srcdir=/usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat
> --prefix=/usr/local/conference/ --exec-prefix=${prefix}
> --libdir=${exec_prefix}/lib
> --includedir=${prefix}/include/apr-${APRUTIL_MAJOR_VERSION}
> --bindir=${exec_prefix}/bin
> configure failed for xml/expat
> configure: error: /bin/sh './configure.gnu' failed for libs/apr-util
>
>
>
> i was using ./configure --prefix=/usr/local/conference/
>
>
>
> Regards
> Sam
>
>
>
> On Thu, Dec 23, 2010 at 3:49 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> Try enabling siptrace (sofia profile external siptrace on) - does it
>> show an INVITE being sent in the logs?
>>
>> -Steve
>>
>>
>> On 23 December 2010 09:20, Sam <u2nsam at gmail.com> wrote:
>> > Hello friends,
>> >
>> > I am using the dialplan,
>> >
>> > <action application="bridge" data="sofia/external/12345 at 192.168.2.3"/>
>> >
>> > its not sending INVITE to the server 192.168.2.3
>> >
>> > i am getting the below logs:-
>> >
>> > 2010-12-23 14:03:02.509500 [NOTICE] switch_channel.c:784 New Channel
>> > sofia/external/12345 at 192.168.2.3 [418e0dfa-0e6f-11e0-a5e1-33ac6c473a4e]
>> > 2010-12-23 14:03:02.509500 [DEBUG] mod_sofia.c:4052
>> > (sofia/external/12345 at 192.168.2.3) State Change CS_NEW -> CS_INIT
>> > 2010-12-23 14:03:02.509500 [DEBUG] switch_core_session.c:1083 Send
>> > signal
>> > sofia/external/12345 at 192.168.2.3 [BREAK]
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
>> > (sofia/external/12345 at 192.168.2.3) Running State Change CS_INIT
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
>> > (sofia/external/12345 at 192.168.2.3) State INIT
>> > 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:86
>> > sofia/external/12345 at 192.168.2.3 SOFIA INIT
>> > 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:126
>> > (sofia/external/12345 at 192.168.2.3) State Change CS_INIT -> CS_ROUTING
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_session.c:1083 Send
>> > signal
>> > sofia/external/12345 at 192.168.2.3 [BREAK]
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
>> > (sofia/external/12345 at 192.168.2.3) State INIT going to sleep
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
>> > (sofia/external/12345 at 192.168.2.3) Running State Change CS_ROUTING
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_channel.c:1615
>> > (sofia/external/12345 at 192.168.2.3) Callstate Change DOWN -> RINGING
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
>> > (sofia/external/12345 at 192.168.2.3) State ROUTING
>> > 2010-12-23 14:03:02.511888 [DEBUG] mod_sofia.c:149
>> > sofia/external/12345 at 192.168.2.3 SOFIA ROUTING
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_ivr_originate.c:66
>> > (sofia/external/12345 at 192.168.2.3) State Change CS_ROUTING ->
>> > CS_CONSUME_MEDIA
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_session.c:1083 Send
>> > signal
>> > sofia/external/12345 at 192.168.2.3 [BREAK]
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
>> > (sofia/external/12345 at 192.168.2.3) State ROUTING going to sleep
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:320
>> > (sofia/external/12345 at 192.168.2.3) Running State Change CS_CONSUME_MEDIA
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:378
>> > (sofia/external/12345 at 192.168.2.3) State CONSUME_MEDIA
>> > 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
>> > sofia/external/12345 at 192.168.2.3 entering state [calling][0]
>> > 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
>> > sofia/external/12345 at 192.168.2.3 entering state [terminated][503]
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_channel.c:2493
>> > (sofia/external/12345 at 192.168.2.3) Callstate Change RINGING -> HANGUP
>> > 2010-12-23 14:03:02.511888 [NOTICE] sofia.c:5246 Hangup
>> > sofia/external/12345 at 192.168.2.3 [CS_CONSUME_MEDIA]
>> > [NORMAL_TEMPORARY_FAILURE]
>> >
>> >
>> > Regards
>> > Sam
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From a.afzali2003 at gmail.com  Thu Dec 23 14:19:51 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Thu, 23 Dec 2010 14:49:51 +0330
Subject: [Freeswitch-users] Loosing DTMF Digits in XML IVR
Message-ID: <AANLkTim98bvSVJQWdKFcvTsZk8uJd3OsPZvK-omLTh4=@mail.gmail.com>

Hi Guys,

After updating to latest freeswitch (git-34a0ca5 2010-12-22 20-38-57 -0600)
, I've noticed that I'm loosing some of the dtmf digits in my IVR. Actually
freeswitch looses most of them! My freeswitch hosted by ubuntu 10.04 64bit
Server.

Appreciate all comments,

Regards,
-- afshin
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From mkane02 at harris.com  Thu Dec 23 16:46:39 2010
From: mkane02 at harris.com (Kane, Michael (mkane02))
Date: Thu, 23 Dec 2010 08:46:39 -0500
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AANLkTikD8J_-EgtrheOfqNWOgDMpSYr+sGDkBw6-w_Pp@mail.gmail.com>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net><AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net><AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net><018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org><AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net><41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr><AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net>
	<AANLkTikD8J_-EgtrheOfqNWOgDMpSYr+sGDkBw6-w_Pp@mail.gmail.com>
Message-ID: <AF824FE3F385E543ACA98ADFD670BA6A0184D009@mlbe2k9.cs.myharris.net>

Hey Steve/David, I'm a bit confused about having to add the X- prefix to the Subject header.  Can you please point me in the direction of which RFC talks to experimental headers?  

Thanks Mike

Ps...thanks for looking into this for me.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Wednesday, December 22, 2010 5:04 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropping the Subject: header

David,

Change your custom app to use X-Subject not Subject (AFAIK the RFC
says any custom headers should be prefixed X-). My previous example
will then work if you export sip_h_X-Subject.

-Steve


On 22 December 2010 19:40, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
> Thanks David, I really appreciate your clarification.
>
>
>
>
>
> Brian, please contact me regarding how we can get a patch incorporated into
> FS to pass the Subject header.
>
>
>
> Thanks Mike
>
>
>
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David
> Ponzone
> Sent: Wednesday, December 22, 2010 2:31 PM
>
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>
>
>
> Michael,
>
>
>
> what Brian meant is that FreeSWICH can only passes the X-something and
> P-something headers, which are the only allowed ones.
>
> Any system requiring custom headers should implement:
>
> X-Subject
>
> and not
>
> Subject
>
>
>
>
>
> David Ponzone ?Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel: ? ? ?01 74 03 18 97
>
> gsm: ? 06 66 98 76 34
>
>
>
> Service Client?IPeva
>
> tel: ? ? ?0811 46 26 26
>
> www.ipeva.fr? -? ?www.ipeva-studio.com
>
>
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
>
> Le 22/12/2010 ? 20:09, Kane, Michael (mkane02) a ?crit :
>
> We have a custom app that requires the Subject header with values so our
> endpoints can route the audio accordingly. ?I'm ok with using sip_h_X
> and or sip_h_P, but can't seem to get FS to pass the Subject header.
> Any direction would be greatly appreciated.
>
> Mike
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Brian West
> Sent: Wednesday, December 22, 2010 1:58 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>
> What is the sudden obsession with the subject header? ?Looking thru the
> code it appears as if sip_h_X and sip_h_P are the only two things that
> you can do... a patch will have to be made if you wish to set the
> subject on an outbound invite... see sofia_glue.c ?in do_invite.
>
> /b
>
> On Dec 22, 2010, at 12:50 PM, Kane, Michael (mkane02) wrote:
>
>
> Hey Steve, the Subject is still not regenerated through the switch. ?I
>
> posted a new pastbin.
>
>
>
> 14858
>
>
>
> Mike
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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


From steveayre at gmail.com  Thu Dec 23 18:46:07 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 23 Dec 2010 15:46:07 +0000
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AF824FE3F385E543ACA98ADFD670BA6A0184D009@mlbe2k9.cs.myharris.net>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net>
	<AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net>
	<AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net>
	<018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net>
	<41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net>
	<AANLkTikD8J_-EgtrheOfqNWOgDMpSYr+sGDkBw6-w_Pp@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D009@mlbe2k9.cs.myharris.net>
Message-ID: <AANLkTikXa8bXORjRkxgHUcvRBeg7pJ4zLBczK5pOpH_i@mail.gmail.com>

http://www.rfc-editor.org/rfc/rfc3427.txt
The RFC recognises X- has been used in other protocols (eg SMTP) and
says prefix P- should be used for SIP. In practice both P- and X- are
used.

The reason is that if you are setting a custom header, it avoids
conflicts if they revise the SIP specification in future (eg for
SIPv3) and add a header with the same name as yours but a different
purpose.

See also:
http://blogs.voxeo.com/speakingofstandards/2008/05/06/what-is-a-p-header-in-sip-and-whyhow-would-you-use-one/
http://www.the-asterisk-book.com/unstable/applikationen-sipaddheader.html

Note that the X-header functionality is specifically coded within the
Sofia-SIP library:
http://sofia-sip.sourceforge.net/refdocs/sip/group__sip__header__x.html
To handle custom headers not prefixed with X- or P- would involve
considerable rewriting of the library to expose the ability. Since the
library is written by Nokia that would result in a large difference
between the FS and upstream versions, making it hard to maintain.

Regards,
Steve


On 23 December 2010 13:46, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
> Hey Steve/David, I'm a bit confused about having to add the X- prefix to the Subject header. ?Can you please point me in the direction of which RFC talks to experimental headers?
>
> Thanks Mike
>
> Ps...thanks for looking into this for me.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
> Sent: Wednesday, December 22, 2010 5:04 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>
> David,
>
> Change your custom app to use X-Subject not Subject (AFAIK the RFC
> says any custom headers should be prefixed X-). My previous example
> will then work if you export sip_h_X-Subject.
>
> -Steve
>
>
> On 22 December 2010 19:40, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
>> Thanks David, I really appreciate your clarification.
>>
>>
>>
>>
>>
>> Brian, please contact me regarding how we can get a patch incorporated into
>> FS to pass the Subject header.
>>
>>
>>
>> Thanks Mike
>>
>>
>>
>>
>>
>>
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David
>> Ponzone
>> Sent: Wednesday, December 22, 2010 2:31 PM
>>
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>>
>>
>>
>> Michael,
>>
>>
>>
>> what Brian meant is that FreeSWICH can only passes the X-something and
>> P-something headers, which are the only allowed ones.
>>
>> Any system requiring custom headers should implement:
>>
>> X-Subject
>>
>> and not
>>
>> Subject
>>
>>
>>
>>
>>
>> David Ponzone ?Direction Technique
>>
>> email: david.ponzone at ipeva.fr
>>
>> tel: ? ? ?01 74 03 18 97
>>
>> gsm: ? 06 66 98 76 34
>>
>>
>>
>> Service Client?IPeva
>>
>> tel: ? ? ?0811 46 26 26
>>
>> www.ipeva.fr? -? ?www.ipeva-studio.com
>>
>>
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>>
>>
>> Le 22/12/2010 ? 20:09, Kane, Michael (mkane02) a ?crit :
>>
>> We have a custom app that requires the Subject header with values so our
>> endpoints can route the audio accordingly. ?I'm ok with using sip_h_X
>> and or sip_h_P, but can't seem to get FS to pass the Subject header.
>> Any direction would be greatly appreciated.
>>
>> Mike
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Brian West
>> Sent: Wednesday, December 22, 2010 1:58 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>>
>> What is the sudden obsession with the subject header? ?Looking thru the
>> code it appears as if sip_h_X and sip_h_P are the only two things that
>> you can do... a patch will have to be made if you wish to set the
>> subject on an outbound invite... see sofia_glue.c ?in do_invite.
>>
>> /b
>>
>> On Dec 22, 2010, at 12:50 PM, Kane, Michael (mkane02) wrote:
>>
>>
>> Hey Steve, the Subject is still not regenerated through the switch. ?I
>>
>> posted a new pastbin.
>>
>>
>>
>> 14858
>>
>>
>>
>> Mike
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Thu Dec 23 19:04:51 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 23 Dec 2010 10:04:51 -0600
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AANLkTikXa8bXORjRkxgHUcvRBeg7pJ4zLBczK5pOpH_i@mail.gmail.com>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net>
	<AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net>
	<AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net>
	<018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net>
	<41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net>
	<AANLkTikD8J_-EgtrheOfqNWOgDMpSYr+sGDkBw6-w_Pp@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D009@mlbe2k9.cs.myharris.net>
	<AANLkTikXa8bXORjRkxgHUcvRBeg7pJ4zLBczK5pOpH_i@mail.gmail.com>
Message-ID: <AD5272CF-5528-4525-A7B4-DE5202E2A8EA@freeswitch.org>

This makes me laugh... could they fuck it up any worse then it already is?

/b

On Dec 23, 2010, at 9:46 AM, Steven Ayre wrote:

> SIPv3



From infos at madovsky.org  Thu Dec 23 19:10:48 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 23 Dec 2010 11:10:48 -0500
Subject: [Freeswitch-users] no invite send
References: <AANLkTimBh+CqTz1O5qNmd7=z+Z6WZGeY1ua8kdZAwzH5@mail.gmail.com><AANLkTikpL3pfFieW3Yv20iFxDo+yNdYA9zoyd3tgT-iK@mail.gmail.com>
	<AANLkTim0hhJBaJSg9ghh8cwrbSY8_rC+4oOP79vT-Hw4@mail.gmail.com>
Message-ID: <038BD7F658C942DAB81DD149475CF8A2@e1705>

you have to install or correct path of apr-utils package
  ----- Original Message ----- 
  From: Sam 
  To: FreeSWITCH Users Help 
  Sent: Thursday, December 23, 2010 5:28 AM
  Subject: Re: [Freeswitch-users] no invite send


  I check it in sip trace ... it do not sends invite, probably i might have miss configured something,
  on the another FS server it works fine, this happened on the new installation where i was using prefix,

  I was trying to reinstall FS on the new server with prefix it gave me below error.

  ./configure: line 38199: 30565 Segmentation fault      /bin/sh /usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat/configure '--prefix=/usr/local/conference/' 'CONFIGURE_CFLAGS=-g -O2' 'CONFIGURE_CXXFLAGS=-g -O2' 'CONFIGURE_LDFLAGS=' '--cache-file=/dev/null' '--srcdir=.' '--with-apr=../apr' '--disable-shared' '--with-pic' '--without-sqlite2' '--without-sqlite3' '--with-expat=builtin' --cache-file=/dev/null --srcdir=/usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat --prefix=/usr/local/conference/ --exec-prefix=${prefix} --libdir=${exec_prefix}/lib --includedir=${prefix}/include/apr-${APRUTIL_MAJOR_VERSION} --bindir=${exec_prefix}/bin
  configure failed for xml/expat
  configure: error: /bin/sh './configure.gnu' failed for libs/apr-util



  i was using ./configure --prefix=/usr/local/conference/



  Regards
  Sam




  On Thu, Dec 23, 2010 at 3:49 PM, Steven Ayre <steveayre at gmail.com> wrote:

    Try enabling siptrace (sofia profile external siptrace on) - does it
    show an INVITE being sent in the logs?

    -Steve



    On 23 December 2010 09:20, Sam <u2nsam at gmail.com> wrote:
    > Hello friends,
    >
    > I am using the dialplan,
    >
    > <action application="bridge" data="sofia/external/12345 at 192.168.2.3"/>
    >
    > its not sending INVITE to the server 192.168.2.3
    >
    > i am getting the below logs:-
    >
    > 2010-12-23 14:03:02.509500 [NOTICE] switch_channel.c:784 New Channel
    > sofia/external/12345 at 192.168.2.3 [418e0dfa-0e6f-11e0-a5e1-33ac6c473a4e]
    > 2010-12-23 14:03:02.509500 [DEBUG] mod_sofia.c:4052
    > (sofia/external/12345 at 192.168.2.3) State Change CS_NEW -> CS_INIT
    > 2010-12-23 14:03:02.509500 [DEBUG] switch_core_session.c:1083 Send signal
    > sofia/external/12345 at 192.168.2.3 [BREAK]
    > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
    > (sofia/external/12345 at 192.168.2.3) Running State Change CS_INIT
    > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
    > (sofia/external/12345 at 192.168.2.3) State INIT
    > 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:86
    > sofia/external/12345 at 192.168.2.3 SOFIA INIT
    > 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:126
    > (sofia/external/12345 at 192.168.2.3) State Change CS_INIT -> CS_ROUTING
    > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_session.c:1083 Send signal
    > sofia/external/12345 at 192.168.2.3 [BREAK]
    > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
    > (sofia/external/12345 at 192.168.2.3) State INIT going to sleep
    > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
    > (sofia/external/12345 at 192.168.2.3) Running State Change CS_ROUTING
    > 2010-12-23 14:03:02.510864 [DEBUG] switch_channel.c:1615
    > (sofia/external/12345 at 192.168.2.3) Callstate Change DOWN -> RINGING
    > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
    > (sofia/external/12345 at 192.168.2.3) State ROUTING
    > 2010-12-23 14:03:02.511888 [DEBUG] mod_sofia.c:149
    > sofia/external/12345 at 192.168.2.3 SOFIA ROUTING
    > 2010-12-23 14:03:02.511888 [DEBUG] switch_ivr_originate.c:66
    > (sofia/external/12345 at 192.168.2.3) State Change CS_ROUTING ->
    > CS_CONSUME_MEDIA
    > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_session.c:1083 Send signal
    > sofia/external/12345 at 192.168.2.3 [BREAK]
    > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
    > (sofia/external/12345 at 192.168.2.3) State ROUTING going to sleep
    > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:320
    > (sofia/external/12345 at 192.168.2.3) Running State Change CS_CONSUME_MEDIA
    > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:378
    > (sofia/external/12345 at 192.168.2.3) State CONSUME_MEDIA
    > 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
    > sofia/external/12345 at 192.168.2.3 entering state [calling][0]
    > 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
    > sofia/external/12345 at 192.168.2.3 entering state [terminated][503]
    > 2010-12-23 14:03:02.511888 [DEBUG] switch_channel.c:2493
    > (sofia/external/12345 at 192.168.2.3) Callstate Change RINGING -> HANGUP
    > 2010-12-23 14:03:02.511888 [NOTICE] sofia.c:5246 Hangup
    > sofia/external/12345 at 192.168.2.3 [CS_CONSUME_MEDIA]
    > [NORMAL_TEMPORARY_FAILURE]
    >
    >
    > Regards
    > Sam
    >

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

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





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From steveayre at gmail.com  Thu Dec 23 19:24:50 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 23 Dec 2010 16:24:50 +0000
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AD5272CF-5528-4525-A7B4-DE5202E2A8EA@freeswitch.org>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net>
	<AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net>
	<AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net>
	<018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net>
	<41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net>
	<AANLkTikD8J_-EgtrheOfqNWOgDMpSYr+sGDkBw6-w_Pp@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D009@mlbe2k9.cs.myharris.net>
	<AANLkTikXa8bXORjRkxgHUcvRBeg7pJ4zLBczK5pOpH_i@mail.gmail.com>
	<AD5272CF-5528-4525-A7B4-DE5202E2A8EA@freeswitch.org>
Message-ID: <AANLkTikePmGWBcdihqsdVB4H7jMevEWp8L9TV=MaOrpC@mail.gmail.com>

I'm sure they could find a way...

On 23 December 2010 16:04, Brian West <brian at freeswitch.org> wrote:
> This makes me laugh... could they fuck it up any worse then it already is?
>
> /b
>
> On Dec 23, 2010, at 9:46 AM, Steven Ayre wrote:
>
>> SIPv3
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Thu Dec 23 19:28:09 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 23 Dec 2010 10:28:09 -0600
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AANLkTikePmGWBcdihqsdVB4H7jMevEWp8L9TV=MaOrpC@mail.gmail.com>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net>
	<AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net>
	<AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net>
	<018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net>
	<41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net>
	<AANLkTikD8J_-EgtrheOfqNWOgDMpSYr+sGDkBw6-w_Pp@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D009@mlbe2k9.cs.myharris.net>
	<AANLkTikXa8bXORjRkxgHUcvRBeg7pJ4zLBczK5pOpH_i@mail.gmail.com>
	<AD5272CF-5528-4525-A7B4-DE5202E2A8EA@freeswitch.org>
	<AANLkTikePmGWBcdihqsdVB4H7jMevEWp8L9TV=MaOrpC@mail.gmail.com>
Message-ID: <F2B723D7-7463-47F0-AE4D-133FCC9348C6@freeswitch.org>

This is how http://tools.ietf.org/html/draft-shirasaki-nat444-isp-shared-addr-04

/b

On Dec 23, 2010, at 10:24 AM, Steven Ayre wrote:

> I'm sure they could find a way...
> 
> On 23 December 2010 16:04, Brian West <brian at freeswitch.org> wrote:
>> This makes me laugh... could they fuck it up any worse then it already is?
>> 
>> /b
>> 
>> On Dec 23, 2010, at 9:46 AM, Steven Ayre wrote:
>> 
>>> SIPv3
> 

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From mkane02 at harris.com  Thu Dec 23 19:27:39 2010
From: mkane02 at harris.com (Kane, Michael (mkane02))
Date: Thu, 23 Dec 2010 11:27:39 -0500
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AANLkTikXa8bXORjRkxgHUcvRBeg7pJ4zLBczK5pOpH_i@mail.gmail.com>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net><AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net><AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net><018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org><AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net><41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr><AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net><AANLkTikD8J_-EgtrheOfqNWOgDMpSYr+sGDkBw6-w_Pp@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184D009@mlbe2k9.cs.myharris.net>
	<AANLkTikXa8bXORjRkxgHUcvRBeg7pJ4zLBczK5pOpH_i@mail.gmail.com>
Message-ID: <AF824FE3F385E543ACA98ADFD670BA6A0184D00A@mlbe2k9.cs.myharris.net>

Thanks Steve.

Mike

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Thursday, December 23, 2010 10:46 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropping the Subject: header

http://www.rfc-editor.org/rfc/rfc3427.txt
The RFC recognises X- has been used in other protocols (eg SMTP) and
says prefix P- should be used for SIP. In practice both P- and X- are
used.

The reason is that if you are setting a custom header, it avoids
conflicts if they revise the SIP specification in future (eg for
SIPv3) and add a header with the same name as yours but a different
purpose.

See also:
http://blogs.voxeo.com/speakingofstandards/2008/05/06/what-is-a-p-header-in-sip-and-whyhow-would-you-use-one/
http://www.the-asterisk-book.com/unstable/applikationen-sipaddheader.html

Note that the X-header functionality is specifically coded within the
Sofia-SIP library:
http://sofia-sip.sourceforge.net/refdocs/sip/group__sip__header__x.html
To handle custom headers not prefixed with X- or P- would involve
considerable rewriting of the library to expose the ability. Since the
library is written by Nokia that would result in a large difference
between the FS and upstream versions, making it hard to maintain.

Regards,
Steve


On 23 December 2010 13:46, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
> Hey Steve/David, I'm a bit confused about having to add the X- prefix to the Subject header. ?Can you please point me in the direction of which RFC talks to experimental headers?
>
> Thanks Mike
>
> Ps...thanks for looking into this for me.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
> Sent: Wednesday, December 22, 2010 5:04 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>
> David,
>
> Change your custom app to use X-Subject not Subject (AFAIK the RFC
> says any custom headers should be prefixed X-). My previous example
> will then work if you export sip_h_X-Subject.
>
> -Steve
>
>
> On 22 December 2010 19:40, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
>> Thanks David, I really appreciate your clarification.
>>
>>
>>
>>
>>
>> Brian, please contact me regarding how we can get a patch incorporated into
>> FS to pass the Subject header.
>>
>>
>>
>> Thanks Mike
>>
>>
>>
>>
>>
>>
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David
>> Ponzone
>> Sent: Wednesday, December 22, 2010 2:31 PM
>>
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>>
>>
>>
>> Michael,
>>
>>
>>
>> what Brian meant is that FreeSWICH can only passes the X-something and
>> P-something headers, which are the only allowed ones.
>>
>> Any system requiring custom headers should implement:
>>
>> X-Subject
>>
>> and not
>>
>> Subject
>>
>>
>>
>>
>>
>> David Ponzone ?Direction Technique
>>
>> email: david.ponzone at ipeva.fr
>>
>> tel: ? ? ?01 74 03 18 97
>>
>> gsm: ? 06 66 98 76 34
>>
>>
>>
>> Service Client?IPeva
>>
>> tel: ? ? ?0811 46 26 26
>>
>> www.ipeva.fr? -? ?www.ipeva-studio.com
>>
>>
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>>
>>
>> Le 22/12/2010 ? 20:09, Kane, Michael (mkane02) a ?crit :
>>
>> We have a custom app that requires the Subject header with values so our
>> endpoints can route the audio accordingly. ?I'm ok with using sip_h_X
>> and or sip_h_P, but can't seem to get FS to pass the Subject header.
>> Any direction would be greatly appreciated.
>>
>> Mike
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Brian West
>> Sent: Wednesday, December 22, 2010 1:58 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>>
>> What is the sudden obsession with the subject header? ?Looking thru the
>> code it appears as if sip_h_X and sip_h_P are the only two things that
>> you can do... a patch will have to be made if you wish to set the
>> subject on an outbound invite... see sofia_glue.c ?in do_invite.
>>
>> /b
>>
>> On Dec 22, 2010, at 12:50 PM, Kane, Michael (mkane02) wrote:
>>
>>
>> Hey Steve, the Subject is still not regenerated through the switch. ?I
>>
>> posted a new pastbin.
>>
>>
>>
>> 14858
>>
>>
>>
>> Mike
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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From mkane02 at harris.com  Thu Dec 23 19:28:23 2010
From: mkane02 at harris.com (Kane, Michael (mkane02))
Date: Thu, 23 Dec 2010 11:28:23 -0500
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AD5272CF-5528-4525-A7B4-DE5202E2A8EA@freeswitch.org>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net><AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net><AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net><018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org><AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net><41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr><AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net><AANLkTikD8J_-EgtrheOfqNWOgDMpSYr+sGDkBw6-w_Pp@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184D009@mlbe2k9.cs.myharris.net><AANLkTikXa8bXORjRkxgHUcvRBeg7pJ4zLBczK5pOpH_i@mail.gmail.com>
	<AD5272CF-5528-4525-A7B4-DE5202E2A8EA@freeswitch.org>
Message-ID: <AF824FE3F385E543ACA98ADFD670BA6A0184D00B@mlbe2k9.cs.myharris.net>

Sorry Brian, I'm didn't follow(understand) your thread.

Mike

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West
Sent: Thursday, December 23, 2010 11:05 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropping the Subject: header

This makes me laugh... could they fuck it up any worse then it already
is?

/b

On Dec 23, 2010, at 9:46 AM, Steven Ayre wrote:

> SIPv3


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From mkane02 at harris.com  Thu Dec 23 19:46:12 2010
From: mkane02 at harris.com (Kane, Michael (mkane02))
Date: Thu, 23 Dec 2010 11:46:12 -0500
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AANLkTikXa8bXORjRkxgHUcvRBeg7pJ4zLBczK5pOpH_i@mail.gmail.com>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net><AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net><AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net><018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org><AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net><41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr><AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net><AANLkTikD8J_-EgtrheOfqNWOgDMpSYr+sGDkBw6-w_Pp@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184D009@mlbe2k9.cs.myharris.net>
	<AANLkTikXa8bXORjRkxgHUcvRBeg7pJ4zLBczK5pOpH_i@mail.gmail.com>
Message-ID: <AF824FE3F385E543ACA98ADFD670BA6A0184D00E@mlbe2k9.cs.myharris.net>

Hey Steve, I was under the impression Subject header wasn't a custom header.  We're using Media5's stack and it's readily available in their library and is defined in 3261.  

Mike

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Thursday, December 23, 2010 10:46 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropping the Subject: header

http://www.rfc-editor.org/rfc/rfc3427.txt
The RFC recognises X- has been used in other protocols (eg SMTP) and
says prefix P- should be used for SIP. In practice both P- and X- are
used.

The reason is that if you are setting a custom header, it avoids
conflicts if they revise the SIP specification in future (eg for
SIPv3) and add a header with the same name as yours but a different
purpose.

See also:
http://blogs.voxeo.com/speakingofstandards/2008/05/06/what-is-a-p-header-in-sip-and-whyhow-would-you-use-one/
http://www.the-asterisk-book.com/unstable/applikationen-sipaddheader.html

Note that the X-header functionality is specifically coded within the
Sofia-SIP library:
http://sofia-sip.sourceforge.net/refdocs/sip/group__sip__header__x.html
To handle custom headers not prefixed with X- or P- would involve
considerable rewriting of the library to expose the ability. Since the
library is written by Nokia that would result in a large difference
between the FS and upstream versions, making it hard to maintain.

Regards,
Steve


On 23 December 2010 13:46, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
> Hey Steve/David, I'm a bit confused about having to add the X- prefix to the Subject header. ?Can you please point me in the direction of which RFC talks to experimental headers?
>
> Thanks Mike
>
> Ps...thanks for looking into this for me.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
> Sent: Wednesday, December 22, 2010 5:04 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>
> David,
>
> Change your custom app to use X-Subject not Subject (AFAIK the RFC
> says any custom headers should be prefixed X-). My previous example
> will then work if you export sip_h_X-Subject.
>
> -Steve
>
>
> On 22 December 2010 19:40, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
>> Thanks David, I really appreciate your clarification.
>>
>>
>>
>>
>>
>> Brian, please contact me regarding how we can get a patch incorporated into
>> FS to pass the Subject header.
>>
>>
>>
>> Thanks Mike
>>
>>
>>
>>
>>
>>
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David
>> Ponzone
>> Sent: Wednesday, December 22, 2010 2:31 PM
>>
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>>
>>
>>
>> Michael,
>>
>>
>>
>> what Brian meant is that FreeSWICH can only passes the X-something and
>> P-something headers, which are the only allowed ones.
>>
>> Any system requiring custom headers should implement:
>>
>> X-Subject
>>
>> and not
>>
>> Subject
>>
>>
>>
>>
>>
>> David Ponzone ?Direction Technique
>>
>> email: david.ponzone at ipeva.fr
>>
>> tel: ? ? ?01 74 03 18 97
>>
>> gsm: ? 06 66 98 76 34
>>
>>
>>
>> Service Client?IPeva
>>
>> tel: ? ? ?0811 46 26 26
>>
>> www.ipeva.fr? -? ?www.ipeva-studio.com
>>
>>
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>>
>>
>> Le 22/12/2010 ? 20:09, Kane, Michael (mkane02) a ?crit :
>>
>> We have a custom app that requires the Subject header with values so our
>> endpoints can route the audio accordingly. ?I'm ok with using sip_h_X
>> and or sip_h_P, but can't seem to get FS to pass the Subject header.
>> Any direction would be greatly appreciated.
>>
>> Mike
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Brian West
>> Sent: Wednesday, December 22, 2010 1:58 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>>
>> What is the sudden obsession with the subject header? ?Looking thru the
>> code it appears as if sip_h_X and sip_h_P are the only two things that
>> you can do... a patch will have to be made if you wish to set the
>> subject on an outbound invite... see sofia_glue.c ?in do_invite.
>>
>> /b
>>
>> On Dec 22, 2010, at 12:50 PM, Kane, Michael (mkane02) wrote:
>>
>>
>> Hey Steve, the Subject is still not regenerated through the switch. ?I
>>
>> posted a new pastbin.
>>
>>
>>
>> 14858
>>
>>
>>
>> Mike
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From chris at cloudtel.com  Thu Dec 23 19:48:50 2010
From: chris at cloudtel.com (Chris Burns)
Date: Thu, 23 Dec 2010 11:48:50 -0500
Subject: [Freeswitch-users] Asynchronous PTIME
In-Reply-To: <4D127733.8010409@nevian.org>
References: <4D127733.8010409@nevian.org>
Message-ID: <AANLkTi=UY1RDLD=vACoYR+AGssOf=q1KYGtmxHZMFOps@mail.gmail.com>

The switch is receiving audio frames larger than it should for the
negotiated codec, which causes it to analyze the timing of received frames
and correct the issue. To stop the switch from trying to fix the timing you
should be able to uncomment the rtp-autofix-timing line in your related
sofia profile, but I cant speak to whether that will fix your audio.

There is a logfile.conf.xml to alter any of the settings of how the switch
logs to file. For debugging SIP to log file you should do it outside of the
switch console, for instance with ngrep.

On Wed, Dec 22, 2010 at 5:09 PM, Serge S. Yuriev <me at nevian.org> wrote:

> Hello,
>
> 2010-12-20 16:51:13.960502 [WARNING] mod_sofia.c:1036 Asynchronous PTIME
> not supported, changing our end from 20 to 60
>
> I'm getting this warning and client hears chopped sound :(
> That is "Our end"?
>
> Call flow is Cisco (sip) FS (sip) YATE (h323) MeraTransitSoftSwitch
> All but MVTS under my control.
>
> I doesn't see any clue in logs and can't reproduce this with my testing
> via Blink/PortSIP/PhonerLite/Twinkle or Grandstream devices
>
> Which debug/logs I should take? Any ideas?
>
> Thanks a lot.
>
> btw how I can save debug into log not only console?
> --
> wbr,
> Serge
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20101223/0d74ad92/attachment.html 

From me at nevian.org  Thu Dec 23 19:56:36 2010
From: me at nevian.org (Serge S. Yuriev)
Date: Thu, 23 Dec 2010 19:56:36 +0300
Subject: [Freeswitch-users] Asynchronous PTIME
In-Reply-To: <4D127733.8010409@nevian.org>
References: <4D127733.8010409@nevian.org>
Message-ID: <20101223195636.755ab1df@nevian.org>

Hello,

On Thu, 23 Dec 2010 01:09:55 +0300 "Serge S. Yuriev" <me at nevian.org>
wrote:

Any one?
Captured console log here
http://pastebin.freeswitch.org/14871

I wonna clear on side of this problem - should I beat provider or have
beaten myself..

> 2010-12-20 16:51:13.960502 [WARNING] mod_sofia.c:1036 Asynchronous
> PTIME not supported, changing our end from 20 to 60
> 
> I'm getting this warning and client hears chopped sound :(
> That is "Our end"?
> 
> Call flow is Cisco (sip) FS (sip) YATE (h323) MeraTransitSoftSwitch
> All but MVTS under my control.
> 
> I doesn't see any clue in logs and can't reproduce this with my
> testing via Blink/PortSIP/PhonerLite/Twinkle or Grandstream devices
> 
> Which debug/logs I should take? Any ideas?
> 
> Thanks a lot.
> 
> btw how I can save debug into log not only console?

-- 
Serge S. Yuriev
Lead VoIP engineer



From me at nevian.org  Thu Dec 23 20:08:34 2010
From: me at nevian.org (Serge S. Yuriev)
Date: Thu, 23 Dec 2010 20:08:34 +0300
Subject: [Freeswitch-users] Asynchronous PTIME
In-Reply-To: <AANLkTi=UY1RDLD=vACoYR+AGssOf=q1KYGtmxHZMFOps@mail.gmail.com>
References: <4D127733.8010409@nevian.org>
	<AANLkTi=UY1RDLD=vACoYR+AGssOf=q1KYGtmxHZMFOps@mail.gmail.com>
Message-ID: <20101223200834.271583b7@nevian.org>

?????? ????,

On Thu, 23 Dec 2010 11:48:50 -0500 Chris Burns <chris at cloudtel.com>
wrote:

Thank you so much for answers!

> The switch is receiving audio frames larger than it should for the
> negotiated codec, which causes it to analyze the timing of received
> frames and correct the issue.

So it's not wrong signaling problem it's somewhere in RTP, right?

> To stop the switch from trying to fix
> the timing you should be able to uncomment the rtp-autofix-timing
> line in your related sofia profile, but I cant speak to whether that
> will fix your audio.

How I can analyse side "bad" RTP from? WireShark?
 
> There is a logfile.conf.xml to alter any of the settings of how the
> switch logs to file. For debugging SIP to log file you should do it
> outside of the switch console, for instance with ngrep.

Ok, got it.

> On Wed, Dec 22, 2010 at 5:09 PM, Serge S. Yuriev <me at nevian.org>
> wrote:
> 
> > Hello,
> >
> > 2010-12-20 16:51:13.960502 [WARNING] mod_sofia.c:1036 Asynchronous
> > PTIME not supported, changing our end from 20 to 60
> >
> > I'm getting this warning and client hears chopped sound :(
> > That is "Our end"?
> >
> > Call flow is Cisco (sip) FS (sip) YATE (h323) MeraTransitSoftSwitch
> > All but MVTS under my control.
> >
> > I doesn't see any clue in logs and can't reproduce this with my
> > testing via Blink/PortSIP/PhonerLite/Twinkle or Grandstream devices
> >
> > Which debug/logs I should take? Any ideas?
> >
> > Thanks a lot.
> >
> > btw how I can save debug into log not only console?
> > --
> > wbr,
> > Serge
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >



-- 
Serge S. Yuriev
Lead VoIP engineer



From steveayre at gmail.com  Thu Dec 23 20:25:42 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 23 Dec 2010 17:25:42 +0000
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AF824FE3F385E543ACA98ADFD670BA6A0184D00E@mlbe2k9.cs.myharris.net>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net>
	<AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net>
	<AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net>
	<018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net>
	<41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net>
	<AANLkTikD8J_-EgtrheOfqNWOgDMpSYr+sGDkBw6-w_Pp@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D009@mlbe2k9.cs.myharris.net>
	<AANLkTikXa8bXORjRkxgHUcvRBeg7pJ4zLBczK5pOpH_i@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D00E@mlbe2k9.cs.myharris.net>
Message-ID: <AANLkTi=OdTTbp+HJLHiwG609pA7pQsvjRQ_kQUyent0V@mail.gmail.com>

Ah, you're quite right... sorry, I locked onto you saying that you
were setting it for a custom app and assumed you were trying to do
something outside of the specification.

Looking at the source code, it looks like FreeSWITCH doesn't currently
provide any way to use that field on an INVITE from mod_sofia... if
you wanted it you'd have to open a Jira ticket.

http://jira.freeswitch.org/

Regards,
-Steve



On 23 December 2010 16:46, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
> Hey Steve, I was under the impression Subject header wasn't a custom header. ?We're using Media5's stack and it's readily available in their library and is defined in 3261.
>
> Mike
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
> Sent: Thursday, December 23, 2010 10:46 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>
> http://www.rfc-editor.org/rfc/rfc3427.txt
> The RFC recognises X- has been used in other protocols (eg SMTP) and
> says prefix P- should be used for SIP. In practice both P- and X- are
> used.
>
> The reason is that if you are setting a custom header, it avoids
> conflicts if they revise the SIP specification in future (eg for
> SIPv3) and add a header with the same name as yours but a different
> purpose.
>
> See also:
> http://blogs.voxeo.com/speakingofstandards/2008/05/06/what-is-a-p-header-in-sip-and-whyhow-would-you-use-one/
> http://www.the-asterisk-book.com/unstable/applikationen-sipaddheader.html
>
> Note that the X-header functionality is specifically coded within the
> Sofia-SIP library:
> http://sofia-sip.sourceforge.net/refdocs/sip/group__sip__header__x.html
> To handle custom headers not prefixed with X- or P- would involve
> considerable rewriting of the library to expose the ability. Since the
> library is written by Nokia that would result in a large difference
> between the FS and upstream versions, making it hard to maintain.
>
> Regards,
> Steve
>
>
> On 23 December 2010 13:46, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
>> Hey Steve/David, I'm a bit confused about having to add the X- prefix to the Subject header. ?Can you please point me in the direction of which RFC talks to experimental headers?
>>
>> Thanks Mike
>>
>> Ps...thanks for looking into this for me.
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
>> Sent: Wednesday, December 22, 2010 5:04 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>>
>> David,
>>
>> Change your custom app to use X-Subject not Subject (AFAIK the RFC
>> says any custom headers should be prefixed X-). My previous example
>> will then work if you export sip_h_X-Subject.
>>
>> -Steve
>>
>>
>> On 22 December 2010 19:40, Kane, Michael (mkane02) <mkane02 at harris.com> wrote:
>>> Thanks David, I really appreciate your clarification.
>>>
>>>
>>>
>>>
>>>
>>> Brian, please contact me regarding how we can get a patch incorporated into
>>> FS to pass the Subject header.
>>>
>>>
>>>
>>> Thanks Mike
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David
>>> Ponzone
>>> Sent: Wednesday, December 22, 2010 2:31 PM
>>>
>>> To: FreeSWITCH Users Help
>>> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>>>
>>>
>>>
>>> Michael,
>>>
>>>
>>>
>>> what Brian meant is that FreeSWICH can only passes the X-something and
>>> P-something headers, which are the only allowed ones.
>>>
>>> Any system requiring custom headers should implement:
>>>
>>> X-Subject
>>>
>>> and not
>>>
>>> Subject
>>>
>>>
>>>
>>>
>>>
>>> David Ponzone ?Direction Technique
>>>
>>> email: david.ponzone at ipeva.fr
>>>
>>> tel: ? ? ?01 74 03 18 97
>>>
>>> gsm: ? 06 66 98 76 34
>>>
>>>
>>>
>>> Service Client?IPeva
>>>
>>> tel: ? ? ?0811 46 26 26
>>>
>>> www.ipeva.fr? -? ?www.ipeva-studio.com
>>>
>>>
>>>
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>
>>>
>>>
>>>
>>>
>>> Le 22/12/2010 ? 20:09, Kane, Michael (mkane02) a ?crit :
>>>
>>> We have a custom app that requires the Subject header with values so our
>>> endpoints can route the audio accordingly. ?I'm ok with using sip_h_X
>>> and or sip_h_P, but can't seem to get FS to pass the Subject header.
>>> Any direction would be greatly appreciated.
>>>
>>> Mike
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>> Brian West
>>> Sent: Wednesday, December 22, 2010 1:58 PM
>>> To: FreeSWITCH Users Help
>>> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>>>
>>> What is the sudden obsession with the subject header? ?Looking thru the
>>> code it appears as if sip_h_X and sip_h_P are the only two things that
>>> you can do... a patch will have to be made if you wish to set the
>>> subject on an outbound invite... see sofia_glue.c ?in do_invite.
>>>
>>> /b
>>>
>>> On Dec 22, 2010, at 12:50 PM, Kane, Michael (mkane02) wrote:
>>>
>>>
>>> Hey Steve, the Subject is still not regenerated through the switch. ?I
>>>
>>> posted a new pastbin.
>>>
>>>
>>>
>>> 14858
>>>
>>>
>>>
>>> Mike
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Thu Dec 23 20:32:50 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 23 Dec 2010 11:32:50 -0600
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AANLkTi=OdTTbp+HJLHiwG609pA7pQsvjRQ_kQUyent0V@mail.gmail.com>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net>
	<AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net>
	<AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net>
	<018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net>
	<41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net>
	<AANLkTikD8J_-EgtrheOfqNWOgDMpSYr+sGDkBw6-w_Pp@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D009@mlbe2k9.cs.myharris.net>
	<AANLkTikXa8bXORjRkxgHUcvRBeg7pJ4zLBczK5pOpH_i@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D00E@mlbe2k9.cs.myharris.net>
	<AANLkTi=OdTTbp+HJLHiwG609pA7pQsvjRQ_kQUyent0V@mail.gmail.com>
Message-ID: <559A89A6-DD9A-4002-ACB9-5D367D8E2F9C@freeswitch.org>

He did open a jira as a patch request but no bounty amount on it.  So someone will have to patch it for him for free or he'll have to post a bounty to raise the priority of the bug a bit because its a non-critical issue.  Its at the bottom of my list.

/b

On Dec 23, 2010, at 11:25 AM, Steven Ayre wrote:

> Ah, you're quite right... sorry, I locked onto you saying that you
> were setting it for a custom app and assumed you were trying to do
> something outside of the specification.
> 
> Looking at the source code, it looks like FreeSWITCH doesn't currently
> provide any way to use that field on an INVITE from mod_sofia... if
> you wanted it you'd have to open a Jira ticket.
> 
> http://jira.freeswitch.org/
> 
> Regards,
> -Steve

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From rafonline at hotmail.com  Thu Dec 23 20:42:57 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Thu, 23 Dec 2010 17:42:57 +0000
Subject: [Freeswitch-users] ESL outbound /inbound
Message-ID: <COL122-W2256918ACD58AE62DC48CDB51C0@phx.gbl>


Hi,

I am writing a calling card app using lua and java.  

As advised, Java will predominatley do all the billing side of things (via ESL) leaving lua do simple things like asking for the calling card pin number etc.

When a call comes in, my lua script answers the call and asks the user for a pin.  Instead of querying the DB inline from within lua, I would like my app server to do this (please let me know if this should be done inline instead).  I understand my app server (Java ESL inbound socket) can register for a pin checking custom event and I can generate such an event from within my lua script.  My lua script would then wait for an appropriate repsonse event:

-- Check if pin is valid
local event = freeswitch.Event("CUSTOM", "check_pin_request");
event:addHeader("pin_number", digits);
event:fire();

-- wait for response
con = freeswitch.EventConsumer("CUSTOM", "check_pin_reponse");
con:pop(1);
print("event\n" .. e:serialize("xml"));

I was wondering if the above is the right way of doing things, or should I be using ESLOutboundSocket and have lua script do something like this instead:

session:execute("set", "pin_to_check=12345");
session:execute("socket", "192.168.0.2:8084");

Not sure how lua will be told whether the pin is valid or not in this scenario.

I appreciate FreeSWITCH is very flexible and would like to make sure I develop a scalable and performant application.

Any help will be much appreciated.

Cheers

Raf



 		 	   		  
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From mkane02 at harris.com  Thu Dec 23 20:51:52 2010
From: mkane02 at harris.com (Kane, Michael (mkane02))
Date: Thu, 23 Dec 2010 12:51:52 -0500
Subject: [Freeswitch-users] Dropping the Subject: header
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net><AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net><AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net><018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org><AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net><41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr><AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net><AANLkTikD8J_-EgtrheOfqNWOgDMpSYr+sGDkBw6-w_Pp@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184D009@mlbe2k9.cs.myharris.net><AANLkTikXa8bXORjRkxgHUcvRBeg7pJ4zLBczK5pOpH_i@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184D00E@mlbe2k9.cs.myharris.net><AANLkTi=OdTTbp+HJLHiwG609pA7pQsvjRQ_kQUyent0V@mail.gmail.com>
	<559A89A6-DD9A-4002-ACB9-5D367D8E2F9C@freeswitch.org>
Message-ID: <AF824FE3F385E543ACA98ADFD670BA6A02A07F81@mlbe2k9.cs.myharris.net>

Ok, Brian can you call me at some point to discuss.  321-223-5511.  I have no problem posting  as a bounty.  

We can roll this into the discussions we've had in the past for support and programming support.  The problem is we usually do not deal with independent contractors as a rule.  When one posts a bounty are they dealing with FreeSWITCH solutions or independent contractors?  

Thanks Mike 


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Brian West
Sent: Thu 12/23/2010 12:32 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropping the Subject: header
 
He did open a jira as a patch request but no bounty amount on it.  So someone will have to patch it for him for free or he'll have to post a bounty to raise the priority of the bug a bit because its a non-critical issue.  Its at the bottom of my list.

/b

On Dec 23, 2010, at 11:25 AM, Steven Ayre wrote:

> Ah, you're quite right... sorry, I locked onto you saying that you
> were setting it for a custom app and assumed you were trying to do
> something outside of the specification.
> 
> Looking at the source code, it looks like FreeSWITCH doesn't currently
> provide any way to use that field on an INVITE from mod_sofia... if
> you wanted it you'd have to open a Jira ticket.
> 
> http://jira.freeswitch.org/
> 
> Regards,
> -Steve




From george at ezuce.com  Thu Dec 23 12:40:05 2010
From: george at ezuce.com (George Niculae)
Date: Thu, 23 Dec 2010 11:40:05 +0200
Subject: [Freeswitch-users] no invite send
In-Reply-To: <AANLkTimBh+CqTz1O5qNmd7=z+Z6WZGeY1ua8kdZAwzH5@mail.gmail.com>
References: <AANLkTimBh+CqTz1O5qNmd7=z+Z6WZGeY1ua8kdZAwzH5@mail.gmail.com>
Message-ID: <AANLkTik5c65RBdq6K3JP9jJyXxywWEuAg8Yuq9hrY96t@mail.gmail.com>

On Thu, Dec 23, 2010 at 11:20 AM, Sam <u2nsam at gmail.com> wrote:
> Hello friends,
>
> I am using the dialplan,
>
> <action application="bridge" data="sofia/external/12345 at 192.168.2.3"/>
>
> its not sending INVITE to the server 192.168.2.3
>
> i am getting the below logs:-
>
> 2010-12-23 14:03:02.509500 [NOTICE] switch_channel.c:784 New Channel
> sofia/external/12345 at 192.168.2.3 [418e0dfa-0e6f-11e0-a5e1-33ac6c473a4e]
> 2010-12-23 14:03:02.509500 [DEBUG] mod_sofia.c:4052
> (sofia/external/12345 at 192.168.2.3) State Change CS_NEW -> CS_INIT
> 2010-12-23 14:03:02.509500 [DEBUG] switch_core_session.c:1083 Send signal
> sofia/external/12345 at 192.168.2.3 [BREAK]
> 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/12345 at 192.168.2.3) Running State Change CS_INIT
> 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/12345 at 192.168.2.3) State INIT
> 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:86
> sofia/external/12345 at 192.168.2.3 SOFIA INIT
> 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:126
> (sofia/external/12345 at 192.168.2.3) State Change CS_INIT -> CS_ROUTING
> 2010-12-23 14:03:02.510864 [DEBUG] switch_core_session.c:1083 Send signal
> sofia/external/12345 at 192.168.2.3 [BREAK]
> 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
> (sofia/external/12345 at 192.168.2.3) State INIT going to sleep
> 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/12345 at 192.168.2.3) Running State Change CS_ROUTING
> 2010-12-23 14:03:02.510864 [DEBUG] switch_channel.c:1615
> (sofia/external/12345 at 192.168.2.3) Callstate Change DOWN -> RINGING
> 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
> (sofia/external/12345 at 192.168.2.3) State ROUTING
> 2010-12-23 14:03:02.511888 [DEBUG] mod_sofia.c:149
> sofia/external/12345 at 192.168.2.3 SOFIA ROUTING
> 2010-12-23 14:03:02.511888 [DEBUG] switch_ivr_originate.c:66
> (sofia/external/12345 at 192.168.2.3) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2010-12-23 14:03:02.511888 [DEBUG] switch_core_session.c:1083 Send signal
> sofia/external/12345 at 192.168.2.3 [BREAK]
> 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
> (sofia/external/12345 at 192.168.2.3) State ROUTING going to sleep
> 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:320
> (sofia/external/12345 at 192.168.2.3) Running State Change CS_CONSUME_MEDIA
> 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:378
> (sofia/external/12345 at 192.168.2.3) State CONSUME_MEDIA
> 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
> sofia/external/12345 at 192.168.2.3 entering state [calling][0]
> 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
> sofia/external/12345 at 192.168.2.3 entering state [terminated][503]
> 2010-12-23 14:03:02.511888 [DEBUG] switch_channel.c:2493
> (sofia/external/12345 at 192.168.2.3) Callstate Change RINGING -> HANGUP
> 2010-12-23 14:03:02.511888 [NOTICE] sofia.c:5246 Hangup
> sofia/external/12345 at 192.168.2.3 [CS_CONSUME_MEDIA]
> [NORMAL_TEMPORARY_FAILURE]
>

I'm new to FS and I can be wrong here, but as it says in
http://wiki.freeswitch.org/wiki/Hangup_causes NORMAL_TEMPORARY_FAILURE
indicates that the network is not functioning correctly and that the
condition is not likely to last a long period of time; e.g. the user
may wish to try another call attempt almost immediately.
Maybe setting continue_on_fail will help here:
http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail

George


From kbdfck at gmail.com  Thu Dec 23 13:59:54 2010
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 23 Dec 2010 05:59:54 -0500
Subject: [Freeswitch-users] bind_meta_app and bind_digit_action with
 inband DTMF?
In-Reply-To: <AANLkTik1Fuds_ephKqeOQfwi-pB=PpCR=wFnEvdEkiHo@mail.gmail.com>
References: <AANLkTimiDuz8iHLkWL2Z1osyAz1pFQr3jV-hvXw=iGAR@mail.gmail.com>
	<AANLkTik1Fuds_ephKqeOQfwi-pB=PpCR=wFnEvdEkiHo@mail.gmail.com>
Message-ID: <AANLkTimqArJRjqb1M=j=wdw84=b-KCYuAwhuQG+XfHK=@mail.gmail.com>

Thanks for update on bind_digit_action page! But still no luck with it.

My local profile has dtmf-type set to inband, does this setting have
any effect?
I'm trying to turn DTMF detection on with start_dtmf to convert it to
RFC2833 for external profile. Both profiles has rfc2388-pt turned on,
FS complaining that pt will not work on trascoded call when start_dtmf
is present, I think it is just an info message.

I see DTMF events on console being transcoded to RFC2833 but no action
launched when I press 500 during call.

<extension name="local2pstn">
  <condition field="destination_number" expression="^8\d\d\d+$">
  <action application="start_dtmf"/>
  <action application="bind_digit_action"
data="myrealm,500,exec:playback,en/us/callie/ivr/ivr-welcome_to_freeswitch.wav"/>
  <action application="digit_action_set_realm" data="myrealm"/>
  <action application="bridge"
data="sofia/gateway/mediant_2/${destination_number}"/>

</condition>


2010/12/22 Michael Collins <msc at freeswitch.org>:
> FYI,
> I just tested bind_digit_action with inband dtmf and it works splendidly. I
> recommend using bind_digit_action. Also, in case you didn't see it, Brian
> West added a magic extension that looks for the presence of
> "telephone-event" in the SDP and then turns on inband dtmf detection if the
> rtpmap is missing. Look in conf/dialplan/default.xml for the "global"
> extension. The lines that do the auto dtmf thing are commented out by
> default. I just uncommented them and tested with Eyebeam configured for
> inband dtmf. (It also works when Eyebeam is set for 2833...)
> Also, on the FS conf call today the community did a tag-team effort to test
> and document the bind_digit_action app. It has a lot more information than
> it did previously:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_digit_action
> Hope this helps.
> -MC
>
> On Wed, Dec 22, 2010 at 8:01 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>
>> Hi! We are trying to migrate our Asterisk system to Freeswitch, and
>> faced with some trouble:
>>
>> We use inband at user side, and want Freeswitch to regenerate it to
>> RFC2833 before sending to media gateway.
>>
>> When user endpoints use rfc2833 bind_meta_app and bind_digit_action
>> work as expected, but not with start_dtmf (as needed to parse inband
>> DTMF). We can't use rfc2833 DTMF on user endpoints as there seems to
>> be a problem in freeswitch or our media gateway leading to audible
>> DTMF duplicates.
>>
>> When user endpoints are set to inband, I see messages on console about
>> DTMF detection:
>>
>> 2010-12-22 18:50:30.802253 [DEBUG] switch_ivr_async.c:2089 DTMF DETECTED:
>> [*]
>> 2010-12-22 18:50:30.803253 [DEBUG] switch_rtp.c:1985 Send start packet
>> for [*] ts=503795131 dur=160/160/2000 seq=43701
>> 2010-12-22 18:50:30.823282 [DEBUG] switch_rtp.c:1921 Send middle
>> packet for [*] ts=503795131 dur=320/320/2000 seq=43702
>> ...skip...
>> 2010-12-22 18:50:30.923393 [DEBUG] switch_rtp.c:1921 Send middle
>> packet for [*] ts=503795131 dur=1920/1920/2000 seq=43712
>> 2010-12-22 18:50:30.923393 [DEBUG] switch_rtp.c:1921 Send end packet
>> for [*] ts=503795131 dur=2080/2080/2000 seq=43713
>> ...skip...
>> 2010-12-22 18:50:31.242695 [DEBUG] switch_ivr_async.c:2089 DTMF DETECTED:
>> [1]
>> 2010-12-22 18:50:31.242695 [DEBUG] switch_rtp.c:1985 Send start packet
>> for [1] ts=503798651 dur=160/160/2000 seq=43732
>>
>> So digits are received but no in-call action launched by
>> bind_meta_app. I tried start_dtmf before and after bind_meta_app but
>> with no success. Is there a way to make inband dtmf launch in-call
>> actions while transcoding it to RFC2833 on trunk side?
>> I can't make inband dtmf trigger my actions :(
>>
>> --
>> Best regards,
>>
>> Dmitry Sytchev,
>> IT Engineer
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer


From george at ezuce.com  Thu Dec 23 15:36:46 2010
From: george at ezuce.com (George Niculae)
Date: Thu, 23 Dec 2010 14:36:46 +0200
Subject: [Freeswitch-users] call dropped while trying to transfer
In-Reply-To: <AANLkTin0Lmz64n+chbcQ6UqYY704bcim+yOf5AOcV6sv@mail.gmail.com>
References: <AANLkTiku8ZRDxFxQECys14BW94X2J28UpxKZu+p_BOXq@mail.gmail.com>
	<AANLkTin0Lmz64n+chbcQ6UqYY704bcim+yOf5AOcV6sv@mail.gmail.com>
Message-ID: <AANLkTintrgfrhB33pRydJzniksont2BPtu0eVpj+=fUG@mail.gmail.com>

Michael,

the commands are written on socket using PrintWriter.printf() and in
this case is something like:
api uuid_deflect f5539b24-0e8e-11e0-9a0e-c37fe40448c1 sip:101 at dizzy.dizzysip.ro
Please see here all commands sent (prefixed with FSES::cmd):
http://pastebin.freeswitch.org/14868 , uuid deflect at line 27
New console output (for correlating uuid's if needed):
http://pastebin.freeswitch.org/14867

Thanks,
George

On Thu, Dec 23, 2010 at 2:07 AM, Michael Collins <msc at freeswitch.org> wrote:
> Please pastebin the code that performs the uuid_deflect so that we can see
> what you are doing to produce this symptom.
> -MC
>
> On Wed, Dec 22, 2010 at 8:35 AM, George Niculae <george at ezuce.com> wrote:
>>
>> Hi All,
>>
>> I am working on an IVR application based on FS (running FreeSWITCH
>> Version 1.0.head (git-43393f2 2010-12-15 20-59-42 -0600) where the
>> following scenario fails:
>> user 201 calls to 100 (autoattendant), hears menu then press # to
>> transfer to voicemail (101), but the call is dropped (transfer is made
>> using uuid_deflect api command)
>> Dialplan extension configured like:
>>
>> ?<extension name="IVR">
>> ? ?<condition field="destination_number" expression="^IVR${dollar}">
>> ? ? ?<action application="socket" data="192.168.0.114:8084 async full"/>
>> ? ?</condition>
>> ?</extension>
>>
>> Actions taken are:
>> - when call arrives to extension 100 call is bridged
>> (hangup_after_bridge=true)
>> - answer the call, autoattendant menu is played and DTMF collected
>> - when # pressed, call is transfered to 101 using uuid_deflect
>> - call arrives to voicemail extension and is again bridged
>> - call is answered - at this point in time the initial bridge hangs up
>> and the whole call is dropped
>> Please see console output http://pastebin.freeswitch.org/14855
>>
>> When debugging the application, If I keep the first channel connected
>> transfer works just fine without dropping the call.
>> Pretty sure I'm missing something here, any suggestion highly appreciated
>>
>> Thanks,
>> George
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Thu Dec 23 22:15:08 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Dec 2010 11:15:08 -0800
Subject: [Freeswitch-users] Dropping the Subject: header
In-Reply-To: <AF824FE3F385E543ACA98ADFD670BA6A02A07F81@mlbe2k9.cs.myharris.net>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CFF4@mlbe2k9.cs.myharris.net>
	<AANLkTi=zqCe_e5yYMQk1w_U3yEjiCmRm48iFEmuj6iAD@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFF8@mlbe2k9.cs.myharris.net>
	<AANLkTik1oK23B_FzjgZVcPchBnB49BkWO0E7rYJQiytm@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CFFF@mlbe2k9.cs.myharris.net>
	<018408A0-B537-42A2-A8F5-A10B4F0B46FF@freeswitch.org>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D000@mlbe2k9.cs.myharris.net>
	<41C17985-26B1-4DED-B3CA-348DC26FC133@ipeva.fr>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D003@mlbe2k9.cs.myharris.net>
	<AANLkTikD8J_-EgtrheOfqNWOgDMpSYr+sGDkBw6-w_Pp@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D009@mlbe2k9.cs.myharris.net>
	<AANLkTikXa8bXORjRkxgHUcvRBeg7pJ4zLBczK5pOpH_i@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184D00E@mlbe2k9.cs.myharris.net>
	<AANLkTi=OdTTbp+HJLHiwG609pA7pQsvjRQ_kQUyent0V@mail.gmail.com>
	<559A89A6-DD9A-4002-ACB9-5D367D8E2F9C@freeswitch.org>
	<AF824FE3F385E543ACA98ADFD670BA6A02A07F81@mlbe2k9.cs.myharris.net>
Message-ID: <AANLkTi=cT0Zt+EzOeKLG+q5ReP_7d+0WubYQ_NQPNkwh@mail.gmail.com>

A bounty is generally open to any who choose to accept the challenge. If you
specifically want FSS to add the feature then you'd need to contact
support at freeswitch.org and discuss a specific contract, price, etc.

-MC

On Thu, Dec 23, 2010 at 9:51 AM, Kane, Michael (mkane02) <mkane02 at harris.com
> wrote:

> Ok, Brian can you call me at some point to discuss.  321-223-5511.  I have
> no problem posting  as a bounty.
>
> We can roll this into the discussions we've had in the past for support and
> programming support.  The problem is we usually do not deal with independent
> contractors as a rule.  When one posts a bounty are they dealing with
> FreeSWITCH solutions or independent contractors?
>
> Thanks Mike
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Brian
> West
> Sent: Thu 12/23/2010 12:32 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Dropping the Subject: header
>
> He did open a jira as a patch request but no bounty amount on it.  So
> someone will have to patch it for him for free or he'll have to post a
> bounty to raise the priority of the bug a bit because its a non-critical
> issue.  Its at the bottom of my list.
>
> /b
>
> On Dec 23, 2010, at 11:25 AM, Steven Ayre wrote:
>
> > Ah, you're quite right... sorry, I locked onto you saying that you
> > were setting it for a custom app and assumed you were trying to do
> > something outside of the specification.
> >
> > Looking at the source code, it looks like FreeSWITCH doesn't currently
> > provide any way to use that field on an INVITE from mod_sofia... if
> > you wanted it you'd have to open a Jira ticket.
> >
> > http://jira.freeswitch.org/
> >
> > Regards,
> > -Steve
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Dec 23 22:16:01 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Dec 2010 11:16:01 -0800
Subject: [Freeswitch-users] Can't get demo_ivr to work - just goes to
	Music-on-hold
In-Reply-To: <AANLkTimTX_DMzJK97ACJ_9Xmf4BQu4-6f3f4QmVfiVwE@mail.gmail.com>
References: <AANLkTimTX_DMzJK97ACJ_9Xmf4BQu4-6f3f4QmVfiVwE@mail.gmail.com>
Message-ID: <AANLkTi=RckL05HSfuVeGht-tqcFwsc1bNTr_Vf7VVpBX@mail.gmail.com>

I thought the demo IVR was 5000? Also, the screaming monkeys guy shut that
extension down. :(

-MC

On Wed, Dec 22, 2010 at 10:47 PM, Tom C <mel0torme at gmail.com> wrote:

> When I dial extension 6000 to use the demo_ivr, I hear the "please dial it
> now" message.  But no matter what number I dial, it just sends me to Music
> On Hold.
>
> I've tried this on my Dockstar (debian squeeze) and my P4 (debian lenny)
> with the same results.
>
> I want my screaming monkeys!!!  :-)   What am I doing wrong?  Is there
> another module I need to load?  I had this working a week ago.
>
> module_exists mod_flite
> true
>
> Here's the debug log after it plays the initial greeting and I start to
> dial:
> 2010-12-22 22:33:13.592915 [DEBUG] switch_rtp.c:3018 RTP RECV DTMF 5:480
> 2010-12-22 22:33:14.272565 [DEBUG] switch_rtp.c:3018 RTP RECV DTMF #:480
> 2010-12-22 22:33:14.272565 [DEBUG] mod_local_stream.c:421 Opening Stream
> [moh/8000] 8000hz
> 2010-12-22 22:33:14.272565 [DEBUG] switch_ivr_play_say.c:1236 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
> I've tried with internally registered clients as well as an external SIP
> call.  I've re-copied the demo_ivr.xml from the recently git-pulled source
> directory, no help.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Dec 23 22:20:41 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Dec 2010 11:20:41 -0800
Subject: [Freeswitch-users] call dropped while trying to transfer
In-Reply-To: <AANLkTintrgfrhB33pRydJzniksont2BPtu0eVpj+=fUG@mail.gmail.com>
References: <AANLkTiku8ZRDxFxQECys14BW94X2J28UpxKZu+p_BOXq@mail.gmail.com>
	<AANLkTin0Lmz64n+chbcQ6UqYY704bcim+yOf5AOcV6sv@mail.gmail.com>
	<AANLkTintrgfrhB33pRydJzniksont2BPtu0eVpj+=fUG@mail.gmail.com>
Message-ID: <AANLkTinEiSKfA3Aw4LjXQ6cEYdEqiGN87thUUHQEh7QQ@mail.gmail.com>

Try turning on the siptrace as well so we can see the sip traffic:

sofia profile internal siptrace on

Then do another test & pastebin the debug output.
-MC

On Thu, Dec 23, 2010 at 4:36 AM, George Niculae <george at ezuce.com> wrote:

> Michael,
>
> the commands are written on socket using PrintWriter.printf() and in
> this case is something like:
> api uuid_deflect f5539b24-0e8e-11e0-9a0e-c37fe40448c1
> sip:101 at dizzy.dizzysip.ro <sip%3A101 at dizzy.dizzysip.ro>
> Please see here all commands sent (prefixed with FSES::cmd):
> http://pastebin.freeswitch.org/14868 , uuid deflect at line 27
> New console output (for correlating uuid's if needed):
> http://pastebin.freeswitch.org/14867
>
> Thanks,
> George
>
> On Thu, Dec 23, 2010 at 2:07 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> > Please pastebin the code that performs the uuid_deflect so that we can
> see
> > what you are doing to produce this symptom.
> > -MC
> >
> > On Wed, Dec 22, 2010 at 8:35 AM, George Niculae <george at ezuce.com>
> wrote:
> >>
> >> Hi All,
> >>
> >> I am working on an IVR application based on FS (running FreeSWITCH
> >> Version 1.0.head (git-43393f2 2010-12-15 20-59-42 -0600) where the
> >> following scenario fails:
> >> user 201 calls to 100 (autoattendant), hears menu then press # to
> >> transfer to voicemail (101), but the call is dropped (transfer is made
> >> using uuid_deflect api command)
> >> Dialplan extension configured like:
> >>
> >>  <extension name="IVR">
> >>    <condition field="destination_number" expression="^IVR${dollar}">
> >>      <action application="socket" data="192.168.0.114:8084 async
> full"/>
> >>    </condition>
> >>  </extension>
> >>
> >> Actions taken are:
> >> - when call arrives to extension 100 call is bridged
> >> (hangup_after_bridge=true)
> >> - answer the call, autoattendant menu is played and DTMF collected
> >> - when # pressed, call is transfered to 101 using uuid_deflect
> >> - call arrives to voicemail extension and is again bridged
> >> - call is answered - at this point in time the initial bridge hangs up
> >> and the whole call is dropped
> >> Please see console output http://pastebin.freeswitch.org/14855
> >>
> >> When debugging the application, If I keep the first channel connected
> >> transfer works just fine without dropping the call.
> >> Pretty sure I'm missing something here, any suggestion highly
> appreciated
> >>
> >> Thanks,
> >> George
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Dec 23 22:23:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Dec 2010 11:23:16 -0800
Subject: [Freeswitch-users] bind_meta_app and bind_digit_action with
 inband DTMF?
In-Reply-To: <AANLkTimqArJRjqb1M=j=wdw84=b-KCYuAwhuQG+XfHK=@mail.gmail.com>
References: <AANLkTimiDuz8iHLkWL2Z1osyAz1pFQr3jV-hvXw=iGAR@mail.gmail.com>
	<AANLkTik1Fuds_ephKqeOQfwi-pB=PpCR=wFnEvdEkiHo@mail.gmail.com>
	<AANLkTimqArJRjqb1M=j=wdw84=b-KCYuAwhuQG+XfHK=@mail.gmail.com>
Message-ID: <AANLkTikxORmBxY+nJqsiPbpWGp8UFsoKzjdpQ+y56o0S@mail.gmail.com>

FYI, I just added bunch of stuff the the bind_digit_action page yesterday. I
also fixed a few typos. Change the filename in your bind_digit_action line
because this is wrong:
en/us/callie/ivr/ivr-welcome_to_freeswitch.wav

Make it this:
ivr/ivr-welcome_to_freeswitch.wav

Then try again. Also, pastebin the debug output and we'll have a look. (
pastebin.freeswitch.org)
-MC

On Thu, Dec 23, 2010 at 2:59 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> Thanks for update on bind_digit_action page! But still no luck with it.
>
> My local profile has dtmf-type set to inband, does this setting have
> any effect?
> I'm trying to turn DTMF detection on with start_dtmf to convert it to
> RFC2833 for external profile. Both profiles has rfc2388-pt turned on,
> FS complaining that pt will not work on trascoded call when start_dtmf
> is present, I think it is just an info message.
>
> I see DTMF events on console being transcoded to RFC2833 but no action
> launched when I press 500 during call.
>
> <extension name="local2pstn">
>  <condition field="destination_number" expression="^8\d\d\d+$">
>  <action application="start_dtmf"/>
>  <action application="bind_digit_action"
>
> data="myrealm,500,exec:playback,en/us/callie/ivr/ivr-welcome_to_freeswitch.wav"/>
>  <action application="digit_action_set_realm" data="myrealm"/>
>  <action application="bridge"
> data="sofia/gateway/mediant_2/${destination_number}"/>
>
> </condition>
>
>
> 2010/12/22 Michael Collins <msc at freeswitch.org>:
> > FYI,
> > I just tested bind_digit_action with inband dtmf and it works splendidly.
> I
> > recommend using bind_digit_action. Also, in case you didn't see it, Brian
> > West added a magic extension that looks for the presence of
> > "telephone-event" in the SDP and then turns on inband dtmf detection if
> the
> > rtpmap is missing. Look in conf/dialplan/default.xml for the "global"
> > extension. The lines that do the auto dtmf thing are commented out by
> > default. I just uncommented them and tested with Eyebeam configured for
> > inband dtmf. (It also works when Eyebeam is set for 2833...)
> > Also, on the FS conf call today the community did a tag-team effort to
> test
> > and document the bind_digit_action app. It has a lot more information
> than
> > it did previously:
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_digit_action
> > Hope this helps.
> > -MC
> >
> > On Wed, Dec 22, 2010 at 8:01 AM, Dmitry Sytchev <kbdfck at gmail.com>
> wrote:
> >>
> >> Hi! We are trying to migrate our Asterisk system to Freeswitch, and
> >> faced with some trouble:
> >>
> >> We use inband at user side, and want Freeswitch to regenerate it to
> >> RFC2833 before sending to media gateway.
> >>
> >> When user endpoints use rfc2833 bind_meta_app and bind_digit_action
> >> work as expected, but not with start_dtmf (as needed to parse inband
> >> DTMF). We can't use rfc2833 DTMF on user endpoints as there seems to
> >> be a problem in freeswitch or our media gateway leading to audible
> >> DTMF duplicates.
> >>
> >> When user endpoints are set to inband, I see messages on console about
> >> DTMF detection:
> >>
> >> 2010-12-22 18:50:30.802253 [DEBUG] switch_ivr_async.c:2089 DTMF
> DETECTED:
> >> [*]
> >> 2010-12-22 18:50:30.803253 [DEBUG] switch_rtp.c:1985 Send start packet
> >> for [*] ts=503795131 dur=160/160/2000 seq=43701
> >> 2010-12-22 18:50:30.823282 [DEBUG] switch_rtp.c:1921 Send middle
> >> packet for [*] ts=503795131 dur=320/320/2000 seq=43702
> >> ...skip...
> >> 2010-12-22 18:50:30.923393 [DEBUG] switch_rtp.c:1921 Send middle
> >> packet for [*] ts=503795131 dur=1920/1920/2000 seq=43712
> >> 2010-12-22 18:50:30.923393 [DEBUG] switch_rtp.c:1921 Send end packet
> >> for [*] ts=503795131 dur=2080/2080/2000 seq=43713
> >> ...skip...
> >> 2010-12-22 18:50:31.242695 [DEBUG] switch_ivr_async.c:2089 DTMF
> DETECTED:
> >> [1]
> >> 2010-12-22 18:50:31.242695 [DEBUG] switch_rtp.c:1985 Send start packet
> >> for [1] ts=503798651 dur=160/160/2000 seq=43732
> >>
> >> So digits are received but no in-call action launched by
> >> bind_meta_app. I tried start_dtmf before and after bind_meta_app but
> >> with no success. Is there a way to make inband dtmf launch in-call
> >> actions while transcoding it to RFC2833 on trunk side?
> >> I can't make inband dtmf trigger my actions :(
> >>
> >> --
> >> Best regards,
> >>
> >> Dmitry Sytchev,
> >> IT Engineer
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Dec 23 22:25:39 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Dec 2010 11:25:39 -0800
Subject: [Freeswitch-users] Loosing DTMF Digits in XML IVR
In-Reply-To: <AANLkTim98bvSVJQWdKFcvTsZk8uJd3OsPZvK-omLTh4=@mail.gmail.com>
References: <AANLkTim98bvSVJQWdKFcvTsZk8uJd3OsPZvK-omLTh4=@mail.gmail.com>
Message-ID: <AANLkTimt8wrSVDBy480R9XwP2WpV3GKdeQ7VCLQQpLeN@mail.gmail.com>

Are you using inband or 2833 for DTMFs? How are you creating your IVR - with
the XML IVR or with something else? Get a console debug and put it on
pastebin.freeswitch.org so we can see what's going on.

-MC

On Thu, Dec 23, 2010 at 3:19 AM, afshin afzali <a.afzali2003 at gmail.com>wrote:

> Hi Guys,
>
> After updating to latest freeswitch (git-34a0ca5 2010-12-22 20-38-57 -0600)
> , I've noticed that I'm loosing some of the dtmf digits in my IVR. Actually
> freeswitch looses most of them! My freeswitch hosted by ubuntu 10.04 64bit
> Server.
>
> Appreciate all comments,
>
> Regards,
> -- afshin
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dmitry.bely at gmail.com  Thu Dec 23 22:51:42 2010
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Thu, 23 Dec 2010 22:51:42 +0300
Subject: [Freeswitch-users] FreeSWITCH as Debian Package?
In-Reply-To: <BB497725-7989-44D6-8321-27B424861945@jerris.com>
References: <20100119103139.GR4767@tamay-dogan.net>
	<2D5C208F-292E-462A-BFA9-DFD20CB09954@jerris.com>
	<20100120180720.GH4767@tamay-dogan.net>
	<BB497725-7989-44D6-8321-27B424861945@jerris.com>
Message-ID: <AANLkTi=mNg3m=ZSAXyCnN6_tONFX0oc1fjoNMKUPfr-Y@mail.gmail.com>

On Thu, Jan 21, 2010 at 9:48 AM, Michael Jerris <mike at jerris.com> wrote:
> We have a hudson instance doing builds, we just need boxes that can be the build drones. ?And someone with a little time > to set it up and bandwidth to handle it. ?we build every 30 min that there is a change to svn, which is depending how long > the builds take, usually 20+ times a day and upload the build results to the hudson server.

Although http://www.freeswitch.de:8080 hudson seems to regularly build
FreeSWITCH .deb packages from git sources they are not available for
download anymore. Is there another place to grab them?

