From mgg at giagnocavo.net  Thu Sep  1 00:21:10 2011
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Wed, 31 Aug 2011 16:21:10 -0400
Subject: [Freeswitch-users] problem using mod_managed under linux
Message-ID: <rjxwrb84m9olrbg6m7ghyb97.1314822061391@email.android.com>

Yea I have been meaning to get around to it. I'll put it on my todo and see if I can figure it out. Sorry for the inconvenience.

"covici at ccs.covici.com" <covici at ccs.covici.com> wrote:


Any chance of you fixing the bug?  I think you must be right because in
the server stack trace I see a lot of things involving serialization.

Michael Giagnocavo <mgg at giagnocavo.net> wrote:

> It's probably more related to some cross-appdomain/serialization stuff that's specific to Mono. I wrote mod_managed against Mono 2.4 or so, and I think last time I ran it was 2.6, and only on CentOS 5. Probably something changed and is triggering a bug in mod_managed in the newer builds.
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim McQueen
> Sent: Wednesday, August 31, 2011 9:08 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] problem using mod_managed under linux
>
> I'm having the same problem.  (http://pastebin.freeswitch.org/17216)
>
> Have you tried the Demo.csx to see if it will even run? In my case it won't.  It is my opinon that it's because the FreeSwitch.Managed.dll is dependent specifically on C# 4.0, since the same code will run correctly on Windows.
>
> I sent a message out to the freeswitch-devs group but I don't think it was approved by the moderator, because I haven't seen it come back through the list.  Someone on IRC said that mod_managed isn't maintained anymore, but I see activity in FishEye from last week.
> On Tue, Aug 30, 2011 at 5:39 AM, <covici at ccs.covici.com<mailto:covici at ccs.covici.com>> wrote:
> Hi.  I am trying to use mod_managed on my linux box -- gentoo
> distribution -- and its driving me ...
>
>
> I wrote a very simple program for a test.  The program just sets a dtmf
> callback and then streams a file and writes a log entry when it sees the
> dtmf.
>
> Under Windows 7 net framework 4, it works just fine.  Under Linux, using
> mono versions 2.8.2 qand 2.10.4, I get the following exception which I
> will put in a pastebin.
>
> http://pastebin.freeswitch.org/17232
>
> And the app is in this one
>
> http://pastebin.freeswitch.org/17233
>
> I have tried using gmcs and dmcs thinking it may be a .net framework
> issue, and if I do it wrong, I can bring down fs itself, but the best I
> can get under Linux is the exception shown above.
>
> Thanks in advance for anything you can come up with on this.
>
> --
> 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<mailto:covici at ccs.covici.com>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ----------------------------------------------------
> 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


From anthony.minessale at gmail.com  Thu Sep  1 00:48:38 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 31 Aug 2011 15:48:38 -0500
Subject: [Freeswitch-users] Condition based on custom sip header
In-Reply-To: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A8B6E34@jjex01.jajah.dublin>
References: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A75B7CD@jjex01.jajah.dublin>
	<CAKbxfG-pD89e2p+JtYi__kngCt1azbNkCvmnHDUsLGfjbuB-qA@mail.gmail.com>
	<F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A75B951@jjex01.jajah.dublin>
	<CAKbxfG_dCxQ9qfmtPJMPfDJYr6ZFLyttmxDJmyyRFczhvCN1-g@mail.gmail.com>
	<1314787698150-6745987.post@n2.nabble.com>
	<F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A8B6E34@jjex01.jajah.dublin>
Message-ID: <CAKbxfG8Kif9VZqq6woA=92oqBBRY6sm87LVKFjVZcie2t=TmNg@mail.gmail.com>

and that's what I would have told you had you not made a comment about
how we are "strange" because we do not comb over every sip message and
turn every header into a into a variable in case somebody wants one
even when we have only had 2 requests for it in 6 years.  if a
particular non-standard header becomes popular then you can submit a
patch adding that specific header to the list..............



2011/8/31 Alex Massover <alex at jajah.com>:
> Hi,
>
> That's great! Exactly what I need.
>
> Thanks!
>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
>> users-bounces at lists.freeswitch.org] On Behalf Of Dissident
>> Sent: ????? 31 ?????? 2011 13:48
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Condition based on custom sip header
>>
>> Hello Alex,
>>
>> I had to face the same situation weeks ago...
>> Yes, It's shame that many hardware/software makers are ignoring the
>> RFCs and
>> doing things their own way but what can you do...
>> Here is how I sorted it out.
>>
>> http://freeswitch-users.2379917.n2.nabble.com/Sip-Headers-advice-Not-
>> parsing-properly-td6606461.html
>>
>> good luck
>>
>>
>> --
>> View this message in context: http://freeswitch-
>> users.2379917.n2.nabble.com/Condition-based-on-custom-sip-header-
>> tp6738108p6745987.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
>>
>> This mail was received via Mail-SeCure System.
>>
>
>
> This mail was sent via Mail-SeCure System.
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Sep  1 00:48:37 2011
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Wed, 31 Aug 2011 16:48:37 -0400
Subject: [Freeswitch-users] VoiceMail Issue - Call Disconnect
Message-ID: <CAGvuWYZZD0zaLgqvDnOxbQ_HC6DRv034fQQLaiG2Tqm+BLi7tA@mail.gmail.com>

Hi All,

I am having difficulty to find the cause of the problem. Voicemail deliver
in partially.

IVR -> Extension -> Voicemail

While leaving the message (voicemail) the calls disconnect. This is
happening randomly.

<action application="bridge" data="loopback/app=voicemail:default
${domain_name} ${dialed_extension}"/>


How to fix this issue or How to troubleshoot this problem or is there any
parameter to fix this problem.

Thanks
Lloyd
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/4c940f7a/attachment.html 

From covici at ccs.covici.com  Thu Sep  1 01:16:06 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 31 Aug 2011 17:16:06 -0400
Subject: [Freeswitch-users] problem using mod_managed under linux
In-Reply-To: <rjxwrb84m9olrbg6m7ghyb97.1314822061391@email.android.com>
References: <rjxwrb84m9olrbg6m7ghyb97.1314822061391@email.android.com>
Message-ID: <20320.1314825366@ccs.covici.com>

Thanks so much for working on this.

Michael Giagnocavo <mgg at giagnocavo.net> wrote:

> Yea I have been meaning to get around to it. I'll put it on my todo and see if I can figure it out. Sorry for the inconvenience.
> 
> "covici at ccs.covici.com" <covici at ccs.covici.com> wrote:
> 
> 
> Any chance of you fixing the bug?  I think you must be right because in
> the server stack trace I see a lot of things involving serialization.
> 
> Michael Giagnocavo <mgg at giagnocavo.net> wrote:
> 
> > It's probably more related to some cross-appdomain/serialization stuff that's specific to Mono. I wrote mod_managed against Mono 2.4 or so, and I think last time I ran it was 2.6, and only on CentOS 5. Probably something changed and is triggering a bug in mod_managed in the newer builds.
> >
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim McQueen
> > Sent: Wednesday, August 31, 2011 9:08 AM
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] problem using mod_managed under linux
> >
> > I'm having the same problem.  (http://pastebin.freeswitch.org/17216)
> >
> > Have you tried the Demo.csx to see if it will even run? In my case it won't.  It is my opinon that it's because the FreeSwitch.Managed.dll is dependent specifically on C# 4.0, since the same code will run correctly on Windows.
> >
> > I sent a message out to the freeswitch-devs group but I don't think it was approved by the moderator, because I haven't seen it come back through the list.  Someone on IRC said that mod_managed isn't maintained anymore, but I see activity in FishEye from last week.
> > On Tue, Aug 30, 2011 at 5:39 AM, <covici at ccs.covici.com<mailto:covici at ccs.covici.com>> wrote:
> > Hi.  I am trying to use mod_managed on my linux box -- gentoo
> > distribution -- and its driving me ...
> >
> >
> > I wrote a very simple program for a test.  The program just sets a dtmf
> > callback and then streams a file and writes a log entry when it sees the
> > dtmf.
> >
> > Under Windows 7 net framework 4, it works just fine.  Under Linux, using
> > mono versions 2.8.2 qand 2.10.4, I get the following exception which I
> > will put in a pastebin.
> >
> > http://pastebin.freeswitch.org/17232
> >
> > And the app is in this one
> >
> > http://pastebin.freeswitch.org/17233
> >
> > I have tried using gmcs and dmcs thinking it may be a .net framework
> > issue, and if I do it wrong, I can bring down fs itself, but the best I
> > can get under Linux is the exception shown above.
> >
> > Thanks in advance for anything you can come up with on this.
> >
> > --
> > 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<mailto:covici at ccs.covici.com>
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > ----------------------------------------------------
> > 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
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mario_fs at mgtech.com  Thu Sep  1 01:17:33 2011
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 31 Aug 2011 14:17:33 -0700
Subject: [Freeswitch-users] OS X PPC Leopard error on "make"
In-Reply-To: <8319344044664188306@unknownmsgid>
References: <8319344044664188306@unknownmsgid>
Message-ID: <088B3EEE-10C1-4639-9896-E5E747229E3B@mgtech.com>

I have not updated FS in a while but find it's prereqs seem to change. I need to update the wiki with the info below about pkg_config which may be your problem. When it was missing I got multiple errors later on. Check to see if there is a problem with this in earlier messages. I plan the update the wiki when I install osX 10.7 Lion but that will be a few months. Also will update devtools info, etc. On the other hand this could be related to the fact that's it's a PowerPC which I no longer have. Hope this helps.
Mario G


src directory, build and install:
    cd ~/Downloads
    mv pkg-config-0.25 /usr/local/src
    cd /usr/local/src/pkg-config-0.25
    ./configure
    make
    sudo make install	<? sudo is required because binaries are loaded into /usr/bin



On Aug 31, 2011, at 4:25 AM, Peter Schrock wrote:

> I am new to FreeSWITCH (sort of, tried it a while back and dropped it).
> I have a powerpc Apple running Leopard on it. I followed the directions by Mario G and managed to get all the way up to where he discusses error's with FLITE. I managed to go without the error message because I followed the instructions on how to fix it. However, I have come up with this error message and can't seem to make any sense of it. Help Please.
> 
> Here is the error I get:
> make[8]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'.  Stop.
> make[7]: *** [all-recursive] Error 1
> Making all in packages
> make[6]: *** [all-recursive] Error 1
> make[5]: *** [all] Error 2
> make[4]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2
> make[3]: *** [mod_sofia-all] Error 1
> make[2]: *** [all-recursive] Error 1
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
> 
> 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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/ed43be0e/attachment-0001.html 

From moises.silva at gmail.com  Thu Sep  1 02:44:33 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 31 Aug 2011 18:44:33 -0400
Subject: [Freeswitch-users] Sangoma is hiring!
Message-ID: <CAA4nhyCo0se-UfBZRKenDj_jdjVhK7YiQo7pW466U29GfzguAQ@mail.gmail.com>

Hello everyone, I apologize in advance for the spam.

Sangoma is looking for talented software engineers for development and
QA. Roughly speaking we would love to have people with open source
mentality and that knows his/her way around Linux and multithreaded C.
Kernel level coding is an asset. They should be willing to relocate to
work in our offices at Toronto, Canada.

Full description in LinkedIn:
http://ca.linkedin.com/jobs/jobs-Software-Engineer-VoIPTDMVideoNetworking-1879437

Interested candidates can contact me directly.

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


From msc at freeswitch.org  Thu Sep  1 04:46:29 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 31 Aug 2011 17:46:29 -0700
Subject: [Freeswitch-users] Investigating build testing for FreeSWITCH
Message-ID: <CAKzWOxX5w7E4JFRRgKwmwXH-6YsdeHANO4aZha91V1_Y+79SyQ@mail.gmail.com>

Hey all,

On today's conference call someone pointed out that there is a utility
called BuildBot that helps to automate testing of new commits to git, svn,
etc. You can find it at buildbot.net. The catch: it's written in Python.
(blech)

I tried to build it on 3 different systems and I get 3 different
undecipherable Python-ish errors. (Stuff like syntax errors during the build
or syntax errors in twstd while it's running, dependency mismatches for
python-virtualenv, etc.) If you have a system that can test this out, and
you can stand to use Python this much (:D) then please try out the steps
outlined here:

http://buildbot.net/buildbot/docs/current/tutorial/firstrun.html

Supposedly this thing will let us run multiple tests on multiple boxes and
aggregate the results. I'll believe it when I see it. :) If you get this
working please let me know. Also, if you can set up a test platform that
would also be helpful. Ideally we would have multiple Linuxes, OS X,
Windows, etc. and test on all these platforms. Let me know what you come up
with.

Thanks,
Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/40b4675a/attachment.html 

From jmesquita at freeswitch.org  Thu Sep  1 04:59:39 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 31 Aug 2011 21:59:39 -0300
Subject: [Freeswitch-users] [Freeswitch-dev] Investigating build testing
	for FreeSWITCH
In-Reply-To: <CAKzWOxX5w7E4JFRRgKwmwXH-6YsdeHANO4aZha91V1_Y+79SyQ@mail.gmail.com>
References: <CAKzWOxX5w7E4JFRRgKwmwXH-6YsdeHANO4aZha91V1_Y+79SyQ@mail.gmail.com>
Message-ID: <CABH8QX3QFdP6eqXdxWecMMW8GWJhS+foz0hL9LYAMYq6jw-cig@mail.gmail.com>

I can promise to give it a serious try. I am out of the country this week
but I will try to get my hands dirty either way. I kinda know python despite
of what bkw_ has to say about it. LOL

Regards,

JM

On Wednesday, August 31, 2011, Michael Collins wrote:

> Hey all,
>
> On today's conference call someone pointed out that there is a utility
> called BuildBot that helps to automate testing of new commits to git, svn,
> etc. You can find it at buildbot.net. The catch: it's written in Python.
> (blech)
>
> I tried to build it on 3 different systems and I get 3 different
> undecipherable Python-ish errors. (Stuff like syntax errors during the build
> or syntax errors in twstd while it's running, dependency mismatches for
> python-virtualenv, etc.) If you have a system that can test this out, and
> you can stand to use Python this much (:D) then please try out the steps
> outlined here:
>
> http://buildbot.net/buildbot/docs/current/tutorial/firstrun.html
>
> Supposedly this thing will let us run multiple tests on multiple boxes and
> aggregate the results. I'll believe it when I see it. :) If you get this
> working please let me know. Also, if you can set up a test platform that
> would also be helpful. Ideally we would have multiple Linuxes, OS X,
> Windows, etc. and test on all these platforms. Let me know what you come up
> with.
>
> Thanks,
> Michael
>


-- 
Jo?o Mesquita
FreeSWITCH? Solutions
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/7fd8a702/attachment.html 

From xing2kin at yahoo.com  Thu Sep  1 06:21:02 2011
From: xing2kin at yahoo.com (king2kin)
Date: Wed, 31 Aug 2011 19:21:02 -0700 (PDT)
Subject: [Freeswitch-users] session:recordFile(-) always creates empty
	wav file during outbound IVR call
In-Reply-To: <CAKbxfG89m2cDYkWCqJrHAwse_2Ei26d7CBbOTcuykc4h9Eqw8g@mail.gmail.com>
Message-ID: <1314843662.40400.YahooMailClassic@web39701.mail.mud.yahoo.com>

Anthony,

Thank you for help. I tried outbound IVR call in multiple ways again based on your advice,  session:recordFile(-) still doesn't work normally, still created empty wav file .