- Dmitry Bely


From george.niculae79 at gmail.com  Thu Dec 23 22:56:45 2010
From: george.niculae79 at gmail.com (George Niculae)
Date: Thu, 23 Dec 2010 21:56:45 +0200
Subject: [Freeswitch-users] call dropped while trying to transfer
In-Reply-To: <AANLkTinEiSKfA3Aw4LjXQ6cEYdEqiGN87thUUHQEh7QQ@mail.gmail.com>
References: <AANLkTiku8ZRDxFxQECys14BW94X2J28UpxKZu+p_BOXq@mail.gmail.com>
	<AANLkTin0Lmz64n+chbcQ6UqYY704bcim+yOf5AOcV6sv@mail.gmail.com>
	<AANLkTintrgfrhB33pRydJzniksont2BPtu0eVpj+=fUG@mail.gmail.com>
	<AANLkTinEiSKfA3Aw4LjXQ6cEYdEqiGN87thUUHQEh7QQ@mail.gmail.com>
Message-ID: <AANLkTi==EKk_igs7kNnzgiS8XZs9S5L_Dp1=NcXBma0+@mail.gmail.com>

Here it is: http://pastebin.freeswitch.org/14873

<http://pastebin.freeswitch.org/14873>Thanks,
George

2010/12/23 Michael Collins <msc at freeswitch.org>

> Try turning on the siptrace as well so we can see the sip traffic:
>
> sofia profile internal siptrace on
>
> Then do another test & pastebin the debug output.
> -MC
>
>
> On Thu, Dec 23, 2010 at 4:36 AM, George Niculae <george at ezuce.com> wrote:
>
>> Michael,
>>
>> the commands are written on socket using PrintWriter.printf() and in
>> this case is something like:
>> api uuid_deflect f5539b24-0e8e-11e0-9a0e-c37fe40448c1
>> sip:101 at dizzy.dizzysip.ro <sip%3A101 at dizzy.dizzysip.ro>
>> Please see here all commands sent (prefixed with FSES::cmd):
>> http://pastebin.freeswitch.org/14868 , uuid deflect at line 27
>> New console output (for correlating uuid's if needed):
>> http://pastebin.freeswitch.org/14867
>>
>> Thanks,
>> George
>>
>> On Thu, Dec 23, 2010 at 2:07 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > Please pastebin the code that performs the uuid_deflect so that we can
>> see
>> > what you are doing to produce this symptom.
>> > -MC
>> >
>> > On Wed, Dec 22, 2010 at 8:35 AM, George Niculae <george at ezuce.com>
>> wrote:
>> >>
>> >> Hi All,
>> >>
>> >> I am working on an IVR application based on FS (running FreeSWITCH
>> >> Version 1.0.head (git-43393f2 2010-12-15 20-59-42 -0600) where the
>> >> following scenario fails:
>> >> user 201 calls to 100 (autoattendant), hears menu then press # to
>> >> transfer to voicemail (101), but the call is dropped (transfer is made
>> >> using uuid_deflect api command)
>> >> Dialplan extension configured like:
>> >>
>> >>  <extension name="IVR">
>> >>    <condition field="destination_number" expression="^IVR${dollar}">
>> >>      <action application="socket" data="192.168.0.114:8084 async
>> full"/>
>> >>    </condition>
>> >>  </extension>
>> >>
>> >> Actions taken are:
>> >> - when call arrives to extension 100 call is bridged
>> >> (hangup_after_bridge=true)
>> >> - answer the call, autoattendant menu is played and DTMF collected
>> >> - when # pressed, call is transfered to 101 using uuid_deflect
>> >> - call arrives to voicemail extension and is again bridged
>> >> - call is answered - at this point in time the initial bridge hangs up
>> >> and the whole call is dropped
>> >> Please see console output http://pastebin.freeswitch.org/14855
>> >>
>> >> When debugging the application, If I keep the first channel connected
>> >> transfer works just fine without dropping the call.
>> >> Pretty sure I'm missing something here, any suggestion highly
>> appreciated
>> >>
>> >> Thanks,
>> >> George
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From a.afzali2003 at gmail.com  Thu Dec 23 23:25:34 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Thu, 23 Dec 2010 23:55:34 +0330
Subject: [Freeswitch-users] Loosing DTMF Digits in XML IVR
In-Reply-To: <AANLkTimt8wrSVDBy480R9XwP2WpV3GKdeQ7VCLQQpLeN@mail.gmail.com>
References: <AANLkTim98bvSVJQWdKFcvTsZk8uJd3OsPZvK-omLTh4=@mail.gmail.com>
	<AANLkTimt8wrSVDBy480R9XwP2WpV3GKdeQ7VCLQQpLeN@mail.gmail.com>
Message-ID: <AANLkTinNuSc4MFfz9rCP1Qj6WYmmoWW3eAqmNL5_YQvr@mail.gmail.com>

Hi,

I use eyeBeam to test my application. As I checked, it is enabled in both by
default. my IVR is in XML format, also I experience this behavior by the
demo IVR ( 5000 ) as well. I tried to test digit detection just by one of
inbound / 2833 format. The result was that one of those did not work at all.
I'll provide you debug log soon.

Thanks MC,
-- afshin

On Thu, Dec 23, 2010 at 10:55 PM, Michael Collins <msc at freeswitch.org>wrote:

> Are you using inband or 2833 for DTMFs? How are you creating your IVR -
> with the XML IVR or with something else? Get a console debug and put it on
> pastebin.freeswitch.org so we can see what's going on.
>
> -MC
>
> On Thu, Dec 23, 2010 at 3:19 AM, afshin afzali <a.afzali2003 at gmail.com>wrote:
>
>> Hi Guys,
>>
>> After updating to latest freeswitch (git-34a0ca5 2010-12-22 20-38-57
>> -0600) , I've noticed that I'm loosing some of the dtmf digits in my IVR.
>> Actually freeswitch looses most of them! My freeswitch hosted by ubuntu
>> 10.04 64bit Server.
>>
>> Appreciate all comments,
>>
>> Regards,
>> -- afshin
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Dec 23 23:37:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Dec 2010 12:37:26 -0800
Subject: [Freeswitch-users] ESL outbound /inbound
In-Reply-To: <COL122-W2256918ACD58AE62DC48CDB51C0@phx.gbl>
References: <COL122-W2256918ACD58AE62DC48CDB51C0@phx.gbl>
Message-ID: <AANLkTi=etPHjn63mMxNFYZJRH+Pch+o_3RMuU2xrViK_@mail.gmail.com>

What is wrong with having Lua do the db lookup?
-MC

On Thu, Dec 23, 2010 at 9:42 AM, Rafqat . <rafonline at hotmail.com> wrote:

>  Hi,
>
> I am writing a calling card app using lua and java.
>
> As advised, Java will predominatley do all the billing side of things (via
> ESL) leaving lua do simple things like asking for the calling card pin
> number etc.
>
> When a call comes in, my lua script answers the call and asks the user for
> a pin.  Instead of querying the DB inline from within lua, I would like my
> app server to do this (please let me know if this should be done inline
> instead).  I understand my app server (Java ESL inbound socket) can register
> for a pin checking custom event and I can generate such an event from within
> my lua script.  My lua script would then wait for an appropriate repsonse
> event:
>
> -- Check if pin is valid
> local event = freeswitch.Event("CUSTOM", "check_pin_request");
> event:addHeader("pin_number", digits);
> event:fire();
>
> -- wait for response
> con = freeswitch.EventConsumer("CUSTOM", "check_pin_reponse");
> con:pop(1);
> print("event\n" .. e:serialize("xml"));
>
> I was wondering if the above is the right way of doing things, or should I
> be using ESLOutboundSocket and have lua script do something like this
> instead:
>
> session:execute("set", "pin_to_check=12345");
> session:execute("socket", "192.168.0.2:8084");
>
> Not sure how lua will be told whether the pin is valid or not in this
> scenario.
>
> I appreciate FreeSWITCH is very flexible and would like to make sure I
> develop a scalable and performant application.
>
> Any help will be much appreciated.
>
> Cheers
>
> Raf
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rafonline at hotmail.com  Fri Dec 24 00:07:17 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Thu, 23 Dec 2010 21:07:17 +0000
Subject: [Freeswitch-users] ESL outbound /inbound
In-Reply-To: <AANLkTi=etPHjn63mMxNFYZJRH+Pch+o_3RMuU2xrViK_@mail.gmail.com>
References: <COL122-W2256918ACD58AE62DC48CDB51C0@phx.gbl>,
	<AANLkTi=etPHjn63mMxNFYZJRH+Pch+o_3RMuU2xrViK_@mail.gmail.com>
Message-ID: <COL122-W13FC929C897D9C705EFDAEB51C0@phx.gbl>


Hi,

I could get lua to do the lookup, I was wondering i guess where does one draw the line between when to do stuff inline and when to do stuff outside of freeswitch (in terms of load and scalability of solution etc.)

Cheers

Raf

Date: Thu, 23 Dec 2010 12:37:26 -0800
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] ESL outbound /inbound

What is wrong with having Lua do the db lookup? -MC

On Thu, Dec 23, 2010 at 9:42 AM, Rafqat . <rafonline at hotmail.com> wrote:






Hi,

I am writing a calling card app using lua and java.  

As advised, Java will predominatley do all the billing side of things (via ESL) leaving lua do simple things like asking for the calling card pin number etc.


When a call comes in, my lua script answers the call and asks the user for a pin.  Instead of querying the DB inline from within lua, I would like my app server to do this (please let me know if this should be done inline instead).  I understand my app server (Java ESL inbound socket) can register for a pin checking custom event and I can generate such an event from within my lua script.  My lua script would then wait for an appropriate repsonse event:


-- Check if pin is valid
local event = freeswitch.Event("CUSTOM", "check_pin_request");
event:addHeader("pin_number", digits);
event:fire();

-- wait for response
con = freeswitch.EventConsumer("CUSTOM", "check_pin_reponse");

con:pop(1);
print("event\n" .. e:serialize("xml"));

I was wondering if the above is the right way of doing things, or should I be using ESLOutboundSocket and have lua script do something like this instead:


session:execute("set", "pin_to_check=12345");
session:execute("socket", "192.168.0.2:8084");

Not sure how lua will be told whether the pin is valid or not in this scenario.


I appreciate FreeSWITCH is very flexible and would like to make sure I develop a scalable and performant application.

Any help will be much appreciated.

Cheers

Raf



 		 	   		  


_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
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From steveayre at gmail.com  Fri Dec 24 00:27:28 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 23 Dec 2010 21:27:28 +0000
Subject: [Freeswitch-users] FreeSWITCH as Debian Package?
In-Reply-To: <AANLkTi=mNg3m=ZSAXyCnN6_tONFX0oc1fjoNMKUPfr-Y@mail.gmail.com>
References: <20100119103139.GR4767@tamay-dogan.net>
	<2D5C208F-292E-462A-BFA9-DFD20CB09954@jerris.com>
	<20100120180720.GH4767@tamay-dogan.net>
	<BB497725-7989-44D6-8321-27B424861945@jerris.com>
	<AANLkTi=mNg3m=ZSAXyCnN6_tONFX0oc1fjoNMKUPfr-Y@mail.gmail.com>
Message-ID: <AANLkTikE2ZSd0UuUj6LwHApMpAi5voecE0Y9Fu2osm3E@mail.gmail.com>

You can build them yourself from the source (which is how Hudson does it).

$ git clone git://git.freeswitch.org/freeswitch.git freeswitch
$ cd freeswitch
$ dpkg-buildpackage -uc -us

(There are of course a few -dev packages you'll need to install first,
dpkg-buildpackage should stop and tell you which ones are missing).

Warm regards,
-Steve



On 23 December 2010 19:51, Dmitry Bely <dmitry.bely at gmail.com> wrote:
> On Thu, Jan 21, 2010 at 9:48 AM, Michael Jerris <mike at jerris.com> wrote:
>> We have a hudson instance doing builds, we just need boxes that can be the build drones. ?And someone with a little time > to set it up and bandwidth to handle it. ?we build every 30 min that there is a change to svn, which is depending how long > the builds take, usually 20+ times a day and upload the build results to the hudson server.
>
> Although http://www.freeswitch.de:8080 hudson seems to regularly build
> FreeSWITCH .deb packages from git sources they are not available for
> download anymore. Is there another place to grab them?
>
> - Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From michal.bielicki at seventhsignal.de  Fri Dec 24 00:46:17 2010
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Thu, 23 Dec 2010 22:46:17 +0100
Subject: [Freeswitch-users] FreeSWITCH as Debian Package?
In-Reply-To: <AANLkTikE2ZSd0UuUj6LwHApMpAi5voecE0Y9Fu2osm3E@mail.gmail.com>
References: <20100119103139.GR4767@tamay-dogan.net>
	<2D5C208F-292E-462A-BFA9-DFD20CB09954@jerris.com>
	<20100120180720.GH4767@tamay-dogan.net>
	<BB497725-7989-44D6-8321-27B424861945@jerris.com>
	<AANLkTi=mNg3m=ZSAXyCnN6_tONFX0oc1fjoNMKUPfr-Y@mail.gmail.com>
	<AANLkTikE2ZSd0UuUj6LwHApMpAi5voecE0Y9Fu2osm3E@mail.gmail.com>
Message-ID: <55D85001-65F9-41F2-9C8A-C940C5D56A9A@seventhsignal.de>

the ones from www.freeswitch.de:8080 are being pushed into this place:
http://repo.freeswitch.de/apt

Am 23.12.2010 um 22:27 schrieb Steven Ayre:

> You can build them yourself from the source (which is how Hudson does it).
> 
> $ git clone git://git.freeswitch.org/freeswitch.git freeswitch
> $ cd freeswitch
> $ dpkg-buildpackage -uc -us
> 
> (There are of course a few -dev packages you'll need to install first,
> dpkg-buildpackage should stop and tell you which ones are missing).
> 
> Warm regards,
> -Steve
> 
> 
> 
> On 23 December 2010 19:51, Dmitry Bely <dmitry.bely at gmail.com> wrote:
>> On Thu, Jan 21, 2010 at 9:48 AM, Michael Jerris <mike at jerris.com> wrote:
>>> We have a hudson instance doing builds, we just need boxes that can be the build drones.  And someone with a little time > to set it up and bandwidth to handle it.  we build every 30 min that there is a change to svn, which is depending how long > the builds take, usually 20+ times a day and upload the build results to the hudson server.
>> 
>> Although http://www.freeswitch.de:8080 hudson seems to regularly build
>> FreeSWITCH .deb packages from git sources they are not available for
>> download anymore. Is there another place to grab them?
>> 
>> - Dmitry Bely
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115
D-12059 Berlin
Voice: +49 30 60988730
Amtsgericht Berlin Charlottenburg HRA 44413 B, Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From michal.bielicki at seventhsignal.de  Fri Dec 24 00:47:33 2010
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Thu, 23 Dec 2010 22:47:33 +0100
Subject: [Freeswitch-users] FreeSWITCH debs and rpms
Message-ID: <CD470134-1990-4F78-8617-55C0FB1BD2EC@seventhsignal.de>

debs and rpms are available from git builds from
http://repo.freeswitch.de/
the files are kept for 14 days

the centos rpms are build with sangoma libsng_isdn and libsng_ss7 as well as wanpipe.
radius and openr2 are next on my list to be added.


Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115
D-12059 Berlin
Voice: +49 30 60988730
Amtsgericht Berlin Charlottenburg HRA 44413 B, Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From lloyd.aloysius at gmail.com  Fri Dec 24 00:49:15 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 23 Dec 2010 16:49:15 -0500
Subject: [Freeswitch-users] CentOS 5.5 - FreeSWITCH - Audio Quality issues
Message-ID: <AANLkTik3y8EfTi5s_a_1BURnUJe8Nv0p+5y8Ph6Jkcg3@mail.gmail.com>

Hi All,

*FreeSWITCH environment*

Hardware - Intel Core 2 Quad 2.4GHz , 4GB RAM
OS - CentOS 5.5
FreeSWITCH Version 1.0.head (git-fcd6c54 2010-12-19 00-13-08 -0500)
Ethernet : Public IP + 10Mbps port


Calls Between Phones and outside calls perfect.


But Voice Quality is not great in the following scenarios

1. Voice Mails ... most of the voicemails breaking voices .. lots of static
2. IVR ... breaking voices
3. System Automatically Hangup ... while people leaving messages..



Do I need to adjust any parameters ?

Any help is appreciated.

Thanks
Lloyd
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From mel0torme at gmail.com  Fri Dec 24 01:29:59 2010
From: mel0torme at gmail.com (Tom C)
Date: Thu, 23 Dec 2010 14:29:59 -0800
Subject: [Freeswitch-users] Can't get demo_ivr to work - just goes to
	Music-on-hold
In-Reply-To: <AANLkTi=RckL05HSfuVeGht-tqcFwsc1bNTr_Vf7VVpBX@mail.gmail.com>
References: <AANLkTimTX_DMzJK97ACJ_9Xmf4BQu4-6f3f4QmVfiVwE@mail.gmail.com>
	<AANLkTi=RckL05HSfuVeGht-tqcFwsc1bNTr_Vf7VVpBX@mail.gmail.com>
Message-ID: <AANLkTikRgXNKg5uofnTEVbxymc3Z5cA+2+VVv=ggdfZ1@mail.gmail.com>

>>>  I thought the demo IVR was 5000?

AAAAAAAAAAAAAAAAAARGHHHHHHHHH!!!!!!!!!!!!!!!!!

Crawling under rock.

Thank you.  :-)  I can't believe I wasted 3 hours trying to debug this.



On Thu, Dec 23, 2010 at 11:16 AM, Michael Collins <msc at freeswitch.org>wrote:

> I thought the demo IVR was 5000? Also, the screaming monkeys guy shut that
> extension down. :(
>
> -MC
>
>   On Wed, Dec 22, 2010 at 10:47 PM, Tom C <mel0torme at gmail.com> wrote:
>
>>  When I dial extension 6000 to use the demo_ivr, I hear the "please dial
>> it now" message.  But no matter what number I dial, it just sends me to
>> Music On Hold.
>>
>> I've tried this on my Dockstar (debian squeeze) and my P4 (debian lenny)
>> with the same results.
>>
>> I want my screaming monkeys!!!  :-)   What am I doing wrong?  Is there
>> another module I need to load?  I had this working a week ago.
>>
>> module_exists mod_flite
>> true
>>
>> Here's the debug log after it plays the initial greeting and I start to
>> dial:
>> 2010-12-22 22:33:13.592915 [DEBUG] switch_rtp.c:3018 RTP RECV DTMF 5:480
>> 2010-12-22 22:33:14.272565 [DEBUG] switch_rtp.c:3018 RTP RECV DTMF #:480
>> 2010-12-22 22:33:14.272565 [DEBUG] mod_local_stream.c:421 Opening Stream
>> [moh/8000] 8000hz
>> 2010-12-22 22:33:14.272565 [DEBUG] switch_ivr_play_say.c:1236 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>>
>> I've tried with internally registered clients as well as an external SIP
>> call.  I've re-copied the demo_ivr.xml from the recently git-pulled source
>> directory, no help.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gevlichenko at gmail.com  Fri Dec 24 01:10:05 2010
From: gevlichenko at gmail.com (Alexandr Gevlichenko)
Date: Fri, 24 Dec 2010 01:10:05 +0300
Subject: [Freeswitch-users] X-FS-SUPPORT header
Message-ID: <4D13C8BD.1070504@gmail.com>

Hi list

Freeswitch adds to all INVITE messages header:
X-FS-Support: update_display.

How can I remove it from the INVITE message?
Thanks in advance.


From chris at cloudtel.com  Fri Dec 24 02:39:19 2010
From: chris at cloudtel.com (Chris Burns)
Date: Thu, 23 Dec 2010 18:39:19 -0500
Subject: [Freeswitch-users] ESL outbound /inbound
In-Reply-To: <COL122-W13FC929C897D9C705EFDAEB51C0@phx.gbl>
References: <COL122-W2256918ACD58AE62DC48CDB51C0@phx.gbl>
	<AANLkTi=etPHjn63mMxNFYZJRH+Pch+o_3RMuU2xrViK_@mail.gmail.com>
	<COL122-W13FC929C897D9C705EFDAEB51C0@phx.gbl>
Message-ID: <AANLkTi=CUMxo357GyBk2Q08kzA5FWtFRMnyrhi0aprQs@mail.gmail.com>

Billing is not so much a question of scalability, but accuracy. Inline lua
can verify the PIN against the DB, but it cant accurately calculate billing.

Anything making a calculation on how much money your customer just spent on
minutes should not be done inline, as many tricks can happen and channels
have their own state (CS_REPORTING) when the optimal information for
accounting is available. For prompting the user for their calling card PIN,
there is nothing wrong with lua connecting to your database. For subtracting
minutes from an account while the user is on call then event socket or a
module listening to events is a better method. The latter is how
mod_nibblebill works.

Personally, for several large projects I chose to monitor events across
multiple switches via event socket and it worked out very nicely.

On Thu, Dec 23, 2010 at 4:07 PM, Rafqat . <rafonline at hotmail.com> wrote:

>  Hi,
>
> I could get lua to do the lookup, I was wondering i guess where does one
> draw the line between when to do stuff inline and when to do stuff outside
> of freeswitch (in terms of load and scalability of solution etc.)
>
> Cheers
>
> Raf
>
> ------------------------------
> Date: Thu, 23 Dec 2010 12:37:26 -0800
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] ESL outbound /inbound
>
>
> What is wrong with having Lua do the db lookup?
> -MC
>
> On Thu, Dec 23, 2010 at 9:42 AM, Rafqat . <rafonline at hotmail.com> wrote:
>
>  Hi,
>
> I am writing a calling card app using lua and java.
>
> As advised, Java will predominatley do all the billing side of things (via
> ESL) leaving lua do simple things like asking for the calling card pin
> number etc.
>
> When a call comes in, my lua script answers the call and asks the user for
> a pin.  Instead of querying the DB inline from within lua, I would like my
> app server to do this (please let me know if this should be done inline
> instead).  I understand my app server (Java ESL inbound socket) can register
> for a pin checking custom event and I can generate such an event from within
> my lua script.  My lua script would then wait for an appropriate repsonse
> event:
>
> -- Check if pin is valid
> local event = freeswitch.Event("CUSTOM", "check_pin_request");
> event:addHeader("pin_number", digits);
> event:fire();
>
> -- wait for response
> con = freeswitch.EventConsumer("CUSTOM", "check_pin_reponse");
> con:pop(1);
> print("event\n" .. e:serialize("xml"));
>
> I was wondering if the above is the right way of doing things, or should I
> be using ESLOutboundSocket and have lua script do something like this
> instead:
>
> session:execute("set", "pin_to_check=12345");
> session:execute("socket", "192.168.0.2:8084");
>
> Not sure how lua will be told whether the pin is valid or not in this
> scenario.
>
> I appreciate FreeSWITCH is very flexible and would like to make sure I
> develop a scalable and performant application.
>
> Any help will be much appreciated.
>
> Cheers
>
> Raf
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dmitry.bely at gmail.com  Fri Dec 24 02:59:01 2010
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Fri, 24 Dec 2010 02:59:01 +0300
Subject: [Freeswitch-users] FreeSWITCH as Debian Package?
In-Reply-To: <55D85001-65F9-41F2-9C8A-C940C5D56A9A@seventhsignal.de>
References: <20100119103139.GR4767@tamay-dogan.net>
	<2D5C208F-292E-462A-BFA9-DFD20CB09954@jerris.com>
	<20100120180720.GH4767@tamay-dogan.net>
	<BB497725-7989-44D6-8321-27B424861945@jerris.com>
	<AANLkTi=mNg3m=ZSAXyCnN6_tONFX0oc1fjoNMKUPfr-Y@mail.gmail.com>
	<AANLkTikE2ZSd0UuUj6LwHApMpAi5voecE0Y9Fu2osm3E@mail.gmail.com>
	<55D85001-65F9-41F2-9C8A-C940C5D56A9A@seventhsignal.de>
Message-ID: <AANLkTi=Ypp_kgghv=bTTY96f4XXrpJ4VdPgTwfY=ig8T@mail.gmail.com>

On Fri, Dec 24, 2010 at 12:46 AM, Michal Bielicki
<michal.bielicki at seventhsignal.de> wrote:
> the ones from www.freeswitch.de:8080 are being pushed into this place:
> http://repo.freeswitch.de/apt

Thanks, that's exactly I was looking for.

- Dmitry Bely


From michal.bielicki at seventhsignal.de  Fri Dec 24 03:27:18 2010
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Fri, 24 Dec 2010 01:27:18 +0100
Subject: [Freeswitch-users] FreeSWITCH as Debian Package?
In-Reply-To: <AANLkTi=Ypp_kgghv=bTTY96f4XXrpJ4VdPgTwfY=ig8T@mail.gmail.com>
References: <20100119103139.GR4767@tamay-dogan.net>
	<2D5C208F-292E-462A-BFA9-DFD20CB09954@jerris.com>
	<20100120180720.GH4767@tamay-dogan.net>
	<BB497725-7989-44D6-8321-27B424861945@jerris.com>
	<AANLkTi=mNg3m=ZSAXyCnN6_tONFX0oc1fjoNMKUPfr-Y@mail.gmail.com>
	<AANLkTikE2ZSd0UuUj6LwHApMpAi5voecE0Y9Fu2osm3E@mail.gmail.com>
	<55D85001-65F9-41F2-9C8A-C940C5D56A9A@seventhsignal.de>
	<AANLkTi=Ypp_kgghv=bTTY96f4XXrpJ4VdPgTwfY=ig8T@mail.gmail.com>
Message-ID: <EA6A3218-93C4-46EB-B061-D16D97AA3EF3@seventhsignal.de>

et me know how it goes, I am pretty pro at rpm and sysv pkg but debian/ubuntu is still strange to me ;)

Am 24.12.2010 um 00:59 schrieb Dmitry Bely:

> On Fri, Dec 24, 2010 at 12:46 AM, Michal Bielicki
> <michal.bielicki at seventhsignal.de> wrote:
>> the ones from www.freeswitch.de:8080 are being pushed into this place:
>> http://repo.freeswitch.de/apt
> 
> Thanks, that's exactly I was looking for.
> 
> - Dmitry Bely
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115
D-12059 Berlin
Voice: +49 30 60988730
Amtsgericht Berlin Charlottenburg HRA 44413 B, Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From msc at freeswitch.org  Fri Dec 24 03:27:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Dec 2010 16:27:26 -0800
Subject: [Freeswitch-users] CentOS 5.5 - FreeSWITCH - Audio Quality
	issues
In-Reply-To: <AANLkTik3y8EfTi5s_a_1BURnUJe8Nv0p+5y8Ph6Jkcg3@mail.gmail.com>
References: <AANLkTik3y8EfTi5s_a_1BURnUJe8Nv0p+5y8Ph6Jkcg3@mail.gmail.com>
Message-ID: <AANLkTi=6shTJg0C2eFsar+mdjstT330e2kaUtFxsPQ1F@mail.gmail.com>

Just a casual observation, but it looks like all three of these symptoms
have disk I/O in common. Perhaps your disk performance isn't up to snuff?
You could probably test this by moving some of your sound files into a
ramdisk and then calling in to listen to them...

-MC

On Thu, Dec 23, 2010 at 1:49 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> *FreeSWITCH environment*
>
> Hardware - Intel Core 2 Quad 2.4GHz , 4GB RAM
> OS - CentOS 5.5
> FreeSWITCH Version 1.0.head (git-fcd6c54 2010-12-19 00-13-08 -0500)
> Ethernet : Public IP + 10Mbps port
>
>
> Calls Between Phones and outside calls perfect.
>
>
> But Voice Quality is not great in the following scenarios
>
> 1. Voice Mails ... most of the voicemails breaking voices .. lots of static
> 2. IVR ... breaking voices
> 3. System Automatically Hangup ... while people leaving messages..
>
>
>
> Do I need to adjust any parameters ?
>
> Any help is appreciated.
>
> Thanks
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Dec 24 03:28:19 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Dec 2010 16:28:19 -0800
Subject: [Freeswitch-users] Can't get demo_ivr to work - just goes to
	Music-on-hold
In-Reply-To: <AANLkTikRgXNKg5uofnTEVbxymc3Z5cA+2+VVv=ggdfZ1@mail.gmail.com>
References: <AANLkTimTX_DMzJK97ACJ_9Xmf4BQu4-6f3f4QmVfiVwE@mail.gmail.com>
	<AANLkTi=RckL05HSfuVeGht-tqcFwsc1bNTr_Vf7VVpBX@mail.gmail.com>
	<AANLkTikRgXNKg5uofnTEVbxymc3Z5cA+2+VVv=ggdfZ1@mail.gmail.com>
Message-ID: <AANLkTimh2LfZdSuV3_zb_bT4CatAV6EWEjf9gk5mtJ4X@mail.gmail.com>

No prob. If I charged you by the hour to fix this then my bill would be,
appropriately, $0.02. :)
-MC

On Thu, Dec 23, 2010 at 2:29 PM, Tom C <mel0torme at gmail.com> wrote:

> >>>  I thought the demo IVR was 5000?
>
> AAAAAAAAAAAAAAAAAARGHHHHHHHHH!!!!!!!!!!!!!!!!!
>
> Crawling under rock.
>
> Thank you.  :-)  I can't believe I wasted 3 hours trying to debug this.
>
>
>
> On Thu, Dec 23, 2010 at 11:16 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> I thought the demo IVR was 5000? Also, the screaming monkeys guy shut that
>> extension down. :(
>>
>> -MC
>>
>>   On Wed, Dec 22, 2010 at 10:47 PM, Tom C <mel0torme at gmail.com> wrote:
>>
>>>  When I dial extension 6000 to use the demo_ivr, I hear the "please dial
>>> it now" message.  But no matter what number I dial, it just sends me to
>>> Music On Hold.
>>>
>>> I've tried this on my Dockstar (debian squeeze) and my P4 (debian lenny)
>>> with the same results.
>>>
>>> I want my screaming monkeys!!!  :-)   What am I doing wrong?  Is there
>>> another module I need to load?  I had this working a week ago.
>>>
>>> module_exists mod_flite
>>> true
>>>
>>> Here's the debug log after it plays the initial greeting and I start to
>>> dial:
>>> 2010-12-22 22:33:13.592915 [DEBUG] switch_rtp.c:3018 RTP RECV DTMF 5:480
>>> 2010-12-22 22:33:14.272565 [DEBUG] switch_rtp.c:3018 RTP RECV DTMF #:480
>>> 2010-12-22 22:33:14.272565 [DEBUG] mod_local_stream.c:421 Opening Stream
>>> [moh/8000] 8000hz
>>> 2010-12-22 22:33:14.272565 [DEBUG] switch_ivr_play_say.c:1236 Codec
>>> Activated L16 at 8000hz 1 channels 20ms
>>>
>>> I've tried with internally registered clients as well as an external SIP
>>> call.  I've re-copied the demo_ivr.xml from the recently git-pulled source
>>> directory, no help.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Dec 24 03:31:27 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Dec 2010 16:31:27 -0800
Subject: [Freeswitch-users] X-FS-SUPPORT header
In-Reply-To: <4D13C8BD.1070504@gmail.com>
References: <4D13C8BD.1070504@gmail.com>
Message-ID: <AANLkTikcsCYoeDXzrHECgxGjYcBGo8Q8wucGCcsAN_Yj@mail.gmail.com>

Is there a problem with this particular X header? If you have a device that
is choking on this header then you need to go after the developers with a
machine gun. :)

-MC

On Thu, Dec 23, 2010 at 2:10 PM, Alexandr Gevlichenko <gevlichenko at gmail.com
> wrote:

> Hi list
>
> Freeswitch adds to all INVITE messages header:
> X-FS-Support: update_display.
>
> How can I remove it from the INVITE message?
> Thanks in advance.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Fri Dec 24 03:34:52 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 23 Dec 2010 18:34:52 -0600
Subject: [Freeswitch-users] X-FS-SUPPORT header
In-Reply-To: <AANLkTikcsCYoeDXzrHECgxGjYcBGo8Q8wucGCcsAN_Yj@mail.gmail.com>
References: <4D13C8BD.1070504@gmail.com>
	<AANLkTikcsCYoeDXzrHECgxGjYcBGo8Q8wucGCcsAN_Yj@mail.gmail.com>
Message-ID: <617FB25D-F12A-457D-A803-6CFD4CCCDC8A@freeswitch.org>

I have a baseball bat he can go beat the device to death with... its legal but some stuff does croak on it... an option exists to disable it but I can't recall it off the top of my head.

/b

On Dec 23, 2010, at 6:31 PM, Michael Collins wrote:

> Is there a problem with this particular X header? If you have a device that is choking on this header then you need to go after the developers with a machine gun. :)
> 
> -MC



From msc at freeswitch.org  Fri Dec 24 03:37:08 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Dec 2010 16:37:08 -0800
Subject: [Freeswitch-users] X-FS-SUPPORT header
In-Reply-To: <617FB25D-F12A-457D-A803-6CFD4CCCDC8A@freeswitch.org>
References: <4D13C8BD.1070504@gmail.com>
	<AANLkTikcsCYoeDXzrHECgxGjYcBGo8Q8wucGCcsAN_Yj@mail.gmail.com>
	<617FB25D-F12A-457D-A803-6CFD4CCCDC8A@freeswitch.org>
Message-ID: <AANLkTimNgwr3qLfntJ0ojK6pUQa4LSyuRHuwkbo2YgLw@mail.gmail.com>

okay, i'm digging through configs, trying to find it...
-MC

On Thu, Dec 23, 2010 at 4:34 PM, Brian West <brian at freeswitch.org> wrote:

> I have a baseball bat he can go beat the device to death with... its legal
> but some stuff does croak on it... an option exists to disable it but I
> can't recall it off the top of my head.
>
> /b
>
> On Dec 23, 2010, at 6:31 PM, Michael Collins wrote:
>
> > Is there a problem with this particular X header? If you have a device
> that is choking on this header then you need to go after the developers with
> a machine gun. :)
> >
> > -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Dec 24 03:52:14 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Dec 2010 16:52:14 -0800
Subject: [Freeswitch-users] X-FS-SUPPORT header
In-Reply-To: <AANLkTimNgwr3qLfntJ0ojK6pUQa4LSyuRHuwkbo2YgLw@mail.gmail.com>
References: <4D13C8BD.1070504@gmail.com>
	<AANLkTikcsCYoeDXzrHECgxGjYcBGo8Q8wucGCcsAN_Yj@mail.gmail.com>
	<617FB25D-F12A-457D-A803-6CFD4CCCDC8A@freeswitch.org>
	<AANLkTimNgwr3qLfntJ0ojK6pUQa4LSyuRHuwkbo2YgLw@mail.gmail.com>
Message-ID: <AANLkTikC-_Z1PtgmWRgkX-kzRO2xcs8vG2408OEtDjoV@mail.gmail.com>

Try this, hopefully it will work:
http://wiki.freeswitch.org/wiki/Variable_ignore_display_updates

-MC

On Thu, Dec 23, 2010 at 4:37 PM, Michael Collins <msc at freeswitch.org> wrote:

> okay, i'm digging through configs, trying to find it...
> -MC
>
>
> On Thu, Dec 23, 2010 at 4:34 PM, Brian West <brian at freeswitch.org> wrote:
>
>> I have a baseball bat he can go beat the device to death with... its legal
>> but some stuff does croak on it... an option exists to disable it but I
>> can't recall it off the top of my head.
>>
>> /b
>>
>> On Dec 23, 2010, at 6:31 PM, Michael Collins wrote:
>>
>> > Is there a problem with this particular X header? If you have a device
>> that is choking on this header then you need to go after the developers with
>> a machine gun. :)
>> >
>> > -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From infos at madovsky.org  Fri Dec 24 07:27:42 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 23 Dec 2010 23:27:42 -0500
Subject: [Freeswitch-users] nibblebilll low balance
Message-ID: <BEF8B14394C04087A1569578BEE53DA3@e1705>

I have this config in nibblebill.conf.xml
-----

    <!-- By default, warn a caller when their balance is at $5.00. You can set this to a negative number. -->
    <param name="lowbal_amt" value="1"/>
    <param name="lowbal_action" value="lowbalance XML default"/>

----

But no effect if the cash field starts to be under 1
I'm wondering if the cash field has to be exactly 1 if not it fails ?

THanks
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From dmitry.bely at gmail.com  Fri Dec 24 11:41:18 2010
From: dmitry.bely at gmail.com (Dmitry Bely)
Date: Fri, 24 Dec 2010 11:41:18 +0300
Subject: [Freeswitch-users] FreeSWITCH as Debian Package?
In-Reply-To: <EA6A3218-93C4-46EB-B061-D16D97AA3EF3@seventhsignal.de>
References: <20100119103139.GR4767@tamay-dogan.net>
	<2D5C208F-292E-462A-BFA9-DFD20CB09954@jerris.com>
	<20100120180720.GH4767@tamay-dogan.net>
	<BB497725-7989-44D6-8321-27B424861945@jerris.com>
	<AANLkTi=mNg3m=ZSAXyCnN6_tONFX0oc1fjoNMKUPfr-Y@mail.gmail.com>
	<AANLkTikE2ZSd0UuUj6LwHApMpAi5voecE0Y9Fu2osm3E@mail.gmail.com>
	<55D85001-65F9-41F2-9C8A-C940C5D56A9A@seventhsignal.de>
	<AANLkTi=Ypp_kgghv=bTTY96f4XXrpJ4VdPgTwfY=ig8T@mail.gmail.com>
	<EA6A3218-93C4-46EB-B061-D16D97AA3EF3@seventhsignal.de>
Message-ID: <AANLkTi=emCRO1DMQ2Z2MEBA=2YAtRvRa+ba7=wTk7OJS@mail.gmail.com>

On Fri, Dec 24, 2010 at 3:27 AM, Michal Bielicki
<michal.bielicki at seventhsignal.de> wrote:
> et me know how it goes, I am pretty pro at rpm and sysv pkg but
> debian/ubuntu is still strange to me ;)

Seems to be OK. The only problem encountered so far was packages
version/naming: I had to force "downgrade" in aptitude while replacing
the September hudson packages with the current ones.

- Dmitry Bely


From u2nsam at gmail.com  Fri Dec 24 13:27:20 2010
From: u2nsam at gmail.com (Sam)
Date: Fri, 24 Dec 2010 15:57:20 +0530
Subject: [Freeswitch-users] no invite send
In-Reply-To: <AANLkTik5c65RBdq6K3JP9jJyXxywWEuAg8Yuq9hrY96t@mail.gmail.com>
References: <AANLkTimBh+CqTz1O5qNmd7=z+Z6WZGeY1ua8kdZAwzH5@mail.gmail.com>
	<AANLkTik5c65RBdq6K3JP9jJyXxywWEuAg8Yuq9hrY96t@mail.gmail.com>
Message-ID: <AANLkTi=Thxn4FBBwaZZ-aMqUhZT8eXuifV1u8tCMEcTp@mail.gmail.com>

Hi,

How to find current version of freeswitch installed ?

Regards
Sam


On Thu, Dec 23, 2010 at 3:10 PM, George Niculae <george at ezuce.com> wrote:

> On Thu, Dec 23, 2010 at 11:20 AM, Sam <u2nsam at gmail.com> wrote:
> > Hello friends,
> >
> > I am using the dialplan,
> >
> > <action application="bridge" data="sofia/external/12345 at 192.168.2.3"/>
> >
> > its not sending INVITE to the server 192.168.2.3
> >
> > i am getting the below logs:-
> >
> > 2010-12-23 14:03:02.509500 [NOTICE] switch_channel.c:784 New Channel
> > sofia/external/12345 at 192.168.2.3 [418e0dfa-0e6f-11e0-a5e1-33ac6c473a4e]
> > 2010-12-23 14:03:02.509500 [DEBUG] mod_sofia.c:4052
> > (sofia/external/12345 at 192.168.2.3) State Change CS_NEW -> CS_INIT
> > 2010-12-23 14:03:02.509500 [DEBUG] switch_core_session.c:1083 Send signal
> > sofia/external/12345 at 192.168.2.3 [BREAK]
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
> > (sofia/external/12345 at 192.168.2.3) Running State Change CS_INIT
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
> > (sofia/external/12345 at 192.168.2.3) State INIT
> > 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:86
> > sofia/external/12345 at 192.168.2.3 SOFIA INIT
> > 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:126
> > (sofia/external/12345 at 192.168.2.3) State Change CS_INIT -> CS_ROUTING
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_session.c:1083 Send signal
> > sofia/external/12345 at 192.168.2.3 [BREAK]
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
> > (sofia/external/12345 at 192.168.2.3) State INIT going to sleep
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
> > (sofia/external/12345 at 192.168.2.3) Running State Change CS_ROUTING
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_channel.c:1615
> > (sofia/external/12345 at 192.168.2.3) Callstate Change DOWN -> RINGING
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
> > (sofia/external/12345 at 192.168.2.3) State ROUTING
> > 2010-12-23 14:03:02.511888 [DEBUG] mod_sofia.c:149
> > sofia/external/12345 at 192.168.2.3 SOFIA ROUTING
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_ivr_originate.c:66
> > (sofia/external/12345 at 192.168.2.3) State Change CS_ROUTING ->
> > CS_CONSUME_MEDIA
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_session.c:1083 Send signal
> > sofia/external/12345 at 192.168.2.3 [BREAK]
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
> > (sofia/external/12345 at 192.168.2.3) State ROUTING going to sleep
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:320
> > (sofia/external/12345 at 192.168.2.3) Running State Change CS_CONSUME_MEDIA
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:378
> > (sofia/external/12345 at 192.168.2.3) State CONSUME_MEDIA
> > 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
> > sofia/external/12345 at 192.168.2.3 entering state [calling][0]
> > 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
> > sofia/external/12345 at 192.168.2.3 entering state [terminated][503]
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_channel.c:2493
> > (sofia/external/12345 at 192.168.2.3) Callstate Change RINGING -> HANGUP
> > 2010-12-23 14:03:02.511888 [NOTICE] sofia.c:5246 Hangup
> > sofia/external/12345 at 192.168.2.3 [CS_CONSUME_MEDIA]
> > [NORMAL_TEMPORARY_FAILURE]
> >
>
> I'm new to FS and I can be wrong here, but as it says in
> http://wiki.freeswitch.org/wiki/Hangup_causes NORMAL_TEMPORARY_FAILURE
> indicates that the network is not functioning correctly and that the
> condition is not likely to last a long period of time; e.g. the user
> may wish to try another call attempt almost immediately.
> Maybe setting continue_on_fail will help here:
> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>
> George
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From u2nsam at gmail.com  Fri Dec 24 13:40:13 2010
From: u2nsam at gmail.com (Sam)
Date: Fri, 24 Dec 2010 16:10:13 +0530
Subject: [Freeswitch-users] DTMF missing
Message-ID: <AANLkTik=NLa+injL5hF=VsymwoE9wd1Wb2joXJDhvfFZ@mail.gmail.com>

Hi,,


I have installed the latest ver of freeswitch and i have configured the
conference.

now when i punch in the digits for password , i could see that the DTMF
digits are missed on fs_cli.

it only happens when i dial it from polycom or cisco phones.

I have tried with and without these values below:-

<param name="dtmf-type" value="rfc2833"/>
    <param name="pass-rfc2833" value="true"/>



traces fetched:
192.168.2.49:5060 -> 192.168.2.190:5060
  INVITE sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190> SIP/2.0..Via:
SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From: "7028" <
sip:7028 at 192.168.2.190
<sip%3A7028 at 192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To:
<sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>..Call-ID:
0017592a-eb33001a-
  63da3294-1a7bfdfa at 192.168.2.49..Max-Forwards: 70..Date: Fri, 24 Dec 2010
10:13:52 GMT..CSeq: 102 INVITE..User-Agent: Cisco-CP7940G/8.0..Contact:
<sip:7028 at 192.168.2.49:5060;transport=udp>..Proxy-Authorization: Digest
username="7028"
  ,realm="192.168.2.190",uri="sip:7050 at 192.168.2.190<sip%3A7050 at 192.168.2.190>",response="a668f5c480285b35e7ff6bcd446879f0",nonce="d2c540f2-8487-4d87-bdab-871585253eb8",cnonce="0a6c4176",qop=auth,nc=00000001,algorithm=MD5..Expires:
180..Accept: application/sdp
  ..Allow:
ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported:
replaces,join,norefersub..Content-Length: 220..Content-Type:
application/sdp..Content-Disposition:
session;handling=optional....v=0..o=Cisco-SIPUA 16102
   0 IN IP4 192.168.2.49..s=SIP Call..t=0 0..m=audio 17298 RTP/AVP 0 8
18..c=IN IP4 192.168.2.49..a=rtpmap:0 PCMU/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=sendrecv..


192.168.2.190:5060 -> 192.168.2.49:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.49:5060;branch=z9hG4bK1531f395..From:
"7028" <sip:7028 at 192.168.2.190
<sip%3A7028 at 192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To:
<sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;tag=2XXUZpgr1rvgc..Call-ID:
0017592a-eb33001a-63da3
  294-1a7bfdfa at 192.168.2.49..CSeq: 102 INVITE..Contact:
<sip:7050 at 192.168.2.190:5060;transport=udp>..User-Agent: NOVANET..Accept:
application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
INFO, REGISTER, REFER, NOTIF
  Y, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path,
replaces..Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
sla, include-session-description, presence.winfo, message-summary,
refer..Session-Expires: 180
  0;refresher=uas..Min-SE: 120..Content-Type:
application/sdp..Content-Disposition: session..Content-Length:
249..Remote-Party-ID: "7050"
<sip:7050 at 192.168.2.190<sip%3A7050 at 192.168.2.190>>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH
1293163579 129
  3163580 IN IP4 192.168.2.190..s=FreeSWITCH..c=IN IP4 192.168.2.190..t=0
0..m=audio 22050 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - -
-..a=ptime:20..

But the dtmf are not missed when punched on eyebeam softphone.

And all the phones have RFC 2833.

traces fetched for softphone:-

192.168.2.17:6182 -> 192.168.2.190:5060
  INVITE sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190> SIP/2.0..To: <
sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>..From: 7001<
sip:7001 at 192.168.2.190 <sip%3A7001 at 192.168.2.190>>;tag=6c557c1e..Via:
SIP/2.0/UDP 192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport..Call-ID:
d32ffe546570a77e..CS
  eq: 2 INVITE..Contact: <sip:7001 at 192.168.2.17:6182>..Max-Forwards:
70..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Authorization: Digest
username="7001",rea
  lm="192.168.2.190",nonce="d0b4db6e-bd76-447f-9076-2e6b7809cb54",uri="
sip:7050 at 192.168.2.190
<sip%3A7050 at 192.168.2.190>",response="45fb9eb4f0e0e4fffd87a22769a007ba",cnonce="1c27f3687059b16d",nc=00000001,qop=auth,algorithm=MD5..User-Agent:
eyeBeam release 3007n
   stamp 17816..Content-Length: 233....v=0..o=- 27833664 27833670 IN IP4
192.168.2.17..s=eyeBeam..c=IN IP4 192.168.2.17..t=0 0..m=audio 6398 RTP/AVP
0 18 101..a=alt:1 1 : 2C830AD9 0000004F 192.168.2.17 6398..a=fmtp:101
0-15..a=rtpmap:
  101 telephone-event/8000..a=sendrecv..


192.168.2.190:5060 -> 192.168.2.17:6182
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport=6182..From:
7001 <sip:7001 at 192.168.2.190 <sip%3A7001 at 192.168.2.190>>;tag=6c557c1e..To: <
sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;tag=H0ctQv7KNgU2j..Call-ID:
d32ffe546570a77e..C
  Seq: 2 INVITE..Contact:
<sip:7050 at 192.168.2.190:5060;transport=udp>..User-Agent:
NOVANET..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Support
  ed: timer, precondition, path, replaces..Allow-Events: talk, hold,
presence, dialog, line-seize, call-info, sla, include-session-description,
presence.winfo, message-summary, refer..Session-Expires:
1800;refresher=uas..Min-SE: 120..
  Content-Type: application/sdp..Content-Disposition:
session..Content-Length: 249..Remote-Party-ID: "7050" <
sip:7050 at 192.168.2.190
<sip%3A7050 at 192.168.2.190>>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH
1293161640 1293161641 IN IP4 192.168.2.190..
  s=FreeSWITCH..c=IN IP4 192.168.2.190..t=0 0..m=audio 24852 RTP/AVP 0
101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-16..a=silenceSupp:off - - - -..a=ptime:20..


Any thing you can think how it can happen?


Regards
Sam
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From rafonline at hotmail.com  Fri Dec 24 14:55:55 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Fri, 24 Dec 2010 11:55:55 +0000
Subject: [Freeswitch-users] ESL outbound /inbound
In-Reply-To: <AANLkTi=CUMxo357GyBk2Q08kzA5FWtFRMnyrhi0aprQs@mail.gmail.com>
References: <COL122-W2256918ACD58AE62DC48CDB51C0@phx.gbl>,
	<AANLkTi=etPHjn63mMxNFYZJRH+Pch+o_3RMuU2xrViK_@mail.gmail.com>,
	<COL122-W13FC929C897D9C705EFDAEB51C0@phx.gbl>,
	<AANLkTi=CUMxo357GyBk2Q08kzA5FWtFRMnyrhi0aprQs@mail.gmail.com>
Message-ID: <COL122-W4463935CBB0A4B25F9BE75B51D0@phx.gbl>


Hi Chris

In terms of the projects you eluded too, do you use inbound or outbound event sockets?  The reason I am asking is that it seems to me that either could be used for billing monitoring.  At the moment I am thinking in terms of purely using inbound mode to listen to events (and controlling what freeswitch does via sendmsg).  

I would appreciate it if you could clarify this point for me as I am failing to understand the true difference between inbound and outbound mode and when each should be used (there seems to be alot of overlap imo - correct me if i am wrong).

Cheers

Raf

Date: Thu, 23 Dec 2010 18:39:19 -0500
From: chris at cloudtel.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] ESL outbound /inbound

Billing is not so much a question of scalability, but accuracy.
Inline lua can verify the PIN against the DB, but it cant accurately calculate billing.

Anything making a calculation on how much money your customer just spent on minutes should not be done inline, as many tricks can happen and channels have their own state (CS_REPORTING) when the optimal information for accounting is available. For prompting the user for their calling card PIN, there is nothing wrong with lua connecting to your database. For subtracting minutes from an account while the user is on call then event socket or a module listening to events is a better method. The latter is how mod_nibblebill works.


Personally, for several large projects I chose to monitor events across multiple switches via event socket and it worked out very nicely.

On Thu, Dec 23, 2010 at 4:07 PM, Rafqat . <rafonline at hotmail.com> wrote:






Hi,

I could get lua to do the lookup, I was wondering i guess where does one draw the line between when to do stuff inline and when to do stuff outside of freeswitch (in terms of load and scalability of solution etc.)


Cheers

Raf

Date: Thu, 23 Dec 2010 12:37:26 -0800
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] ESL outbound /inbound

What is wrong with having Lua do the db lookup? -MC

On Thu, Dec 23, 2010 at 9:42 AM, Rafqat . <rafonline at hotmail.com> wrote:







Hi,

I am writing a calling card app using lua and java.  

As advised, Java will predominatley do all the billing side of things (via ESL) leaving lua do simple things like asking for the calling card pin number etc.



When a call comes in, my lua script answers the call and asks the user for a pin.  Instead of querying the DB inline from within lua, I would like my app server to do this (please let me know if this should be done inline instead).  I understand my app server (Java ESL inbound socket) can register for a pin checking custom event and I can generate such an event from within my lua script.  My lua script would then wait for an appropriate repsonse event:



-- Check if pin is valid
local event = freeswitch.Event("CUSTOM", "check_pin_request");
event:addHeader("pin_number", digits);
event:fire();

-- wait for response
con = freeswitch.EventConsumer("CUSTOM", "check_pin_reponse");


con:pop(1);
print("event\n" .. e:serialize("xml"));

I was wondering if the above is the right way of doing things, or should I be using ESLOutboundSocket and have lua script do something like this instead:



session:execute("set", "pin_to_check=12345");
session:execute("socket", "192.168.0.2:8084");

Not sure how lua will be told whether the pin is valid or not in this scenario.



I appreciate FreeSWITCH is very flexible and would like to make sure I develop a scalable and performant application.

Any help will be much appreciated.

Cheers

Raf



 		 	   		  



_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





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

_______________________________________________

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FreeSWITCH-users at lists.freeswitch.org

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

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From steveayre at gmail.com  Fri Dec 24 16:37:49 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 24 Dec 2010 13:37:49 +0000
Subject: [Freeswitch-users] no invite send
In-Reply-To: <AANLkTi=Thxn4FBBwaZZ-aMqUhZT8eXuifV1u8tCMEcTp@mail.gmail.com>
References: <AANLkTimBh+CqTz1O5qNmd7=z+Z6WZGeY1ua8kdZAwzH5@mail.gmail.com>
	<AANLkTik5c65RBdq6K3JP9jJyXxywWEuAg8Yuq9hrY96t@mail.gmail.com>
	<AANLkTi=Thxn4FBBwaZZ-aMqUhZT8eXuifV1u8tCMEcTp@mail.gmail.com>
Message-ID: <AANLkTik+Gj2zV4c3Sj+c_JWt=QELeMGEKSxmHyzNOWOf@mail.gmail.com>

>From console/fs_cli run 'version'

-Steve



On 24 December 2010 10:27, Sam <u2nsam at gmail.com> wrote:
> Hi,
>
> How to find current version of freeswitch installed ?
>
> Regards
> Sam
>
>
> On Thu, Dec 23, 2010 at 3:10 PM, George Niculae <george at ezuce.com> wrote:
>>
>> On Thu, Dec 23, 2010 at 11:20 AM, Sam <u2nsam at gmail.com> wrote:
>> > Hello friends,
>> >
>> > I am using the dialplan,
>> >
>> > <action application="bridge" data="sofia/external/12345 at 192.168.2.3"/>
>> >
>> > its not sending INVITE to the server 192.168.2.3
>> >
>> > i am getting the below logs:-
>> >
>> > 2010-12-23 14:03:02.509500 [NOTICE] switch_channel.c:784 New Channel
>> > sofia/external/12345 at 192.168.2.3 [418e0dfa-0e6f-11e0-a5e1-33ac6c473a4e]
>> > 2010-12-23 14:03:02.509500 [DEBUG] mod_sofia.c:4052
>> > (sofia/external/12345 at 192.168.2.3) State Change CS_NEW -> CS_INIT
>> > 2010-12-23 14:03:02.509500 [DEBUG] switch_core_session.c:1083 Send
>> > signal
>> > sofia/external/12345 at 192.168.2.3 [BREAK]
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
>> > (sofia/external/12345 at 192.168.2.3) Running State Change CS_INIT
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
>> > (sofia/external/12345 at 192.168.2.3) State INIT
>> > 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:86
>> > sofia/external/12345 at 192.168.2.3 SOFIA INIT
>> > 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:126
>> > (sofia/external/12345 at 192.168.2.3) State Change CS_INIT -> CS_ROUTING
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_session.c:1083 Send
>> > signal
>> > sofia/external/12345 at 192.168.2.3 [BREAK]
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
>> > (sofia/external/12345 at 192.168.2.3) State INIT going to sleep
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
>> > (sofia/external/12345 at 192.168.2.3) Running State Change CS_ROUTING
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_channel.c:1615
>> > (sofia/external/12345 at 192.168.2.3) Callstate Change DOWN -> RINGING
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
>> > (sofia/external/12345 at 192.168.2.3) State ROUTING
>> > 2010-12-23 14:03:02.511888 [DEBUG] mod_sofia.c:149
>> > sofia/external/12345 at 192.168.2.3 SOFIA ROUTING
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_ivr_originate.c:66
>> > (sofia/external/12345 at 192.168.2.3) State Change CS_ROUTING ->
>> > CS_CONSUME_MEDIA
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_session.c:1083 Send
>> > signal
>> > sofia/external/12345 at 192.168.2.3 [BREAK]
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
>> > (sofia/external/12345 at 192.168.2.3) State ROUTING going to sleep
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:320
>> > (sofia/external/12345 at 192.168.2.3) Running State Change CS_CONSUME_MEDIA
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:378
>> > (sofia/external/12345 at 192.168.2.3) State CONSUME_MEDIA
>> > 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
>> > sofia/external/12345 at 192.168.2.3 entering state [calling][0]
>> > 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
>> > sofia/external/12345 at 192.168.2.3 entering state [terminated][503]
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_channel.c:2493
>> > (sofia/external/12345 at 192.168.2.3) Callstate Change RINGING -> HANGUP
>> > 2010-12-23 14:03:02.511888 [NOTICE] sofia.c:5246 Hangup
>> > sofia/external/12345 at 192.168.2.3 [CS_CONSUME_MEDIA]
>> > [NORMAL_TEMPORARY_FAILURE]
>> >
>>
>> I'm new to FS and I can be wrong here, but as it says in
>> http://wiki.freeswitch.org/wiki/Hangup_causes NORMAL_TEMPORARY_FAILURE
>> indicates that the network is not functioning correctly and that the
>> condition is not likely to last a long period of time; e.g. the user
>> may wish to try another call attempt almost immediately.
>> Maybe setting continue_on_fail will help here:
>> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>
>> George
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From michal.bielicki at seventhsignal.de  Fri Dec 24 16:46:26 2010
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Fri, 24 Dec 2010 14:46:26 +0100
Subject: [Freeswitch-users] FreeSWITCH as Debian Package?
In-Reply-To: <AANLkTi=emCRO1DMQ2Z2MEBA=2YAtRvRa+ba7=wTk7OJS@mail.gmail.com>
References: <20100119103139.GR4767@tamay-dogan.net>
	<2D5C208F-292E-462A-BFA9-DFD20CB09954@jerris.com>
	<20100120180720.GH4767@tamay-dogan.net>
	<BB497725-7989-44D6-8321-27B424861945@jerris.com>
	<AANLkTi=mNg3m=ZSAXyCnN6_tONFX0oc1fjoNMKUPfr-Y@mail.gmail.com>
	<AANLkTikE2ZSd0UuUj6LwHApMpAi5voecE0Y9Fu2osm3E@mail.gmail.com>
	<55D85001-65F9-41F2-9C8A-C940C5D56A9A@seventhsignal.de>
	<AANLkTi=Ypp_kgghv=bTTY96f4XXrpJ4VdPgTwfY=ig8T@mail.gmail.com>
	<EA6A3218-93C4-46EB-B061-D16D97AA3EF3@seventhsignal.de>
	<AANLkTi=emCRO1DMQ2Z2MEBA=2YAtRvRa+ba7=wTk7OJS@mail.gmail.com>
Message-ID: <E334A2CB-68E4-47A4-949B-0943ACAEE86A@seventhsignal.de>

yah I had to redo the struct so that it does all the packages the same. Not sure I am doing it correctly but since it seems to work I will leave it like it is for now.
The line I use in sources.list is:
deb http://repo.freeswitch.de/apt lenny non-free main contrib

Am 24.12.2010 um 09:41 schrieb Dmitry Bely:

> On Fri, Dec 24, 2010 at 3:27 AM, Michal Bielicki
> <michal.bielicki at seventhsignal.de> wrote:
>> et me know how it goes, I am pretty pro at rpm and sysv pkg but
>> debian/ubuntu is still strange to me ;)
> 
> Seems to be OK. The only problem encountered so far was packages
> version/naming: I had to force "downgrade" in aptitude while replacing
> the September hudson packages with the current ones.
> 
> - Dmitry Bely

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115
D-12059 Berlin
Voice: +49 30 60988730
Amtsgericht Berlin Charlottenburg HRA 44413 B, Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From infos at madovsky.org  Fri Dec 24 20:58:53 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 24 Dec 2010 12:58:53 -0500
Subject: [Freeswitch-users] nibblebill schedule
Message-ID: <F07B572C409042239691B1F3DE08575E@e1705>

Is it possible to schedule the transfer to nibblebill nobal_action ?
I'd like to schedule of 30s before the call is cut and go to nobal_action extension

thanks
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From chris at cloudtel.com  Fri Dec 24 23:20:00 2010
From: chris at cloudtel.com (Chris Burns)
Date: Fri, 24 Dec 2010 15:20:00 -0500
Subject: [Freeswitch-users] ESL outbound /inbound
In-Reply-To: <COL122-W4463935CBB0A4B25F9BE75B51D0@phx.gbl>
References: <COL122-W2256918ACD58AE62DC48CDB51C0@phx.gbl>
	<AANLkTi=etPHjn63mMxNFYZJRH+Pch+o_3RMuU2xrViK_@mail.gmail.com>
	<COL122-W13FC929C897D9C705EFDAEB51C0@phx.gbl>
	<AANLkTi=CUMxo357GyBk2Q08kzA5FWtFRMnyrhi0aprQs@mail.gmail.com>
	<COL122-W4463935CBB0A4B25F9BE75B51D0@phx.gbl>
Message-ID: <AANLkTikZiZ7VBasTMOkvU5Fsxn_z+o7H6_kqNx0UGA35@mail.gmail.com>

I use inbound to monitor switches as a socket client and mod_xml_curl to
direct call flow from a seperate web app. This allows me to receive events
across multiple channels across multiple switches without worrying about
call flow, which is a good design for what I am doing in this example ...
monitoring call flow and conferencing events for realtime display in a UI.
Our call flow systems are critical and monitoring is non-critical so it
works well to seperate them. As a calling card app your system will probably
require tighter integration of call flow and billing, so combining them
could make sense.

The main benefit of outbound is you can use a socket server and each of your
clients represents a distinct channel on one of your switches. It can make
directing call flow easy and it is simple to require your socket server or
play an "oops we are busy" message when it is down ... and it helped
everyone quickly port their FAGI apps to this project :)

You are correct that you can just use event socket and sendmsg to control
call flow and not use lua at all. Yes there is overlap of functionality
between inbound/outbound because both allow you to do basically the same
thing, but as different roles in the socket connection.

On Fri, Dec 24, 2010 at 6:55 AM, Rafqat . <rafonline at hotmail.com> wrote:

>  Hi Chris
>
> In terms of the projects you eluded too, do you use inbound or outbound
> event sockets?  The reason I am asking is that it seems to me that either
> could be used for billing monitoring.  At the moment I am thinking in terms
> of purely using inbound mode to listen to events (and controlling what
> freeswitch does via sendmsg).
>
> I would appreciate it if you could clarify this point for me as I am
> failing to understand the true difference between inbound and outbound mode
> and when each should be used (there seems to be alot of overlap imo -
> correct me if i am wrong).
>
> Cheers
>
> Raf
>
> ------------------------------
> Date: Thu, 23 Dec 2010 18:39:19 -0500
> From: chris at cloudtel.com
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] ESL outbound /inbound
>
> Billing is not so much a question of scalability, but accuracy. Inline lua
> can verify the PIN against the DB, but it cant accurately calculate billing.
>
> Anything making a calculation on how much money your customer just spent on
> minutes should not be done inline, as many tricks can happen and channels
> have their own state (CS_REPORTING) when the optimal information for
> accounting is available. For prompting the user for their calling card PIN,
> there is nothing wrong with lua connecting to your database. For subtracting
> minutes from an account while the user is on call then event socket or a
> module listening to events is a better method. The latter is how
> mod_nibblebill works.
>
> Personally, for several large projects I chose to monitor events across
> multiple switches via event socket and it worked out very nicely.
>
> On Thu, Dec 23, 2010 at 4:07 PM, Rafqat . <rafonline at hotmail.com> wrote:
>
>  Hi,
>
> I could get lua to do the lookup, I was wondering i guess where does one
> draw the line between when to do stuff inline and when to do stuff outside
> of freeswitch (in terms of load and scalability of solution etc.)
>
> Cheers
>
> Raf
>
> ------------------------------
> Date: Thu, 23 Dec 2010 12:37:26 -0800
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] ESL outbound /inbound
>
>
> What is wrong with having Lua do the db lookup?
> -MC
>
> On Thu, Dec 23, 2010 at 9:42 AM, Rafqat . <rafonline at hotmail.com> wrote:
>
>  Hi,
>
> I am writing a calling card app using lua and java.
>
> As advised, Java will predominatley do all the billing side of things (via
> ESL) leaving lua do simple things like asking for the calling card pin
> number etc.
>
> When a call comes in, my lua script answers the call and asks the user for
> a pin.  Instead of querying the DB inline from within lua, I would like my
> app server to do this (please let me know if this should be done inline
> instead).  I understand my app server (Java ESL inbound socket) can register
> for a pin checking custom event and I can generate such an event from within
> my lua script.  My lua script would then wait for an appropriate repsonse
> event:
>
> -- Check if pin is valid
> local event = freeswitch.Event("CUSTOM", "check_pin_request");
> event:addHeader("pin_number", digits);
> event:fire();
>
> -- wait for response
> con = freeswitch.EventConsumer("CUSTOM", "check_pin_reponse");
> con:pop(1);
> print("event\n" .. e:serialize("xml"));
>
> I was wondering if the above is the right way of doing things, or should I
> be using ESLOutboundSocket and have lua script do something like this
> instead:
>
> session:execute("set", "pin_to_check=12345");
> session:execute("socket", "192.168.0.2:8084");
>
> Not sure how lua will be told whether the pin is valid or not in this
> scenario.
>
> I appreciate FreeSWITCH is very flexible and would like to make sure I
> develop a scalable and performant application.
>
> Any help will be much appreciated.
>
> Cheers
>
> Raf
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rafonline at hotmail.com  Sat Dec 25 00:09:36 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Fri, 24 Dec 2010 21:09:36 +0000
Subject: [Freeswitch-users] ESL outbound /inbound
In-Reply-To: <AANLkTikZiZ7VBasTMOkvU5Fsxn_z+o7H6_kqNx0UGA35@mail.gmail.com>
References: <COL122-W2256918ACD58AE62DC48CDB51C0@phx.gbl>,
	<AANLkTi=etPHjn63mMxNFYZJRH+Pch+o_3RMuU2xrViK_@mail.gmail.com>,
	<COL122-W13FC929C897D9C705EFDAEB51C0@phx.gbl>,
	<AANLkTi=CUMxo357GyBk2Q08kzA5FWtFRMnyrhi0aprQs@mail.gmail.com>,
	<COL122-W4463935CBB0A4B25F9BE75B51D0@phx.gbl>,
	<AANLkTikZiZ7VBasTMOkvU5Fsxn_z+o7H6_kqNx0UGA35@mail.gmail.com>
Message-ID: <COL122-W60C04326D9849EE2C4A06EB51D0@phx.gbl>



Hi Chris,

Thanks for your advice. 

In terms of tighter integration and control, do you think using inbound event socket for billing and 
sendmsg to control call flow would be a good approach for a calling card application.? I have no problem using lua but feel that once i get passed the trivial stuff of checking account funds etc. I may have logic scattered around in more than one place.

Sorry for being a pita but your advice (and the freeswitch community's advice)  is invaluable to a newbie like me.

Thanks

Raf


Date: Fri, 24 Dec 2010 15:20:00 -0500
From: chris at cloudtel.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] ESL outbound /inbound

I use inbound to monitor switches as a socket client and mod_xml_curl to direct call flow from a seperate web app. This allows me to receive events across multiple channels across multiple switches without worrying about call flow, which is a good design for what I am doing in this example ... monitoring call flow and conferencing events for realtime display in a UI. Our call flow systems are critical and monitoring is non-critical so it works well to seperate them. As a calling card app your system will probably require tighter integration of call flow and billing, so combining them could make sense.


The main benefit of outbound is you can use a socket server and each of your clients represents a distinct channel on one of your switches. It can make directing call flow easy and it is simple to require your socket server or play an "oops we are busy" message when it is down ... and it helped everyone quickly port their FAGI apps to this project :)


You are correct that you can just use event socket and sendmsg to control call flow and not use lua at all. Yes there is overlap of functionality between inbound/outbound because both allow you to do basically the same thing, but as different roles in the socket connection.



 		 	   		  

From chris at cloudtel.com  Sat Dec 25 01:53:35 2010
From: chris at cloudtel.com (Chris Burns)
Date: Fri, 24 Dec 2010 17:53:35 -0500
Subject: [Freeswitch-users] ESL outbound /inbound
In-Reply-To: <COL122-W60C04326D9849EE2C4A06EB51D0@phx.gbl>
References: <COL122-W2256918ACD58AE62DC48CDB51C0@phx.gbl>
	<AANLkTi=etPHjn63mMxNFYZJRH+Pch+o_3RMuU2xrViK_@mail.gmail.com>
	<COL122-W13FC929C897D9C705EFDAEB51C0@phx.gbl>
	<AANLkTi=CUMxo357GyBk2Q08kzA5FWtFRMnyrhi0aprQs@mail.gmail.com>
	<COL122-W4463935CBB0A4B25F9BE75B51D0@phx.gbl>
	<AANLkTikZiZ7VBasTMOkvU5Fsxn_z+o7H6_kqNx0UGA35@mail.gmail.com>
	<COL122-W60C04326D9849EE2C4A06EB51D0@phx.gbl>
Message-ID: <AANLkTi=DGdZ7c7t-=C+0bjpT+KrO_ZuE-cn4ox9_6b8V@mail.gmail.com>

I think inbound or outbound event socket both are fine choices. Either way
you can keep your logic all in java and not bother with lua if you dont want
it.

On Fri, Dec 24, 2010 at 4:09 PM, Rafqat . <rafonline at hotmail.com> wrote:

>
>
> Hi Chris,
>
> Thanks for your advice.
>
> In terms of tighter integration and control, do you think using inbound
> event socket for billing and
> sendmsg to control call flow would be a good approach for a calling card
> application.  I have no problem using lua but feel that once i get passed
> the trivial stuff of checking account funds etc. I may have logic scattered
> around in more than one place.
>
> Sorry for being a pita but your advice (and the freeswitch community's
> advice)  is invaluable to a newbie like me.
>
> Thanks
>
> Raf
>
>
> Date: Fri, 24 Dec 2010 15:20:00 -0500
> From: chris at cloudtel.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] ESL outbound /inbound
>
> I use inbound to monitor switches as a socket client and mod_xml_curl to
> direct call flow from a seperate web app. This allows me to receive events
> across multiple channels across multiple switches without worrying about
> call flow, which is a good design for what I am doing in this example ...
> monitoring call flow and conferencing events for realtime display in a UI.
> Our call flow systems are critical and monitoring is non-critical so it
> works well to seperate them. As a calling card app your system will probably
> require tighter integration of call flow and billing, so combining them
> could make sense.
>
>
> The main benefit of outbound is you can use a socket server and each of
> your clients represents a distinct channel on one of your switches. It can
> make directing call flow easy and it is simple to require your socket server
> or play an "oops we are busy" message when it is down ... and it helped
> everyone quickly port their FAGI apps to this project :)
>
>
> You are correct that you can just use event socket and sendmsg to control
> call flow and not use lua at all. Yes there is overlap of functionality
> between inbound/outbound because both allow you to do basically the same
> thing, but as different roles in the socket connection.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From riedinger at sns.eu  Sat Dec 25 02:02:03 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Sat, 25 Dec 2010 00:02:03 +0100
Subject: [Freeswitch-users] Asynchronous PTIME
In-Reply-To: <4D127733.8010409@nevian.org>
References: <4D127733.8010409@nevian.org>
Message-ID: <4D15266B.3080903@sns.eu>

Hi Serge,

I had similar problems, look for my thread "Problematic Behaviour of FS 
regarding ptime negotiation" in October. I could solve the problem by 
setting "rtp-autofix-timing=false", which disabled the (too) smart 
behaviour of FreeSwitch.

BR
     Jan


Am 22.12.2010 23:09, schrieb Serge S. Yuriev:
> Hello,
>
> 2010-12-20 16:51:13.960502 [WARNING] mod_sofia.c:1036 Asynchronous PTIME
> not supported, changing our end from 20 to 60
>
> I'm getting this warning and client hears chopped sound :(
> That is "Our end"?
>
> Call flow is Cisco (sip) FS (sip) YATE (h323) MeraTransitSoftSwitch
> All but MVTS under my control.
>
> I doesn't see any clue in logs and can't reproduce this with my testing
> via Blink/PortSIP/PhonerLite/Twinkle or Grandstream devices
>
> Which debug/logs I should take? Any ideas?
>
> Thanks a lot.
>
> btw how I can save debug into log not only console?


From riedinger at sns.eu  Sat Dec 25 02:07:20 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Sat, 25 Dec 2010 00:07:20 +0100
Subject: [Freeswitch-users] calling card  app design decision
In-Reply-To: <COL122-W40EEE9A9EEB0A4C5B52283B51B0@phx.gbl>
References: <COL122-W40EEE9A9EEB0A4C5B52283B51B0@phx.gbl>
Message-ID: <4D1527A8.7010607@sns.eu>

Did you check, if ASTPP can satisfy your needs? There is support for 
calling cards with  FreeSwitch in beta state.

BR
    Jan

BR
    Jan

Am 22.12.2010 16:29, schrieb Rafqat .:
> Hi,
>
> I am wanting to develop a calling card service using FreeSWITCH.  It's 
> seems FreeSWITCH is very flexible in terms of how one could go about 
> implementing this service.    I am intending to use mod_nibblebill and 
> mod_lcr to do the core stuff.  I would like to use Java for any 
> programming (as it's the language I am most familiar with) however, I 
> am unsure whether to take an event socket based approach to the 
> application or to use something like mod_java.
>
> Any advice/suggestions will be most welcome.
>
> Cheers
>
> Raf
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From brian at freeswitch.org  Sat Dec 25 02:16:41 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 24 Dec 2010 17:16:41 -0600
Subject: [Freeswitch-users] Asynchronous PTIME
In-Reply-To: <4D15266B.3080903@sns.eu>
References: <4D127733.8010409@nevian.org> <4D15266B.3080903@sns.eu>
Message-ID: <F37A07D0-0A6D-4A79-A2EE-04EFA0DC95F5@freeswitch.org>

sigh... You could also try setting your codec to XXX at 60i

But in the end if the device is broken make them fix it... if we don't start making people fix broken stuff we'll never see the end of this crap.

/b

On Dec 24, 2010, at 5:02 PM, Jan Riedinger wrote:

> Hi Serge,
> 
> I had similar problems, look for my thread "Problematic Behaviour of FS 
> regarding ptime negotiation" in October. I could solve the problem by 
> setting "rtp-autofix-timing=false", which disabled the (too) smart 
> behaviour of FreeSwitch.
> 
> BR
>     Jan



From jonyoung111 at gmail.com  Sat Dec 25 02:20:07 2010
From: jonyoung111 at gmail.com (Jon Young)
Date: Fri, 24 Dec 2010 16:20:07 -0700
Subject: [Freeswitch-users] Can FreeSWITCH be setup to look like a SIP
	carrier?
Message-ID: <AANLkTinTi0qGHFkougazm4p_uCEPAusQwSU8REf5sZ4m@mail.gmail.com>

I am trying to determine if using FS would work for what I need.

I have a training class where I have 10-15 students with SIP PBXs.  I
would like each student to have access to a SIP trunk. What I would
like to do is have each student Register with FS using a 10 digit
number.  I have a SIP Trunk group with 6 call paths that I would like
to have FS Register to the SIP provider.

This will allow them to make outbound calls and I may have inbound
calls hit the IVR to select the student.

Thanks for any feedback.


From brian at freeswitch.org  Sat Dec 25 02:39:02 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 24 Dec 2010 17:39:02 -0600
Subject: [Freeswitch-users] Can FreeSWITCH be setup to look like a SIP
	carrier?
In-Reply-To: <AANLkTinTi0qGHFkougazm4p_uCEPAusQwSU8REf5sZ4m@mail.gmail.com>
References: <AANLkTinTi0qGHFkougazm4p_uCEPAusQwSU8REf5sZ4m@mail.gmail.com>
Message-ID: <A14E8B84-2447-40AD-A8D3-B1C15BB6BC05@freeswitch.org>

People do exactly this every day... our default config works as a PBX.  Have you read the FreeSWITCH book yet?

/b

On Dec 24, 2010, at 5:20 PM, Jon Young wrote:

> I am trying to determine if using FS would work for what I need.
> 
> I have a training class where I have 10-15 students with SIP PBXs.  I
> would like each student to have access to a SIP trunk. What I would
> like to do is have each student Register with FS using a 10 digit
> number.  I have a SIP Trunk group with 6 call paths that I would like
> to have FS Register to the SIP provider.
> 
> This will allow them to make outbound calls and I may have inbound
> calls hit the IVR to select the student.
> 
> Thanks for any feedback.



From jonyoung111 at gmail.com  Sat Dec 25 03:10:35 2010
From: jonyoung111 at gmail.com (Jon Young)
Date: Fri, 24 Dec 2010 17:10:35 -0700
Subject: [Freeswitch-users] Can FreeSWITCH be setup to look like a SIP
	carrier?
In-Reply-To: <A14E8B84-2447-40AD-A8D3-B1C15BB6BC05@freeswitch.org>
References: <AANLkTinTi0qGHFkougazm4p_uCEPAusQwSU8REf5sZ4m@mail.gmail.com>
	<A14E8B84-2447-40AD-A8D3-B1C15BB6BC05@freeswitch.org>
Message-ID: <AANLkTin8ohyLyX_=QSAwKAgoq5fpvjjCMZhaxXGuMQC7@mail.gmail.com>

I haven't read the book yet.  I am familiar with using FS as a PBX but
maybe I am over-complicating what I am trying to do.  I'll start
putting it together and see if I get stuck.  Thanks.

On Fri, Dec 24, 2010 at 4:39 PM, Brian West <brian at freeswitch.org> wrote:
> People do exactly this every day... our default config works as a PBX. ?Have you read the FreeSWITCH book yet?
>
> /b
>
> On Dec 24, 2010, at 5:20 PM, Jon Young wrote:
>
>> I am trying to determine if using FS would work for what I need.
>>
>> I have a training class where I have 10-15 students with SIP PBXs. ?I
>> would like each student to have access to a SIP trunk. What I would
>> like to do is have each student Register with FS using a 10 digit
>> number. ?I have a SIP Trunk group with 6 call paths that I would like
>> to have FS Register to the SIP provider.
>>
>> This will allow them to make outbound calls and I may have inbound
>> calls hit the IVR to select the student.
>>
>> Thanks for any feedback.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From edpimentl at gmail.com  Sat Dec 25 05:24:54 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Fri, 24 Dec 2010 21:24:54 -0500
Subject: [Freeswitch-users] calling card app design decision
In-Reply-To: <4D1527A8.7010607@sns.eu>
References: <COL122-W40EEE9A9EEB0A4C5B52283B51B0@phx.gbl>
	<4D1527A8.7010607@sns.eu>
Message-ID: <AANLkTim4nPi+tiPBurNknt2au7XTbj0j8XJ8uCuKy-xp@mail.gmail.com>

Can someone "please" post the link to the latest version of astpp for
freeswitch?

Thanks in advance,
-E
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From a.afzali2003 at gmail.com  Sat Dec 25 08:54:28 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sat, 25 Dec 2010 09:24:28 +0330
Subject: [Freeswitch-users] Loosing DTMF Digits in XML IVR
In-Reply-To: <AANLkTinNuSc4MFfz9rCP1Qj6WYmmoWW3eAqmNL5_YQvr@mail.gmail.com>
References: <AANLkTim98bvSVJQWdKFcvTsZk8uJd3OsPZvK-omLTh4=@mail.gmail.com>
	<AANLkTimt8wrSVDBy480R9XwP2WpV3GKdeQ7VCLQQpLeN@mail.gmail.com>
	<AANLkTinNuSc4MFfz9rCP1Qj6WYmmoWW3eAqmNL5_YQvr@mail.gmail.com>
Message-ID: <AANLkTikCiH9ZNDSr_MttAx_3c7XSWxtEEOeSuaWmmpqY@mail.gmail.com>

merry xmas,

After update to latest freeswitch (it seems related to switch_rtp.c) the
issue resolved.

Happy New Year to all you good guys.

BEST,

--afshin



On Thu, Dec 23, 2010 at 11:55 PM, afshin afzali <a.afzali2003 at gmail.com>wrote:

> Hi,
>
> I use eyeBeam to test my application. As I checked, it is enabled in both
> by default. my IVR is in XML format, also I experience this behavior by the
> demo IVR ( 5000 ) as well. I tried to test digit detection just by one of
> inbound / 2833 format. The result was that one of those did not work at all.
> I'll provide you debug log soon.
>
> Thanks MC,
> -- afshin
>
>
> On Thu, Dec 23, 2010 at 10:55 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Are you using inband or 2833 for DTMFs? How are you creating your IVR -
>> with the XML IVR or with something else? Get a console debug and put it on
>> pastebin.freeswitch.org so we can see what's going on.
>>
>> -MC
>>
>> On Thu, Dec 23, 2010 at 3:19 AM, afshin afzali <a.afzali2003 at gmail.com>wrote:
>>
>>> Hi Guys,
>>>
>>> After updating to latest freeswitch (git-34a0ca5 2010-12-22 20-38-57
>>> -0600) , I've noticed that I'm loosing some of the dtmf digits in my IVR.
>>> Actually freeswitch looses most of them! My freeswitch hosted by ubuntu
>>> 10.04 64bit Server.
>>>
>>> Appreciate all comments,
>>>
>>> Regards,
>>> -- afshin
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From rupa at rupa.com  Sat Dec 25 16:38:05 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 25 Dec 2010 07:38:05 -0600
Subject: [Freeswitch-users] nibblebill schedule
In-Reply-To: <F07B572C409042239691B1F3DE08575E@e1705>
References: <F07B572C409042239691B1F3DE08575E@e1705>
Message-ID: <AANLkTi=W9BV+JnD4K=Oqv_fGSt-SpVM2jYV3AyL7xzO2@mail.gmail.com>

I don't see how without change.

You have no idea how fast the $$ is being nibbled away by other calls.  So,
while you can account for a 30s buffer in the current call using the current
rate, the more calls that are up for that account the more "off" the 30s
estimate will be.

On Fri, Dec 24, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:

>  Is it possible to schedule the transfer to nibblebill nobal_action ?
> I'd like to schedule of 30s before the call is cut and go to nobal_action
> extension
>
> thanks
>
>
-- 
-Rupa
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From infos at madovsky.org  Sat Dec 25 19:22:18 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 25 Dec 2010 11:22:18 -0500
Subject: [Freeswitch-users] nibblebill schedule
References: <F07B572C409042239691B1F3DE08575E@e1705>
	<AANLkTi=W9BV+JnD4K=Oqv_fGSt-SpVM2jYV3AyL7xzO2@mail.gmail.com>
Message-ID: <DA3EFAD250384C26947B906385BE23D1@e1705>

ha ok,
the concept I thought is to let user to call 30s as a trial and
start niblebill after this time.... does it need to modify the mod_nibblebill source code ?

Thanks

F
  ----- Original Message ----- 
  From: Rupa Schomaker 
  To: FreeSWITCH Users Help 
  Sent: Saturday, December 25, 2010 8:38 AM
  Subject: Re: [Freeswitch-users] nibblebill schedule


  I don't see how without change.


  You have no idea how fast the $$ is being nibbled away by other calls.  So, while you can account for a 30s buffer in the current call using the current rate, the more calls that are up for that account the more "off" the 30s estimate will be.  


  On Fri, Dec 24, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:

    Is it possible to schedule the transfer to nibblebill nobal_action ?
    I'd like to schedule of 30s before the call is cut and go to nobal_action extension

    thanks



  -- 
  -Rupa



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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From Avi at aMarcus.com  Sat Dec 25 19:30:55 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Sat, 25 Dec 2010 18:30:55 +0200
Subject: [Freeswitch-users] nibblebill schedule
In-Reply-To: <DA3EFAD250384C26947B906385BE23D1@e1705>
References: <F07B572C409042239691B1F3DE08575E@e1705>
	<AANLkTi=W9BV+JnD4K=Oqv_fGSt-SpVM2jYV3AyL7xzO2@mail.gmail.com>
	<DA3EFAD250384C26947B906385BE23D1@e1705>
Message-ID: <AANLkTimRgSA7EJLU1HO2pydgfYGta-D+KQf_Tf079hdk@mail.gmail.com>

Call 30 seconds as a trial?
Perhaps skip nibblebill and use:
sched_hangup<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_hangup>
And count it somewhere that they used their trial period (using
Mod_db<http://wiki.freeswitch.org/wiki/Mod_db> with
odbc  or mod_odbc_query from git contrib)

-Avi

On Sat, Dec 25, 2010 at 6:22 PM, Madovsky <infos at madovsky.org> wrote:
> ha ok,
> the concept I thought is to let user to call 30s as a trial and
> start niblebill after this time.... does it need to modify the
> mod_nibblebill source code ?
>
> Thanks
> F
>
> ----- Original Message -----
> From: Rupa Schomaker
> To: FreeSWITCH Users Help
> Sent: Saturday, December 25, 2010 8:38 AM
> Subject: Re: [Freeswitch-users] nibblebill schedule
> I don't see how without change.
> You have no idea how fast the $$ is being nibbled away by other calls.
 So,
> while you can account for a 30s buffer in the current call using the
current
> rate, the more calls that are up for that account the more "off" the 30s
> estimate will be.
>
> On Fri, Dec 24, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:
>>
>> Is it possible to schedule the transfer to nibblebill nobal_action ?
>> I'd like to schedule of 30s before the call is cut and go to nobal_action
>> extension
>>
>> thanks
>
> --
> -Rupa
>
> ________________________________
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From riedinger at sns.eu  Sat Dec 25 19:32:40 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Sat, 25 Dec 2010 17:32:40 +0100
Subject: [Freeswitch-users] calling card app design decision
In-Reply-To: <AANLkTim4nPi+tiPBurNknt2au7XTbj0j8XJ8uCuKy-xp@mail.gmail.com>
References: <COL122-W40EEE9A9EEB0A4C5B52283B51B0@phx.gbl>	<4D1527A8.7010607@sns.eu>
	<AANLkTim4nPi+tiPBurNknt2au7XTbj0j8XJ8uCuKy-xp@mail.gmail.com>
Message-ID: <4D161CA8.6040703@sns.eu>

Hi Ed,

just look at www.astpp.org. I wrote to the author about two months ago, 
see below.

Unfortunately, I hadn't had time for further examination. I think, I 
will start the project within the next months. If you want we can try it 
both and we can exchange expericences.

BR
     Jan

    /
    Am 13.10.2010 06:13, schrieb Darren Wiebe:
      On 12/10/2010 1:48 AM, Jan Riedinger wrote:
    /
>     / Hi Darren,
>
>     I'm examining for a customer the usage of ASTPP for a Freeswitch
>     based calling card Platform. What's the status of this project?
>     The last entries of on the ASTPP web site for Freeswitch Calling
>     Card support are pretty old (about 1 year) and I still read
>     "testing" for this topic on the ASTPP features web site. Is there
>     ongoing work? Can you recommend the usage of ASTPP for this
>     purpose at this moment. / /
>     Thank you in advance
>                       Jan
>
>     / 
    /
    Hello,
    Yes, there is still ongoing work on callingcard support.  As to
    recommending it for calling card use I'm not sure how to respond. 
    It's just starting to get deployed for that.  However, the last
    person that I set it up for has been testing that for several weeks
    and has not reported any problems except some missing sound files. 
    He's supposed to be getting me copies of the correct sound files.

    Darren Wiebe
    /


Am 25.12.2010 03:24, schrieb EdPimentl:
> Can someone "please" post the link to the latest version of astpp for 
> freeswitch?
>
> Thanks in advance,
> -E
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From infos at madovsky.org  Sat Dec 25 19:42:51 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 25 Dec 2010 11:42:51 -0500
Subject: [Freeswitch-users] nibblebill schedule
References: <F07B572C409042239691B1F3DE08575E@e1705><AANLkTi=W9BV+JnD4K=Oqv_fGSt-SpVM2jYV3AyL7xzO2@mail.gmail.com><DA3EFAD250384C26947B906385BE23D1@e1705>
	<AANLkTimRgSA7EJLU1HO2pydgfYGta-D+KQf_Tf079hdk@mail.gmail.com>
Message-ID: <3D74E1421A48499EAFC8A653D5EF4EF0@e1705>

yes I thought to this but it would be easier to do it through nibblebill
  ----- Original Message ----- 
  From: Avi Marcus 
  To: FreeSWITCH Users Help 
  Sent: Saturday, December 25, 2010 11:30 AM
  Subject: Re: [Freeswitch-users] nibblebill schedule


  Call 30 seconds as a trial?
  Perhaps skip nibblebill and use: sched_hangup
  And count it somewhere that they used their trial period (using Mod_db with odbc  or mod_odbc_query from git contrib)


  -Avi

  On Sat, Dec 25, 2010 at 6:22 PM, Madovsky <infos at madovsky.org> wrote:
  > ha ok,
  > the concept I thought is to let user to call 30s as a trial and
  > start niblebill after this time.... does it need to modify the
  > mod_nibblebill source code ?
  >  
  > Thanks
  > F
  >
  > ----- Original Message -----
  > From: Rupa Schomaker
  > To: FreeSWITCH Users Help
  > Sent: Saturday, December 25, 2010 8:38 AM
  > Subject: Re: [Freeswitch-users] nibblebill schedule
  > I don't see how without change.
  > You have no idea how fast the $$ is being nibbled away by other calls.  So,
  > while you can account for a 30s buffer in the current call using the current
  > rate, the more calls that are up for that account the more "off" the 30s
  > estimate will be.  
  >
  > On Fri, Dec 24, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:
  >>
  >> Is it possible to schedule the transfer to nibblebill nobal_action ?
  >> I'd like to schedule of 30s before the call is cut and go to nobal_action
  >> extension
  >>  
  >> thanks
  >
  > --
  > -Rupa
  >
  > ________________________________
  >
  > _______________________________________________
  > FreeSWITCH-users mailing list
  > FreeSWITCH-users at lists.freeswitch.org
  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  > http://www.freeswitch.org
  >
  > _______________________________________________
  > FreeSWITCH-users mailing list
  > FreeSWITCH-users at lists.freeswitch.org
  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  > http://www.freeswitch.org
  >
  >




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


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From Avi at aMarcus.com  Sat Dec 25 19:47:30 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Sat, 25 Dec 2010 18:47:30 +0200
Subject: [Freeswitch-users] nibblebill schedule
In-Reply-To: <3D74E1421A48499EAFC8A653D5EF4EF0@e1705>
References: <F07B572C409042239691B1F3DE08575E@e1705>
	<AANLkTi=W9BV+JnD4K=Oqv_fGSt-SpVM2jYV3AyL7xzO2@mail.gmail.com>
	<DA3EFAD250384C26947B906385BE23D1@e1705>
	<AANLkTimRgSA7EJLU1HO2pydgfYGta-D+KQf_Tf079hdk@mail.gmail.com>
	<3D74E1421A48499EAFC8A653D5EF4EF0@e1705>
Message-ID: <AANLkTim=Xx=sM2RHJcSfXcoJCsCPu_Vq1fusUfgD736V@mail.gmail.com>

Hmm. Alternatively..
You could set a user-variable that shows they are on the trial.
Then, before you set the price, you check for that variable and if it's
true, you set, say, nibble_rate=.02 and make sure they start with a balance
of .01
You probably need to change the heartbeat to 30 seconds to do that.
Good luck :)

On Sat, Dec 25, 2010 at 6:42 PM, Madovsky <infos at madovsky.org> wrote:

>  yes I thought to this but it would be easier to do it through nibblebill
>
> ----- Original Message -----
> *From:* Avi Marcus <Avi at aMarcus.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Saturday, December 25, 2010 11:30 AM
> *Subject:* Re: [Freeswitch-users] nibblebill schedule
>
> Call 30 seconds as a trial?
> Perhaps skip nibblebill and use: sched_hangup<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_hangup>
> And count it somewhere that they used their trial period (using Mod_db<http://wiki.freeswitch.org/wiki/Mod_db> with
> odbc  or mod_odbc_query from git contrib)
>
> -Avi
>
> On Sat, Dec 25, 2010 at 6:22 PM, Madovsky <infos at madovsky.org> wrote:
> > ha ok,
> > the concept I thought is to let user to call 30s as a trial and
> > start niblebill after this time.... does it need to modify the
> > mod_nibblebill source code ?
> >
> > Thanks
> > F
> >
> > ----- Original Message -----
> > From: Rupa Schomaker
> > To: FreeSWITCH Users Help
> > Sent: Saturday, December 25, 2010 8:38 AM
> > Subject: Re: [Freeswitch-users] nibblebill schedule
> > I don't see how without change.
> > You have no idea how fast the $$ is being nibbled away by other calls.
>  So,
> > while you can account for a 30s buffer in the current call using the
> current
> > rate, the more calls that are up for that account the more "off" the 30s
> > estimate will be.
> >
> > On Fri, Dec 24, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:
> >>
> >> Is it possible to schedule the transfer to nibblebill nobal_action ?
> >> I'd like to schedule of 30s before the call is cut and go to
> nobal_action
> >> extension
> >>
> >> thanks
> >
> > --
> > -Rupa
> >
> > ________________________________
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveu at coppice.org  Sat Dec 25 20:16:09 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 26 Dec 2010 01:16:09 +0800
Subject: [Freeswitch-users] Asynchronous PTIME
In-Reply-To: <F37A07D0-0A6D-4A79-A2EE-04EFA0DC95F5@freeswitch.org>
References: <4D127733.8010409@nevian.org> <4D15266B.3080903@sns.eu>
	<F37A07D0-0A6D-4A79-A2EE-04EFA0DC95F5@freeswitch.org>
Message-ID: <4D1626D9.4060601@coppice.org>

If the VoIP industry develops like the PSTN industry, you will *never* 
see the end of this crap. The only FAX modems I've seen which fully 
comply with the ITU specs are my own ones, in spandsp. All the others 
did some wrong things in the early days of the specs. The competitors 
learned to tolerate each other's errors. After that nobody ever bothered 
to fix those errors. The same can be said almost every other group of 
ITU specs.

Steve


On 12/25/2010 07:16 AM, Brian West wrote:
> sigh... You could also try setting your codec to XXX at 60i
>
> But in the end if the device is broken make them fix it... if we don't start making people fix broken stuff we'll never see the end of this crap.
>
> /b
>
> On Dec 24, 2010, at 5:02 PM, Jan Riedinger wrote:
>
>> Hi Serge,
>>
>> I had similar problems, look for my thread "Problematic Behaviour of FS
>> regarding ptime negotiation" in October. I could solve the problem by
>> setting "rtp-autofix-timing=false", which disabled the (too) smart
>> behaviour of FreeSwitch.
>>
>> BR
>>      Jan



From infos at madovsky.org  Sat Dec 25 20:16:30 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 25 Dec 2010 12:16:30 -0500
Subject: [Freeswitch-users] nibblebill schedule
References: <F07B572C409042239691B1F3DE08575E@e1705><AANLkTi=W9BV+JnD4K=Oqv_fGSt-SpVM2jYV3AyL7xzO2@mail.gmail.com><DA3EFAD250384C26947B906385BE23D1@e1705><AANLkTimRgSA7EJLU1HO2pydgfYGta-D+KQf_Tf079hdk@mail.gmail.com><3D74E1421A48499EAFC8A653D5EF4EF0@e1705>
	<AANLkTim=Xx=sM2RHJcSfXcoJCsCPu_Vq1fusUfgD736V@mail.gmail.com>
Message-ID: <95B12D2DC36442F388CA538F29898C16@e1705>

yes already thought also ;)
and already set nibblebill hearbeat to 30s.
but in fact the user is never in "trial" mode.
it's in case that if he has less than nobal so he will
always call for 30sec only.
Or maybe make a real trial when he registers and once he bought his
first prepaid I will remove it.
  ----- Original Message ----- 
  From: Avi Marcus 
  To: FreeSWITCH Users Help 
  Sent: Saturday, December 25, 2010 11:47 AM
  Subject: Re: [Freeswitch-users] nibblebill schedule


  Hmm. Alternatively..
  You could set a user-variable that shows they are on the trial.
  Then, before you set the price, you check for that variable and if it's true, you set, say, nibble_rate=.02 and make sure they start with a balance of .01
  You probably need to change the heartbeat to 30 seconds to do that.
  Good luck :)


  On Sat, Dec 25, 2010 at 6:42 PM, Madovsky <infos at madovsky.org> wrote:

    yes I thought to this but it would be easier to do it through nibblebill
      ----- Original Message ----- 
      From: Avi Marcus 
      To: FreeSWITCH Users Help 
      Sent: Saturday, December 25, 2010 11:30 AM
      Subject: Re: [Freeswitch-users] nibblebill schedule


      Call 30 seconds as a trial?
      Perhaps skip nibblebill and use: sched_hangup
      And count it somewhere that they used their trial period (using Mod_db with odbc  or mod_odbc_query from git contrib) 


      -Avi

      On Sat, Dec 25, 2010 at 6:22 PM, Madovsky <infos at madovsky.org> wrote:
      > ha ok,
      > the concept I thought is to let user to call 30s as a trial and
      > start niblebill after this time.... does it need to modify the
      > mod_nibblebill source code ?
      >  
      > Thanks
      > F
      >
      > ----- Original Message -----
      > From: Rupa Schomaker
      > To: FreeSWITCH Users Help
      > Sent: Saturday, December 25, 2010 8:38 AM
      > Subject: Re: [Freeswitch-users] nibblebill schedule
      > I don't see how without change.
      > You have no idea how fast the $$ is being nibbled away by other calls.  So,
      > while you can account for a 30s buffer in the current call using the current
      > rate, the more calls that are up for that account the more "off" the 30s
      > estimate will be.  
      >
      > On Fri, Dec 24, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:
      >>
      >> Is it possible to schedule the transfer to nibblebill nobal_action ?
      >> I'd like to schedule of 30s before the call is cut and go to nobal_action
      >> extension
      >>  
      >> thanks
      >
      > --
      > -Rupa
      >
      > ________________________________
      >
      > _______________________________________________
      > FreeSWITCH-users mailing list
      > FreeSWITCH-users at lists.freeswitch.org
      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      > http://www.freeswitch.org
      >
      > _______________________________________________
      > FreeSWITCH-users mailing list
      > FreeSWITCH-users at lists.freeswitch.org
      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      > http://www.freeswitch.org
      >
      >




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


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



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






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


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From a.afzali2003 at gmail.com  Sat Dec 25 21:54:03 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sat, 25 Dec 2010 22:24:03 +0330
Subject: [Freeswitch-users] Converting WAV to MP3 by LAME
Message-ID: <AANLkTimmTjP0Zaj9u_neY9=+QT1XVR9v8RgPzURqXgc1@mail.gmail.com>

Hi Guys,

I want to use LAME to convert session recorded call logs to mp3 format in a
Lua script.
Is there a better way for this?

BEST,

-- afshin
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From curriegrad2004 at gmail.com  Sun Dec 26 03:47:10 2010
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 25 Dec 2010 16:47:10 -0800
Subject: [Freeswitch-users] Converting WAV to MP3 by LAME
In-Reply-To: <AANLkTimmTjP0Zaj9u_neY9=+QT1XVR9v8RgPzURqXgc1@mail.gmail.com>
References: <AANLkTimmTjP0Zaj9u_neY9=+QT1XVR9v8RgPzURqXgc1@mail.gmail.com>
Message-ID: <AANLkTinhKD6fR9ke0MbejetEXu00TQSV4dPN_J5R-=au@mail.gmail.com>

You can do this with a shell script easily through the dialplan.
api_hangup_hook and system should do what you want it to do.

On Sat, Dec 25, 2010 at 10:54 AM, afshin afzali <a.afzali2003 at gmail.com> wrote:
> Hi Guys,
>
> I want to use LAME to convert session recorded call logs to mp3 format in a
> Lua script.
> Is there a better way for this?
>
> BEST,
>
> -- afshin
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Sun Dec 26 04:13:22 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 25 Dec 2010 20:13:22 -0500
Subject: [Freeswitch-users] Converting WAV to MP3 by LAME
References: <AANLkTimmTjP0Zaj9u_neY9=+QT1XVR9v8RgPzURqXgc1@mail.gmail.com>
	<AANLkTinhKD6fR9ke0MbejetEXu00TQSV4dPN_J5R-=au@mail.gmail.com>
Message-ID: <F6F081015F18495AA7CFA8873D8F72DF@e1705>

or use Perl application

----- Original Message ----- 
From: "curriegrad2004" <curriegrad2004 at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, December 25, 2010 7:47 PM
Subject: Re: [Freeswitch-users] Converting WAV to MP3 by LAME


> You can do this with a shell script easily through the dialplan.
> api_hangup_hook and system should do what you want it to do.
>
> On Sat, Dec 25, 2010 at 10:54 AM, afshin afzali <a.afzali2003 at gmail.com> 
> wrote:
>> Hi Guys,
>>
>> I want to use LAME to convert session recorded call logs to mp3 format in 
>> a
>> Lua script.
>> Is there a better way for this?
>>
>> BEST,
>>
>> -- afshin
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From Nabble at slickdeals.endjunk.com  Sun Dec 26 04:23:40 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 25 Dec 2010 17:23:40 -0800 (PST)
Subject: [Freeswitch-users] libg722_1: Cross Compilation of
	src/make_dct4_tables.c
Message-ID: <1293326620424-5867008.post@n2.nabble.com>


I am facing a dilemma trying to cross-compile libg722_1 package from
FreeSWITCH git version for an ARM5 platform. First of all, the compilation
is fine up to a point to produce a cross-platform src/make_dct4_tables
binary file. From there on, the compilation process tries to execute the
newly produced (cross-compiled) src/make_dct4_tables binary file on the host
to produce a src/dct4.h file that contains tables. This certainly will crash
the compilation process because the src/make_dct4_tables binary file is for
an ARM5 platform and not an x86 platform where the host is. I can manually
compile the src/make_dct4_tables.c on the host to produce an x86 binary file
and use it to manually generate the src/dct4.h file, but this poses a
dilemma. Even though the src/dct4.h produced in this way will not have a
problem to produce the libg722_1 codes to run/execute on an ARM5 platform,
but it will probably not be able to correctly process the G722_1 audio
streams. The reason is simply I suspect the src/dct4.h files generated by
the src/make_dct4_tables binary file from both the x86 and ARM5 platforms
are not identical. I hope experts here will be able to see this and come up
with the solution.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/libg722-1-Cross-Compilation-of-src-make-dct4-tables-c-tp5867008p5867008.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From u2nsam at gmail.com  Sun Dec 26 07:03:28 2010
From: u2nsam at gmail.com (Sam)
Date: Sun, 26 Dec 2010 09:33:28 +0530
Subject: [Freeswitch-users] DTMF missing
In-Reply-To: <AANLkTik=NLa+injL5hF=VsymwoE9wd1Wb2joXJDhvfFZ@mail.gmail.com>
References: <AANLkTik=NLa+injL5hF=VsymwoE9wd1Wb2joXJDhvfFZ@mail.gmail.com>
Message-ID: <AANLkTincAApO4JDwzt-5oX0iD2VozFNVCtembOumL4iS@mail.gmail.com>

Hello

Just like Afshin, I am also missing DTMF digits ,using FreeSWITCH Version
1.0.head (git-34a0ca5 2010-12-22 20-38-57 -0600) ;
should i upgrade to latest ?

Regards
Sam



On Fri, Dec 24, 2010 at 4:10 PM, Sam <u2nsam at gmail.com> wrote:

> Hi,,
>
>
> I have installed the latest ver of freeswitch and i have configured the
> conference.
>
> now when i punch in the digits for password , i could see that the DTMF
> digits are missed on fs_cli.
>
> it only happens when i dial it from polycom or cisco phones.
>
> I have tried with and without these values below:-
>
> <param name="dtmf-type" value="rfc2833"/>
>     <param name="pass-rfc2833" value="true"/>
>
>
>
> traces fetched:
> 192.168.2.49:5060 -> 192.168.2.190:5060
>   INVITE sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190> SIP/2.0..Via:
> SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From: "7028" <
> sip:7028 at 192.168.2.190 <sip%3A7028 at 192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To:
> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>..Call-ID:
> 0017592a-eb33001a-
>   63da3294-1a7bfdfa at 192.168.2.49..Max-Forwards: 70..Date: Fri, 24 Dec 2010
> 10:13:52 GMT..CSeq: 102 INVITE..User-Agent: Cisco-CP7940G/8.0..Contact:
> <sip:7028 at 192.168.2.49:5060;transport=udp>..Proxy-Authorization: Digest
> username="7028"
>   ,realm="192.168.2.190",uri="sip:7050 at 192.168.2.190<sip%3A7050 at 192.168.2.190>",response="a668f5c480285b35e7ff6bcd446879f0",nonce="d2c540f2-8487-4d87-bdab-871585253eb8",cnonce="0a6c4176",qop=auth,nc=00000001,algorithm=MD5..Expires:
> 180..Accept: application/sdp
>   ..Allow:
> ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported:
> replaces,join,norefersub..Content-Length: 220..Content-Type:
> application/sdp..Content-Disposition:
> session;handling=optional....v=0..o=Cisco-SIPUA 16102
>    0 IN IP4 192.168.2.49..s=SIP Call..t=0 0..m=audio 17298 RTP/AVP 0 8
> 18..c=IN IP4 192.168.2.49..a=rtpmap:0 PCMU/8000..a=rtpmap:8
> PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=sendrecv..
>
>
> 192.168.2.190:5060 -> 192.168.2.49:5060
>   SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From:
> "7028" <sip:7028 at 192.168.2.190 <sip%3A7028 at 192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To:
> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;tag=2XXUZpgr1rvgc..Call-ID:
> 0017592a-eb33001a-63da3
>   294-1a7bfdfa at 192.168.2.49..CSeq: 102 INVITE..Contact:
> <sip:7050 at 192.168.2.190:5060;transport=udp>..User-Agent: NOVANET..Accept:
> application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
> INFO, REGISTER, REFER, NOTIF
>   Y, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path,
> replaces..Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
> sla, include-session-description, presence.winfo, message-summary,
> refer..Session-Expires: 180
>   0;refresher=uas..Min-SE: 120..Content-Type:
> application/sdp..Content-Disposition: session..Content-Length:
> 249..Remote-Party-ID: "7050" <sip:7050 at 192.168.2.190<sip%3A7050 at 192.168.2.190>>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH
> 1293163579 129
>   3163580 IN IP4 192.168.2.190..s=FreeSWITCH..c=IN IP4 192.168.2.190..t=0
> 0..m=audio 22050 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101
> telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - -
> -..a=ptime:20..
>
> But the dtmf are not missed when punched on eyebeam softphone.
>
> And all the phones have RFC 2833.
>
> traces fetched for softphone:-
>
> 192.168.2.17:6182 -> 192.168.2.190:5060
>   INVITE sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190> SIP/2.0..To: <
> sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>..From: 7001<
> sip:7001 at 192.168.2.190 <sip%3A7001 at 192.168.2.190>>;tag=6c557c1e..Via:
> SIP/2.0/UDP 192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport..Call-ID:
> d32ffe546570a77e..CS
>   eq: 2 INVITE..Contact: <sip:7001 at 192.168.2.17:6182>..Max-Forwards:
> 70..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Authorization: Digest
> username="7001",rea
>   lm="192.168.2.190",nonce="d0b4db6e-bd76-447f-9076-2e6b7809cb54",uri="
> sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>",response="45fb9eb4f0e0e4fffd87a22769a007ba",cnonce="1c27f3687059b16d",nc=00000001,qop=auth,algorithm=MD5..User-Agent:
> eyeBeam release 3007n
>    stamp 17816..Content-Length: 233....v=0..o=- 27833664 27833670 IN IP4
> 192.168.2.17..s=eyeBeam..c=IN IP4 192.168.2.17..t=0 0..m=audio 6398 RTP/AVP
> 0 18 101..a=alt:1 1 : 2C830AD9 0000004F 192.168.2.17 6398..a=fmtp:101
> 0-15..a=rtpmap:
>   101 telephone-event/8000..a=sendrecv..
>
>
> 192.168.2.190:5060 -> 192.168.2.17:6182
>   SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport=6182..From:
> 7001 <sip:7001 at 192.168.2.190 <sip%3A7001 at 192.168.2.190>>;tag=6c557c1e..To:
> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;tag=H0ctQv7KNgU2j..Call-ID:
> d32ffe546570a77e..C
>   Seq: 2 INVITE..Contact: <sip:7050 at 192.168.2.190:5060;transport=udp>..User-Agent:
> NOVANET..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
> MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Support
>   ed: timer, precondition, path, replaces..Allow-Events: talk, hold,
> presence, dialog, line-seize, call-info, sla, include-session-description,
> presence.winfo, message-summary, refer..Session-Expires:
> 1800;refresher=uas..Min-SE: 120..
>   Content-Type: application/sdp..Content-Disposition:
> session..Content-Length: 249..Remote-Party-ID: "7050" <
> sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH
> 1293161640 1293161641 IN IP4 192.168.2.190..
>   s=FreeSWITCH..c=IN IP4 192.168.2.190..t=0 0..m=audio 24852 RTP/AVP 0
> 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
> 0-16..a=silenceSupp:off - - - -..a=ptime:20..
>
>
> Any thing you can think how it can happen?
>
>
> Regards
> Sam
>
>
>
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From u2nsam at gmail.com  Sun Dec 26 07:20:12 2010
From: u2nsam at gmail.com (Sam)
Date: Sun, 26 Dec 2010 09:50:12 +0530
Subject: [Freeswitch-users] no invite send
In-Reply-To: <AANLkTim0hhJBaJSg9ghh8cwrbSY8_rC+4oOP79vT-Hw4@mail.gmail.com>
References: <AANLkTimBh+CqTz1O5qNmd7=z+Z6WZGeY1ua8kdZAwzH5@mail.gmail.com>
	<AANLkTikpL3pfFieW3Yv20iFxDo+yNdYA9zoyd3tgT-iK@mail.gmail.com>
	<AANLkTim0hhJBaJSg9ghh8cwrbSY8_rC+4oOP79vT-Hw4@mail.gmail.com>
Message-ID: <AANLkTim2O1mQkfbWX6cqzFE9xJoQL7+oDs4v43g7+T+J@mail.gmail.com>

Further to this question:

I have installed FS without prefix and when installing with prefix
./configure --prefix=/usr/local/conference/ ;it gives me an error,

how is the best method to install FS with prefix and can we install more
than 1 FS instance on the same server as i can do them to listen on
different ports?

Regds
Sam



On Thu, Dec 23, 2010 at 3:58 PM, Sam <u2nsam at gmail.com> wrote:

> I check it in sip trace ... it do not sends invite, probably i might have
> miss configured something,
> on the another FS server it works fine, this happened on the new
> installation where i was using prefix,
>
> I was trying to reinstall FS on the new server with prefix it gave me below
> error.
>
> ./configure: line 38199: 30565 Segmentation fault      /bin/sh
> /usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat/configure
> '--prefix=/usr/local/conference/' 'CONFIGURE_CFLAGS=-g -O2'
> 'CONFIGURE_CXXFLAGS=-g -O2' 'CONFIGURE_LDFLAGS=' '--cache-file=/dev/null'
> '--srcdir=.' '--with-apr=../apr' '--disable-shared' '--with-pic'
> '--without-sqlite2' '--without-sqlite3' '--with-expat=builtin'
> --cache-file=/dev/null
> --srcdir=/usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat
> --prefix=/usr/local/conference/ --exec-prefix=${prefix}
> --libdir=${exec_prefix}/lib
> --includedir=${prefix}/include/apr-${APRUTIL_MAJOR_VERSION}
> --bindir=${exec_prefix}/bin
> configure failed for xml/expat
> configure: error: /bin/sh './configure.gnu' failed for libs/apr-util
>
>
>
> i was using ./configure --prefix=/usr/local/conference/
>
>
>
> Regards
> Sam
>
>
>
>
> On Thu, Dec 23, 2010 at 3:49 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Try enabling siptrace (sofia profile external siptrace on) - does it
>> show an INVITE being sent in the logs?
>>
>> -Steve
>>
>>
>> On 23 December 2010 09:20, Sam <u2nsam at gmail.com> wrote:
>> > Hello friends,
>> >
>> > I am using the dialplan,
>> >
>> > <action application="bridge" data="sofia/external/12345 at 192.168.2.3"/>
>> >
>> > its not sending INVITE to the server 192.168.2.3
>> >
>> > i am getting the below logs:-
>> >
>> > 2010-12-23 14:03:02.509500 [NOTICE] switch_channel.c:784 New Channel
>> > sofia/external/12345 at 192.168.2.3 [418e0dfa-0e6f-11e0-a5e1-33ac6c473a4e]
>> > 2010-12-23 14:03:02.509500 [DEBUG] mod_sofia.c:4052
>> > (sofia/external/12345 at 192.168.2.3) State Change CS_NEW -> CS_INIT
>> > 2010-12-23 14:03:02.509500 [DEBUG] switch_core_session.c:1083 Send
>> signal
>> > sofia/external/12345 at 192.168.2.3 [BREAK]
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
>> > (sofia/external/12345 at 192.168.2.3) Running State Change CS_INIT
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
>> > (sofia/external/12345 at 192.168.2.3) State INIT
>> > 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:86
>> > sofia/external/12345 at 192.168.2.3 SOFIA INIT
>> > 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:126
>> > (sofia/external/12345 at 192.168.2.3) State Change CS_INIT -> CS_ROUTING
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_session.c:1083 Send
>> signal
>> > sofia/external/12345 at 192.168.2.3 [BREAK]
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
>> > (sofia/external/12345 at 192.168.2.3) State INIT going to sleep
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
>> > (sofia/external/12345 at 192.168.2.3) Running State Change CS_ROUTING
>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_channel.c:1615
>> > (sofia/external/12345 at 192.168.2.3) Callstate Change DOWN -> RINGING
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
>> > (sofia/external/12345 at 192.168.2.3) State ROUTING
>> > 2010-12-23 14:03:02.511888 [DEBUG] mod_sofia.c:149
>> > sofia/external/12345 at 192.168.2.3 SOFIA ROUTING
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_ivr_originate.c:66
>> > (sofia/external/12345 at 192.168.2.3) State Change CS_ROUTING ->
>> > CS_CONSUME_MEDIA
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_session.c:1083 Send
>> signal
>> > sofia/external/12345 at 192.168.2.3 [BREAK]
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
>> > (sofia/external/12345 at 192.168.2.3) State ROUTING going to sleep
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:320
>> > (sofia/external/12345 at 192.168.2.3) Running State Change
>> CS_CONSUME_MEDIA
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:378
>> > (sofia/external/12345 at 192.168.2.3) State CONSUME_MEDIA
>> > 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
>> > sofia/external/12345 at 192.168.2.3 entering state [calling][0]
>> > 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
>> > sofia/external/12345 at 192.168.2.3 entering state [terminated][503]
>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_channel.c:2493
>> > (sofia/external/12345 at 192.168.2.3) Callstate Change RINGING -> HANGUP
>> > 2010-12-23 14:03:02.511888 [NOTICE] sofia.c:5246 Hangup
>> > sofia/external/12345 at 192.168.2.3 [CS_CONSUME_MEDIA]
>> > [NORMAL_TEMPORARY_FAILURE]
>> >
>> >
>> > Regards
>> > Sam
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From steveu at coppice.org  Sun Dec 26 08:43:23 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 26 Dec 2010 13:43:23 +0800
Subject: [Freeswitch-users] libg722_1: Cross Compilation
	of	src/make_dct4_tables.c
In-Reply-To: <1293326620424-5867008.post@n2.nabble.com>
References: <1293326620424-5867008.post@n2.nabble.com>
Message-ID: <4D16D5FB.40001@coppice.org>

On 12/26/2010 09:23 AM, mazilo wrote:
> I am facing a dilemma trying to cross-compile libg722_1 package from
> FreeSWITCH git version for an ARM5 platform. First of all, the compilation
> is fine up to a point to produce a cross-platform src/make_dct4_tables
> binary file. From there on, the compilation process tries to execute the
> newly produced (cross-compiled) src/make_dct4_tables binary file on the host
> to produce a src/dct4.h file that contains tables. This certainly will crash
> the compilation process because the src/make_dct4_tables binary file is for
> an ARM5 platform and not an x86 platform where the host is. I can manually
> compile the src/make_dct4_tables.c on the host to produce an x86 binary file
> and use it to manually generate the src/dct4.h file, but this poses a
> dilemma. Even though the src/dct4.h produced in this way will not have a
> problem to produce the libg722_1 codes to run/execute on an ARM5 platform,
> but it will probably not be able to correctly process the G722_1 audio
> streams. The reason is simply I suspect the src/dct4.h files generated by
> the src/make_dct4_tables binary file from both the x86 and ARM5 platforms
> are not identical. I hope experts here will be able to see this and come up
> with the solution.
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
I just committed some updates to the G.722.1 codec in the FreeSwitch 
repository. Can you see if that builds OK in your cross compiling 
environment?

Steve



From infos at madovsky.org  Sun Dec 26 09:03:53 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 26 Dec 2010 01:03:53 -0500
Subject: [Freeswitch-users] nibblebill schedule
Message-ID: <22BBEDA83CB1421397A8C2F19F40C1A0@e1705>

Or maybe a channel variable of
cash amount of the account like
nibblebill_cash before the bridge would be great
  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Saturday, December 25, 2010 12:16 PM
  Subject: Re: [Freeswitch-users] nibblebill schedule


  yes already thought also ;)
  and already set nibblebill hearbeat to 30s.
  but in fact the user is never in "trial" mode.
  it's in case that if he has less than nobal so he will
  always call for 30sec only.
  Or maybe make a real trial when he registers and once he bought his
  first prepaid I will remove it.
    ----- Original Message ----- 
    From: Avi Marcus 
    To: FreeSWITCH Users Help 
    Sent: Saturday, December 25, 2010 11:47 AM
    Subject: Re: [Freeswitch-users] nibblebill schedule


    Hmm. Alternatively.. 
    You could set a user-variable that shows they are on the trial.
    Then, before you set the price, you check for that variable and if it's true, you set, say, nibble_rate=.02 and make sure they start with a balance of .01
    You probably need to change the heartbeat to 30 seconds to do that.
    Good luck :)


    On Sat, Dec 25, 2010 at 6:42 PM, Madovsky <infos at madovsky.org> wrote:

      yes I thought to this but it would be easier to do it through nibblebill
        ----- Original Message ----- 
        From: Avi Marcus 
        To: FreeSWITCH Users Help 
        Sent: Saturday, December 25, 2010 11:30 AM
        Subject: Re: [Freeswitch-users] nibblebill schedule


        Call 30 seconds as a trial?
        Perhaps skip nibblebill and use: sched_hangup
        And count it somewhere that they used their trial period (using Mod_db with odbc  or mod_odbc_query from git contrib) 


        -Avi

        On Sat, Dec 25, 2010 at 6:22 PM, Madovsky <infos at madovsky.org> wrote:
        > ha ok,
        > the concept I thought is to let user to call 30s as a trial and
        > start niblebill after this time.... does it need to modify the
        > mod_nibblebill source code ?
        >  
        > Thanks
        > F
        >
        > ----- Original Message -----
        > From: Rupa Schomaker
        > To: FreeSWITCH Users Help
        > Sent: Saturday, December 25, 2010 8:38 AM
        > Subject: Re: [Freeswitch-users] nibblebill schedule
        > I don't see how without change.
        > You have no idea how fast the $$ is being nibbled away by other calls.  So,
        > while you can account for a 30s buffer in the current call using the current
        > rate, the more calls that are up for that account the more "off" the 30s
        > estimate will be.  
        >
        > On Fri, Dec 24, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:
        >>
        >> Is it possible to schedule the transfer to nibblebill nobal_action ?
        >> I'd like to schedule of 30s before the call is cut and go to nobal_action
        >> extension
        >>  
        >> thanks
        >
        > --
        > -Rupa
        >
        > ________________________________
        >
        > _______________________________________________
        > FreeSWITCH-users mailing list
        > FreeSWITCH-users at lists.freeswitch.org
        > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        > http://www.freeswitch.org
        >
        > _______________________________________________
        > FreeSWITCH-users mailing list
        > FreeSWITCH-users at lists.freeswitch.org
        > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        > http://www.freeswitch.org
        >
        >




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


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



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






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


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From a.afzali2003 at gmail.com  Sun Dec 26 11:42:10 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sun, 26 Dec 2010 12:12:10 +0330
Subject: [Freeswitch-users] Converting WAV to MP3 by LAME
In-Reply-To: <F6F081015F18495AA7CFA8873D8F72DF@e1705>
References: <AANLkTimmTjP0Zaj9u_neY9=+QT1XVR9v8RgPzURqXgc1@mail.gmail.com>
	<AANLkTinhKD6fR9ke0MbejetEXu00TQSV4dPN_J5R-=au@mail.gmail.com>
	<F6F081015F18495AA7CFA8873D8F72DF@e1705>
Message-ID: <AANLkTim2osK1nG2WUk4Yq-H7uH8bNS0jFcm5dea9X6k4@mail.gmail.com>

Thanks for suggestions,

-- afshin

On Sun, Dec 26, 2010 at 4:43 AM, Madovsky <infos at madovsky.org> wrote:

> or use Perl application
>
> ----- Original Message -----
> From: "curriegrad2004" <curriegrad2004 at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, December 25, 2010 7:47 PM
> Subject: Re: [Freeswitch-users] Converting WAV to MP3 by LAME
>
>
> > You can do this with a shell script easily through the dialplan.
> > api_hangup_hook and system should do what you want it to do.
> >
> > On Sat, Dec 25, 2010 at 10:54 AM, afshin afzali <a.afzali2003 at gmail.com>
> > wrote:
> >> Hi Guys,
> >>
> >> I want to use LAME to convert session recorded call logs to mp3 format
> in
> >> a
> >> Lua script.
> >> Is there a better way for this?
> >>
> >> BEST,
> >>
> >> -- afshin
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From wstephen80 at gmail.com  Sun Dec 26 13:45:52 2010
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Sun, 26 Dec 2010 11:45:52 +0100
Subject: [Freeswitch-users] hangup_after_bridge and failure_causes: what is
	wrong in my dialpan?
Message-ID: <AANLkTin579attOO86yhPdOoRbrwf2p-hxA96NY1q-AoW@mail.gmail.com>

My dialplan is:

        <action application="set"
data="failure_causes=NO_ROUTE_DESTINATION,NORMAL_CIRCUIT_CONGESTION,SWITCH_CONGESTION,NORMAL_TEMPORARY_FAILURE"
/>
        <action application="set" data="hangup_after_bridge=true" />
        <action application="limit_execute" data="hash outbound ${hash0}
${limit0} bridge ${dialstring0}" />
        <action application="limit_execute" data="hash outbound ${hash1}
${limit1} bridge ${dialstring1}" />
        <action application="limit_execute" data="hash outbound ${hash2}
${limit2} bridge ${dialstring2}" />
        <action application="limit_execute" data="hash outbound ${hash3}
${limit3} bridge ${dialstring3}" />
        <action application="limit_execute" data="hash outbound ${hash4}
${limit4} bridge ${dialstring4}" />
        <action application="hangup" data="${bridge_hangup_cause}"/>

What I expect is that the incoming call is hangup when the bridge fails for
any reason except the reasons specified in "failure_causes", and in case the
bridge fail for a reason specified in "failure_causes" the dialplan
execution continues.

My problem is that, for example, I see in my CDR that there is 1 inbound and
5 related outbound calls (one for each bridge in my dialplan)
for NO_USER_RESPONSE, USER_BUSY, NORMAL_TEMPORARY_FAILURE, UNALLOCATED_NUMBER
so the behaviour is different from what I expect: what is wrong in my
dialplan?
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From steveayre at gmail.com  Sun Dec 26 16:46:17 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 26 Dec 2010 13:46:17 +0000
Subject: [Freeswitch-users] no invite send
In-Reply-To: <AANLkTim2O1mQkfbWX6cqzFE9xJoQL7+oDs4v43g7+T+J@mail.gmail.com>
References: <AANLkTimBh+CqTz1O5qNmd7=z+Z6WZGeY1ua8kdZAwzH5@mail.gmail.com>
	<AANLkTikpL3pfFieW3Yv20iFxDo+yNdYA9zoyd3tgT-iK@mail.gmail.com>
	<AANLkTim0hhJBaJSg9ghh8cwrbSY8_rC+4oOP79vT-Hw4@mail.gmail.com>
	<AANLkTim2O1mQkfbWX6cqzFE9xJoQL7+oDs4v43g7+T+J@mail.gmail.com>
Message-ID: <8DD82FC8-48B4-438D-BE4C-90D3162C4F91@gmail.com>

That is correct, and yes you can run more than one if they use different ports. You can use a single installation to save space but use the arguments to freeswitch to use different config/run/log directories.

Steve on iPhone

On 26 Dec 2010, at 04:20, Sam <u2nsam at gmail.com> wrote:

> Further to this question:
> 
> I have installed FS without prefix and when installing with prefix ./configure --prefix=/usr/local/conference/ ;it gives me an error,
> 
> how is the best method to install FS with prefix and can we install more than 1 FS instance on the same server as i can do them to listen on different ports?
> 
> Regds
> Sam
> 
> 
> 
> On Thu, Dec 23, 2010 at 3:58 PM, Sam <u2nsam at gmail.com> wrote:
> I check it in sip trace ... it do not sends invite, probably i might have miss configured something,
> on the another FS server it works fine, this happened on the new installation where i was using prefix,
> 
> I was trying to reinstall FS on the new server with prefix it gave me below error.
> 
> ./configure: line 38199: 30565 Segmentation fault      /bin/sh /usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat/configure '--prefix=/usr/local/conference/' 'CONFIGURE_CFLAGS=-g -O2' 'CONFIGURE_CXXFLAGS=-g -O2' 'CONFIGURE_LDFLAGS=' '--cache-file=/dev/null' '--srcdir=.' '--with-apr=../apr' '--disable-shared' '--with-pic' '--without-sqlite2' '--without-sqlite3' '--with-expat=builtin' --cache-file=/dev/null --srcdir=/usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat --prefix=/usr/local/conference/ --exec-prefix=${prefix} --libdir=${exec_prefix}/lib --includedir=${prefix}/include/apr-${APRUTIL_MAJOR_VERSION} --bindir=${exec_prefix}/bin
> configure failed for xml/expat
> configure: error: /bin/sh './configure.gnu' failed for libs/apr-util
> 
> 
> 
> i was using ./configure --prefix=/usr/local/conference/
> 
> 
> 
> Regards
> Sam
> 
> 
> 
> 
> On Thu, Dec 23, 2010 at 3:49 PM, Steven Ayre <steveayre at gmail.com> wrote:
> Try enabling siptrace (sofia profile external siptrace on) - does it
> show an INVITE being sent in the logs?
> 
> -Steve
> 
> 
> On 23 December 2010 09:20, Sam <u2nsam at gmail.com> wrote:
> > Hello friends,
> >
> > I am using the dialplan,
> >
> > <action application="bridge" data="sofia/external/12345 at 192.168.2.3"/>
> >
> > its not sending INVITE to the server 192.168.2.3
> >
> > i am getting the below logs:-
> >
> > 2010-12-23 14:03:02.509500 [NOTICE] switch_channel.c:784 New Channel
> > sofia/external/12345 at 192.168.2.3 [418e0dfa-0e6f-11e0-a5e1-33ac6c473a4e]
> > 2010-12-23 14:03:02.509500 [DEBUG] mod_sofia.c:4052
> > (sofia/external/12345 at 192.168.2.3) State Change CS_NEW -> CS_INIT
> > 2010-12-23 14:03:02.509500 [DEBUG] switch_core_session.c:1083 Send signal
> > sofia/external/12345 at 192.168.2.3 [BREAK]
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
> > (sofia/external/12345 at 192.168.2.3) Running State Change CS_INIT
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
> > (sofia/external/12345 at 192.168.2.3) State INIT
> > 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:86
> > sofia/external/12345 at 192.168.2.3 SOFIA INIT
> > 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:126
> > (sofia/external/12345 at 192.168.2.3) State Change CS_INIT -> CS_ROUTING
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_session.c:1083 Send signal
> > sofia/external/12345 at 192.168.2.3 [BREAK]
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
> > (sofia/external/12345 at 192.168.2.3) State INIT going to sleep
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
> > (sofia/external/12345 at 192.168.2.3) Running State Change CS_ROUTING
> > 2010-12-23 14:03:02.510864 [DEBUG] switch_channel.c:1615
> > (sofia/external/12345 at 192.168.2.3) Callstate Change DOWN -> RINGING
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
> > (sofia/external/12345 at 192.168.2.3) State ROUTING
> > 2010-12-23 14:03:02.511888 [DEBUG] mod_sofia.c:149
> > sofia/external/12345 at 192.168.2.3 SOFIA ROUTING
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_ivr_originate.c:66
> > (sofia/external/12345 at 192.168.2.3) State Change CS_ROUTING ->
> > CS_CONSUME_MEDIA
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_session.c:1083 Send signal
> > sofia/external/12345 at 192.168.2.3 [BREAK]
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
> > (sofia/external/12345 at 192.168.2.3) State ROUTING going to sleep
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:320
> > (sofia/external/12345 at 192.168.2.3) Running State Change CS_CONSUME_MEDIA
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:378
> > (sofia/external/12345 at 192.168.2.3) State CONSUME_MEDIA
> > 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
> > sofia/external/12345 at 192.168.2.3 entering state [calling][0]
> > 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
> > sofia/external/12345 at 192.168.2.3 entering state [terminated][503]
> > 2010-12-23 14:03:02.511888 [DEBUG] switch_channel.c:2493
> > (sofia/external/12345 at 192.168.2.3) Callstate Change RINGING -> HANGUP
> > 2010-12-23 14:03:02.511888 [NOTICE] sofia.c:5246 Hangup
> > sofia/external/12345 at 192.168.2.3 [CS_CONSUME_MEDIA]
> > [NORMAL_TEMPORARY_FAILURE]
> >
> >
> > Regards
> > Sam
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From infos at madovsky.org  Sun Dec 26 17:08:39 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 26 Dec 2010 09:08:39 -0500
Subject: [Freeswitch-users] deactivate nibblebill in dialplan
Message-ID: <2ACB47EEB82F4D2AA8C3CAF903A9CC74@e1705>

Is it possilbe to deactivate nibblebill in dialplan ?
I'd like to allow the caller to reach the voice machine of the callee in case of the bridge
fails and if his balance is under the nobal value in nibblebill.conf.xml


        <extension name="inbound">
                <condition field="${sip_to_uri}" expression="^([2-9]\d{10,15})@$${domain}$">
                        <action application="set" data="hangup_after_bridge=true"/>
                        <action inline="true" application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,SUBSCRIBER_ABSENT,CALL_REJECTED,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
                        <action application="set" data="instant_ringback=true"/>
                        <action application="ring_ready"/>
                        <action inline="true" application="lcr" data="$1 default"/>
                        <action inline="true" application="set" data="nibble_rate=0.01"/>
                        <action application="export" data="originate_timeout=19"/>
                        <action application="bridge" data="sofia/gateway/$1"/>
                        <!-- how to deactivate nibblebill here or allow user to continue even if his balance is to 0 ? -->
                        <action application="sleep" data="3000"/>
                        <action application="answer"/>
                        <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
                </condition>
        </extension>


Thanks
-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

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From Avi at aMarcus.com  Sun Dec 26 17:24:19 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Sun, 26 Dec 2010 16:24:19 +0200
Subject: [Freeswitch-users] deactivate nibblebill in dialplan
In-Reply-To: <2ACB47EEB82F4D2AA8C3CAF903A9CC74@e1705>
References: <2ACB47EEB82F4D2AA8C3CAF903A9CC74@e1705>
Message-ID: <AANLkTik7KaTDuugyFTPy8rsQYUj9verajU=Zf+XQyRHP@mail.gmail.com>

I *suppose* it's a 2 step process.
You need to set the nobal to transfer them to an extension that checks if
it's relevant to send them to voicemail, else, it send them to a "Empty
balance" error message.
Then, on the voicemail part, you simple unset nibble_account and
mod_nibblebill ignores the channel. No account = no billing.

-Avi Marcus

On Sun, Dec 26, 2010 at 4:08 PM, Madovsky <infos at madovsky.org> wrote:

>  Is it possilbe to deactivate nibblebill in dialplan ?
> I'd like to allow the caller to reach the voice machine of the callee in
> case of the bridge
> fails and if his balance is under the nobal value in nibblebill.conf.xml
>
>
>         <extension name="inbound">
>                 <condition field="${sip_to_uri}"
> expression="^([2-9]\d{10,15})@$${domain}$">
>                         <action application="set"
> data="hangup_after_bridge=true"/>
>                         <action inline="true" application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,SUBSCRIBER_ABSENT,CALL_REJECTED,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
>                         <action application="set"
> data="instant_ringback=true"/>
>                         <action application="ring_ready"/>
>                         <action inline="true" application="lcr" data="$1
> default"/>
>                         <action inline="true" application="set"
> data="nibble_rate=0.01"/>
>                         <action application="export"
> data="originate_timeout=19"/>
>                         <action application="bridge"
> data="sofia/gateway/$1"/>
>                         <!-- how to deactivate nibblebill here or allow
> user to continue even if his balance is to 0 ? -->
>                         <action application="sleep" data="3000"/>
>                         <action application="answer"/>
>                         <action application="voicemail" data="default
> ${domain_name} ${dialed_extension}"/>
>                 </condition>
>         </extension>
>
> Thanks
>
> --
> This message has been scanned for viruses and
> dangerous content by *MailScanner* <http://www.mailscanner.info/>, and is
> believed to be clean.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Sun Dec 26 17:41:12 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 26 Dec 2010 09:41:12 -0500
Subject: [Freeswitch-users] condition weirdness
Message-ID: <A983598584834A44B3CD83DF95F07B4F@e1705>

If i call the number 000123456789 at domain

and in my dialplan I have something like

...
<condition field="${sip_to_uri}" expression="^([1|51|758|597|868|598|58]\d{10,15})@$${domain}$">
...

in the log I can see condition passed... I'm confused
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From infos at madovsky.org  Sun Dec 26 17:48:05 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 26 Dec 2010 09:48:05 -0500
Subject: [Freeswitch-users] deactivate nibblebill in dialplan
References: <2ACB47EEB82F4D2AA8C3CAF903A9CC74@e1705>
	<AANLkTik7KaTDuugyFTPy8rsQYUj9verajU=Zf+XQyRHP@mail.gmail.com>
Message-ID: <3A8FE2E7591F4BC69DB025A1B32AC8FC@e1705>

ok thanks,
to unset nibble vars I need to set it to empty ?
  ----- Original Message ----- 
  From: Avi Marcus 
  To: FreeSWITCH Users Help 
  Sent: Sunday, December 26, 2010 9:24 AM
  Subject: Re: [Freeswitch-users] deactivate nibblebill in dialplan


  I suppose it's a 2 step process.
  You need to set the nobal to transfer them to an extension that checks if it's relevant to send them to voicemail, else, it send them to a "Empty balance" error message.
  Then, on the voicemail part, you simple unset nibble_account and mod_nibblebill ignores the channel. No account = no billing.


  -Avi Marcus


  On Sun, Dec 26, 2010 at 4:08 PM, Madovsky <infos at madovsky.org> wrote:

    Is it possilbe to deactivate nibblebill in dialplan ?
    I'd like to allow the caller to reach the voice machine of the callee in case of the bridge
    fails and if his balance is under the nobal value in nibblebill.conf.xml


            <extension name="inbound">
                    <condition field="${sip_to_uri}" expression="^([2-9]\d{10,15})@$${domain}$">
                            <action application="set" data="hangup_after_bridge=true"/>
                            <action inline="true" application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,SUBSCRIBER_ABSENT,CALL_REJECTED,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
                            <action application="set" data="instant_ringback=true"/>
                            <action application="ring_ready"/>
                            <action inline="true" application="lcr" data="$1 default"/>
                            <action inline="true" application="set" data="nibble_rate=0.01"/>
                            <action application="export" data="originate_timeout=19"/>
                            <action application="bridge" data="sofia/gateway/$1"/>
                            <!-- how to deactivate nibblebill here or allow user to continue even if his balance is to 0 ? -->
                            <action application="sleep" data="3000"/>
                            <action application="answer"/>
                            <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
                    </condition>
            </extension>


    Thanks

    -- 
    This message has been scanned for viruses and 
    dangerous content by MailScanner, and is 
    believed to be clean. 

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






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


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From tayeb.meftah at gmail.com  Sun Dec 26 19:29:52 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sun, 26 Dec 2010 17:29:52 +0100
Subject: [Freeswitch-users] deactivate nibblebill in dialplan
In-Reply-To: <3A8FE2E7591F4BC69DB025A1B32AC8FC@e1705>
References: <2ACB47EEB82F4D2AA8C3CAF903A9CC74@e1705>	<AANLkTik7KaTDuugyFTPy8rsQYUj9verajU=Zf+XQyRHP@mail.gmail.com>
	<3A8FE2E7591F4BC69DB025A1B32AC8FC@e1705>
Message-ID: <4D176D80.70505@gmail.com>

set nibble_rate to 0
Le 26/12/2010 15:48, Madovsky a ?crit :
> ok thanks,
> to unset nibble vars I need to set it to empty ?
>
>     ----- Original Message -----
>     *From:* Avi Marcus <mailto:Avi at aMarcus.com>
>     *To:* FreeSWITCH Users Help
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     *Sent:* Sunday, December 26, 2010 9:24 AM
>     *Subject:* Re: [Freeswitch-users] deactivate nibblebill in dialplan
>
>     I /suppose/ it's a 2 step process.
>     You need to set the nobal to transfer them to an extension that
>     checks if it's relevant to send them to voicemail, else, it send
>     them to a "Empty balance" error message.
>     Then, on the voicemail part, you simple unset nibble_account and
>     mod_nibblebill ignores the channel. No account = no billing.
>
>     -Avi Marcus
>
>     On Sun, Dec 26, 2010 at 4:08 PM, Madovsky <infos at madovsky.org
>     <mailto:infos at madovsky.org>> wrote:
>
>         Is it possilbe to deactivate nibblebill in dialplan ?
>         I'd like to allow the caller to reach the voice machine of the
>         callee in case of the bridge
>         fails and if his balance is under the nobal value in
>         nibblebill.conf.xml
>         <extension name="inbound">
>         <condition field="${sip_to_uri}"
>         expression="^([2-9]\d{10,15})@$${domain}$">
>         <action application="set" data="hangup_after_bridge=true"/>
>         <action inline="true" application="set"
>         data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,SUBSCRIBER_ABSENT,CALL_REJECTED,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
>         <action application="set" data="instant_ringback=true"/>
>         <action application="ring_ready"/>
>         <action inline="true" application="lcr" data="$1 default"/>
>         <action inline="true" application="set" data="nibble_rate=0.01"/>
>         <action application="export" data="originate_timeout=19"/>
>         <action application="bridge" data="sofia/gateway/$1"/>
>         <!-- how to deactivate nibblebill here or allow user to
>         continue even if his balance is to 0 ? -->
>         <action application="sleep" data="3000"/>
>         <action application="answer"/>
>         <action application="voicemail" data="default ${domain_name}
>         ${dialed_extension}"/>
>         </condition>
>         </extension>
>         Thanks
>
>         -- 
>         This message has been scanned for viruses and
>         dangerous content by *MailScanner*
>         <http://www.mailscanner.info/>, and is
>         believed to be clean.
>
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>     ------------------------------------------------------------------------
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313

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From Nabble at slickdeals.endjunk.com  Sun Dec 26 19:34:54 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 26 Dec 2010 08:34:54 -0800 (PST)
Subject: [Freeswitch-users] libg722_1: Cross Compilation
	of	src/make_dct4_tables.c
In-Reply-To: <4D16D5FB.40001@coppice.org>
References: <1293326620424-5867008.post@n2.nabble.com>
	<4D16D5FB.40001@coppice.org>
Message-ID: <1293381294171-5867851.post@n2.nabble.com>


Steve,

Thanks for your quick response with patches. The libg722_1 is now
cross-compilable for ARM5 platform to produce a host
libg722_1/src/make_dct4_tables binary file. This will let the compilation
process to generate the libg722_1/src/dct4.h file. However, I wonder if the 
libg722_1 binary code based on such dct4.h file will yield some logical
errors in processing the G722 audio stream. Anyway, 
http://pastebin.com/BQC1iuYr here  is the dct4.h file generated from the
make_dct4_tables on an ARM5 platform (a Seagate 
http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
DockStar ) for your perusal.

BTW, if anyone here knows of a freely available Linux softphone that
supports G722 CoDec, I sure will appreciate that. This way, I can use it to
test with this libg722_1.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/libg722-1-Cross-Compilation-of-src-make-dct4-tables-c-tp5867008p5867851.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From u2nsam at gmail.com  Sun Dec 26 19:44:57 2010
From: u2nsam at gmail.com (Sam)
Date: Sun, 26 Dec 2010 22:14:57 +0530
Subject: [Freeswitch-users] no invite send
In-Reply-To: <8DD82FC8-48B4-438D-BE4C-90D3162C4F91@gmail.com>
References: <AANLkTimBh+CqTz1O5qNmd7=z+Z6WZGeY1ua8kdZAwzH5@mail.gmail.com>
	<AANLkTikpL3pfFieW3Yv20iFxDo+yNdYA9zoyd3tgT-iK@mail.gmail.com>
	<AANLkTim0hhJBaJSg9ghh8cwrbSY8_rC+4oOP79vT-Hw4@mail.gmail.com>
	<AANLkTim2O1mQkfbWX6cqzFE9xJoQL7+oDs4v43g7+T+J@mail.gmail.com>
	<8DD82FC8-48B4-438D-BE4C-90D3162C4F91@gmail.com>
Message-ID: <AANLkTinvnjsUWjgEHavqBiTOZ108Op3-qL80o4L9oxbC@mail.gmail.com>

Hi Steve,

Can you pls elaborate "arguments to freeswitch to use different
config/run/log directories" as how to

Thanks
Sam

On Sun, Dec 26, 2010 at 7:16 PM, Steven Ayre <steveayre at gmail.com> wrote:

> That is correct, and yes you can run more than one if they use different
> ports. You can use a single installation to save space but use the arguments
> to freeswitch to use different config/run/log directories.
>
> Steve on iPhone
>
> On 26 Dec 2010, at 04:20, Sam <u2nsam at gmail.com> wrote:
>
> Further to this question:
>
> I have installed FS without prefix and when installing with prefix
> ./configure --prefix=/usr/local/conference/ ;it gives me an error,
>
> how is the best method to install FS with prefix and can we install more
> than 1 FS instance on the same server as i can do them to listen on
> different ports?
>
> Regds
> Sam
>
>
>
> On Thu, Dec 23, 2010 at 3:58 PM, Sam < <u2nsam at gmail.com>u2nsam at gmail.com>wrote:
>
>> I check it in sip trace ... it do not sends invite, probably i might have
>> miss configured something,
>> on the another FS server it works fine, this happened on the new
>> installation where i was using prefix,
>>
>> I was trying to reinstall FS on the new server with prefix it gave me
>> below error.
>>
>> ./configure: line 38199: 30565 Segmentation fault      /bin/sh
>> /usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat/configure
>> '--prefix=/usr/local/conference/' 'CONFIGURE_CFLAGS=-g -O2'
>> 'CONFIGURE_CXXFLAGS=-g -O2' 'CONFIGURE_LDFLAGS=' '--cache-file=/dev/null'
>> '--srcdir=.' '--with-apr=../apr' '--disable-shared' '--with-pic'
>> '--without-sqlite2' '--without-sqlite3' '--with-expat=builtin'
>> --cache-file=/dev/null
>> --srcdir=/usr/src/freeswitch-1.0.6/libs/apr-util/xml/expat
>> --prefix=/usr/local/conference/ --exec-prefix=${prefix}
>> --libdir=${exec_prefix}/lib
>> --includedir=${prefix}/include/apr-${APRUTIL_MAJOR_VERSION}
>> --bindir=${exec_prefix}/bin
>> configure failed for xml/expat
>> configure: error: /bin/sh './configure.gnu' failed for libs/apr-util
>>
>>
>>
>> i was using ./configure --prefix=/usr/local/conference/
>>
>>
>>
>> Regards
>> Sam
>>
>>
>>
>>
>> On Thu, Dec 23, 2010 at 3:49 PM, Steven Ayre < <steveayre at gmail.com>
>> steveayre at gmail.com> wrote:
>>
>>> Try enabling siptrace (sofia profile external siptrace on) - does it
>>> show an INVITE being sent in the logs?
>>>
>>> -Steve
>>>
>>>
>>> On 23 December 2010 09:20, Sam < <u2nsam at gmail.com>u2nsam at gmail.com>
>>> wrote:
>>> > Hello friends,
>>> >
>>> > I am using the dialplan,
>>> >
>>> > <action application="bridge" data="sofia/external/ <12345 at 192.168.2.3>
>>> 12345 at 192.168.2.3"/>
>>> >
>>> > its not sending INVITE to the server 192.168.2.3
>>> >
>>> > i am getting the below logs:-
>>> >
>>> > 2010-12-23 14:03:02.509500 [NOTICE] switch_channel.c:784 New Channel
>>> > sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3[418e0dfa-0e6f-11e0-a5e1-33ac6c473a4e]
>>> > 2010-12-23 14:03:02.509500 [DEBUG] mod_sofia.c:4052
>>> > (sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3) State Change
>>> CS_NEW -> CS_INIT
>>> > 2010-12-23 14:03:02.509500 [DEBUG] switch_core_session.c:1083 Send
>>> signal
>>> > sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3 [BREAK]
>>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
>>> > (sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3) Running State
>>> Change CS_INIT
>>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
>>> > (sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3) State INIT
>>> > 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:86
>>> > sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3 SOFIA INIT
>>> > 2010-12-23 14:03:02.510864 [DEBUG] mod_sofia.c:126
>>> > (sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3) State Change
>>> CS_INIT -> CS_ROUTING
>>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_session.c:1083 Send
>>> signal
>>> > sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3 [BREAK]
>>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:356
>>> > (sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3) State INIT
>>> going to sleep
>>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_core_state_machine.c:320
>>> > (sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3) Running State
>>> Change CS_ROUTING
>>> > 2010-12-23 14:03:02.510864 [DEBUG] switch_channel.c:1615
>>> > (sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3) Callstate
>>> Change DOWN -> RINGING
>>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
>>> > (sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3) State ROUTING
>>> > 2010-12-23 14:03:02.511888 [DEBUG] mod_sofia.c:149
>>> > sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3 SOFIA ROUTING
>>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_ivr_originate.c:66
>>> > (sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3) State Change
>>> CS_ROUTING ->
>>> > CS_CONSUME_MEDIA
>>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_session.c:1083 Send
>>> signal
>>> > sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3 [BREAK]
>>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:359
>>> > (sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3) State ROUTING
>>> going to sleep
>>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:320
>>> > (sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3) Running State
>>> Change CS_CONSUME_MEDIA
>>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_core_state_machine.c:378
>>> > (sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3) State
>>> CONSUME_MEDIA
>>> > 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
>>> > sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3 entering state
>>> [calling][0]
>>> > 2010-12-23 14:03:02.511888 [DEBUG] sofia.c:4606 Channel
>>> > sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3 entering state
>>> [terminated][503]
>>> > 2010-12-23 14:03:02.511888 [DEBUG] switch_channel.c:2493
>>> > (sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3) Callstate
>>> Change RINGING -> HANGUP
>>> > 2010-12-23 14:03:02.511888 [NOTICE] sofia.c:5246 Hangup
>>> > sofia/external/ <12345 at 192.168.2.3>12345 at 192.168.2.3[CS_CONSUME_MEDIA]
>>> > [NORMAL_TEMPORARY_FAILURE]
>>> >
>>> >
>>> > Regards
>>> > Sam
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From linux4michelle at tamay-dogan.net  Sun Dec 26 19:46:55 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Sun, 26 Dec 2010 17:46:55 +0100
Subject: [Freeswitch-users] FreeSWITCH debs and rpms
In-Reply-To: <CD470134-1990-4F78-8617-55C0FB1BD2EC@seventhsignal.de>
References: <CD470134-1990-4F78-8617-55C0FB1BD2EC@seventhsignal.de>
Message-ID: <20101226164655.GI6539@michelle1>

Hello Michal Bielicki,

Am 2010-12-23 22:47:33, hacktest Du folgendes herunter:
> debs and rpms are available from git builds from
> http://repo.freeswitch.de/

Das ist ja genial  :-D

Gr??e und Sch?ne Weihnachten
    Michelle Konzack

-- 
##################### Debian GNU/Linux Consultant ######################
   Development of Intranet and Embedded Systems with Debian GNU/Linux

itsystems at tdnet France EURL       itsystems at tdnet UG (limited liability)
Owner Michelle Konzack            Owner Michelle Konzack

Apt. 917 (homeoffice)
50, rue de Soultz                 Kinzigstra?e 17
67100 Strasbourg/France           77694 Kehl/Germany
Tel: +33-6-61925193 mobil         Tel: +49-177-9351947 mobil
Tel: +33-9-52705884 fix

<http://www.itsystems.tamay-dogan.net/>  <http://www.flexray4linux.org/>
<http://www.debian.tamay-dogan.net/>         <http://www.can4linux.org/>

Jabber linux4michelle at jabber.ccc.de
ICQ    #328449886

Linux-User #280138 with the Linux Counter, http://counter.li.org/
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From linux4michelle at tamay-dogan.net  Sun Dec 26 19:52:02 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Sun, 26 Dec 2010 17:52:02 +0100
Subject: [Freeswitch-users] GSM-Modems and support
Message-ID: <20101226165202.GJ6539@michelle1>

Hello *,

sometimes ago I already had asked but the answer was not very  satisfait
at the time so I ask now again:

I want to build a local GSM-to-SIP-Gateway with 4 SIMs.  Does FreeSwitch
now support USB 1.1 GSM-Modems and if yes, which are recommened.

I realy mean inexpensive non UMTS/HSPA ones.

Thanks, Greetings and nice Day/Evening
    Michelle Konzack

-- 
##################### Debian GNU/Linux Consultant ######################
   Development of Intranet and Embedded Systems with Debian GNU/Linux

itsystems at tdnet France EURL       itsystems at tdnet UG (limited liability)
Owner Michelle Konzack            Owner Michelle Konzack

Apt. 917 (homeoffice)
50, rue de Soultz                 Kinzigstra?e 17
67100 Strasbourg/France           77694 Kehl/Germany
Tel: +33-6-61925193 mobil         Tel: +49-177-9351947 mobil
Tel: +33-9-52705884 fix

<http://www.itsystems.tamay-dogan.net/>  <http://www.flexray4linux.org/>
<http://www.debian.tamay-dogan.net/>         <http://www.can4linux.org/>

Jabber linux4michelle at jabber.ccc.de
ICQ    #328449886

Linux-User #280138 with the Linux Counter, http://counter.li.org/
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From steveu at coppice.org  Sun Dec 26 20:21:12 2010
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 27 Dec 2010 01:21:12 +0800
Subject: [Freeswitch-users] libg722_1: Cross
	Compilation	of	src/make_dct4_tables.c
In-Reply-To: <1293381294171-5867851.post@n2.nabble.com>
References: <1293326620424-5867008.post@n2.nabble.com>	<4D16D5FB.40001@coppice.org>
	<1293381294171-5867851.post@n2.nabble.com>
Message-ID: <4D177988.5000503@coppice.org>

On 12/27/2010 12:34 AM, mazilo wrote:
> Steve,
>
> Thanks for your quick response with patches. The libg722_1 is now
> cross-compilable for ARM5 platform to produce a host
> libg722_1/src/make_dct4_tables binary file. This will let the compilation
> process to generate the libg722_1/src/dct4.h file. However, I wonder if the
> libg722_1 binary code based on such dct4.h file will yield some logical
> errors in processing the G722 audio stream. Anyway,
> http://pastebin.com/BQC1iuYr here  is the dct4.h file generated from the
> make_dct4_tables on an ARM5 platform (a Seagate
> http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
> DockStar ) for your perusal.
Although that dct4.h full of floating point values will be generated in 
a cross build for an ARM, building for an ARM should automatically set 
fixed point mode. Therefore that dct4.h file will not actually be used. 
Do you have any reason to believe the codec which has been built is bad? 
I think others have cross-built this code successfully for ARMs.

Lots of people doing cross-builds hit problems, solve them on their own, 
but fail to report these issues up stream to the package developer. 
Working around things on your own is great, to get things going, but 
please do report these issues up stream. I think most library developers 
want their code to "just build and run" on the widest possible range of 
platforms, without any manual tweaking. I know I certainly do. No 
developer has every platform available, so it takes input from people 
hitting these problems to eliminate them.