Could anyone please give me a hand on FreeSwitch Record? It always fails to record audio file during any outbound IVR call (auto dialer) although it works well during any inbound ivr call. 

Session:streamFile(-) works well to play back prompt files, dtmf keypress also works, ... during outbound ivr call. 

x.k.

--- On Wed, 8/31/11, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] session:recordFile(-) always creates empty wav file during outbound IVR call
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Wednesday, August 31, 2011, 12:17 PM
> try this dial string instead
> 
> {ignore_early_media=true}sofia/gateway/sip.tpad.com/1726011
> 
> On Wed, Aug 31, 2011 at 12:52 PM, king2kin <xing2kin at yahoo.com>
> wrote:
> > Hi folks,
> >
> > With Lua script and/or originate command, I have tried
> recording a message file during outbound IVR call over and
> over, session:recordFile(-) inside Lua script does create a
> wav file during each of my testings but the recorded audio
> file is always empty.
> >
> > However, session:recordFile(-) works well for inbound
> IVR call.
> >
> > I tried the session:recordFile(-) via Lua script in
> three ways:
> >
> > 1. run lua script "test_outcall_ivr.lua" at freeswitch
> command-line:
> >
> > luarun test_outcall_ivr.lua
> >
> >
> > -- [test_outcall_ivr.lua]
> > {
> > local sessionx =
> freeswitch.Session("sofia/gateway/sip.tpad.com/1726011",
> session)
> >
> > -- Set the path separator
> > pathsep = '/'
> >
> > -- Windows users do this instead:
> > -- pathsep = '\'
> >
> > -- Answer the call
> > -- sessionx:answer()
> >
> > --Create a string with path and filename of a sound
> file
> > prompt = "ivr" .. pathsep ..
> "ivr-welcome_to_freeswitch.wav"
> >
> > -- Print a log message
> > freeswitch.consoleLog("INFO","Prompt file is '" ..
> prompt .. "'\n")
> >
> > --Play the prompt
> > sessionx:streamFile(prompt)
> >
> > -- Record record file
> >
> sessionx:streamFile("phrase:voicemail_record_message")
> >
> > -- Play a ""bong"" tone prior to recording
> >
> sessionx:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
> 0, 640)")
> >
> > -- record a message
> > filename = sessionx:getVariable('sounds_dir') ..
> pathsep .. "123.wav"
> > sessionx:recordFile(filename,300,100,10)
> >
> > -- play back the recorded msg
> > sessionx:streamFile(filename)
> >
> > -- Hangup
> > sessionx:hangup()
> >
> > }
> >
> > 2. I also tried it differently by submitting the
> following commands at the FreeSwitch command-line
> interface:
> >
> > originate user/1005 &transfer(8887 xml default)
> >
> > originate user/1005 &lua('test1.lua')
> >
> > originate sofia/gateway/sip.tpad.com/1726011
> &lua('test1.lua')
> >
> >
> > -- [test1.lua]
> > {
> > -- Set the path separator
> > pathsep = '/'
> >
> > -- Windows users do this instead:
> > -- pathsep = '\'
> >
> > --Answer the call
> > session:answer()
> >
> > --Create a string with path and filename of a sound
> file
> > prompt = "ivr" .. pathsep ..
> "ivr-welcome_to_freeswitch.wav"
> >
> > -- Print a log message
> > freeswitch.consoleLog("INFO","Prompt file is '" ..
> prompt .. "'\n")
> >
> > --Play the prompt
> > session:streamFile(prompt)
> >
> > -- Record record file
> > session:streamFile("phrase:voicemail_record_message")
> >
> > -- Play a ""bong"" tone prior to recording
> >
> session:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
> 0, 640)")
> >
> > -- record a message
> > filename = session:getVariable('sounds_dir') ..
> pathsep .. "123.wav"
> > session:recordFile(filename,300,100,10)
> >
> > -- play back the recorded msg
> > session:streamFile(filename)
> >
> > -- Hangup
> > session:hangup()
> > }
> >
> > -- [xml dialplan for extension 8887]:
> > {
> > ? ? ? ?<extension name="Simple Lua Test">
> > ? ? ? ? ? ? ? ?<condition
> field="destination_number" expression="^(8887)$">
> > ? ? ? ? ? ? ? ? ? ? ? ?<action
> application="set" data="record_waste_resources=true"/>
> > ? ? ? ? ? ? ? ? ? ? ? ?<action
> application="lua" data="test1.lua"/>
> > ? ? ? ? ? ? ? ?</condition>
> > ? ? ? ?</extension>
> > }
> >
> >
> ======================================================
> >
> > For all the above testing cases, session:recordFile(-)
> always creates an empty wav file for each of outbound IVR
> calls, however, if I make an inbound IVR?call to run Lua
> script "test1.lua", session:recordFile(-) always works
> perfect to generate a normal wav file.
> >
> > So, what's wrong with [session:recordFile(-)] during
> an outbound IVR call?
> >
> > x.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
> >
> 
> 
> 
> -- 
> 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 jeff at jefflenk.com  Thu Sep  1 06:27:20 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 31 Aug 2011 19:27:20 -0700 (PDT)
Subject: [Freeswitch-users] [Freeswitch-dev] Investigating build testing
	for FreeSWITCH
In-Reply-To: <CABH8QX3QFdP6eqXdxWecMMW8GWJhS+foz0hL9LYAMYq6jw-cig@mail.gmail.com>
References: <CAKzWOxX5w7E4JFRRgKwmwXH-6YsdeHANO4aZha91V1_Y+79SyQ@mail.gmail.com>
	<CABH8QX3QFdP6eqXdxWecMMW8GWJhS+foz0hL9LYAMYq6jw-cig@mail.gmail.com>
Message-ID: <1314844040696-6748773.post@n2.nabble.com>

cypromis runs a continous integration server (hudson) which is a very popular
framework written in Java. He has done some cool things with it such as IRC
notifications of build problems to the dev channel and lots more. Maybe he
will pipe in here. I also run a continous integration server for the windows
builds (CruiseControl.Net) of which I receive email notifications of build
related problems if they occur for windows.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Investigating-build-testing-for-FreeSWITCH-tp6748651p6748773.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Thu Sep  1 06:46:30 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 31 Aug 2011 21:46:30 -0500
Subject: [Freeswitch-users] session:recordFile(-) always creates empty
 wav file during outbound IVR call
In-Reply-To: <1314843662.40400.YahooMailClassic@web39701.mail.mud.yahoo.com>
References: <CAKbxfG89m2cDYkWCqJrHAwse_2Ei26d7CBbOTcuykc4h9Eqw8g@mail.gmail.com>
	<1314843662.40400.YahooMailClassic@web39701.mail.mud.yahoo.com>
Message-ID: <CAKbxfG_Bi1P1Ykbf2v2XyLdSdFgugz+LDqrejzQvbLOXmF6itg@mail.gmail.com>

Does the empty file contain silence that corresponds to the duration
of the time it's recording?
Are you producing the audio yourself into the recording and can you
verify with a pcap that there is actually any audio to record?


On Wed, Aug 31, 2011 at 9:21 PM, king2kin <xing2kin at yahoo.com> wrote:
> Anthony,
>
> Thank you for help. I tried outbound IVR call in multiple ways again based on your advice, ?session:recordFile(-) still doesn't work normally, still created empty wav file .
>
> Could anyone please give me a hand on FreeSwitch Record? It always fails to record audio file during any outbound IVR call (auto dialer) although it works well during any inbound ivr call.
>
> Session:streamFile(-) works well to play back prompt files, dtmf keypress also works, ... during outbound ivr call.
>
> x.k.
>
> --- On Wed, 8/31/11, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> Subject: Re: [Freeswitch-users] session:recordFile(-) always creates empty wav file during outbound IVR call
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Date: Wednesday, August 31, 2011, 12:17 PM
>> try this dial string instead
>>
>> {ignore_early_media=true}sofia/gateway/sip.tpad.com/1726011
>>
>> On Wed, Aug 31, 2011 at 12:52 PM, king2kin <xing2kin at yahoo.com>
>> wrote:
>> > Hi folks,
>> >
>> > With Lua script and/or originate command, I have tried
>> recording a message file during outbound IVR call over and
>> over, session:recordFile(-) inside Lua script does create a
>> wav file during each of my testings but the recorded audio
>> file is always empty.
>> >
>> > However, session:recordFile(-) works well for inbound
>> IVR call.
>> >
>> > I tried the session:recordFile(-) via Lua script in
>> three ways:
>> >
>> > 1. run lua script "test_outcall_ivr.lua" at freeswitch
>> command-line:
>> >
>> > luarun test_outcall_ivr.lua
>> >
>> >
>> > -- [test_outcall_ivr.lua]
>> > {
>> > local sessionx =
>> freeswitch.Session("sofia/gateway/sip.tpad.com/1726011",
>> session)
>> >
>> > -- Set the path separator
>> > pathsep = '/'
>> >
>> > -- Windows users do this instead:
>> > -- pathsep = '\'
>> >
>> > -- Answer the call
>> > -- sessionx:answer()
>> >
>> > --Create a string with path and filename of a sound
>> file
>> > prompt = "ivr" .. pathsep ..
>> "ivr-welcome_to_freeswitch.wav"
>> >
>> > -- Print a log message
>> > freeswitch.consoleLog("INFO","Prompt file is '" ..
>> prompt .. "'\n")
>> >
>> > --Play the prompt
>> > sessionx:streamFile(prompt)
>> >
>> > -- Record record file
>> >
>> sessionx:streamFile("phrase:voicemail_record_message")
>> >
>> > -- Play a ""bong"" tone prior to recording
>> >
>> sessionx:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
>> 0, 640)")
>> >
>> > -- record a message
>> > filename = sessionx:getVariable('sounds_dir') ..
>> pathsep .. "123.wav"
>> > sessionx:recordFile(filename,300,100,10)
>> >
>> > -- play back the recorded msg
>> > sessionx:streamFile(filename)
>> >
>> > -- Hangup
>> > sessionx:hangup()
>> >
>> > }
>> >
>> > 2. I also tried it differently by submitting the
>> following commands at the FreeSwitch command-line
>> interface:
>> >
>> > originate user/1005 &transfer(8887 xml default)
>> >
>> > originate user/1005 &lua('test1.lua')
>> >
>> > originate sofia/gateway/sip.tpad.com/1726011
>> &lua('test1.lua')
>> >
>> >
>> > -- [test1.lua]
>> > {
>> > -- Set the path separator
>> > pathsep = '/'
>> >
>> > -- Windows users do this instead:
>> > -- pathsep = '\'
>> >
>> > --Answer the call
>> > session:answer()
>> >
>> > --Create a string with path and filename of a sound
>> file
>> > prompt = "ivr" .. pathsep ..
>> "ivr-welcome_to_freeswitch.wav"
>> >
>> > -- Print a log message
>> > freeswitch.consoleLog("INFO","Prompt file is '" ..
>> prompt .. "'\n")
>> >
>> > --Play the prompt
>> > session:streamFile(prompt)
>> >
>> > -- Record record file
>> > session:streamFile("phrase:voicemail_record_message")
>> >
>> > -- Play a ""bong"" tone prior to recording
>> >
>> session:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
>> 0, 640)")
>> >
>> > -- record a message
>> > filename = session:getVariable('sounds_dir') ..
>> pathsep .. "123.wav"
>> > session:recordFile(filename,300,100,10)
>> >
>> > -- play back the recorded msg
>> > session:streamFile(filename)
>> >
>> > -- Hangup
>> > session:hangup()
>> > }
>> >
>> > -- [xml dialplan for extension 8887]:
>> > {
>> > ? ? ? ?<extension name="Simple Lua Test">
>> > ? ? ? ? ? ? ? ?<condition
>> field="destination_number" expression="^(8887)$">
>> > ? ? ? ? ? ? ? ? ? ? ? ?<action
>> application="set" data="record_waste_resources=true"/>
>> > ? ? ? ? ? ? ? ? ? ? ? ?<action
>> application="lua" data="test1.lua"/>
>> > ? ? ? ? ? ? ? ?</condition>
>> > ? ? ? ?</extension>
>> > }
>> >
>> >
>> ======================================================
>> >
>> > For all the above testing cases, session:recordFile(-)
>> always creates an empty wav file for each of outbound IVR
>> calls, however, if I make an inbound IVR?call to run Lua
>> script "test1.lua", session:recordFile(-) always works
>> perfect to generate a normal wav file.
>> >
>> > So, what's wrong with [session:recordFile(-)] during
>> an outbound IVR call?
>> >
>> > x.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
>> >
>>
>>
>>
>> --
>> 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 xing2kin at yahoo.com  Thu Sep  1 10:38:25 2011
From: xing2kin at yahoo.com (king2kin)
Date: Wed, 31 Aug 2011 23:38:25 -0700 (PDT)
Subject: [Freeswitch-users] session:recordFile(-) always creates empty
	wav file during outbound IVR call
In-Reply-To: <CAKbxfG_Bi1P1Ykbf2v2XyLdSdFgugz+LDqrejzQvbLOXmF6itg@mail.gmail.com>
Message-ID: <1314859105.73725.YahooMailClassic@web39707.mail.mud.yahoo.com>

Anthony,

Thank you again for helping me!

> Does the empty file contain silence
> that corresponds to the duration
> of the time it's recording?

No! It's always an empty wav file with its file size 68 bytes, even don't contain any silence samples; 

When I open the wav file with audio tool (e.g. wavesurfer.exe or windows media player), I don't see or hear it contains any single sample of audio data. Probably it contains only the wave header structure of ms wav file format because its size is always 68 bytes by checking its file properties on windows system. By the way, I tried two versions of FreeSwitch (1.0-head-2011-05-23 and 1.0-head-2011-08-31) on windows server 2003. 

> Are you producing the audio yourself into the recording and

Yes, I produced audio myself each time by talking to microphone via X-Lite.
Again, there is never a problem in session:recordFile(-) during any inbound IVR call which may even run the same Lua script as the outbound IVR call.

> can you
> verify with a pcap that there is actually any audio to
> record?
> 
sorry, I have no idea on 'pcap' and don't know to how to make use of it for this debug.

I basically think there is probably a BUG in session:recordFile(-) for outbound IVR call.

Could you please do me a favour trying the following simple outbound IVR call on your site to see if session:recordFile(-) really has a BUG for outbound IVR call?

originate /sofia/gateway/mygateway/xxxx &lua('test.lua')
or
originate user/1005  &lua('test.lua')

where "test.lua" is defined as follows:

{
-- test.lua

-- Set the path separator
pathsep = '/'

-- Windows users do this instead:
-- pathsep = '\'

--Answer the call
session:answer()

--Create a string with path and filename of a sound file
prompt = "ivr" .. pathsep .. "ivr-welcome_to_freeswitch.wav"

-- Print a log message
freeswitch.consoleLog("INFO","Prompt file is '" .. prompt .. "'\n")

--Play the prompt
session:streamFile(prompt)

-- Record record file
session:streamFile("phrase:voicemail_record_message")
			
-- Play a ""bong"" tone prior to recording
session:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000, 0, 640)")

filename = session:getVariable('sounds_dir') .. pathsep .. "123.wav"
session:recordFile(filename,300,100,10)
session:streamFile(filename)

goodbye = "voicemail" .. pathsep .. "vm-goodbye.wav"
session:sleep(250)
session:streamFile(goodbye)

session:hangup()
}


--- On Wed, 8/31/11, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] session:recordFile(-) always creates empty wav file during outbound IVR call
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Wednesday, August 31, 2011, 7:46 PM

> Does the empty file contain silence
> that corresponds to the duration
> of the time it's recording?
> Are you producing the audio yourself into the recording and
> can you
> verify with a pcap that there is actually any audio to
> record?
> 
> 
> On Wed, Aug 31, 2011 at 9:21 PM, king2kin <xing2kin at yahoo.com>
> wrote:
> > Anthony,
> >
> > Thank you for help. I tried outbound IVR call in
> multiple ways again based on your advice,
> ?session:recordFile(-) still doesn't work normally, still
> created empty wav file .
> >
> > Could anyone please give me a hand on FreeSwitch
> Record? It always fails to record audio file during any
> outbound IVR call (auto dialer) although it works well
> during any inbound ivr call.
> >
> > Session:streamFile(-) works well to play back prompt
> files, dtmf keypress also works, ... during outbound ivr
> call.
> >
> > x.k.
> >
> > --- On Wed, 8/31/11, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
> >
> >> From: Anthony Minessale <anthony.minessale at gmail.com>
> >> Subject: Re: [Freeswitch-users]
> session:recordFile(-) always creates empty wav file during
> outbound IVR call
> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> >> Date: Wednesday, August 31, 2011, 12:17 PM
> >> try this dial string instead
> >>
> >>
> {ignore_early_media=true}sofia/gateway/sip.tpad.com/1726011
> >>
> >> On Wed, Aug 31, 2011 at 12:52 PM, king2kin <xing2kin at yahoo.com>
> >> wrote:
> >> > Hi folks,
> >> >
> >> > With Lua script and/or originate command, I
> have tried
> >> recording a message file during outbound IVR call
> over and
> >> over, session:recordFile(-) inside Lua script does
> create a
> >> wav file during each of my testings but the
> recorded audio
> >> file is always empty.
> >> >
> >> > However, session:recordFile(-) works well for
> inbound
> >> IVR call.
> >> >
> >> > I tried the session:recordFile(-) via Lua
> script in
> >> three ways:
> >> >
> >> > 1. run lua script "test_outcall_ivr.lua" at
> freeswitch
> >> command-line:
> >> >
> >> > luarun test_outcall_ivr.lua
> >> >
> >> >
> >> > -- [test_outcall_ivr.lua]
> >> > {
> >> > local sessionx =
> >>
> freeswitch.Session("sofia/gateway/sip.tpad.com/1726011",
> >> session)
> >> >
> >> > -- Set the path separator
> >> > pathsep = '/'
> >> >
> >> > -- Windows users do this instead:
> >> > -- pathsep = '\'
> >> >
> >> > -- Answer the call
> >> > -- sessionx:answer()
> >> >
> >> > --Create a string with path and filename of a
> sound
> >> file
> >> > prompt = "ivr" .. pathsep ..
> >> "ivr-welcome_to_freeswitch.wav"
> >> >
> >> > -- Print a log message
> >> > freeswitch.consoleLog("INFO","Prompt file is
> '" ..
> >> prompt .. "'\n")
> >> >
> >> > --Play the prompt
> >> > sessionx:streamFile(prompt)
> >> >
> >> > -- Record record file
> >> >
> >>
> sessionx:streamFile("phrase:voicemail_record_message")
> >> >
> >> > -- Play a ""bong"" tone prior to recording
> >> >
> >>
> sessionx:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
> >> 0, 640)")
> >> >
> >> > -- record a message
> >> > filename = sessionx:getVariable('sounds_dir')
> ..
> >> pathsep .. "123.wav"
> >> > sessionx:recordFile(filename,300,100,10)
> >> >
> >> > -- play back the recorded msg
> >> > sessionx:streamFile(filename)
> >> >
> >> > -- Hangup
> >> > sessionx:hangup()
> >> >
> >> > }
> >> >
> >> > 2. I also tried it differently by submitting
> the
> >> following commands at the FreeSwitch command-line
> >> interface:
> >> >
> >> > originate user/1005 &transfer(8887 xml
> default)
> >> >
> >> > originate user/1005 &lua('test1.lua')
> >> >
> >> > originate sofia/gateway/sip.tpad.com/1726011
> >> &lua('test1.lua')
> >> >
> >> >
> >> > -- [test1.lua]
> >> > {
> >> > -- Set the path separator
> >> > pathsep = '/'
> >> >
> >> > -- Windows users do this instead:
> >> > -- pathsep = '\'
> >> >
> >> > --Answer the call
> >> > session:answer()
> >> >
> >> > --Create a string with path and filename of a
> sound
> >> file
> >> > prompt = "ivr" .. pathsep ..
> >> "ivr-welcome_to_freeswitch.wav"
> >> >
> >> > -- Print a log message
> >> > freeswitch.consoleLog("INFO","Prompt file is
> '" ..
> >> prompt .. "'\n")
> >> >
> >> > --Play the prompt
> >> > session:streamFile(prompt)
> >> >
> >> > -- Record record file
> >> >
> session:streamFile("phrase:voicemail_record_message")
> >> >
> >> > -- Play a ""bong"" tone prior to recording
> >> >
> >>
> session:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
> >> 0, 640)")
> >> >
> >> > -- record a message
> >> > filename = session:getVariable('sounds_dir')
> ..
> >> pathsep .. "123.wav"
> >> > session:recordFile(filename,300,100,10)
> >> >
> >> > -- play back the recorded msg
> >> > session:streamFile(filename)
> >> >
> >> > -- Hangup
> >> > session:hangup()
> >> > }
> >> >
> >> > -- [xml dialplan for extension 8887]:
> >> > {
> >> > ? ? ? ?<extension name="Simple Lua
> Test">
> >> > ? ? ? ? ? ? ? ?<condition
> >> field="destination_number"
> expression="^(8887)$">
> >> > ? ? ? ? ? ? ? ? ? ? ?
> ?<action
> >> application="set"
> data="record_waste_resources=true"/>
> >> > ? ? ? ? ? ? ? ? ? ? ?
> ?<action
> >> application="lua" data="test1.lua"/>
> >> > ? ? ? ? ? ? ? ?</condition>
> >> > ? ? ? ?</extension>
> >> > }
> >> >
> >> >
> >>
> ======================================================
> >> >
> >> > For all the above testing cases,
> session:recordFile(-)
> >> always creates an empty wav file for each of
> outbound IVR
> >> calls, however, if I make an inbound IVR?call to
> run Lua
> >> script "test1.lua", session:recordFile(-) always
> works
> >> perfect to generate a normal wav file.
> >> >
> >> > So, what's wrong with [session:recordFile(-)]
> during
> >> an outbound IVR call?
> >> >
> >> > x.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
> >> >
> >>
> >>
> >>
> >> --
> >> 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
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From ales.zelenik at it-tim.si  Thu Sep  1 11:27:36 2011
From: ales.zelenik at it-tim.si (=?UTF-8?Q?Ale=C5=A1?= Zelenik)
Date: Thu, 1 Sep 2011 09:27:36 +0200
Subject: [Freeswitch-users] Start FreeSWITCH as a Windows service	with
 -nonat option
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0D8@cooper>
References: <1314783946.2217.12.camel@ales-Latitude-D630>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0D8@cooper>
Message-ID: <1314862056.4855.3.camel@ales-Latitude-D630>

It works, thank you for quick reply


On sre, 2011-08-31 at 18:09 +0200, Peter Olsson wrote:
> Change the command line for the service, from "-service FreeSWITCH" to "-service FreeSWITCH -nonat" - I believe that the extra options must come after the service argument.
> 
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Ale? Zelenik [ales.zelenik at it-tim.si]
> Skickat: den 31 augusti 2011 11:45
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] Start FreeSWITCH as a Windows service with     -nonat option
> 
> Hello all,
> 
> I am having trouble starting fs as a windows service with -nonat option
> 
> If I run from command prompt "FreeSwitchConsole -nonat" it works ok, but
> I am left with open window in logged-in session.
> 
> Command to run as a service is "...path...\FreeSwitchConsole.exe
> -service FreeSWITCH", which can be also verified with "sc qc FreeSWITCH"
> 
> If I want to add an argument -nonat with command:
> sc config FreeSWITCH binPath= "C:\Program Files (x86)\FreeSWITCH
> \FreeSwitchConsole.exe -nonat -service FreeSWITCH"
> 
> service won't start anymore
> 
> also, fs cannot be installed as a service with optional arguments, eg
> FreeSwitchConsole -install -nonat
> 
> Anyone managed to run fs like this?
> 
> Thanks,
> --
> Ales Zelenik,
> 
> 
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:4e5e50df32763458912694!
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ale? Zelenik,

sistemski in?enir:: system engineer

tel: +386 (0) 2 251 34 44
     +386 (0) 59 210 263
sip: ales at voip.it-tim.si
fax: +386 (0) 2 251 34 45
gsm: +386 (0) 40 556 982
eml: ales.zelenik at it-tim.si
www: www.it-tim.si


I.T. tim d.o.o. ::  Ulica heroja ?aranovi?a 37 :: 2000 Maribor
ID zavezanca za DDV: SI33519439
TRR: 03121-1000371323



From davidwaf at gmail.com  Thu Sep  1 12:14:56 2011
From: davidwaf at gmail.com (David Wafula)
Date: Thu, 1 Sep 2011 10:14:56 +0200
Subject: [Freeswitch-users] /usr/src/freeswitch.git/libs/ldns/configure: No
	such file or directory
Message-ID: <CAK9Ws2dMnTPxb7aRgg==h+6Gn39MAnOVR8eSQnaKL80sGu9Uog@mail.gmail.com>

Hi all,
I am updating my freeswitch installation ..after months.

Am unable to get past this...have never encountered it before:

/bin/bash: /usr/src/freeswitch.git/libs/ldns/configure: No such file or
directory
make[5]: *** [/usr/src/freeswitch.git/libs/ldns/Makefile] Error 127
make[4]: *** [install] Error 1
make[3]: *** [mod_enum-install] Error 1
make[2]: *** [install-recursive] Error 1
make[1]: *** [install-recursive] Error 1
make: *** [install] Error 2



Environment:

Ubuntu 10.04  i686, running in Linode.com

Regards.
-- 
David Wafula
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/340d0663/attachment.html 

From neilp at cs.stanford.edu  Thu Sep  1 12:55:33 2011
From: neilp at cs.stanford.edu (Neil Patel)
Date: Thu, 1 Sep 2011 01:55:33 -0700
Subject: [Freeswitch-users] recordFile max_len not obeyed
Message-ID: <CAAcoRq1_NFPDpUgXMpwazN165pO7tKBP+y8X03bBjVi+pyY-CQ@mail.gmail.com>

Hi All,

Based on the documentation<http://wiki.freeswitch.org/wiki/Mod_lua#session:recordFile>,
I am executing the following in a lua script:
session:recordFile(filename, 10000, 100, 5);

I expect that the recording will terminate automatically after 10 seconds.
But it's not, the recording stays engaged long after 10s till I either
terminate with '#' or there is enough trailing silence. What's the problem?

Thanks in advance,
Neil
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/d70cc05a/attachment.html 

From mrene_lists at avgs.ca  Thu Sep  1 13:03:08 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 1 Sep 2011 11:03:08 +0200
Subject: [Freeswitch-users] /usr/src/freeswitch.git/libs/ldns/configure:
	No such file or directory
In-Reply-To: <CAK9Ws2dMnTPxb7aRgg==h+6Gn39MAnOVR8eSQnaKL80sGu9Uog@mail.gmail.com>
References: <CAK9Ws2dMnTPxb7aRgg==h+6Gn39MAnOVR8eSQnaKL80sGu9Uog@mail.gmail.com>
Message-ID: <1A1CEF14-7FBB-495B-8F43-F5D8057F192B@avgs.ca>

You need to re-run bootstrap.sh

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




On 2011-09-01, at 10:14 AM, David Wafula wrote:

> Hi all,
> I am updating my freeswitch installation ..after months. 
> 
> Am unable to get past this...have never encountered it before:
> 
> /bin/bash: /usr/src/freeswitch.git/libs/ldns/configure: No such file or directory
> make[5]: *** [/usr/src/freeswitch.git/libs/ldns/Makefile] Error 127
> make[4]: *** [install] Error 1
> make[3]: *** [mod_enum-install] Error 1
> make[2]: *** [install-recursive] Error 1
> make[1]: *** [install-recursive] Error 1
> make: *** [install] Error 2
> 
> 
> 
> Environment: 
> 
> Ubuntu 10.04  i686, running in Linode.com
> 
> Regards.
> -- 
> David Wafula
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From davidwaf at gmail.com  Thu Sep  1 13:30:01 2011
From: davidwaf at gmail.com (David Wafula)
Date: Thu, 1 Sep 2011 11:30:01 +0200
Subject: [Freeswitch-users] /usr/src/freeswitch.git/libs/ldns/configure:
 No such file or directory
In-Reply-To: <1A1CEF14-7FBB-495B-8F43-F5D8057F192B@avgs.ca>
References: <CAK9Ws2dMnTPxb7aRgg==h+6Gn39MAnOVR8eSQnaKL80sGu9Uog@mail.gmail.com>
	<1A1CEF14-7FBB-495B-8F43-F5D8057F192B@avgs.ca>
Message-ID: <CAK9Ws2fEFukWreep+sAbOVu_U5cTvF0zGXCo1=z-UMXmPu6O0g@mail.gmail.com>

Thanks. That fixed it.
Regards

On Thu, Sep 1, 2011 at 11:03 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> You need to re-run bootstrap.sh
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2011-09-01, at 10:14 AM, David Wafula wrote:
>
> > Hi all,
> > I am updating my freeswitch installation ..after months.
> >
> > Am unable to get past this...have never encountered it before:
> >
> > /bin/bash: /usr/src/freeswitch.git/libs/ldns/configure: No such file or
> directory
> > make[5]: *** [/usr/src/freeswitch.git/libs/ldns/Makefile] Error 127
> > make[4]: *** [install] Error 1
> > make[3]: *** [mod_enum-install] Error 1
> > make[2]: *** [install-recursive] Error 1
> > make[1]: *** [install-recursive] Error 1
> > make: *** [install] Error 2
> >
> >
> >
> > Environment:
> >
> > Ubuntu 10.04  i686, running in Linode.com
> >
> > Regards.
> > --
> > David Wafula
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
David Wafula
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/ad5ff946/attachment-0001.html 

From steveayre at gmail.com  Thu Sep  1 13:47:19 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 1 Sep 2011 10:47:19 +0100
Subject: [Freeswitch-users] Variables in Dialplan - Problem with getting
 Variables from User Directory
In-Reply-To: <4E5E7578.2000803@omeco.de>
References: <4E5D50BD.8010208@omeco.de>
	<A1E3DB7E-64B5-4E74-89FF-7F87E7F8D501@seventhsignal.de>
	<4E5E7578.2000803@omeco.de>
Message-ID: <CAFiqYun2Dx+nUZ5C39HUL9zbJkiE_KhMYLdgcOEM1Q9G+jy3vQ@mail.gmail.com>

>
> the question is "why" are the vars and params unavailable without set_user
> or user_data thingies
>

If set_user works ok, then I guess you're probably not authenticating as
that user without running that app. Perhaps some config file was changed
during the update?