Steve



From Avi at aMarcus.com  Sun Dec 26 20:39:34 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Sun, 26 Dec 2010 19:39:34 +0200
Subject: [Freeswitch-users] deactivate nibblebill in dialplan
In-Reply-To: <4D176D80.70505@gmail.com>
References: <2ACB47EEB82F4D2AA8C3CAF903A9CC74@e1705>
	<AANLkTik7KaTDuugyFTPy8rsQYUj9verajU=Zf+XQyRHP@mail.gmail.com>
	<3A8FE2E7591F4BC69DB025A1B32AC8FC@e1705> <4D176D80.70505@gmail.com>
Message-ID: <AANLkTi=P9mVfE8tqvF20HigP8uNi2N9bidVJiU2TL7Zp@mail.gmail.com>

If you set nibble_rate to 0, it still runs and issues db queries. Maybe it
even cuts them off at nobal, even. Rather, unset the nibble_account.
-Avi

On Sun, Dec 26, 2010 at 6:29 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:

>  set nibble_rate to 0
> Le 26/12/2010 15:48, Madovsky a ?crit :
>
> ok thanks,
> to unset nibble vars I need to set it to empty ?
>
> ----- Original Message -----
>  *From:* Avi Marcus <Avi at aMarcus.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Sunday, December 26, 2010 9:24 AM
> *Subject:* Re: [Freeswitch-users] deactivate nibblebill in dialplan
>
>  I *suppose* it's a 2 step process.
> You need to set the nobal to transfer them to an extension that checks if
> it's relevant to send them to voicemail, else, it send them to a "Empty
> balance" error message.
> Then, on the voicemail part, you simple unset nibble_account and
> mod_nibblebill ignores the channel. No account = no billing.
>
>  -Avi Marcus
>
> On Sun, Dec 26, 2010 at 4:08 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  Is it possilbe to deactivate nibblebill in dialplan ?
>> I'd like to allow the caller to reach the voice machine of the callee in
>> case of the bridge
>> fails and if his balance is under the nobal value in nibblebill.conf.xml
>>
>>
>>         <extension name="inbound">
>>                 <condition field="${sip_to_uri}"
>> expression="^([2-9]\d{10,15})@$${domain}$">
>>                         <action application="set"
>> data="hangup_after_bridge=true"/>
>>                         <action inline="true" application="set"
>> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,SUBSCRIBER_ABSENT,CALL_REJECTED,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
>>                         <action application="set"
>> data="instant_ringback=true"/>
>>                         <action application="ring_ready"/>
>>                         <action inline="true" application="lcr" data="$1
>> default"/>
>>                         <action inline="true" application="set"
>> data="nibble_rate=0.01"/>
>>                         <action application="export"
>> data="originate_timeout=19"/>
>>                         <action application="bridge"
>> data="sofia/gateway/$1"/>
>>                         <!-- how to deactivate nibblebill here or allow
>> user to continue even if his balance is to 0 ? -->
>>                         <action application="sleep" data="3000"/>
>>                         <action application="answer"/>
>>                         <action application="voicemail" data="default
>> ${domain_name} ${dialed_extension}"/>
>>                 </condition>
>>         </extension>
>>
>> Thanks
>>
>> --
>> This message has been scanned for viruses and
>> dangerous content by *MailScanner* <http://www.mailscanner.info/>, and is
>>
>> believed to be clean.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>   ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Meftah Tayeb
> inum: +883510001288000
> Phone: +13602276297
> Fax: +12538020313
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From a.afzali2003 at gmail.com  Sun Dec 26 20:46:18 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sun, 26 Dec 2010 21:16:18 +0330
Subject: [Freeswitch-users] DTMF missing
In-Reply-To: <AANLkTincAApO4JDwzt-5oX0iD2VozFNVCtembOumL4iS@mail.gmail.com>
References: <AANLkTik=NLa+injL5hF=VsymwoE9wd1Wb2joXJDhvfFZ@mail.gmail.com>
	<AANLkTincAApO4JDwzt-5oX0iD2VozFNVCtembOumL4iS@mail.gmail.com>
Message-ID: <AANLkTin14RN-WxkYzqkQPRZctYt09xkWguAoCr8eeUWe@mail.gmail.com>

Sam,

Yes, The issue has resolved in last git :)
-- afshin

On Sun, Dec 26, 2010 at 7:33 AM, Sam <u2nsam at gmail.com> wrote:

> Hello
>
> Just like Afshin, I am also missing DTMF digits ,using FreeSWITCH Version
> 1.0.head (git-34a0ca5 2010-12-22 20-38-57 -0600) ;
> should i upgrade to latest ?
>
> Regards
> Sam
>
>
>
> On Fri, Dec 24, 2010 at 4:10 PM, Sam <u2nsam at gmail.com> wrote:
>
>> Hi,,
>>
>>
>> I have installed the latest ver of freeswitch and i have configured the
>> conference.
>>
>> now when i punch in the digits for password , i could see that the DTMF
>> digits are missed on fs_cli.
>>
>> it only happens when i dial it from polycom or cisco phones.
>>
>> I have tried with and without these values below:-
>>
>> <param name="dtmf-type" value="rfc2833"/>
>>     <param name="pass-rfc2833" value="true"/>
>>
>>
>>
>> traces fetched:
>> 192.168.2.49:5060 -> 192.168.2.190:5060
>>   INVITE sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190> SIP/2.0..Via:
>> SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From: "7028" <
>> sip:7028 at 192.168.2.190 <sip%3A7028 at 192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To:
>> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>..Call-ID:
>> 0017592a-eb33001a-
>>   63da3294-1a7bfdfa at 192.168.2.49..Max-Forwards: 70..Date: Fri, 24 Dec
>> 2010 10:13:52 GMT..CSeq: 102 INVITE..User-Agent: Cisco-CP7940G/8.0..Contact:
>> <sip:7028 at 192.168.2.49:5060;transport=udp>..Proxy-Authorization: Digest
>> username="7028"
>>   ,realm="192.168.2.190",uri="sip:7050 at 192.168.2.190<sip%3A7050 at 192.168.2.190>",response="a668f5c480285b35e7ff6bcd446879f0",nonce="d2c540f2-8487-4d87-bdab-871585253eb8",cnonce="0a6c4176",qop=auth,nc=00000001,algorithm=MD5..Expires:
>> 180..Accept: application/sdp
>>   ..Allow:
>> ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported:
>> replaces,join,norefersub..Content-Length: 220..Content-Type:
>> application/sdp..Content-Disposition:
>> session;handling=optional....v=0..o=Cisco-SIPUA 16102
>>    0 IN IP4 192.168.2.49..s=SIP Call..t=0 0..m=audio 17298 RTP/AVP 0 8
>> 18..c=IN IP4 192.168.2.49..a=rtpmap:0 PCMU/8000..a=rtpmap:8
>> PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=sendrecv..
>>
>>
>> 192.168.2.190:5060 -> 192.168.2.49:5060
>>   SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From:
>> "7028" <sip:7028 at 192.168.2.190 <sip%3A7028 at 192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To:
>> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;tag=2XXUZpgr1rvgc..Call-ID:
>> 0017592a-eb33001a-63da3
>>   294-1a7bfdfa at 192.168.2.49..CSeq: 102 INVITE..Contact:
>> <sip:7050 at 192.168.2.190:5060;transport=udp>..User-Agent: NOVANET..Accept:
>> application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
>> INFO, REGISTER, REFER, NOTIF
>>   Y, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path,
>> replaces..Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> sla, include-session-description, presence.winfo, message-summary,
>> refer..Session-Expires: 180
>>   0;refresher=uas..Min-SE: 120..Content-Type:
>> application/sdp..Content-Disposition: session..Content-Length:
>> 249..Remote-Party-ID: "7050" <sip:7050 at 192.168.2.190<sip%3A7050 at 192.168.2.190>>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH
>> 1293163579 129
>>   3163580 IN IP4 192.168.2.190..s=FreeSWITCH..c=IN IP4 192.168.2.190..t=0
>> 0..m=audio 22050 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101
>> telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - -
>> -..a=ptime:20..
>>
>> But the dtmf are not missed when punched on eyebeam softphone.
>>
>> And all the phones have RFC 2833.
>>
>> traces fetched for softphone:-
>>
>> 192.168.2.17:6182 -> 192.168.2.190:5060
>>   INVITE sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190> SIP/2.0..To: <
>> sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>..From: 7001<
>> sip:7001 at 192.168.2.190 <sip%3A7001 at 192.168.2.190>>;tag=6c557c1e..Via:
>> SIP/2.0/UDP 192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport..Call-ID:
>> d32ffe546570a77e..CS
>>   eq: 2 INVITE..Contact: <sip:7001 at 192.168.2.17:6182>..Max-Forwards:
>> 70..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>> SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Authorization: Digest
>> username="7001",rea
>>   lm="192.168.2.190",nonce="d0b4db6e-bd76-447f-9076-2e6b7809cb54",uri="
>> sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>",response="45fb9eb4f0e0e4fffd87a22769a007ba",cnonce="1c27f3687059b16d",nc=00000001,qop=auth,algorithm=MD5..User-Agent:
>> eyeBeam release 3007n
>>    stamp 17816..Content-Length: 233....v=0..o=- 27833664 27833670 IN IP4
>> 192.168.2.17..s=eyeBeam..c=IN IP4 192.168.2.17..t=0 0..m=audio 6398 RTP/AVP
>> 0 18 101..a=alt:1 1 : 2C830AD9 0000004F 192.168.2.17 6398..a=fmtp:101
>> 0-15..a=rtpmap:
>>   101 telephone-event/8000..a=sendrecv..
>>
>>
>> 192.168.2.190:5060 -> 192.168.2.17:6182
>>   SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport=6182..From:
>> 7001 <sip:7001 at 192.168.2.190 <sip%3A7001 at 192.168.2.190>>;tag=6c557c1e..To:
>> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;tag=H0ctQv7KNgU2j..Call-ID:
>> d32ffe546570a77e..C
>>   Seq: 2 INVITE..Contact: <sip:7050 at 192.168.2.190:5060;transport=udp>..User-Agent:
>> NOVANET..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
>> MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Support
>>   ed: timer, precondition, path, replaces..Allow-Events: talk, hold,
>> presence, dialog, line-seize, call-info, sla, include-session-description,
>> presence.winfo, message-summary, refer..Session-Expires:
>> 1800;refresher=uas..Min-SE: 120..
>>   Content-Type: application/sdp..Content-Disposition:
>> session..Content-Length: 249..Remote-Party-ID: "7050" <
>> sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH
>> 1293161640 1293161641 IN IP4 192.168.2.190..
>>   s=FreeSWITCH..c=IN IP4 192.168.2.190..t=0 0..m=audio 24852 RTP/AVP 0
>> 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
>> 0-16..a=silenceSupp:off - - - -..a=ptime:20..
>>
>>
>> Any thing you can think how it can happen?
>>
>>
>> Regards
>> Sam
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From Avi at aMarcus.com  Sun Dec 26 20:47:35 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Sun, 26 Dec 2010 19:47:35 +0200
Subject: [Freeswitch-users] condition weirdness
In-Reply-To: <A983598584834A44B3CD83DF95F07B4F@e1705>
References: <A983598584834A44B3CD83DF95F07B4F@e1705>
Message-ID: <AANLkTinxvUsfcjfJ74iweWL-eJqmd+O36R9WO4Y4TV=J@mail.gmail.com>

You probably meant (1|51|758|597|868|598|58) rather than the [character
class].
So try <condition field="${sip_to_uri}"
expression="^((1|51|758|597|868|598|58)\d{10,15})@$${domain}$">
-Avi Marcus

On Sun, Dec 26, 2010 at 4:41 PM, Madovsky <infos at madovsky.org> wrote:

>  If i call the number 000123456789 at domain
>
> and in my dialplan I have something like
>
> ...
> <condition field="${sip_to_uri}"
> expression="^([1|51|758|597|868|598|58]\d{10,15})@$${domain}$">
> ...
>
> in the log I can see condition passed... I'm confused
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Nabble at slickdeals.endjunk.com  Sun Dec 26 20:54:51 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 26 Dec 2010 09:54:51 -0800 (PST)
Subject: [Freeswitch-users] libg722_1: Cross
	Compilation	of	src/make_dct4_tables.c
In-Reply-To: <4D177988.5000503@coppice.org>
References: <1293326620424-5867008.post@n2.nabble.com>
	<4D16D5FB.40001@coppice.org>
	<1293381294171-5867851.post@n2.nabble.com>
	<4D177988.5000503@coppice.org>
Message-ID: <1293386091895-5867940.post@n2.nabble.com>



Steve Underwood wrote:
> Although that dct4.h full of floating point values will be generated in 
> a cross build for an ARM, building for an ARM should automatically set 
> fixed point mode. Therefore that dct4.h file will not actually be used. 
> Do you have any reason to believe the codec which has been built is bad? 
> I think others have cross-built this code successfully for ARMs.

Thank Steve and am glad to hear this. Now that you explained above, I have
no concerns any more unless I run into such a problem (if there is).

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/libg722-1-Cross-Compilation-of-src-make-dct4-tables-c-tp5867008p5867940.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Sun Dec 26 22:02:18 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 26 Dec 2010 14:02:18 -0500
Subject: [Freeswitch-users] condition weirdness
References: <A983598584834A44B3CD83DF95F07B4F@e1705>
	<AANLkTinxvUsfcjfJ74iweWL-eJqmd+O36R9WO4Y4TV=J@mail.gmail.com>
Message-ID: <62399DC6F5EC4EDA8E4302A024CC206E@e1705>

oops, ok thanks
  ----- Original Message ----- 
  From: Avi Marcus 
  To: FreeSWITCH Users Help 
  Sent: Sunday, December 26, 2010 12:47 PM
  Subject: Re: [Freeswitch-users] condition weirdness


  You probably meant (1|51|758|597|868|598|58) rather than the [character class].
  So try <condition field="${sip_to_uri}" expression="^((1|51|758|597|868|598|58)\d{10,15})@$${domain}$">

  -Avi Marcus


  On Sun, Dec 26, 2010 at 4:41 PM, Madovsky <infos at madovsky.org> wrote:

    If i call the number 000123456789 at domain

    and in my dialplan I have something like

    ...
    <condition field="${sip_to_uri}" expression="^([1|51|758|597|868|598|58]\d{10,15})@$${domain}$">
    ...

    in the log I can see condition passed... I'm confused

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






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From infos at madovsky.org  Sun Dec 26 22:34:32 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 26 Dec 2010 14:34:32 -0500
Subject: [Freeswitch-users] operator-extension in voicemail
Message-ID: <0BF2E45D256C4981A808669BB3760CE9@e1705>

is operator-extension free to configure
any extension transfer ?

I'd like to us for example "9" while the voicemail is running
to redirect in a conference extension

Thanks

Franck
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From gmaruzz at gmail.com  Sun Dec 26 22:56:54 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 26 Dec 2010 20:56:54 +0100
Subject: [Freeswitch-users] GSM-Modems and support
In-Reply-To: <20101226165202.GJ6539@michelle1>
References: <20101226165202.GJ6539@michelle1>
Message-ID: <AANLkTi=wWURHxbY+tDKtj-xP9FG67a33dFsqO+fsdYv6@mail.gmail.com>

In the wiki, please look for gsmopen
please note is pre-beta (but works very nice for me)
example device: www.mobigater.com
(beware, their model with 4 sims does give problems. You need to use a
multi-tt hub to use more than 3 usb audio device)
if you are seriously interested in this kind of things, and are
able/willing/have time to collaborate, I would like to.
-giovanni

On 12/26/10, Michelle Konzack <linux4michelle at tamay-dogan.net> wrote:
> Hello *,
>
> sometimes ago I already had asked but the answer was not very  satisfait
> at the time so I ask now again:
>
> I want to build a local GSM-to-SIP-Gateway with 4 SIMs.  Does FreeSwitch
> now support USB 1.1 GSM-Modems and if yes, which are recommened.
>
> I realy mean inexpensive non UMTS/HSPA ones.
>
> Thanks, Greetings and nice Day/Evening
>     Michelle Konzack
>
> --
> ##################### Debian GNU/Linux Consultant ######################
>    Development of Intranet and Embedded Systems with Debian GNU/Linux
>
> itsystems at tdnet France EURL       itsystems at tdnet UG (limited liability)
> Owner Michelle Konzack            Owner Michelle Konzack
>
> Apt. 917 (homeoffice)
> 50, rue de Soultz                 Kinzigstra?e 17
> 67100 Strasbourg/France           77694 Kehl/Germany
> Tel: +33-6-61925193 mobil         Tel: +49-177-9351947 mobil
> Tel: +33-9-52705884 fix
>
> <http://www.itsystems.tamay-dogan.net/>  <http://www.flexray4linux.org/>
> <http://www.debian.tamay-dogan.net/>         <http://www.can4linux.org/>
>
> Jabber linux4michelle at jabber.ccc.de
> ICQ    #328449886
>
> Linux-User #280138 with the Linux Counter, http://counter.li.org/
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From xduvox at gmail.com  Sun Dec 26 23:37:45 2010
From: xduvox at gmail.com (Octavio Duarte)
Date: Sun, 26 Dec 2010 14:37:45 -0600
Subject: [Freeswitch-users] how to know gateway used by distributor
Message-ID: <AANLkTimtavZ4HJ-J7TbW4fNp66CZCwTDAkz8VTBWvYzb@mail.gmail.com>

Hello everybody

I have this problem , when i use this

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

1   <action application="bridge" data="sofia/internal/${
distributor(gwusers)}%${domain_name}^$1"  loop="4"/>
or
2   <action application="bridge"
data="sofia/gateway/${distributor(gwproviders)}/$1"
loop="10"/>

both work perfectly

but i don't know for with gateway call went out

i have tried with ${bridge_channel} the result for the first example is

sofia/internal/sip:1002 at 1xx.142.1x7.xxx  it is enough for me!

but for the second example this is the result

sofia/internal/5533186003 and i can't know the gateway that was used!

i use this <param name="master-file-only" value="true"/>   in cdr_csv.conf.xml

so my question is:  how can i know the gateway was used by distributor?

thanks in advance for your help
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From lloyd.aloysius at gmail.com  Sun Dec 26 23:37:45 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 26 Dec 2010 15:37:45 -0500
Subject: [Freeswitch-users] group_confirm - Question
Message-ID: <AANLkTikPXY7073P3=3Gto+ArfBY41TUxkikKRAkGe13A@mail.gmail.com>

Hi All,

Dial plan below use group_confrim,  work for both internal and external
(PSTN) Calls without any issues

<action application="bridge" data="{ignore_early_media=true,*
group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1
*}user/${dialed_extension}@
${domain_name},sofia/gateway/voipms/${user_data(${dialed_extension}@${domain_name}
var followme_number)}"/>


-----

But I would like to use the group_confirm feature for external PSTN (Calls
only). The following Dial plan is not working and the console log below.

<action application="bridge"
data="{ignore_early_media=true}user/${dialed_extension}@${domain_name},{*
group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1
*}sofia/gateway/voipms/${user_data(${dialed_extension}@${domain_name} var
followme_number)}"/>


*Console Log*

2010-12-26 15:23:21.796308 [ERR] switch_core_session.c:380 Could not locate
channel type {group_confirm_file=
2010-12-26 15:23:21.796308 [ERR] switch_ivr_originate.c:2614 Cannot create
outgoing channel of type [{group_confirm_file=] cause:
[CHAN_NOT_IMPLEMENTED]
2010-12-26 15:23:21.796308 [ERR] switch_core_session.c:380 Could not locate
channel type group_confirm_key=1}sofia
2010-12-26 15:23:21.796308 [ERR] switch_ivr_originate.c:2614 Cannot create
outgoing channel of type [group_confirm_key=1}sofia] cause:
[CHAN_NOT_IMPLEMENTED]

--------

Any help is appreciated.

Thanks
Lloyd
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From Avi at aMarcus.com  Mon Dec 27 00:29:52 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Sun, 26 Dec 2010 23:29:52 +0200
Subject: [Freeswitch-users] group_confirm - Question
In-Reply-To: <AANLkTikPXY7073P3=3Gto+ArfBY41TUxkikKRAkGe13A@mail.gmail.com>
References: <AANLkTikPXY7073P3=3Gto+ArfBY41TUxkikKRAkGe13A@mail.gmail.com>
Message-ID: <AANLkTi=f+MsRTD=du8YUHmVp=ANrDRTzOmD_4XCx48az@mail.gmail.com>

I don't think you are supposed to be using {} variables in the middle of
your bridge string. It's only used at the beginning for setting variables
for EVERY bridge string, else, you should enclose it in []. I think. Give it
a try.
-Avi Marcus

On Sun, Dec 26, 2010 at 10:37 PM, Aloysius Lloyd
<lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> Dial plan below use group_confrim,  work for both internal and external
> (PSTN) Calls without any issues
>
> <action application="bridge" data="{ignore_early_media=true,*
> group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1
> *}user/${dialed_extension}@
> ${domain_name},sofia/gateway/voipms/${user_data(${dialed_extension}@${domain_name}
> var followme_number)}"/>
>
>
> -----
>
> But I would like to use the group_confirm feature for external PSTN (Calls
> only). The following Dial plan is not working and the console log below.
>
> <action application="bridge"
> data="{ignore_early_media=true}user/${dialed_extension}@${domain_name},{*
> group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1
> *}sofia/gateway/voipms/${user_data(${dialed_extension}@${domain_name} var
> followme_number)}"/>
>
>
> *Console Log*
>
> 2010-12-26 15:23:21.796308 [ERR] switch_core_session.c:380 Could not locate
> channel type {group_confirm_file=
> 2010-12-26 15:23:21.796308 [ERR] switch_ivr_originate.c:2614 Cannot create
> outgoing channel of type [{group_confirm_file=] cause:
> [CHAN_NOT_IMPLEMENTED]
> 2010-12-26 15:23:21.796308 [ERR] switch_core_session.c:380 Could not locate
> channel type group_confirm_key=1}sofia
> 2010-12-26 15:23:21.796308 [ERR] switch_ivr_originate.c:2614 Cannot create
> outgoing channel of type [group_confirm_key=1}sofia] cause:
> [CHAN_NOT_IMPLEMENTED]
>
> --------
>
> Any help is appreciated.
>
> Thanks
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Mon Dec 27 02:38:39 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 26 Dec 2010 18:38:39 -0500
Subject: [Freeswitch-users] how to know gateway used by distributor
References: <AANLkTimtavZ4HJ-J7TbW4fNp66CZCwTDAkz8VTBWvYzb@mail.gmail.com>
Message-ID: <DB4B6706CB844BDC93BFC8A9C32A34DB@e1705>

try to make a condition cond() in your dialplan
  ----- Original Message ----- 
  From: Octavio Duarte 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, December 26, 2010 3:37 PM
  Subject: [Freeswitch-users] how to know gateway used by distributor


  Hello everybody

  I have this problem , when i use this

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

  1   <action application="bridge" data="sofia/internal/${distributor(gwusers)}%${domain_name}^$1"  loop="4"/>
  or
  2   <action application="bridge" data="sofia/gateway/${distributor(gwproviders)}/$1"  loop="10"/>

  both work perfectly 

  but i don't know for with gateway call went out

  i have tried with ${bridge_channel} the result for the first example is

sofia/internal/sip:1002 at 1xx.142.1x7.xxx  it is enough for me! but for the second example this is the result 
sofia/internal/5533186003 and i can't know the gateway that was used!i use this <param name="master-file-only" value="true"/>   in cdr_csv.conf.xmlso my question is:  how can i know the gateway was used by distributor?
thanks in advance for your help



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


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From Avi at aMarcus.com  Mon Dec 27 02:49:28 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Mon, 27 Dec 2010 01:49:28 +0200
Subject: [Freeswitch-users] Time Conditions & Break - Weird Behavior? &
	Anti-actions Usefulness?
Message-ID: <AANLkTinQEpW_O=zMRaeU77K_ng43f27+vv93m9wmc=DG@mail.gmail.com>

I have two xml config files with time of day routing, and I couldn't imagine
why when it reached the first one, it just skipped the rest of the xml
files, as such:

Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 parsing [default->YLE1]
continue=false
Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Regex (*PASS*) [*YLE1*]
destination_number(t1105) =~ /^t1105$/ break=on-false
Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Date/Time Match (*FAIL*)
[*YLE1*] break=on-false
2010-12-27 01:36:09.410521 [DEBUG] switch_core_state_machine.c:119
(sofia/internal/sip:1000 at 109.67.185.37:5072) State Change CS_ROUTING ->
CS_EXECUTE

But when I just edit the first and set continue=true, it suddenly now
properly moves on. But it never matched the entire extension's condition
tags, so why did it think it should not continue?
(xml files here: http://pastebin.freeswitch.org/14885)

Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 parsing [default->YLE1]
continue=true
Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Regex (PASS) [YLE1]
destination_number(t1105) =~ /^t1105$/ break=on-false
Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Date/Time Match (FAIL)
[YLE1] break=on-false
Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 parsing [default->YLE2]
continue=false
Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Regex (PASS) [YLE2]
destination_number(t1105) =~ /^t1105$/ break=on-false
Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Date/Time Match (PASS)
[YLE2] break=on-false
Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Regex (FAIL) [YLE2]
time-of-day() =~ /08:00-22:00/ break=on-false
Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 ANTI-Action
voicemail(default ${domain} 1105)
2010-12-27 01:37:24.754671 [DEBUG] switch_core_state_machine.c:119
(sofia/internal/sip:1000 at 109.67.185.37:5072) State Change CS_ROUTING ->
CS_EXECUTE

And - it seems the anti-action is triggered no matter how "off" we are, is
there a way to have a less agressive anti action? E.g. it only activates if
the first condition is true? It's stopping the rest of my dial plan from
executing.
When would one ever want an anti-action other than as his last <extension>
in which case it could just be a new last extension with no condition tag. I
don't understand why you would ever want them.

Understanding would be appreciated :)
-Avi Marcus
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From Avi at aMarcus.com  Mon Dec 27 03:29:02 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Mon, 27 Dec 2010 02:29:02 +0200
Subject: [Freeswitch-users] Time-of-day broken?
Message-ID: <AANLkTi=mad51Fz4z4m2wHd7JODev7xp=eHKGKoNR9uK0@mail.gmail.com>

This is my first day using time-based routing, and I'm not having fun.

Regex (*FAIL*) [YLE3] time-of-day() =~ /00:00-07:59:59/ break=on-false
freeswitch at internal> strftime 2010-12-27 02:14:41

This is on the latest GIT.
-Avi
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From linux4michelle at tamay-dogan.net  Mon Dec 27 03:43:00 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Mon, 27 Dec 2010 01:43:00 +0100
Subject: [Freeswitch-users] GSM-Modems and support
In-Reply-To: <AANLkTi=wWURHxbY+tDKtj-xP9FG67a33dFsqO+fsdYv6@mail.gmail.com>
References: <20101226165202.GJ6539@michelle1>
	<AANLkTi=wWURHxbY+tDKtj-xP9FG67a33dFsqO+fsdYv6@mail.gmail.com>
Message-ID: <20101227004300.GK6539@michelle1>

Hello Giovanni Maruzzelli,

Am 2010-12-26 20:56:54, hacktest Du folgendes herunter:
> In the wiki, please look for gsmopen
> please note is pre-beta (but works very nice for me)
> example device: www.mobigater.com

Cool...

Do you now the price for the MobiGater?
On the Website is nothing written...

Hmmm, maybe I should call them in  Bulgaria  with  my  Ortel/e-Plus SIM,
because it cost only 0.079 Euro/min it  is  much  cheaper  as  my  fixed
Telephone line in France.

> (beware, their model with 4 sims does give problems. You need to use a
> multi-tt hub to use more than 3 usb audio device)
  ^^^^^^^^
A what please?

It would be already perfect, if I could connect my first to  cards  from
Ortel and O2.

> if you are seriously interested in this kind of things, and are
> able/willing/have time to collaborate, I would like to.

If you go to the website <http://www.ortelmobile.de/> you  can  see  the
prices and because my Dual-SIM cellphone Made-In-China is not  the  best
quality, it is time to get a realy good GSM-Transmiter which  permit  me
to use an external antenna (My Mobil-Home/Office are made of Inox).

Which help do you need?

> -giovanni

Thanks, Greetings and nice Day/Evening
    Michelle Konzack

-- 
##################### Debian GNU/Linux Consultant ######################
   Development of Intranet and Embedded Systems with Debian GNU/Linux

itsystems at tdnet France EURL       itsystems at tdnet UG (limited liability)
Owner Michelle Konzack            Owner Michelle Konzack

Apt. 917 (homeoffice)
50, rue de Soultz                 Kinzigstra?e 17
67100 Strasbourg/France           77694 Kehl/Germany
Tel: +33-6-61925193 mobil         Tel: +49-177-9351947 mobil
Tel: +33-9-52705884 fix

<http://www.itsystems.tamay-dogan.net/>  <http://www.flexray4linux.org/>
<http://www.debian.tamay-dogan.net/>         <http://www.can4linux.org/>

Jabber linux4michelle at jabber.ccc.de
ICQ    #328449886

Linux-User #280138 with the Linux Counter, http://counter.li.org/
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From xduvox at gmail.com  Mon Dec 27 07:39:47 2010
From: xduvox at gmail.com (xduvox)
Date: Sun, 26 Dec 2010 20:39:47 -0800 (PST)
Subject: [Freeswitch-users] how to know gateway used by distributor
In-Reply-To: <DB4B6706CB844BDC93BFC8A9C32A34DB@e1705>
References: <AANLkTimtavZ4HJ-J7TbW4fNp66CZCwTDAkz8VTBWvYzb@mail.gmail.com>
	<DB4B6706CB844BDC93BFC8A9C32A34DB@e1705>
Message-ID: <1293424787581-5868787.post@n2.nabble.com>


thanks for your reply

i have solved the problem by using b-leg cdr and exporting a-leg variable so
know i have the cdrs as i wanted to

thanks
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/how-to-know-gateway-used-by-distributor-tp5868195p5868787.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From lloyd.aloysius at sunteltech.ca  Mon Dec 27 07:57:44 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Sun, 26 Dec 2010 23:57:44 -0500
Subject: [Freeswitch-users] group_confirm - Question
In-Reply-To: <AANLkTi=f+MsRTD=du8YUHmVp=ANrDRTzOmD_4XCx48az@mail.gmail.com>
References: <AANLkTikPXY7073P3=3Gto+ArfBY41TUxkikKRAkGe13A@mail.gmail.com>
	<AANLkTi=f+MsRTD=du8YUHmVp=ANrDRTzOmD_4XCx48az@mail.gmail.com>
Message-ID: <AANLkTi=tDcLWG6V6fQHN1d95GKvOaB-NOR3BQFB38PeP@mail.gmail.com>

I try the [] now no errors. But

-  group_confirm_file not play
- call directly connected

basically group_confirm not working.

Thanks
Lloyd

On Sun, Dec 26, 2010 at 4:29 PM, Avi Marcus <Avi at amarcus.com> wrote:

> I don't think you are supposed to be using {} variables in the middle of
> your bridge string. It's only used at the beginning for setting variables
> for EVERY bridge string, else, you should enclose it in []. I think. Give it
> a try.
> -Avi Marcus
>
> On Sun, Dec 26, 2010 at 10:37 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com
> > wrote:
>
>> Hi All,
>>
>> Dial plan below use group_confrim,  work for both internal and external
>> (PSTN) Calls without any issues
>>
>> <action application="bridge" data="{ignore_early_media=true,*
>> group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1
>> *}user/${dialed_extension}@
>> ${domain_name},sofia/gateway/voipms/${user_data(${dialed_extension}@${domain_name}
>> var followme_number)}"/>
>>
>>
>> -----
>>
>> But I would like to use the group_confirm feature for external PSTN (Calls
>> only). The following Dial plan is not working and the console log below.
>>
>> <action application="bridge"
>> data="{ignore_early_media=true}user/${dialed_extension}@${domain_name},{*
>> group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1
>> *}sofia/gateway/voipms/${user_data(${dialed_extension}@${domain_name} var
>> followme_number)}"/>
>>
>>
>> *Console Log*
>>
>> 2010-12-26 15:23:21.796308 [ERR] switch_core_session.c:380 Could not
>> locate channel type {group_confirm_file=
>> 2010-12-26 15:23:21.796308 [ERR] switch_ivr_originate.c:2614 Cannot create
>> outgoing channel of type [{group_confirm_file=] cause:
>> [CHAN_NOT_IMPLEMENTED]
>> 2010-12-26 15:23:21.796308 [ERR] switch_core_session.c:380 Could not
>> locate channel type group_confirm_key=1}sofia
>> 2010-12-26 15:23:21.796308 [ERR] switch_ivr_originate.c:2614 Cannot create
>> outgoing channel of type [group_confirm_key=1}sofia] cause:
>> [CHAN_NOT_IMPLEMENTED]
>>
>> --------
>>
>> Any help is appreciated.
>>
>> Thanks
>> Lloyd
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Avi at aMarcus.com  Mon Dec 27 09:04:12 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Mon, 27 Dec 2010 08:04:12 +0200
Subject: [Freeswitch-users] group_confirm - Question
In-Reply-To: <AANLkTi=tDcLWG6V6fQHN1d95GKvOaB-NOR3BQFB38PeP@mail.gmail.com>
References: <AANLkTikPXY7073P3=3Gto+ArfBY41TUxkikKRAkGe13A@mail.gmail.com>
	<AANLkTi=f+MsRTD=du8YUHmVp=ANrDRTzOmD_4XCx48az@mail.gmail.com>
	<AANLkTi=tDcLWG6V6fQHN1d95GKvOaB-NOR3BQFB38PeP@mail.gmail.com>
Message-ID: <AANLkTikF9FqqFUdi2htpNKcbNNYC8rS3ZRF-JeG0QVrZ@mail.gmail.com>

Can you paste the bridge for us again just to confirm the change, and the
new log?
And to clarify, you mean it's not working for the PSTN recipients, correct?
-Avi

On Mon, Dec 27, 2010 at 6:57 AM, Aloysius Lloyd <
lloyd.aloysius at sunteltech.ca> wrote:

> I try the [] now no errors. But
>
> -  group_confirm_file not play
> - call directly connected
>
> basically group_confirm not working.
>
> Thanks
> Lloyd
>
> On Sun, Dec 26, 2010 at 4:29 PM, Avi Marcus <Avi at amarcus.com> wrote:
>
>> I don't think you are supposed to be using {} variables in the middle of
>> your bridge string. It's only used at the beginning for setting variables
>> for EVERY bridge string, else, you should enclose it in []. I think. Give it
>> a try.
>> -Avi Marcus
>>
>> On Sun, Dec 26, 2010 at 10:37 PM, Aloysius Lloyd <
>> lloyd.aloysius at gmail.com> wrote:
>>
>>> Hi All,
>>>
>>> Dial plan below use group_confrim,  work for both internal and external
>>> (PSTN) Calls without any issues
>>>
>>> <action application="bridge" data="{ignore_early_media=true,*
>>> group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1
>>> *}user/${dialed_extension}@
>>> ${domain_name},sofia/gateway/voipms/${user_data(${dialed_extension}@${domain_name}
>>> var followme_number)}"/>
>>>
>>>
>>> -----
>>>
>>> But I would like to use the group_confirm feature for external PSTN
>>> (Calls only). The following Dial plan is not working and the console log
>>> below.
>>>
>>> <action application="bridge"
>>> data="{ignore_early_media=true}user/${dialed_extension}@${domain_name},{
>>> *
>>> group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1
>>> *}sofia/gateway/voipms/${user_data(${dialed_extension}@${domain_name}
>>> var followme_number)}"/>
>>>
>>>
>>> *Console Log*
>>>
>>> 2010-12-26 15:23:21.796308 [ERR] switch_core_session.c:380 Could not
>>> locate channel type {group_confirm_file=
>>> 2010-12-26 15:23:21.796308 [ERR] switch_ivr_originate.c:2614 Cannot
>>> create outgoing channel of type [{group_confirm_file=] cause:
>>> [CHAN_NOT_IMPLEMENTED]
>>> 2010-12-26 15:23:21.796308 [ERR] switch_core_session.c:380 Could not
>>> locate channel type group_confirm_key=1}sofia
>>> 2010-12-26 15:23:21.796308 [ERR] switch_ivr_originate.c:2614 Cannot
>>> create outgoing channel of type [group_confirm_key=1}sofia] cause:
>>> [CHAN_NOT_IMPLEMENTED]
>>>
>>> --------
>>>
>>> Any help is appreciated.
>>>
>>> Thanks
>>> Lloyd
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Mon Dec 27 10:56:23 2010
From: u2nsam at gmail.com (Sam)
Date: Mon, 27 Dec 2010 13:26:23 +0530
Subject: [Freeswitch-users] DTMF missing
In-Reply-To: <AANLkTin14RN-WxkYzqkQPRZctYt09xkWguAoCr8eeUWe@mail.gmail.com>
References: <AANLkTik=NLa+injL5hF=VsymwoE9wd1Wb2joXJDhvfFZ@mail.gmail.com>
	<AANLkTincAApO4JDwzt-5oX0iD2VozFNVCtembOumL4iS@mail.gmail.com>
	<AANLkTin14RN-WxkYzqkQPRZctYt09xkWguAoCr8eeUWe@mail.gmail.com>
Message-ID: <AANLkTik3N+F6Li6OL9EmyDH34eowxR+ttBFZx=1e83Eg@mail.gmail.com>

Hi ,

I have now upgraded to ver (git-4e95227 2010-12-26 09-09-14 -0600) , now the
dtmfs in IVR are working !


When i used bind meta application then its not executing, any comments for
me ?

            <action application="set" data="export_vars=#,*"/>
            <action application="export" data="#=true"/>
            <action application="set" data="bind_meta_key=#"/>
            <action application="bind_meta_app" data="1 ab s
execute_extension::dx XML features"/>




Regards
Sam



On Sun, Dec 26, 2010 at 11:16 PM, afshin afzali <a.afzali2003 at gmail.com>wrote:

> Sam,
>
> Yes, The issue has resolved in last git :)
> -- afshin
>
> On Sun, Dec 26, 2010 at 7:33 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Hello
>>
>> Just like Afshin, I am also missing DTMF digits ,using FreeSWITCH Version
>> 1.0.head (git-34a0ca5 2010-12-22 20-38-57 -0600) ;
>> should i upgrade to latest ?
>>
>> Regards
>> Sam
>>
>>
>>
>> On Fri, Dec 24, 2010 at 4:10 PM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Hi,,
>>>
>>>
>>> I have installed the latest ver of freeswitch and i have configured the
>>> conference.
>>>
>>> now when i punch in the digits for password , i could see that the DTMF
>>> digits are missed on fs_cli.
>>>
>>> it only happens when i dial it from polycom or cisco phones.
>>>
>>> I have tried with and without these values below:-
>>>
>>> <param name="dtmf-type" value="rfc2833"/>
>>>     <param name="pass-rfc2833" value="true"/>
>>>
>>>
>>>
>>> traces fetched:
>>> 192.168.2.49:5060 -> 192.168.2.190:5060
>>>   INVITE sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190> SIP/2.0..Via:
>>> SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From: "7028" <
>>> sip:7028 at 192.168.2.190 <sip%3A7028 at 192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To:
>>> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>..Call-ID:
>>> 0017592a-eb33001a-
>>>   63da3294-1a7bfdfa at 192.168.2.49..Max-Forwards: 70..Date: Fri, 24 Dec
>>> 2010 10:13:52 GMT..CSeq: 102 INVITE..User-Agent: Cisco-CP7940G/8.0..Contact:
>>> <sip:7028 at 192.168.2.49:5060;transport=udp>..Proxy-Authorization: Digest
>>> username="7028"
>>>   ,realm="192.168.2.190",uri="sip:7050 at 192.168.2.190<sip%3A7050 at 192.168.2.190>",response="a668f5c480285b35e7ff6bcd446879f0",nonce="d2c540f2-8487-4d87-bdab-871585253eb8",cnonce="0a6c4176",qop=auth,nc=00000001,algorithm=MD5..Expires:
>>> 180..Accept: application/sdp
>>>   ..Allow:
>>> ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported:
>>> replaces,join,norefersub..Content-Length: 220..Content-Type:
>>> application/sdp..Content-Disposition:
>>> session;handling=optional....v=0..o=Cisco-SIPUA 16102
>>>    0 IN IP4 192.168.2.49..s=SIP Call..t=0 0..m=audio 17298 RTP/AVP 0 8
>>> 18..c=IN IP4 192.168.2.49..a=rtpmap:0 PCMU/8000..a=rtpmap:8
>>> PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=sendrecv..
>>>
>>>
>>> 192.168.2.190:5060 -> 192.168.2.49:5060
>>>   SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From:
>>> "7028" <sip:7028 at 192.168.2.190 <sip%3A7028 at 192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To:
>>> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;tag=2XXUZpgr1rvgc..Call-ID:
>>> 0017592a-eb33001a-63da3
>>>   294-1a7bfdfa at 192.168.2.49..CSeq: 102 INVITE..Contact:
>>> <sip:7050 at 192.168.2.190:5060;transport=udp>..User-Agent:
>>> NOVANET..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
>>> MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIF
>>>   Y, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path,
>>> replaces..Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>> sla, include-session-description, presence.winfo, message-summary,
>>> refer..Session-Expires: 180
>>>   0;refresher=uas..Min-SE: 120..Content-Type:
>>> application/sdp..Content-Disposition: session..Content-Length:
>>> 249..Remote-Party-ID: "7050" <sip:7050 at 192.168.2.190<sip%3A7050 at 192.168.2.190>>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH
>>> 1293163579 129
>>>   3163580 IN IP4 192.168.2.190..s=FreeSWITCH..c=IN IP4 192.168.2.190..t=0
>>> 0..m=audio 22050 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101
>>> telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - -
>>> -..a=ptime:20..
>>>
>>> But the dtmf are not missed when punched on eyebeam softphone.
>>>
>>> And all the phones have RFC 2833.
>>>
>>> traces fetched for softphone:-
>>>
>>> 192.168.2.17:6182 -> 192.168.2.190:5060
>>>   INVITE sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190> SIP/2.0..To:
>>> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>..From: 7001<
>>> sip:7001 at 192.168.2.190 <sip%3A7001 at 192.168.2.190>>;tag=6c557c1e..Via:
>>> SIP/2.0/UDP 192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport..Call-ID:
>>> d32ffe546570a77e..CS
>>>   eq: 2 INVITE..Contact: <sip:7001 at 192.168.2.17:6182>..Max-Forwards:
>>> 70..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>>> SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Authorization: Digest
>>> username="7001",rea
>>>   lm="192.168.2.190",nonce="d0b4db6e-bd76-447f-9076-2e6b7809cb54",uri="
>>> sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>",response="45fb9eb4f0e0e4fffd87a22769a007ba",cnonce="1c27f3687059b16d",nc=00000001,qop=auth,algorithm=MD5..User-Agent:
>>> eyeBeam release 3007n
>>>    stamp 17816..Content-Length: 233....v=0..o=- 27833664 27833670 IN IP4
>>> 192.168.2.17..s=eyeBeam..c=IN IP4 192.168.2.17..t=0 0..m=audio 6398 RTP/AVP
>>> 0 18 101..a=alt:1 1 : 2C830AD9 0000004F 192.168.2.17 6398..a=fmtp:101
>>> 0-15..a=rtpmap:
>>>   101 telephone-event/8000..a=sendrecv..
>>>
>>>
>>> 192.168.2.190:5060 -> 192.168.2.17:6182
>>>   SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport=6182..From:
>>> 7001 <sip:7001 at 192.168.2.190 <sip%3A7001 at 192.168.2.190>>;tag=6c557c1e..To:
>>> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;tag=H0ctQv7KNgU2j..Call-ID:
>>> d32ffe546570a77e..C
>>>   Seq: 2 INVITE..Contact: <sip:7050 at 192.168.2.190:5060;transport=udp>..User-Agent:
>>> NOVANET..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
>>> MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Support
>>>   ed: timer, precondition, path, replaces..Allow-Events: talk, hold,
>>> presence, dialog, line-seize, call-info, sla, include-session-description,
>>> presence.winfo, message-summary, refer..Session-Expires:
>>> 1800;refresher=uas..Min-SE: 120..
>>>   Content-Type: application/sdp..Content-Disposition:
>>> session..Content-Length: 249..Remote-Party-ID: "7050" <
>>> sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH
>>> 1293161640 1293161641 IN IP4 192.168.2.190..
>>>   s=FreeSWITCH..c=IN IP4 192.168.2.190..t=0 0..m=audio 24852 RTP/AVP 0
>>> 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
>>> 0-16..a=silenceSupp:off - - - -..a=ptime:20..
>>>
>>>
>>> Any thing you can think how it can happen?
>>>
>>>
>>> Regards
>>> Sam
>>>
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rajkumar.kmry at gmail.com  Mon Dec 27 12:42:29 2010
From: rajkumar.kmry at gmail.com (rajkumar)
Date: Mon, 27 Dec 2010 01:42:29 -0800 (PST)
Subject: [Freeswitch-users] mod_callcenter features
Message-ID: <1293442949323-5869082.post@n2.nabble.com>


Hi,

  I am developing an application with mod_callcenter. I need to know the
following about mod_callcenter.

  * Is it possible to add/update/delete the queue configurations dynamically
without using static xml configurations.
  * How can I control the call flow (for playback message and recording)
before and after bridging the call.

Thanks in advance

regards
rajkumar k
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-callcenter-features-tp5869082p5869082.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From peter.olsson at visionutveckling.se  Mon Dec 27 14:44:32 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 27 Dec 2010 12:44:32 +0100
Subject: [Freeswitch-users] mod_conference produces bad audio after commit
 75198fe4cb130f3d6d0d43f50882188879a59f6f
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57ECD48498@cooper>

It seems like commit 75198fe4cb130f3d6d0d43f50882188879a59f6f causes bad audio within the conference. When commenting out some of the new code it works fine again. Jira is not up for the moment, but I will add a ticket there when it's up and running again.

The changes below forces to use the timer (like it was before).

-if (mux_used < bytes * 2) {
+//if (mux_used < bytes * 2) {
                            use_timer = 1;
-}
+//}

/Peter Olsson

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From linux4michelle at tamay-dogan.net  Mon Dec 27 17:00:43 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Mon, 27 Dec 2010 15:00:43 +0100
Subject: [Freeswitch-users] GSM-Modems and support
In-Reply-To: <AANLkTi=wWURHxbY+tDKtj-xP9FG67a33dFsqO+fsdYv6@mail.gmail.com>
References: <20101226165202.GJ6539@michelle1>
	<AANLkTi=wWURHxbY+tDKtj-xP9FG67a33dFsqO+fsdYv6@mail.gmail.com>
Message-ID: <20101227140043.GB25187@michelle1>

Ciao Giovanni,

Am 2010-12-26 20:56:54, hacktest Du folgendes herunter:
> In the wiki, please look for gsmopen
> please note is pre-beta (but works very nice for me)
> example device: www.mobigater.com

I have contacted Eurodesign and they have respond very fast!

> (beware, their model with 4 sims does give problems. You need to use a
> multi-tt hub to use more than 3 usb audio device)

They have currently

MobiGater Pro                    65 Euro
MobiGater S3 (19", 1U, 3 SIMs)  320 Euro
MobiGater S6 (19", 2U, 6 SIMs)  620 Euro

and I am intrested in the Pro and S6... ;-)  but for now and testing the
Pro version only because I must be sure, my FreeSwitch  installation  is
working properly

> if you are seriously interested in this kind of things, and are
> able/willing/have time to collaborate, I would like to.

I think, I will have my "Pro" in two weeks.

> -giovanni

Thanks, Greetings and nice Day/Evening
    Michelle Konzack

-- 
##################### Debian GNU/Linux Consultant ######################
   Development of Intranet and Embedded Systems with Debian GNU/Linux

itsystems at tdnet France EURL       itsystems at tdnet UG (limited liability)
Owner Michelle Konzack            Owner Michelle Konzack

Apt. 917 (homeoffice)
50, rue de Soultz                 Kinzigstra?e 17
67100 Strasbourg/France           77694 Kehl/Germany
Tel: +33-6-61925193 mobil         Tel: +49-177-9351947 mobil
Tel: +33-9-52705884 fix

<http://www.itsystems.tamay-dogan.net/>  <http://www.flexray4linux.org/>
<http://www.debian.tamay-dogan.net/>         <http://www.can4linux.org/>

Jabber linux4michelle at jabber.ccc.de
ICQ    #328449886

Linux-User #280138 with the Linux Counter, http://counter.li.org/
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From gmaruzz at gmail.com  Mon Dec 27 17:48:32 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 27 Dec 2010 15:48:32 +0100
Subject: [Freeswitch-users] GSM-Modems and support
In-Reply-To: <20101227140043.GB25187@michelle1>
References: <20101226165202.GJ6539@michelle1>
	<AANLkTi=wWURHxbY+tDKtj-xP9FG67a33dFsqO+fsdYv6@mail.gmail.com>
	<20101227140043.GB25187@michelle1>
Message-ID: <AANLkTinwRb2fZ55K-MNRC54x7_ifCyaZd3NRsbb96P0=@mail.gmail.com>

Hi Michelle,

I'm rushing out now, but in short:

with the "pro" model you'll have no problem.

If you want to have more lines, have many pro connected to a multi-tt
hub (google for multi-tt usb hub, is a special kind of hub, btw is
cheap).

Talk to you soon, at least when you got your device and begin testing.

Ciao for now,

-giovanni


On Mon, Dec 27, 2010 at 3:00 PM, Michelle Konzack
<linux4michelle at tamay-dogan.net> wrote:
> Ciao Giovanni,
>
> Am 2010-12-26 20:56:54, hacktest Du folgendes herunter:
>> In the wiki, please look for gsmopen
>> please note is pre-beta (but works very nice for me)
>> example device: www.mobigater.com
>
> I have contacted Eurodesign and they have respond very fast!
>
>> (beware, their model with 4 sims does give problems. You need to use a
>> multi-tt hub to use more than 3 usb audio device)
>
> They have currently
>
> MobiGater Pro ? ? ? ? ? ? ? ? ? ?65 Euro
> MobiGater S3 (19", 1U, 3 SIMs) ?320 Euro
> MobiGater S6 (19", 2U, 6 SIMs) ?620 Euro
>
> and I am intrested in the Pro and S6... ;-) ?but for now and testing the
> Pro version only because I must be sure, my FreeSwitch ?installation ?is
> working properly
>
>> if you are seriously interested in this kind of things, and are
>> able/willing/have time to collaborate, I would like to.
>
> I think, I will have my "Pro" in two weeks.
>
>> -giovanni
>
> Thanks, Greetings and nice Day/Evening
> ? ?Michelle Konzack
>
> --
> ##################### Debian GNU/Linux Consultant ######################
> ? Development of Intranet and Embedded Systems with Debian GNU/Linux
>
> itsystems at tdnet France EURL ? ? ? itsystems at tdnet UG (limited liability)
> Owner Michelle Konzack ? ? ? ? ? ?Owner Michelle Konzack
>
> Apt. 917 (homeoffice)
> 50, rue de Soultz ? ? ? ? ? ? ? ? Kinzigstra?e 17
> 67100 Strasbourg/France ? ? ? ? ? 77694 Kehl/Germany
> Tel: +33-6-61925193 mobil ? ? ? ? Tel: +49-177-9351947 mobil
> Tel: +33-9-52705884 fix
>
> <http://www.itsystems.tamay-dogan.net/> ?<http://www.flexray4linux.org/>
> <http://www.debian.tamay-dogan.net/> ? ? ? ? <http://www.can4linux.org/>
>
> Jabber linux4michelle at jabber.ccc.de
> ICQ ? ?#328449886
>
> Linux-User #280138 with the Linux Counter, http://counter.li.org/
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From anthony.minessale at gmail.com  Mon Dec 27 18:37:03 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 27 Dec 2010 09:37:03 -0600
Subject: [Freeswitch-users] mod_conference produces bad audio after
	commit 75198fe4cb130f3d6d0d43f50882188879a59f6f
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57ECD48498@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C57ECD48498@cooper>
Message-ID: <AANLkTikEZTa2Viid=n7bnvo6jF40AOs2a91ih9mY+xWd@mail.gmail.com>

don't bother
commit 430dfdacabcb07feb4f4c1aa6ad5a57e97a3e250
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Mon Dec 27 09:27:21 2010 -0600



On Mon, Dec 27, 2010 at 5:44 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> It seems like commit 75198fe4cb130f3d6d0d43f50882188879a59f6f causes bad
> audio within the conference. When commenting out some of the new code it
> works fine again. Jira is not up for the moment, but I will add a ticket
> there when it?s up and running again.
>
>
>
> The changes below forces to use the timer (like it was before).
>
>
>
> -if (mux_used < bytes * 2) {
>
> +//if (mux_used < bytes * 2) {
>
> ??????????????????????????? use_timer = 1;
>
> -}
>
> +//}
>
>
>
> /Peter Olsson
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Mon Dec 27 18:47:26 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 27 Dec 2010 09:47:26 -0600
Subject: [Freeswitch-users] Time-of-day broken?
In-Reply-To: <AANLkTi=mad51Fz4z4m2wHd7JODev7xp=eHKGKoNR9uK0@mail.gmail.com>
References: <AANLkTi=mad51Fz4z4m2wHd7JODev7xp=eHKGKoNR9uK0@mail.gmail.com>
Message-ID: <AANLkTimUCYQM3GSg2xaJBLgQ0cK1QGmbUAh_bLKq7qz6@mail.gmail.com>

the () implies whatever you are testing evaluated to an empty string
so you are testing if NOTHING matches the regex.
What did the xml look like?



On Sun, Dec 26, 2010 at 6:29 PM, Avi Marcus <Avi at amarcus.com> wrote:
> This is my first day using time-based routing, and I'm not having fun.
> Regex (FAIL) [YLE3] time-of-day() =~ /00:00-07:59:59/ break=on-false
> freeswitch at internal> strftime?2010-12-27 02:14:41
> This is on the latest GIT.
> -Avi
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From peter.olsson at visionutveckling.se  Mon Dec 27 18:48:38 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 27 Dec 2010 16:48:38 +0100
Subject: [Freeswitch-users] mod_conference produces bad audio
	after	commit 75198fe4cb130f3d6d0d43f50882188879a59f6f
In-Reply-To: <AANLkTikEZTa2Viid=n7bnvo6jF40AOs2a91ih9mY+xWd@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C57ECD48498@cooper>
	<AANLkTikEZTa2Viid=n7bnvo6jF40AOs2a91ih9mY+xWd@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57ECD48508@cooper>

Thanks! :)

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale
Skickat: den 27 december 2010 16:37
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] mod_conference produces bad audio after commit 75198fe4cb130f3d6d0d43f50882188879a59f6f

don't bother
commit 430dfdacabcb07feb4f4c1aa6ad5a57e97a3e250
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Mon Dec 27 09:27:21 2010 -0600



On Mon, Dec 27, 2010 at 5:44 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> It seems like commit 75198fe4cb130f3d6d0d43f50882188879a59f6f causes bad
> audio within the conference. When commenting out some of the new code it
> works fine again. Jira is not up for the moment, but I will add a ticket
> there when it's up and running again.
>
>
>
> The changes below forces to use the timer (like it was before).
>
>
>
> -if (mux_used < bytes * 2) {
>
> +//if (mux_used < bytes * 2) {
>
> ??????????????????????????? use_timer = 1;
>
> -}
>
> +//}
>
>
>
> /Peter Olsson
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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

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

!DSPAM:4d18b46132765174615982!



From Avi at amarcus.com  Mon Dec 27 19:39:36 2010
From: Avi at amarcus.com (Avi Marcus)
Date: Mon, 27 Dec 2010 18:39:36 +0200
Subject: [Freeswitch-users] Time-of-day broken?
In-Reply-To: <AANLkTimUCYQM3GSg2xaJBLgQ0cK1QGmbUAh_bLKq7qz6@mail.gmail.com>
References: <AANLkTi=mad51Fz4z4m2wHd7JODev7xp=eHKGKoNR9uK0@mail.gmail.com>
	<AANLkTimUCYQM3GSg2xaJBLgQ0cK1QGmbUAh_bLKq7qz6@mail.gmail.com>
Message-ID: <AANLkTimBkn3xmWSo1iWcQDC14G7TPZuxDjsq7o_q_-2t@mail.gmail.com>

I deleted that file, argh. let's see:
I tried <condition time-of-day="00:00-21:59:59"  break="on-true">

and now I get
Date/Time Match (PASS) [YLE-tod] break=on-true

Not sure what changed.

But if it showed Regex (*FAIL*) [YLE3] time-of-day() =~ /00:00-07:59:59/
break=on-false
last time - it seemed to be matching it to something...?
Why the switch from regex to date/time? Did I have the () in my XML?

Oh well, if I can't reproduce it, then I guess there's nothing to do about
it.

-Avi

On Mon, Dec 27, 2010 at 5:47 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> the () implies whatever you are testing evaluated to an empty string
> so you are testing if NOTHING matches the regex.
> What did the xml look like?
>
>
>
> On Sun, Dec 26, 2010 at 6:29 PM, Avi Marcus <Avi at amarcus.com> wrote:
> > This is my first day using time-based routing, and I'm not having fun.
> > Regex (FAIL) [YLE3] time-of-day() =~ /00:00-07:59:59/ break=on-false
> > freeswitch at internal> strftime 2010-12-27 02:14:41
> > This is on the latest GIT.
> > -Avi
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From phone.bytes at gmail.com  Mon Dec 27 20:24:09 2010
From: phone.bytes at gmail.com (Phone)
Date: Mon, 27 Dec 2010 10:24:09 -0700
Subject: [Freeswitch-users] FreeTDM hangs on Partial PRI
Message-ID: <4D18CBB9.5080909@gmail.com>

We are using

FreeSWITCH Version 1.0.head (git-8825b6e 2010-11-28 17-15-39 -0500)

WANPIPE Release: 3.5.14

Hardware is Sangoma A101

Partial PRI D-chan=24 B-chans=16-23

We are having trouble with the Partial PRI (channel 16) getting stuck in 
a state where it cannot take calls.  This happens intermittently.  The 
system may run several days or even a week or two without having the 
problem.  It seems that the channel is getting stuck in some odd state 
when trying to hang up.

To resolve this, we unload the FreeTDM mod, restarting Wanrouter, and 
then reload the FreeTDM mod.  This clears the channels and allows the 
system to take calls normally again.

In testing we have also observed the following conditions:

1. Place a call in.  Cannot complete. Hang up.
2. Place a call in.  It completes and works correctly.
3. Place a call in.  Cannot complete.  Hang up.
4. Place a call in.  It completes and works correctly.

The above pattern can be repeated multiple times, with the same result.

Here is some log info:

2010-12-26 20:51:20.868083 [DEBUG] ftmod_sangoma_boost.c:1889 RX EVENT
(N): CALL_START:(80) [w1g16] CSid=0 Seq=390 Cn=[JONES J     ]
Cd=[7878030] Ci=[8015542086] Rdnis=[]
2010-12-26 20:51:20.868083 [DEBUG] ftmod_sangoma_boost.c:274 Channel 1:1
~ 1:16 is already in use in state TERMINATING
2010-12-26 20:51:20.868083 [ERR] ftmod_sangoma_boost.c:1008 s1c16:
rejecting incoming call in channel state TERMINATING
2010-12-26 20:51:20.868083 [DEBUG] sangoma_boost_client.c:252 TX EVENT
(N): CALL_START_NACK:(82) [s1c16] Rc=44 CSid=0 Seq=5
2010-12-26 20:51:28.563720 [DEBUG] ftmod_sangoma_boost.c:1889 RX EVENT
(N): CALL_START_NACK:(82) [s1c16] Rc=16 CSid=0 Seq=391
2010-12-26 20:51:28.563720 [WARNING] ftmod_sangoma_boost.c:811
[s1c1][1:16] Why bother changing state from TERMINATING to TERMINATING

Hope this information is useful.

Thanks.




From mranga at gmail.com  Mon Dec 27 20:35:26 2010
From: mranga at gmail.com (M. Ranganathan)
Date: Mon, 27 Dec 2010 12:35:26 -0500
Subject: [Freeswitch-users] Bad P-Asserted-Identity header from FS
	conference.
Message-ID: <AANLkTikxVS5_kMGSaHyAmbsBwDAZze1ecPcZ7BbwW9N9@mail.gmail.com>

Hello,

I am not sure what triggers this but I get a bad P-Asserted-Identity
header from Free-SWITCH in an UPDATE request when I create a
conference and send an INVITE to that conference. Here is the
offending Request :

UPDATE sip:502 at 192.168.5.75:6060;x-sipX-nonat SIP/2.0
Via: SIP/2.0/UDP 192.168.5.75:15060;rport;branch=z9hG4bK2ZtpZ94m4Zg7F
Route: <sip:192.168.5.75:5060;lr;sipXecs-rs=*auth~.*from~OUQ1MENEMEQtODAzNjU5QkU%60.900_ntap*id~MTQ0MDMtOQ%60%60!34e982a809955a84b401940d3c785d61>
Max-Forwards: 70
From: <sip:502 at sipxtest.sipxtest.net;user=phone>;tag=2ecF2jHpjjH6c
To: "user1" <sip:user1 at sipxtest.sipxtest.net>;tag=9D50CD0D-803659BE
Call-ID: 41745e29-669e33ca-619711cf at 192.168.5.242
CSeq: 6399341 UPDATE
Contact: <sip:502 at 192.168.5.75:15060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-2
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 297
P-Asserted-Identity: "conference" <conference>


p-asserted-identity should specify a SIP URI and it does not.

Is this indicative of a misconfiguration somewhere or is it a bug?
Where is the P-A-I header constructed from?

Thanks

Regards,

Ranga


-- 
M. Ranganathan


From anthony.minessale at gmail.com  Mon Dec 27 20:53:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 27 Dec 2010 11:53:00 -0600
Subject: [Freeswitch-users] Bad P-Asserted-Identity header from FS
	conference.
In-Reply-To: <AANLkTikxVS5_kMGSaHyAmbsBwDAZze1ecPcZ7BbwW9N9@mail.gmail.com>
References: <AANLkTikxVS5_kMGSaHyAmbsBwDAZze1ecPcZ7BbwW9N9@mail.gmail.com>
Message-ID: <AANLkTikOTXO9ET6b7FAYuv4eT2Gcqmw04TH21Ui55A7S@mail.gmail.com>

What is the client?
Those should only be sent to supported devices which are stupid phones
who actually want it that way.
There is some code to detect it based on the user agent.


On Mon, Dec 27, 2010 at 11:35 AM, M. Ranganathan <mranga at gmail.com> wrote:
> Hello,
>
> I am not sure what triggers this but I get a bad P-Asserted-Identity
> header from Free-SWITCH in an UPDATE request when I create a
> conference and send an INVITE to that conference. Here is the
> offending Request :
>
> UPDATE sip:502 at 192.168.5.75:6060;x-sipX-nonat SIP/2.0
> Via: SIP/2.0/UDP 192.168.5.75:15060;rport;branch=z9hG4bK2ZtpZ94m4Zg7F
> Route: <sip:192.168.5.75:5060;lr;sipXecs-rs=*auth~.*from~OUQ1MENEMEQtODAzNjU5QkU%60.900_ntap*id~MTQ0MDMtOQ%60%60!34e982a809955a84b401940d3c785d61>
> Max-Forwards: 70
> From: <sip:502 at sipxtest.sipxtest.net;user=phone>;tag=2ecF2jHpjjH6c
> To: "user1" <sip:user1 at sipxtest.sipxtest.net>;tag=9D50CD0D-803659BE
> Call-ID: 41745e29-669e33ca-619711cf at 192.168.5.242
> CSeq: 6399341 UPDATE
> Contact: <sip:502 at 192.168.5.75:15060;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-2
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 297
> P-Asserted-Identity: "conference" <conference>
>
>
> p-asserted-identity should specify a SIP URI and it does not.
>
> Is this indicative of a misconfiguration somewhere or is it a bug?
> Where is the P-A-I header constructed from?
>
> Thanks
>
> Regards,
>
> Ranga
>
>
> --
> M. Ranganathan
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From mranga at gmail.com  Mon Dec 27 21:10:53 2010
From: mranga at gmail.com (M. Ranganathan)
Date: Mon, 27 Dec 2010 13:10:53 -0500
Subject: [Freeswitch-users] Bad P-Asserted-Identity header from FS
	conference.
In-Reply-To: <AANLkTikOTXO9ET6b7FAYuv4eT2Gcqmw04TH21Ui55A7S@mail.gmail.com>
References: <AANLkTikxVS5_kMGSaHyAmbsBwDAZze1ecPcZ7BbwW9N9@mail.gmail.com>
	<AANLkTikOTXO9ET6b7FAYuv4eT2Gcqmw04TH21Ui55A7S@mail.gmail.com>
Message-ID: <AANLkTimXLMhVTvXrbN_g02jrOE=ebeSX5_1fQ5m7sikF@mail.gmail.com>

The user agent header for the Phone that sends the INVITE in this case is

User-Agent: PolycomSoundPointIP-SPIP_320-UA/3.3.0.1098

Thanks

Ranga



On Mon, Dec 27, 2010 at 12:53 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> What is the client?
> Those should only be sent to supported devices which are stupid phones
> who actually want it that way.
> There is some code to detect it based on the user agent.
>
>
> On Mon, Dec 27, 2010 at 11:35 AM, M. Ranganathan <mranga at gmail.com> wrote:
>> Hello,
>>
>> I am not sure what triggers this but I get a bad P-Asserted-Identity
>> header from Free-SWITCH in an UPDATE request when I create a
>> conference and send an INVITE to that conference. Here is the
>> offending Request :
>>
>> UPDATE sip:502 at 192.168.5.75:6060;x-sipX-nonat SIP/2.0
>> Via: SIP/2.0/UDP 192.168.5.75:15060;rport;branch=z9hG4bK2ZtpZ94m4Zg7F
>> Route: <sip:192.168.5.75:5060;lr;sipXecs-rs=*auth~.*from~OUQ1MENEMEQtODAzNjU5QkU%60.900_ntap*id~MTQ0MDMtOQ%60%60!34e982a809955a84b401940d3c785d61>
>> Max-Forwards: 70
>> From: <sip:502 at sipxtest.sipxtest.net;user=phone>;tag=2ecF2jHpjjH6c
>> To: "user1" <sip:user1 at sipxtest.sipxtest.net>;tag=9D50CD0D-803659BE
>> Call-ID: 41745e29-669e33ca-619711cf at 192.168.5.242
>> CSeq: 6399341 UPDATE
>> Contact: <sip:502 at 192.168.5.75:15060;transport=udp>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-2
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> Supported: timer, precondition, path, replaces
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 297
>> P-Asserted-Identity: "conference" <conference>
>>
>>
>> p-asserted-identity should specify a SIP URI and it does not.
>>
>> Is this indicative of a misconfiguration somewhere or is it a bug?
>> Where is the P-A-I header constructed from?
>>
>> Thanks
>>
>> Regards,
>>
>> Ranga
>>
>>
>> --
>> M. Ranganathan
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
M. Ranganathan


From anthony.minessale at gmail.com  Mon Dec 27 21:17:36 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 27 Dec 2010 12:17:36 -0600
Subject: [Freeswitch-users] Bad P-Asserted-Identity header from FS
	conference.
In-Reply-To: <AANLkTimXLMhVTvXrbN_g02jrOE=ebeSX5_1fQ5m7sikF@mail.gmail.com>
References: <AANLkTikxVS5_kMGSaHyAmbsBwDAZze1ecPcZ7BbwW9N9@mail.gmail.com>
	<AANLkTikOTXO9ET6b7FAYuv4eT2Gcqmw04TH21Ui55A7S@mail.gmail.com>
	<AANLkTimXLMhVTvXrbN_g02jrOE=ebeSX5_1fQ5m7sikF@mail.gmail.com>
Message-ID: <AANLkTikOCJOvytR-gYxfGd8wCaOvaANh+_Tv1APJP1cg@mail.gmail.com>

This is exactly how polycom expects the header.  anything else is
incorrectly displayed on the screen.
We actually force the packet to look this way when user agent says polycom.

Judging the contents of the header when acting as a proxy is a bit over-zealous.



On Mon, Dec 27, 2010 at 12:10 PM, M. Ranganathan <mranga at gmail.com> wrote:
> The user agent header for the Phone that sends the INVITE in this case is
>
> User-Agent: PolycomSoundPointIP-SPIP_320-UA/3.3.0.1098
>
> Thanks
>
> Ranga
>
>
>
> On Mon, Dec 27, 2010 at 12:53 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> What is the client?
>> Those should only be sent to supported devices which are stupid phones
>> who actually want it that way.
>> There is some code to detect it based on the user agent.
>>
>>
>> On Mon, Dec 27, 2010 at 11:35 AM, M. Ranganathan <mranga at gmail.com> wrote:
>>> Hello,
>>>
>>> I am not sure what triggers this but I get a bad P-Asserted-Identity
>>> header from Free-SWITCH in an UPDATE request when I create a
>>> conference and send an INVITE to that conference. Here is the
>>> offending Request :
>>>
>>> UPDATE sip:502 at 192.168.5.75:6060;x-sipX-nonat SIP/2.0
>>> Via: SIP/2.0/UDP 192.168.5.75:15060;rport;branch=z9hG4bK2ZtpZ94m4Zg7F
>>> Route: <sip:192.168.5.75:5060;lr;sipXecs-rs=*auth~.*from~OUQ1MENEMEQtODAzNjU5QkU%60.900_ntap*id~MTQ0MDMtOQ%60%60!34e982a809955a84b401940d3c785d61>
>>> Max-Forwards: 70
>>> From: <sip:502 at sipxtest.sipxtest.net;user=phone>;tag=2ecF2jHpjjH6c
>>> To: "user1" <sip:user1 at sipxtest.sipxtest.net>;tag=9D50CD0D-803659BE
>>> Call-ID: 41745e29-669e33ca-619711cf at 192.168.5.242
>>> CSeq: 6399341 UPDATE
>>> Contact: <sip:502 at 192.168.5.75:15060;transport=udp>
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-2
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>> Supported: timer, precondition, path, replaces
>>> Content-Type: application/sdp
>>> Content-Disposition: session
>>> Content-Length: 297
>>> P-Asserted-Identity: "conference" <conference>
>>>
>>>
>>> p-asserted-identity should specify a SIP URI and it does not.
>>>
>>> Is this indicative of a misconfiguration somewhere or is it a bug?
>>> Where is the P-A-I header constructed from?
>>>
>>> Thanks
>>>
>>> Regards,
>>>
>>> Ranga
>>>
>>>
>>> --
>>> M. Ranganathan
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> M. Ranganathan
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Mon Dec 27 22:50:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 27 Dec 2010 13:50:10 -0600
Subject: [Freeswitch-users] Time Conditions & Break - Weird Behavior? &
 Anti-actions Usefulness?
In-Reply-To: <AANLkTinQEpW_O=zMRaeU77K_ng43f27+vv93m9wmc=DG@mail.gmail.com>
References: <AANLkTinQEpW_O=zMRaeU77K_ng43f27+vv93m9wmc=DG@mail.gmail.com>
Message-ID: <AANLkTik1DmpQMR366p5kyvFQXeLbmCt2busuD8qYHrhm@mail.gmail.com>

anti-actions are like the xml file equiv of else

the actions are executed when the condition is matched
the anti-actions are executed when it does not match

are you aware that there is time of day matching built right into the
xml that does not require doing regex?

http://wiki.freeswitch.org/wiki/Time_of_Day_Routing

On Sun, Dec 26, 2010 at 5:49 PM, Avi Marcus <Avi at amarcus.com> wrote:
> I have two xml config files with time of day routing, and I couldn't imagine
> why when it reached the first one, it just skipped the rest of the xml
> files, as such:
> Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 parsing [default->YLE1]
> continue=false
> Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Regex (PASS) [YLE1]
> destination_number(t1105) =~ /^t1105$/ break=on-false
> Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Date/Time Match (FAIL)
> [YLE1] break=on-false
> 2010-12-27 01:36:09.410521 [DEBUG] switch_core_state_machine.c:119
> (sofia/internal/sip:1000 at 109.67.185.37:5072) State Change CS_ROUTING ->
> CS_EXECUTE
> But when I just edit the first and set continue=true, it suddenly now
> properly moves on. But it never matched the entire extension's condition
> tags, so why did it think it should not continue?
> (xml files here:?http://pastebin.freeswitch.org/14885)
> Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 parsing [default->YLE1]
> continue=true
> Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Regex (PASS) [YLE1]
> destination_number(t1105) =~ /^t1105$/ break=on-false
> Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Date/Time Match (FAIL)
> [YLE1] break=on-false
> Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 parsing [default->YLE2]
> continue=false
> Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Regex (PASS) [YLE2]
> destination_number(t1105) =~ /^t1105$/ break=on-false
> Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Date/Time Match (PASS)
> [YLE2] break=on-false
> Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Regex (FAIL) [YLE2]
> time-of-day() =~ /08:00-22:00/ break=on-false
> Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 ANTI-Action
> voicemail(default ${domain} 1105)
> 2010-12-27 01:37:24.754671 [DEBUG] switch_core_state_machine.c:119
> (sofia/internal/sip:1000 at 109.67.185.37:5072) State Change CS_ROUTING ->
> CS_EXECUTE
> And - it seems the anti-action is?triggered?no matter how "off" we are, is
> there a way to have a less agressive anti action? E.g. it only activates if
> the first condition is true? It's stopping the rest of my dial plan from
> executing.
> When would one ever want an anti-action other than as his last <extension>
> in which case it could just be a new last extension with no condition tag. I
> don't?understand?why you would ever want them.
> Understanding would be appreciated :)
> -Avi Marcus
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From Avi at aMarcus.com  Mon Dec 27 23:19:32 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Mon, 27 Dec 2010 22:19:32 +0200
Subject: [Freeswitch-users] Time Conditions & Break - Weird Behavior? &
 Anti-actions Usefulness?
In-Reply-To: <AANLkTik1DmpQMR366p5kyvFQXeLbmCt2busuD8qYHrhm@mail.gmail.com>
References: <AANLkTinQEpW_O=zMRaeU77K_ng43f27+vv93m9wmc=DG@mail.gmail.com>
	<AANLkTik1DmpQMR366p5kyvFQXeLbmCt2busuD8qYHrhm@mail.gmail.com>
Message-ID: <AANLkTikPs8vc-eD+KKY4vFrZBppjmshTAnj6r+8cJDPK@mail.gmail.com>

On Mon, Dec 27, 2010 at 9:50 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> anti-actions are like the xml file equiv of else
>
> the actions are executed when the condition is matched
> the anti-actions are executed when it does not match
>
> are you aware that there is time of day matching built right into the
> xml that does not require doing regex?
>
> http://wiki.freeswitch.org/wiki/Time_of_Day_Routing

It seems that was a typo of an added (). I don't have the file so I can't be
sue, but it must have been. (as I figured out in the last email)
Thanks,
Avi



>
>
> On Sun, Dec 26, 2010 at 5:49 PM, Avi Marcus <Avi at amarcus.com> wrote:
> > I have two xml config files with time of day routing, and I couldn't
> imagine
> > why when it reached the first one, it just skipped the rest of the xml
> > files, as such:
> > Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 parsing
> [default->YLE1]
> > continue=false
> > Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Regex (PASS) [YLE1]
> > destination_number(t1105) =~ /^t1105$/ break=on-false
> > Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Date/Time Match
> (FAIL)
> > [YLE1] break=on-false
> > 2010-12-27 01:36:09.410521 [DEBUG] switch_core_state_machine.c:119
> > (sofia/internal/sip:1000 at 109.67.185.37:5072) State Change CS_ROUTING ->
> > CS_EXECUTE
> > But when I just edit the first and set continue=true, it suddenly now
> > properly moves on. But it never matched the entire extension's condition
> > tags, so why did it think it should not continue?
> > (xml files here: http://pastebin.freeswitch.org/14885)
> > Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 parsing
> [default->YLE1]
> > continue=true
> > Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Regex (PASS) [YLE1]
> > destination_number(t1105) =~ /^t1105$/ break=on-false
> > Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Date/Time Match
> (FAIL)
> > [YLE1] break=on-false
> > Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 parsing
> [default->YLE2]
> > continue=false
> > Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Regex (PASS) [YLE2]
> > destination_number(t1105) =~ /^t1105$/ break=on-false
> > Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Date/Time Match
> (PASS)
> > [YLE2] break=on-false
> > Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 Regex (FAIL) [YLE2]
> > time-of-day() =~ /08:00-22:00/ break=on-false
> > Dialplan: sofia/internal/sip:1000 at 109.67.185.37:5072 ANTI-Action
> > voicemail(default ${domain} 1105)
> > 2010-12-27 01:37:24.754671 [DEBUG] switch_core_state_machine.c:119
> > (sofia/internal/sip:1000 at 109.67.185.37:5072) State Change CS_ROUTING ->
> > CS_EXECUTE
> > And - it seems the anti-action is triggered no matter how "off" we are,
> is
> > there a way to have a less agressive anti action? E.g. it only activates
> if
> > the first condition is true? It's stopping the rest of my dial plan from
> > executing.
> > When would one ever want an anti-action other than as his last
> <extension>
> > in which case it could just be a new last extension with no condition
> tag. I
> > don't understand why you would ever want them.
> > Understanding would be appreciated :)
> > -Avi Marcus
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From rupa at rupa.com  Tue Dec 28 01:27:31 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 27 Dec 2010 16:27:31 -0600
Subject: [Freeswitch-users] nibblebill schedule
In-Reply-To: <DA3EFAD250384C26947B906385BE23D1@e1705>
References: <F07B572C409042239691B1F3DE08575E@e1705>
	<AANLkTi=W9BV+JnD4K=Oqv_fGSt-SpVM2jYV3AyL7xzO2@mail.gmail.com>
	<DA3EFAD250384C26947B906385BE23D1@e1705>
Message-ID: <AANLkTinwV_vFXqSLqwTnyXJZopLzdYQ_22Lodm9DqKn5@mail.gmail.com>

Look at setting up nibblebill in paused mode, then use the schedule api to
take nibblebill out of pause mode after 30s.

http://wiki.freeswitch.org/wiki/Mod_nibblebill#Pause
<http://wiki.freeswitch.org/wiki/Mod_nibblebill#Pause>
http://wiki.freeswitch.org/wiki/Mod_commands (look for sched_api)

On Sat, Dec 25, 2010 at 10:22 AM, Madovsky <infos at madovsky.org> wrote:

>  ha ok,
> the concept I thought is to let user to call 30s as a trial and
> start niblebill after this time.... does it need to modify the
> mod_nibblebill source code ?
>
> Thanks
>
> F
>
> ----- Original Message -----
> *From:* Rupa Schomaker <rupa at rupa.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Saturday, December 25, 2010 8:38 AM
> *Subject:* Re: [Freeswitch-users] nibblebill schedule
>
> I don't see how without change.
>
> You have no idea how fast the $$ is being nibbled away by other calls.  So,
> while you can account for a 30s buffer in the current call using the current
> rate, the more calls that are up for that account the more "off" the 30s
> estimate will be.
>
> On Fri, Dec 24, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  Is it possible to schedule the transfer to nibblebill nobal_action ?
>> I'd like to schedule of 30s before the call is cut and go to nobal_action
>> extension
>>
>> thanks
>>
>>
> --
> -Rupa
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From lloyd.aloysius at sunteltech.ca  Tue Dec 28 04:20:08 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Mon, 27 Dec 2010 20:20:08 -0500
Subject: [Freeswitch-users] group_confirm - Question
In-Reply-To: <AANLkTikF9FqqFUdi2htpNKcbNNYC8rS3ZRF-JeG0QVrZ@mail.gmail.com>
References: <AANLkTikPXY7073P3=3Gto+ArfBY41TUxkikKRAkGe13A@mail.gmail.com>
	<AANLkTi=f+MsRTD=du8YUHmVp=ANrDRTzOmD_4XCx48az@mail.gmail.com>
	<AANLkTi=tDcLWG6V6fQHN1d95GKvOaB-NOR3BQFB38PeP@mail.gmail.com>
	<AANLkTikF9FqqFUdi2htpNKcbNNYC8rS3ZRF-JeG0QVrZ@mail.gmail.com>
Message-ID: <AANLkTim5vrjumTps0KZ2_8NZ1Ea6DK4LBCJNbt7dB4gF@mail.gmail.com>

Dialplan and Log - http://pastebin.freeswitch.org/14887

Here is the dial plan. No Errors ... but group_confirm not working and call
directly connected.