-Steve



On 31 August 2011 18:55, Silvio Escher <sescher_ml at omeco.de> wrote:

> Am 31.08.11 15:20, schrieb Michal Bielicki:
> >
> > Am 30.08.2011 um 23:06 schrieb Silvio Escher:
> >
> >
> > if you set the user with set_user there is no requirement to use
> user_data since it should get the
> > params for the respective user automatically.
> >
> yes - i already got this - i just want to show 2 ways to solve my issue
> partially ( maybe helpfull
> for others after google indexing ;) )
> >> params are checked on call, so only there when required and checked each
> time, vars are fixed.
> this wont help me much at this point - the question is "why" are the vars
> and params unavailable
> without set_user or user_data thingies ..
> or at this special point - how to get the params available during the
> voicemail event
>
> Best Regards,
> Silvio
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/fe033810/attachment.html 

From xing2kin at yahoo.com  Thu Sep  1 13:48:12 2011
From: xing2kin at yahoo.com (king2kin)
Date: Thu, 1 Sep 2011 02:48:12 -0700 (PDT)
Subject: [Freeswitch-users] session:recordFile(-) always creates empty
	wav file during outbound IVR call
In-Reply-To: <CAKbxfG_Bi1P1Ykbf2v2XyLdSdFgugz+LDqrejzQvbLOXmF6itg@mail.gmail.com>
Message-ID: <1314870492.70429.YahooMailClassic@web39708.mail.mud.yahoo.com>

Hi Anthony,

Actually FreeSwitch application "record" doesn't work either for outbound IVR call (see below), created an empty wav file with its size 68 bytes, the wav file doesn't contain any samples of audio data; 

<action application="record" data="C:/c4dev/freeswitch/Debug/sounds/test.wav 20 200"/>

earlier I reported that "session:recordFile(-)" doesn't work inside Lua Script for any outbound IVR call.

- My CLI commands to make an outbound IVR call:

originate {ignore_early_media=true}sofia/gateway/mygateway/1726011  8884
or
originate user/1005   8884

- xml diaplan for extension 8884:
{
	<extension name="Play and Record Sound File">
		<condition field="destination_number" expression="^(8884)$">
			<action application="answer"/>
			<action application="sleep" data="1000"/>
			<action application="playback" data="ivr/ivr-thank_you.wav"/>	
			
			<action application="record" data="C:/c4dev/freeswitch/Debug/sounds/test.wav 20 200"/>
			
			<action application="sleep" data="1000"/>
			<action application="playback" data="voicemail/vm-goodbye.wav"/>	
			
			<action application="sleep" data="1000"/>		
			<action application="hangup"/>			
		</condition>
	</extension>
}

--- On Wed, 8/31/11, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] session:recordFile(-) always creates empty wav file during outbound IVR call
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Wednesday, August 31, 2011, 7:46 PM
> Does the empty file contain silence
> that corresponds to the duration
> of the time it's recording?
> Are you producing the audio yourself into the recording and
> can you
> verify with a pcap that there is actually any audio to
> record?
> 
> 
> On Wed, Aug 31, 2011 at 9:21 PM, king2kin <xing2kin at yahoo.com>
> wrote:
> > Anthony,
> >
> > Thank you for help. I tried outbound IVR call in
> multiple ways again based on your advice,
> ?session:recordFile(-) still doesn't work normally, still
> created empty wav file .
> >
> > Could anyone please give me a hand on FreeSwitch
> Record? It always fails to record audio file during any
> outbound IVR call (auto dialer) although it works well
> during any inbound ivr call.
> >
> > Session:streamFile(-) works well to play back prompt
> files, dtmf keypress also works, ... during outbound ivr
> call.
> >
> > x.k.
> >
> > --- On Wed, 8/31/11, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
> >
> >> From: Anthony Minessale <anthony.minessale at gmail.com>
> >> Subject: Re: [Freeswitch-users]
> session:recordFile(-) always creates empty wav file during
> outbound IVR call
> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> >> Date: Wednesday, August 31, 2011, 12:17 PM
> >> try this dial string instead
> >>
> >>
> {ignore_early_media=true}sofia/gateway/sip.tpad.com/1726011
> >>
> >> On Wed, Aug 31, 2011 at 12:52 PM, king2kin <xing2kin at yahoo.com>
> >> wrote:
> >> > Hi folks,
> >> >
> >> > With Lua script and/or originate command, I
> have tried
> >> recording a message file during outbound IVR call
> over and
> >> over, session:recordFile(-) inside Lua script does
> create a
> >> wav file during each of my testings but the
> recorded audio
> >> file is always empty.
> >> >
> >> > However, session:recordFile(-) works well for
> inbound
> >> IVR call.
> >> >
> >> > I tried the session:recordFile(-) via Lua
> script in
> >> three ways:
> >> >
> >> > 1. run lua script "test_outcall_ivr.lua" at
> freeswitch
> >> command-line:
> >> >
> >> > luarun test_outcall_ivr.lua
> >> >
> >> >
> >> > -- [test_outcall_ivr.lua]
> >> > {
> >> > local sessionx =
> >>
> freeswitch.Session("sofia/gateway/sip.tpad.com/1726011",
> >> session)
> >> >
> >> > -- Set the path separator
> >> > pathsep = '/'
> >> >
> >> > -- Windows users do this instead:
> >> > -- pathsep = '\'
> >> >
> >> > -- Answer the call
> >> > -- sessionx:answer()
> >> >
> >> > --Create a string with path and filename of a
> sound
> >> file
> >> > prompt = "ivr" .. pathsep ..
> >> "ivr-welcome_to_freeswitch.wav"
> >> >
> >> > -- Print a log message
> >> > freeswitch.consoleLog("INFO","Prompt file is
> '" ..
> >> prompt .. "'\n")
> >> >
> >> > --Play the prompt
> >> > sessionx:streamFile(prompt)
> >> >
> >> > -- Record record file
> >> >
> >>
> sessionx:streamFile("phrase:voicemail_record_message")
> >> >
> >> > -- Play a ""bong"" tone prior to recording
> >> >
> >>
> sessionx:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
> >> 0, 640)")
> >> >
> >> > -- record a message
> >> > filename = sessionx:getVariable('sounds_dir')
> ..
> >> pathsep .. "123.wav"
> >> > sessionx:recordFile(filename,300,100,10)
> >> >
> >> > -- play back the recorded msg
> >> > sessionx:streamFile(filename)
> >> >
> >> > -- Hangup
> >> > sessionx:hangup()
> >> >
> >> > }
> >> >
> >> > 2. I also tried it differently by submitting
> the
> >> following commands at the FreeSwitch command-line
> >> interface:
> >> >
> >> > originate user/1005 &transfer(8887 xml
> default)
> >> >
> >> > originate user/1005 &lua('test1.lua')
> >> >
> >> > originate sofia/gateway/sip.tpad.com/1726011
> >> &lua('test1.lua')
> >> >
> >> >
> >> > -- [test1.lua]
> >> > {
> >> > -- Set the path separator
> >> > pathsep = '/'
> >> >
> >> > -- Windows users do this instead:
> >> > -- pathsep = '\'
> >> >
> >> > --Answer the call
> >> > session:answer()
> >> >
> >> > --Create a string with path and filename of a
> sound
> >> file
> >> > prompt = "ivr" .. pathsep ..
> >> "ivr-welcome_to_freeswitch.wav"
> >> >
> >> > -- Print a log message
> >> > freeswitch.consoleLog("INFO","Prompt file is
> '" ..
> >> prompt .. "'\n")
> >> >
> >> > --Play the prompt
> >> > session:streamFile(prompt)
> >> >
> >> > -- Record record file
> >> >
> session:streamFile("phrase:voicemail_record_message")
> >> >
> >> > -- Play a ""bong"" tone prior to recording
> >> >
> >>
> session:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
> >> 0, 640)")
> >> >
> >> > -- record a message
> >> > filename = session:getVariable('sounds_dir')
> ..
> >> pathsep .. "123.wav"
> >> > session:recordFile(filename,300,100,10)
> >> >
> >> > -- play back the recorded msg
> >> > session:streamFile(filename)
> >> >
> >> > -- Hangup
> >> > session:hangup()
> >> > }
> >> >
> >> > -- [xml dialplan for extension 8887]:
> >> > {
> >> > ? ? ? ?<extension name="Simple Lua
> Test">
> >> > ? ? ? ? ? ? ? ?<condition
> >> field="destination_number"
> expression="^(8887)$">
> >> > ? ? ? ? ? ? ? ? ? ? ?
> ?<action
> >> application="set"
> data="record_waste_resources=true"/>
> >> > ? ? ? ? ? ? ? ? ? ? ?
> ?<action
> >> application="lua" data="test1.lua"/>
> >> > ? ? ? ? ? ? ? ?</condition>
> >> > ? ? ? ?</extension>
> >> > }
> >> >
> >> >
> >>
> ======================================================
> >> >
> >> > For all the above testing cases,
> session:recordFile(-)
> >> always creates an empty wav file for each of
> outbound IVR
> >> calls, however, if I make an inbound IVR?call to
> run Lua
> >> script "test1.lua", session:recordFile(-) always
> works
> >> perfect to generate a normal wav file.
> >> >
> >> > So, what's wrong with [session:recordFile(-)]
> during
> >> an outbound IVR call?
> >> >
> >> > x.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
> >> >
> >>
> >>
> >>
> >> --
> >> 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
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From sescher_ml at omeco.de  Thu Sep  1 15:10:00 2011
From: sescher_ml at omeco.de (Silvio Escher)
Date: Thu, 01 Sep 2011 13:10:00 +0200
Subject: [Freeswitch-users] Variables in Dialplan - Problem with getting
 Variables from User Directory
In-Reply-To: <CAFiqYun2Dx+nUZ5C39HUL9zbJkiE_KhMYLdgcOEM1Q9G+jy3vQ@mail.gmail.com>
References: <4E5D50BD.8010208@omeco.de>
	<A1E3DB7E-64B5-4E74-89FF-7F87E7F8D501@seventhsignal.de>
	<4E5E7578.2000803@omeco.de>
	<CAFiqYun2Dx+nUZ5C39HUL9zbJkiE_KhMYLdgcOEM1Q9G+jy3vQ@mail.gmail.com>
Message-ID: <4E5F6808.1070707@omeco.de>

Am 01.09.11 11:47, schrieb Steven Ayre:
>
>     the question is "why" are the vars and params unavailable without set_user or user_data thingies
>
>
> If set_user works ok, then I guess you're probably not authenticating as that user without running 
> that app. Perhaps some config file was changed during the update?
>
> -Steve
>
Regarding the mod_voicemail issue there isn't an authenticated user ( internal ) in place.

The Calls comes from external and joins the public dp ...

<extension name="internDW">
<condition field="${sip_to_user}" expression="^063168035(\d{1,2})$">
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="call_timeout=20"/>
<action application="bridge" data="user/$1@${domain_name}"/>
<action application="answer"/>
<action application="playback" data="niemand_da_ohnezeit.wav"/>
<action application="sleep" data="1000"/>
<action application="set" data="default_language=de"/>
<action application="set" data="skip_greeting=true"/>
<action application="voicemail" data="default $${domain} $1"/>
</condition>
</extension>

As i understand the freeswitch wiki, mod_voicemail should take the Params for $1 ( the dialed 
extension ) somehow from the Directory User.

But an unanswered Call to DID 40 enters the Voicemail for that User correctly ( VM is stored, Phone 
Button/LED for VM Notify works as expected ) - but i get no Notify Message by Mail
because mod_voicemail cant get the "vm-mailto" from my Directoryentry.

2011-09-01 13:03:52.924206 [DEBUG] mod_voicemail.c:2578 Deliver VM to 40 at pbx.omeco.voice
2011-09-01 13:03:52.924206 [DEBUG] switch_utils.c:709 Emailed data to [(null)]
2011-09-01 13:03:52.924206 [DEBUG] mod_voicemail.c:2809 Sending notify message to (null)

Regards,
Silvio


-- 
Silvio Escher
omeco GmbH

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/e7d441fd/attachment.html 

From ocset at the800group.com  Thu Sep  1 13:54:29 2011
From: ocset at the800group.com (ocset)
Date: Thu, 01 Sep 2011 17:54:29 +0800
Subject: [Freeswitch-users] Yealink t28p setup - please help
Message-ID: <4E5F5655.9050705@the800group.com>

Hi All

I am new to FreeSwitch and this is my first post on this forum/mailing-list.

I have a default install of FS on Ubuntu 10.10 and have two Yealink T28P 
phones. I am trying to understand the best way to config these phones 
for FS. What I am fighting with at the moment is that if I assign the 
same extension to both phones, only one phone will ring when the shared 
extension is called.

The phones have the option to assign 6 SIP accounts/extensions and this 
is my current test config.

Phone 1
ext 1001
ext 1010
ext 1011

Phone 2
ext 1002
ext 1010
ext 1011

My thinking behind this is that I want to assign a POST line to the 1010 
extension (using an SPA3102)  and a VOIP line to the 1011 extension. 
That way, when an external call comes in on one of those lines, all 
parties have the option to pick up the call. What I have found however 
is that only one of the phones rings and is able to answer the 1010 and 
1011 extensions. Is this expected behaviour? Should I be doing this 
another way?

I am very new to VOIP, Freeswitch etc so my terminology may be wrong - 
hopefully you understand my explanation.

Thanks in advance


From qiaoqiao7036 at gmail.com  Thu Sep  1 05:24:14 2011
From: qiaoqiao7036 at gmail.com (qiaoqiao7036)
Date: Thu, 1 Sep 2011 09:24:14 +0800
Subject: [Freeswitch-users] freeswitch-1.0.7 Conference set auto outcall bug?
Message-ID: <007c01cc6845$e0d3c9b0$a27b5d10$@gmail.com>

Hi,

  I have used the function of conference set auto outcall in freeswitch
1.0.6, it is very good,

several days ago, I installed a new freeswitch with git, the version is
1.0.7, I wrote a very simple 

dialplan to test the conference auto outcall function, and the problem
happened:

 

This is my dialplan, I have tested it in my freeswitch 1.0.6 successfully:

 

   <extension name="Demonstrate conference_set_auto_outcall">

     <condition field="destination_number" expression="^(12345)$">

 

       <action application="answer"/>

 

       <action application="set" data="conference_auto_outcall_timeout=5"/>

       <action application="set" data="conference_auto_outcall_flags=none"/>

       <action application="set"
data="conference_auto_outcall_caller_id_name=$${ caller_id_name}"/>

       <action application="set"
data="conference_auto_outcall_caller_id_number=$${ caller_id_number}"/>

       <action application="set"
data="conference_auto_outcall_profile=default"/>

 

       <action application="conference_set_auto_outcall"
data="[call_timeout=15]user/1001@$${domain}"/>

       <action application="conference_set_auto_outcall"
data="[call_timeout=15]user/1002@$${domain}"/>

 

       <action application="conference" data="$1 at default"/>

 

     </condition>

   </extension>

 

Problem 1:  When I use a softphone to dial number 12345, in my old
freeswitch I can hear the moh music when the phone 1001 and 1002 is ring,

But in freeswitch 1.0.7 I can't hear anything, and I found the debug
informations in freeswitch cli:

 

2011-08-31 18:59:14.520108 [DEBUG] switch_core_session.c:1156 Send signal
sofia/internal/sip:1001 at 192.168.2.226:5060 [BREAK]

2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:369
(sofia/internal/sip:1001 at 192.168.2.226:5060) State INIT going to sleep

2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/sip:1001 at 192.168.2.226:5060) Running State Change CS_ROUTING

2011-08-31 18:59:14.520108 [DEBUG] switch_channel.c:1828
(sofia/internal/sip:1001 at 192.168.2.226:5060) Callstate Change DOWN ->
RINGING

2011-08-31 18:59:14.520108 [DEBUG] switch_core_session.c:857 Send signal
sofia/internal/sip:1001 at 192.168.2.226:5060 [BREAK]

2011-08-31 18:59:14.520108 [DEBUG] sofia.c:5128 Channel
sofia/internal/sip:1001 at 192.168.2.226:5060 entering state [calling][0]

2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:378
(sofia/internal/sip:1001 at 192.168.2.226:5060) State ROUTING

2011-08-31 18:59:14.520108 [DEBUG] mod_sofia.c:148
sofia/internal/sip:1001 at 192.168.2.226:5060 SOFIA ROUTING

2011-08-31 18:59:14.520108 [DEBUG] switch_ivr_originate.c:66
(sofia/internal/sip:1001 at 192.168.2.226:5060) State Change CS_ROUTING ->
CS_CONSUME_MEDIA

2011-08-31 18:59:14.520108 [DEBUG] switch_core_session.c:1156 Send signal
sofia/internal/sip:1001 at 192.168.2.226:5060 [BREAK]

2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:378
(sofia/internal/sip:1001 at 192.168.2.226:5060) State ROUTING going to sleep

2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/sip:1001 at 192.168.2.226:5060) Running State Change
CS_CONSUME_MEDIA

2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/sip:1001 at 192.168.2.226:5060) State CONSUME_MEDIA

2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/sip:1001 at 192.168.2.226:5060) State CONSUME_MEDIA going to
sleep

2011-08-31 18:59:14.540080 [DEBUG] mod_local_stream.c:421 Opening Stream
[moh/8000] 8000hz

2011-08-31 18:59:14.540080 [DEBUG] switch_core_session.c:857 Send signal
sofia/internal/1000 at 192.168.2.242 [BREAK]

2011-08-31 18:59:14.540080 [DEBUG] switch_core_session.c:857 Send signal
sofia/internal/1000 at 192.168.2.242 [BREAK]

2011-08-31 18:59:14.540080 [DEBUG] switch_core_session.c:857 Send signal
sofia/internal/1000 at 192.168.2.242 [BREAK]

2011-08-31 18:59:14.559978 [DEBUG] sofia.c:5128 Channel
sofia/internal/1000 at 192.168.2.242 entering state [ready][200]

 

       That mean the freeswitch is sending the moh music to the caller 1000,
is that right? But I can't hear anything.

 

       Problem 2: when just one phone pickup, like phone 1001, and phone
1002 don't, (I use phone 1000 to call number 12345) I can't hear anything
from phone 1001 when 1002 is ringing except phone 1002 pick up or when the
phone 1002 is timeout to ring, it's very stranger!!! I tested this situation
in freeswitch 1.0.6, when just only person pickup the phone the conference
will work, 1000 will hear the phone 1001's voice, don't need the 1002 to
pickup or timeout.

 

By the way: I have tested this with the newest version: FreeSWITCH Version
1.0.head (git-6a5f6e5 2011-08-30 15-00-07 -0500)

The problem is still exist.

Thank you for advising me how to solve this problem.

 

 

Regards!

Dennis

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/86547f0d/attachment-0001.html 

From tomasz at hyziak.pl  Thu Sep  1 13:03:13 2011
From: tomasz at hyziak.pl (Tomasz Hyziak)
Date: Thu, 1 Sep 2011 11:03:13 +0200
Subject: [Freeswitch-users] FreeSwitch and core dumps
Message-ID: <CAFv7ub1ciPKA9ceWcJrUx3fQROXrAxsdDyc0Q355gRB-ZtU52g@mail.gmail.com>

Hi

I've got a problem with FS core dump - http://pastebin.freeswitch.org/17255

When FS shuts down with a core dump it restarts but all calls are
disconnected after 30 seconds. Only restart helps for some time.

In CDR i see at these calls hangup_cause NORMAL_UNSPECIFIED and
billsec 33 or 34 seconds.

I cannot find anything interesting in logs... Here it is:

b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[NOTICE] switch_channel.c:907 New Channel
sofia/external/501XXXXXX at trunk.dialog.pl
[b541f405-1a39-4cbc-80dc-8bdbc92ed070]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] sofia.c:5142 Channel sofia/external/501XXXXXX at trunk.dialog.pl
entering state [received][100]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_core_state_machine.c:330
(sofia/external/501XXXXXX at trunk.dialog.pl) Running State Change CS_NEW
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] sofia.c:5153 Remote SDP:
v=0
o=BroadWorks 51808039 1 IN IP4 10.165.64.36
s=-
c=IN IP4 10.165.64.36
t=0 0
m=audio 22558 RTP/AVP 0 8 101
b=AS:82
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_core_state_machine.c:348
(sofia/external/501XXXXXX at trunk.dialog.pl) State NEW
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] sofia_glue.c:4731 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] sofia_glue.c:4731 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] sofia_glue.c:2839 Set Codec
sofia/external/501XXXXXX at trunk.dialog.pl PCMU/8000 20 ms 160 samples
64000 bits
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] sofia_glue.c:4845 Set 2833 dtmf send/recv payload to 101
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] sofia.c:5343 (sofia/external/501XXXXXX at trunk.dialog.pl) State
Change CS_NEW -> CS_INIT
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_core_session.c:1156 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_core_state_machine.c:330
(sofia/external/501XXXXXX at trunk.dialog.pl) Running State Change
CS_INIT
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_core_state_machine.c:369
(sofia/external/501XXXXXX at trunk.dialog.pl) State INIT
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] mod_sofia.c:85 sofia/external/501XXXXXX at trunk.dialog.pl SOFIA
INIT
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] mod_sofia.c:125 (sofia/external/501XXXXXX at trunk.dialog.pl)
State Change CS_INIT -> CS_ROUTING
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_core_session.c:1156 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_core_state_machine.c:369
(sofia/external/501XXXXXX at trunk.dialog.pl) State INIT going to sleep
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_core_state_machine.c:330
(sofia/external/501XXXXXX at trunk.dialog.pl) Running State Change
CS_ROUTING
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_channel.c:1836
(sofia/external/501XXXXXX at trunk.dialog.pl) Callstate Change DOWN ->
RINGING
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_core_state_machine.c:378
(sofia/external/501XXXXXX at trunk.dialog.pl) State ROUTING
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] mod_sofia.c:148 sofia/external/501XXXXXX at trunk.dialog.pl SOFIA
ROUTING
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_core_state_machine.c:78
sofia/external/501XXXXXX at trunk.dialog.pl Standard ROUTING
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283 [INFO]
mod_dialplan_xml.c:336 Processing 501XXXXXX <501XXXXXX>->71XXXXXXX in
context public
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl parsing [public->unloop]
continue=false
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl parsing
[public->outside_call] continue=true
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Absolute Condition
[outside_call]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action set(outside_call=true)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl parsing [public->call_debug]
continue=true
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Regex (FAIL) [call_debug]
${call_debug}(false) =~ /^true$/ break=never
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl parsing
[public->public_extensions] continue=false
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Regex (FAIL)
[public_extensions] destination_number(71XXXXXXX) =~
/^(1[0-9][0-9][0-9])$/ break=on-false
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl parsing [public->in_dialog_1]
continue=false
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Date/Time Match (FAIL)
[in_dialog_1] break=on-true
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Regex (PASS) [in_dialog_1]
destination_number(71XXXXXXX) =~ /^(717739100|71XXXXXXX|717854012)$/
break=on-false
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(RECORD_STEREO=true)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
record_session(/srv/nagrania/aktualne/IN_${strftime(%Y%m%d-%H%M%S)}_${destination_number}_${caller_id_number}.wav)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(media_bug_answer_req=true)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(domain_name=192.168.0.8)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
export(absolute_codec_string=PCMU,PCMA)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
export(codec_string=PCMU,PCMA)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(jitterbuffer_msec=180)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action set_audio_level(read
1)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(hangup_after_bridge=true)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action set(bind_meta_key=9)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action bind_meta_app(1 b s
execute_extension::dx XML features)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action bind_meta_app(2 b s
execute_extension::cf XML features)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action bind_meta_app(3 b s
execute_extension::att_xfer XML features)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action set(call_timeout=60)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(continue_on_fail=NO_ANSWER)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(limit_ignore_transfer=true)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action answer()
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action sleep(1000)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(fifo_music=/srv/nagrania/ivr/ponce-preludio-in-e-major-10s.wav)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action ivr(ivr)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action fifo(fifo_ogolne in)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action hangup()
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_core_state_machine.c:122
(sofia/external/501XXXXXX at trunk.dialog.pl) State Change CS_ROUTING ->
CS_EXECUTE
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_core_session.c:1156 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_core_state_machine.c:378
(sofia/external/501XXXXXX at trunk.dialog.pl) State ROUTING going to
sleep
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_core_state_machine.c:330
(sofia/external/501XXXXXX at trunk.dialog.pl) Running State Change
CS_EXECUTE
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_core_state_machine.c:385
(sofia/external/501XXXXXX at trunk.dialog.pl) State EXECUTE
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] mod_sofia.c:241 sofia/external/501XXXXXX at trunk.dialog.pl SOFIA
EXECUTE
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_core_state_machine.c:160
sofia/external/501XXXXXX at trunk.dialog.pl Standard EXECUTE
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set(outside_call=true)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [outside_call]=[true]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set(RFC2822_DATE=Thu, 01 Sep
2011 08:36:08 +0200)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [RFC2822_DATE]=[Thu, 01 Sep 2011 08:36:08 +0200]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set(RECORD_STEREO=true)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [RECORD_STEREO]=[true]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.103283
[DEBUG] switch_core_session.c:1983 Application record_session Requires
media! pre_answering channel sofia/external/501XXXXXX at trunk.dialog.pl
2011-09-01 08:36:08.103283 [INFO] switch_core_session.c:1985 Sending early media
2011-09-01 08:36:08.143282 [DEBUG] switch_nat.c:510 mapped public port
29830 protocol UDP to localport 29830
2011-09-01 08:36:08.163284 [DEBUG] switch_nat.c:510 mapped public port
29831 protocol UDP to localport 29831
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.163284
[DEBUG] sofia_glue.c:3091 AUDIO RTP
[sofia/external/501XXXXXX at trunk.dialog.pl] 172.20.0.22 port 29830 ->
10.165.64.36 port 22558 codec: 0 ms: 20
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.163284
[DEBUG] switch_rtp.c:1637 Starting timer [soft] 160 bytes per 20ms
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.163284
[DEBUG] sofia_glue.c:3353 Set 2833 dtmf send payload to 101
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.163284
[DEBUG] sofia_glue.c:3358 Set 2833 dtmf receive payload to 101
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.163284
[DEBUG] mod_sofia.c:2436 Ring SDP:
v=0
o=FreeSWITCH 1314829138 1314829139 IN IP4 10.64.64.64
s=FreeSWITCH
c=IN IP4 10.64.64.64
t=0 0
m=audio 29830 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

b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.163284
[NOTICE] mod_sofia.c:2439 Pre-Answer
sofia/external/501XXXXXX at trunk.dialog.pl!
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.163284
[DEBUG] switch_channel.c:2851
(sofia/external/501XXXXXX at trunk.dialog.pl) Callstate Change RINGING ->
EARLY
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] switch_core_session.c:713 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl
record_session(/srv/nagrania/aktualne/IN_20110901-083608_71XXXXXXX_501XXXXXX.wav)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] switch_core_session.c:857 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] sofia.c:5135 Channel sofia/external/501XXXXXX at trunk.dialog.pl
skipping state [early][183]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] switch_core_media_bug.c:360 Attaching BUG to
sofia/external/501XXXXXX at trunk.dialog.pl
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl
set(media_bug_answer_req=true)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [media_bug_answer_req]=[true]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set(domain_name=192.168.0.8)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [domain_name]=[192.168.0.8]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl
export(absolute_codec_string=PCMU,PCMA)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] switch_channel.c:1074 EXPORT (export_vars)
[absolute_codec_string]=[PCMU,PCMA]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl
export(codec_string=PCMU,PCMA)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] switch_channel.c:1074 EXPORT (export_vars)
[codec_string]=[PCMU,PCMA]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set(jitterbuffer_msec=180)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [jitterbuffer_msec]=[180]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set_audio_level(read 1)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] switch_core_media_bug.c:360 Attaching BUG to
sofia/external/501XXXXXX at trunk.dialog.pl
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set(hangup_after_bridge=true)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [hangup_after_bridge]=[true]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set(bind_meta_key=9)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [bind_meta_key]=[9]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl bind_meta_app(1 b s
execute_extension::dx XML features)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289 [INFO]
switch_ivr_async.c:3069 Bound B-Leg: 91 execute_extension::dx XML
features
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl bind_meta_app(2 b s
execute_extension::cf XML features)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289 [INFO]
switch_ivr_async.c:3069 Bound B-Leg: 92 execute_extension::cf XML
features
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl bind_meta_app(3 b s
execute_extension::att_xfer XML features)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289 [INFO]
switch_ivr_async.c:3069 Bound B-Leg: 93 execute_extension::att_xfer
XML features
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set(call_timeout=60)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [call_timeout]=[60]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl
set(continue_on_fail=NO_ANSWER)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [continue_on_fail]=[NO_ANSWER]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl
set(limit_ignore_transfer=true)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [limit_ignore_transfer]=[true]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl answer()
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] mod_sofia.c:738 Local SDP
sofia/external/501XXXXXX at trunk.dialog.pl:
v=0
o=FreeSWITCH 1314829138 1314829140 IN IP4 10.64.64.64
s=FreeSWITCH
c=IN IP4 10.64.64.64
t=0 0
m=audio 29830 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