 <http://pastebin.freeswitch.org/14887>
<extension name="Local_Extension_FollowMe">
   <condition field="${user_data(${destination_number}@${domain_name} var
followme_status)}" expression="enable"/>
   <condition field="destination_number" expression="^(20[0-9])$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s]
<app> -->
        <action application="bind_meta_app" data="1 b s
execute_extension::dx XML features"/>
        <action application="bind_meta_app" data="2 b s
record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
        <action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>
        <action application="bind_meta_app" data="4 b s
execute_extension::att_xfer XML features"/>
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="call_timeout=20"/>
        <!-- <action application="set"
data="sip_exclude_contact=${network_addr}"/> -->
        <action application="set" data="hangup_after_bridge=true"/>
        <!--<action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
-->
        <action application="set" data="continue_on_fail=true"/>
        <action application="hash"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
        <action application="set"
data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}"/>
        <!--<action application="export"
data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name}
var sip_secure_media)}"/>-->
        <action application="hash"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
       * <action application="bridge"
data="{ignore_early_media=true}[leg_timeout=20]user/${dialed_extension}@
${domain_name},[ignore_early_media=true,group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1]sofia/gateway/utel/${user_data(${dialed_extension}@${domain_name}
var followme_number)}"/>*
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="bridge" data="loopback/app=voicemail:default
${domain_name} ${dialed_extension}"/>
    </condition>
</extension>

Thanks
Lloyd

On Mon, Dec 27, 2010 at 1:04 AM, Avi Marcus <Avi at amarcus.com> wrote:

> Can you paste the bridge for us again just to confirm the change, and the
> new log?
> And to clarify, you mean it's not working for the PSTN recipients, correct?
> -Avi
>
>
> On Mon, Dec 27, 2010 at 6:57 AM, Aloysius Lloyd <
> lloyd.aloysius at sunteltech.ca> wrote:
>
>> I try the [] now no errors. But
>>
>> -  group_confirm_file not play
>> - call directly connected
>>
>> basically group_confirm not working.
>>
>> Thanks
>> Lloyd
>>
>> On Sun, Dec 26, 2010 at 4:29 PM, Avi Marcus <Avi at amarcus.com> wrote:
>>
>>> I don't think you are supposed to be using {} variables in the middle of
>>> your bridge string. It's only used at the beginning for setting variables
>>> for EVERY bridge string, else, you should enclose it in []. I think. Give it
>>> a try.
>>> -Avi Marcus
>>>
>>> On Sun, Dec 26, 2010 at 10:37 PM, Aloysius Lloyd <
>>> lloyd.aloysius at gmail.com> wrote:
>>>
>>>> Hi All,
>>>>
>>>> Dial plan below use group_confrim,  work for both internal and external
>>>> (PSTN) Calls without any issues
>>>>
>>>> <action application="bridge" data="{ignore_early_media=true,*
>>>> group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1
>>>> *}user/${dialed_extension}@
>>>> ${domain_name},sofia/gateway/voipms/${user_data(${dialed_extension}@${domain_name}
>>>> var followme_number)}"/>
>>>>
>>>>
>>>> -----
>>>>
>>>> But I would like to use the group_confirm feature for external PSTN
>>>> (Calls only). The following Dial plan is not working and the console log
>>>> below.
>>>>
>>>> <action application="bridge"
>>>> data="{ignore_early_media=true}user/${dialed_extension}@
>>>> ${domain_name},{*
>>>> group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1
>>>> *}sofia/gateway/voipms/${user_data(${dialed_extension}@${domain_name}
>>>> var followme_number)}"/>
>>>>
>>>>
>>>> *Console Log*
>>>>
>>>> 2010-12-26 15:23:21.796308 [ERR] switch_core_session.c:380 Could not
>>>> locate channel type {group_confirm_file=
>>>> 2010-12-26 15:23:21.796308 [ERR] switch_ivr_originate.c:2614 Cannot
>>>> create outgoing channel of type [{group_confirm_file=] cause:
>>>> [CHAN_NOT_IMPLEMENTED]
>>>> 2010-12-26 15:23:21.796308 [ERR] switch_core_session.c:380 Could not
>>>> locate channel type group_confirm_key=1}sofia
>>>> 2010-12-26 15:23:21.796308 [ERR] switch_ivr_originate.c:2614 Cannot
>>>> create outgoing channel of type [group_confirm_key=1}sofia] cause:
>>>> [CHAN_NOT_IMPLEMENTED]
>>>>
>>>> --------
>>>>
>>>> Any help is appreciated.
>>>>
>>>> Thanks
>>>> Lloyd
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Tue Dec 28 05:13:41 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 27 Dec 2010 21:13:41 -0500
Subject: [Freeswitch-users] voicemail operator-extension - Multi-tenant
	Environment
Message-ID: <AANLkTimm99xdzt2cW6zTpFZY+FtiRCaHCQSm=Rv__o4G@mail.gmail.com>

Hi All,

I am trying to use the operator extension in Multi-tenant environment. I
modified the voicemail.conf.xml file as below.

  <param name="operator-extension" value="operator XML *${domain_name}*"/>

Looks to me this is not the right way to do ... when I press the 9
(voicemail operator extension) call ended. Below is the log...


2010-12-27 21:07:27.664362 [NOTICE] switch_ivr.c:1603 Transfer
loopback/voicemail-b to XML[operator@${domain_name}]
2010-12-27 21:07:27.664362 [DEBUG] switch_core_state_machine.c:366
(loopback/voicemail-b) State EXECUTE going to sleep
2010-12-27 21:07:27.664362 [DEBUG] switch_core_state_machine.c:320
(loopback/voicemail-b) Running State Change CS_ROUTING
2010-12-27 21:07:27.664362 [DEBUG] switch_channel.c:1615
(loopback/voicemail-b) Callstate Change EARLY -> RINGING
2010-12-27 21:07:27.664362 [DEBUG] switch_core_state_machine.c:359
(loopback/voicemail-b) State ROUTING
2010-12-27 21:07:27.665397 [DEBUG] mod_loopback.c:323 loopback/voicemail-b
CHANNEL ROUTING
2010-12-27 21:07:27.665397 [DEBUG] switch_core_state_machine.c:77
loopback/voicemail-b Standard ROUTING
2010-12-27 21:07:27.665397 [INFO] mod_dialplan_xml.c:331 Processing Ext 202
<202>->*operator in context ${domain_name}*
2010-12-27 21:07:27.665397 [WARNING] mod_dialplan_xml.c:361 *Context
${domain_name} not found*
2010-12-27 21:07:27.665397 [INFO] switch_core_state_machine.c:142 No Route,
Aborting


Is there any way to use the *${domain_name} *in the voicemail
operator-extension parameter. What is the recommended way of using
operator-extension in Multi-tenant environment.
*
*
*
*
*
*
Thanks and regards,
Lloyd
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From infos at madovsky.org  Tue Dec 28 05:59:12 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 27 Dec 2010 21:59:12 -0500
Subject: [Freeswitch-users] voicemail operator-extension -
	Multi-tenantEnvironment
References: <AANLkTimm99xdzt2cW6zTpFZY+FtiRCaHCQSm=Rv__o4G@mail.gmail.com>
Message-ID: <614459D4C10248C8B913E4422CFC7C35@e1705>

maybe $${domain}
  ----- Original Message ----- 
  From: Aloysius Lloyd 
  To: FreeSWITCH Users Help 
  Sent: Monday, December 27, 2010 9:13 PM
  Subject: [Freeswitch-users] voicemail operator-extension - Multi-tenantEnvironment


  Hi All, 


  I am trying to use the operator extension in Multi-tenant environment. I modified the voicemail.conf.xml file as below.


    <param name="operator-extension" value="operator XML ${domain_name}"/>


  Looks to me this is not the right way to do ... when I press the 9 (voicemail operator extension) call ended. Below is the log...




  2010-12-27 21:07:27.664362 [NOTICE] switch_ivr.c:1603 Transfer loopback/voicemail-b to XML[operator@${domain_name}]
  2010-12-27 21:07:27.664362 [DEBUG] switch_core_state_machine.c:366 (loopback/voicemail-b) State EXECUTE going to sleep
  2010-12-27 21:07:27.664362 [DEBUG] switch_core_state_machine.c:320 (loopback/voicemail-b) Running State Change CS_ROUTING
  2010-12-27 21:07:27.664362 [DEBUG] switch_channel.c:1615 (loopback/voicemail-b) Callstate Change EARLY -> RINGING
  2010-12-27 21:07:27.664362 [DEBUG] switch_core_state_machine.c:359 (loopback/voicemail-b) State ROUTING
  2010-12-27 21:07:27.665397 [DEBUG] mod_loopback.c:323 loopback/voicemail-b CHANNEL ROUTING
  2010-12-27 21:07:27.665397 [DEBUG] switch_core_state_machine.c:77 loopback/voicemail-b Standard ROUTING
  2010-12-27 21:07:27.665397 [INFO] mod_dialplan_xml.c:331 Processing Ext 202 <202>->operator in context ${domain_name}
  2010-12-27 21:07:27.665397 [WARNING] mod_dialplan_xml.c:361 Context ${domain_name} not found
  2010-12-27 21:07:27.665397 [INFO] switch_core_state_machine.c:142 No Route, Aborting




  Is there any way to use the ${domain_name} in the voicemail operator-extension parameter. What is the recommended way of using operator-extension in Multi-tenant environment.






  Thanks and regards,
  Lloyd


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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From acosgrov at gmail.com  Tue Dec 28 06:10:15 2010
From: acosgrov at gmail.com (Anthony Cosgrove)
Date: Mon, 27 Dec 2010 22:10:15 -0500
Subject: [Freeswitch-users] voicemail operator-extension -
	Multi-tenantEnvironment
In-Reply-To: <614459D4C10248C8B913E4422CFC7C35@e1705>
References: <AANLkTimm99xdzt2cW6zTpFZY+FtiRCaHCQSm=Rv__o4G@mail.gmail.com>
	<614459D4C10248C8B913E4422CFC7C35@e1705>
Message-ID: <D330BE13-B0A1-4D76-9DFA-B80B7F3534D8@gmail.com>

Aren't $${VARIABLES} read one-time when the XML is parsed? I believe you are going to need to use ${domain_name} but it needs to be set when the channel is created.


Anthony C

On Dec 27, 2010, at 9:59 PM, Madovsky wrote:

> maybe $${domain}
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lloyd.aloysius at sunteltech.ca  Tue Dec 28 07:08:05 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Mon, 27 Dec 2010 23:08:05 -0500
Subject: [Freeswitch-users] voicemail operator-extension -
	Multi-tenantEnvironment
In-Reply-To: <D330BE13-B0A1-4D76-9DFA-B80B7F3534D8@gmail.com>
References: <AANLkTimm99xdzt2cW6zTpFZY+FtiRCaHCQSm=Rv__o4G@mail.gmail.com>
	<614459D4C10248C8B913E4422CFC7C35@e1705>
	<D330BE13-B0A1-4D76-9DFA-B80B7F3534D8@gmail.com>
Message-ID: <AANLkTinATfPEbDkzfZSyXHgVYM0mjtsgvzk28xYBotXt@mail.gmail.com>

$${domain} - incorrect

It is already set when the Chanel is created. here is the dialplan when I
reach to voicemail box already belong to a domain.

<action application="bridge" data="loopback/app=voicemail:default
${domain_name} ${dialed_extension}"/>

when is press 9 - I am already in the domain extension voice mail box so
there is ${domain_name} variable have the value.

looks to me voicemail application does not understand the ${domain_name}

Thanks
Lloyd


On Mon, Dec 27, 2010 at 10:10 PM, Anthony Cosgrove <acosgrov at gmail.com>wrote:

> Aren't $${VARIABLES} read one-time when the XML is parsed? I believe you
> are going to need to use ${domain_name} but it needs to be set when the
> channel is created.
>
>
> Anthony C
>
> On Dec 27, 2010, at 9:59 PM, Madovsky wrote:
>
> maybe $${domain}
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From godson.g at gmail.com  Tue Dec 28 09:51:57 2010
From: godson.g at gmail.com (Godson Gera)
Date: Tue, 28 Dec 2010 12:21:57 +0530
Subject: [Freeswitch-users] =?windows-1252?q?ANN_=3A_PySWITCH_Release_=96_?=
	=?windows-1252?q?0=2E1alpha?=
Message-ID: <AANLkTi=FDcobUchZZsq8uWzfSe99MgTHW5FvZV=d2kzM@mail.gmail.com>

Hi All,

I am glad to announce the first alpha release of PySWITCH.

http://pyswitch.sf.net


The idea of PySWITCH is to offer a complete library to Python and Twisted
programmers for interacting with FreeSWITCH using EventSocket interface. The
target is to cover all FreeSWITCH API commands and Dialplan tools. PySWITCH
handles all the low level details in executing FreeSWITCH commands, so the
programmer can easily concentrate on quickly building FreeSWITCH
applications. As an example, the API functions offered by PySWITCH often
executes many FreeSWITCH commands under the hood and finally returns the
desired result. Suppose you execute a background job, PySWITCH API will
automatically wait and catch the backgroundjob event parse the result and
will fire the deferred.

The current release covers good amount of API commands and a few Dialplan
tools. The protocol communication issues are ironed out. It has a nice event
call back interface. I?ll present its usage in couple of tutorials soon.


-- 
Thanks & Regards,
Godson Gera
http://godson.in
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From msc at freeswitch.org  Tue Dec 28 10:51:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 27 Dec 2010 23:51:16 -0800
Subject: [Freeswitch-users] voicemail operator-extension -
	Multi-tenantEnvironment
In-Reply-To: <AANLkTinATfPEbDkzfZSyXHgVYM0mjtsgvzk28xYBotXt@mail.gmail.com>
References: <AANLkTimm99xdzt2cW6zTpFZY+FtiRCaHCQSm=Rv__o4G@mail.gmail.com>
	<614459D4C10248C8B913E4422CFC7C35@e1705>
	<D330BE13-B0A1-4D76-9DFA-B80B7F3534D8@gmail.com>
	<AANLkTinATfPEbDkzfZSyXHgVYM0mjtsgvzk28xYBotXt@mail.gmail.com>
Message-ID: <AANLkTik=afD7Zjsb_jPw1aNSXsxLCtHeshiVbK-aHTbA@mail.gmail.com>

On Mon, Dec 27, 2010 at 8:08 PM, Aloysius Lloyd <
lloyd.aloysius at sunteltech.ca> wrote:

> $${domain} - incorrect
>
> It is already set when the Chanel is created. here is the dialplan when I
> reach to voicemail box already belong to a domain.
>
> <action application="bridge" data="loopback/app=voicemail:default
> ${domain_name} ${dialed_extension}"/>
>
> when is press 9 - I am already in the domain extension voice mail box so
> there is ${domain_name} variable have the value.
>
> looks to me voicemail application does not understand the ${domain_name}
>

This is almost a true statement. In actuality, you don't put chan vars in
params like this and have them interpolated with each call. Interpolation of
channel variables like this is best done in the dialplan. The easiest way
for you to do this would be to set the value back to the default of:
 <param name="operator-extension" value="operator XML default"/>
...and then create the "operator" extension which handles the transfer:
<extension name="operator transfer">
  <condition field="destination_number" expression="^(operator)$">
    <!-- ${domain_name} should be available here -->
    <action application="transfer" data="operator XML ${domain_name}"/>
  </condition>
</extension>

Just be sure that you don't have a domain name "default" or this will blow
up. :) You could also put this "utility" operator extension in the
'features' context and then as long as you didn't have a domain named
'features' then you would be just fine.

-MC
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From u2nsam at gmail.com  Tue Dec 28 12:08:36 2010
From: u2nsam at gmail.com (Sam)
Date: Tue, 28 Dec 2010 14:38:36 +0530
Subject: [Freeswitch-users] call pickup
In-Reply-To: <AANLkTinHoVMzeP6eq34grTQoqB-7u3FJrrn+1=7os9SD@mail.gmail.com>
References: <1292391773589-5837233.post@n2.nabble.com>
	<665DE8BC-65A7-46D2-8F8E-002649C59C21@ipeva.fr>
	<AANLkTimZeMcBbDkW-W=O8661Vx4ZXququv8qPtLFTDgJ@mail.gmail.com>
	<AANLkTi=KM=Nhj-D7SyyVi1hMfT8Luja3chcUfg=NwByd@mail.gmail.com>
	<AANLkTimsAshk3=k8BvbWX=XeiAoVXCO5vC8kG50T3_KQ@mail.gmail.com>
	<AANLkTinHoVMzeP6eq34grTQoqB-7u3FJrrn+1=7os9SD@mail.gmail.com>
Message-ID: <AANLkTiknLoq884pAnZxTUwePWTBVCiYutjnP40rAtqhW@mail.gmail.com>

Thanks all,

what i inserted , i selected on intercept hash .

Regards
Sam

On Sat, Dec 18, 2010 at 4:57 AM, Chris Burns <chris at cloudtel.com> wrote:

> You are doing a hash select to determine the data sent the the intercept
> app ... is it selecting out meaningful data that you are inserting
> elsewhere? The intercept example in the default dialplan works in concert
> with other elements which may not be present in your own.
>
>
> On Fri, Dec 17, 2010 at 11:03 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Yes those are define for respective extension groups.
>>
>> Was the dialplan correct or anything needs to be updated ?
>>
>> Thanks & regds
>> Sam
>>
>>
>> On Fri, Dec 17, 2010 at 7:44 PM, Frank Park <frank at telonium.com> wrote:
>>
>>> Have you defined the "callgroup" variable in the directory?
>>> <variable name="callgroup" value="VALUE"/>
>>>
>>>
>>>
>>> ----=======================----
>>> Frank Park
>>> Telonium Communications, LLC
>>> frank at telonium.com
>>> http://www.telonium.com
>>> Follow Us on Twitter: @GetTelonium
>>> 404-566-8888 x1001 Office
>>> 404-939-4242 Cell
>>> ----=======================----
>>>
>>>
>>> On Wed, Dec 15, 2010 at 10:34 PM, Sam <u2nsam at gmail.com> wrote:
>>> >
>>> > Yes David ,
>>> >
>>> > You are right from yesterday i was trying to post it to the forum and
>>> it was giving me message that your post is not posted in the forum.
>>> > And at last it did with multiple emails !! may be it could have
>>> happened as i created the account day before yesterday and it takes time.
>>> > Sorry all for any inconvenience caused !
>>> >
>>> > Regds
>>> >
>>> > On Wed, Dec 15, 2010 at 3:47 PM, David Ponzone <david.ponzone at ipeva.fr>
>>> wrote:
>>> >>
>>> >> Samir,
>>> >> Perhaps next time, you could think before sending your mail.
>>> >> It will avoid you sending multiple emails for the same question.
>>> >> David Ponzone  Direction Technique
>>> >> email: david.ponzone at ipeva.fr
>>> >> tel:      01 74 03 18 97
>>> >> gsm:   06 66 98 76 34
>>> >> Service Client IPeva
>>> >> tel:      0811 46 26 26
>>> >> www.ipeva.fr  -   www.ipeva-studio.com
>>> >> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>> >>
>>> >>
>>> >>
>>> >> Le 15/12/2010 ? 06:42, samir a ?crit :
>>> >>
>>> >> hello, I am trying call "pickup" for the incoming calls with intercept
>>> function but it do not seems to work, am i missing something ? Regards Sam
>>> >> ________________________________
>>> >> View this message in context: call pickup
>>> >> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From u2nsam at gmail.com  Tue Dec 28 12:11:07 2010
From: u2nsam at gmail.com (Sam)
Date: Tue, 28 Dec 2010 14:41:07 +0530
Subject: [Freeswitch-users] DTMF missing
In-Reply-To: <AANLkTik3N+F6Li6OL9EmyDH34eowxR+ttBFZx=1e83Eg@mail.gmail.com>
References: <AANLkTik=NLa+injL5hF=VsymwoE9wd1Wb2joXJDhvfFZ@mail.gmail.com>
	<AANLkTincAApO4JDwzt-5oX0iD2VozFNVCtembOumL4iS@mail.gmail.com>
	<AANLkTin14RN-WxkYzqkQPRZctYt09xkWguAoCr8eeUWe@mail.gmail.com>
	<AANLkTik3N+F6Li6OL9EmyDH34eowxR+ttBFZx=1e83Eg@mail.gmail.com>
Message-ID: <AANLkTikAKqXyFDQGTaBxW6OhS7pVzUzT_m3n7mZmAfxc@mail.gmail.com>

When i do " #1 " i cannot see the dtmf on the fs_cli , is it because of that
?

Regds
Sam

On Mon, Dec 27, 2010 at 1:26 PM, Sam <u2nsam at gmail.com> wrote:

> Hi ,
>
> I have now upgraded to ver (git-4e95227 2010-12-26 09-09-14 -0600) , now
> the dtmfs in IVR are working !
>
>
> When i used bind meta application then its not executing, any comments for
> me ?
>
>             <action application="set" data="export_vars=#,*"/>
>             <action application="export" data="#=true"/>
>             <action application="set" data="bind_meta_key=#"/>
>             <action application="bind_meta_app" data="1 ab s
> execute_extension::dx XML features"/>
>
>
>
>
> Regards
> Sam
>
>
>
>
> On Sun, Dec 26, 2010 at 11:16 PM, afshin afzali <a.afzali2003 at gmail.com>wrote:
>
>> Sam,
>>
>> Yes, The issue has resolved in last git :)
>> -- afshin
>>
>> On Sun, Dec 26, 2010 at 7:33 AM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Hello
>>>
>>> Just like Afshin, I am also missing DTMF digits ,using FreeSWITCH Version
>>> 1.0.head (git-34a0ca5 2010-12-22 20-38-57 -0600) ;
>>> should i upgrade to latest ?
>>>
>>> Regards
>>> Sam
>>>
>>>
>>>
>>> On Fri, Dec 24, 2010 at 4:10 PM, Sam <u2nsam at gmail.com> wrote:
>>>
>>>> Hi,,
>>>>
>>>>
>>>> I have installed the latest ver of freeswitch and i have configured the
>>>> conference.
>>>>
>>>> now when i punch in the digits for password , i could see that the DTMF
>>>> digits are missed on fs_cli.
>>>>
>>>> it only happens when i dial it from polycom or cisco phones.
>>>>
>>>> I have tried with and without these values below:-
>>>>
>>>> <param name="dtmf-type" value="rfc2833"/>
>>>>     <param name="pass-rfc2833" value="true"/>
>>>>
>>>>
>>>>
>>>> traces fetched:
>>>> 192.168.2.49:5060 -> 192.168.2.190:5060
>>>>   INVITE sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>SIP/2.0..Via: SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From:
>>>> "7028" <sip:7028 at 192.168.2.190 <sip%3A7028 at 192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To:
>>>> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>..Call-ID:
>>>> 0017592a-eb33001a-
>>>>   63da3294-1a7bfdfa at 192.168.2.49..Max-Forwards: 70..Date: Fri, 24 Dec
>>>> 2010 10:13:52 GMT..CSeq: 102 INVITE..User-Agent: Cisco-CP7940G/8.0..Contact:
>>>> <sip:7028 at 192.168.2.49:5060;transport=udp>..Proxy-Authorization: Digest
>>>> username="7028"
>>>>   ,realm="192.168.2.190",uri="sip:7050 at 192.168.2.190<sip%3A7050 at 192.168.2.190>",response="a668f5c480285b35e7ff6bcd446879f0",nonce="d2c540f2-8487-4d87-bdab-871585253eb8",cnonce="0a6c4176",qop=auth,nc=00000001,algorithm=MD5..Expires:
>>>> 180..Accept: application/sdp
>>>>   ..Allow:
>>>> ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported:
>>>> replaces,join,norefersub..Content-Length: 220..Content-Type:
>>>> application/sdp..Content-Disposition:
>>>> session;handling=optional....v=0..o=Cisco-SIPUA 16102
>>>>    0 IN IP4 192.168.2.49..s=SIP Call..t=0 0..m=audio 17298 RTP/AVP 0 8
>>>> 18..c=IN IP4 192.168.2.49..a=rtpmap:0 PCMU/8000..a=rtpmap:8
>>>> PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=sendrecv..
>>>>
>>>>
>>>> 192.168.2.190:5060 -> 192.168.2.49:5060
>>>>   SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From:
>>>> "7028" <sip:7028 at 192.168.2.190 <sip%3A7028 at 192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To:
>>>> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;tag=2XXUZpgr1rvgc..Call-ID:
>>>> 0017592a-eb33001a-63da3
>>>>   294-1a7bfdfa at 192.168.2.49..CSeq: 102 INVITE..Contact:
>>>> <sip:7050 at 192.168.2.190:5060;transport=udp>..User-Agent:
>>>> NOVANET..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
>>>> MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIF
>>>>   Y, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path,
>>>> replaces..Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>>> sla, include-session-description, presence.winfo, message-summary,
>>>> refer..Session-Expires: 180
>>>>   0;refresher=uas..Min-SE: 120..Content-Type:
>>>> application/sdp..Content-Disposition: session..Content-Length:
>>>> 249..Remote-Party-ID: "7050" <sip:7050 at 192.168.2.190<sip%3A7050 at 192.168.2.190>>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH
>>>> 1293163579 129
>>>>   3163580 IN IP4 192.168.2.190..s=FreeSWITCH..c=IN IP4
>>>> 192.168.2.190..t=0 0..m=audio 22050 RTP/AVP 0 101..a=rtpmap:0
>>>> PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
>>>> 0-16..a=silenceSupp:off - - - -..a=ptime:20..
>>>>
>>>> But the dtmf are not missed when punched on eyebeam softphone.
>>>>
>>>> And all the phones have RFC 2833.
>>>>
>>>> traces fetched for softphone:-
>>>>
>>>> 192.168.2.17:6182 -> 192.168.2.190:5060
>>>>   INVITE sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190> SIP/2.0..To:
>>>> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>..From: 7001<
>>>> sip:7001 at 192.168.2.190 <sip%3A7001 at 192.168.2.190>>;tag=6c557c1e..Via:
>>>> SIP/2.0/UDP 192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport..Call-ID:
>>>> d32ffe546570a77e..CS
>>>>   eq: 2 INVITE..Contact: <sip:7001 at 192.168.2.17:6182>..Max-Forwards:
>>>> 70..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>>>> SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Authorization: Digest
>>>> username="7001",rea
>>>>   lm="192.168.2.190",nonce="d0b4db6e-bd76-447f-9076-2e6b7809cb54",uri="
>>>> sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>",response="45fb9eb4f0e0e4fffd87a22769a007ba",cnonce="1c27f3687059b16d",nc=00000001,qop=auth,algorithm=MD5..User-Agent:
>>>> eyeBeam release 3007n
>>>>    stamp 17816..Content-Length: 233....v=0..o=- 27833664 27833670 IN IP4
>>>> 192.168.2.17..s=eyeBeam..c=IN IP4 192.168.2.17..t=0 0..m=audio 6398 RTP/AVP
>>>> 0 18 101..a=alt:1 1 : 2C830AD9 0000004F 192.168.2.17 6398..a=fmtp:101
>>>> 0-15..a=rtpmap:
>>>>   101 telephone-event/8000..a=sendrecv..
>>>>
>>>>
>>>> 192.168.2.190:5060 -> 192.168.2.17:6182
>>>>   SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport=6182..From:
>>>> 7001 <sip:7001 at 192.168.2.190 <sip%3A7001 at 192.168.2.190>>;tag=6c557c1e..To:
>>>> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;tag=H0ctQv7KNgU2j..Call-ID:
>>>> d32ffe546570a77e..C
>>>>   Seq: 2 INVITE..Contact: <sip:7050 at 192.168.2.190:5060;transport=udp>..User-Agent:
>>>> NOVANET..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
>>>> MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Support
>>>>   ed: timer, precondition, path, replaces..Allow-Events: talk, hold,
>>>> presence, dialog, line-seize, call-info, sla, include-session-description,
>>>> presence.winfo, message-summary, refer..Session-Expires:
>>>> 1800;refresher=uas..Min-SE: 120..
>>>>   Content-Type: application/sdp..Content-Disposition:
>>>> session..Content-Length: 249..Remote-Party-ID: "7050" <
>>>> sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH
>>>> 1293161640 1293161641 IN IP4 192.168.2.190..
>>>>   s=FreeSWITCH..c=IN IP4 192.168.2.190..t=0 0..m=audio 24852 RTP/AVP 0
>>>> 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
>>>> 0-16..a=silenceSupp:off - - - -..a=ptime:20..
>>>>
>>>>
>>>> Any thing you can think how it can happen?
>>>>
>>>>
>>>> Regards
>>>> Sam
>>>>
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From dujinfang at gmail.com  Tue Dec 28 16:08:39 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 28 Dec 2010 21:08:39 +0800
Subject: [Freeswitch-users] skypopen load problem
Message-ID: <AANLkTimY-L=gA65vsRdfLTM1BEaza8PjA98DoiWZpSiH@mail.gmail.com>

Hi,

So finally I get a chance to move to CentOS 64bit.

git HEAD on CentOS 5.5 64bit. skype 2.0.0.72 static.

Load error:

http://pastebin.freeswitch.org/14888

Seems sk_2 read extra AUTOAWAY OFF and ERROR 68, however, client.c
never shows those extra lines on my test against all my 20 instances.
I use multi-instances-same-username btw.

./client :302
Initialized XInitThreads!
PROTOCOL 7
#ciapalino PONG
CONNSTATUS ONLINE
CURRENTUSERHANDLE idapted_voip_10
USERSTATUS ONLINE


Sometimes I can load till sk_8, sometimes sk_1 doesn't load. Any clue?

Thanks.

-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From dujinfang at gmail.com  Tue Dec 28 16:12:37 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 28 Dec 2010 21:12:37 +0800
Subject: [Freeswitch-users] skypopen load problem
In-Reply-To: <AANLkTimY-L=gA65vsRdfLTM1BEaza8PjA98DoiWZpSiH@mail.gmail.com>
References: <AANLkTimY-L=gA65vsRdfLTM1BEaza8PjA98DoiWZpSiH@mail.gmail.com>
Message-ID: <AANLkTi=UEWRSjhf=8+jEAbyptaPP4Sp1NBk5yzP8Um4k@mail.gmail.com>

see also http://pastebin.freeswitch.org/14889

sorry flooding.

On Tue, Dec 28, 2010 at 9:08 PM, Seven Du <dujinfang at gmail.com> wrote:
> Hi,
>
> So finally I get a chance to move to CentOS 64bit.
>
> git HEAD on CentOS 5.5 64bit. skype 2.0.0.72 static.
>
> Load error:
>
> http://pastebin.freeswitch.org/14888
>
> Seems sk_2 read extra AUTOAWAY OFF and ERROR 68, however, client.c
> never shows those extra lines on my test against all my 20 instances.
> I use multi-instances-same-username btw.
>
> ./client :302
> Initialized XInitThreads!
> PROTOCOL 7
> #ciapalino PONG
> CONNSTATUS ONLINE
> CURRENTUSERHANDLE idapted_voip_10
> USERSTATUS ONLINE
>
>
> Sometimes I can load till sk_8, sometimes sk_1 doesn't load. Any clue?
>
> Thanks.
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:? http://www.freeswitch.org.cn
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From Nabble at slickdeals.endjunk.com  Tue Dec 28 17:01:11 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 28 Dec 2010 06:01:11 -0800 (PST)
Subject: [Freeswitch-users] Some outdated/obsolete modules?
Message-ID: <1293544871342-5871879.post@n2.nabble.com>


While porting and compiling FS git (2010/12/27) to an ARM platform, I noticed
the build/modules.conf.in has left out several modules, i.e.
applications/mod_fax, applications/mod_limit, and codecs/mod_voipcodecs.
After a little chat with stkn on the FS IRC channel, I learnt that these
modules are no longer in used and have been considered obsolete. If so, why
not start doing a housekeeping and remove the pertinent modules' source
files from the circulation? This way, the obsolete modules won't cause any
further confusion to new comers to FS git like it did to me. Just my 2?. 

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Some-outdated-obsolete-modules-tp5871879p5871879.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gmaruzz at gmail.com  Tue Dec 28 17:00:32 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 28 Dec 2010 15:00:32 +0100
Subject: [Freeswitch-users] skypopen load problem
In-Reply-To: <AANLkTi=UEWRSjhf=8+jEAbyptaPP4Sp1NBk5yzP8Um4k@mail.gmail.com>
References: <AANLkTimY-L=gA65vsRdfLTM1BEaza8PjA98DoiWZpSiH@mail.gmail.com>
	<AANLkTi=UEWRSjhf=8+jEAbyptaPP4Sp1NBk5yzP8Um4k@mail.gmail.com>
Message-ID: <AANLkTi=AXuqR0VvBCoPqAx3mLPrZ=GASF0=2fvEX-8G6@mail.gmail.com>

Ciao Seven,

is a pleasure to hear from you.

The "error 68" is that the skypeclient does not recognize-authorize
you. This is because (with all probability) it has not yet connected
to the skype p2p network. Give the client some time before loading
mod_skypopen. Skype network had "some problems" last days (hehehe).

If not because of network problems, can be because the files in the
skypeclient directory (where it keeps config data) has been trashed,
but if client.c goes through is not that, is the network problem.

Anyway, CentOS 64 bit is supersupported, actually is the main
reference platform (with Ubuntu LTS the second platform).

Also, as soon as you overcome this problem (and please write again on
this if it persists) I would like you (if you got time) to test the
new OSS driver (to be used with skypeclient static for oss 2.0.72, a
little bit easier to find around the net than the skypeclient static
for alsa 2.0.72). I believe it can give you more performances (but
mileage can vary). More on this will follow in the next days.

Ciao for now,

-giovanni

On Tue, Dec 28, 2010 at 2:12 PM, Seven Du <dujinfang at gmail.com> wrote:
> see also http://pastebin.freeswitch.org/14889
>
> sorry flooding.
>
> On Tue, Dec 28, 2010 at 9:08 PM, Seven Du <dujinfang at gmail.com> wrote:
>> Hi,
>>
>> So finally I get a chance to move to CentOS 64bit.
>>
>> git HEAD on CentOS 5.5 64bit. skype 2.0.0.72 static.
>>
>> Load error:
>>
>> http://pastebin.freeswitch.org/14888
>>
>> Seems sk_2 read extra AUTOAWAY OFF and ERROR 68, however, client.c
>> never shows those extra lines on my test against all my 20 instances.
>> I use multi-instances-same-username btw.
>>
>> ./client :302
>> Initialized XInitThreads!
>> PROTOCOL 7
>> #ciapalino PONG
>> CONNSTATUS ONLINE
>> CURRENTUSERHANDLE idapted_voip_10
>> USERSTATUS ONLINE
>>
>>
>> Sometimes I can load till sk_8, sometimes sk_1 doesn't load. Any clue?
>>
>> Thanks.
>>
>> --
>> About: http://about.me/dujinfang
>> Blog: http://www.dujinfang.com
>> Proj:? http://www.freeswitch.org.cn
>>
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:? http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From rupa at rupa.com  Tue Dec 28 17:14:49 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 28 Dec 2010 08:14:49 -0600
Subject: [Freeswitch-users] Some outdated/obsolete modules?
In-Reply-To: <1293544871342-5871879.post@n2.nabble.com>
References: <1293544871342-5871879.post@n2.nabble.com>
Message-ID: <AANLkTimuy485ivhmvRsSF3OqANPoDtNCQ57c=Wt6YGrJ@mail.gmail.com>

mod_limit provides compatibility with... well, the old mod_limit.  There is
a reason it still exists in tree.  It isn't compiled by default since I'd
prefer that people not use the compatibility shim if at all possible.

On Tue, Dec 28, 2010 at 8:01 AM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> While porting and compiling FS git (2010/12/27) to an ARM platform, I
> noticed
> the build/modules.conf.in has left out several modules, i.e.
> applications/mod_fax, applications/mod_limit, and codecs/mod_voipcodecs.
> After a little chat with stkn on the FS IRC channel, I learnt that these
> modules are no longer in used and have been considered obsolete. If so, why
> not start doing a housekeeping and remove the pertinent modules' source
> files from the circulation? This way, the obsolete modules won't cause any
> further confusion to new comers to FS git like it did to me. Just my 2?.
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Some-outdated-obsolete-modules-tp5871879p5871879.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From anthony.minessale at gmail.com  Tue Dec 28 17:26:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 28 Dec 2010 08:26:14 -0600
Subject: [Freeswitch-users] group_confirm - Question
In-Reply-To: <AANLkTikPXY7073P3=3Gto+ArfBY41TUxkikKRAkGe13A@mail.gmail.com>
References: <AANLkTikPXY7073P3=3Gto+ArfBY41TUxkikKRAkGe13A@mail.gmail.com>
Message-ID: <AANLkTi=p50Y8xaGVHNJ1xkD-Cb_BXn91xrEXbcgG7=au@mail.gmail.com>

Your syntax is wrong you put the group confirm vars inside the same. {} as
the ignore early media the {} encloses several comma sep name val pairs.
Review the wiki on originate syntax.
On Dec 26, 2010 2:39 PM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com> wrote:
> Hi All,
>
> Dial plan below use group_confrim, work for both internal and external
> (PSTN) Calls without any issues
>
> <action application="bridge" data="{ignore_early_media=true,*
>
group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1
> *}user/${dialed_extension}@
> ${domain_name},sofia/gateway/voipms/${user_data(${dialed_extension}@
${domain_name}
> var followme_number)}"/>
>
>
> -----
>
> But I would like to use the group_confirm feature for external PSTN (Calls
> only). The following Dial plan is not working and the console log below.
>
> <action application="bridge"
> data="{ignore_early_media=true}user/${dialed_extension}@${domain_name},{*
>
group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1
> *}sofia/gateway/voipms/${user_data(${dialed_extension}@${domain_name} var
> followme_number)}"/>
>
>
> *Console Log*
>
> 2010-12-26 15:23:21.796308 [ERR] switch_core_session.c:380 Could not
locate
> channel type {group_confirm_file=
> 2010-12-26 15:23:21.796308 [ERR] switch_ivr_originate.c:2614 Cannot create
> outgoing channel of type [{group_confirm_file=] cause:
> [CHAN_NOT_IMPLEMENTED]
> 2010-12-26 15:23:21.796308 [ERR] switch_core_session.c:380 Could not
locate
> channel type group_confirm_key=1}sofia
> 2010-12-26 15:23:21.796308 [ERR] switch_ivr_originate.c:2614 Cannot create
> outgoing channel of type [group_confirm_key=1}sofia] cause:
> [CHAN_NOT_IMPLEMENTED]
>
> --------
>
> Any help is appreciated.
>
> Thanks
> Lloyd
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From linux4michelle at tamay-dogan.net  Tue Dec 28 20:05:34 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Tue, 28 Dec 2010 18:05:34 +0100
Subject: [Freeswitch-users] [Semi-OT] MobiGater Pro in buy collective from
	Germany/France
Message-ID: <20101228170534.GJ25187@michelle1>

Hello *,

I am now in  contact  with  Eurodesign  which  is  the  Manufacturer  of
"MobiGater Pro".   Since hey have no disributor corrently in Germany and
France and I am in creation of a new OnlineStore, I a thinking on sellig
the MobiGater (all versions) in Germany and France.

Since I am not very finace-stable currently I like to offer  you  a  buy
collective for a exceptionel special price.  Minimum quantity is for  me
12pcs/Karton and any Waranties are handled by "itsystems at tdnet UG i.G.".

If you are interested please send a mail to
        <sales at itsystems.tamay-dogan.net>

or me directly.

Thanks, Greetings and nice Day/Evening
    Michelle Konzack

-- 
##################### Debian GNU/Linux Consultant ######################
   Development of Intranet and Embedded Systems with Debian GNU/Linux

itsystems at tdnet France EURL       itsystems at tdnet UG (limited liability)
Owner Michelle Konzack            Owner Michelle Konzack

Apt. 917 (homeoffice)
50, rue de Soultz                 Kinzigstra?e 17
67100 Strasbourg/France           77694 Kehl/Germany
Tel: +33-6-61925193 mobil         Tel: +49-177-9351947 mobil
Tel: +33-9-52705884 fix

<http://www.itsystems.tamay-dogan.net/>  <http://www.flexray4linux.org/>
<http://www.debian.tamay-dogan.net/>         <http://www.can4linux.org/>

Jabber linux4michelle at jabber.ccc.de

Linux-User #280138 with the Linux Counter, http://counter.li.org/
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From lloyd.aloysius at gmail.com  Tue Dec 28 20:06:20 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 28 Dec 2010 12:06:20 -0500
Subject: [Freeswitch-users] group_confirm - Question
In-Reply-To: <AANLkTi=p50Y8xaGVHNJ1xkD-Cb_BXn91xrEXbcgG7=au@mail.gmail.com>
References: <AANLkTikPXY7073P3=3Gto+ArfBY41TUxkikKRAkGe13A@mail.gmail.com>
	<AANLkTi=p50Y8xaGVHNJ1xkD-Cb_BXn91xrEXbcgG7=au@mail.gmail.com>
Message-ID: <AANLkTinoEHJUnvL+vWgJu_6pM1yOuQ8o6A8bF2uHE03Z@mail.gmail.com>

Anthony,

Thank you for the direction. The following syntax works

<action application="bridge"
data="{ignore_early_media=true}[leg_timeout=20]user/${dialed_extension}@
${domain_name}:_:{ignore_early_media=true,group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1}[leg_timeout=20]sofia/gateway/utel/${user_data(${dialed_extension}@${domain_name}
var followme_number)}"/>

I assume this is the right way of doing. Please let me know if this is
wrong.

Thanks
Lloyd


On Tue, Dec 28, 2010 at 9:26 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Your syntax is wrong you put the group confirm vars inside the same. {} as
> the ignore early media the {} encloses several comma sep name val pairs.
> Review the wiki on originate syntax.
> On Dec 26, 2010 2:39 PM, "Aloysius Lloyd" <lloyd.aloysius at gmail.com>
> wrote:
> > Hi All,
> >
> > Dial plan below use group_confrim, work for both internal and external
> > (PSTN) Calls without any issues
> >
> > <action application="bridge" data="{ignore_early_media=true,*
> >
> group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1
> > *}user/${dialed_extension}@
> > ${domain_name},sofia/gateway/voipms/${user_data(${dialed_extension}@
> ${domain_name}
> > var followme_number)}"/>
> >
> >
> > -----
> >
> > But I would like to use the group_confirm feature for external PSTN
> (Calls
> > only). The following Dial plan is not working and the console log below.
> >
> > <action application="bridge"
> > data="{ignore_early_media=true}user/${dialed_extension}@
> ${domain_name},{*
> >
> group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1
> > *}sofia/gateway/voipms/${user_data(${dialed_extension}@${domain_name}
> var
> > followme_number)}"/>
> >
> >
> > *Console Log*
> >
> > 2010-12-26 15:23:21.796308 [ERR] switch_core_session.c:380 Could not
> locate
> > channel type {group_confirm_file=
> > 2010-12-26 15:23:21.796308 [ERR] switch_ivr_originate.c:2614 Cannot
> create
> > outgoing channel of type [{group_confirm_file=] cause:
> > [CHAN_NOT_IMPLEMENTED]
> > 2010-12-26 15:23:21.796308 [ERR] switch_core_session.c:380 Could not
> locate
> > channel type group_confirm_key=1}sofia
> > 2010-12-26 15:23:21.796308 [ERR] switch_ivr_originate.c:2614 Cannot
> create
> > outgoing channel of type [group_confirm_key=1}sofia] cause:
> > [CHAN_NOT_IMPLEMENTED]
> >
> > --------
> >
> > Any help is appreciated.
> >
> > Thanks
> > Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Tue Dec 28 20:57:08 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 28 Dec 2010 12:57:08 -0500
Subject: [Freeswitch-users] voicemail operator-extension -
	Multi-tenantEnvironment
In-Reply-To: <AANLkTik=afD7Zjsb_jPw1aNSXsxLCtHeshiVbK-aHTbA@mail.gmail.com>
References: <AANLkTimm99xdzt2cW6zTpFZY+FtiRCaHCQSm=Rv__o4G@mail.gmail.com>
	<614459D4C10248C8B913E4422CFC7C35@e1705>
	<D330BE13-B0A1-4D76-9DFA-B80B7F3534D8@gmail.com>
	<AANLkTinATfPEbDkzfZSyXHgVYM0mjtsgvzk28xYBotXt@mail.gmail.com>
	<AANLkTik=afD7Zjsb_jPw1aNSXsxLCtHeshiVbK-aHTbA@mail.gmail.com>
Message-ID: <AANLkTikfETbZCi080u1uuLQj-VkezQf_2+=eWBUv5mZ+@mail.gmail.com>

Michael,

Thank you for the suggestion but this is not working . I think because of
the the  <param name="callback-context" value="default"/>

But in the voicemail.conf.xml there is one parameter like below.

 *<param name="email-from" value="${voicemail_account}@
${voicemail_domain}"/>*

The above working with multiple domain without any issue. why the param
name="operator-extension" does not understand the variable ${domain_name} or
${voicemail_domain}

Any help is appreciated.

Thanks and regards,
Lloyd

On Tue, Dec 28, 2010 at 2:51 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Dec 27, 2010 at 8:08 PM, Aloysius Lloyd <
> lloyd.aloysius at sunteltech.ca> wrote:
>
>> $${domain} - incorrect
>>
>> It is already set when the Chanel is created. here is the dialplan when I
>> reach to voicemail box already belong to a domain.
>>
>> <action application="bridge" data="loopback/app=voicemail:default
>> ${domain_name} ${dialed_extension}"/>
>>
>> when is press 9 - I am already in the domain extension voice mail box so
>> there is ${domain_name} variable have the value.
>>
>> looks to me voicemail application does not understand the ${domain_name}
>>
>
> This is almost a true statement. In actuality, you don't put chan vars in
> params like this and have them interpolated with each call. Interpolation of
> channel variables like this is best done in the dialplan. The easiest way
> for you to do this would be to set the value back to the default of:
>  <param name="operator-extension" value="operator XML default"/>
> ...and then create the "operator" extension which handles the transfer:
> <extension name="operator transfer">
>   <condition field="destination_number" expression="^(operator)$">
>     <!-- ${domain_name} should be available here -->
>     <action application="transfer" data="operator XML ${domain_name}"/>
>   </condition>
> </extension>
>
> Just be sure that you don't have a domain name "default" or this will blow
> up. :) You could also put this "utility" operator extension in the
> 'features' context and then as long as you didn't have a domain named
> 'features' then you would be just fine.
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Nabble at slickdeals.endjunk.com  Tue Dec 28 21:25:17 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 28 Dec 2010 10:25:17 -0800 (PST)
Subject: [Freeswitch-users] Cross-compilation crash on
	src/mod/languages/mod_perl
Message-ID: <1293560717568-5872539.post@n2.nabble.com>


I ran into this  http://pastebin.com/1JNVfwM1 problem  when trying to
cross-compile FS git (2010/12/27) src/mod/languages/mod_perl for an ARM
platform. The error (line# 6) indicates the compilation was trying to use
host perl and its library. Can FS developers please take a look and see if
that can be resolved? Thanks.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Cross-compilation-crash-on-src-mod-languages-mod-perl-tp5872539p5872539.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From lloyd.aloysius at sunteltech.ca  Tue Dec 28 21:50:26 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Tue, 28 Dec 2010 13:50:26 -0500
Subject: [Freeswitch-users] CentOS 5.5 - FreeSWITCH - Audio Quality
	issues
In-Reply-To: <AANLkTi=6shTJg0C2eFsar+mdjstT330e2kaUtFxsPQ1F@mail.gmail.com>
References: <AANLkTik3y8EfTi5s_a_1BURnUJe8Nv0p+5y8Ph6Jkcg3@mail.gmail.com>
	<AANLkTi=6shTJg0C2eFsar+mdjstT330e2kaUtFxsPQ1F@mail.gmail.com>
Message-ID: <AANLkTi=3Yi_Phq6Zv2Ep-YEgf=3orNmOEpLVyww=7x=L@mail.gmail.com>

MOH - playing without any issue. I do not think Disk activity is a issue
here.

Box running Intel Core 2 Quad 2.4GHz + 4GB Ram - CentOS FreeSWITCH

IVR - Start low volume then increase ... sometime high volume then go to low
volume
Voice mails - Noice, breaking voices etc ...

Any help is appreciated.

Thanks
Lloyd


On Thu, Dec 23, 2010 at 7:27 PM, Michael Collins <msc at freeswitch.org> wrote:

> Just a casual observation, but it looks like all three of these symptoms
> have disk I/O in common. Perhaps your disk performance isn't up to snuff?
> You could probably test this by moving some of your sound files into a
> ramdisk and then calling in to listen to them...
>
> -MC
>
> On Thu, Dec 23, 2010 at 1:49 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Hi All,
>>
>> *FreeSWITCH environment*
>>
>> Hardware - Intel Core 2 Quad 2.4GHz , 4GB RAM
>> OS - CentOS 5.5
>> FreeSWITCH Version 1.0.head (git-fcd6c54 2010-12-19 00-13-08 -0500)
>> Ethernet : Public IP + 10Mbps port
>>
>>
>> Calls Between Phones and outside calls perfect.
>>
>>
>> But Voice Quality is not great in the following scenarios
>>
>> 1. Voice Mails ... most of the voicemails breaking voices .. lots of
>> static
>> 2. IVR ... breaking voices
>> 3. System Automatically Hangup ... while people leaving messages..
>>
>>
>>
>> Do I need to adjust any parameters ?
>>
>> Any help is appreciated.
>>
>> Thanks
>> Lloyd
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Dec 28 21:54:39 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 28 Dec 2010 12:54:39 -0600
Subject: [Freeswitch-users] CentOS 5.5 - FreeSWITCH - Audio Quality
	issues
In-Reply-To: <AANLkTi=3Yi_Phq6Zv2Ep-YEgf=3orNmOEpLVyww=7x=L@mail.gmail.com>
References: <AANLkTik3y8EfTi5s_a_1BURnUJe8Nv0p+5y8Ph6Jkcg3@mail.gmail.com>
	<AANLkTi=6shTJg0C2eFsar+mdjstT330e2kaUtFxsPQ1F@mail.gmail.com>
	<AANLkTi=3Yi_Phq6Zv2Ep-YEgf=3orNmOEpLVyww=7x=L@mail.gmail.com>
Message-ID: <316084A7-BCA6-4490-8990-9894331F999F@freeswitch.org>

I run on a Intel(R) Core(TM)2 Quad  CPU   Q8200 ... its fine with 5.5 so what exactly are you running on?  And where are these calls coming from? codec? sample rate and such?

/b

On Dec 28, 2010, at 12:50 PM, Aloysius Lloyd wrote:

> 
> MOH - playing without any issue. I do not think Disk activity is a issue here. 
> 
> Box running Intel Core 2 Quad 2.4GHz + 4GB Ram - CentOS FreeSWITCH 
> 
> IVR - Start low volume then increase ... sometime high volume then go to low volume
> Voice mails - Noice, breaking voices etc ...
> 
> Any help is appreciated.
> 
> Thanks
> Lloyd
> 

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From lloyd.aloysius at gmail.com  Tue Dec 28 22:05:19 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 28 Dec 2010 14:05:19 -0500
Subject: [Freeswitch-users] CentOS 5.5 - FreeSWITCH - Audio Quality
	issues
In-Reply-To: <316084A7-BCA6-4490-8990-9894331F999F@freeswitch.org>
References: <AANLkTik3y8EfTi5s_a_1BURnUJe8Nv0p+5y8Ph6Jkcg3@mail.gmail.com>
	<AANLkTi=6shTJg0C2eFsar+mdjstT330e2kaUtFxsPQ1F@mail.gmail.com>
	<AANLkTi=3Yi_Phq6Zv2Ep-YEgf=3orNmOEpLVyww=7x=L@mail.gmail.com>
	<316084A7-BCA6-4490-8990-9894331F999F@freeswitch.org>
Message-ID: <AANLkTimREKiD=3t12WT7fMWWR7JPyOT+3S4jm7SPY6UT@mail.gmail.com>

Brian,

Here is information about the box and codec

processor       : 2
vendor_id       : GenuineIntel
cpu family      : 6
model           : 15
model name      : Intel(R) Core(TM)2 Quad CPU    Q6600  @ 2.40GHz
stepping        : 11
cpu MHz         : 2394.057
cache size      : 4096 KB
physical id     : 0
siblings        : 4
core id         : 2
cpu cores       : 4
apicid          : 2
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca
cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm
constant_tsc pni monitor ds_cpl vmx est tm2 ssse3 cx16 xtpr lahf_lm
bogomips        : 4787.33

---

Applications

/usr/local/freeswitch/bin/freeswitch -nc
mysql
Apache

Apache and mysql use only for xml_cdr. There is no other applications.

--

top - 14:01:32 up 8 days, 12:57,  2 users,  load average: 0.10, 0.05, 0.01
Tasks: 104 total,   1 running, 102 sleeping,   0 stopped,   1 zombie
Cpu(s):  0.0%us,  0.1%sy,  0.0%ni, 99.9%id,  0.0%wa,  0.0%hi,  0.0%si,
 0.0%st
Mem:   4137172k total,   730956k used,  3406216k free,   164412k buffers
Swap:  4192944k total,        0k used,  4192944k free,   463228k cached

---

             total       used       free     shared    buffers     cached
Mem:          4040        713       3326          0        160        452
-/+ buffers/cache:        101       3939
Swap:         4094          0       4094

--

v=0
o=FreeSWITCH 1293532168 1293532169 IN IP4 209.172.34.154
s=FreeSWITCH
c=IN IP4 209.172.34.154
t=0 0
m=audio 30754 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

FreeSWITCH Version 1.0.head (git-fcd6c54 2010-12-19 00-13-08 -0500)

Thanks for your help.

Lloyd


On Tue, Dec 28, 2010 at 1:54 PM, Brian West <brian at freeswitch.org> wrote:

> I run on a Intel(R) Core(TM)2 Quad  CPU   Q8200 ... its fine with 5.5 so
> what exactly are you running on?  And where are these calls coming from?
> codec? sample rate and such?
>
> /b
>
> On Dec 28, 2010, at 12:50 PM, Aloysius Lloyd wrote:
>
>
> MOH - playing without any issue. I do not think Disk activity is a issue
> here.
>
> Box running Intel Core 2 Quad 2.4GHz + 4GB Ram - CentOS FreeSWITCH
>
> IVR - Start low volume then increase ... sometime high volume then go to
> low volume
> Voice mails - Noice, breaking voices etc ...
>
> Any help is appreciated.
>
> Thanks
> Lloyd
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue Dec 28 23:02:01 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 28 Dec 2010 15:02:01 -0500
Subject: [Freeswitch-users] voicemail operator-extension
	-Multi-tenantEnvironment
References: <AANLkTimm99xdzt2cW6zTpFZY+FtiRCaHCQSm=Rv__o4G@mail.gmail.com><614459D4C10248C8B913E4422CFC7C35@e1705><D330BE13-B0A1-4D76-9DFA-B80B7F3534D8@gmail.com><AANLkTinATfPEbDkzfZSyXHgVYM0mjtsgvzk28xYBotXt@mail.gmail.com><AANLkTik=afD7Zjsb_jPw1aNSXsxLCtHeshiVbK-aHTbA@mail.gmail.com>
	<AANLkTikfETbZCi080u1uuLQj-VkezQf_2+=eWBUv5mZ+@mail.gmail.com>
Message-ID: <FE2ED4566C464422A885CE886E991F10@e1705>

and if you set it inline="true" a domain_name var ?
  ----- Original Message ----- 
  From: Aloysius Lloyd 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, December 28, 2010 12:57 PM
  Subject: Re: [Freeswitch-users] voicemail operator-extension -Multi-tenantEnvironment


  Michael,


  Thank you for the suggestion but this is not working . I think because of the the  <param name="callback-context" value="default"/>


  But in the voicemail.conf.xml there is one parameter like below. 


   <param name="email-from" value="${voicemail_account}@${voicemail_domain}"/>


  The above working with multiple domain without any issue. why the param name="operator-extension" does not understand the variable ${domain_name} or ${voicemail_domain}


  Any help is appreciated.


  Thanks and regards,
  Lloyd


  On Tue, Dec 28, 2010 at 2:51 AM, Michael Collins <msc at freeswitch.org> wrote:




    On Mon, Dec 27, 2010 at 8:08 PM, Aloysius Lloyd <lloyd.aloysius at sunteltech.ca> wrote:

      $${domain} - incorrect


      It is already set when the Chanel is created. here is the dialplan when I reach to voicemail box already belong to a domain.


      <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>


      when is press 9 - I am already in the domain extension voice mail box so there is ${domain_name} variable have the value.


      looks to me voicemail application does not understand the ${domain_name}


    This is almost a true statement. In actuality, you don't put chan vars in params like this and have them interpolated with each call. Interpolation of channel variables like this is best done in the dialplan. The easiest way for you to do this would be to set the value back to the default of:
     <param name="operator-extension" value="operator XML default"/>
    ...and then create the "operator" extension which handles the transfer:
    <extension name="operator transfer">
      <condition field="destination_number" expression="^(operator)$">
        <!-- ${domain_name} should be available here -->
        <action application="transfer" data="operator XML ${domain_name}"/>
      </condition>
    </extension>


    Just be sure that you don't have a domain name "default" or this will blow up. :) You could also put this "utility" operator extension in the 'features' context and then as long as you didn't have a domain named 'features' then you would be just fine.


    -MC



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






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


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From brian at freeswitch.org  Tue Dec 28 23:08:37 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 28 Dec 2010 14:08:37 -0600
Subject: [Freeswitch-users] voicemail operator-extension
	-Multi-tenantEnvironment
In-Reply-To: <FE2ED4566C464422A885CE886E991F10@e1705>
References: <AANLkTimm99xdzt2cW6zTpFZY+FtiRCaHCQSm=Rv__o4G@mail.gmail.com><614459D4C10248C8B913E4422CFC7C35@e1705><D330BE13-B0A1-4D76-9DFA-B80B7F3534D8@gmail.com><AANLkTinATfPEbDkzfZSyXHgVYM0mjtsgvzk28xYBotXt@mail.gmail.com><AANLkTik=afD7Zjsb_jPw1aNSXsxLCtHeshiVbK-aHTbA@mail.gmail.com>
	<AANLkTikfETbZCi080u1uuLQj-VkezQf_2+=eWBUv5mZ+@mail.gmail.com>
	<FE2ED4566C464422A885CE886E991F10@e1705>
Message-ID: <9D544C24-66A6-4BB5-9FF8-D1FCA8B08F5F@freeswitch.org>

xml_curl.

/b

On Dec 28, 2010, at 2:02 PM, Madovsky wrote:

> and if you set it inline="true" a domain_name var ?
> ----- Original Message -----
> From: Aloysius Lloyd
> To: FreeSWITCH Users Help
> Sent: Tuesday, December 28, 2010 12:57 PM
> Subject: Re: [Freeswitch-users] voicemail operator-extension -Multi-tenantEnvironment
> 
> Michael,
> 
> Thank you for the suggestion but this is not working . I think because of the the  <param name="callback-context" value="default"/>
> 
> But in the voicemail.conf.xml there is one parameter like below. 
> 
>  <param name="email-from" value="${voicemail_account}@${voicemail_domain}"/>

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From Nabble at slickdeals.endjunk.com  Wed Dec 29 00:04:53 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 28 Dec 2010 13:04:53 -0800 (PST)
Subject: [Freeswitch-users] spindermonkey: Host execution of a
 cross-compilation of libs/js/nsprpub/config/nsinstall.c
Message-ID: <1293570293681-5872965.post@n2.nabble.com>


Here goes again. When I tried to cross-compile src/mod/languages/spidermonkey
module for an ARM platform, the compilation crashes as shown 
http://pastebin.com/Qc0g8jQ5 here  (see line # 16) when it tries to execute
the newly compiled libs/js/nsprpub/config/nsinstall file (a cross-compiled
binary code for an ARM platform). This looks similar to this 
http://freeswitch-users.2379917.n2.nabble.com/libg722-1-Cross-Compilation-of-src-make-dct4-tables-c-td5867008.html
case  (libg722_1: Cross Compilation of src/make_dct4_tables.c). I hope FS
developers will be able to fix this easily with patches. Thanks.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/spindermonkey-Host-execution-of-a-cross-compilation-of-libs-js-nsprpub-config-nsinstall-c-tp5872965p5872965.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From moises.silva at gmail.com  Wed Dec 29 00:18:14 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 28 Dec 2010 16:18:14 -0500
Subject: [Freeswitch-users] FreeTDM hangs on Partial PRI
In-Reply-To: <4D18CBB9.5080909@gmail.com>
References: <4D18CBB9.5080909@gmail.com>
Message-ID: <AANLkTimNVn4X-_TGKJCbkuY0X7ORgQ0MBE04UpPGyqbZ@mail.gmail.com>

On Mon, Dec 27, 2010 at 12:24 PM, Phone <phone.bytes at gmail.com> wrote:

> We are using
>
> FreeSWITCH Version 1.0.head (git-8825b6e 2010-11-28 17-15-39 -0500)
>
> WANPIPE Release: 3.5.14
>
> Hardware is Sangoma A101
>
> Partial PRI D-chan=24 B-chans=16-23
>
> We are having trouble with the Partial PRI (channel 16) getting stuck in
> a state where it cannot take calls.  This happens intermittently.  The
> system may run several days or even a week or two without having the
> problem.  It seems that the channel is getting stuck in some odd state
> when trying to hang up.
>
> To resolve this, we unload the FreeTDM mod, restarting Wanrouter, and
> then reload the FreeTDM mod.  This clears the channels and allows the
> system to take calls normally again.
>
> In testing we have also observed the following conditions:
>
> 1. Place a call in.  Cannot complete. Hang up.
> 2. Place a call in.  It completes and works correctly.
> 3. Place a call in.  Cannot complete.  Hang up.
> 4. Place a call in.  It completes and works correctly.
>
> The above pattern can be repeated multiple times, with the same result.
>
>
Are you aware sangoma_boost module is deprecated? you should be using
ftmod_sangoma_isdn

When the channel gets stuck you can do ftdm dump <span> <channel> and
pastebin the output (there should be a state history at the bottom that
helps to fix this type of problems).

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com
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From brian at freeswitch.org  Wed Dec 29 00:22:09 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 28 Dec 2010 15:22:09 -0600
Subject: [Freeswitch-users] FreeTDM hangs on Partial PRI
In-Reply-To: <AANLkTimNVn4X-_TGKJCbkuY0X7ORgQ0MBE04UpPGyqbZ@mail.gmail.com>
References: <4D18CBB9.5080909@gmail.com>
	<AANLkTimNVn4X-_TGKJCbkuY0X7ORgQ0MBE04UpPGyqbZ@mail.gmail.com>
Message-ID: <68B6C94C-5C6B-4839-948E-E8701B79CABB@freeswitch.org>

I have never once seen a T1 configured in this manner.   I have seen 24 ad a D channel and 1-7 as B channels but this is just odd.

/b

On Dec 28, 2010, at 3:18 PM, Moises Silva wrote:

> 
> Partial PRI D-chan=24 B-chans=16-23
> 
> We are having trouble with the Partial PRI (channel 16) getting stuck in
> a state where it cannot take calls.  This happens intermittently.  The
> system may run several days or even a week or two without having the
> problem.  It seems that the channel is getting stuck in some odd state
> when trying to hang up.
> 
> To resolve this, we unload the FreeTDM mod, restarting Wanrouter, and
> then reload the FreeTDM mod.  This clears the channels and allows the
> system to take calls normally again.
> 
> In testing we have also observed the following conditions:
> 
> 1. Place a call in.  Cannot complete. Hang up.
> 2. Place a call in.  It completes and works correctly.
> 3. Place a call in.  Cannot complete.  Hang up.
> 4. Place a call in.  It completes and works correctly.
> 
> The above pattern can be repeated multiple times, with the same result.
> 
> 
> Are you aware sangoma_boost module is deprecated? you should be using ftmod_sangoma_isdn
> 
> When the channel gets stuck you can do ftdm dump <span> <channel> and pastebin the output (there should be a state history at the bottom that helps to fix this type of problems).
> 
> Moises Silva

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From msc at freeswitch.org  Wed Dec 29 00:29:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 28 Dec 2010 13:29:32 -0800
Subject: [Freeswitch-users] voicemail operator-extension -
	Multi-tenantEnvironment
In-Reply-To: <AANLkTikfETbZCi080u1uuLQj-VkezQf_2+=eWBUv5mZ+@mail.gmail.com>
References: <AANLkTimm99xdzt2cW6zTpFZY+FtiRCaHCQSm=Rv__o4G@mail.gmail.com>
	<614459D4C10248C8B913E4422CFC7C35@e1705>
	<D330BE13-B0A1-4D76-9DFA-B80B7F3534D8@gmail.com>
	<AANLkTinATfPEbDkzfZSyXHgVYM0mjtsgvzk28xYBotXt@mail.gmail.com>
	<AANLkTik=afD7Zjsb_jPw1aNSXsxLCtHeshiVbK-aHTbA@mail.gmail.com>
	<AANLkTikfETbZCi080u1uuLQj-VkezQf_2+=eWBUv5mZ+@mail.gmail.com>
Message-ID: <AANLkTimbpDJ3QrkfzvVygtSA1s-KmpV_f8TKYU4RLgf7@mail.gmail.com>

On Tue, Dec 28, 2010 at 9:57 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Michael,
>
> Thank you for the suggestion but this is not working . I think because of
> the the  <param name="callback-context" value="default"/>
>

Please define "not working" - either you press 9 and the call is x-fer'd or
it is not. Once it is x-fer'd to the dialplan you should be able to do
whatever you want with the call. Or do what bkw says and use xml_curl.
-MC
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From msc at freeswitch.org  Wed Dec 29 00:43:29 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 28 Dec 2010 13:43:29 -0800
Subject: [Freeswitch-users]
	=?windows-1252?q?ANN_=3A_PySWITCH_Release_=96_?=
	=?windows-1252?q?0=2E1alpha?=
In-Reply-To: <AANLkTi=FDcobUchZZsq8uWzfSe99MgTHW5FvZV=d2kzM@mail.gmail.com>
References: <AANLkTi=FDcobUchZZsq8uWzfSe99MgTHW5FvZV=d2kzM@mail.gmail.com>
Message-ID: <AANLkTikGS5wK3fzArXVCiZOFN8HKKyWFLfL1gUstWZx7@mail.gmail.com>

Thanks for the heads up. Perhaps in the near future you could call in to our
weekly FS conference and give us a demonstration of your API in action.
-MC

On Mon, Dec 27, 2010 at 10:51 PM, Godson Gera <godson.g at gmail.com> wrote:

>
> Hi All,
>
> I am glad to announce the first alpha release of PySWITCH.
>
> http://pyswitch.sf.net
>
>
> The idea of PySWITCH is to offer a complete library to Python and Twisted
> programmers for interacting with FreeSWITCH using EventSocket interface. The
> target is to cover all FreeSWITCH API commands and Dialplan tools. PySWITCH
> handles all the low level details in executing FreeSWITCH commands, so the
> programmer can easily concentrate on quickly building FreeSWITCH
> applications. As an example, the API functions offered by PySWITCH often
> executes many FreeSWITCH commands under the hood and finally returns the
> desired result. Suppose you execute a background job, PySWITCH API will
> automatically wait and catch the backgroundjob event parse the result and
> will fire the deferred.
>
> The current release covers good amount of API commands and a few Dialplan
> tools. The protocol communication issues are ironed out. It has a nice event
> call back interface. I?ll present its usage in couple of tutorials soon.
>
>
> --
> Thanks & Regards,
> Godson Gera
> http://godson.in
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Dec 29 00:51:43 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 28 Dec 2010 13:51:43 -0800
Subject: [Freeswitch-users] DTMF missing
In-Reply-To: <AANLkTikAKqXyFDQGTaBxW6OhS7pVzUzT_m3n7mZmAfxc@mail.gmail.com>
References: <AANLkTik=NLa+injL5hF=VsymwoE9wd1Wb2joXJDhvfFZ@mail.gmail.com>
	<AANLkTincAApO4JDwzt-5oX0iD2VozFNVCtembOumL4iS@mail.gmail.com>
	<AANLkTin14RN-WxkYzqkQPRZctYt09xkWguAoCr8eeUWe@mail.gmail.com>
	<AANLkTik3N+F6Li6OL9EmyDH34eowxR+ttBFZx=1e83Eg@mail.gmail.com>
	<AANLkTikAKqXyFDQGTaBxW6OhS7pVzUzT_m3n7mZmAfxc@mail.gmail.com>
Message-ID: <AANLkTi=3N59_nwtiej=Xbtiodp95U_FYUpOPamN=fzYH@mail.gmail.com>

Assuming your console in on DEBUG level output then yes, you should see the
DTMFs show up on the screen. If you're not seeing DTMFs then yeah, something
is wrong. Confirm whether the client is sending the DTMFs inband or not.