b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] switch_core_session.c:713 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] switch_channel.c:3043
(sofia/external/501XXXXXX at trunk.dialog.pl) Callstate Change EARLY ->
ACTIVE
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] switch_core_session.c:857 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[NOTICE] mod_dptools.c:930 Channel
[sofia/external/501XXXXXX at trunk.dialog.pl] has been answered
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:08.183289
[DEBUG] sofia.c:5142 Channel sofia/external/501XXXXXX at trunk.dialog.pl
entering state [completed][200]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl sleep(1000)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl
set(fifo_music=/srv/nagrania/ivr/ponce-preludio-in-e-major-10s.wav)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:09.203286
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [fifo_music]=[/srv/nagrania/ivr/ponce-preludio-in-e-major-10s.wav]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl ivr(ivr)
2011-09-01 08:36:09.203286 [DEBUG] switch_ivr_menu.c:660
switch_ivr_menu_stack_xml_add binding 'menu-exit'
2011-09-01 08:36:09.203286 [DEBUG] switch_ivr_menu.c:660
switch_ivr_menu_stack_xml_add binding 'menu-sub'
2011-09-01 08:36:09.203286 [DEBUG] switch_ivr_menu.c:660
switch_ivr_menu_stack_xml_add binding 'menu-exec-app'
2011-09-01 08:36:09.203286 [DEBUG] switch_ivr_menu.c:660
switch_ivr_menu_stack_xml_add binding 'menu-play-sound'
2011-09-01 08:36:09.203286 [DEBUG] switch_ivr_menu.c:660
switch_ivr_menu_stack_xml_add binding 'menu-back'
2011-09-01 08:36:09.203286 [DEBUG] switch_ivr_menu.c:660
switch_ivr_menu_stack_xml_add binding 'menu-top'
2011-09-01 08:36:09.203286 [DEBUG] switch_ivr_menu.c:791 building menu 'ivr'
2011-09-01 08:36:09.203286 [DEBUG] switch_ivr_menu.c:852 binding menu
action 'menu-exec-app' to '1'
2011-09-01 08:36:09.203286 [DEBUG] switch_ivr_menu.c:852 binding menu
action 'menu-exec-app' to '2'
2011-09-01 08:36:09.203286 [DEBUG] switch_ivr_menu.c:852 binding menu
action 'menu-top' to '9'
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:09.203286
[DEBUG] switch_ivr_menu.c:428 Executing IVR menu ivr
2011-09-01 08:36:09.203286 [DEBUG] switch_core_file.c:180 File
/srv/nagrania/ivr/powitanie_pelne.wav sample rate 22050 doesn't match
requested rate 8000
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:09.203286
[DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1
channels 20ms
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:09.203286
[DEBUG] switch_ivr_play_say.c:1351 Setup timer success 320 bytes per
20 ms!
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:26.823286
[DEBUG] switch_ivr_play_say.c:1672 done playing file
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:26.823286
[DEBUG] switch_ivr_menu.c:343 waiting for 1/1 digits t/o 2000
2011-09-01 08:36:30.443283 [DEBUG] switch_nat.c:306 got UPnP keep alive packet:
NOTIFY * HTTP/1.1
HOST: 239.255.255.250:1900
CACHE-CONTROL: max-age=180
Location: http://192.168.0.6:5431/dyndev/uuid:001c1091-9dfb-001c-1091-9dfb00005800
NT: urn:schemas-upnp-org:service:WANPPPConnection:1
NTS: ssdp:alive
SERVER:LINUX/2.4 UPnP/1.0 BRCM400/1.0
USN: uuid:001c1091-9dfb-001c-1091-9dfb02005800::urn:schemas-upnp-org:service:WANPPPConnection:1


b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:31.843284
[DEBUG] switch_ivr_menu.c:390 digits ''
2011-09-01 08:36:31.843284 [DEBUG] switch_core_file.c:180 File
/srv/nagrania/ivr/powitanie_skrocone.wav sample rate 22050 doesn't
match requested rate 8000
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:31.843284
[DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1
channels 20ms
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:31.843284
[DEBUG] switch_ivr_play_say.c:1351 Setup timer success 320 bytes per
20 ms!
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[NOTICE] switch_channel.c:907 New Channel
sofia/external/501XXXXXX at trunk.dialog.pl
[28f96f4c-4fd0-4d19-bd69-6b08e6887db2]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] sofia.c:5142 Channel sofia/external/501XXXXXX at trunk.dialog.pl
entering state [received][100]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_core_state_machine.c:330
(sofia/external/501XXXXXX at trunk.dialog.pl) Running State Change CS_NEW
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] sofia.c:5153 Remote SDP:
v=0
o=BroadWorks 51808287 1 IN IP4 10.165.64.36
s=-
c=IN IP4 10.165.64.36
t=0 0
m=audio 22914 RTP/AVP 0 8 101
b=AS:82
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_core_state_machine.c:348
(sofia/external/501XXXXXX at trunk.dialog.pl) State NEW
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] sofia_glue.c:4731 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] sofia_glue.c:4731 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] sofia_glue.c:2839 Set Codec
sofia/external/501XXXXXX at trunk.dialog.pl PCMU/8000 20 ms 160 samples
64000 bits
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] sofia_glue.c:4845 Set 2833 dtmf send/recv payload to 101
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] sofia.c:5343 (sofia/external/501XXXXXX at trunk.dialog.pl) State
Change CS_NEW -> CS_INIT
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_core_session.c:1156 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_core_state_machine.c:330
(sofia/external/501XXXXXX at trunk.dialog.pl) Running State Change
CS_INIT
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_core_state_machine.c:369
(sofia/external/501XXXXXX at trunk.dialog.pl) State INIT
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] mod_sofia.c:85 sofia/external/501XXXXXX at trunk.dialog.pl SOFIA
INIT
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] mod_sofia.c:125 (sofia/external/501XXXXXX at trunk.dialog.pl)
State Change CS_INIT -> CS_ROUTING
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_core_session.c:1156 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_core_state_machine.c:369
(sofia/external/501XXXXXX at trunk.dialog.pl) State INIT going to sleep
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_core_state_machine.c:330
(sofia/external/501XXXXXX at trunk.dialog.pl) Running State Change
CS_ROUTING
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_channel.c:1836
(sofia/external/501XXXXXX at trunk.dialog.pl) Callstate Change DOWN ->
RINGING
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_core_state_machine.c:378
(sofia/external/501XXXXXX at trunk.dialog.pl) State ROUTING
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] mod_sofia.c:148 sofia/external/501XXXXXX at trunk.dialog.pl SOFIA
ROUTING
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_core_state_machine.c:78
sofia/external/501XXXXXX at trunk.dialog.pl Standard ROUTING
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283 [INFO]
mod_dialplan_xml.c:336 Processing 501XXXXXX <501XXXXXX>->71XXXXXXX in
context public
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl parsing [public->unloop]
continue=false
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl parsing
[public->outside_call] continue=true
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Absolute Condition
[outside_call]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action set(outside_call=true)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl parsing [public->call_debug]
continue=true
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Regex (FAIL) [call_debug]
${call_debug}(false) =~ /^true$/ break=never
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl parsing
[public->public_extensions] continue=false
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Regex (FAIL)
[public_extensions] destination_number(71XXXXXXX) =~
/^(1[0-9][0-9][0-9])$/ break=on-false
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl parsing [public->in_dialog_1]
continue=false
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Date/Time Match (FAIL)
[in_dialog_1] break=on-true
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Regex (PASS) [in_dialog_1]
destination_number(71XXXXXXX) =~ /^(717739100|71XXXXXXX|717854012)$/
break=on-false
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(RECORD_STEREO=true)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
record_session(/srv/nagrania/aktualne/IN_${strftime(%Y%m%d-%H%M%S)}_${destination_number}_${caller_id_number}.wav)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(media_bug_answer_req=true)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(domain_name=192.168.0.8)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
export(absolute_codec_string=PCMU,PCMA)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
export(codec_string=PCMU,PCMA)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(jitterbuffer_msec=180)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action set_audio_level(read
1)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(hangup_after_bridge=true)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action set(bind_meta_key=9)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action bind_meta_app(1 b s
execute_extension::dx XML features)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action bind_meta_app(2 b s
execute_extension::cf XML features)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action bind_meta_app(3 b s
execute_extension::att_xfer XML features)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action set(call_timeout=60)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(continue_on_fail=NO_ANSWER)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(limit_ignore_transfer=true)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action answer()
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action sleep(1000)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action
set(fifo_music=/srv/nagrania/ivr/ponce-preludio-in-e-major-10s.wav)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action ivr(ivr)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action fifo(fifo_ogolne in)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 Dialplan:
sofia/external/501XXXXXX at trunk.dialog.pl Action hangup()
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_core_state_machine.c:122
(sofia/external/501XXXXXX at trunk.dialog.pl) State Change CS_ROUTING ->
CS_EXECUTE
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_core_session.c:1156 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_core_state_machine.c:378
(sofia/external/501XXXXXX at trunk.dialog.pl) State ROUTING going to
sleep
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_core_state_machine.c:330
(sofia/external/501XXXXXX at trunk.dialog.pl) Running State Change
CS_EXECUTE
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_core_state_machine.c:385
(sofia/external/501XXXXXX at trunk.dialog.pl) State EXECUTE
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] mod_sofia.c:241 sofia/external/501XXXXXX at trunk.dialog.pl SOFIA
EXECUTE
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_core_state_machine.c:160
sofia/external/501XXXXXX at trunk.dialog.pl Standard EXECUTE
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set(outside_call=true)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [outside_call]=[true]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set(RFC2822_DATE=Thu, 01 Sep
2011 08:36:35 +0200)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [RFC2822_DATE]=[Thu, 01 Sep 2011 08:36:35 +0200]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set(RECORD_STEREO=true)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [RECORD_STEREO]=[true]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.283283
[DEBUG] switch_core_session.c:1983 Application record_session Requires
media! pre_answering channel sofia/external/501XXXXXX at trunk.dialog.pl
2011-09-01 08:36:35.283283 [INFO] switch_core_session.c:1985 Sending early media
2011-09-01 08:36:35.323283 [DEBUG] switch_nat.c:510 mapped public port
30160 protocol UDP to localport 30160
2011-09-01 08:36:35.363285 [DEBUG] switch_nat.c:510 mapped public port
30161 protocol UDP to localport 30161
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] sofia_glue.c:3091 AUDIO RTP
[sofia/external/501XXXXXX at trunk.dialog.pl] 172.20.0.22 port 30160 ->
10.165.64.36 port 22914 codec: 0 ms: 20
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] switch_rtp.c:1637 Starting timer [soft] 160 bytes per 20ms
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] sofia_glue.c:3353 Set 2833 dtmf send payload to 101
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] sofia_glue.c:3358 Set 2833 dtmf receive payload to 101
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] mod_sofia.c:2436 Ring SDP:
v=0
o=FreeSWITCH 1314828835 1314828836 IN IP4 10.64.64.64
s=FreeSWITCH
c=IN IP4 10.64.64.64
t=0 0
m=audio 30160 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

28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[NOTICE] mod_sofia.c:2439 Pre-Answer
sofia/external/501XXXXXX at trunk.dialog.pl!
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] switch_channel.c:2851
(sofia/external/501XXXXXX at trunk.dialog.pl) Callstate Change RINGING ->
EARLY
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] switch_core_session.c:713 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl
record_session(/srv/nagrania/aktualne/IN_20110901-083635_71XXXXXXX_501XXXXXX.wav)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] switch_core_session.c:857 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] sofia.c:5135 Channel sofia/external/501XXXXXX at trunk.dialog.pl
skipping state [early][183]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] switch_core_media_bug.c:360 Attaching BUG to
sofia/external/501XXXXXX at trunk.dialog.pl
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl
set(media_bug_answer_req=true)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [media_bug_answer_req]=[true]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set(domain_name=192.168.0.8)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [domain_name]=[192.168.0.8]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl
export(absolute_codec_string=PCMU,PCMA)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] switch_channel.c:1074 EXPORT (export_vars)
[absolute_codec_string]=[PCMU,PCMA]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl
export(codec_string=PCMU,PCMA)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] switch_channel.c:1074 EXPORT (export_vars)
[codec_string]=[PCMU,PCMA]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set(jitterbuffer_msec=180)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [jitterbuffer_msec]=[180]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set_audio_level(read 1)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] switch_core_media_bug.c:360 Attaching BUG to
sofia/external/501XXXXXX at trunk.dialog.pl
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set(hangup_after_bridge=true)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [hangup_after_bridge]=[true]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set(bind_meta_key=9)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [bind_meta_key]=[9]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl bind_meta_app(1 b s
execute_extension::dx XML features)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285 [INFO]
switch_ivr_async.c:3069 Bound B-Leg: 91 execute_extension::dx XML
features
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl bind_meta_app(2 b s
execute_extension::cf XML features)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285 [INFO]
switch_ivr_async.c:3069 Bound B-Leg: 92 execute_extension::cf XML
features
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl bind_meta_app(3 b s
execute_extension::att_xfer XML features)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285 [INFO]
switch_ivr_async.c:3069 Bound B-Leg: 93 execute_extension::att_xfer
XML features
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl set(call_timeout=60)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [call_timeout]=[60]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl
set(continue_on_fail=NO_ANSWER)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [continue_on_fail]=[NO_ANSWER]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl
set(limit_ignore_transfer=true)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [limit_ignore_transfer]=[true]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl answer()
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] mod_sofia.c:738 Local SDP
sofia/external/501XXXXXX at trunk.dialog.pl:
v=0
o=FreeSWITCH 1314828835 1314828837 IN IP4 10.64.64.64
s=FreeSWITCH
c=IN IP4 10.64.64.64
t=0 0
m=audio 30160 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

28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] switch_core_session.c:713 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] switch_channel.c:3043
(sofia/external/501XXXXXX at trunk.dialog.pl) Callstate Change EARLY ->
ACTIVE
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] switch_core_session.c:857 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[NOTICE] mod_dptools.c:930 Channel
[sofia/external/501XXXXXX at trunk.dialog.pl] has been answered
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:35.363285
[DEBUG] sofia.c:5142 Channel sofia/external/501XXXXXX at trunk.dialog.pl
entering state [completed][200]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl sleep(1000)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl
set(fifo_music=/srv/nagrania/ivr/ponce-preludio-in-e-major-10s.wav)
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:36.363287
[DEBUG] mod_dptools.c:1063 sofia/external/501XXXXXX at trunk.dialog.pl
SET [fifo_music]=[/srv/nagrania/ivr/ponce-preludio-in-e-major-10s.wav]
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 EXECUTE
sofia/external/501XXXXXX at trunk.dialog.pl ivr(ivr)
2011-09-01 08:36:36.363287 [DEBUG] switch_ivr_menu.c:660
switch_ivr_menu_stack_xml_add binding 'menu-exit'
2011-09-01 08:36:36.363287 [DEBUG] switch_ivr_menu.c:660
switch_ivr_menu_stack_xml_add binding 'menu-sub'
2011-09-01 08:36:36.363287 [DEBUG] switch_ivr_menu.c:660
switch_ivr_menu_stack_xml_add binding 'menu-exec-app'
2011-09-01 08:36:36.363287 [DEBUG] switch_ivr_menu.c:660
switch_ivr_menu_stack_xml_add binding 'menu-play-sound'
2011-09-01 08:36:36.363287 [DEBUG] switch_ivr_menu.c:660
switch_ivr_menu_stack_xml_add binding 'menu-back'
2011-09-01 08:36:36.363287 [DEBUG] switch_ivr_menu.c:660
switch_ivr_menu_stack_xml_add binding 'menu-top'
2011-09-01 08:36:36.363287 [DEBUG] switch_ivr_menu.c:791 building menu 'ivr'
2011-09-01 08:36:36.363287 [DEBUG] switch_ivr_menu.c:852 binding menu
action 'menu-exec-app' to '1'
2011-09-01 08:36:36.363287 [DEBUG] switch_ivr_menu.c:852 binding menu
action 'menu-exec-app' to '2'
2011-09-01 08:36:36.363287 [DEBUG] switch_ivr_menu.c:852 binding menu
action 'menu-top' to '9'
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:36.363287
[DEBUG] switch_ivr_menu.c:428 Executing IVR menu ivr
2011-09-01 08:36:36.363287 [DEBUG] switch_core_file.c:180 File
/srv/nagrania/ivr/powitanie_pelne.wav sample rate 22050 doesn't match
requested rate 8000
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:36.363287
[DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1
channels 20ms
28f96f4c-4fd0-4d19-bd69-6b08e6887db2 2011-09-01 08:36:36.363287
[DEBUG] switch_ivr_play_say.c:1351 Setup timer success 320 bytes per
20 ms!
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:40.183284
[DEBUG] switch_core_session.c:857 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:40.183284
[DEBUG] switch_core_session.c:857 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:40.203284
[DEBUG] sofia.c:5142 Channel sofia/external/501XXXXXX at trunk.dialog.pl
entering state [terminating][0]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:40.203284
[DEBUG] switch_channel.c:2775
(sofia/external/501XXXXXX at trunk.dialog.pl) Callstate Change ACTIVE ->
HANGUP
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:40.203284
[NOTICE] sofia.c:5867 Hangup sofia/external/501XXXXXX at trunk.dialog.pl
[CS_EXECUTE] [NORMAL_UNSPECIFIED]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:40.203284
[DEBUG] switch_channel.c:2791 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [KILL]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:40.203284
[DEBUG] switch_core_session.c:1156 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:40.203284
[DEBUG] switch_ivr_play_say.c:1672 done playing file
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:40.203284
[DEBUG] switch_ivr_menu.c:343 waiting for 1/1 digits t/o 2000
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:40.203284
[DEBUG] switch_ivr_menu.c:390 digits ''
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:40.203284
[DEBUG] switch_ivr_menu.c:580 IVR menu 'ivr' no input detected
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.203285
[DEBUG] switch_ivr_menu.c:594 exit-sound
'/srv/nagrania/ivr/laczenie_z_operatorem.wav'
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.203285
[DEBUG] switch_core_session.c:2133
sofia/external/501XXXXXX at trunk.dialog.pl skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.203285
[DEBUG] switch_core_state_machine.c:385
(sofia/external/501XXXXXX at trunk.dialog.pl) State EXECUTE going to
sleep
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.203285
[DEBUG] switch_core_state_machine.c:330
(sofia/external/501XXXXXX at trunk.dialog.pl) Running State Change
CS_HANGUP
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.203285
[DEBUG] switch_core_media_bug.c:480 Removing BUG from
sofia/external/501XXXXXX at trunk.dialog.pl
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.203285
[DEBUG] switch_ivr_async.c:937 Stop recording file
/srv/nagrania/aktualne/IN_20110901-083608_71XXXXXXX_501XXXXXX.wav
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.203285
[DEBUG] switch_core_media_bug.c:480 Removing BUG from
sofia/external/501XXXXXX at trunk.dialog.pl
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.203285
[DEBUG] switch_core_state_machine.c:580
(sofia/external/501XXXXXX at trunk.dialog.pl) State HANGUP
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.203285
[DEBUG] mod_sofia.c:458 Channel
sofia/external/501XXXXXX at trunk.dialog.pl hanging up, cause:
NORMAL_UNSPECIFIED
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.203285
[DEBUG] switch_core_state_machine.c:46
sofia/external/501XXXXXX at trunk.dialog.pl Standard HANGUP, cause:
NORMAL_UNSPECIFIED
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.203285
[DEBUG] switch_core_state_machine.c:580
(sofia/external/501XXXXXX at trunk.dialog.pl) State HANGUP going to sleep
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.203285
[DEBUG] switch_core_state_machine.c:361
(sofia/external/501XXXXXX at trunk.dialog.pl) State Change CS_HANGUP ->
CS_REPORTING
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.203285
[DEBUG] switch_core_session.c:1156 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.203285
[DEBUG] switch_core_state_machine.c:330
(sofia/external/501XXXXXX at trunk.dialog.pl) Running State Change
CS_REPORTING
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.203285
[DEBUG] switch_core_state_machine.c:640
(sofia/external/501XXXXXX at trunk.dialog.pl) State REPORTING
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.263295
[DEBUG] switch_core_state_machine.c:53
sofia/external/501XXXXXX at trunk.dialog.pl Standard REPORTING, cause:
NORMAL_UNSPECIFIED
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.263295
[DEBUG] switch_core_state_machine.c:640
(sofia/external/501XXXXXX at trunk.dialog.pl) State REPORTING going to
sleep
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.263295
[DEBUG] switch_core_state_machine.c:355
(sofia/external/501XXXXXX at trunk.dialog.pl) State Change CS_REPORTING
-> CS_DESTROY
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.263295
[DEBUG] switch_core_session.c:1156 Send signal
sofia/external/501XXXXXX at trunk.dialog.pl [BREAK]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.263295
[DEBUG] switch_core_session.c:1328 Session 2
(sofia/external/501XXXXXX at trunk.dialog.pl) Locked, Waiting on external
entities
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.263295
[NOTICE] switch_core_session.c:1346 Session 2
(sofia/external/501XXXXXX at trunk.dialog.pl) Ended
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.263295
[NOTICE] switch_core_session.c:1348 Close Channel
sofia/external/501XXXXXX at trunk.dialog.pl [CS_DESTROY]
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.263295
[DEBUG] switch_core_state_machine.c:469
(sofia/external/501XXXXXX at trunk.dialog.pl) Callstate Change HANGUP ->
DOWN
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.263295
[DEBUG] switch_core_state_machine.c:472
(sofia/external/501XXXXXX at trunk.dialog.pl) Running State Change
CS_DESTROY
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.263295
[DEBUG] switch_core_state_machine.c:482
(sofia/external/501XXXXXX at trunk.dialog.pl) State DESTROY
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.263295
[DEBUG] mod_sofia.c:363 sofia/external/501XXXXXX at trunk.dialog.pl SOFIA
DESTROY
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.343286
[DEBUG] switch_core_state_machine.c:60
sofia/external/501XXXXXX at trunk.dialog.pl Standard DESTROY
b541f405-1a39-4cbc-80dc-8bdbc92ed070 2011-09-01 08:36:41.343286
[DEBUG] switch_core_state_machine.c:482
(sofia/external/501XXXXXX at trunk.dialog.pl) State DESTROY going to
sleep


Any ideas ?

--
Greetings - Tomasz Hyziak


From dgarcia at anew.com.ve  Thu Sep  1 17:41:02 2011
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Thu, 01 Sep 2011 09:11:02 -0430
Subject: [Freeswitch-users] Yealink t28p setup - please help
In-Reply-To: <4E5F5655.9050705@the800group.com>
References: <4E5F5655.9050705@the800group.com>
Message-ID: <4E5F8B6E.4040002@anew.com.ve>

mmm,
I am also new with FS

I think you can't doit in that way. Each phone will register the same 
extensi?n, so FS will receive the register the sip account from two 
different phones, FS will associate the sip account with the last phone 
who receive the requirement

You could try this:
1. Program each phone with 3 different accounts: Phone A : 1001, 1003, 
1004, ; Phone B: 1002, 1005,1006
2. Create 2 hunt group: 2000, 2001. HG 2000 will ring in 1003 and 1005. 
HG 2001 in 1004 and 1006
3. Direct the POST line to one HG, Direct the extension to the second HG.

You are trying to implement a feature boss/secretary?

On 9/1/2011 5:24 AM, ocset wrote:
> Hi All
>
> I am new to FreeSwitch and this is my first post on this forum/mailing-list.
>
> I have a default install of FS on Ubuntu 10.10 and have two Yealink T28P
> phones. I am trying to understand the best way to config these phones
> for FS. What I am fighting with at the moment is that if I assign the
> same extension to both phones, only one phone will ring when the shared
> extension is called.
>
> The phones have the option to assign 6 SIP accounts/extensions and this
> is my current test config.
>
> Phone 1
> ext 1001
> ext 1010
> ext 1011
>
> Phone 2
> ext 1002
> ext 1010
> ext 1011
>
> My thinking behind this is that I want to assign a POST line to the 1010
> extension (using an SPA3102)  and a VOIP line to the 1011 extension.
> That way, when an external call comes in on one of those lines, all
> parties have the option to pick up the call. What I have found however
> is that only one of the phones rings and is able to answer the 1010 and
> 1011 extensions. Is this expected behaviour? Should I be doing this
> another way?
>
> I am very new to VOIP, Freeswitch etc so my terminology may be wrong -
> hopefully you understand my explanation.
>
> 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
>
>
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 10.0.1392 / Virus Database: 1520/3869 - Release Date: 08/31/11
>
>


-- 
Atentamente,
*Dario Garc?a*
Consultor.

CCCT, Nivel C2, Sector Yarey, Mz,
Ofc. MZ03a.
Caracas-Venezuela.
Tel?fono: +58 212 9081842
Cel: +58 412 2221515
dgarcia at anew.com.ve
http://www.anew.com.ve
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/fda845b5/attachment.html 

From jeff at jefflenk.com  Thu Sep  1 17:45:10 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 1 Sep 2011 06:45:10 -0700 (PDT)
Subject: [Freeswitch-users] FreeSwitch and core dumps
In-Reply-To: <CAFv7ub1ciPKA9ceWcJrUx3fQROXrAxsdDyc0Q355gRB-ZtU52g@mail.gmail.com>
References: <CAFv7ub1ciPKA9ceWcJrUx3fQROXrAxsdDyc0Q355gRB-ZtU52g@mail.gmail.com>
Message-ID: <1314884710492-6750074.post@n2.nabble.com>

Is this with Git Head? It appears that something is wrong with curl.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-and-core-dumps-tp6750013p6750074.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From avi at avimarcus.net  Thu Sep  1 17:45:13 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 1 Sep 2011 16:45:13 +0300
Subject: [Freeswitch-users] Yealink t28p setup - please help
In-Reply-To: <4E5F8B6E.4040002@anew.com.ve>
References: <4E5F5655.9050705@the800group.com> <4E5F8B6E.4040002@anew.com.ve>
Message-ID: <CALNZuwmEEZwaaJ4DU8eGY2Y3TTAcA+_ROm-xCO-gRwQSQ=rBnQ@mail.gmail.com>

Hi.
A) if you wanted both phones to be able to share one SIP account, you need
to turn on multiple registrations in your internal sip profile.
B) I know the yealink supports it, but why would you need more than one SIP
account per phone? You can simply set up a group and all the routing on the
server-side rather than the phone, to have it ring the phones of your
choice.

-Avi Marcus


On Thu, Sep 1, 2011 at 4:41 PM, Saugort Dario Garcia Tovar <
dgarcia at anew.com.ve> wrote:

>  mmm,
> I am also new with FS
>
> I think you can't doit in that way. Each phone will register the same
> extensi?n, so FS will receive the register the sip account from two
> different phones, FS will associate the sip account with the last phone who
> receive the requirement
>
> You could try this:
> 1. Program each phone with 3 different accounts: Phone A : 1001, 1003,
> 1004, ; Phone B: 1002, 1005,1006
> 2. Create 2 hunt group: 2000, 2001. HG 2000 will ring in 1003 and 1005. HG
> 2001 in 1004 and 1006
> 3. Direct the POST line to one HG, Direct the extension to the second HG.
>
> You are trying to implement a feature boss/secretary?
>
> On 9/1/2011 5:24 AM, ocset wrote:
>
> Hi All
>
> I am new to FreeSwitch and this is my first post on this forum/mailing-list.
>
> I have a default install of FS on Ubuntu 10.10 and have two Yealink T28P
> phones. I am trying to understand the best way to config these phones
> for FS. What I am fighting with at the moment is that if I assign the
> same extension to both phones, only one phone will ring when the shared
> extension is called.
>
> The phones have the option to assign 6 SIP accounts/extensions and this
> is my current test config.
>
> Phone 1
> ext 1001
> ext 1010
> ext 1011
>
> Phone 2
> ext 1002
> ext 1010
> ext 1011
>
> My thinking behind this is that I want to assign a POST line to the 1010
> extension (using an SPA3102)  and a VOIP line to the 1011 extension.
> That way, when an external call comes in on one of those lines, all
> parties have the option to pick up the call. What I have found however
> is that only one of the phones rings and is able to answer the 1010 and
> 1011 extensions. Is this expected behaviour? Should I be doing this
> another way?
>
> I am very new to VOIP, Freeswitch etc so my terminology may be wrong -
> hopefully you understand my explanation.
>
> Thanks in advance
>
>
> 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
>
>
>
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 10.0.1392 / Virus Database: 1520/3869 - Release Date: 08/31/11
>
>
>
>
>
> --
> Atentamente,
> *Dario Garc?a*
> Consultor.
>
> CCCT, Nivel C2, Sector Yarey, Mz,
> Ofc. MZ03a.
> Caracas-Venezuela.
> Tel?fono: +58 212 9081842
> Cel: +58 412 2221515
> dgarcia at anew.com.ve
> http://www.anew.com.ve
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/a53bb614/attachment.html 

From tomasz at hyziak.pl  Thu Sep  1 17:57:59 2011
From: tomasz at hyziak.pl (Tomasz Hyziak)
Date: Thu, 1 Sep 2011 15:57:59 +0200
Subject: [Freeswitch-users] FreeSwitch and core dumps
In-Reply-To: <1314884710492-6750074.post@n2.nabble.com>
References: <CAFv7ub1ciPKA9ceWcJrUx3fQROXrAxsdDyc0Q355gRB-ZtU52g@mail.gmail.com>
	<1314884710492-6750074.post@n2.nabble.com>
Message-ID: <CAFv7ub0crGLj13WSu8s0zEgP759zGP=bFFaVjwMtx0X=oKZ30g@mail.gmail.com>

Hi Jeef.