-MC

On Tue, Dec 28, 2010 at 1:11 AM, Sam <u2nsam at gmail.com> wrote:

> When i do " #1 " i cannot see the dtmf on the fs_cli , is it because of
> that ?
>
> Regds
> Sam
>
>
> On Mon, Dec 27, 2010 at 1:26 PM, Sam <u2nsam at gmail.com> wrote:
>
>> Hi ,
>>
>> I have now upgraded to ver (git-4e95227 2010-12-26 09-09-14 -0600) , now
>> the dtmfs in IVR are working !
>>
>>
>> When i used bind meta application then its not executing, any comments for
>> me ?
>>
>>             <action application="set" data="export_vars=#,*"/>
>>             <action application="export" data="#=true"/>
>>             <action application="set" data="bind_meta_key=#"/>
>>             <action application="bind_meta_app" data="1 ab s
>> execute_extension::dx XML features"/>
>>
>>
>>
>>
>> Regards
>> Sam
>>
>>
>>
>>
>> On Sun, Dec 26, 2010 at 11:16 PM, afshin afzali <a.afzali2003 at gmail.com>wrote:
>>
>>> Sam,
>>>
>>> Yes, The issue has resolved in last git :)
>>> -- afshin
>>>
>>> On Sun, Dec 26, 2010 at 7:33 AM, Sam <u2nsam at gmail.com> wrote:
>>>
>>>> Hello
>>>>
>>>> Just like Afshin, I am also missing DTMF digits ,using FreeSWITCH
>>>> Version 1.0.head (git-34a0ca5 2010-12-22 20-38-57 -0600) ;
>>>> should i upgrade to latest ?
>>>>
>>>> Regards
>>>> Sam
>>>>
>>>>
>>>>
>>>> On Fri, Dec 24, 2010 at 4:10 PM, Sam <u2nsam at gmail.com> wrote:
>>>>
>>>>> Hi,,
>>>>>
>>>>>
>>>>> I have installed the latest ver of freeswitch and i have configured the
>>>>> conference.
>>>>>
>>>>> now when i punch in the digits for password , i could see that the DTMF
>>>>> digits are missed on fs_cli.
>>>>>
>>>>> it only happens when i dial it from polycom or cisco phones.
>>>>>
>>>>> I have tried with and without these values below:-
>>>>>
>>>>> <param name="dtmf-type" value="rfc2833"/>
>>>>>     <param name="pass-rfc2833" value="true"/>
>>>>>
>>>>>
>>>>>
>>>>> traces fetched:
>>>>> 192.168.2.49:5060 -> 192.168.2.190:5060
>>>>>   INVITE sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>SIP/2.0..Via: SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From:
>>>>> "7028" <sip:7028 at 192.168.2.190 <sip%3A7028 at 192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To:
>>>>> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>..Call-ID:
>>>>> 0017592a-eb33001a-
>>>>>   63da3294-1a7bfdfa at 192.168.2.49..Max-Forwards: 70..Date: Fri, 24 Dec
>>>>> 2010 10:13:52 GMT..CSeq: 102 INVITE..User-Agent: Cisco-CP7940G/8.0..Contact:
>>>>> <sip:7028 at 192.168.2.49:5060;transport=udp>..Proxy-Authorization:
>>>>> Digest username="7028"
>>>>>   ,realm="192.168.2.190",uri="sip:7050 at 192.168.2.190<sip%3A7050 at 192.168.2.190>",response="a668f5c480285b35e7ff6bcd446879f0",nonce="d2c540f2-8487-4d87-bdab-871585253eb8",cnonce="0a6c4176",qop=auth,nc=00000001,algorithm=MD5..Expires:
>>>>> 180..Accept: application/sdp
>>>>>   ..Allow:
>>>>> ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported:
>>>>> replaces,join,norefersub..Content-Length: 220..Content-Type:
>>>>> application/sdp..Content-Disposition:
>>>>> session;handling=optional....v=0..o=Cisco-SIPUA 16102
>>>>>    0 IN IP4 192.168.2.49..s=SIP Call..t=0 0..m=audio 17298 RTP/AVP 0 8
>>>>> 18..c=IN IP4 192.168.2.49..a=rtpmap:0 PCMU/8000..a=rtpmap:8
>>>>> PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=sendrecv..
>>>>>
>>>>>
>>>>> 192.168.2.190:5060 -> 192.168.2.49:5060
>>>>>   SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From:
>>>>> "7028" <sip:7028 at 192.168.2.190 <sip%3A7028 at 192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To:
>>>>> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;tag=2XXUZpgr1rvgc..Call-ID:
>>>>> 0017592a-eb33001a-63da3
>>>>>   294-1a7bfdfa at 192.168.2.49..CSeq: 102 INVITE..Contact:
>>>>> <sip:7050 at 192.168.2.190:5060;transport=udp>..User-Agent:
>>>>> NOVANET..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
>>>>> MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIF
>>>>>   Y, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path,
>>>>> replaces..Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>>>> sla, include-session-description, presence.winfo, message-summary,
>>>>> refer..Session-Expires: 180
>>>>>   0;refresher=uas..Min-SE: 120..Content-Type:
>>>>> application/sdp..Content-Disposition: session..Content-Length:
>>>>> 249..Remote-Party-ID: "7050" <sip:7050 at 192.168.2.190<sip%3A7050 at 192.168.2.190>>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH
>>>>> 1293163579 129
>>>>>   3163580 IN IP4 192.168.2.190..s=FreeSWITCH..c=IN IP4
>>>>> 192.168.2.190..t=0 0..m=audio 22050 RTP/AVP 0 101..a=rtpmap:0
>>>>> PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
>>>>> 0-16..a=silenceSupp:off - - - -..a=ptime:20..
>>>>>
>>>>> But the dtmf are not missed when punched on eyebeam softphone.
>>>>>
>>>>> And all the phones have RFC 2833.
>>>>>
>>>>> traces fetched for softphone:-
>>>>>
>>>>> 192.168.2.17:6182 -> 192.168.2.190:5060
>>>>>   INVITE sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>SIP/2.0..To: <
>>>>> sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>..From: 7001<
>>>>> sip:7001 at 192.168.2.190 <sip%3A7001 at 192.168.2.190>>;tag=6c557c1e..Via:
>>>>> SIP/2.0/UDP 192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport..Call-ID:
>>>>> d32ffe546570a77e..CS
>>>>>   eq: 2 INVITE..Contact: <sip:7001 at 192.168.2.17:6182>..Max-Forwards:
>>>>> 70..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>>>>> SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Authorization: Digest
>>>>> username="7001",rea
>>>>>   lm="192.168.2.190",nonce="d0b4db6e-bd76-447f-9076-2e6b7809cb54",uri="
>>>>> sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>",response="45fb9eb4f0e0e4fffd87a22769a007ba",cnonce="1c27f3687059b16d",nc=00000001,qop=auth,algorithm=MD5..User-Agent:
>>>>> eyeBeam release 3007n
>>>>>    stamp 17816..Content-Length: 233....v=0..o=- 27833664 27833670 IN
>>>>> IP4 192.168.2.17..s=eyeBeam..c=IN IP4 192.168.2.17..t=0 0..m=audio 6398
>>>>> RTP/AVP 0 18 101..a=alt:1 1 : 2C830AD9 0000004F 192.168.2.17
>>>>> 6398..a=fmtp:101 0-15..a=rtpmap:
>>>>>   101 telephone-event/8000..a=sendrecv..
>>>>>
>>>>>
>>>>> 192.168.2.190:5060 -> 192.168.2.17:6182
>>>>>   SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport=6182..From:
>>>>> 7001 <sip:7001 at 192.168.2.190 <sip%3A7001 at 192.168.2.190>>;tag=6c557c1e..To:
>>>>> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;tag=H0ctQv7KNgU2j..Call-ID:
>>>>> d32ffe546570a77e..C
>>>>>   Seq: 2 INVITE..Contact: <sip:7050 at 192.168.2.190:5060;transport=udp>..User-Agent:
>>>>> NOVANET..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
>>>>> MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Support
>>>>>   ed: timer, precondition, path, replaces..Allow-Events: talk, hold,
>>>>> presence, dialog, line-seize, call-info, sla, include-session-description,
>>>>> presence.winfo, message-summary, refer..Session-Expires:
>>>>> 1800;refresher=uas..Min-SE: 120..
>>>>>   Content-Type: application/sdp..Content-Disposition:
>>>>> session..Content-Length: 249..Remote-Party-ID: "7050" <
>>>>> sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH
>>>>> 1293161640 1293161641 IN IP4 192.168.2.190..
>>>>>   s=FreeSWITCH..c=IN IP4 192.168.2.190..t=0 0..m=audio 24852 RTP/AVP 0
>>>>> 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
>>>>> 0-16..a=silenceSupp:off - - - -..a=ptime:20..
>>>>>
>>>>>
>>>>> Any thing you can think how it can happen?
>>>>>
>>>>>
>>>>> Regards
>>>>> Sam
>>>>>
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From george.niculae79 at gmail.com  Wed Dec 29 00:55:41 2010
From: george.niculae79 at gmail.com (George Niculae)
Date: Tue, 28 Dec 2010 23:55:41 +0200
Subject: [Freeswitch-users] call dropped while trying to transfer
In-Reply-To: <AANLkTi==EKk_igs7kNnzgiS8XZs9S5L_Dp1=NcXBma0+@mail.gmail.com>
References: <AANLkTiku8ZRDxFxQECys14BW94X2J28UpxKZu+p_BOXq@mail.gmail.com>
	<AANLkTin0Lmz64n+chbcQ6UqYY704bcim+yOf5AOcV6sv@mail.gmail.com>
	<AANLkTintrgfrhB33pRydJzniksont2BPtu0eVpj+=fUG@mail.gmail.com>
	<AANLkTinEiSKfA3Aw4LjXQ6cEYdEqiGN87thUUHQEh7QQ@mail.gmail.com>
	<AANLkTi==EKk_igs7kNnzgiS8XZs9S5L_Dp1=NcXBma0+@mail.gmail.com>
Message-ID: <AANLkTikxbsAKhsybVp2SKG7_t3CmB+Hv_HZ_fxSAgbEb@mail.gmail.com>

Hi Michael,

is the provided trace OK or should I collect new logs?

Thanks,
George

2010/12/23 George Niculae <george.niculae79 at gmail.com>

> Here it is: http://pastebin.freeswitch.org/14873
>
> <http://pastebin.freeswitch.org/14873>Thanks,
> George
>
> 2010/12/23 Michael Collins <msc at freeswitch.org>
>
> Try turning on the siptrace as well so we can see the sip traffic:
>>
>> sofia profile internal siptrace on
>>
>> Then do another test & pastebin the debug output.
>> -MC
>>
>>
>> On Thu, Dec 23, 2010 at 4:36 AM, George Niculae <george at ezuce.com> wrote:
>>
>>> Michael,
>>>
>>> the commands are written on socket using PrintWriter.printf() and in
>>> this case is something like:
>>> api uuid_deflect f5539b24-0e8e-11e0-9a0e-c37fe40448c1
>>> sip:101 at dizzy.dizzysip.ro <sip%3A101 at dizzy.dizzysip.ro>
>>> Please see here all commands sent (prefixed with FSES::cmd):
>>> http://pastebin.freeswitch.org/14868 , uuid deflect at line 27
>>> New console output (for correlating uuid's if needed):
>>> http://pastebin.freeswitch.org/14867
>>>
>>> Thanks,
>>> George
>>>
>>> On Thu, Dec 23, 2010 at 2:07 AM, Michael Collins <msc at freeswitch.org>
>>> wrote:
>>> > Please pastebin the code that performs the uuid_deflect so that we can
>>> see
>>> > what you are doing to produce this symptom.
>>> > -MC
>>> >
>>> > On Wed, Dec 22, 2010 at 8:35 AM, George Niculae <george at ezuce.com>
>>> wrote:
>>> >>
>>> >> Hi All,
>>> >>
>>> >> I am working on an IVR application based on FS (running FreeSWITCH
>>> >> Version 1.0.head (git-43393f2 2010-12-15 20-59-42 -0600) where the
>>> >> following scenario fails:
>>> >> user 201 calls to 100 (autoattendant), hears menu then press # to
>>> >> transfer to voicemail (101), but the call is dropped (transfer is made
>>> >> using uuid_deflect api command)
>>> >> Dialplan extension configured like:
>>> >>
>>> >>  <extension name="IVR">
>>> >>    <condition field="destination_number" expression="^IVR${dollar}">
>>> >>      <action application="socket" data="192.168.0.114:8084 async
>>> full"/>
>>> >>    </condition>
>>> >>  </extension>
>>> >>
>>> >> Actions taken are:
>>> >> - when call arrives to extension 100 call is bridged
>>> >> (hangup_after_bridge=true)
>>> >> - answer the call, autoattendant menu is played and DTMF collected
>>> >> - when # pressed, call is transfered to 101 using uuid_deflect
>>> >> - call arrives to voicemail extension and is again bridged
>>> >> - call is answered - at this point in time the initial bridge hangs up
>>> >> and the whole call is dropped
>>> >> Please see console output http://pastebin.freeswitch.org/14855
>>> >>
>>> >> When debugging the application, If I keep the first channel connected
>>> >> transfer works just fine without dropping the call.
>>> >> Pretty sure I'm missing something here, any suggestion highly
>>> appreciated
>>> >>
>>> >> Thanks,
>>> >> George
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From lloyd.aloysius at gmail.com  Wed Dec 29 01:44:00 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 28 Dec 2010 17:44:00 -0500
Subject: [Freeswitch-users] voicemail operator-extension -
	Multi-tenantEnvironment
In-Reply-To: <AANLkTimbpDJ3QrkfzvVygtSA1s-KmpV_f8TKYU4RLgf7@mail.gmail.com>
References: <AANLkTimm99xdzt2cW6zTpFZY+FtiRCaHCQSm=Rv__o4G@mail.gmail.com>
	<614459D4C10248C8B913E4422CFC7C35@e1705>
	<D330BE13-B0A1-4D76-9DFA-B80B7F3534D8@gmail.com>
	<AANLkTinATfPEbDkzfZSyXHgVYM0mjtsgvzk28xYBotXt@mail.gmail.com>
	<AANLkTik=afD7Zjsb_jPw1aNSXsxLCtHeshiVbK-aHTbA@mail.gmail.com>
	<AANLkTikfETbZCi080u1uuLQj-VkezQf_2+=eWBUv5mZ+@mail.gmail.com>
	<AANLkTimbpDJ3QrkfzvVygtSA1s-KmpV_f8TKYU4RLgf7@mail.gmail.com>
Message-ID: <AANLkTin5nFvEUwUmNj9kf1cHx7wTgcNKsLWKF8D64iw3@mail.gmail.com>

When I press 9 the call get transfered to the default context. Then I try
get the ${domain_name} that is giving the default domain_name. I could not
find a way to get the correct voicemal domain_name from the from default
context.

xml_curl .... right now I am using xml_curl. All users defined in mysql,
 then I use a php script for the user informations.

How to use xml_curl for voicemail ? please let me know if there any
help/docs on this.

Thanks
LLoyd


On Tue, Dec 28, 2010 at 4:29 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Dec 28, 2010 at 9:57 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Michael,
>>
>> Thank you for the suggestion but this is not working . I think because of
>> the the  <param name="callback-context" value="default"/>
>>
>
> Please define "not working" - either you press 9 and the call is x-fer'd or
> it is not. Once it is x-fer'd to the dialplan you should be able to do
> whatever you want with the call. Or do what bkw says and use xml_curl.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From godson.g at gmail.com  Wed Dec 29 05:30:12 2010
From: godson.g at gmail.com (Godson Gera)
Date: Wed, 29 Dec 2010 08:00:12 +0530
Subject: [Freeswitch-users]
	=?windows-1252?q?ANN_=3A_PySWITCH_Release_=96_?=
	=?windows-1252?q?0=2E1alpha?=
In-Reply-To: <AANLkTikGS5wK3fzArXVCiZOFN8HKKyWFLfL1gUstWZx7@mail.gmail.com>
References: <AANLkTi=FDcobUchZZsq8uWzfSe99MgTHW5FvZV=d2kzM@mail.gmail.com>
	<AANLkTikGS5wK3fzArXVCiZOFN8HKKyWFLfL1gUstWZx7@mail.gmail.com>
Message-ID: <AANLkTi=yW_gy+u_ZCoaZx8itJRvjLkewZyoxbOZkn-Tx@mail.gmail.com>

Hi Michael,

Thanks for the reply. To make it easy for developers, I am in the process of
setting up some tutorials and examples. Once I am done with that, I sure
would give a demo of it in FS conference.

On Wed, Dec 29, 2010 at 3:13 AM, Michael Collins <msc at freeswitch.org> wrote:

> Thanks for the heads up. Perhaps in the near future you could call in to
> our weekly FS conference and give us a demonstration of your API in action.
> -MC
>
> On Mon, Dec 27, 2010 at 10:51 PM, Godson Gera <godson.g at gmail.com> wrote:
>
>>
>> Hi All,
>>
>> I am glad to announce the first alpha release of PySWITCH.
>>
>> http://pyswitch.sf.net
>>
>>
>> The idea of PySWITCH is to offer a complete library to Python and Twisted
>> programmers for interacting with FreeSWITCH using EventSocket interface. The
>> target is to cover all FreeSWITCH API commands and Dialplan tools. PySWITCH
>> handles all the low level details in executing FreeSWITCH commands, so the
>> programmer can easily concentrate on quickly building FreeSWITCH
>> applications. As an example, the API functions offered by PySWITCH often
>> executes many FreeSWITCH commands under the hood and finally returns the
>> desired result. Suppose you execute a background job, PySWITCH API will
>> automatically wait and catch the backgroundjob event parse the result and
>> will fire the deferred.
>>
>> The current release covers good amount of API commands and a few Dialplan
>> tools. The protocol communication issues are ironed out. It has a nice event
>> call back interface. I?ll present its usage in couple of tutorials soon.
>>
>>
>> --
>> Thanks & Regards,
>> Godson Gera
>> http://godson.in
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Thanks & Regards,
Godson Gera
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From brian at freeswitch.org  Wed Dec 29 05:34:01 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 28 Dec 2010 20:34:01 -0600
Subject: [Freeswitch-users] files.freeswitch.org
Message-ID: <5E5E14C7-564E-404C-A976-3A658A53A4B6@freeswitch.org>

Our CDN is having some issues in the past 4 hours so I have disabled it for now please let me know if you continue to have issues with files.freeswitch.org

Thanks,
Brian



From darren at aleph-com.net  Wed Dec 29 06:46:15 2010
From: darren at aleph-com.net (Darren Wiebe)
Date: Tue, 28 Dec 2010 20:46:15 -0700
Subject: [Freeswitch-users] calling card app design decision
In-Reply-To: <AANLkTim4nPi+tiPBurNknt2au7XTbj0j8XJ8uCuKy-xp@mail.gmail.com>
References: <COL122-W40EEE9A9EEB0A4C5B52283B51B0@phx.gbl>	<4D1527A8.7010607@sns.eu>
	<AANLkTim4nPi+tiPBurNknt2au7XTbj0j8XJ8uCuKy-xp@mail.gmail.com>
Message-ID: <4D1AAF07.8080506@aleph-com.net>

I'd suggest using the latest code in -trunk.

http://sourceforge.net/projects/astpp/develop

Darren Wiebe

On 10-12-24 07:24 PM, EdPimentl wrote:
> Can someone "please" post the link to the latest version of astpp for 
> freeswitch?
>
> Thanks in advance,
> -E
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From u2nsam at gmail.com  Wed Dec 29 07:29:32 2010
From: u2nsam at gmail.com (Sam)
Date: Wed, 29 Dec 2010 09:59:32 +0530
Subject: [Freeswitch-users] DTMF missing
In-Reply-To: <AANLkTi=3N59_nwtiej=Xbtiodp95U_FYUpOPamN=fzYH@mail.gmail.com>
References: <AANLkTik=NLa+injL5hF=VsymwoE9wd1Wb2joXJDhvfFZ@mail.gmail.com>
	<AANLkTincAApO4JDwzt-5oX0iD2VozFNVCtembOumL4iS@mail.gmail.com>
	<AANLkTin14RN-WxkYzqkQPRZctYt09xkWguAoCr8eeUWe@mail.gmail.com>
	<AANLkTik3N+F6Li6OL9EmyDH34eowxR+ttBFZx=1e83Eg@mail.gmail.com>
	<AANLkTikAKqXyFDQGTaBxW6OhS7pVzUzT_m3n7mZmAfxc@mail.gmail.com>
	<AANLkTi=3N59_nwtiej=Xbtiodp95U_FYUpOPamN=fzYH@mail.gmail.com>
Message-ID: <AANLkTikPrBiJ500DxzVkqJjHht0RBLq_WjsOEEWJtSOL@mail.gmail.com>

The client is sending DTMF inband

also have used <variable name="bypass_media" value="0"/>

and also have used

<param name="dtmf-type" value="rfc2833"/>
    <param name="pass-rfc2833" value="true"/>

for IVR i could see the dtmf ; but when after bridging extension to
extension i do not see the dtmf when used for attn transfer.

Regards
Sam



On Wed, Dec 29, 2010 at 3:21 AM, Michael Collins <msc at freeswitch.org> wrote:

> Assuming your console in on DEBUG level output then yes, you should see the
> DTMFs show up on the screen. If you're not seeing DTMFs then yeah, something
> is wrong. Confirm whether the client is sending the DTMFs inband or not.
>
> -MC
>
>
> On Tue, Dec 28, 2010 at 1:11 AM, Sam <u2nsam at gmail.com> wrote:
>
>> When i do " #1 " i cannot see the dtmf on the fs_cli , is it because of
>> that ?
>>
>> Regds
>> Sam
>>
>>
>> On Mon, Dec 27, 2010 at 1:26 PM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Hi ,
>>>
>>> I have now upgraded to ver (git-4e95227 2010-12-26 09-09-14 -0600) , now
>>> the dtmfs in IVR are working !
>>>
>>>
>>> When i used bind meta application then its not executing, any comments
>>> for me ?
>>>
>>>             <action application="set" data="export_vars=#,*"/>
>>>             <action application="export" data="#=true"/>
>>>             <action application="set" data="bind_meta_key=#"/>
>>>             <action application="bind_meta_app" data="1 ab s
>>> execute_extension::dx XML features"/>
>>>
>>>
>>>
>>>
>>> Regards
>>> Sam
>>>
>>>
>>>
>>>
>>> On Sun, Dec 26, 2010 at 11:16 PM, afshin afzali <a.afzali2003 at gmail.com>wrote:
>>>
>>>> Sam,
>>>>
>>>> Yes, The issue has resolved in last git :)
>>>> -- afshin
>>>>
>>>> On Sun, Dec 26, 2010 at 7:33 AM, Sam <u2nsam at gmail.com> wrote:
>>>>
>>>>> Hello
>>>>>
>>>>> Just like Afshin, I am also missing DTMF digits ,using FreeSWITCH
>>>>> Version 1.0.head (git-34a0ca5 2010-12-22 20-38-57 -0600) ;
>>>>> should i upgrade to latest ?
>>>>>
>>>>> Regards
>>>>> Sam
>>>>>
>>>>>
>>>>>
>>>>> On Fri, Dec 24, 2010 at 4:10 PM, Sam <u2nsam at gmail.com> wrote:
>>>>>
>>>>>> Hi,,
>>>>>>
>>>>>>
>>>>>> I have installed the latest ver of freeswitch and i have configured
>>>>>> the conference.
>>>>>>
>>>>>> now when i punch in the digits for password , i could see that the
>>>>>> DTMF digits are missed on fs_cli.
>>>>>>
>>>>>> it only happens when i dial it from polycom or cisco phones.
>>>>>>
>>>>>> I have tried with and without these values below:-
>>>>>>
>>>>>> <param name="dtmf-type" value="rfc2833"/>
>>>>>>     <param name="pass-rfc2833" value="true"/>
>>>>>>
>>>>>>
>>>>>>
>>>>>> traces fetched:
>>>>>> 192.168.2.49:5060 -> 192.168.2.190:5060
>>>>>>   INVITE sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>SIP/2.0..Via: SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From:
>>>>>> "7028" <sip:7028 at 192.168.2.190 <sip%3A7028 at 192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To:
>>>>>> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>..Call-ID:
>>>>>> 0017592a-eb33001a-
>>>>>>   63da3294-1a7bfdfa at 192.168.2.49..Max-Forwards: 70..Date: Fri, 24 Dec
>>>>>> 2010 10:13:52 GMT..CSeq: 102 INVITE..User-Agent: Cisco-CP7940G/8.0..Contact:
>>>>>> <sip:7028 at 192.168.2.49:5060;transport=udp>..Proxy-Authorization:
>>>>>> Digest username="7028"
>>>>>>   ,realm="192.168.2.190",uri="sip:7050 at 192.168.2.190<sip%3A7050 at 192.168.2.190>",response="a668f5c480285b35e7ff6bcd446879f0",nonce="d2c540f2-8487-4d87-bdab-871585253eb8",cnonce="0a6c4176",qop=auth,nc=00000001,algorithm=MD5..Expires:
>>>>>> 180..Accept: application/sdp
>>>>>>   ..Allow:
>>>>>> ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported:
>>>>>> replaces,join,norefersub..Content-Length: 220..Content-Type:
>>>>>> application/sdp..Content-Disposition:
>>>>>> session;handling=optional....v=0..o=Cisco-SIPUA 16102
>>>>>>    0 IN IP4 192.168.2.49..s=SIP Call..t=0 0..m=audio 17298 RTP/AVP 0 8
>>>>>> 18..c=IN IP4 192.168.2.49..a=rtpmap:0 PCMU/8000..a=rtpmap:8
>>>>>> PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=sendrecv..
>>>>>>
>>>>>>
>>>>>> 192.168.2.190:5060 -> 192.168.2.49:5060
>>>>>>   SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From:
>>>>>> "7028" <sip:7028 at 192.168.2.190 <sip%3A7028 at 192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To:
>>>>>> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;tag=2XXUZpgr1rvgc..Call-ID:
>>>>>> 0017592a-eb33001a-63da3
>>>>>>   294-1a7bfdfa at 192.168.2.49..CSeq: 102 INVITE..Contact:
>>>>>> <sip:7050 at 192.168.2.190:5060;transport=udp>..User-Agent:
>>>>>> NOVANET..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
>>>>>> MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIF
>>>>>>   Y, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path,
>>>>>> replaces..Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>>>>>> sla, include-session-description, presence.winfo, message-summary,
>>>>>> refer..Session-Expires: 180
>>>>>>   0;refresher=uas..Min-SE: 120..Content-Type:
>>>>>> application/sdp..Content-Disposition: session..Content-Length:
>>>>>> 249..Remote-Party-ID: "7050" <sip:7050 at 192.168.2.190<sip%3A7050 at 192.168.2.190>>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH
>>>>>> 1293163579 129
>>>>>>   3163580 IN IP4 192.168.2.190..s=FreeSWITCH..c=IN IP4
>>>>>> 192.168.2.190..t=0 0..m=audio 22050 RTP/AVP 0 101..a=rtpmap:0
>>>>>> PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
>>>>>> 0-16..a=silenceSupp:off - - - -..a=ptime:20..
>>>>>>
>>>>>> But the dtmf are not missed when punched on eyebeam softphone.
>>>>>>
>>>>>> And all the phones have RFC 2833.
>>>>>>
>>>>>> traces fetched for softphone:-
>>>>>>
>>>>>> 192.168.2.17:6182 -> 192.168.2.190:5060
>>>>>>   INVITE sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>SIP/2.0..To: <
>>>>>> sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>..From: 7001<
>>>>>> sip:7001 at 192.168.2.190 <sip%3A7001 at 192.168.2.190>>;tag=6c557c1e..Via:
>>>>>> SIP/2.0/UDP 192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport..Call-ID:
>>>>>> d32ffe546570a77e..CS
>>>>>>   eq: 2 INVITE..Contact: <sip:7001 at 192.168.2.17:6182>..Max-Forwards:
>>>>>> 70..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>>>>>> SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Authorization: Digest
>>>>>> username="7001",rea
>>>>>>
>>>>>> lm="192.168.2.190",nonce="d0b4db6e-bd76-447f-9076-2e6b7809cb54",uri="
>>>>>> sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>",response="45fb9eb4f0e0e4fffd87a22769a007ba",cnonce="1c27f3687059b16d",nc=00000001,qop=auth,algorithm=MD5..User-Agent:
>>>>>> eyeBeam release 3007n
>>>>>>    stamp 17816..Content-Length: 233....v=0..o=- 27833664 27833670 IN
>>>>>> IP4 192.168.2.17..s=eyeBeam..c=IN IP4 192.168.2.17..t=0 0..m=audio 6398
>>>>>> RTP/AVP 0 18 101..a=alt:1 1 : 2C830AD9 0000004F 192.168.2.17
>>>>>> 6398..a=fmtp:101 0-15..a=rtpmap:
>>>>>>   101 telephone-event/8000..a=sendrecv..
>>>>>>
>>>>>>
>>>>>> 192.168.2.190:5060 -> 192.168.2.17:6182
>>>>>>   SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport=6182..From:
>>>>>> 7001 <sip:7001 at 192.168.2.190 <sip%3A7001 at 192.168.2.190>>;tag=6c557c1e..To:
>>>>>> <sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;tag=H0ctQv7KNgU2j..Call-ID:
>>>>>> d32ffe546570a77e..C
>>>>>>   Seq: 2 INVITE..Contact: <sip:7050 at 192.168.2.190:5060;transport=udp>..User-Agent:
>>>>>> NOVANET..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
>>>>>> MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Support
>>>>>>   ed: timer, precondition, path, replaces..Allow-Events: talk, hold,
>>>>>> presence, dialog, line-seize, call-info, sla, include-session-description,
>>>>>> presence.winfo, message-summary, refer..Session-Expires:
>>>>>> 1800;refresher=uas..Min-SE: 120..
>>>>>>   Content-Type: application/sdp..Content-Disposition:
>>>>>> session..Content-Length: 249..Remote-Party-ID: "7050" <
>>>>>> sip:7050 at 192.168.2.190 <sip%3A7050 at 192.168.2.190>>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH
>>>>>> 1293161640 1293161641 IN IP4 192.168.2.190..
>>>>>>   s=FreeSWITCH..c=IN IP4 192.168.2.190..t=0 0..m=audio 24852 RTP/AVP 0
>>>>>> 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
>>>>>> 0-16..a=silenceSupp:off - - - -..a=ptime:20..
>>>>>>
>>>>>>
>>>>>> Any thing you can think how it can happen?
>>>>>>
>>>>>>
>>>>>> Regards
>>>>>> Sam
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Dec 29 07:39:37 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 28 Dec 2010 22:39:37 -0600
Subject: [Freeswitch-users] DTMF missing
In-Reply-To: <AANLkTikPrBiJ500DxzVkqJjHht0RBLq_WjsOEEWJtSOL@mail.gmail.com>
References: <AANLkTik=NLa+injL5hF=VsymwoE9wd1Wb2joXJDhvfFZ@mail.gmail.com>
	<AANLkTincAApO4JDwzt-5oX0iD2VozFNVCtembOumL4iS@mail.gmail.com>
	<AANLkTin14RN-WxkYzqkQPRZctYt09xkWguAoCr8eeUWe@mail.gmail.com>
	<AANLkTik3N+F6Li6OL9EmyDH34eowxR+ttBFZx=1e83Eg@mail.gmail.com>
	<AANLkTikAKqXyFDQGTaBxW6OhS7pVzUzT_m3n7mZmAfxc@mail.gmail.com>
	<AANLkTi=3N59_nwtiej=Xbtiodp95U_FYUpOPamN=fzYH@mail.gmail.com>
	<AANLkTikPrBiJ500DxzVkqJjHht0RBLq_WjsOEEWJtSOL@mail.gmail.com>
Message-ID: <48F39398-0A59-4108-80F2-B6663B961DED@freeswitch.org>

Ok here is the deal... RFC2833 is in the media... if you have bypass on how do you ever expect to have the DTMF in FreeSWITCH?

/b

On Dec 28, 2010, at 10:29 PM, Sam wrote:

> also have used <variable name="bypass_media" value="0"/>



From u2nsam at gmail.com  Wed Dec 29 08:59:08 2010
From: u2nsam at gmail.com (Sam)
Date: Wed, 29 Dec 2010 11:29:08 +0530
Subject: [Freeswitch-users] DTMF missing
In-Reply-To: <48F39398-0A59-4108-80F2-B6663B961DED@freeswitch.org>
References: <AANLkTik=NLa+injL5hF=VsymwoE9wd1Wb2joXJDhvfFZ@mail.gmail.com>
	<AANLkTincAApO4JDwzt-5oX0iD2VozFNVCtembOumL4iS@mail.gmail.com>
	<AANLkTin14RN-WxkYzqkQPRZctYt09xkWguAoCr8eeUWe@mail.gmail.com>
	<AANLkTik3N+F6Li6OL9EmyDH34eowxR+ttBFZx=1e83Eg@mail.gmail.com>
	<AANLkTikAKqXyFDQGTaBxW6OhS7pVzUzT_m3n7mZmAfxc@mail.gmail.com>
	<AANLkTi=3N59_nwtiej=Xbtiodp95U_FYUpOPamN=fzYH@mail.gmail.com>
	<AANLkTikPrBiJ500DxzVkqJjHht0RBLq_WjsOEEWJtSOL@mail.gmail.com>
	<48F39398-0A59-4108-80F2-B6663B961DED@freeswitch.org>
Message-ID: <AANLkTimPDPNQsbjQZCfGgRTFLeQd2zUA85HCU8e5K_fL@mail.gmail.com>

Hi,

All Kool ! it was because of <variable name="bypass_media" value="0"/> ,
what is the meaning of the "value" here ? or its a wrong syntax

Regds
Sam




On Wed, Dec 29, 2010 at 10:09 AM, Brian West <brian at freeswitch.org> wrote:

> Ok here is the deal... RFC2833 is in the media... if you have bypass on how
> do you ever expect to have the DTMF in FreeSWITCH?
>
> /b
>
> On Dec 28, 2010, at 10:29 PM, Sam wrote:
>
> > also have used <variable name="bypass_media" value="0"/>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From telconettech at yahoo.com  Wed Dec 29 04:35:02 2010
From: telconettech at yahoo.com (Joel Mange)
Date: Tue, 28 Dec 2010 17:35:02 -0800 (PST)
Subject: [Freeswitch-users] event i_media_error 988 Incomplete offer/answer
Message-ID: <589498.32134.qm@web110801.mail.gq1.yahoo.com>

Hello.? New to FS and in need of help.

Environment is as follows:

Linksys SPA-962 (x94140) <---> FS <---sip---> Cisco3845 <---qsig---> TDM PBX <--> TDM phone
 (x91237)

Can complete call from TDM phone to Linksys SPA-962.? Can't complete call from Linksys to TDM phone:? TDM phone rings, but call fails when it goes off-hook.? Getting the following sofia error:? nua(0x8b60cf0): event i_media_error 988 Incomplete offer/answer.

I think the above message tells me that the other side of the call sent an invalid SDP.? Having trouble identifying the invalid SDP and making necessary correction.

Siptrace and router running-config at:??http://pastebin.freeswitch.org/14896

Any help appreciated.


      
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From u2nsam at gmail.com  Wed Dec 29 12:38:09 2010
From: u2nsam at gmail.com (Sam)
Date: Wed, 29 Dec 2010 15:08:09 +0530
Subject: [Freeswitch-users] DTMF missing
In-Reply-To: <AANLkTimPDPNQsbjQZCfGgRTFLeQd2zUA85HCU8e5K_fL@mail.gmail.com>
References: <AANLkTik=NLa+injL5hF=VsymwoE9wd1Wb2joXJDhvfFZ@mail.gmail.com>
	<AANLkTincAApO4JDwzt-5oX0iD2VozFNVCtembOumL4iS@mail.gmail.com>
	<AANLkTin14RN-WxkYzqkQPRZctYt09xkWguAoCr8eeUWe@mail.gmail.com>
	<AANLkTik3N+F6Li6OL9EmyDH34eowxR+ttBFZx=1e83Eg@mail.gmail.com>
	<AANLkTikAKqXyFDQGTaBxW6OhS7pVzUzT_m3n7mZmAfxc@mail.gmail.com>
	<AANLkTi=3N59_nwtiej=Xbtiodp95U_FYUpOPamN=fzYH@mail.gmail.com>
	<AANLkTikPrBiJ500DxzVkqJjHht0RBLq_WjsOEEWJtSOL@mail.gmail.com>
	<48F39398-0A59-4108-80F2-B6663B961DED@freeswitch.org>
	<AANLkTimPDPNQsbjQZCfGgRTFLeQd2zUA85HCU8e5K_fL@mail.gmail.com>
Message-ID: <AANLkTi=S6YBwVQ4Bk6xSRMiPqqHDRuTKfhJwqwzKTFbr@mail.gmail.com>

With the same setting which was working few hours ago and now its not
working.
I donot see DTMF on fs cli now

Regds
Sam

On Wed, Dec 29, 2010 at 11:29 AM, Sam <u2nsam at gmail.com> wrote:

> Hi,
>
> All Kool ! it was because of <variable name="bypass_media" value="0"/> ,
> what is the meaning of the "value" here ? or its a wrong syntax
>
> Regds
> Sam
>
>
>
>
>
> On Wed, Dec 29, 2010 at 10:09 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Ok here is the deal... RFC2833 is in the media... if you have bypass on
>> how do you ever expect to have the DTMF in FreeSWITCH?
>>
>> /b
>>
>> On Dec 28, 2010, at 10:29 PM, Sam wrote:
>>
>> > also have used <variable name="bypass_media" value="0"/>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From rajkumar.kmry at gmail.com  Wed Dec 29 12:32:25 2010
From: rajkumar.kmry at gmail.com (rajkumar)
Date: Wed, 29 Dec 2010 01:32:25 -0800 (PST)
Subject: [Freeswitch-users] bridge application dial string
Message-ID: <1293615145403-5874073.post@n2.nabble.com>


Hi,

  Normally we can specify the channel in bridge application.
  Is it possible to specify the dial plan extension in bridge dial string?
  
regards
rajkumar k
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/bridge-application-dial-string-tp5874073p5874073.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Wed Dec 29 14:27:12 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 29 Dec 2010 11:27:12 +0000
Subject: [Freeswitch-users] bridge application dial string
In-Reply-To: <1293615145403-5874073.post@n2.nabble.com>
References: <1293615145403-5874073.post@n2.nabble.com>
Message-ID: <39788A22-DE3A-46C9-AC45-1597CC339A19@gmail.com>

Do you mean you want to send the call to a specific extension? If so, look at transfer or loopback.

Steve on iPhone

On 29 Dec 2010, at 09:32, rajkumar <rajkumar.kmry at gmail.com> wrote:

> 
> Hi,
> 
>  Normally we can specify the channel in bridge application.
>  Is it possible to specify the dial plan extension in bridge dial string?
> 
> regards
> rajkumar k
> -- 
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/bridge-application-dial-string-tp5874073p5874073.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From kris at kriskinc.com  Wed Dec 29 17:38:53 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 29 Dec 2010 09:38:53 -0500
Subject: [Freeswitch-users] event i_media_error 988 Incomplete
	offer/answer
In-Reply-To: <589498.32134.qm@web110801.mail.gq1.yahoo.com>
References: <589498.32134.qm@web110801.mail.gq1.yahoo.com>
Message-ID: <AANLkTimnq1g-TpsU2AW73YDs1K1FzFLB7cmoWhQ5UUhe@mail.gmail.com>

Joel,

? Why do you have SIP GTD enabled on your Cisco router?? FreeSWITCH
won't do anything with this information anyway, you might as well
disable it and multipart handling in general.? It's probably enabled
by default because you're using QSIG (I'm not sure) but this doc
should be able to help:

http://www.cisco.com/en/US/docs/ios/voice/sip/configuration/guide/sip_cg-isdn.html#wp1047570

? Or you could try to tweak multipart handling in FS using this variable:

http://wiki.freeswitch.org/wiki/Variable_sip_copy_multipart

  Also - why do you have compact headers turned on?

On Tue, Dec 28, 2010 at 8:35 PM, Joel Mange <telconettech at yahoo.com> wrote:
>
> Hello.? New to FS and in need of help.
>
> Environment is as follows:
>
> Linksys SPA-962 (x94140) <---> FS <---sip---> Cisco3845 <---qsig---> TDM PBX <--> TDM phone (x91237)
>
> Can complete call from TDM phone to Linksys SPA-962.? Can't complete call from Linksys to TDM phone:? TDM phone rings, but call fails when it goes off-hook.? Getting the following sofia error:? nua(0x8b60cf0): event i_media_error 988 Incomplete offer/answer.
>
> I think the above message tells me that the other side of the call sent an invalid SDP.? Having trouble identifying the invalid SDP and making necessary correction.
>
> Siptrace and router running-config at:??http://pastebin.freeswitch.org/14896
>
> Any help appreciated.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From u2nsam at gmail.com  Wed Dec 29 17:48:34 2010
From: u2nsam at gmail.com (Sam)
Date: Wed, 29 Dec 2010 20:18:34 +0530
Subject: [Freeswitch-users] mod callcenter
Message-ID: <AANLkTikdRNMo4CgFcuon0X4KkyZSzZYHK6RECaDxrNwa@mail.gmail.com>

Hello,

Was testing callcenter module and found out that at times it gives error "
invalid application callcenter " and after
reloading the module it works fine.
Some time also happens that if I reload the module it do not reloads the
parameters of the callcenter like the agents & tires.
It just unload & reloads even if there are changes to the specifications of
agents.

Regds
Sam
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From nagalenoj at gmail.com  Wed Dec 29 18:15:43 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 29 Dec 2010 20:45:43 +0530
Subject: [Freeswitch-users] ftmod_sangoma_isdn: dialtone in overlap dialing
Message-ID: <AANLkTi=qD8nqq2yqpKPHcs88oVDxcBN4wSqiaKkQweHb@mail.gmail.com>

Dear friends,
  I would want to know how to configure ftmod_sangoma_isdn to send dialtone.

My setup is as follows,

Telcom <--->  Freeswitch <---> PBX <----> Extensions 1..n


My pbx has got overlap dial enabled and it is configured to seize the trunk
when it receives '0' from extensions.

When I connect the PRI to my pbx, and when '0' is dialled from an extension,
my pbx seizes the trunk and I'm able to hear dialtone. But, when I connect
the PRI to freeswitch and connect freeswitch to pbx and when '0' is dialled
from an extension, I'm not hearing dialtone.

Whether some configuration is available to ask ftmod_sangoma_isdn to send
dialtone before started to collect digits in overlap mode? or any other way
to give dialtone??


-- 
Regards,
Nagalenoj H.
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From lloyd.aloysius at gmail.com  Wed Dec 29 18:46:31 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 29 Dec 2010 10:46:31 -0500
Subject: [Freeswitch-users] CentOS 5.5 - FreeSWITCH - Audio Quality
	issues
In-Reply-To: <AANLkTimREKiD=3t12WT7fMWWR7JPyOT+3S4jm7SPY6UT@mail.gmail.com>
References: <AANLkTik3y8EfTi5s_a_1BURnUJe8Nv0p+5y8Ph6Jkcg3@mail.gmail.com>
	<AANLkTi=6shTJg0C2eFsar+mdjstT330e2kaUtFxsPQ1F@mail.gmail.com>
	<AANLkTi=3Yi_Phq6Zv2Ep-YEgf=3orNmOEpLVyww=7x=L@mail.gmail.com>
	<316084A7-BCA6-4490-8990-9894331F999F@freeswitch.org>
	<AANLkTimREKiD=3t12WT7fMWWR7JPyOT+3S4jm7SPY6UT@mail.gmail.com>
Message-ID: <AANLkTintspeXDQiyHXsu2GmuyiK0yqVKrBKyBmwmY-fB@mail.gmail.com>

Hi All,

I am replying to my last email. Apologies if this cause any problems.

Please see the Disk performance below. Looks to me there is no issue on the
Disk IO.

Right now I am using CentOS 5.5 32 bit and kernel 2.6.18-194.26.1.el5PAE #1
SMP . I could not find any other way to figure out what is causing this
problem. is it a good idea to use CentOS 64 bit ?

[root at hd-t1463cl ~]# sar
Linux 2.6.18-194.26.1.el5PAE
12/29/2010

12:00:01 AM       CPU     %user     %nice   %system   %iowait
%steal     %idle
12:10:01 AM       all      0.02      0.00      0.02      0.12
0.00     99.85
12:20:01 AM       all      0.02      0.00      0.02      0.12
0.00     99.84
12:30:01 AM       all      0.01      0.00      0.02      0.12
0.00     99.85
12:40:01 AM       all      0.02      0.00      0.02      0.13
0.00     99.83
12:50:01 AM       all      0.02      0.00      0.02      0.12
0.00     99.83
01:00:01 AM       all      0.02      0.00      0.02      0.11
0.00     99.85


[root at hd-t1463cl ~]# cat /proc/mdstat
Personalities : [raid1]
md0 : active raid1 sdb1[1] sda1[0]
     104320 blocks [2/2] [UU]

md1 : active raid1 sdb2[1] sda2[0]
     2096384 blocks [2/2] [UU]

md2 : active raid1 sdb5[1] sda5[0]
     308271168 blocks [2/2] [UU]

[root at hd-t1463cl ~]# hdparm -tT /dev/sda

/dev/sda:
 Timing cached reads:   15900 MB in  1.99 seconds = 7974.92 MB/sec
 Timing buffered disk reads:  330 MB in  3.01 seconds = 109.70 MB/sec

[root at hd-t1463cl ~]# hdparm -tT /dev/sdb

/dev/sdb:
 Timing cached reads:   15000 MB in  1.99 seconds = 7522.76 MB/sec
 Timing buffered disk reads:  334 MB in  3.01 seconds = 110.79 MB/sec

Thanks and regards,
Lloyd

On Tue, Dec 28, 2010 at 2:05 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Brian,
>
> Here is information about the box and codec
>
> processor       : 2
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 15
> model name      : Intel(R) Core(TM)2 Quad CPU    Q6600  @ 2.40GHz
> stepping        : 11
> cpu MHz         : 2394.057
> cache size      : 4096 KB
> physical id     : 0
> siblings        : 4
> core id         : 2
> cpu cores       : 4
> apicid          : 2
> fdiv_bug        : no
> hlt_bug         : no
> f00f_bug        : no
> coma_bug        : no
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca
> cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm
> constant_tsc pni monitor ds_cpl vmx est tm2 ssse3 cx16 xtpr lahf_lm
> bogomips        : 4787.33
>
> ---
>
> Applications
>
> /usr/local/freeswitch/bin/freeswitch -nc
> mysql
> Apache
>
> Apache and mysql use only for xml_cdr. There is no other applications.
>
> --
>
> top - 14:01:32 up 8 days, 12:57,  2 users,  load average: 0.10, 0.05, 0.01
> Tasks: 104 total,   1 running, 102 sleeping,   0 stopped,   1 zombie
> Cpu(s):  0.0%us,  0.1%sy,  0.0%ni, 99.9%id,  0.0%wa,  0.0%hi,  0.0%si,
>  0.0%st
> Mem:   4137172k total,   730956k used,  3406216k free,   164412k buffers
> Swap:  4192944k total,        0k used,  4192944k free,   463228k cached
>
> ---
>
>              total       used       free     shared    buffers     cached
> Mem:          4040        713       3326          0        160        452
> -/+ buffers/cache:        101       3939
> Swap:         4094          0       4094
>
> --
>
> v=0
> o=FreeSWITCH 1293532168 1293532169 IN IP4 209.172.34.154
> s=FreeSWITCH
> c=IN IP4 209.172.34.154
> t=0 0
> m=audio 30754 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> ---
>
> FreeSWITCH Version 1.0.head (git-fcd6c54 2010-12-19 00-13-08 -0500)
>
> Thanks for your help.
>
> Lloyd
>
>
> On Tue, Dec 28, 2010 at 1:54 PM, Brian West <brian at freeswitch.org> wrote:
>
>> I run on a Intel(R) Core(TM)2 Quad  CPU   Q8200 ... its fine with 5.5 so
>> what exactly are you running on?  And where are these calls coming from?
>> codec? sample rate and such?
>>
>> /b
>>
>> On Dec 28, 2010, at 12:50 PM, Aloysius Lloyd wrote:
>>
>>
>> MOH - playing without any issue. I do not think Disk activity is a issue
>> here.
>>
>> Box running Intel Core 2 Quad 2.4GHz + 4GB Ram - CentOS FreeSWITCH
>>
>> IVR - Start low volume then increase ... sometime high volume then go to
>> low volume
>> Voice mails - Noice, breaking voices etc ...
>>
>> Any help is appreciated.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From infos at madovsky.org  Wed Dec 29 19:18:25 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 29 Dec 2010 11:18:25 -0500
Subject: [Freeswitch-users] multiple-registration for same extension
Message-ID: <568973BF49BD4CE39418F547BA3E5712@e1705>

I enabled multiple-registrations to true in internal.xml (which is the difference between true and contact ?)
added this line to my dialplan

<action application="bridge" data="${sofia_contact(${dialed_extension}@${domain_name})}"/>

but only the phone registered lastly rings...

thanks

Franck


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From lloyd.aloysius at gmail.com  Wed Dec 29 19:42:15 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 29 Dec 2010 11:42:15 -0500
Subject: [Freeswitch-users] multiple-registration for same extension
In-Reply-To: <568973BF49BD4CE39418F547BA3E5712@e1705>
References: <568973BF49BD4CE39418F547BA3E5712@e1705>
Message-ID: <AANLkTimiL7ObO07Gc6oGDcmOaVSy+F+Lz3SBmdmEyar6@mail.gmail.com>

reload mod_sofia

give a try. should work.

Thanks
Lloyd


On Wed, Dec 29, 2010 at 11:18 AM, Madovsky <infos at madovsky.org> wrote:

>  I enabled multiple-registrations to true in internal.xml (which is the
> difference between true and contact ?)
> added this line to my dialplan
>
> <action application="bridge" data="${sofia_contact(${dialed_extension}@
> ${domain_name})}"/>
>
> but only the phone registered lastly rings...
>
> thanks
>
> Franck
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Avi at aMarcus.com  Wed Dec 29 19:44:31 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Wed, 29 Dec 2010 18:44:31 +0200
Subject: [Freeswitch-users] multiple-registration for same extension
In-Reply-To: <568973BF49BD4CE39418F547BA3E5712@e1705>
References: <568973BF49BD4CE39418F547BA3E5712@e1705>
Message-ID: <AANLkTimhuvjtrcSBWgy=kt4HkGhqc8Y1MJDVo+j6x-E2@mail.gmail.com>

You set:
<param name="multiple-registrations" value="true"/>
and did a reloadxml, and waited for both to register again?
It's worked for me several times, however, I usually bridge to
"user/$extension".
-Avi

On Wed, Dec 29, 2010 at 6:18 PM, Madovsky <infos at madovsky.org> wrote:

>  I enabled multiple-registrations to true in internal.xml (which is the
> difference between true and contact ?)
> added this line to my dialplan
>
> <action application="bridge" data="${sofia_contact(${dialed_extension}@
> ${domain_name})}"/>
>
> but only the phone registered lastly rings...
>
> thanks
>
> Franck
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Wed Dec 29 19:54:47 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 29 Dec 2010 10:54:47 -0600
Subject: [Freeswitch-users] CentOS 5.5 - FreeSWITCH - Audio Quality
	issues
In-Reply-To: <AANLkTintspeXDQiyHXsu2GmuyiK0yqVKrBKyBmwmY-fB@mail.gmail.com>
References: <AANLkTik3y8EfTi5s_a_1BURnUJe8Nv0p+5y8Ph6Jkcg3@mail.gmail.com>
	<AANLkTi=6shTJg0C2eFsar+mdjstT330e2kaUtFxsPQ1F@mail.gmail.com>
	<AANLkTi=3Yi_Phq6Zv2Ep-YEgf=3orNmOEpLVyww=7x=L@mail.gmail.com>
	<316084A7-BCA6-4490-8990-9894331F999F@freeswitch.org>
	<AANLkTimREKiD=3t12WT7fMWWR7JPyOT+3S4jm7SPY6UT@mail.gmail.com>
	<AANLkTintspeXDQiyHXsu2GmuyiK0yqVKrBKyBmwmY-fB@mail.gmail.com>
Message-ID: <AANLkTikHJRuCYTWKOm0d_UpU340nze3kKE3xeBJad7Z4@mail.gmail.com>

What is the client you are using to make the calls?

This sounds like an error in the RTP timing caused by malfunctioning devices.
Can you get a full trace of a single test call

sofia global siptrace on
console loglevel 7

REPRODUCE and post at http://pastebin.freeswitch.org



On Wed, Dec 29, 2010 at 9:46 AM, Aloysius Lloyd
<lloyd.aloysius at gmail.com> wrote:
> Hi All,
> I am replying to my last email.?Apologies?if this cause any problems.
> Please see the Disk performance below. Looks to me there is no issue on the
> Disk IO.
> Right now I am using CentOS 5.5 32 bit and kernel?2.6.18-194.26.1.el5PAE #1
> SMP . I could not find any other way to?figure out?what is causing this
> problem. is it a good idea to use CentOS 64 bit ?
> [root at hd-t1463cl ~]# sar
> Linux 2.6.18-194.26.1.el5PAE
> 12/29/2010
> 12:00:01 AM ? ? ? CPU ? ? %user ? ? %nice ? %system ? %iowait
> %steal ? ? %idle
> 12:10:01 AM ? ? ? all ? ? ?0.02 ? ? ?0.00 ? ? ?0.02 ? ? ?0.12
> 0.00 ? ? 99.85
> 12:20:01 AM ? ? ? all ? ? ?0.02 ? ? ?0.00 ? ? ?0.02 ? ? ?0.12
> 0.00 ? ? 99.84
> 12:30:01 AM ? ? ? all ? ? ?0.01 ? ? ?0.00 ? ? ?0.02 ? ? ?0.12
> 0.00 ? ? 99.85
> 12:40:01 AM ? ? ? all ? ? ?0.02 ? ? ?0.00 ? ? ?0.02 ? ? ?0.13
> 0.00 ? ? 99.83
> 12:50:01 AM ? ? ? all ? ? ?0.02 ? ? ?0.00 ? ? ?0.02 ? ? ?0.12
> 0.00 ? ? 99.83
> 01:00:01 AM ? ? ? all ? ? ?0.02 ? ? ?0.00 ? ? ?0.02 ? ? ?0.11
> 0.00 ? ? 99.85
>
> [root at hd-t1463cl ~]# cat /proc/mdstat
> Personalities : [raid1]
> md0 : active raid1 sdb1[1] sda1[0]
> ?? ? 104320 blocks [2/2] [UU]
> md1 : active raid1 sdb2[1] sda2[0]
> ?? ? 2096384 blocks [2/2] [UU]
> md2 : active raid1 sdb5[1] sda5[0]
> ?? ? 308271168 blocks [2/2] [UU]
>
> [root at hd-t1463cl ~]# hdparm -tT /dev/sda
> /dev/sda:
> ?Timing cached reads: ? 15900 MB in ?1.99 seconds = 7974.92 MB/sec
> ?Timing buffered disk reads: ?330 MB in ?3.01 seconds = 109.70 MB/sec
> [root at hd-t1463cl ~]# hdparm -tT /dev/sdb
> /dev/sdb:
> ?Timing cached reads: ? 15000 MB in ?1.99 seconds = 7522.76 MB/sec
> ?Timing buffered disk reads: ?334 MB in ?3.01 seconds = 110.79 MB/sec
> Thanks and regards,
> Lloyd
>
> On Tue, Dec 28, 2010 at 2:05 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>
> wrote:
>>
>> Brian,
>> Here is information about the box and codec
>> processor ? ? ? : 2
>> vendor_id ? ? ? : GenuineIntel
>> cpu family ? ? ?: 6
>> model ? ? ? ? ? : 15
>> model name ? ? ?: Intel(R) Core(TM)2 Quad CPU ? ?Q6600 ?@ 2.40GHz
>> stepping ? ? ? ?: 11
>> cpu MHz ? ? ? ? : 2394.057
>> cache size ? ? ?: 4096 KB
>> physical id ? ? : 0
>> siblings ? ? ? ?: 4
>> core id ? ? ? ? : 2
>> cpu cores ? ? ? : 4
>> apicid ? ? ? ? ?: 2
>> fdiv_bug ? ? ? ?: no
>> hlt_bug ? ? ? ? : no
>> f00f_bug ? ? ? ?: no
>> coma_bug ? ? ? ?: no
>> fpu ? ? ? ? ? ? : yes
>> fpu_exception ? : yes
>> cpuid level ? ? : 10
>> wp ? ? ? ? ? ? ?: yes
>> flags ? ? ? ? ? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca
>> cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm
>> constant_tsc pni monitor ds_cpl vmx est tm2 ssse3 cx16 xtpr lahf_lm
>> bogomips ? ? ? ?: 4787.33
>> ---
>> Applications
>> /usr/local/freeswitch/bin/freeswitch -nc
>> mysql
>> Apache
>> Apache and mysql use only for xml_cdr. There is no other applications.
>> --
>> top - 14:01:32 up 8 days, 12:57, ?2 users, ?load average: 0.10, 0.05, 0.01
>> Tasks: 104 total, ? 1 running, 102 sleeping, ? 0 stopped, ? 1 zombie
>> Cpu(s): ?0.0%us, ?0.1%sy, ?0.0%ni, 99.9%id, ?0.0%wa, ?0.0%hi, ?0.0%si,
>> ?0.0%st
>> Mem: ? 4137172k total, ? 730956k used, ?3406216k free, ? 164412k buffers
>> Swap: ?4192944k total, ? ? ? ?0k used, ?4192944k free, ? 463228k cached
>> ---
>> ?? ? ? ? ? ? total ? ? ? used ? ? ? free ? ? shared ? ?buffers ? ? cached
>> Mem: ? ? ? ? ?4040 ? ? ? ?713 ? ? ? 3326 ? ? ? ? ?0 ? ? ? ?160 ? ? ? ?452
>> -/+ buffers/cache: ? ? ? ?101 ? ? ? 3939
>> Swap: ? ? ? ? 4094 ? ? ? ? ?0 ? ? ? 4094
>> --
>> v=0
>> o=FreeSWITCH 1293532168 1293532169 IN IP4 209.172.34.154
>> s=FreeSWITCH
>> c=IN IP4 209.172.34.154
>> t=0 0
>> m=audio 30754 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>> ---
>> FreeSWITCH Version 1.0.head (git-fcd6c54 2010-12-19 00-13-08 -0500)
>> Thanks for your help.
>> Lloyd
>>
>> On Tue, Dec 28, 2010 at 1:54 PM, Brian West <brian at freeswitch.org> wrote:
>>>
>>> I run on a?Intel(R) Core(TM)2 Quad ?CPU ? Q8200 ... its fine with 5.5 so
>>> what exactly are you running on? ?And where are these calls coming from?
>>> codec? sample rate and such?
>>> /b
>>> On Dec 28, 2010, at 12:50 PM, Aloysius Lloyd wrote:
>>>
>>> MOH - playing without any issue. I do not think Disk?activity?is a issue
>>> here.
>>> Box running?Intel Core 2 Quad 2.4GHz + 4GB Ram - CentOS FreeSWITCH
>>> IVR - Start low volume then increase ... sometime high volume then go to
>>> low volume
>>> Voice mails?- Noice, breaking voices etc ...
>>> Any help is appreciated.
>>> Thanks
>>> Lloyd
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From brian at freeswitch.org  Wed Dec 29 19:55:28 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 29 Dec 2010 10:55:28 -0600
Subject: [Freeswitch-users] CentOS 5.5 - FreeSWITCH - Audio Quality
	issues
In-Reply-To: <AANLkTintspeXDQiyHXsu2GmuyiK0yqVKrBKyBmwmY-fB@mail.gmail.com>
References: <AANLkTik3y8EfTi5s_a_1BURnUJe8Nv0p+5y8Ph6Jkcg3@mail.gmail.com>
	<AANLkTi=6shTJg0C2eFsar+mdjstT330e2kaUtFxsPQ1F@mail.gmail.com>
	<AANLkTi=3Yi_Phq6Zv2Ep-YEgf=3orNmOEpLVyww=7x=L@mail.gmail.com>
	<316084A7-BCA6-4490-8990-9894331F999F@freeswitch.org>
	<AANLkTimREKiD=3t12WT7fMWWR7JPyOT+3S4jm7SPY6UT@mail.gmail.com>
	<AANLkTintspeXDQiyHXsu2GmuyiK0yqVKrBKyBmwmY-fB@mail.gmail.com>
Message-ID: <AE8F4FDB-95EF-49E6-80A1-A91315DCF2E4@freeswitch.org>

YES.  You should always run a 64bit OS on a 64bit CPU... unless you like wasting have your cpu registers... but I bet you the problem will go away if you switch.

/b

On Dec 29, 2010, at 9:46 AM, Aloysius Lloyd wrote:

>  CentOS 64 bit ?

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From Avi at aMarcus.com  Wed Dec 29 20:02:29 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Wed, 29 Dec 2010 19:02:29 +0200
Subject: [Freeswitch-users] multiple-registration for same extension
In-Reply-To: <AANLkTimiL7ObO07Gc6oGDcmOaVSy+F+Lz3SBmdmEyar6@mail.gmail.com>
References: <568973BF49BD4CE39418F547BA3E5712@e1705>
	<AANLkTimiL7ObO07Gc6oGDcmOaVSy+F+Lz3SBmdmEyar6@mail.gmail.com>
Message-ID: <AANLkTi=jX=8bQ_Ui2xmzH=LrPopmEH2S2FBcx_zV=GGw@mail.gmail.com>

Oh right, not reload xml.
Reload mod_sofia is overkill, however  - "sofia profile internal restart"
does the trick.
-Avi

On Wed, Dec 29, 2010 at 6:42 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> reload mod_sofia
>
> give a try. should work.
>
> Thanks
> Lloyd
>
>
> On Wed, Dec 29, 2010 at 11:18 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  I enabled multiple-registrations to true in internal.xml (which is the
>> difference between true and contact ?)
>> added this line to my dialplan
>>
>> <action application="bridge" data="${sofia_contact(${dialed_extension}@
>> ${domain_name})}"/>
>>
>> but only the phone registered lastly rings...
>>
>> thanks
>>
>> Franck
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Dec 29 20:06:35 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Dec 2010 09:06:35 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <AANLkTinXy1y_NNb4UK286WipX3X2rzya-0MVKnbvCjQ6@mail.gmail.com>

Hey gang,

Here is today's agenda page:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_29

It is light since we have lots of people on vacation and otherwise taking
holiday. If everyone is up to it I'd like to do what we did last week and
have everyone help out with some documentation. We have several things that
we can work on:

New jitterbuffer API
FIFO params and vars
Voicemail page needs to be updated

Thanks! Talk to you shortly.
-MC
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From infos at madovsky.org  Wed Dec 29 20:11:20 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 29 Dec 2010 12:11:20 -0500
Subject: [Freeswitch-users] multiple-registration for same extension
References: <568973BF49BD4CE39418F547BA3E5712@e1705>
	<AANLkTimhuvjtrcSBWgy=kt4HkGhqc8Y1MJDVo+j6x-E2@mail.gmail.com>
Message-ID: <7BCEC2EFD5EE410594C3C47309FFDE84@e1705>

yes, 

<param name="multiple-registrations" value="true"/>
and restart FS.
also is it need to put this line in internal.xml or external.xml knowing that
registrations are effective only in internal.xml ?

> It's worked for me several times, however, I usually bridge to "user/$extension".
but if you use sofia_contact in the dialplan bridge it gives to you sofia/$${domain}/extension@$${domain}
so how to use user/extension in mutliple-registrations ?

as I use a cluster with SRV NAPTR for outbound calls maybe it's the problem.
I noticed that if the call arrives in the same node as the one where phones are registered it works.
if not, only 1 phone or no phone rings. weirdly there is nothing abnormal on debug log
  ----- Original Message ----- 
  From: Avi Marcus 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, December 29, 2010 11:44 AM
  Subject: Re: [Freeswitch-users] multiple-registration for same extension


  You set:
  <param name="multiple-registrations" value="true"/>
  and did a reloadxml, and waited for both to register again?
  It's worked for me several times, however, I usually bridge to "user/$extension".
  -Avi


  On Wed, Dec 29, 2010 at 6:18 PM, Madovsky <infos at madovsky.org> wrote:

    I enabled multiple-registrations to true in internal.xml (which is the difference between true and contact ?)
    added this line to my dialplan

    <action application="bridge" data="${sofia_contact(${dialed_extension}@${domain_name})}"/>

    but only the phone registered lastly rings...

    thanks

    Franck




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






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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From Avi at aMarcus.com  Wed Dec 29 20:26:58 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Wed, 29 Dec 2010 19:26:58 +0200
Subject: [Freeswitch-users] multiple-registration for same extension
In-Reply-To: <7BCEC2EFD5EE410594C3C47309FFDE84@e1705>
References: <568973BF49BD4CE39418F547BA3E5712@e1705>
	<AANLkTimhuvjtrcSBWgy=kt4HkGhqc8Y1MJDVo+j6x-E2@mail.gmail.com>
	<7BCEC2EFD5EE410594C3C47309FFDE84@e1705>
Message-ID: <AANLkTikm1i5qSOCib0BRwo2gLPjuufFoFcUSQ4O-767s@mail.gmail.com>

On Wed, Dec 29, 2010 at 7:11 PM, Madovsky <infos at madovsky.org> wrote:

>  yes,
>
> <param name="multiple-registrations" value="true"/>
> and restart FS.
> also is it need to put this line in internal.xml or external.xml knowing
> that
> registrations are effective only in internal.xml ?
>
> > It's worked for me several times, however, I usually bridge to
> "user/$extension".
> but if you use sofia_contact in the dialplan bridge it gives to you
> sofia/$${domain}/extension@$${domain<sofia/$$%7Bdomain%7D/extension@$$%7Bdomain>
> }
> so how to use user/extension in mutliple-registrations ?
>
> as I use a cluster with SRV NAPTR for outbound calls maybe it's the
> problem.
> I noticed that if the call arrives in the same node as the one where phones
> are registered it works.
> if not, only 1 phone or no phone rings. weirdly there is nothing abnormal
> on debug log
>
I guess you're not clustering the DB and replicating the internal
registrations across the servers?



>  ----- Original Message -----
> *From:* Avi Marcus <Avi at aMarcus.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, December 29, 2010 11:44 AM
> *Subject:* Re: [Freeswitch-users] multiple-registration for same extension
>
> You set:
> <param name="multiple-registrations" value="true"/>
> and did a reloadxml, and waited for both to register again?
> It's worked for me several times, however, I usually bridge to
> "user/$extension".
> -Avi
>
> On Wed, Dec 29, 2010 at 6:18 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  I enabled multiple-registrations to true in internal.xml (which is the
>> difference between true and contact ?)
>> added this line to my dialplan
>>
>> <action application="bridge" data="${sofia_contact(${dialed_extension}@
>> ${domain_name})}"/>
>>
>> but only the phone registered lastly rings...
>>
>> thanks
>>
>> Franck
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Wed Dec 29 20:35:21 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 29 Dec 2010 12:35:21 -0500
Subject: [Freeswitch-users] multiple-registration for same extension
References: <568973BF49BD4CE39418F547BA3E5712@e1705><AANLkTimhuvjtrcSBWgy=kt4HkGhqc8Y1MJDVo+j6x-E2@mail.gmail.com><7BCEC2EFD5EE410594C3C47309FFDE84@e1705>
	<AANLkTikm1i5qSOCib0BRwo2gLPjuufFoFcUSQ4O-767s@mail.gmail.com>
Message-ID: <1DC63D7EFF8F41BA897BF0C13A48EAB9@e1705>

yes it is. use ODBC-postgresql with shared presence
registrations are ok on all nodes. (it works well if only one phone is registered)
  ----- Original Message ----- 
  From: Avi Marcus 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, December 29, 2010 12:26 PM
  Subject: Re: [Freeswitch-users] multiple-registration for same extension


  On Wed, Dec 29, 2010 at 7:11 PM, Madovsky <infos at madovsky.org> wrote:

    yes, 

    <param name="multiple-registrations" value="true"/>
    and restart FS.
    also is it need to put this line in internal.xml or external.xml knowing that
    registrations are effective only in internal.xml ?

    > It's worked for me several times, however, I usually bridge to "user/$extension".
    but if you use sofia_contact in the dialplan bridge it gives to you sofia/$${domain}/extension@$${domain}
    so how to use user/extension in mutliple-registrations ?

    as I use a cluster with SRV NAPTR for outbound calls maybe it's the problem.
    I noticed that if the call arrives in the same node as the one where phones are registered it works.
    if not, only 1 phone or no phone rings. weirdly there is nothing abnormal on debug log
  I guess you're not clustering the DB and replicating the internal registrations across the servers?



      ----- Original Message ----- 
      From: Avi Marcus 
      To: FreeSWITCH Users Help 
      Sent: Wednesday, December 29, 2010 11:44 AM
      Subject: Re: [Freeswitch-users] multiple-registration for same extension


      You set: 
      <param name="multiple-registrations" value="true"/>
      and did a reloadxml, and waited for both to register again?
      It's worked for me several times, however, I usually bridge to "user/$extension".
      -Avi


      On Wed, Dec 29, 2010 at 6:18 PM, Madovsky <infos at madovsky.org> wrote:

        I enabled multiple-registrations to true in internal.xml (which is the difference between true and contact ?)
        added this line to my dialplan

        <action application="bridge" data="${sofia_contact(${dialed_extension}@${domain_name})}"/>

        but only the phone registered lastly rings...

        thanks

        Franck




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






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


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      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org



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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From lloyd.aloysius at gmail.com  Wed Dec 29 20:47:47 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 29 Dec 2010 12:47:47 -0500
Subject: [Freeswitch-users] CentOS 5.5 - FreeSWITCH - Audio Quality
	issues
In-Reply-To: <AE8F4FDB-95EF-49E6-80A1-A91315DCF2E4@freeswitch.org>
References: <AANLkTik3y8EfTi5s_a_1BURnUJe8Nv0p+5y8Ph6Jkcg3@mail.gmail.com>
	<AANLkTi=6shTJg0C2eFsar+mdjstT330e2kaUtFxsPQ1F@mail.gmail.com>
	<AANLkTi=3Yi_Phq6Zv2Ep-YEgf=3orNmOEpLVyww=7x=L@mail.gmail.com>
	<316084A7-BCA6-4490-8990-9894331F999F@freeswitch.org>
	<AANLkTimREKiD=3t12WT7fMWWR7JPyOT+3S4jm7SPY6UT@mail.gmail.com>
	<AANLkTintspeXDQiyHXsu2GmuyiK0yqVKrBKyBmwmY-fB@mail.gmail.com>
	<AE8F4FDB-95EF-49E6-80A1-A91315DCF2E4@freeswitch.org>
Message-ID: <AANLkTinLXcLEbw7qcyFwgyaX5zoa0uENpy9YK3UraND2@mail.gmail.com>

Anthony & Brian - Thanks.

1. Client Make the calls ..

Regular Bell Canada Land line or Cell Phone ---> SIP Provider DID --->
FreeSWITCH --> IVR ->Voicemail

or

Regular Bell Canada Land line or Cell Phone ---> SIP Provider DID --->
FreeSWITCH --> IVR


2. I will capture the trace when there is no calls. This is a production box
. Is there any way to capture a trace for a call when there is other calls
in the box?

3. I decided to switch to 64bit today night let see that will help me.

Thanks and regards,
Lloyd



On Wed, Dec 29, 2010 at 11:55 AM, Brian West <brian at freeswitch.org> wrote:

> YES.  You should always run a 64bit OS on a 64bit CPU... unless you like
> wasting have your cpu registers... but I bet you the problem will go away if
> you switch.
>
> /b
>
> On Dec 29, 2010, at 9:46 AM, Aloysius Lloyd wrote:
>
>  CentOS 64 bit ?
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Dec 29 20:51:42 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 29 Dec 2010 11:51:42 -0600
Subject: [Freeswitch-users] multiple-registration for same extension
In-Reply-To: <1DC63D7EFF8F41BA897BF0C13A48EAB9@e1705>
References: <568973BF49BD4CE39418F547BA3E5712@e1705><AANLkTimhuvjtrcSBWgy=kt4HkGhqc8Y1MJDVo+j6x-E2@mail.gmail.com><7BCEC2EFD5EE410594C3C47309FFDE84@e1705>
	<AANLkTikm1i5qSOCib0BRwo2gLPjuufFoFcUSQ4O-767s@mail.gmail.com>
	<1DC63D7EFF8F41BA897BF0C13A48EAB9@e1705>
Message-ID: <3ABD9AC4-9C33-42AA-9AFD-6F7F22943C5D@freeswitch.org>

What kind of phone are you using again?

/b

On Dec 29, 2010, at 11:35 AM, Madovsky wrote:

> yes it is. use ODBC-postgresql with shared presence
> registrations are ok on all nodes. (it works well if only one phone is registered)
> 

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From infos at madovsky.org  Wed Dec 29 21:05:26 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 29 Dec 2010 13:05:26 -0500
Subject: [Freeswitch-users] multiple-registration for same extension
References: <568973BF49BD4CE39418F547BA3E5712@e1705><AANLkTimhuvjtrcSBWgy=kt4HkGhqc8Y1MJDVo+j6x-E2@mail.gmail.com><7BCEC2EFD5EE410594C3C47309FFDE84@e1705><AANLkTikm1i5qSOCib0BRwo2gLPjuufFoFcUSQ4O-767s@mail.gmail.com><1DC63D7EFF8F41BA897BF0C13A48EAB9@e1705>
	<3ABD9AC4-9C33-42AA-9AFD-6F7F22943C5D@freeswitch.org>
Message-ID: <66892B5022FA462CB91FCCA81EAA50BF@e1705>

I try with x-lite v3 and v4
  ----- Original Message ----- 
  From: Brian West 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, December 29, 2010 12:51 PM
  Subject: Re: [Freeswitch-users] multiple-registration for same extension


  What kind of phone are you using again?


  /b


  On Dec 29, 2010, at 11:35 AM, Madovsky wrote:


    yes it is. use ODBC-postgresql with shared presence
    registrations are ok on all nodes. (it works well if only one phone is registered)






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From rajkumar.kmry at gmail.com  Wed Dec 29 15:54:01 2010
From: rajkumar.kmry at gmail.com (rajkumar)
Date: Wed, 29 Dec 2010 04:54:01 -0800 (PST)
Subject: [Freeswitch-users] bridge application dial string
In-Reply-To: <39788A22-DE3A-46C9-AC45-1597CC339A19@gmail.com>
References: <1293615145403-5874073.post@n2.nabble.com>
	<39788A22-DE3A-46C9-AC45-1597CC339A19@gmail.com>
Message-ID: <1293627241963-5874416.post@n2.nabble.com>


Thanks, loopback works fine to land the call on particular extension.

regards
rajkumar k
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/bridge-application-dial-string-tp5874073p5874416.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From telconettech at yahoo.com  Wed Dec 29 21:56:16 2010
From: telconettech at yahoo.com (Joel Mange)
Date: Wed, 29 Dec 2010 10:56:16 -0800 (PST)
Subject: [Freeswitch-users] event i_media_error 988 Incomplete
	offer/answer
Message-ID: <520133.78693.qm@web110803.mail.gq1.yahoo.com>

Problem resolved -- thank you for pointing me in the right direction.

Was
following a Cisco application guide for router configuration, which
specified the following voice service voip command:? signaling forward
unconditional.? This enabled SIP GTD.

Following command disabled SIP GTD and corrected issue for me:? signaling forward none.

Thanks again.

--- On Tue, 12/28/10, Joel Mange <telconettech at yahoo.com> wrote:

From: Joel Mange <telconettech at yahoo.com>
Subject: event i_media_error 988 Incomplete offer/answer
To: freeswitch-users at lists.freeswitch.org
Date: Tuesday, December 28, 2010, 7:35 PM

Hello.? New to FS and in need of help.

Environment is as follows:

Linksys SPA-962 (x94140) <---> FS <---sip---> Cisco3845 <---qsig---> TDM PBX <--> TDM phone
 (x91237)

Can complete call from TDM phone to Linksys SPA-962.? Can't complete call from Linksys to TDM phone:? TDM phone rings, but call fails when it goes off-hook.? Getting the following sofia error:? nua(0x8b60cf0): event i_media_error 988 Incomplete offer/answer.

I think the above message tells me that the other side of the call sent an invalid SDP.? Having trouble identifying the invalid SDP and making necessary correction.

Siptrace and router running-config at:??http://pastebin.freeswitch.org/14896

Any help appreciated.


      


      
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From infos at madovsky.org  Wed Dec 29 22:00:01 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 29 Dec 2010 14:00:01 -0500
Subject: [Freeswitch-users] multiple-registration for same extension
References: <568973BF49BD4CE39418F547BA3E5712@e1705><AANLkTimhuvjtrcSBWgy=kt4HkGhqc8Y1MJDVo+j6x-E2@mail.gmail.com><7BCEC2EFD5EE410594C3C47309FFDE84@e1705>
	<AANLkTikm1i5qSOCib0BRwo2gLPjuufFoFcUSQ4O-767s@mail.gmail.com>
Message-ID: <D7128006F867444E82E49A17F279D541@e1705>

I have this in directory/default.xml

<param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>

tried this in my dialplan :
<action application="bridge" data="user/${dialed_extension}"/>
<action application="bridge" data="${sofia_contact(${dialed_user}@${dialed_domain})}"/>

this in internal.xml :
<param name="multiple-registrations" value="true"/>
    <param name="manage-presence" value="true"/>
    <param name="manage-shared-appearance" value="true"/>
    <param name="dbname" value="share_presence"/>
    <param name="presence-hosts" value="$${domain}"/>

so when an outbound call arrives :
- if one phone with one extension is registered it works well, it receives the call no matter from whatever node
- if 2 phones with same extension are registered on 2 different nodes it works well now
- if 2 phones with same extension are registered on the same node only 1 phone rings (the latest registered)

tell if I need to pastebin the logs (don't remember the link)

Thanks


  ----- Original Message ----- 
  From: Avi Marcus 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, December 29, 2010 12:26 PM
  Subject: Re: [Freeswitch-users] multiple-registration for same extension


  On Wed, Dec 29, 2010 at 7:11 PM, Madovsky <infos at madovsky.org> wrote:

    yes, 

    <param name="multiple-registrations" value="true"/>
    and restart FS.
    also is it need to put this line in internal.xml or external.xml knowing that
    registrations are effective only in internal.xml ?

    > It's worked for me several times, however, I usually bridge to "user/$extension".
    but if you use sofia_contact in the dialplan bridge it gives to you sofia/$${domain}/extension@$${domain}
    so how to use user/extension in mutliple-registrations ?

    as I use a cluster with SRV NAPTR for outbound calls maybe it's the problem.
    I noticed that if the call arrives in the same node as the one where phones are registered it works.
    if not, only 1 phone or no phone rings. weirdly there is nothing abnormal on debug log
  I guess you're not clustering the DB and replicating the internal registrations across the servers?



      ----- Original Message ----- 
      From: Avi Marcus 
      To: FreeSWITCH Users Help 
      Sent: Wednesday, December 29, 2010 11:44 AM
      Subject: Re: [Freeswitch-users] multiple-registration for same extension


      You set: 
      <param name="multiple-registrations" value="true"/>
      and did a reloadxml, and waited for both to register again?
      It's worked for me several times, however, I usually bridge to "user/$extension".
      -Avi


      On Wed, Dec 29, 2010 at 6:18 PM, Madovsky <infos at madovsky.org> wrote:

        I enabled multiple-registrations to true in internal.xml (which is the difference between true and contact ?)
        added this line to my dialplan

        <action application="bridge" data="${sofia_contact(${dialed_extension}@${domain_name})}"/>

        but only the phone registered lastly rings...

        thanks

        Franck




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From saeedahmad1981 at gmail.com  Wed Dec 29 22:04:14 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Wed, 29 Dec 2010 20:04:14 +0100
Subject: [Freeswitch-users] Asynchronous PTIME
In-Reply-To: <4D15266B.3080903@sns.eu>
References: <4D127733.8010409@nevian.org> <4D15266B.3080903@sns.eu>
Message-ID: <AANLkTimzmg1kfM4+75e-Y9vG6wErhXjX6f83+4-voNyu@mail.gmail.com>

on which profile did you put that param?

I've allnet phone if i use g729 with ptime:40 then i can't hear called party
properly, its choppy voice, i put autofix param in both internal and
external profiles, but it didn't help.

if i use ptime: 20 then it works fine.

On Sat, Dec 25, 2010 at 12:02 AM, Jan Riedinger <riedinger at sns.eu> wrote:

> Hi Serge,
>
> I had similar problems, look for my thread "Problematic Behaviour of FS
> regarding ptime negotiation" in October. I could solve the problem by
> setting "rtp-autofix-timing=false", which disabled the (too) smart
> behaviour of FreeSwitch.
>
> BR
>     Jan
>
>
> Am 22.12.2010 23:09, schrieb Serge S. Yuriev:
> > Hello,
> >
> > 2010-12-20 16:51:13.960502 [WARNING] mod_sofia.c:1036 Asynchronous PTIME
> > not supported, changing our end from 20 to 60
> >
> > I'm getting this warning and client hears chopped sound :(
> > That is "Our end"?
> >
> > Call flow is Cisco (sip) FS (sip) YATE (h323) MeraTransitSoftSwitch
> > All but MVTS under my control.
> >
> > I doesn't see any clue in logs and can't reproduce this with my testing
> > via Blink/PortSIP/PhonerLite/Twinkle or Grandstream devices
> >
> > Which debug/logs I should take? Any ideas?
> >
> > Thanks a lot.
> >
> > btw how I can save debug into log not only console?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Wed Dec 29 23:01:08 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 29 Dec 2010 15:01:08 -0500
Subject: [Freeswitch-users] multiple-registration for same extension
Message-ID: <E2834B28618E4F58899C450744741998@e1705>

ok I think the problem is not FS...

thanks

  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, December 29, 2010 2:00 PM
  Subject: Re: [Freeswitch-users] multiple-registration for same extension


  I have this in directory/default.xml

  <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>

  tried this in my dialplan :
  <action application="bridge" data="user/${dialed_extension}"/>
  <action application="bridge" data="${sofia_contact(${dialed_user}@${dialed_domain})}"/>

  this in internal.xml :
  <param name="multiple-registrations" value="true"/>
      <param name="manage-presence" value="true"/>
      <param name="manage-shared-appearance" value="true"/>
      <param name="dbname" value="share_presence"/>
      <param name="presence-hosts" value="$${domain}"/>

  so when an outbound call arrives :
  - if one phone with one extension is registered it works well, it receives the call no matter from whatever node
  - if 2 phones with same extension are registered on 2 different nodes it works well now
  - if 2 phones with same extension are registered on the same node only 1 phone rings (the latest registered)

  tell if I need to pastebin the logs (don't remember the link)

  Thanks


    ----- Original Message ----- 
    From: Avi Marcus 
    To: FreeSWITCH Users Help 
    Sent: Wednesday, December 29, 2010 12:26 PM
    Subject: Re: [Freeswitch-users] multiple-registration for same extension


    On Wed, Dec 29, 2010 at 7:11 PM, Madovsky <infos at madovsky.org> wrote:

      yes, 

      <param name="multiple-registrations" value="true"/>
      and restart FS.
      also is it need to put this line in internal.xml or external.xml knowing that
      registrations are effective only in internal.xml ?

      > It's worked for me several times, however, I usually bridge to "user/$extension".
      but if you use sofia_contact in the dialplan bridge it gives to you sofia/$${domain}/extension@$${domain}
      so how to use user/extension in mutliple-registrations ?

      as I use a cluster with SRV NAPTR for outbound calls maybe it's the problem.
      I noticed that if the call arrives in the same node as the one where phones are registered it works.
      if not, only 1 phone or no phone rings. weirdly there is nothing abnormal on debug log
    I guess you're not clustering the DB and replicating the internal registrations across the servers?



        ----- Original Message ----- 
        From: Avi Marcus 
        To: FreeSWITCH Users Help 
        Sent: Wednesday, December 29, 2010 11:44 AM
        Subject: Re: [Freeswitch-users] multiple-registration for same extension


        You set: 
        <param name="multiple-registrations" value="true"/>
        and did a reloadxml, and waited for both to register again?
        It's worked for me several times, however, I usually bridge to "user/$extension".
        -Avi


        On Wed, Dec 29, 2010 at 6:18 PM, Madovsky <infos at madovsky.org> wrote:

          I enabled multiple-registrations to true in internal.xml (which is the difference between true and contact ?)
          added this line to my dialplan

          <action application="bridge" data="${sofia_contact(${dialed_extension}@${domain_name})}"/>

          but only the phone registered lastly rings...

          thanks

          Franck




          _______________________________________________
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          FreeSWITCH-users at lists.freeswitch.org
          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From moises.silva at gmail.com  Wed Dec 29 23:09:03 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 29 Dec 2010 15:09:03 -0500
Subject: [Freeswitch-users] ftmod_sangoma_isdn: dialtone in overlap
	dialing
In-Reply-To: <AANLkTi=qD8nqq2yqpKPHcs88oVDxcBN4wSqiaKkQweHb@mail.gmail.com>
References: <AANLkTi=qD8nqq2yqpKPHcs88oVDxcBN4wSqiaKkQweHb@mail.gmail.com>
Message-ID: <AANLkTi=PZw7r0uYbmtbgO6-qGN64tMCgENU+GiS7=VLV@mail.gmail.com>

On Wed, Dec 29, 2010 at 10:15 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Dear friends,
>   I would want to know how to configure ftmod_sangoma_isdn to send
> dialtone.
>
> My setup is as follows,
>
> Telcom <--->  Freeswitch <---> PBX <----> Extensions 1..n
>
>
> My pbx has got overlap dial enabled and it is configured to seize the trunk
> when it receives '0' from extensions.
>
> When I connect the PRI to my pbx, and when '0' is dialled from an
> extension, my pbx seizes the trunk and I'm able to hear dialtone. But, when
> I connect the PRI to freeswitch and connect freeswitch to pbx and when '0'
> is dialled from an extension, I'm not hearing dialtone.
>
> Whether some configuration is available to ask ftmod_sangoma_isdn to send
> dialtone before started to collect digits in overlap mode? or any other way
> to give dialtone??
>
>
Not supported with the current code. Some changes need to be performed in
the ftmod_sangoma_isdn module to do this. I'd recommend you to open a jira
report to ask for this feature, as I know is something is needed and would
like to keep track of it and put it in our next development cycle.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com
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From msc at freeswitch.org  Thu Dec 30 01:24:14 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Dec 2010 14:24:14 -0800
Subject: [Freeswitch-users] mod callcenter
In-Reply-To: <AANLkTikdRNMo4CgFcuon0X4KkyZSzZYHK6RECaDxrNwa@mail.gmail.com>
References: <AANLkTikdRNMo4CgFcuon0X4KkyZSzZYHK6RECaDxrNwa@mail.gmail.com>
Message-ID: <AANLkTin0_cUT1Wb-SanDj_sD_JWnrzJ_QC1YpUMbBDc0@mail.gmail.com>

Open a tick on jira.freeswitch.org and Moc will take a look. Be sure to
provide as much info as possible.
-MC

On Wed, Dec 29, 2010 at 6:48 AM, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
> Was testing callcenter module and found out that at times it gives error "
> invalid application callcenter " and after
> reloading the module it works fine.
> Some time also happens that if I reload the module it do not reloads the
> parameters of the callcenter like the agents & tires.
> It just unload & reloads even if there are changes to the specifications of
> agents.
>
> Regds
> Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ralph at eckhard.nl  Thu Dec 30 01:26:23 2010
From: ralph at eckhard.nl (Ralph Eckhard)
Date: Wed, 29 Dec 2010 23:26:23 +0100
Subject: [Freeswitch-users] Briding two sip-trunks
Message-ID: <C1F84D446E640840881E12DDC1D7BD6C062391@server01.eckhard.local>

Hi there,

 

I'm trying to use FreeSwitch to connect my MS Lync (previously OCS) to a
SIP-trunk. Problem is that Lync only works using TCP-based trunks, while
my SIP-provider uses UDP.

I understand that by using FreeSwitch I can bridge these two protocols. 

I installed FreeSwitch on my Lync server and configured both the
external (to sip-provider) and internal (to Lync) sip profiles. These
both work; when i dial in from remote to a DDI on my trunk, i can see
the incoming call in the FS CLI. Same goes for outgoing calls; when
dialing a number from my lync-client, the FS CLI logs the call. 

 

Now, what do i config to bridge the two trunks? I don't need any extra
extensions or otherwise, all call handling is done by Lync. All calls
coming in from my external sip trunk need to be forwarded to my internal
sip, and all calls originating from my internal sip need to be forwarded
to my external sip trunk.

 

Can someone point me in the right direction?

 

Thanks in advance ;)

 

Ralph

 

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From bcxml at hotmail.com  Thu Dec 30 03:15:27 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Wed, 29 Dec 2010 19:15:27 -0500
Subject: [Freeswitch-users] Briding two sip-trunks
In-Reply-To: <C1F84D446E640840881E12DDC1D7BD6C062391@server01.eckhard.local>
References: <C1F84D446E640840881E12DDC1D7BD6C062391@server01.eckhard.local>
Message-ID: <SNT135-w144634BD17C0EC497DB4E5B7030@phx.gbl>


Hi Ralph
 
I have done a fair bit of work hooking FreeSwitch to Microsoft Speech Server 2007..same issue...Speech Server only does SIP over TCP
 
I have documented the whole process that I went through in a 6 part blog series over at www.gotspeech.net. 
 
Part 1 of the series can be found here http://gotspeech.net/blogs/verbalinput/archive/2009/12.aspx
 
I am still getting up to speed with Lync 2010, so I dont know whether the process is the same as it was for Speech Server 2007, but hopefully the blog postings will help you out
 
Cheers
 
 
Brian Campbell
Speech Developer
Micro Design Centre Inc
905-918-3027
 


Date: Wed, 29 Dec 2010 23:26:23 +0100
From: ralph at eckhard.nl
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Briding two sip-trunks






Hi there,
 
I?m trying to use FreeSwitch to connect my MS Lync (previously OCS) to a SIP-trunk. Problem is that Lync only works using TCP-based trunks, while my SIP-provider uses UDP.
I understand that by using FreeSwitch I can bridge these two protocols. 
I installed FreeSwitch on my Lync server and configured both the external (to sip-provider) and internal (to Lync) sip profiles. These both work; when i dial in from remote to a DDI on my trunk, i can see the incoming call in the FS CLI. Same goes for outgoing calls; when dialing a number from my lync-client, the FS CLI logs the call. 
 
Now, what do i config to bridge the two trunks? I don?t need any extra extensions or otherwise, all call handling is done by Lync. All calls coming in from my external sip trunk need to be forwarded to my internal sip, and all calls originating from my internal sip need to be forwarded to my external sip trunk.
 
Can someone point me in the right direction?
 
Thanks in advance ;)
 
Ralph
 
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 		 	   		  
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From msc at freeswitch.org  Thu Dec 30 03:16:44 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Dec 2010 16:16:44 -0800
Subject: [Freeswitch-users] New FS sounds testing
Message-ID: <AANLkTi=bRueAXGhqoi7svhkeLeREDsORKL6=L-n3PkqZ@mail.gmail.com>

FreeSWITCHers,

We have the new sounds rolled and I'd like to do some expanded testing
before I update in git. If anyone wants to get the latest sounds then go to
your src directory and edit build/sounds_version.txt. Change en-us-callie
from 1.0.13 to 1.0.14 and then do "make cd-sounds-install". The latest
sounds should get installed. Please try this on your non-production systems
first! :) Please reply to me off-list and let me know if you run into any
problems or if the sounds are okay. I would like feedback either way. If we
don't have any issues then I'll bump the sounds version in git and everyone
will be good to go.

Thanks!
-Michael

P.S. - If you would like to review which sound prompts are new to 1.0.14
then check this commit:
http://fisheye.freeswitch.org/browse/freeswitch.git/docs/phrase/phrase_en.xml?r1=ee051faef39ff78d7805f1eb8c9f6330a4258032&r2=257c7edaf7d7e6151d11e5ef924d87a77f2c369b
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From msc at freeswitch.org  Thu Dec 30 03:28:22 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Dec 2010 16:28:22 -0800
Subject: [Freeswitch-users] Briding two sip-trunks
In-Reply-To: <C1F84D446E640840881E12DDC1D7BD6C062391@server01.eckhard.local>
References: <C1F84D446E640840881E12DDC1D7BD6C062391@server01.eckhard.local>
Message-ID: <AANLkTinyoO2fgROU3tcxfnUh5D75k6FkS=yQXfZiUHPF@mail.gmail.com>

Is security an issue? Or will all calls behave exactly the same? You can use
acl.conf.xml for letting in only certain IP addrs if need be. In any case,
you could add something like this in conf/dialplan/public.xml:

<extension name="fwd to external profile">
  <condition field="destination_number" expression="([your DID number])">
    <action application="bridge" data="sofia/internal/$1 at x.x.x.x:yyyy"/>
  </condition>
</extension>

Where x.x.x.x:yyyy is the IP address and port that Lync is listening on.
Give that a try and see what happens... :)

-MC

On Wed, Dec 29, 2010 at 2:26 PM, Ralph Eckhard <ralph at eckhard.nl> wrote:

> Hi there,
>
>
>
> I?m trying to use FreeSwitch to connect my MS Lync (previously OCS) to a
> SIP-trunk. Problem is that Lync only works using TCP-based trunks, while my
> SIP-provider uses UDP.
>
> I understand that by using FreeSwitch I can bridge these two protocols.
>
> I installed FreeSwitch on my Lync server and configured both the external
> (to sip-provider) and internal (to Lync) sip profiles. These both work; when
> i dial in from remote to a DDI on my trunk, i can see the incoming call in
> the FS CLI. Same goes for outgoing calls; when dialing a number from my
> lync-client, the FS CLI logs the call.
>
>
>
> Now, what do i config to bridge the two trunks? I don?t need any extra
> extensions or otherwise, all call handling is done by Lync. All calls coming
> in from my external sip trunk need to be forwarded to my internal sip, and
> all calls originating from my internal sip need to be forwarded to my
> external sip trunk.
>
>
>
> Can someone point me in the right direction?
>
>
>
> Thanks in advance ;)
>
>
>
> Ralph
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Dec 30 03:32:14 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Dec 2010 16:32:14 -0800
Subject: [Freeswitch-users] DTMF missing
In-Reply-To: <AANLkTi=S6YBwVQ4Bk6xSRMiPqqHDRuTKfhJwqwzKTFbr@mail.gmail.com>
References: <AANLkTik=NLa+injL5hF=VsymwoE9wd1Wb2joXJDhvfFZ@mail.gmail.com>
	<AANLkTincAApO4JDwzt-5oX0iD2VozFNVCtembOumL4iS@mail.gmail.com>
	<AANLkTin14RN-WxkYzqkQPRZctYt09xkWguAoCr8eeUWe@mail.gmail.com>
	<AANLkTik3N+F6Li6OL9EmyDH34eowxR+ttBFZx=1e83Eg@mail.gmail.com>
	<AANLkTikAKqXyFDQGTaBxW6OhS7pVzUzT_m3n7mZmAfxc@mail.gmail.com>
	<AANLkTi=3N59_nwtiej=Xbtiodp95U_FYUpOPamN=fzYH@mail.gmail.com>
	<AANLkTikPrBiJ500DxzVkqJjHht0RBLq_WjsOEEWJtSOL@mail.gmail.com>
	<48F39398-0A59-4108-80F2-B6663B961DED@freeswitch.org>
	<AANLkTimPDPNQsbjQZCfGgRTFLeQd2zUA85HCU8e5K_fL@mail.gmail.com>
	<AANLkTi=S6YBwVQ4Bk6xSRMiPqqHDRuTKfhJwqwzKTFbr@mail.gmail.com>
Message-ID: <AANLkTikGyY19dtKNnY3S-qRBmgZwhm6BrrZBBw0OGvkL@mail.gmail.com>

I do not believe you have ruled out the client misbehaving. Record the call
and listen to the wav to see if the DTMFs are happening in band or not. You
can also tcpdump the traffic to/from the client and analyze in Wireshark.
You can look at VoIP calls and if you do a graph it will show any DTMFs sent
via RFC2833.

The likelihood that FS is broken is extremely small compared to the
likelihood that your client is doing something silly.
-MC

On Wed, Dec 29, 2010 at 1:38 AM, Sam <u2nsam at gmail.com> wrote:

> With the same setting which was working few hours ago and now its not
> working.
> I donot see DTMF on fs cli now
>
> Regds
> Sam
>
>
> On Wed, Dec 29, 2010 at 11:29 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Hi,
>>
>> All Kool ! it was because of <variable name="bypass_media" value="0"/> ,
>> what is the meaning of the "value" here ? or its a wrong syntax
>>
>> Regds
>> Sam
>>
>>
>>
>>
>>
>> On Wed, Dec 29, 2010 at 10:09 AM, Brian West <brian at freeswitch.org>wrote:
>>
>>> Ok here is the deal... RFC2833 is in the media... if you have bypass on
>>> how do you ever expect to have the DTMF in FreeSWITCH?
>>>
>>> /b
>>>
>>> On Dec 28, 2010, at 10:29 PM, Sam wrote:
>>>
>>> > also have used <variable name="bypass_media" value="0"/>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Thu Dec 30 03:34:55 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 29 Dec 2010 18:34:55 -0600
Subject: [Freeswitch-users] DTMF missing
In-Reply-To: <AANLkTikGyY19dtKNnY3S-qRBmgZwhm6BrrZBBw0OGvkL@mail.gmail.com>
References: <AANLkTik=NLa+injL5hF=VsymwoE9wd1Wb2joXJDhvfFZ@mail.gmail.com>
	<AANLkTincAApO4JDwzt-5oX0iD2VozFNVCtembOumL4iS@mail.gmail.com>
	<AANLkTin14RN-WxkYzqkQPRZctYt09xkWguAoCr8eeUWe@mail.gmail.com>
	<AANLkTik3N+F6Li6OL9EmyDH34eowxR+ttBFZx=1e83Eg@mail.gmail.com>
	<AANLkTikAKqXyFDQGTaBxW6OhS7pVzUzT_m3n7mZmAfxc@mail.gmail.com>
	<AANLkTi=3N59_nwtiej=Xbtiodp95U_FYUpOPamN=fzYH@mail.gmail.com>
	<AANLkTikPrBiJ500DxzVkqJjHht0RBLq_WjsOEEWJtSOL@mail.gmail.com>
	<48F39398-0A59-4108-80F2-B6663B961DED@freeswitch.org>
	<AANLkTimPDPNQsbjQZCfGgRTFLeQd2zUA85HCU8e5K_fL@mail.gmail.com>
	<AANLkTi=S6YBwVQ4Bk6xSRMiPqqHDRuTKfhJwqwzKTFbr@mail.gmail.com>
	<AANLkTikGyY19dtKNnY3S-qRBmgZwhm6BrrZBBw0OGvkL@mail.gmail.com>
Message-ID: <8A916367-6FB9-48BC-AA3F-819473B39253@freeswitch.org>

Unless he updates to the very latest code cuz it was not quite right for a small window last week.

/b

On Dec 29, 2010, at 6:32 PM, Michael Collins wrote:

> I do not believe you have ruled out the client misbehaving. Record the call and listen to the wav to see if the DTMFs are happening in band or not. You can also tcpdump the traffic to/from the client and analyze in Wireshark. You can look at VoIP calls and if you do a graph it will show any DTMFs sent via RFC2833.
> 
> The likelihood that FS is broken is extremely small compared to the likelihood that your client is doing something silly.
> -MC
> 
> On Wed, Dec 29, 2010 at 1:38 AM, Sam <u2nsam at gmail.com> wrote:
> With the same setting which was working few hours ago and now its not working.
> I donot see DTMF on fs cli now
> 
> Regds
> Sam

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From infos at madovsky.org  Thu Dec 30 06:02:05 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 29 Dec 2010 22:02:05 -0500
Subject: [Freeswitch-users] unregister extension from cli
Message-ID: <AA6CDEE43E4F48DCA7EEA9E43DF6BFAA@e1705>

is there any way to unregister an extension from CLI ?
for example I register my phone with 123456 and from CLI
I disconnect it..

Thanks
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From hwnorman at hotmail.com  Thu Dec 30 06:08:48 2010
From: hwnorman at hotmail.com (Norman Lam)
Date: Thu, 30 Dec 2010 11:08:48 +0800
Subject: [Freeswitch-users] mutilple destination phone ringing on incoming
	call, skypopen dial plan from Sip X-lite
Message-ID: <SNT109-DS23F413FBA3EA84EED0CE72C1030@phx.gbl>

Hi Everyone 

 

1.       Could anybody  tell me how to have multiple destination ringing
from an incoming skypopen, 

Currently is set to  <param name="destination" value="1000"/> 

      But I want it to ring on 1001, 1002,1003, and so on. 

2.        Does anybody have a outgoing  dial plan from sip X-lite to
skypopen ( skype network), I try looking at the Mod skypopen Skype Endpoint
and Trunk wiki site but couldn't figure it out. 

Please advise , Thank in advance 

Norman 

 

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From nagalenoj at gmail.com  Thu Dec 30 07:54:02 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Thu, 30 Dec 2010 10:24:02 +0530
Subject: [Freeswitch-users] ftmod_libpri: support for overlap dial
Message-ID: <AANLkTikxnBQk5=3p0_HRJ4LySpUWSLSjhz-=THzUQmJb@mail.gmail.com>

Dear Friends,
    When I tried ftmod_libpri to check whether it supports overlap dial. It
didn't work. So, is there anyone trying to implement overlap dial in
ftmod_libpri?

-- 
Regards,
Nagalenoj H.
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From u2nsam at gmail.com  Thu Dec 30 08:12:48 2010
From: u2nsam at gmail.com (Sam)
Date: Thu, 30 Dec 2010 10:42:48 +0530
Subject: [Freeswitch-users] CPS
Message-ID: <AANLkTi=GEYTXe4MREfxeSaZc5y==Mv4e9rQCoJVHE2ES@mail.gmail.com>

Hello,

Was testing the environment with media bypass, could see the CPU for FS
application shooting to 200% on quadracore 64 bits debian for CPS of 70.

Is it true for the environment or can be scaled further ?

Regards
Sam
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From nagalenoj at gmail.com  Thu Dec 30 08:19:15 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Thu, 30 Dec 2010 10:49:15 +0530
Subject: [Freeswitch-users] ftmod_sangoma_isdn: dialtone in overlap
	dialing
In-Reply-To: <AANLkTi=PZw7r0uYbmtbgO6-qGN64tMCgENU+GiS7=VLV@mail.gmail.com>
References: <AANLkTi=qD8nqq2yqpKPHcs88oVDxcBN4wSqiaKkQweHb@mail.gmail.com>
	<AANLkTi=PZw7r0uYbmtbgO6-qGN64tMCgENU+GiS7=VLV@mail.gmail.com>
Message-ID: <AANLkTin7NhEAV1=YMkBGU=-nJtB8RscS_kpYbPT_wu7y@mail.gmail.com>

http://jira.freeswitch.org/browse/OPENZAP-134

Thanks.

On Thu, Dec 30, 2010 at 1:39 AM, Moises Silva <moises.silva at gmail.com>wrote:

> On Wed, Dec 29, 2010 at 10:15 AM, Nagalenoj H. <nagalenoj at gmail.com>wrote:
>
>> Dear friends,
>>   I would want to know how to configure ftmod_sangoma_isdn to send
>> dialtone.
>>
>> My setup is as follows,
>>
>> Telcom <--->  Freeswitch <---> PBX <----> Extensions 1..n
>>
>>
>> My pbx has got overlap dial enabled and it is configured to seize the
>> trunk when it receives '0' from extensions.
>>
>> When I connect the PRI to my pbx, and when '0' is dialled from an
>> extension, my pbx seizes the trunk and I'm able to hear dialtone. But, when
>> I connect the PRI to freeswitch and connect freeswitch to pbx and when '0'
>> is dialled from an extension, I'm not hearing dialtone.
>>
>> Whether some configuration is available to ask ftmod_sangoma_isdn to send
>> dialtone before started to collect digits in overlap mode? or any other way
>> to give dialtone??
>>
>>
> Not supported with the current code. Some changes need to be performed in
> the ftmod_sangoma_isdn module to do this. I'd recommend you to open a jira
> report to ask for this feature, as I know is something is needed and would
> like to keep track of it and put it in our next development cycle.
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R
> 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
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From u2nsam at gmail.com  Thu Dec 30 08:53:48 2010
From: u2nsam at gmail.com (Sam)
Date: Thu, 30 Dec 2010 11:23:48 +0530
Subject: [Freeswitch-users] DTMF missing
In-Reply-To: <8A916367-6FB9-48BC-AA3F-819473B39253@freeswitch.org>
References: <AANLkTik=NLa+injL5hF=VsymwoE9wd1Wb2joXJDhvfFZ@mail.gmail.com>
	<AANLkTincAApO4JDwzt-5oX0iD2VozFNVCtembOumL4iS@mail.gmail.com>
	<AANLkTin14RN-WxkYzqkQPRZctYt09xkWguAoCr8eeUWe@mail.gmail.com>
	<AANLkTik3N+F6Li6OL9EmyDH34eowxR+ttBFZx=1e83Eg@mail.gmail.com>
	<AANLkTikAKqXyFDQGTaBxW6OhS7pVzUzT_m3n7mZmAfxc@mail.gmail.com>
	<AANLkTi=3N59_nwtiej=Xbtiodp95U_FYUpOPamN=fzYH@mail.gmail.com>
	<AANLkTikPrBiJ500DxzVkqJjHht0RBLq_WjsOEEWJtSOL@mail.gmail.com>
	<48F39398-0A59-4108-80F2-B6663B961DED@freeswitch.org>
	<AANLkTimPDPNQsbjQZCfGgRTFLeQd2zUA85HCU8e5K_fL@mail.gmail.com>
	<AANLkTi=S6YBwVQ4Bk6xSRMiPqqHDRuTKfhJwqwzKTFbr@mail.gmail.com>
	<AANLkTikGyY19dtKNnY3S-qRBmgZwhm6BrrZBBw0OGvkL@mail.gmail.com>
	<8A916367-6FB9-48BC-AA3F-819473B39253@freeswitch.org>
Message-ID: <AANLkTimbUY+ifTRsSmSuU6ORB8oPMN027CX3vjf4QGMB@mail.gmail.com>

Currently it working , will get traces/dump when not working.

git-4e95227 2010-12-26 09-09-14 -0600

Regds
Sam

On Thu, Dec 30, 2010 at 6:04 AM, Brian West <brian at freeswitch.org> wrote:

> Unless he updates to the very latest code cuz it was not quite right for a
> small window last week.
>
> /b
>
> On Dec 29, 2010, at 6:32 PM, Michael Collins wrote:
>
> I do not believe you have ruled out the client misbehaving. Record the call
> and listen to the wav to see if the DTMFs are happening in band or not. You
> can also tcpdump the traffic to/from the client and analyze in Wireshark.
> You can look at VoIP calls and if you do a graph it will show any DTMFs sent
> via RFC2833.
>
> The likelihood that FS is broken is extremely small compared to the
> likelihood that your client is doing something silly.
> -MC
>
> On Wed, Dec 29, 2010 at 1:38 AM, Sam <u2nsam at gmail.com> wrote:
>
>> With the same setting which was working few hours ago and now its not
>> working.
>> I donot see DTMF on fs cli now
>>
>> Regds
>> Sam
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lists at infosecurity.ch  Thu Dec 30 11:45:30 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Thu, 30 Dec 2010 09:45:30 +0100
Subject: [Freeswitch-users] Moving RTP relay to another FS BOX?
Message-ID: <4D1C46AA.3060702@infosecurity.ch>

Hi all,

i have an infrastructure with a single FS server that does RTP
conditional RTP relay.

My server is in Europe but now i have VoIP clients that need to use the
server from South America and India.

Obviously two peers coming from "South america" need to bounce all their
traffic trough Europe increasing the delay.

I would like to install one FS server on each continent and implement a
dynamic geo-ip based logic to have two clients use the RTP relay near to
them.

All VoIP clients connect to a single FS server.

How it's possible to install a 2nd FS (or RTPRelay or whatever) so that
if the two clients are coming from a different geographical area,
redirect their RTP relay to the FS near to them?

I am missing how to do it with FS, i just need 1 central server for SIP
signaling and multiple FS acting as RTP relay with a custom server side
components that looks are the SIP client IP address and address the VoIP
clients's RTP flows to a 2nd FS box, near to them.

How to address VoIP client's RTP flow to another FS server?

Fabio


From Avi at aMarcus.com  Thu Dec 30 13:15:24 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Thu, 30 Dec 2010 12:15:24 +0200
Subject: [Freeswitch-users] mutilple destination phone ringing on
 incoming call, skypopen dial plan from Sip X-lite
In-Reply-To: <SNT109-DS23F413FBA3EA84EED0CE72C1030@phx.gbl>
References: <SNT109-DS23F413FBA3EA84EED0CE72C1030@phx.gbl>
Message-ID: <AANLkTim0o1CeMUykcmSYJEPCfkG40d_6-4iYim+ZRxh7@mail.gmail.com>

For #1, I don't know anything about skypopen, but you could set the
destination to be a hunt group or some other extension that you script to
ring multiple places.
-Avi

On Thu, Dec 30, 2010 at 5:08 AM, Norman Lam <hwnorman at hotmail.com> wrote:

>  Hi Everyone
>
>
>
> 1.       Could anybody  tell me how to have multiple destination ringing
> from an incoming skypopen,
>
> Currently is set to  <param name="destination" value="1000"/>
>
>       But I want it to ring on 1001, 1002,1003, and so on.
>
> 2.        Does anybody have a outgoing  dial plan from sip X-lite to
> skypopen ( skype network), I try looking at the Mod skypopen Skype
> Endpoint and Trunk wiki site but couldn?t figure it out.
>
> Please advise , Thank in advance
>
> Norman
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Thu Dec 30 13:43:27 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 30 Dec 2010 10:43:27 +0000
Subject: [Freeswitch-users] Moving RTP relay to another FS BOX?
In-Reply-To: <4D1C46AA.3060702@infosecurity.ch>
References: <4D1C46AA.3060702@infosecurity.ch>
Message-ID: <AANLkTime6GQtbQS3x2GC3yQh_=fGnMafYg12FWwUmgOe@mail.gmail.com>

Try bridging the call from the central SIP FS server to the regional
FS server, with bypass_media=true on the central server and =false on
the regional ones.

C1=client 1, C2=client2, FS1=central, FS2=regional

The SIP flow will be: C1 -> FS1 -> FS2 -> C2
The RTP flow will be C1 -> FS2 -> C2

It will be up to your logic on FS1 to select the correct FS2 server of course.

-Steve


On 30 December 2010 08:45, Fabio Pietrosanti (naif)
<lists at infosecurity.ch> wrote:
> Hi all,
>
> i have an infrastructure with a single FS server that does RTP
> conditional RTP relay.
>
> My server is in Europe but now i have VoIP clients that need to use the
> server from South America and India.
>
> Obviously two peers coming from "South america" need to bounce all their
> traffic trough Europe increasing the delay.
>
> I would like to install one FS server on each continent and implement a
> dynamic geo-ip based logic to have two clients use the RTP relay near to
> them.
>
> All VoIP clients connect to a single FS server.
>
> How it's possible to install a 2nd FS (or RTPRelay or whatever) so that
> if the two clients are coming from a different geographical area,
> redirect their RTP relay to the FS near to them?
>
> I am missing how to do it with FS, i just need 1 central server for SIP
> signaling and multiple FS acting as RTP relay with a custom server side
> components that looks are the SIP client IP address and address the VoIP
> clients's RTP flows to a 2nd FS box, near to them.
>
> How to address VoIP client's RTP flow to another FS server?
>
> Fabio
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From lists at infosecurity.ch  Thu Dec 30 14:00:16 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Thu, 30 Dec 2010 12:00:16 +0100
Subject: [Freeswitch-users] Moving RTP relay to another FS BOX?
In-Reply-To: <AANLkTime6GQtbQS3x2GC3yQh_=fGnMafYg12FWwUmgOe@mail.gmail.com>
References: <4D1C46AA.3060702@infosecurity.ch>
	<AANLkTime6GQtbQS3x2GC3yQh_=fGnMafYg12FWwUmgOe@mail.gmail.com>
Message-ID: <4D1C6640.3050209@infosecurity.ch>

Hi Steven,

if i understand correctly in your scenario "C2" is SIP registered to FS2.

While i would like to have C1 and C2 both SIP registered to FS1, but if
they match certain parameters (that's application logic), i want their
RTP flow to goes proxed trough FS2.

FS1 is in Europe.
FS2 is in India.
C1 and C1 are in India.

C1 and C1 are connected to FS1 in Europe for SIP.

I would like to have the flow as follow:
SIP Flow: C1 -> FS1 -> C2
RTP flow: C1 -> FS2 -> C2

Obviously FS1 need in some way to be able to "instruct" C1 and C2 to go
trough FS2, and FS2 to handle RTP relay.

>From my basic feeling i would need to move to a Kamailio+RTPProxy
solutions, but if FS could have the flexibility to implement such
solution it would be *much better* as i am already FS based.

Also if some custom development is required, i would be happy to sponsor
some bounty about it.

Fabio

On 30/12/10 11.43, Steven Ayre wrote:
> Try bridging the call from the central SIP FS server to the regional
> FS server, with bypass_media=true on the central server and =false on
> the regional ones.
>
> C1=client 1, C2=client2, FS1=central, FS2=regional
>
> The SIP flow will be: C1 -> FS1 -> FS2 -> C2
> The RTP flow will be C1 -> FS2 -> C2
>
> It will be up to your logic on FS1 to select the correct FS2 server of course.
>
> -Steve
>
>
> On 30 December 2010 08:45, Fabio Pietrosanti (naif)
> <lists at infosecurity.ch> wrote:
>> Hi all,
>>
>> i have an infrastructure with a single FS server that does RTP
>> conditional RTP relay.
>>
>> My server is in Europe but now i have VoIP clients that need to use the
>> server from South America and India.
>>
>> Obviously two peers coming from "South america" need to bounce all their
>> traffic trough Europe increasing the delay.
>>
>> I would like to install one FS server on each continent and implement a
>> dynamic geo-ip based logic to have two clients use the RTP relay near to
>> them.
>>
>> All VoIP clients connect to a single FS server.
>>
>> How it's possible to install a 2nd FS (or RTPRelay or whatever) so that
>> if the two clients are coming from a different geographical area,
>> redirect their RTP relay to the FS near to them?
>>
>> I am missing how to do it with FS, i just need 1 central server for SIP
>> signaling and multiple FS acting as RTP relay with a custom server side
>> components that looks are the SIP client IP address and address the VoIP
>> clients's RTP flows to a 2nd FS box, near to them.
>>
>> How to address VoIP client's RTP flow to another FS server?
>>
>> Fabio
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mustafa.pk at gmail.com  Thu Dec 30 15:14:31 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Thu, 30 Dec 2010 17:14:31 +0500
Subject: [Freeswitch-users] FreeSWITCH SBC Setup (HA mode)
Message-ID: <AANLkTi=w5abhESY0TMugvdqbQ84a5RVNDPfvT4uxhd2+@mail.gmail.com>

Hi everyone,

i am trying to setup FreeSWITCH as a SBC in ha mode; for this purpose i have
setup two FS boxes(identical) in HA mode using heartbeat, mysql as odbc
backend and enabled track-calls in both sip profiles. Everything is working
as expected i.e sip sessions successfully recovered during failover in both
directions.

i am just curious and would like to know should i go for odbc or is it
possible to avoid odbc and use drbd for /opt/switch/db directory to keep .db
files in sync between two fs boxes. is it likely to work?

another question is related to performance when track-calls is enabled, we
are supposed to handle ~300 sip concurrent calls around the clock in
proxy_media mode, coming from our sip provider (on external profile) and
forwarding to our call-center server (configured as internal sip gateway),
we have plenty of good hardware (quad-core Xeon servers with 8Gigs ram), is
it good to go :) or can raise alarms?


Thanks and best regards,

-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com
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From justlikeef at gmail.com  Thu Dec 30 15:42:11 2010
From: justlikeef at gmail.com (Rob Hutton)
Date: Thu, 30 Dec 2010 07:42:11 -0500
Subject: [Freeswitch-users] mod_callcenter features
In-Reply-To: <1293442949323-5869082.post@n2.nabble.com>
References: <1293442949323-5869082.post@n2.nabble.com>
Message-ID: <201012300742.11829.justlikeef@gmail.com>

Some of the API commands are on in the Wiki under the heading API:

http://wiki.freeswitch.org/wiki/Mod_callcenter

Everything starts with callcenter_config, so you can feel your way around the command line and everything is fairly self explanatory.

What do you mean control the call flow?  Can you give an example of what you are trying to accomplish?

-- 
Thanks,
Rob
On Monday 27 December 2010 04:42:29 rajkumar wrote:
> 
> Hi,
> 
>   I am developing an application with mod_callcenter. I need to know the
> following about mod_callcenter.
> 
>   * Is it possible to add/update/delete the queue configurations dynamically
> without using static xml configurations.
>   * How can I control the call flow (for playback message and recording)
> before and after bridging the call.
> 
> Thanks in advance
> 
> regards
> rajkumar k
> 


From ashish at vivaconnect.in  Thu Dec 30 14:42:33 2010
From: ashish at vivaconnect.in (Ashish Chavan // Viva)
Date: Thu, 30 Dec 2010 17:12:33 +0530
Subject: [Freeswitch-users] ERR : NORMAL CIRCUIT CONGESTION
Message-ID: <AANLkTik6Ru8T=0Vtvx9vMogq_b-6ZK8eAXf++Cr4HfOq@mail.gmail.com>

Dear Team,

Here I am having 2 port CTI card on FreeSWITCH server. Here I am having
Asterisk server with pep2t.
whenever I am executing originate command on fs_cli like

                    #> originate openzap/smg_prid/a/1001 at g1 &echo

getting --
2010-12-29 14:28:00.530435 [WARNING] ozmod_sangoma_boost.c:341 TX EVENT:
CALL_START:(80) [w1g1] CSid=28 Seq=54 Cn=[N/A] Cd=[1001] Ci=[N/A] Rdnis=[]
2010-12-29 14:28:00.531591 [WARNING] ozmod_sangoma_boost.c:1473 RX EVENT
(N): CALL_START_NACK:(82) [w256g256] Rc=34 CSid=28 Seq=28
2010-12-29 14:28:00.531591 [WARNING] sangoma_boost_client.c:221 TX EVENT
(N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=28 Seq=55
2010-12-29 14:28:00.532579 *[ERR] switch_ivr_originate.c:2430 Cannot create
outgoing channel of type [OpenZAP] cause: [NORMAL_CIRCUIT_CONGESTION]*
2010-12-29 14:28:00.532579 [DEBUG] switch_ivr_originate.c:3228 Originate
Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]



whereas on /var/log/sangoma_mgd.log
getting----
Dec 29 10:12:05 freeswitchtest2 sangoma_prid: HeartBeat timeout
(current:10:12:05 last:10:11:55 grace:0)
Dec 29 10:12:15 freeswitchtest2 sangoma_prid: HeartBeat timeout
(current:10:12:15 last:10:12:05 grace:0)
Dec 29 10:12:15 freeswitchtest2 sangoma_prid: *Assuming application is dead*
Dec 29 10:12:25 freeswitchtest2 sangoma_prid: HeartBeat timeout
(current:10:12:25 last:10:12:15 grace:0)


Incase of any further queries, Please feel free to mail me or contact me on
the numbers provided below.

Thanks & Regards,
Ashish Chavan
Server Administrator

Viva Infomedia Pvt. Ltd.
242, Oshiwara Industrial Centre,
Opp. Oshiwara Bus Depot, New Link Road,
Goregaon West, Mumbai 400104.

Direct: +91.22.4293 0162
Board: +91.22.4293 0100

Email : ashish at vivaconnect.in
Web  : www.vivaconnect.in

Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India
Awards 2009

Disclaimer:
This e-mail and any files transmitted with it are for the sole use of the
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From freiham at splendor.net  Thu Dec 30 14:50:23 2010
From: freiham at splendor.net (Michel  Freiha)
Date: Thu, 30 Dec 2010 13:50:23 +0200
Subject: [Freeswitch-users] Freeswitch with TLS
Message-ID: <921D5B375C81104CB7EB2C498260FBA1015262B6@mail.splendor.net>

Dear Sir,

 

I'm planning to use Freeswitch server as a proxy server before
OpenSIPS...The Job of freeswitch is only when I need to use SIP over TLS
or SRTP when there is a severe firewall on user side...The Client will
send SIP over TLS to freeswitch and the freeswitch will forward these
packets to openSIPS or kamailio or whatever Registrar server...

 

Is that possible?

How many concurrent TLS connection a freeswitch server can handle?

 

Regards

 

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From steveayre at gmail.com  Thu Dec 30 17:33:02 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 30 Dec 2010 14:33:02 +0000
Subject: [Freeswitch-users] Freeswitch with TLS
In-Reply-To: <921D5B375C81104CB7EB2C498260FBA1015262B6@mail.splendor.net>
References: <921D5B375C81104CB7EB2C498260FBA1015262B6@mail.splendor.net>
Message-ID: <AANLkTin=5+Fv2NqWyb_Yj7=obVkE_74xdMr4Ot1p98UO@mail.gmail.com>

FreeSWITCH is a b2bua not proxy server.

However, yes it would be possible to accept a SIP TLS call and bridge
it to a call going to OpenSIPS. It won't forward the original
messages, but will forward the signalling (i.e. you'll get ringing,
answered etc passed along but it'll be in separate calls not just
adding a Via header when forwarding).

OpenSIPS supports TLS itself though - any particular reason you're not
using its own support?

-Steve


On 30 December 2010 11:50, Michel  Freiha <freiham at splendor.net> wrote:
> Dear Sir,
>
>
>
> I?m planning to use Freeswitch server as a proxy server before OpenSIPS?The
> Job of freeswitch is only when I need to use SIP over TLS or SRTP when there
> is a severe firewall on user side?The Client will send SIP over TLS to
> freeswitch and the freeswitch will forward these packets to openSIPS or
> kamailio or whatever Registrar server?
>
>
>
> Is that possible?
>
> How many concurrent TLS connection a freeswitch server can handle?
>
>
>
> Regards
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Thu Dec 30 18:21:43 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 30 Dec 2010 09:21:43 -0600
Subject: [Freeswitch-users] CPS
In-Reply-To: <AANLkTi=GEYTXe4MREfxeSaZc5y==Mv4e9rQCoJVHE2ES@mail.gmail.com>
References: <AANLkTi=GEYTXe4MREfxeSaZc5y==Mv4e9rQCoJVHE2ES@mail.gmail.com>
Message-ID: <33ACD386-5CD2-49E2-87EA-34BBFD985443@freeswitch.org>

This all depends on what you're doing and how hard you're kicking its ass.  Just don't come crying when you run out of threads!  :P

/b

On Dec 29, 2010, at 11:12 PM, Sam wrote:

> Hello,
> 
> Was testing the environment with media bypass, could see the CPU for FS application shooting to 200% on quadracore 64 bits debian for CPS of 70.
> 
> Is it true for the environment or can be scaled further ?
> 
> Regards
> Sam



From freiham at splendor.net  Thu Dec 30 17:40:00 2010
From: freiham at splendor.net (Michel  Freiha)
Date: Thu, 30 Dec 2010 16:40:00 +0200
Subject: [Freeswitch-users] Freeswitch with TLS
References: <921D5B375C81104CB7EB2C498260FBA1015262B6@mail.splendor.net>
	<AANLkTin=5+Fv2NqWyb_Yj7=obVkE_74xdMr4Ot1p98UO@mail.gmail.com>
Message-ID: <921D5B375C81104CB7EB2C498260FBA1015262B7@mail.splendor.net>

Dear Sir,

I know that openSIPS supports TLS...the major issue that we are facing
performance degradation once the number of users using TLS increase and
reach some hundreds...

I just first need to know how many concurrent registered users using TLS
FS can support?
Second, is there any manual that I can follow in order to accomplish
this scenario?

Regards

Michel Freiha 
Technical Manager
Splendor Telecom (www.splendor.net) 
Beirut, Lebanon 
Phone: +961 1 373725 
Fax: +961 1 375554


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Steven Ayre
Sent: Thursday, December 30, 2010 4:33 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Freeswitch with TLS

FreeSWITCH is a b2bua not proxy server.

However, yes it would be possible to accept a SIP TLS call and bridge it
to a call going to OpenSIPS. It won't forward the original messages, but
will forward the signalling (i.e. you'll get ringing, answered etc
passed along but it'll be in separate calls not just adding a Via header
when forwarding).

OpenSIPS supports TLS itself though - any particular reason you're not
using its own support?

-Steve


On 30 December 2010 11:50, Michel  Freiha <freiham at splendor.net> wrote:
> Dear Sir,
>
>
>
> I'm planning to use Freeswitch server as a proxy server before 
> OpenSIPS...The Job of freeswitch is only when I need to use SIP over
TLS 
> or SRTP when there is a severe firewall on user side...The Client will

> send SIP over TLS to freeswitch and the freeswitch will forward these 
> packets to openSIPS or kamailio or whatever Registrar server...
>
>
>
> Is that possible?
>
> How many concurrent TLS connection a freeswitch server can handle?
>
>
>
> Regards
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>

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


From marcdecorny at gmail.com  Thu Dec 30 18:29:22 2010
From: marcdecorny at gmail.com (Marc de Corny)
Date: Thu, 30 Dec 2010 15:29:22 +0000
Subject: [Freeswitch-users] Send email on successful fax sending.
Message-ID: <AANLkTinJ3N+zwg+ZK2RmdAZECEkLgcvOSKT0hf_mbpix@mail.gmail.com>

Hi all,

I have got all the inbound fax working and can get FS to send outbound fax
from the shell by using the commands :
/opt/freeswitch/bin/fs_cli \
--execute="originate {fax_verbose=true}$DEST &txfax($TMPFAX)" as in the wiki

However I'm looking for a way of notifying the sender on the success or
failure of the fax emission. Is there a way of getting a result back from
that command like fax_success 0/1 that will allow me then to send the
relevant emails out ?

Any help is much appreciated.

thanks
Marc
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From rafonline at hotmail.com  Thu Dec 30 19:22:10 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Thu, 30 Dec 2010 16:22:10 +0000
Subject: [Freeswitch-users] mod lcr
Message-ID: <COL122-W41FF54690AE9D0D554E13EB5030@phx.gbl>



Hi,

I am using mod lcr with mod nibble, my example LCR invocation returns the following in lcr_auto_route: 

[lcr_carrier=carrier2,lcr_rate=0.12000,nibble_account=12345,nibble_rate=0.24000,lowbal_amt=0.1,nobal_amt=0]sofia/external/01 at proxy.carrier2.net:5060|[lcr_carrier=carrier1,lcr_rate=0.15000,nibble_account=12345,nibble_rate=0.30000,lowbal_amt=0.1,nobal_amt=0]sofia/gateway/carrier1/$1

I was wondering if there is a quick method of accessing the nibble_rate field, so I can give callers an estimate of how long they before bridging their call?? I was hoping for some session variable to be populated by lcr to give easy access to this field. 

Any help will be appreciated

Cheers


Raf





 		 	   		  

From freeswitch at peely.com  Thu Dec 30 19:25:06 2010
From: freeswitch at peely.com (peely)
Date: Thu, 30 Dec 2010 08:25:06 -0800 (PST)
Subject: [Freeswitch-users] Video codec negotiation
Message-ID: <1293726306918-5877068.post@n2.nabble.com>


Hi,

I have a profile which supports video codecs H264, H263, H261. With even the
latest GIT version it seems that FS doesn't make any attempt to limit the
codecs offered between endpoints.

For example, if User Agent A has H.264 first priority then H.263 and H.261,
but User Agent B has only H.263 and H.261 it seems that FS 200's with H.264
as the codec for UA A and H.263 for UA B, as a result, no video exchanged
between UAs.

I have tried greedy codec negotiation but still the same. If I remove H.264
from User Agent A then both negotiation H.263.

Given that FS only supports passthrough video, shouldn't it enforce the same
video codec end to end or no video codec at all?



Regards,



Neil.


-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Video-codec-negotiation-tp5877068p5877068.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Avi at aMarcus.com  Thu Dec 30 19:27:37 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Thu, 30 Dec 2010 18:27:37 +0200
Subject: [Freeswitch-users] FreeSWITCH SBC Setup (HA mode)
In-Reply-To: <AANLkTi=w5abhESY0TMugvdqbQ84a5RVNDPfvT4uxhd2+@mail.gmail.com>
References: <AANLkTi=w5abhESY0TMugvdqbQ84a5RVNDPfvT4uxhd2+@mail.gmail.com>
Message-ID: <AANLkTik-fA0yJbK=GQpvrR=P3rGMVSOiGPj8agO7CpJ4@mail.gmail.com>

The other option for failover I was considering is the default sqlite - set
up a job to rsync it to/from the slave constantly.
Perhaps even throw that into a ramdisk, if that makes a difference (the
kernal may be caching it anyway).
-Avi

On Thu, Dec 30, 2010 at 2:14 PM, Ghulam Mustafa <mustafa.pk at gmail.com>wrote:

> Hi everyone,
>
> i am trying to setup FreeSWITCH as a SBC in ha mode; for this purpose i
> have setup two FS boxes(identical) in HA mode using heartbeat, mysql as odbc
> backend and enabled track-calls in both sip profiles. Everything is working
> as expected i.e sip sessions successfully recovered during failover in both
> directions.
>
> i am just curious and would like to know should i go for odbc or is it
> possible to avoid odbc and use drbd for /opt/switch/db directory to keep .db
> files in sync between two fs boxes. is it likely to work?
>
> another question is related to performance when track-calls is enabled, we
> are supposed to handle ~300 sip concurrent calls around the clock in
> proxy_media mode, coming from our sip provider (on external profile) and
> forwarding to our call-center server (configured as internal sip gateway),
> we have plenty of good hardware (quad-core Xeon servers with 8Gigs ram), is
> it good to go :) or can raise alarms?
>
>
> Thanks and best regards,
>
> --
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mustafa.pk at gmail.com  Thu Dec 30 19:39:33 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Thu, 30 Dec 2010 21:39:33 +0500
Subject: [Freeswitch-users] FreeSWITCH SBC Setup (HA mode)
In-Reply-To: <AANLkTik-fA0yJbK=GQpvrR=P3rGMVSOiGPj8agO7CpJ4@mail.gmail.com>
References: <AANLkTi=w5abhESY0TMugvdqbQ84a5RVNDPfvT4uxhd2+@mail.gmail.com>
	<AANLkTik-fA0yJbK=GQpvrR=P3rGMVSOiGPj8agO7CpJ4@mail.gmail.com>
Message-ID: <AANLkTimO7zndUfhApf+8Y2n5ui0UoQaKWCSJoMihO4i0@mail.gmail.com>

i think drbd is more HA aware and works very well in HA setups. rsync is not
recommended  for hot/live mirroring and specially when doing it both
directions.

-m

On Thu, Dec 30, 2010 at 9:27 PM, Avi Marcus <Avi at amarcus.com> wrote:

> The other option for failover I was considering is the default sqlite - set
> up a job to rsync it to/from the slave constantly.
> Perhaps even throw that into a ramdisk, if that makes a difference (the
> kernal may be caching it anyway).
> -Avi
>
> On Thu, Dec 30, 2010 at 2:14 PM, Ghulam Mustafa <mustafa.pk at gmail.com>wrote:
>
>> Hi everyone,
>>
>> i am trying to setup FreeSWITCH as a SBC in ha mode; for this purpose i
>> have setup two FS boxes(identical) in HA mode using heartbeat, mysql as odbc
>> backend and enabled track-calls in both sip profiles. Everything is working
>> as expected i.e sip sessions successfully recovered during failover in both
>> directions.
>>
>> i am just curious and would like to know should i go for odbc or is it
>> possible to avoid odbc and use drbd for /opt/switch/db directory to keep .db
>> files in sync between two fs boxes. is it likely to work?
>>
>> another question is related to performance when track-calls is enabled, we
>> are supposed to handle ~300 sip concurrent calls around the clock in
>> proxy_media mode, coming from our sip provider (on external profile) and
>> forwarding to our call-center server (configured as internal sip gateway),
>> we have plenty of good hardware (quad-core Xeon servers with 8Gigs ram), is
>> it good to go :) or can raise alarms?
>>
>>
>> Thanks and best regards,
>>
>> --
>> Ghulam Mustafa
>> cell: +92 333.611.7681
>> sip: cyrenity at ekiga.net
>> mail: mustafa.pk at gmail.com
>> web: cyrenity.wordpress.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com
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From rupa at rupa.com  Thu Dec 30 19:51:39 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 30 Dec 2010 10:51:39 -0600
Subject: [Freeswitch-users] mod lcr
In-Reply-To: <COL122-W41FF54690AE9D0D554E13EB5030@phx.gbl>
References: <COL122-W41FF54690AE9D0D554E13EB5030@phx.gbl>
Message-ID: <AANLkTinu1DmGDZ3mqB-tbQp2Q3CSWan3gNgSX14P=q3y@mail.gmail.com>

I guess my first question would be *which* rate field?  Notice they are all
different.  If you mean after the bridge completes, then the var is set on
the b-leg which you can use to announce the rate or whatever.

On Thu, Dec 30, 2010 at 10:22 AM, Rafqat . <rafonline at hotmail.com> wrote:

>
>
> Hi,
>
> I am using mod lcr with mod nibble, my example LCR invocation returns the
> following in lcr_auto_route:
>
>
> [lcr_carrier=carrier2,lcr_rate=0.12000,nibble_account=12345,nibble_rate=0.24000,lowbal_amt=0.1,nobal_amt=0]sofia/external/01 at proxy.carrier2.net:5060
> |[lcr_carrier=carrier1,lcr_rate=0.15000,nibble_account=12345,nibble_rate=0.30000,lowbal_amt=0.1,nobal_amt=0]sofia/gateway/carrier1/$1
>
> I was wondering if there is a quick method of accessing the nibble_rate
> field, so I can give callers an estimate of how long they before bridging
> their call?  I was hoping for some session variable to be populated by lcr
> to give easy access to this field.
>
> Any help will be appreciated
>
> Cheers
>
>
> Raf
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From rupa at rupa.com  Thu Dec 30 19:52:35 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 30 Dec 2010 10:52:35 -0600
Subject: [Freeswitch-users] FreeSWITCH SBC Setup (HA mode)
In-Reply-To: <AANLkTik-fA0yJbK=GQpvrR=P3rGMVSOiGPj8agO7CpJ4@mail.gmail.com>
References: <AANLkTi=w5abhESY0TMugvdqbQ84a5RVNDPfvT4uxhd2+@mail.gmail.com>
	<AANLkTik-fA0yJbK=GQpvrR=P3rGMVSOiGPj8agO7CpJ4@mail.gmail.com>
Message-ID: <AANLkTikq-+hJDTuo540ktQrsph+y-0n349fdeTeKPG6N@mail.gmail.com>

I'm not sure I'd trust a rsync of a live database.

On Thu, Dec 30, 2010 at 10:27 AM, Avi Marcus <Avi at amarcus.com> wrote:

> The other option for failover I was considering is the default sqlite - set
> up a job to rsync it to/from the slave constantly.
> Perhaps even throw that into a ramdisk, if that makes a difference (the
> kernal may be caching it anyway).
> -Avi
>
> On Thu, Dec 30, 2010 at 2:14 PM, Ghulam Mustafa <mustafa.pk at gmail.com>wrote:
>
>> Hi everyone,
>>
>> i am trying to setup FreeSWITCH as a SBC in ha mode; for this purpose i
>> have setup two FS boxes(identical) in HA mode using heartbeat, mysql as odbc
>> backend and enabled track-calls in both sip profiles. Everything is working
>> as expected i.e sip sessions successfully recovered during failover in both
>> directions.
>>
>> i am just curious and would like to know should i go for odbc or is it
>> possible to avoid odbc and use drbd for /opt/switch/db directory to keep .db
>> files in sync between two fs boxes. is it likely to work?
>>
>> another question is related to performance when track-calls is enabled, we
>> are supposed to handle ~300 sip concurrent calls around the clock in
>> proxy_media mode, coming from our sip provider (on external profile) and
>> forwarding to our call-center server (configured as internal sip gateway),
>> we have plenty of good hardware (quad-core Xeon servers with 8Gigs ram), is
>> it good to go :) or can raise alarms?
>>
>>
>> Thanks and best regards,
>>
>> --
>> Ghulam Mustafa
>> cell: +92 333.611.7681
>> sip: cyrenity at ekiga.net
>> mail: mustafa.pk at gmail.com
>> web: cyrenity.wordpress.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From rafonline at hotmail.com  Thu Dec 30 20:05:45 2010
From: rafonline at hotmail.com (Rafqat .)
Date: Thu, 30 Dec 2010 17:05:45 +0000
Subject: [Freeswitch-users] mod lcr
In-Reply-To: <AANLkTinu1DmGDZ3mqB-tbQp2Q3CSWan3gNgSX14P=q3y@mail.gmail.com>
References: <COL122-W41FF54690AE9D0D554E13EB5030@phx.gbl>,
	<AANLkTinu1DmGDZ3mqB-tbQp2Q3CSWan3gNgSX14P=q3y@mail.gmail.com>
Message-ID: <COL122-W25871C908DFBFB412EE3C9B5030@phx.gbl>



Hi Rupa

I am after the 'nibble_rate' field which is returned from my custom sql:

[lcr_carrier=carrier2,lcr_rate=0.12000,nibble_account=12345,nibble_rate=0.24000,lowbal_amt=0.1,nobal_amt=0]sofia/external/01 at proxy.carrier2.net:5060|[lcr_carrier=carrier1,lcr_rate=0.15000,nibble_account=12345,nibble_rate=0.30000,lowbal_amt=0.1,nobal_amt=0]sofia/gateway/carrier1/$1

I would like to calculate the time the user has for the call before doing any bridging as they might have positive funds, but not enough for a minimum call duration of 5 minutes.

Is there away of getting this field into session scope? 

Cheers

Raf


________________________________
> Date: Thu, 30 Dec 2010 10:51:39 -0600
> From: rupa at rupa.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod lcr
>
> I guess my first question would be *which* rate field? Notice they are
> all different. If you mean after the bridge completes, then the var is
> set on the b-leg which you can use to announce the rate or whatever.
>
> On Thu, Dec 30, 2010 at 10:22 AM, Rafqat .
> > wrote:
>
>
> Hi,
>
> I am using mod lcr with mod nibble, my example LCR invocation returns
> the following in lcr_auto_route:
>
> [lcr_carrier=carrier2,lcr_rate=0.12000,nibble_account=12345,nibble_rate=0.24000,lowbal_amt=0.1,nobal_amt=0]sofia/external/01 at proxy.carrier2.net:5060|[lcr_carrier=carrier1,lcr_rate=0.15000,nibble_account=12345,nibble_rate=0.30000,lowbal_amt=0.1,nobal_amt=0]sofia/gateway/carrier1/$1
>
> I was wondering if there is a quick method of accessing the nibble_rate
> field, so I can give callers an estimate of how long they before
> bridging their call? I was hoping for some session variable to be
> populated by lcr to give easy access to this field.
>
> Any help will be appreciated
>
> Cheers
>
>
> Raf
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> --
> -Rupa
>
> _______________________________________________ FreeSWITCH-users
> mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  

From riedinger at sns.eu  Thu Dec 30 20:06:24 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Thu, 30 Dec 2010 18:06:24 +0100
Subject: [Freeswitch-users] Asynchronous PTIME
In-Reply-To: <AANLkTimzmg1kfM4+75e-Y9vG6wErhXjX6f83+4-voNyu@mail.gmail.com>
References: <4D127733.8010409@nevian.org> <4D15266B.3080903@sns.eu>
	<AANLkTimzmg1kfM4+75e-Y9vG6wErhXjX6f83+4-voNyu@mail.gmail.com>
Message-ID: <4D1CBC10.8080004@sns.eu>

You have to set it the paramater in the profile, which is used for the 
call, e. g. set in external.xml:
<param name="rtp-autofix-timing" value="false"/>

BR
    Jan



Am 29.12.2010 20:04, schrieb Saeed Ahmed:
> on which profile did you put that param?
>
> I've allnet phone if i use g729 with ptime:40 then i can't hear called 
> party properly, its choppy voice, i put autofix param in both internal 
> and external profiles, but it didn't help.
>
> if i use ptime: 20 then it works fine.
>
> On Sat, Dec 25, 2010 at 12:02 AM, Jan Riedinger <riedinger at sns.eu 
> <mailto:riedinger at sns.eu>> wrote:
>
>     Hi Serge,
>
>     I had similar problems, look for my thread "Problematic Behaviour
>     of FS
>     regarding ptime negotiation" in October. I could solve the problem by
>     setting "rtp-autofix-timing=false", which disabled the (too) smart
>     behaviour of FreeSwitch.
>
>     BR
>         Jan
>
>
>     Am 22.12.2010 23:09, schrieb Serge S. Yuriev:
>     > Hello,
>     >
>     > 2010-12-20 16:51:13.960502 [WARNING] mod_sofia.c:1036
>     Asynchronous PTIME
>     > not supported, changing our end from 20 to 60
>     >
>     > I'm getting this warning and client hears chopped sound :(
>     > That is "Our end"?
>     >
>     > Call flow is Cisco (sip) FS (sip) YATE (h323) MeraTransitSoftSwitch
>     > All but MVTS under my control.
>     >
>     > I doesn't see any clue in logs and can't reproduce this with my
>     testing
>     > via Blink/PortSIP/PhonerLite/Twinkle or Grandstream devices
>     >
>     > Which debug/logs I should take? Any ideas?
>     >
>     > Thanks a lot.
>     >
>     > btw how I can save debug into log not only console?
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From u2nsam at gmail.com  Thu Dec 30 20:07:04 2010
From: u2nsam at gmail.com (Sam)
Date: Thu, 30 Dec 2010 22:37:04 +0530
Subject: [Freeswitch-users] FreeSWITCH SBC Setup (HA mode)
In-Reply-To: <AANLkTik-fA0yJbK=GQpvrR=P3rGMVSOiGPj8agO7CpJ4@mail.gmail.com>
References: <AANLkTi=w5abhESY0TMugvdqbQ84a5RVNDPfvT4uxhd2+@mail.gmail.com>
	<AANLkTik-fA0yJbK=GQpvrR=P3rGMVSOiGPj8agO7CpJ4@mail.gmail.com>
Message-ID: <AANLkTimR6OHNT=7DU3krKf_puLJC2AMpxk+2NGbwn5Oa@mail.gmail.com>

Can use Sqlite replication in between 2 instances by sqlite3_update_hook.

Regds
Sam

On Thu, Dec 30, 2010 at 9:57 PM, Avi Marcus <Avi at amarcus.com> wrote:

> The other option for failover I was considering is the default sqlite - set
> up a job to rsync it to/from the slave constantly.
> Perhaps even throw that into a ramdisk, if that makes a difference (the
> kernal may be caching it anyway).
> -Avi
>
> On Thu, Dec 30, 2010 at 2:14 PM, Ghulam Mustafa <mustafa.pk at gmail.com>wrote:
>
>> Hi everyone,
>>
>> i am trying to setup FreeSWITCH as a SBC in ha mode; for this purpose i
>> have setup two FS boxes(identical) in HA mode using heartbeat, mysql as odbc
>> backend and enabled track-calls in both sip profiles. Everything is working
>> as expected i.e sip sessions successfully recovered during failover in both
>> directions.
>>
>> i am just curious and would like to know should i go for odbc or is it
>> possible to avoid odbc and use drbd for /opt/switch/db directory to keep .db
>> files in sync between two fs boxes. is it likely to work?
>>
>> another question is related to performance when track-calls is enabled, we
>> are supposed to handle ~300 sip concurrent calls around the clock in
>> proxy_media mode, coming from our sip provider (on external profile) and
>> forwarding to our call-center server (configured as internal sip gateway),
>> we have plenty of good hardware (quad-core Xeon servers with 8Gigs ram), is
>> it good to go :) or can raise alarms?
>>
>>
>> Thanks and best regards,
>>
>> --
>> Ghulam Mustafa
>> cell: +92 333.611.7681
>> sip: cyrenity at ekiga.net
>> mail: mustafa.pk at gmail.com
>> web: cyrenity.wordpress.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Thu Dec 30 20:20:32 2010
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 31 Dec 2010 01:20:32 +0800
Subject: [Freeswitch-users] skypopen load problem
In-Reply-To: <AANLkTi=AXuqR0VvBCoPqAx3mLPrZ=GASF0=2fvEX-8G6@mail.gmail.com>
References: <AANLkTimY-L=gA65vsRdfLTM1BEaza8PjA98DoiWZpSiH@mail.gmail.com>
	<AANLkTi=UEWRSjhf=8+jEAbyptaPP4Sp1NBk5yzP8Um4k@mail.gmail.com>
	<AANLkTi=AXuqR0VvBCoPqAx3mLPrZ=GASF0=2fvEX-8G6@mail.gmail.com>
Message-ID: <AANLkTim42C-B-cbmqP7Wb8Oev=Bycp1TiQzOy0nFCy9Z@mail.gmail.com>

Problem solved.

It's because I move from skypiax to skypopen and still using the old config.

Thanks Giovanni helped me debug on our server.

On Tue, Dec 28, 2010 at 10:00 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> Ciao Seven,
>
> is a pleasure to hear from you.
>
> The "error 68" is that the skypeclient does not recognize-authorize
> you. This is because (with all probability) it has not yet connected
> to the skype p2p network. Give the client some time before loading
> mod_skypopen. Skype network had "some problems" last days (hehehe).
>
> If not because of network problems, can be because the files in the
> skypeclient directory (where it keeps config data) has been trashed,
> but if client.c goes through is not that, is the network problem.
>
> Anyway, CentOS 64 bit is supersupported, actually is the main
> reference platform (with Ubuntu LTS the second platform).
>
> Also, as soon as you overcome this problem (and please write again on
> this if it persists) I would like you (if you got time) to test the
> new OSS driver (to be used with skypeclient static for oss 2.0.72, a
> little bit easier to find around the net than the skypeclient static
> for alsa 2.0.72). I believe it can give you more performances (but
> mileage can vary). More on this will follow in the next days.
>
> Ciao for now,
>
> -giovanni
>
> On Tue, Dec 28, 2010 at 2:12 PM, Seven Du <dujinfang at gmail.com> wrote:
>> see also http://pastebin.freeswitch.org/14889
>>
>> sorry flooding.
>>
>> On Tue, Dec 28, 2010 at 9:08 PM, Seven Du <dujinfang at gmail.com> wrote:
>>> Hi,
>>>
>>> So finally I get a chance to move to CentOS 64bit.
>>>
>>> git HEAD on CentOS 5.5 64bit. skype 2.0.0.72 static.
>>>
>>> Load error:
>>>
>>> http://pastebin.freeswitch.org/14888
>>>
>>> Seems sk_2 read extra AUTOAWAY OFF and ERROR 68, however, client.c
>>> never shows those extra lines on my test against all my 20 instances.
>>> I use multi-instances-same-username btw.
>>>
>>> ./client :302
>>> Initialized XInitThreads!
>>> PROTOCOL 7
>>> #ciapalino PONG
>>> CONNSTATUS ONLINE
>>> CURRENTUSERHANDLE idapted_voip_10
>>> USERSTATUS ONLINE
>>>
>>>
>>> Sometimes I can load till sk_8, sometimes sk_1 doesn't load. Any clue?
>>>
>>> Thanks.
>>>
>>> --
>>> About: http://about.me/dujinfang
>>> Blog: http://www.dujinfang.com
>>> Proj:? http://www.freeswitch.org.cn
>>>
>>
>>
>>
>> --
>> About: http://about.me/dujinfang
>> Blog: http://www.dujinfang.com
>> Proj:? http://www.freeswitch.org.cn
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From jaugenstine at gmail.com  Thu Dec 30 20:27:52 2010
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Thu, 30 Dec 2010 09:27:52 -0800
Subject: [Freeswitch-users] CPS
In-Reply-To: <33ACD386-5CD2-49E2-87EA-34BBFD985443@freeswitch.org>
References: <AANLkTi=GEYTXe4MREfxeSaZc5y==Mv4e9rQCoJVHE2ES@mail.gmail.com>
	<33ACD386-5CD2-49E2-87EA-34BBFD985443@freeswitch.org>
Message-ID: <AANLkTi=w7b1UnbjYFOQA-rWLPVEbQGOGE_L12otMcXdz@mail.gmail.com>

Sam,

I have been doing some work on a server with 16 cores and I have see
Freeswitch go to 900% and it is working great.

Jonathan

On Thu, Dec 30, 2010 at 7:21 AM, Brian West <brian at freeswitch.org> wrote:

> This all depends on what you're doing and how hard you're kicking its ass.
>  Just don't come crying when you run out of threads!  :P
>
> /b
>
> On Dec 29, 2010, at 11:12 PM, Sam wrote:
>
> > Hello,
> >
> > Was testing the environment with media bypass, could see the CPU for FS
> application shooting to 200% on quadracore 64 bits debian for CPS of 70.
> >
> > Is it true for the environment or can be scaled further ?
> >
> > Regards
> > Sam
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Thu Dec 30 20:44:59 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 30 Dec 2010 17:44:59 +0000
Subject: [Freeswitch-users] FreeSWITCH SBC Setup (HA mode)
In-Reply-To: <AANLkTik-fA0yJbK=GQpvrR=P3rGMVSOiGPj8agO7CpJ4@mail.gmail.com>
References: <AANLkTi=w5abhESY0TMugvdqbQ84a5RVNDPfvT4uxhd2+@mail.gmail.com>
	<AANLkTik-fA0yJbK=GQpvrR=P3rGMVSOiGPj8agO7CpJ4@mail.gmail.com>
Message-ID: <5A3B6C9C-637B-427B-95FF-AA50DF64A1CE@gmail.com>

That'll leave you wide open to having a corrupted db, drbd too.

Personally i would use mysql cluster via odbc - all data on 2 nodes, which may or may not be on separate hardware to fs.

Steve on iPhone

On 30 Dec 2010, at 16:27, Avi Marcus <Avi at aMarcus.com> wrote:

> The other option for failover I was considering is the default sqlite - set up a job to rsync it to/from the slave constantly.
> Perhaps even throw that into a ramdisk, if that makes a difference (the kernal may be caching it anyway).
> -Avi
> 
> On Thu, Dec 30, 2010 at 2:14 PM, Ghulam Mustafa <mustafa.pk at gmail.com> wrote:
> Hi everyone,
> 
> i am trying to setup FreeSWITCH as a SBC in ha mode; for this purpose i have setup two FS boxes(identical) in HA mode using heartbeat, mysql as odbc backend and enabled track-calls in both sip profiles. Everything is working as expected i.e sip sessions successfully recovered during failover in both directions.
> 
> i am just curious and would like to know should i go for odbc or is it possible to avoid odbc and use drbd for /opt/switch/db directory to keep .db files in sync between two fs boxes. is it likely to work?
> 
> another question is related to performance when track-calls is enabled, we are supposed to handle ~300 sip concurrent calls around the clock in proxy_media mode, coming from our sip provider (on external profile) and forwarding to our call-center server (configured as internal sip gateway), we have plenty of good hardware (quad-core Xeon servers with 8Gigs ram), is it good to go :) or can raise alarms?
> 
> 
> Thanks and best regards,
> 
> -- 
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Thu Dec 30 20:53:18 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 30 Dec 2010 11:53:18 -0600
Subject: [Freeswitch-users] CPS
In-Reply-To: <AANLkTi=w7b1UnbjYFOQA-rWLPVEbQGOGE_L12otMcXdz@mail.gmail.com>
References: <AANLkTi=GEYTXe4MREfxeSaZc5y==Mv4e9rQCoJVHE2ES@mail.gmail.com>
	<33ACD386-5CD2-49E2-87EA-34BBFD985443@freeswitch.org>
	<AANLkTi=w7b1UnbjYFOQA-rWLPVEbQGOGE_L12otMcXdz@mail.gmail.com>
Message-ID: <AANLkTikgQ2kXjmQMPN5aWZPE7RvfV88XOb8_qT2GHRbw@mail.gmail.com>

yah cos remember its the the percent times the number of cores for the
max so 4 core box has 400% of cpu.
press '1' on top to see each cpu and how much is being used.


On Thu, Dec 30, 2010 at 11:27 AM, jonathan augenstine
<jaugenstine at gmail.com> wrote:
> Sam,
> I have been doing some work on a server with 16 cores and I have see
> Freeswitch go to 900% and it is working great.
> Jonathan
>
> On Thu, Dec 30, 2010 at 7:21 AM, Brian West <brian at freeswitch.org> wrote:
>>
>> This all depends on what you're doing and how hard you're kicking its ass.
>> ?Just don't come crying when you run out of threads! ?:P
>>
>> /b
>>
>> On Dec 29, 2010, at 11:12 PM, Sam wrote:
>>
>> > Hello,
>> >
>> > Was testing the environment with media bypass, could see the CPU for FS
>> > application shooting to 200% on quadracore 64 bits debian for CPS of 70.
>> >
>> > Is it true for the environment or can be scaled further ?
>> >
>> > Regards
>> > Sam
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

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

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


From lloyd.aloysius at gmail.com  Thu Dec 30 20:57:58 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 30 Dec 2010 12:57:58 -0500
Subject: [Freeswitch-users] CentOS 5.5 - FreeSWITCH - Audio Quality
	issues
In-Reply-To: <AANLkTinLXcLEbw7qcyFwgyaX5zoa0uENpy9YK3UraND2@mail.gmail.com>
References: <AANLkTik3y8EfTi5s_a_1BURnUJe8Nv0p+5y8Ph6Jkcg3@mail.gmail.com>
	<AANLkTi=6shTJg0C2eFsar+mdjstT330e2kaUtFxsPQ1F@mail.gmail.com>
	<AANLkTi=3Yi_Phq6Zv2Ep-YEgf=3orNmOEpLVyww=7x=L@mail.gmail.com>
	<316084A7-BCA6-4490-8990-9894331F999F@freeswitch.org>
	<AANLkTimREKiD=3t12WT7fMWWR7JPyOT+3S4jm7SPY6UT@mail.gmail.com>
	<AANLkTintspeXDQiyHXsu2GmuyiK0yqVKrBKyBmwmY-fB@mail.gmail.com>
	<AE8F4FDB-95EF-49E6-80A1-A91315DCF2E4@freeswitch.org>
	<AANLkTinLXcLEbw7qcyFwgyaX5zoa0uENpy9YK3UraND2@mail.gmail.com>
Message-ID: <AANLkTimrj4xO3n5sMYnsb80w5PE6gb4nJLv7NVkCqw81@mail.gmail.com>

Finally solved the problem. There is a hardware (Motherboard) issue on the
server. Replace the server with the new one and use 64bit ... voice quality
problems goes away.

Thank you for the help.

Thanks and regards,
Lloyd




On Wed, Dec 29, 2010 at 12:47 PM, Aloysius Lloyd
<lloyd.aloysius at gmail.com>wrote:

> Anthony & Brian - Thanks.
>
> 1. Client Make the calls ..
>
> Regular Bell Canada Land line or Cell Phone ---> SIP Provider DID --->
> FreeSWITCH --> IVR ->Voicemail
>
> or
>
> Regular Bell Canada Land line or Cell Phone ---> SIP Provider DID --->
> FreeSWITCH --> IVR
>
>
> 2. I will capture the trace when there is no calls. This is a production
> box . Is there any way to capture a trace for a call when there is other
> calls in the box?
>
> 3. I decided to switch to 64bit today night let see that will help me.
>
> Thanks and regards,
> Lloyd
>
>
>
> On Wed, Dec 29, 2010 at 11:55 AM, Brian West <brian at freeswitch.org> wrote:
>
>> YES.  You should always run a 64bit OS on a 64bit CPU... unless you like
>> wasting have your cpu registers... but I bet you the problem will go away if
>> you switch.
>>
>> /b
>>
>> On Dec 29, 2010, at 9:46 AM, Aloysius Lloyd wrote:
>>
>>  CentOS 64 bit ?
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From gmaruzz at gmail.com  Thu Dec 30 21:00:06 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 30 Dec 2010 19:00:06 +0100
Subject: [Freeswitch-users] FreeSWITCH SBC Setup (HA mode)
In-Reply-To: <AANLkTimR6OHNT=7DU3krKf_puLJC2AMpxk+2NGbwn5Oa@mail.gmail.com>
References: <AANLkTi=w5abhESY0TMugvdqbQ84a5RVNDPfvT4uxhd2+@mail.gmail.com>
	<AANLkTik-fA0yJbK=GQpvrR=P3rGMVSOiGPj8agO7CpJ4@mail.gmail.com>
	<AANLkTimR6OHNT=7DU3krKf_puLJC2AMpxk+2NGbwn5Oa@mail.gmail.com>
Message-ID: <AANLkTi==YRnMpfj1eAZw81A+s4ZfSAEddEAHXX7PyV3c@mail.gmail.com>

DRBD is a very interesting and very performing way to replicate a
block device between two or more machine, usally in an active-passive
way. And is very HA aware.

In classic configuration, you have two machines that control each
other state via heartbeat, and a floating IP address that is assigned
to the "active" machine (in addition to the "real" IP address).

The "active" machine is accessed from the HA application through the
floating IP.

The "passive" machine is there idle, but ready to take over in case
the "active" goes down (for failure or maintenance). In that case the
"passive" machine automatically becomes the "active", eg: gets the
floating IP.

The only drawback I can see in this setup (DRBD+nativesqlite instead
of odbc) is that if the FS instance or the machine itself crashes very
badly, maybe will trash the db, and because the sqlite db is
replicated at filesystem level via DRDB to the other machine, you'll
end up with a trashed sqlite db on the "new active" machine too.

Having instead a logic SQL replica (mysql, postgres, whatever) between
the two machines, spares you this possible point of failure.

Also, if you use another heartbeat and another floating IP address
dedicated to the odbc database access, you get HA database too
(meaning: you can do an upgrade of the database engine on the "active"
FS machine, because you moved only the floating IP of the dabase
access to the other machine).

-giovanni

On Thu, Dec 30, 2010 at 6:07 PM, Sam <u2nsam at gmail.com> wrote:
> Can use Sqlite replication in between 2 instances by sqlite3_update_hook.
>
> Regds
> Sam
>
> On Thu, Dec 30, 2010 at 9:57 PM, Avi Marcus <Avi at amarcus.com> wrote:
>>
>> The other option for failover I was considering is the default sqlite -
>> set up a job to rsync it to/from the slave constantly.
>> Perhaps even throw that into a ramdisk, if that makes a difference (the
>> kernal may be caching it anyway).
>> -Avi
>> On Thu, Dec 30, 2010 at 2:14 PM, Ghulam Mustafa <mustafa.pk at gmail.com>
>> wrote:
>>>
>>> Hi everyone,
>>> i am trying to setup FreeSWITCH as a SBC in ha mode; for this purpose?i
>>> have setup two FS boxes(identical) in HA mode using heartbeat, mysql as odbc
>>> backend and enabled track-calls in both sip profiles. Everything is working
>>> as expected i.e sip sessions successfully recovered during failover in both
>>> directions.
>>> i am just curious and would like to know should i go for odbc or is it
>>> possible to avoid odbc and use drbd for /opt/switch/db directory to keep .db
>>> files in sync between two fs boxes. is it likely to work?
>>> another question is related to performance when track-calls is enabled,
>>> we are supposed to handle ~300 sip concurrent calls around the clock in
>>> proxy_media mode, coming from our sip provider (on external profile) and
>>> forwarding to our call-center server (configured as internal sip gateway),
>>> we have plenty of good hardware (quad-core Xeon servers with 8Gigs ram), is
>>> it good to go :) or can raise alarms?
>>>
>>> Thanks and best regards,
>>> --
>>> Ghulam Mustafa
>>> cell: +92 333.611.7681
>>> sip: cyrenity at ekiga.net
>>> mail: mustafa.pk at gmail.com
>>> web: cyrenity.wordpress.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From lloyd.aloysius at gmail.com  Thu Dec 30 21:04:39 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 30 Dec 2010 13:04:39 -0500
Subject: [Freeswitch-users] group_call - ignore the sip_from_user
Message-ID: <AANLkTinf1X-RRNz9W6vsfaifbQzHtLwpKeCB2WK=uhHa@mail.gmail.com>

Hi All,

If a member of a group calling to the group , I want to ignore the member
from the group. Is there any way to do this ?

Thanks
Lloyd
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From norm at voicenetwork.ca  Thu Dec 30 21:09:49 2010
From: norm at voicenetwork.ca (Norman Tomlins)
Date: Thu, 30 Dec 2010 13:09:49 -0500
Subject: [Freeswitch-users] CentOS 5.5 - FreeSWITCH - Audio Quality
	issues
In-Reply-To: <AANLkTimrj4xO3n5sMYnsb80w5PE6gb4nJLv7NVkCqw81@mail.gmail.com>
References: <AANLkTik3y8EfTi5s_a_1BURnUJe8Nv0p+5y8Ph6Jkcg3@mail.gmail.com>
	<AANLkTi=6shTJg0C2eFsar+mdjstT330e2kaUtFxsPQ1F@mail.gmail.com>
	<AANLkTi=3Yi_Phq6Zv2Ep-YEgf=3orNmOEpLVyww=7x=L@mail.gmail.com>
	<316084A7-BCA6-4490-8990-9894331F999F@freeswitch.org>
	<AANLkTimREKiD=3t12WT7fMWWR7JPyOT+3S4jm7SPY6UT@mail.gmail.com>
	<AANLkTintspeXDQiyHXsu2GmuyiK0yqVKrBKyBmwmY-fB@mail.gmail.com>
	<AE8F4FDB-95EF-49E6-80A1-A91315DCF2E4@freeswitch.org>
	<AANLkTinLXcLEbw7qcyFwgyaX5zoa0uENpy9YK3UraND2@mail.gmail.com>
	<AANLkTimrj4xO3n5sMYnsb80w5PE6gb4nJLv7NVkCqw81@mail.gmail.com>
Message-ID: <AANLkTimc6M2sN3-2cp5as=2b=sREeuWhByjDwvL4L-gJ@mail.gmail.com>

Lloyd,

What motherboard was it that gave you the issues?

Norman Tomlins
Voice Network Inc

On Thu, Dec 30, 2010 at 12:57 PM, Aloysius Lloyd
<lloyd.aloysius at gmail.com>wrote:

> Finally solved the problem. There is a hardware (Motherboard) issue on the
> server. Replace the server with the new one and use 64bit ... voice quality
> problems goes away.
>
> Thank you for the help.
>
> Thanks and regards,
> Lloyd
>
>
>
>
> On Wed, Dec 29, 2010 at 12:47 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com
> > wrote:
>
>> Anthony & Brian - Thanks.
>>
>> 1. Client Make the calls ..
>>
>> Regular Bell Canada Land line or Cell Phone ---> SIP Provider DID --->
>> FreeSWITCH --> IVR ->Voicemail
>>
>> or
>>
>> Regular Bell Canada Land line or Cell Phone ---> SIP Provider DID --->
>> FreeSWITCH --> IVR
>>
>>
>> 2. I will capture the trace when there is no calls. This is a production
>> box . Is there any way to capture a trace for a call when there is other
>> calls in the box?
>>
>> 3. I decided to switch to 64bit today night let see that will help me.
>>
>> Thanks and regards,
>> Lloyd
>>
>>
>>
>> On Wed, Dec 29, 2010 at 11:55 AM, Brian West <brian at freeswitch.org>wrote:
>>
>>> YES.  You should always run a 64bit OS on a 64bit CPU... unless you like
>>> wasting have your cpu registers... but I bet you the problem will go away if
>>> you switch.
>>>
>>> /b
>>>
>>> On Dec 29, 2010, at 9:46 AM, Aloysius Lloyd wrote:
>>>
>>>  CentOS 64 bit ?
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Thu Dec 30 21:14:49 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 30 Dec 2010 13:14:49 -0500
Subject: [Freeswitch-users] CentOS 5.5 - FreeSWITCH - Audio Quality
	issues
In-Reply-To: <AANLkTimc6M2sN3-2cp5as=2b=sREeuWhByjDwvL4L-gJ@mail.gmail.com>
References: <AANLkTik3y8EfTi5s_a_1BURnUJe8Nv0p+5y8Ph6Jkcg3@mail.gmail.com>
	<AANLkTi=6shTJg0C2eFsar+mdjstT330e2kaUtFxsPQ1F@mail.gmail.com>
	<AANLkTi=3Yi_Phq6Zv2Ep-YEgf=3orNmOEpLVyww=7x=L@mail.gmail.com>
	<316084A7-BCA6-4490-8990-9894331F999F@freeswitch.org>
	<AANLkTimREKiD=3t12WT7fMWWR7JPyOT+3S4jm7SPY6UT@mail.gmail.com>
	<AANLkTintspeXDQiyHXsu2GmuyiK0yqVKrBKyBmwmY-fB@mail.gmail.com>
	<AE8F4FDB-95EF-49E6-80A1-A91315DCF2E4@freeswitch.org>
	<AANLkTinLXcLEbw7qcyFwgyaX5zoa0uENpy9YK3UraND2@mail.gmail.com>
	<AANLkTimrj4xO3n5sMYnsb80w5PE6gb4nJLv7NVkCqw81@mail.gmail.com>
	<AANLkTimc6M2sN3-2cp5as=2b=sREeuWhByjDwvL4L-gJ@mail.gmail.com>
Message-ID: <AANLkTimRe-pyehB7zcOsEiUTwd-t0g7eCj-do790DVXJ@mail.gmail.com>

Norman,

I use iWeb ( www.iweb.ca )  dedicated server . Motherboard - Supermicro .

Thanks
Lloyd


On Thu, Dec 30, 2010 at 1:09 PM, Norman Tomlins <norm at voicenetwork.ca>wrote:

> Lloyd,
>
> What motherboard was it that gave you the issues?
>
> Norman Tomlins
> Voice Network Inc
>
>
> On Thu, Dec 30, 2010 at 12:57 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com
> > wrote:
>
>> Finally solved the problem. There is a hardware (Motherboard) issue on the
>> server. Replace the server with the new one and use 64bit ... voice quality
>> problems goes away.
>>
>> Thank you for the help.
>>
>> Thanks and regards,
>> Lloyd
>>
>>
>>
>>
>> On Wed, Dec 29, 2010 at 12:47 PM, Aloysius Lloyd <
>> lloyd.aloysius at gmail.com> wrote:
>>
>>> Anthony & Brian - Thanks.
>>>
>>> 1. Client Make the calls ..
>>>
>>> Regular Bell Canada Land line or Cell Phone ---> SIP Provider DID --->
>>> FreeSWITCH --> IVR ->Voicemail
>>>
>>> or
>>>
>>> Regular Bell Canada Land line or Cell Phone ---> SIP Provider DID --->
>>> FreeSWITCH --> IVR
>>>
>>>
>>> 2. I will capture the trace when there is no calls. This is a production
>>> box . Is there any way to capture a trace for a call when there is other
>>> calls in the box?
>>>
>>> 3. I decided to switch to 64bit today night let see that will help me.
>>>
>>> Thanks and regards,
>>> Lloyd
>>>
>>>
>>>
>>> On Wed, Dec 29, 2010 at 11:55 AM, Brian West <brian at freeswitch.org>wrote:
>>>
>>>> YES.  You should always run a 64bit OS on a 64bit CPU... unless you like
>>>> wasting have your cpu registers... but I bet you the problem will go away if
>>>> you switch.
>>>>
>>>> /b
>>>>
>>>> On Dec 29, 2010, at 9:46 AM, Aloysius Lloyd wrote:
>>>>
>>>>  CentOS 64 bit ?
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Thu Dec 30 21:21:56 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 30 Dec 2010 13:21:56 -0500
Subject: [Freeswitch-users] FreeSWITCH SBC Setup (HA mode)
References: <AANLkTi=w5abhESY0TMugvdqbQ84a5RVNDPfvT4uxhd2+@mail.gmail.com><AANLkTik-fA0yJbK=GQpvrR=P3rGMVSOiGPj8agO7CpJ4@mail.gmail.com><AANLkTimR6OHNT=7DU3krKf_puLJC2AMpxk+2NGbwn5Oa@mail.gmail.com>
	<AANLkTi==YRnMpfj1eAZw81A+s4ZfSAEddEAHXX7PyV3c@mail.gmail.com>
Message-ID: <5335523ED41E4C0D92E979FEEA05A824@e1705>

it needs to compile the kernel with drbd module...

----- Original Message ----- 
From: "Giovanni Maruzzelli" <gmaruzz at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, December 30, 2010 1:00 PM
Subject: Re: [Freeswitch-users] FreeSWITCH SBC Setup (HA mode)


DRBD is a very interesting and very performing way to replicate a
block device between two or more machine, usally in an active-passive
way. And is very HA aware.

In classic configuration, you have two machines that control each
other state via heartbeat, and a floating IP address that is assigned
to the "active" machine (in addition to the "real" IP address).

The "active" machine is accessed from the HA application through the
floating IP.

The "passive" machine is there idle, but ready to take over in case
the "active" goes down (for failure or maintenance). In that case the
"passive" machine automatically becomes the "active", eg: gets the
floating IP.

The only drawback I can see in this setup (DRBD+nativesqlite instead
of odbc) is that if the FS instance or the machine itself crashes very
badly, maybe will trash the db, and because the sqlite db is
replicated at filesystem level via DRDB to the other machine, you'll
end up with a trashed sqlite db on the "new active" machine too.

Having instead a logic SQL replica (mysql, postgres, whatever) between
the two machines, spares you this possible point of failure.

Also, if you use another heartbeat and another floating IP address
dedicated to the odbc database access, you get HA database too
(meaning: you can do an upgrade of the database engine on the "active"
FS machine, because you moved only the floating IP of the dabase
access to the other machine).

-giovanni

On Thu, Dec 30, 2010 at 6:07 PM, Sam <u2nsam at gmail.com> wrote:
> Can use Sqlite replication in between 2 instances by sqlite3_update_hook.
>
> Regds
> Sam
>
> On Thu, Dec 30, 2010 at 9:57 PM, Avi Marcus <Avi at amarcus.com> wrote:
>>
>> The other option for failover I was considering is the default sqlite -
>> set up a job to rsync it to/from the slave constantly.
>> Perhaps even throw that into a ramdisk, if that makes a difference (the
>> kernal may be caching it anyway).
>> -Avi
>> On Thu, Dec 30, 2010 at 2:14 PM, Ghulam Mustafa <mustafa.pk at gmail.com>
>> wrote:
>>>
>>> Hi everyone,
>>> i am trying to setup FreeSWITCH as a SBC in ha mode; for this purpose i
>>> have setup two FS boxes(identical) in HA mode using heartbeat, mysql as 
>>> odbc
>>> backend and enabled track-calls in both sip profiles. Everything is 
>>> working
>>> as expected i.e sip sessions successfully recovered during failover in 
>>> both
>>> directions.
>>> i am just curious and would like to know should i go for odbc or is it
>>> possible to avoid odbc and use drbd for /opt/switch/db directory to keep 
>>> .db
>>> files in sync between two fs boxes. is it likely to work?
>>> another question is related to performance when track-calls is enabled,
>>> we are supposed to handle ~300 sip concurrent calls around the clock in
>>> proxy_media mode, coming from our sip provider (on external profile) and
>>> forwarding to our call-center server (configured as internal sip 
>>> gateway),
>>> we have plenty of good hardware (quad-core Xeon servers with 8Gigs ram), 
>>> is
>>> it good to go :) or can raise alarms?
>>>
>>> Thanks and best regards,
>>> --
>>> Ghulam Mustafa
>>> cell: +92 333.611.7681
>>> sip: cyrenity at ekiga.net
>>> mail: mustafa.pk at gmail.com
>>> web: cyrenity.wordpress.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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



From Avi at aMarcus.com  Thu Dec 30 21:34:18 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Thu, 30 Dec 2010 20:34:18 +0200
Subject: [Freeswitch-users] FreeSWITCH SBC Setup (HA mode)
In-Reply-To: <5335523ED41E4C0D92E979FEEA05A824@e1705>
References: <AANLkTi=w5abhESY0TMugvdqbQ84a5RVNDPfvT4uxhd2+@mail.gmail.com>
	<AANLkTik-fA0yJbK=GQpvrR=P3rGMVSOiGPj8agO7CpJ4@mail.gmail.com>
	<AANLkTimR6OHNT=7DU3krKf_puLJC2AMpxk+2NGbwn5Oa@mail.gmail.com>
	<AANLkTi==YRnMpfj1eAZw81A+s4ZfSAEddEAHXX7PyV3c@mail.gmail.com>
	<5335523ED41E4C0D92E979FEEA05A824@e1705>
Message-ID: <AANLkTi=hjZoMTe+55PD+AJ2tqFg5qFfLNVfagSeyJkqe@mail.gmail.com>

Linode has some great guides for setting it up:
http://library.linode.com/linux-ha/

<http://library.linode.com/linux-ha/>I'm going with master-master mysql for
now because I don't want the second machine to be completely passive, I'd
like the ability to do something with it if I wished.
-Avi

On Thu, Dec 30, 2010 at 8:21 PM, Madovsky <infos at madovsky.org> wrote:

> it needs to compile the kernel with drbd module...
>
> ----- Original Message -----
> From: "Giovanni Maruzzelli" <gmaruzz at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, December 30, 2010 1:00 PM
> Subject: Re: [Freeswitch-users] FreeSWITCH SBC Setup (HA mode)
>
>
> DRBD is a very interesting and very performing way to replicate a
> block device between two or more machine, usally in an active-passive
> way. And is very HA aware.
>
> In classic configuration, you have two machines that control each
> other state via heartbeat, and a floating IP address that is assigned
> to the "active" machine (in addition to the "real" IP address).
>
> The "active" machine is accessed from the HA application through the
> floating IP.
>
> The "passive" machine is there idle, but ready to take over in case
> the "active" goes down (for failure or maintenance). In that case the
> "passive" machine automatically becomes the "active", eg: gets the
> floating IP.
>
> The only drawback I can see in this setup (DRBD+nativesqlite instead
> of odbc) is that if the FS instance or the machine itself crashes very
> badly, maybe will trash the db, and because the sqlite db is
> replicated at filesystem level via DRDB to the other machine, you'll
> end up with a trashed sqlite db on the "new active" machine too.
>
> Having instead a logic SQL replica (mysql, postgres, whatever) between
> the two machines, spares you this possible point of failure.
>
> Also, if you use another heartbeat and another floating IP address
> dedicated to the odbc database access, you get HA database too
> (meaning: you can do an upgrade of the database engine on the "active"
> FS machine, because you moved only the floating IP of the dabase
> access to the other machine).
>
> -giovanni
>
> On Thu, Dec 30, 2010 at 6:07 PM, Sam <u2nsam at gmail.com> wrote:
> > Can use Sqlite replication in between 2 instances by sqlite3_update_hook.
> >
> > Regds
> > Sam
> >
> > On Thu, Dec 30, 2010 at 9:57 PM, Avi Marcus <Avi at amarcus.com> wrote:
> >>
> >> The other option for failover I was considering is the default sqlite -
> >> set up a job to rsync it to/from the slave constantly.
> >> Perhaps even throw that into a ramdisk, if that makes a difference (the
> >> kernal may be caching it anyway).
> >> -Avi
> >> On Thu, Dec 30, 2010 at 2:14 PM, Ghulam Mustafa <mustafa.pk at gmail.com>
> >> wrote:
> >>>
> >>> Hi everyone,
> >>> i am trying to setup FreeSWITCH as a SBC in ha mode; for this purpose i
> >>> have setup two FS boxes(identical) in HA mode using heartbeat, mysql as
> >>> odbc
> >>> backend and enabled track-calls in both sip profiles. Everything is
> >>> working
> >>> as expected i.e sip sessions successfully recovered during failover in
> >>> both
> >>> directions.
> >>> i am just curious and would like to know should i go for odbc or is it
> >>> possible to avoid odbc and use drbd for /opt/switch/db directory to
> keep
> >>> .db
> >>> files in sync between two fs boxes. is it likely to work?
> >>> another question is related to performance when track-calls is enabled,
> >>> we are supposed to handle ~300 sip concurrent calls around the clock in
> >>> proxy_media mode, coming from our sip provider (on external profile)
> and
> >>> forwarding to our call-center server (configured as internal sip
> >>> gateway),
> >>> we have plenty of good hardware (quad-core Xeon servers with 8Gigs
> ram),
> >>> is
> >>> it good to go :) or can raise alarms?
> >>>
> >>> Thanks and best regards,
> >>> --
> >>> Ghulam Mustafa
> >>> cell: +92 333.611.7681
> >>> sip: cyrenity at ekiga.net
> >>> mail: mustafa.pk at gmail.com
> >>> web: cyrenity.wordpress.com
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Dec 30 21:43:49 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Dec 2010 10:43:49 -0800
Subject: [Freeswitch-users] ftmod_libpri: support for overlap dial
In-Reply-To: <AANLkTikxnBQk5=3p0_HRJ4LySpUWSLSjhz-=THzUQmJb@mail.gmail.com>
References: <AANLkTikxnBQk5=3p0_HRJ4LySpUWSLSjhz-=THzUQmJb@mail.gmail.com>
Message-ID: <AANLkTik1N6nEFhJ3ervsMA+x3pAz+w+nmnimU4fiLc2W@mail.gmail.com>

Moises said that this isn't supported yet but hopefully one day soon...

On Wed, Dec 29, 2010 at 8:54 PM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Dear Friends,
>     When I tried ftmod_libpri to check whether it supports overlap dial. It
> didn't work. So, is there anyone trying to implement overlap dial in
> ftmod_libpri?
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Dec 30 21:49:38 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Dec 2010 10:49:38 -0800
Subject: [Freeswitch-users] Freeswitch with TLS
In-Reply-To: <921D5B375C81104CB7EB2C498260FBA1015262B7@mail.splendor.net>
References: <921D5B375C81104CB7EB2C498260FBA1015262B6@mail.splendor.net>
	<AANLkTin=5+Fv2NqWyb_Yj7=obVkE_74xdMr4Ot1p98UO@mail.gmail.com>
	<921D5B375C81104CB7EB2C498260FBA1015262B7@mail.splendor.net>
Message-ID: <AANLkTi=6hkeRoECdp-A0bgrSXNmdN3Y=_VHhqshjVuJa@mail.gmail.com>

How much TLS traffic are you sending? Do your user re-register ever 2
seconds? I find it hard to believe that just SIP signaling via TLS would
cause performance degradation unless there is an inordinate amount of
encryption/decryption going on and your CPU is a wuss...

-MC

On Thu, Dec 30, 2010 at 6:40 AM, Michel Freiha <freiham at splendor.net> wrote:

> Dear Sir,
>
> I know that openSIPS supports TLS...the major issue that we are facing
> performance degradation once the number of users using TLS increase and
> reach some hundreds...
>
> I just first need to know how many concurrent registered users using TLS
> FS can support?
> Second, is there any manual that I can follow in order to accomplish
> this scenario?
>
> Regards
>
> Michel Freiha
> Technical Manager
> Splendor Telecom (www.splendor.net)
> Beirut, Lebanon
> Phone: +961 1 373725
> Fax: +961 1 375554
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Steven Ayre
> Sent: Thursday, December 30, 2010 4:33 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Freeswitch with TLS
>
> FreeSWITCH is a b2bua not proxy server.
>
> However, yes it would be possible to accept a SIP TLS call and bridge it
> to a call going to OpenSIPS. It won't forward the original messages, but
> will forward the signalling (i.e. you'll get ringing, answered etc
> passed along but it'll be in separate calls not just adding a Via header
> when forwarding).
>
> OpenSIPS supports TLS itself though - any particular reason you're not
> using its own support?
>
> -Steve
>
>
> On 30 December 2010 11:50, Michel  Freiha <freiham at splendor.net> wrote:
> > Dear Sir,
> >
> >
> >
> > I'm planning to use Freeswitch server as a proxy server before
> > OpenSIPS...The Job of freeswitch is only when I need to use SIP over
> TLS
> > or SRTP when there is a severe firewall on user side...The Client will
>
> > send SIP over TLS to freeswitch and the freeswitch will forward these
> > packets to openSIPS or kamailio or whatever Registrar server...
> >
> >
> >
> > Is that possible?
> >
> > How many concurrent TLS connection a freeswitch server can handle?
> >
> >
> >
> > Regards
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From saeedahmad1981 at gmail.com  Thu Dec 30 22:06:56 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Thu, 30 Dec 2010 20:06:56 +0100
Subject: [Freeswitch-users] Asynchronous PTIME
In-Reply-To: <4D1CBC10.8080004@sns.eu>
References: <4D127733.8010409@nevian.org> <4D15266B.3080903@sns.eu>
	<AANLkTimzmg1kfM4+75e-Y9vG6wErhXjX6f83+4-voNyu@mail.gmail.com>
	<4D1CBC10.8080004@sns.eu>
Message-ID: <AANLkTim6JbGqr+tid8Unrr5T7n78Zxyr4xUAFtWHJW_i@mail.gmail.com>

Thanks Jan, I tried that, but still with ptime: 40 the voice is choppy :(


On Thu, Dec 30, 2010 at 6:06 PM, Jan Riedinger <riedinger at sns.eu> wrote:

>  You have to set it the paramater in the profile, which is used for the
> call, e. g. set in external.xml:
>  <param name="rtp-autofix-timing" value="false"/>
>
> BR
>    Jan
>
>
>
> Am 29.12.2010 20:04, schrieb Saeed Ahmed:
>
> on which profile did you put that param?
>
> I've allnet phone if i use g729 with ptime:40 then i can't hear called
> party properly, its choppy voice, i put autofix param in both internal and
> external profiles, but it didn't help.
>
>  if i use ptime: 20 then it works fine.
>
> On Sat, Dec 25, 2010 at 12:02 AM, Jan Riedinger <riedinger at sns.eu> wrote:
>
>> Hi Serge,
>>
>> I had similar problems, look for my thread "Problematic Behaviour of FS
>> regarding ptime negotiation" in October. I could solve the problem by
>> setting "rtp-autofix-timing=false", which disabled the (too) smart
>> behaviour of FreeSwitch.
>>
>> BR
>>     Jan
>>
>>
>> Am 22.12.2010 23:09, schrieb Serge S. Yuriev:
>> > Hello,
>> >
>> > 2010-12-20 16:51:13.960502 [WARNING] mod_sofia.c:1036 Asynchronous PTIME
>> > not supported, changing our end from 20 to 60
>> >
>> > I'm getting this warning and client hears chopped sound :(
>> > That is "Our end"?
>> >
>> > Call flow is Cisco (sip) FS (sip) YATE (h323) MeraTransitSoftSwitch
>> > All but MVTS under my control.
>> >
>> > I doesn't see any clue in logs and can't reproduce this with my testing
>> > via Blink/PortSIP/PhonerLite/Twinkle or Grandstream devices
>> >
>> > Which debug/logs I should take? Any ideas?
>> >
>> > Thanks a lot.
>> >
>> > btw how I can save debug into log not only console?
>>
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From robert.hadley at teotech.com  Thu Dec 30 22:44:57 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 30 Dec 2010 11:44:57 -0800
Subject: [Freeswitch-users] CentOS 5.5 - FreeSWITCH - Audio Quality
 issues
In-Reply-To: <AANLkTimRe-pyehB7zcOsEiUTwd-t0g7eCj-do790DVXJ@mail.gmail.com>
References: <AANLkTik3y8EfTi5s_a_1BURnUJe8Nv0p+5y8Ph6Jkcg3@mail.gmail.com>
	<AANLkTi=6shTJg0C2eFsar+mdjstT330e2kaUtFxsPQ1F@mail.gmail.com>
	<AANLkTi=3Yi_Phq6Zv2Ep-YEgf=3orNmOEpLVyww=7x=L@mail.gmail.com>
	<316084A7-BCA6-4490-8990-9894331F999F@freeswitch.org>
	<AANLkTimREKiD=3t12WT7fMWWR7JPyOT+3S4jm7SPY6UT@mail.gmail.com>
	<AANLkTintspeXDQiyHXsu2GmuyiK0yqVKrBKyBmwmY-fB@mail.gmail.com>
	<AE8F4FDB-95EF-49E6-80A1-A91315DCF2E4@freeswitch.org>
	<AANLkTinLXcLEbw7qcyFwgyaX5zoa0uENpy9YK3UraND2@mail.gmail.com>
	<AANLkTimrj4xO3n5sMYnsb80w5PE6gb4nJLv7NVkCqw81@mail.gmail.com>
	<AANLkTimc6M2sN3-2cp5as=2b=sREeuWhByjDwvL4L-gJ@mail.gmail.com>
	<AANLkTimRe-pyehB7zcOsEiUTwd-t0g7eCj-do790DVXJ@mail.gmail.com>
Message-ID: <E826C94A3C6BE44198FAF4D66B5AB4D80338493845@VA3DIAXVS351.RED001.local>


Using CentOS 5.3 64-bit, we initially had audio quality issues with supermicro motherboard but found they came with disk write caching disabled.  Enabling disk write caching significantly improved audio quality.

Regards,
Robert


From: Aloysius Lloyd [mailto:lloyd.aloysius at gmail.com]
Sent: Thursday, December 30, 2010 10:15 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] CentOS 5.5 - FreeSWITCH - Audio Quality issues

Norman,

I use iWeb ( www.iweb.ca<http://www.iweb.ca> )  dedicated server . Motherboard - Supermicro .

Thanks
Lloyd

On Thu, Dec 30, 2010 at 1:09 PM, Norman Tomlins <norm at voicenetwork.ca<mailto:norm at voicenetwork.ca>> wrote:
Lloyd,

What motherboard was it that gave you the issues?

Norman Tomlins
Voice Network Inc

On Thu, Dec 30, 2010 at 12:57 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com<mailto:lloyd.aloysius at gmail.com>> wrote:
Finally solved the problem. There is a hardware (Motherboard) issue on the server. Replace the server with the new one and use 64bit ... voice quality problems goes away.

Thank you for the help.

Thanks and regards,
Lloyd



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From Avi at aMarcus.com  Fri Dec 31 00:33:50 2010
From: Avi at aMarcus.com (Avi Marcus)
Date: Thu, 30 Dec 2010 23:33:50 +0200
Subject: [Freeswitch-users] FreeSWITCH SBC Setup (HA mode)
In-Reply-To: <5A3B6C9C-637B-427B-95FF-AA50DF64A1CE@gmail.com>
References: <AANLkTi=w5abhESY0TMugvdqbQ84a5RVNDPfvT4uxhd2+@mail.gmail.com>
	<AANLkTik-fA0yJbK=GQpvrR=P3rGMVSOiGPj8agO7CpJ4@mail.gmail.com>
	<5A3B6C9C-637B-427B-95FF-AA50DF64A1CE@gmail.com>
Message-ID: <AANLkTimGfZfv+zF17-deFMY5dVkFv3Ltb0XTWCPiJs32@mail.gmail.com>

The CORE db for freeswitch doesn't really matter - it's all state
information (other than voicemail DB), so if it really all goes to hell,
e.g. in a crash, then you're no worse off than opening a clean instance of
freeswitch.
-Avi

On Thu, Dec 30, 2010 at 7:44 PM, Steven Ayre <steveayre at gmail.com> wrote:

> That'll leave you wide open to having a corrupted db, drbd too.
>
> Personally i would use mysql cluster via odbc - all data on 2 nodes, which
> may or may not be on separate hardware to fs.
>
> Steve on iPhone
>
> On 30 Dec 2010, at 16:27, Avi Marcus <Avi at aMarcus.com> wrote:
>
> The other option for failover I was considering is the default sqlite - set
> up a job to rsync it to/from the slave constantly.
> Perhaps even throw that into a ramdisk, if that makes a difference (the
> kernal may be caching it anyway).
> -Avi
>
> On Thu, Dec 30, 2010 at 2:14 PM, Ghulam Mustafa <mustafa.pk@<http://gmail.com>
> gmail.com> wrote:
>
>> Hi everyone,
>>
>> i am trying to setup FreeSWITCH as a SBC in ha mode; for this purpose i
>> have setup two FS boxes(identical) in HA mode using heartbeat, mysql as odbc
>> backend and enabled track-calls in both sip profiles. Everything is working
>> as expected i.e sip sessions successfully recovered during failover in both
>> directions.
>>
>> i am just curious and would like to know should i go for odbc or is it
>> possible to avoid odbc and use drbd for /opt/switch/db directory to keep .db
>> files in sync between two fs boxes. is it likely to work?
>>
>> another question is related to performance when track-calls is enabled, we
>> are supposed to handle ~300 sip concurrent calls around the clock in
>> proxy_media mode, coming from our sip provider (on external profile) and
>> forwarding to our call-center server (configured as internal sip gateway),
>> we have plenty of good hardware (quad-core Xeon servers with 8Gigs ram), is
>> it good to go :) or can raise alarms?
>>
>>
>> Thanks and best regards,
>>
>> --
>> Ghulam Mustafa
>> cell: +92 333.611.7681
>> sip: <cyrenity at ekiga.net>cyrenity at ekiga.net
>> mail: mustafa.pk@ <http://gmail.com>gmail.com
>> web: <http://cyrenity.wordpress.com>cyrenity.wordpress.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sameer2k3t at gmail.com  Fri Dec 31 01:31:13 2010
From: sameer2k3t at gmail.com (Sameer)
Date: Fri, 31 Dec 2010 03:31:13 +0500
Subject: [Freeswitch-users] Group / channel limit
Message-ID: <1c1merv4y36vp0bnre0c5qs2.1293748273602@email.android.com>

Hi guys,
Is there a way to check limit status or number of active calls on a number  from XML curl PHP 
And then return the dial plan commands?
  

From gmaruzz at gmail.com  Fri Dec 31 01:40:36 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 30 Dec 2010 23:40:36 +0100
Subject: [Freeswitch-users] FreeSWITCH SBC Setup (HA mode)
In-Reply-To: <AANLkTimGfZfv+zF17-deFMY5dVkFv3Ltb0XTWCPiJs32@mail.gmail.com>
References: <AANLkTi=w5abhESY0TMugvdqbQ84a5RVNDPfvT4uxhd2+@mail.gmail.com>
	<AANLkTik-fA0yJbK=GQpvrR=P3rGMVSOiGPj8agO7CpJ4@mail.gmail.com>
	<5A3B6C9C-637B-427B-95FF-AA50DF64A1CE@gmail.com>
	<AANLkTimGfZfv+zF17-deFMY5dVkFv3Ltb0XTWCPiJs32@mail.gmail.com>
Message-ID: <AANLkTik-eL1DE-tW69F114bXM2DFd2pQ84FgxKmt7mcq@mail.gmail.com>

The OP was about an HA implementation with live migration of calls
from one machine to another without drop.
The OPoster implemented this kind of HA correctly, and was asking
about the possible drawbacks of changing the db storage from odbc
(with a master-master active-passive replica, I suppose) to a native
sqlite replicated via DRBD.
I believe avoiding a clean restart is one of his requirement, or at
least a strongly desired feature.
-giovanni

On 12/30/10, Avi Marcus <Avi at amarcus.com> wrote:
> The CORE db for freeswitch doesn't really matter - it's all state
> information (other than voicemail DB), so if it really all goes to hell,
> e.g. in a crash, then you're no worse off than opening a clean instance of
> freeswitch.
> -Avi
>
> On Thu, Dec 30, 2010 at 7:44 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> That'll leave you wide open to having a corrupted db, drbd too.
>>
>> Personally i would use mysql cluster via odbc - all data on 2 nodes, which
>> may or may not be on separate hardware to fs.
>>
>> Steve on iPhone
>>
>> On 30 Dec 2010, at 16:27, Avi Marcus <Avi at aMarcus.com> wrote:
>>
>> The other option for failover I was considering is the default sqlite -
>> set
>> up a job to rsync it to/from the slave constantly.
>> Perhaps even throw that into a ramdisk, if that makes a difference (the
>> kernal may be caching it anyway).
>> -Avi
>>
>> On Thu, Dec 30, 2010 at 2:14 PM, Ghulam Mustafa
>> <mustafa.pk@<http://gmail.com>
>> gmail.com> wrote:
>>
>>> Hi everyone,
>>>
>>> i am trying to setup FreeSWITCH as a SBC in ha mode; for this purpose i
>>> have setup two FS boxes(identical) in HA mode using heartbeat, mysql as
>>> odbc
>>> backend and enabled track-calls in both sip profiles. Everything is
>>> working
>>> as expected i.e sip sessions successfully recovered during failover in
>>> both
>>> directions.
>>>
>>> i am just curious and would like to know should i go for odbc or is it
>>> possible to avoid odbc and use drbd for /opt/switch/db directory to keep
>>> .db
>>> files in sync between two fs boxes. is it likely to work?
>>>
>>> another question is related to performance when track-calls is enabled,
>>> we
>>> are supposed to handle ~300 sip concurrent calls around the clock in
>>> proxy_media mode, coming from our sip provider (on external profile) and
>>> forwarding to our call-center server (configured as internal sip
>>> gateway),
>>> we have plenty of good hardware (quad-core Xeon servers with 8Gigs ram),
>>> is
>>> it good to go :) or can raise alarms?
>>>
>>>
>>> Thanks and best regards,
>>>
>>> --
>>> Ghulam Mustafa
>>> cell: +92 333.611.7681
>>> sip: <cyrenity at ekiga.net>cyrenity at ekiga.net
>>> mail: mustafa.pk@ <http://gmail.com>gmail.com
>>> web: <http://cyrenity.wordpress.com>cyrenity.wordpress.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From u2nsam at gmail.com  Fri Dec 31 07:20:01 2010
From: u2nsam at gmail.com (Sam)
Date: Fri, 31 Dec 2010 09:50:01 +0530
Subject: [Freeswitch-users] CPS
In-Reply-To: <AANLkTikgQ2kXjmQMPN5aWZPE7RvfV88XOb8_qT2GHRbw@mail.gmail.com>
References: <AANLkTi=GEYTXe4MREfxeSaZc5y==Mv4e9rQCoJVHE2ES@mail.gmail.com>
	<33ACD386-5CD2-49E2-87EA-34BBFD985443@freeswitch.org>
	<AANLkTi=w7b1UnbjYFOQA-rWLPVEbQGOGE_L12otMcXdz@mail.gmail.com>
	<AANLkTikgQ2kXjmQMPN5aWZPE7RvfV88XOb8_qT2GHRbw@mail.gmail.com>
Message-ID: <AANLkTi=6W+o7jH6sgdDXQVE8GyY=GXY7ox1JALe4K7Hy@mail.gmail.com>

Does it has to use so much of resource of FS for 4 core for 70 cps to
200%cpu because the media was flow around .
Here i was relaying the signaling and passing some 8-9 custom headers to the
next hop.

Regards
Sam

On Thu, Dec 30, 2010 at 11:23 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> yah cos remember its the the percent times the number of cores for the
> max so 4 core box has 400% of cpu.
> press '1' on top to see each cpu and how much is being used.
>
>
> On Thu, Dec 30, 2010 at 11:27 AM, jonathan augenstine
> <jaugenstine at gmail.com> wrote:
> > Sam,
> > I have been doing some work on a server with 16 cores and I have see
> > Freeswitch go to 900% and it is working great.
> > Jonathan
> >
> > On Thu, Dec 30, 2010 at 7:21 AM, Brian West <brian at freeswitch.org>
> wrote:
> >>
> >> This all depends on what you're doing and how hard you're kicking its
> ass.
> >>  Just don't come crying when you run out of threads!  :P
> >>
> >> /b
> >>
> >> On Dec 29, 2010, at 11:12 PM, Sam wrote:
> >>
> >> > Hello,
> >> >
> >> > Was testing the environment with media bypass, could see the CPU for
> FS
> >> > application shooting to 200% on quadracore 64 bits debian for CPS of
> 70.
> >> >
> >> > Is it true for the environment or can be scaled further ?
> >> >
> >> > Regards
> >> > Sam
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mustafa.pk at gmail.com  Fri Dec 31 08:35:29 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Fri, 31 Dec 2010 10:35:29 +0500
Subject: [Freeswitch-users] FreeSWITCH SBC Setup (HA mode)
In-Reply-To: <AANLkTik-eL1DE-tW69F114bXM2DFd2pQ84FgxKmt7mcq@mail.gmail.com>
References: <AANLkTi=w5abhESY0TMugvdqbQ84a5RVNDPfvT4uxhd2+@mail.gmail.com>
	<AANLkTik-fA0yJbK=GQpvrR=P3rGMVSOiGPj8agO7CpJ4@mail.gmail.com>
	<5A3B6C9C-637B-427B-95FF-AA50DF64A1CE@gmail.com>
	<AANLkTimGfZfv+zF17-deFMY5dVkFv3Ltb0XTWCPiJs32@mail.gmail.com>
	<AANLkTik-eL1DE-tW69F114bXM2DFd2pQ84FgxKmt7mcq@mail.gmail.com>
Message-ID: <AANLkTinNUuU3SXSgrJ+a9hty5rRxe=OH-aHQ9ZpHS8jr@mail.gmail.com>

@Giovanni;

Yes you are right, i have my desired setup in place. A 2 node mysql-cluster
is in place, was just thinking about the possibility of sqlite db fs/block
level replication. thank you everyone for putting thoughts on it.

though my second question about track-calls performance went un-answered, i
hope one of dev's can explain how it can degrade performance on a high load
server.

keeping scalability in view i'd love to know how a rpc or curl backend for
core db will perform :) though i know it's not yet implemented neither there
are any chances but just a thought.

Regards,

-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com
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From u2nsam at gmail.com  Fri Dec 31 08:57:17 2010
From: u2nsam at gmail.com (Sam)
Date: Fri, 31 Dec 2010 11:27:17 +0530
Subject: [Freeswitch-users] scaling
Message-ID: <AANLkTi=4-b_eUbKwwhKdJXCN4QD9iE52EB3bsXAD7W0q@mail.gmail.com>

Hi,

Can it be done such that 2 servers having identical configuration for
conference ,
and treating those 2 servers as one whole server .
Such that when the conference room is initiated on one server ,the person on
the
second server can join the same conference room, how such could be
synchronized.
Such that any one logs in on either server can initiate the same conference
room and talk to each other.


Regards
Sam
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From infos at madovsky.org  Fri Dec 31 09:05:16 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 31 Dec 2010 01:05:16 -0500
Subject: [Freeswitch-users] scaling
References: <AANLkTi=4-b_eUbKwwhKdJXCN4QD9iE52EB3bsXAD7W0q@mail.gmail.com>
Message-ID: <14CB9067C33B424DA53AFA73431B0C0A@e1705>

use ODBC
  ----- Original Message ----- 
  From: Sam 
  To: FreeSWITCH Users Help 
  Sent: Friday, December 31, 2010 12:57 AM
  Subject: [Freeswitch-users] scaling


  Hi,

  Can it be done such that 2 servers having identical configuration for conference ,
  and treating those 2 servers as one whole server .
  Such that when the conference room is initiated on one server ,the person on the
  second server can join the same conference room, how such could be synchronized.
  Such that any one logs in on either server can initiate the same conference room and talk to each other.


  Regards
  Sam



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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From justlikeef at gmail.com  Fri Dec 31 10:19:10 2010
From: justlikeef at gmail.com (Rob Hutton)
Date: Fri, 31 Dec 2010 02:19:10 -0500
Subject: [Freeswitch-users] mod_callcenter features
In-Reply-To: <AANLkTin7FDKWoRjB2p=qDtGp37mUjySOmiMP1bUH2DFS@mail.gmail.com>
References: <1293442949323-5869082.post@n2.nabble.com>
	<201012300742.11829.justlikeef@gmail.com>
	<AANLkTin7FDKWoRjB2p=qDtGp37mUjySOmiMP1bUH2DFS@mail.gmail.com>
Message-ID: <201012310219.10813.justlikeef@gmail.com>

1. Try something like:

reloadxml
callcenter_config queue load|reload queue_name

2. I don't believe so.  Hit Moc on the IRC channel.  I am trying, also...


-- 
Thanks,
Rob
On Friday 31 December 2010 00:32:46 Rajkumar K wrote:
> Hi,
> 
> 1. In mod_callcenter, there is no command like "callcenter_config queue add"
> for adding queues dynamically. We need to configure it in
> callcenter.conf.xml file and run "callcenter_config queue reload <queue
> name>". So is there any other ways to add queues.
> 
> 2. The mod_callcenter automatically bridges the call once it identifies the
> available agent. But the requirement is to identify the available agent and
> leave the bridge control to our dial plan. There we need to other operations
> (like announcing caller name to agent) instead of  bridging the call.
> 
> regards
> rajkumar k
> 
> On Thu, Dec 30, 2010 at 6:12 PM, Rob Hutton <justlikeef at gmail.com> wrote:
> 
> > Some of the API commands are on in the Wiki under the heading API:
> >
> > http://wiki.freeswitch.org/wiki/Mod_callcenter
> >
> > Everything starts with callcenter_config, so you can feel your way around
> > the command line and everything is fairly self explanatory.
> >
> > What do you mean control the call flow?  Can you give an example of what
> > you are trying to accomplish?
> >
> > --
> > Thanks,
> > Rob
> > On Monday 27 December 2010 04:42:29 rajkumar wrote:
> > >
> > > Hi,
> > >
> > >   I am developing an application with mod_callcenter. I need to know the
> > > following about mod_callcenter.
> > >
> > >   * Is it possible to add/update/delete the queue configurations
> > dynamically
> > > without using static xml configurations.
> > >   * How can I control the call flow (for playback message and recording)
> > > before and after bridging the call.
> > >
> > > Thanks in advance
> > >
> > > regards
> > > rajkumar k
> > >
> >
> 


From u2nsam at gmail.com  Fri Dec 31 12:50:20 2010
From: u2nsam at gmail.com (Sam)
Date: Fri, 31 Dec 2010 15:20:20 +0530
Subject: [Freeswitch-users] CPS
In-Reply-To: <AANLkTi=6W+o7jH6sgdDXQVE8GyY=GXY7ox1JALe4K7Hy@mail.gmail.com>
References: <AANLkTi=GEYTXe4MREfxeSaZc5y==Mv4e9rQCoJVHE2ES@mail.gmail.com>
	<33ACD386-5CD2-49E2-87EA-34BBFD985443@freeswitch.org>
	<AANLkTi=w7b1UnbjYFOQA-rWLPVEbQGOGE_L12otMcXdz@mail.gmail.com>
	<AANLkTikgQ2kXjmQMPN5aWZPE7RvfV88XOb8_qT2GHRbw@mail.gmail.com>
	<AANLkTi=6W+o7jH6sgdDXQVE8GyY=GXY7ox1JALe4K7Hy@mail.gmail.com>
Message-ID: <AANLkTim-ybHFVdvH67arHT2ZJPEo8eob7srEGHThqizT@mail.gmail.com>

Some mistakes it 8 core and not 4 core,

Does it has to use so much of resource of FS for 4 core for 70 cps to 200%
cpu because the media was flow around .
Here i was relaying the signaling and passing some 8-9 custom headers to the
next hop.

Regards
Sam


On Fri, Dec 31, 2010 at 9:50 AM, Sam <u2nsam at gmail.com> wrote:

> Does it has to use so much of resource of FS for 4 core for 70 cps to
> 200%cpu because the media was flow around .
> Here i was relaying the signaling and passing some 8-9 custom headers to
> the next hop.
>
> Regards
> Sam
>
>
> On Thu, Dec 30, 2010 at 11:23 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> yah cos remember its the the percent times the number of cores for the
>> max so 4 core box has 400% of cpu.
>> press '1' on top to see each cpu and how much is being used.
>>
>>
>> On Thu, Dec 30, 2010 at 11:27 AM, jonathan augenstine
>> <jaugenstine at gmail.com> wrote:
>> > Sam,
>> > I have been doing some work on a server with 16 cores and I have see
>> > Freeswitch go to 900% and it is working great.
>> > Jonathan
>> >
>> > On Thu, Dec 30, 2010 at 7:21 AM, Brian West <brian at freeswitch.org>
>> wrote:
>> >>
>> >> This all depends on what you're doing and how hard you're kicking its
>> ass.
>> >>  Just don't come crying when you run out of threads!  :P
>> >>
>> >> /b
>> >>
>> >> On Dec 29, 2010, at 11:12 PM, Sam wrote:
>> >>
>> >> > Hello,
>> >> >
>> >> > Was testing the environment with media bypass, could see the CPU for
>> FS
>> >> > application shooting to 200% on quadracore 64 bits debian for CPS of
>> 70.
>> >> >
>> >> > Is it true for the environment or can be scaled further ?
>> >> >
>> >> > Regards
>> >> > Sam
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From steveayre at gmail.com  Fri Dec 31 17:07:31 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 31 Dec 2010 14:07:31 +0000
Subject: [Freeswitch-users] Group / channel limit
In-Reply-To: <1c1merv4y36vp0bnre0c5qs2.1293748273602@email.android.com>
References: <1c1merv4y36vp0bnre0c5qs2.1293748273602@email.android.com>
Message-ID: <AANLkTikEp=uS7QBuzi3o0KqwZOvTdueQx9=HQbvFTqm7@mail.gmail.com>

There's two possibly ways.

First, you can connect via ESL from PHP and do 'show channels as xml',
the output will show you all incoming and outgoing channels. You might
be able to work out from that how many calls are going to a particular
number.

The other (better) way would be to use the Limit functionality to
record the number of calls to the number, then check the number of
calls currently recorded. That could be done either via api commands
via ESL or if you use an ODBC backend querying the database directly.

See http://wiki.freeswitch.org/wiki/Limit

Depending on what you're doing, you could also entirely handle the
Limit checks in the dialplan XML you return, which would avoid having
to do anything with ESL/database from PHP. Remember you can return
more than one extension, so transfers will still work within the
dialplan you return.

Regards,
-Steve



On 30 December 2010 22:31, Sameer <sameer2k3t at gmail.com> wrote:
> Hi guys,
> Is there a way to check limit status or number of active calls on a number ?from XML curl PHP
> And then return the dial plan commands?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Fri Dec 31 17:46:29 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 31 Dec 2010 08:46:29 -0600
Subject: [Freeswitch-users] CPS
In-Reply-To: <AANLkTim-ybHFVdvH67arHT2ZJPEo8eob7srEGHThqizT@mail.gmail.com>
References: <AANLkTi=GEYTXe4MREfxeSaZc5y==Mv4e9rQCoJVHE2ES@mail.gmail.com>
	<33ACD386-5CD2-49E2-87EA-34BBFD985443@freeswitch.org>
	<AANLkTi=w7b1UnbjYFOQA-rWLPVEbQGOGE_L12otMcXdz@mail.gmail.com>
	<AANLkTikgQ2kXjmQMPN5aWZPE7RvfV88XOb8_qT2GHRbw@mail.gmail.com>
	<AANLkTi=6W+o7jH6sgdDXQVE8GyY=GXY7ox1JALe4K7Hy@mail.gmail.com>
	<AANLkTim-ybHFVdvH67arHT2ZJPEo8eob7srEGHThqizT@mail.gmail.com>
Message-ID: <AANLkTimRAiYZbWe6pgxKnfXBmsAfuaQXd=PfEk+m+XZ5@mail.gmail.com>

So you are only using 25% of the boxes resources.
Like I said press 1 during top and you'll propably see in reality its even
less.
On Dec 31, 2010 3:51 AM, "Sam" <u2nsam at gmail.com> wrote:
> Some mistakes it 8 core and not 4 core,
>
> Does it has to use so much of resource of FS for 4 core for 70 cps to 200%
> cpu because the media was flow around .
> Here i was relaying the signaling and passing some 8-9 custom headers to
the
> next hop.
>
> Regards
> Sam
>
>
> On Fri, Dec 31, 2010 at 9:50 AM, Sam <u2nsam at gmail.com> wrote:
>
>> Does it has to use so much of resource of FS for 4 core for 70 cps to
>> 200%cpu because the media was flow around .
>> Here i was relaying the signaling and passing some 8-9 custom headers to
>> the next hop.
>>
>> Regards
>> Sam
>>
>>
>> On Thu, Dec 30, 2010 at 11:23 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> yah cos remember its the the percent times the number of cores for the
>>> max so 4 core box has 400% of cpu.
>>> press '1' on top to see each cpu and how much is being used.
>>>
>>>
>>> On Thu, Dec 30, 2010 at 11:27 AM, jonathan augenstine
>>> <jaugenstine at gmail.com> wrote:
>>> > Sam,
>>> > I have been doing some work on a server with 16 cores and I have see
>>> > Freeswitch go to 900% and it is working great.
>>> > Jonathan
>>> >
>>> > On Thu, Dec 30, 2010 at 7:21 AM, Brian West <brian at freeswitch.org>
>>> wrote:
>>> >>
>>> >> This all depends on what you're doing and how hard you're kicking its
>>> ass.
>>> >> Just don't come crying when you run out of threads! :P
>>> >>
>>> >> /b
>>> >>
>>> >> On Dec 29, 2010, at 11:12 PM, Sam wrote:
>>> >>
>>> >> > Hello,
>>> >> >
>>> >> > Was testing the environment with media bypass, could see the CPU
for
>>> FS
>>> >> > application shooting to 200% on quadracore 64 bits debian for CPS
of
>>> 70.
>>> >> >
>>> >> > Is it true for the environment or can be scaled further ?
>>> >> >
>>> >> > Regards
>>> >> > Sam
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com><
MSN%3Aanthony_minessale at hotmail.com <MSN%253Aanthony_minessale at hotmail.com>>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
<PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org><
sip%3A888 at conference.freeswitch.org <sip%253A888 at conference.freeswitch.org>>
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
<googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
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From steveayre at gmail.com  Fri Dec 31 18:09:33 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 31 Dec 2010 15:09:33 +0000
Subject: [Freeswitch-users] CPS
In-Reply-To: <AANLkTi=GEYTXe4MREfxeSaZc5y==Mv4e9rQCoJVHE2ES@mail.gmail.com>
References: <AANLkTi=GEYTXe4MREfxeSaZc5y==Mv4e9rQCoJVHE2ES@mail.gmail.com>
Message-ID: <AANLkTinQ+z_OgJZZv5A83UC0bQS86f_73cc1uxHbT9bT@mail.gmail.com>

On 8 cores you should be able to get up to 800% (100% per core). An
individual core could max out before that though, you'd need to press
1 on top as Anthony was suggesting to see if that's the case.

Yes, you should be able to increase the cps beyond 70. You can
probably expect to handle at least 200 cps.

-Steve



On 30 December 2010 05:12, Sam <u2nsam at gmail.com> wrote:
> Hello,
>
> Was testing the environment with media bypass, could see the CPU for FS
> application shooting to 200% on quadracore 64 bits debian for CPS of 70.
>
> Is it true for the environment or can be scaled further ?
>
> Regards
> Sam
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From freiham at splendor.net  Fri Dec 31 22:28:19 2010
From: freiham at splendor.net (Michel  Freiha)
Date: Fri, 31 Dec 2010 21:28:19 +0200
Subject: [Freeswitch-users] Freeswitch with TLS
References: <921D5B375C81104CB7EB2C498260FBA1015262B6@mail.splendor.net><AANLkTin=5+Fv2NqWyb_Yj7=obVkE_74xdMr4Ot1p98UO@mail.gmail.com><921D5B375C81104CB7EB2C498260FBA1015262B7@mail.splendor.net>
	<AANLkTi=6hkeRoECdp-A0bgrSXNmdN3Y=_VHhqshjVuJa@mail.gmail.com>
Message-ID: <921D5B375C81104CB7EB2C498260FBA1E68FD4@mail.splendor.net>

I can have around 5000 concurrent users that need to connect to OpenSIPs using TLS...I'm sure that will not find any CPU that can handle this amount of traffic...Rerigistration is done every 3600 secs

Regars




-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Michael Collins
Sent: Thu 12/30/2010 8:49 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Freeswitch with TLS
 
How much TLS traffic are you sending? Do your user re-register ever 2
seconds? I find it hard to believe that just SIP signaling via TLS would
cause performance degradation unless there is an inordinate amount of
encryption/decryption going on and your CPU is a wuss...

-MC

On Thu, Dec 30, 2010 at 6:40 AM, Michel Freiha <freiham at splendor.net> wrote:

> Dear Sir,
>
> I know that openSIPS supports TLS...the major issue that we are facing
> performance degradation once the number of users using TLS increase and
> reach some hundreds...
>
> I just first need to know how many concurrent registered users using TLS
> FS can support?
> Second, is there any manual that I can follow in order to accomplish
> this scenario?
>
> Regards
>
> Michel Freiha
> Technical Manager
> Splendor Telecom (www.splendor.net)
> Beirut, Lebanon
> Phone: +961 1 373725
> Fax: +961 1 375554
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Steven Ayre
> Sent: Thursday, December 30, 2010 4:33 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Freeswitch with TLS
>
> FreeSWITCH is a b2bua not proxy server.
>
> However, yes it would be possible to accept a SIP TLS call and bridge it
> to a call going to OpenSIPS. It won't forward the original messages, but
> will forward the signalling (i.e. you'll get ringing, answered etc
> passed along but it'll be in separate calls not just adding a Via header
> when forwarding).
>
> OpenSIPS supports TLS itself though - any particular reason you're not
> using its own support?
>
> -Steve
>
>
> On 30 December 2010 11:50, Michel  Freiha <freiham at splendor.net> wrote:
> > Dear Sir,
> >
> >
> >
> > I'm planning to use Freeswitch server as a proxy server before
> > OpenSIPS...The Job of freeswitch is only when I need to use SIP over
> TLS
> > or SRTP when there is a severe firewall on user side...The Client will
>
> > send SIP over TLS to freeswitch and the freeswitch will forward these
> > packets to openSIPS or kamailio or whatever Registrar server...
> >
> >
> >
> > Is that possible?
> >
> > How many concurrent TLS connection a freeswitch server can handle?
> >
> >
> >
> > Regards
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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From freiham at splendor.net  Fri Dec 31 22:30:42 2010
From: freiham at splendor.net (Michel  Freiha)
Date: Fri, 31 Dec 2010 21:30:42 +0200
Subject: [Freeswitch-users] Freeswitch with TLS
References: <921D5B375C81104CB7EB2C498260FBA1015262B6@mail.splendor.net><AANLkTin=5+Fv2NqWyb_Yj7=obVkE_74xdMr4Ot1p98UO@mail.gmail.com><921D5B375C81104CB7EB2C498260FBA1015262B7@mail.splendor.net>
	<AANLkTi=6hkeRoECdp-A0bgrSXNmdN3Y=_VHhqshjVuJa@mail.gmail.com>
Message-ID: <921D5B375C81104CB7EB2C498260FBA1E68FD5@mail.splendor.net>



Mickael, can I use freeswitch for only proxying SIP and media to OpenSIPS? I do not need it as Registrar server, only a Proxy...Is that possible? Ids there any manual describing this topic?

Regards


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Michael Collins
Sent: Thu 12/30/2010 8:49 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Freeswitch with TLS
 
How much TLS traffic are you sending? Do your user re-register ever 2
seconds? I find it hard to believe that just SIP signaling via TLS would
cause performance degradation unless there is an inordinate amount of
encryption/decryption going on and your CPU is a wuss...

-MC

On Thu, Dec 30, 2010 at 6:40 AM, Michel Freiha <freiham at splendor.net> wrote:

> Dear Sir,
>
> I know that openSIPS supports TLS...the major issue that we are facing
> performance degradation once the number of users using TLS increase and
> reach some hundreds...
>
> I just first need to know how many concurrent registered users using TLS
> FS can support?
> Second, is there any manual that I can follow in order to accomplish
> this scenario?
>
> Regards
>
> Michel Freiha
> Technical Manager
> Splendor Telecom (www.splendor.net)
> Beirut, Lebanon
> Phone: +961 1 373725
> Fax: +961 1 375554
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Steven Ayre
> Sent: Thursday, December 30, 2010 4:33 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Freeswitch with TLS
>
> FreeSWITCH is a b2bua not proxy server.
>
> However, yes it would be possible to accept a SIP TLS call and bridge it
> to a call going to OpenSIPS. It won't forward the original messages, but
> will forward the signalling (i.e. you'll get ringing, answered etc
> passed along but it'll be in separate calls not just adding a Via header
> when forwarding).
>
> OpenSIPS supports TLS itself though - any particular reason you're not
> using its own support?
>
> -Steve
>
>
> On 30 December 2010 11:50, Michel  Freiha <freiham at splendor.net> wrote:
> > Dear Sir,
> >
> >
> >
> > I'm planning to use Freeswitch server as a proxy server before
> > OpenSIPS...The Job of freeswitch is only when I need to use SIP over
> TLS
> > or SRTP when there is a severe firewall on user side...The Client will
>
> > send SIP over TLS to freeswitch and the freeswitch will forward these
> > packets to openSIPS or kamailio or whatever Registrar server...
> >
> >
> >
> > Is that possible?
> >
> > How many concurrent TLS connection a freeswitch server can handle?
> >
> >
> >
> > Regards
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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From infos at madovsky.org  Fri Dec 31 22:45:25 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 31 Dec 2010 14:45:25 -0500
Subject: [Freeswitch-users] Freeswitch with TLS
References: <921D5B375C81104CB7EB2C498260FBA1015262B6@mail.splendor.net><AANLkTin=5+Fv2NqWyb_Yj7=obVkE_74xdMr4Ot1p98UO@mail.gmail.com><921D5B375C81104CB7EB2C498260FBA1015262B7@mail.splendor.net><AANLkTi=6hkeRoECdp-A0bgrSXNmdN3Y=_VHhqshjVuJa@mail.gmail.com>
	<921D5B375C81104CB7EB2C498260FBA1E68FD5@mail.splendor.net>
Message-ID: <4E5DDAC69811402AB6194D10D77DCDD1@e1705>

RE: [Freeswitch-users] Freeswitch with TLSopensips and kamalio are proxies already
you can try also RTPproxy
  ----- Original Message ----- 
  From: Michel Freiha 
  To: FreeSWITCH Users Help ; FreeSWITCH Users Help 
  Sent: Friday, December 31, 2010 2:30 PM
  Subject: Re: [Freeswitch-users] Freeswitch with TLS





  Mickael, can I use freeswitch for only proxying SIP and media to OpenSIPS? I do not need it as Registrar server, only a Proxy...Is that possible? Ids there any manual describing this topic?

  Regards


  -----Original Message-----
  From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Michael Collins
  Sent: Thu 12/30/2010 8:49 PM
  To: FreeSWITCH Users Help
  Subject: Re: [Freeswitch-users] Freeswitch with TLS

  How much TLS traffic are you sending? Do your user re-register ever 2
  seconds? I find it hard to believe that just SIP signaling via TLS would
  cause performance degradation unless there is an inordinate amount of
  encryption/decryption going on and your CPU is a wuss...

  -MC

  On Thu, Dec 30, 2010 at 6:40 AM, Michel Freiha <freiham at splendor.net> wrote:

  > Dear Sir,
  >
  > I know that openSIPS supports TLS...the major issue that we are facing
  > performance degradation once the number of users using TLS increase and
  > reach some hundreds...
  >
  > I just first need to know how many concurrent registered users using TLS
  > FS can support?
  > Second, is there any manual that I can follow in order to accomplish
  > this scenario?
  >
  > Regards
  >
  > Michel Freiha
  > Technical Manager
  > Splendor Telecom (www.splendor.net)
  > Beirut, Lebanon
  > Phone: +961 1 373725
  > Fax: +961 1 375554
  >
  >
  > -----Original Message-----
  > From: freeswitch-users-bounces at lists.freeswitch.org
  > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
  > Steven Ayre
  > Sent: Thursday, December 30, 2010 4:33 PM
  > To: FreeSWITCH Users Help
  > Subject: Re: [Freeswitch-users] Freeswitch with TLS
  >
  > FreeSWITCH is a b2bua not proxy server.
  >
  > However, yes it would be possible to accept a SIP TLS call and bridge it
  > to a call going to OpenSIPS. It won't forward the original messages, but
  > will forward the signalling (i.e. you'll get ringing, answered etc
  > passed along but it'll be in separate calls not just adding a Via header
  > when forwarding).
  >
  > OpenSIPS supports TLS itself though - any particular reason you're not
  > using its own support?
  >
  > -Steve
  >
  >
  > On 30 December 2010 11:50, Michel  Freiha <freiham at splendor.net> wrote:
  > > Dear Sir,
  > >
  > >
  > >
  > > I'm planning to use Freeswitch server as a proxy server before
  > > OpenSIPS...The Job of freeswitch is only when I need to use SIP over
  > TLS
  > > or SRTP when there is a severe firewall on user side...The Client will
  >
  > > send SIP over TLS to freeswitch and the freeswitch will forward these
  > > packets to openSIPS or kamailio or whatever Registrar server...
  > >
  > >
  > >
  > > Is that possible?
  > >
  > > How many concurrent TLS connection a freeswitch server can handle?
  > >
  > >
  > >
  > > Regards
  > >
  > >
  > >
  > > _______________________________________________
  > > FreeSWITCH-users mailing list
  > > FreeSWITCH-users at lists.freeswitch.org
  > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
  > > rs
  > > http://www.freeswitch.org
  > >
  > >
  >
  > _______________________________________________
  > FreeSWITCH-users mailing list
  > FreeSWITCH-users at lists.freeswitch.org
  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  > http://www.freeswitch.org
  >
  > _______________________________________________
  > FreeSWITCH-users mailing list
  > FreeSWITCH-users at lists.freeswitch.org
  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  > http://www.freeswitch.org
  >





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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From steveayre at gmail.com  Fri Dec 31 23:07:30 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 31 Dec 2010 20:07:30 +0000
Subject: [Freeswitch-users] Freeswitch with TLS
In-Reply-To: <921D5B375C81104CB7EB2C498260FBA1E68FD5@mail.splendor.net>
References: <921D5B375C81104CB7EB2C498260FBA1015262B6@mail.splendor.net>
	<AANLkTin=5+Fv2NqWyb_Yj7=obVkE_74xdMr4Ot1p98UO@mail.gmail.com>
	<921D5B375C81104CB7EB2C498260FBA1015262B7@mail.splendor.net>
	<AANLkTi=6hkeRoECdp-A0bgrSXNmdN3Y=_VHhqshjVuJa@mail.gmail.com>
	<921D5B375C81104CB7EB2C498260FBA1E68FD5@mail.splendor.net>
Message-ID: <AANLkTi=CUhMeUM0VgS4_48geJHaaYdp5GOyP1RB8gSO5@mail.gmail.com>

Michel,

OpenSIPS/Kamalio/etc are proxies themselves and support TLS. They
would do so more efficiently than FreeSWITCH since they proxy the
calls rather than being a b2bua which starts a full session for each
call as FS does. MediaProxy can handle RTP, if you need that (but I
doubt that you would).

I would suggest that you have several OpenSIPS proxy servers in front
of your current OpenSIPS server which use TLS on incoming and forward
on to your current box unencrypted.

You would probably be better off looking at how you can create a
cluster of OpenSIPS servers which accept TLS and accept registrations,
so that you can add extra servers to the cluster as the load on them
increases. That would scale far better than everyone registering to a
single box.

-Steve





On 31 December 2010 19:30, Michel  Freiha <freiham at splendor.net> wrote:
>
>
> Mickael, can I use freeswitch for only proxying SIP and media to OpenSIPS? I
> do not need it as Registrar server, only a Proxy...Is that possible? Ids
> there any manual describing this topic?
>
> Regards
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Michael
> Collins
> Sent: Thu 12/30/2010 8:49 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Freeswitch with TLS
>
> How much TLS traffic are you sending? Do your user re-register ever 2
> seconds? I find it hard to believe that just SIP signaling via TLS would
> cause performance degradation unless there is an inordinate amount of
> encryption/decryption going on and your CPU is a wuss...
>
> -MC
>
> On Thu, Dec 30, 2010 at 6:40 AM, Michel Freiha <freiham at splendor.net> wrote:
>
>> Dear Sir,
>>
>> I know that openSIPS supports TLS...the major issue that we are facing
>> performance degradation once the number of users using TLS increase and
>> reach some hundreds...
>>
>> I just first need to know how many concurrent registered users using TLS
>> FS can support?
>> Second, is there any manual that I can follow in order to accomplish
>> this scenario?
>>
>> Regards
>>
>> Michel Freiha
>> Technical Manager
>> Splendor Telecom (www.splendor.net)
>> Beirut, Lebanon
>> Phone: +961 1 373725
>> Fax: +961 1 375554
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Steven Ayre
>> Sent: Thursday, December 30, 2010 4:33 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Freeswitch with TLS
>>
>> FreeSWITCH is a b2bua not proxy server.
>>
>> However, yes it would be possible to accept a SIP TLS call and bridge it
>> to a call going to OpenSIPS. It won't forward the original messages, but
>> will forward the signalling (i.e. you'll get ringing, answered etc
>> passed along but it'll be in separate calls not just adding a Via header
>> when forwarding).
>>
>> OpenSIPS supports TLS itself though - any particular reason you're not
>> using its own support?
>>
>> -Steve
>>
>>
>> On 30 December 2010 11:50, Michel? Freiha <freiham at splendor.net> wrote:
>> > Dear Sir,
>> >
>> >
>> >
>> > I'm planning to use Freeswitch server as a proxy server before
>> > OpenSIPS...The Job of freeswitch is only when I need to use SIP over
>> TLS
>> > or SRTP when there is a severe firewall on user side...The Client will
>>
>> > send SIP over TLS to freeswitch and the freeswitch will forward these
>> > packets to openSIPS or kamailio or whatever Registrar server...
>> >
>> >
>> >
>> > Is that possible?
>> >
>> > How many concurrent TLS connection a freeswitch server can handle?
>> >
>> >
>> >
>> > Regards
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> > rs
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From freiham at splendor.net  Fri Dec 31 23:06:46 2010
From: freiham at splendor.net (Michel  Freiha)
Date: Fri, 31 Dec 2010 22:06:46 +0200
Subject: [Freeswitch-users] Freeswitch with TLS
References: <921D5B375C81104CB7EB2C498260FBA1015262B6@mail.splendor.net><AANLkTin=5+Fv2NqWyb_Yj7=obVkE_74xdMr4Ot1p98UO@mail.gmail.com><921D5B375C81104CB7EB2C498260FBA1015262B7@mail.splendor.net><AANLkTi=6hkeRoECdp-A0bgrSXNmdN3Y=_VHhqshjVuJa@mail.gmail.com><921D5B375C81104CB7EB2C498260FBA1E68FD5@mail.splendor.net>
	<4E5DDAC69811402AB6194D10D77DCDD1@e1705>
Message-ID: <921D5B375C81104CB7EB2C498260FBA1E68FD6@mail.splendor.net>

Yes I know...I need to benefit from TLS support on FreeSwitch to forward SIP and Media to OpenSIPs or Kamailio...Just need to know if this is possible or not

Regards



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Madovsky
Sent: Fri 12/31/2010 9:45 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Freeswitch with TLS
 
RE: [Freeswitch-users] Freeswitch with TLSopensips and kamalio are proxies already
you can try also RTPproxy
  ----- Original Message ----- 
  From: Michel Freiha 
  To: FreeSWITCH Users Help ; FreeSWITCH Users Help 
  Sent: Friday, December 31, 2010 2:30 PM
  Subject: Re: [Freeswitch-users] Freeswitch with TLS





  Mickael, can I use freeswitch for only proxying SIP and media to OpenSIPS? I do not need it as Registrar server, only a Proxy...Is that possible? Ids there any manual describing this topic?

  Regards


  -----Original Message-----
  From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Michael Collins
  Sent: Thu 12/30/2010 8:49 PM
  To: FreeSWITCH Users Help
  Subject: Re: [Freeswitch-users] Freeswitch with TLS

  How much TLS traffic are you sending? Do your user re-register ever 2
  seconds? I find it hard to believe that just SIP signaling via TLS would
  cause performance degradation unless there is an inordinate amount of
  encryption/decryption going on and your CPU is a wuss...

  -MC

  On Thu, Dec 30, 2010 at 6:40 AM, Michel Freiha <freiham at splendor.net> wrote:

  > Dear Sir,
  >
  > I know that openSIPS supports TLS...the major issue that we are facing
  > performance degradation once the number of users using TLS increase and
  > reach some hundreds...
  >
  > I just first need to know how many concurrent registered users using TLS
  > FS can support?
  > Second, is there any manual that I can follow in order to accomplish
  > this scenario?
  >
  > Regards
  >
  > Michel Freiha
  > Technical Manager
  > Splendor Telecom (www.splendor.net)
  > Beirut, Lebanon
  > Phone: +961 1 373725
  > Fax: +961 1 375554
  >
  >
  > -----Original Message-----
  > From: freeswitch-users-bounces at lists.freeswitch.org
  > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
  > Steven Ayre
  > Sent: Thursday, December 30, 2010 4:33 PM
  > To: FreeSWITCH Users Help
  > Subject: Re: [Freeswitch-users] Freeswitch with TLS
  >
  > FreeSWITCH is a b2bua not proxy server.
  >
  > However, yes it would be possible to accept a SIP TLS call and bridge it
  > to a call going to OpenSIPS. It won't forward the original messages, but
  > will forward the signalling (i.e. you'll get ringing, answered etc
  > passed along but it'll be in separate calls not just adding a Via header
  > when forwarding).
  >
  > OpenSIPS supports TLS itself though - any particular reason you're not
  > using its own support?
  >
  > -Steve
  >
  >
  > On 30 December 2010 11:50, Michel  Freiha <freiham at splendor.net> wrote:
  > > Dear Sir,
  > >
  > >
  > >
  > > I'm planning to use Freeswitch server as a proxy server before
  > > OpenSIPS...The Job of freeswitch is only when I need to use SIP over
  > TLS
  > > or SRTP when there is a severe firewall on user side...The Client will
  >
  > > send SIP over TLS to freeswitch and the freeswitch will forward these
  > > packets to openSIPS or kamailio or whatever Registrar server...
  > >
  > >
  > >
  > > Is that possible?
  > >
  > > How many concurrent TLS connection a freeswitch server can handle?
  > >
  > >
  > >
  > > Regards
  > >
  > >
  > >
  > > _______________________________________________
  > > FreeSWITCH-users mailing list
  > > FreeSWITCH-users at lists.freeswitch.org
  > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
  > > rs
  > > http://www.freeswitch.org
  > >
  > >
  >
  > _______________________________________________
  > FreeSWITCH-users mailing list
  > FreeSWITCH-users at lists.freeswitch.org
  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  > http://www.freeswitch.org
  >
  > _______________________________________________
  > FreeSWITCH-users mailing list
  > FreeSWITCH-users at lists.freeswitch.org
  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  > http://www.freeswitch.org
  >





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


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From infos at madovsky.org  Fri Dec 31 23:51:51 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 31 Dec 2010 15:51:51 -0500
Subject: [Freeswitch-users] tone_detect and dinging
Message-ID: <C5A8430009F1461592CDA253B4F64DC9@e1705>

I use this in my dialplan before a bridge

...
<action application="set" data="instant_ringback=true"/>
<action application="ring_ready"/>
<action application="tone_detect" data="fax 1100 r +5000 transfer 'receivefax XML features' 1" />
....

but no ring is back to the caller. if I remove tone_detect the ringback is working again.
is anyone knows why ?

Thanks
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