Yes, that's FreeSWITCH compiled from git version: 1.0.head
(git-cd31633 2011-08-17 19-34-22 -0500).

I use mod_xml_cdr for saving cdrs to database via webservice.

What could I do with it ? Recompiling mod_xml_cdr or curl ?

--
pozdrawiam - Tomasz Hyziak



2011/9/1 Jeff Lenk <jeff at jefflenk.com>:
> Is this with Git Head? It appears that something is wrong with curl.
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-and-core-dumps-tp6750013p6750074.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 ocset at the800group.com  Thu Sep  1 18:45:57 2011
From: ocset at the800group.com (ocset)
Date: Thu, 01 Sep 2011 22:45:57 +0800
Subject: [Freeswitch-users] Yealink t28p setup - please help
In-Reply-To: <CALNZuwmEEZwaaJ4DU8eGY2Y3TTAcA+_ROm-xCO-gRwQSQ=rBnQ@mail.gmail.com>
References: <4E5F5655.9050705@the800group.com> <4E5F8B6E.4040002@anew.com.ve>
	<CALNZuwmEEZwaaJ4DU8eGY2Y3TTAcA+_ROm-xCO-gRwQSQ=rBnQ@mail.gmail.com>
Message-ID: <4E5F9AA5.20900@the800group.com>

Thanks Avi

I changed the parameter "multiple-registrations" to true and it now 
works as I expected. Thanks also for your advice on other ways to 
implement the same solution.

I guess FS is just like a programming language - there are many ways to 
reach the same end. For me, the most important thing is to make sure I 
configure FS within the confines of how it was intended to work so as to 
have a stable, predictable solution.

Regards


On 09/01/2011 09:45 PM, Avi Marcus wrote:
> Hi.
> A) if you wanted both phones to be able to share one SIP account, you 
> need to turn on multiple registrations in your internal sip profile.
> B) I know the yealink supports it, but why would you need more than 
> one SIP account per phone? You can simply set up a group and all the 
> routing on the server-side rather than the phone, to have it ring the 
> phones of your choice.
>
> -Avi Marcus
>
>
> On Thu, Sep 1, 2011 at 4:41 PM, Saugort Dario Garcia Tovar 
> <dgarcia at anew.com.ve <mailto:dgarcia at anew.com.ve>> wrote:
>
>     mmm,
>     I am also new with FS
>
>     I think you can't doit in that way. Each phone will register the
>     same extensi?n, so FS will receive the register the sip account
>     from two different phones, FS will associate the sip account with
>     the last phone who receive the requirement
>
>     You could try this:
>     1. Program each phone with 3 different accounts: Phone A : 1001,
>     1003, 1004, ; Phone B: 1002, 1005,1006
>     2. Create 2 hunt group: 2000, 2001. HG 2000 will ring in 1003 and
>     1005. HG 2001 in 1004 and 1006
>     3. Direct the POST line to one HG, Direct the extension to the
>     second HG.
>
>     You are trying to implement a feature boss/secretary?
>
>     On 9/1/2011 5:24 AM, ocset wrote:
>>     Hi All I am new to FreeSwitch and this is my first post on this
>>     forum/mailing-list. I have a default install of FS on Ubuntu
>>     10.10 and have two Yealink T28P phones. I am trying to understand
>>     the best way to config these phones for FS. What I am fighting
>>     with at the moment is that if I assign the same extension to both
>>     phones, only one phone will ring when the shared extension is
>>     called. The phones have the option to assign 6 SIP
>>     accounts/extensions and this is my current test config. Phone 1
>>     ext 1001 ext 1010 ext 1011 Phone 2 ext 1002 ext 1010 ext 1011 My
>>     thinking behind this is that I want to assign a POST line to the
>>     1010 extension (using an SPA3102) and a VOIP line to the 1011
>>     extension. That way, when an external call comes in on one of
>>     those lines, all parties have the option to pick up the call.
>>     What I have found however is that only one of the phones rings
>>     and is able to answer the 1010 and 1011 extensions. Is this
>>     expected behaviour? Should I be doing this another way? I am very
>>     new to VOIP, Freeswitch etc so my terminology may be wrong -
>>     hopefully you understand my explanation. Thanks in advance
>>     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
>>
>>
>>
>>     -----
>>     No virus found in this message.
>>     Checked by AVG -www.avg.com  <http://www.avg.com>
>>     Version: 10.0.1392 / Virus Database: 1520/3869 - Release Date: 08/31/11
>>
>>
>
>
>     -- 
>     Atentamente,
>     *Dario Garc?a*
>     Consultor.
>
>     CCCT, Nivel C2, Sector Yarey, Mz,
>     Ofc. MZ03a.
>     Caracas-Venezuela.
>     Tel?fono: +58 212 9081842 <tel:%2B58%20212%209081842>
>     Cel: +58 412 2221515 <tel:%2B58%20412%202221515>
>     dgarcia at anew.com.ve <mailto:dgarcia at anew.com.ve>
>     http://www.anew.com.ve
>
>
>     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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/464303d4/attachment-0001.html 

From ocset at the800group.com  Thu Sep  1 19:24:32 2011
From: ocset at the800group.com (ocset)
Date: Thu, 01 Sep 2011 23:24:32 +0800
Subject: [Freeswitch-users] FXO/FXS card advice
Message-ID: <4E5FA3B0.9030606@the800group.com>

Hi

As a new member of the forum, I am curious to know your experience with 
FXO/FXS cards.

I have a SPA3102 and have configured it to work with FS, but it feels a 
bit like I am trying to make a square peg fit a round hole. I am hoping 
to implement FS at a business which currently has 4 POTS lines and would 
prefer to use an internal IDE card for the job of integrating these 
phone lines into FreeSwitch.

Has anyone got some advice on which cards I should be looking at that 
just "work" with FS? What about echo cancellation - is that something I 
should just cater for or does it depend on the client situation? What 
about software cancellation?

This seems like one of those times when too much choice is a bad thing 
and I need some guidance on what has worked for you.

All advice would be greatly appreciated.

Regards


From msc at freeswitch.org  Thu Sep  1 19:49:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 1 Sep 2011 08:49:44 -0700
Subject: [Freeswitch-users] session:recordFile(-) always creates empty
 wav file during outbound IVR call
In-Reply-To: <1314870492.70429.YahooMailClassic@web39708.mail.mud.yahoo.com>
References: <CAKbxfG_Bi1P1Ykbf2v2XyLdSdFgugz+LDqrejzQvbLOXmF6itg@mail.gmail.com>
	<1314870492.70429.YahooMailClassic@web39708.mail.mud.yahoo.com>
Message-ID: <CAKzWOxVB8s+UpVQ58_7gcOA2sZqfPtf3e6_r+a2eu2KBva+fwQ@mail.gmail.com>

Tony mentioned something that might be an issue for you. Try setting this on
the channel:
http://wiki.freeswitch.org/wiki/Channel_Variables#record_waste_resources

Let us know if that makes a difference.

-MC

On Thu, Sep 1, 2011 at 2:48 AM, king2kin <xing2kin at yahoo.com> wrote:

> Hi Anthony,
>
> Actually FreeSwitch application "record" doesn't work either for outbound
> IVR call (see below), created an empty wav file with its size 68 bytes, the
> wav file doesn't contain any samples of audio data;
>
> <action application="record"
> data="C:/c4dev/freeswitch/Debug/sounds/test.wav 20 200"/>
>
> earlier I reported that "session:recordFile(-)" doesn't work inside Lua
> Script for any outbound IVR call.
>
> - My CLI commands to make an outbound IVR call:
>
> originate {ignore_early_media=true}sofia/gateway/mygateway/1726011  8884
> or
> originate user/1005   8884
>
> - xml diaplan for extension 8884:
> {
>        <extension name="Play and Record Sound File">
>                <condition field="destination_number" expression="^(8884)$">
>                        <action application="answer"/>
>                        <action application="sleep" data="1000"/>
>                        <action application="playback"
> data="ivr/ivr-thank_you.wav"/>
>
>                        <action application="record"
> data="C:/c4dev/freeswitch/Debug/sounds/test.wav 20 200"/>
>
>                        <action application="sleep" data="1000"/>
>                        <action application="playback"
> data="voicemail/vm-goodbye.wav"/>
>
>                        <action application="sleep" data="1000"/>
>                        <action application="hangup"/>
>                </condition>
>        </extension>
> }
>
> --- On Wed, 8/31/11, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>
> > From: Anthony Minessale <anthony.minessale at gmail.com>
> > Subject: Re: [Freeswitch-users] session:recordFile(-) always creates
> empty wav file during outbound IVR call
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Date: Wednesday, August 31, 2011, 7:46 PM
> > Does the empty file contain silence
> > that corresponds to the duration
> > of the time it's recording?
> > Are you producing the audio yourself into the recording and
> > can you
> > verify with a pcap that there is actually any audio to
> > record?
> >
> >
> > On Wed, Aug 31, 2011 at 9:21 PM, king2kin <xing2kin at yahoo.com>
> > wrote:
> > > Anthony,
> > >
> > > Thank you for help. I tried outbound IVR call in
> > multiple ways again based on your advice,
> >  session:recordFile(-) still doesn't work normally, still
> > created empty wav file .
> > >
> > > Could anyone please give me a hand on FreeSwitch
> > Record? It always fails to record audio file during any
> > outbound IVR call (auto dialer) although it works well
> > during any inbound ivr call.
> > >
> > > Session:streamFile(-) works well to play back prompt
> > files, dtmf keypress also works, ... during outbound ivr
> > call.
> > >
> > > x.k.
> > >
> > > --- On Wed, 8/31/11, Anthony Minessale <anthony.minessale at gmail.com>
> > wrote:
> > >
> > >> From: Anthony Minessale <anthony.minessale at gmail.com>
> > >> Subject: Re: [Freeswitch-users]
> > session:recordFile(-) always creates empty wav file during
> > outbound IVR call
> > >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > >> Date: Wednesday, August 31, 2011, 12:17 PM
> > >> try this dial string instead
> > >>
> > >>
> > {ignore_early_media=true}sofia/gateway/sip.tpad.com/1726011
> > >>
> > >> On Wed, Aug 31, 2011 at 12:52 PM, king2kin <xing2kin at yahoo.com>
> > >> wrote:
> > >> > Hi folks,
> > >> >
> > >> > With Lua script and/or originate command, I
> > have tried
> > >> recording a message file during outbound IVR call
> > over and
> > >> over, session:recordFile(-) inside Lua script does
> > create a
> > >> wav file during each of my testings but the
> > recorded audio
> > >> file is always empty.
> > >> >
> > >> > However, session:recordFile(-) works well for
> > inbound
> > >> IVR call.
> > >> >
> > >> > I tried the session:recordFile(-) via Lua
> > script in
> > >> three ways:
> > >> >
> > >> > 1. run lua script "test_outcall_ivr.lua" at
> > freeswitch
> > >> command-line:
> > >> >
> > >> > luarun test_outcall_ivr.lua
> > >> >
> > >> >
> > >> > -- [test_outcall_ivr.lua]
> > >> > {
> > >> > local sessionx =
> > >>
> > freeswitch.Session("sofia/gateway/sip.tpad.com/1726011",
> > >> session)
> > >> >
> > >> > -- Set the path separator
> > >> > pathsep = '/'
> > >> >
> > >> > -- Windows users do this instead:
> > >> > -- pathsep = '\'
> > >> >
> > >> > -- Answer the call
> > >> > -- sessionx:answer()
> > >> >
> > >> > --Create a string with path and filename of a
> > sound
> > >> file
> > >> > prompt = "ivr" .. pathsep ..
> > >> "ivr-welcome_to_freeswitch.wav"
> > >> >
> > >> > -- Print a log message
> > >> > freeswitch.consoleLog("INFO","Prompt file is
> > '" ..
> > >> prompt .. "'\n")
> > >> >
> > >> > --Play the prompt
> > >> > sessionx:streamFile(prompt)
> > >> >
> > >> > -- Record record file
> > >> >
> > >>
> > sessionx:streamFile("phrase:voicemail_record_message")
> > >> >
> > >> > -- Play a ""bong"" tone prior to recording
> > >> >
> > >>
> >
> sessionx:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
> > >> 0, 640)")
> > >> >
> > >> > -- record a message
> > >> > filename = sessionx:getVariable('sounds_dir')
> > ..
> > >> pathsep .. "123.wav"
> > >> > sessionx:recordFile(filename,300,100,10)
> > >> >
> > >> > -- play back the recorded msg
> > >> > sessionx:streamFile(filename)
> > >> >
> > >> > -- Hangup
> > >> > sessionx:hangup()
> > >> >
> > >> > }
> > >> >
> > >> > 2. I also tried it differently by submitting
> > the
> > >> following commands at the FreeSwitch command-line
> > >> interface:
> > >> >
> > >> > originate user/1005 &transfer(8887 xml
> > default)
> > >> >
> > >> > originate user/1005 &lua('test1.lua')
> > >> >
> > >> > originate sofia/gateway/sip.tpad.com/1726011
> > >> &lua('test1.lua')
> > >> >
> > >> >
> > >> > -- [test1.lua]
> > >> > {
> > >> > -- Set the path separator
> > >> > pathsep = '/'
> > >> >
> > >> > -- Windows users do this instead:
> > >> > -- pathsep = '\'
> > >> >
> > >> > --Answer the call
> > >> > session:answer()
> > >> >
> > >> > --Create a string with path and filename of a
> > sound
> > >> file
> > >> > prompt = "ivr" .. pathsep ..
> > >> "ivr-welcome_to_freeswitch.wav"
> > >> >
> > >> > -- Print a log message
> > >> > freeswitch.consoleLog("INFO","Prompt file is
> > '" ..
> > >> prompt .. "'\n")
> > >> >
> > >> > --Play the prompt
> > >> > session:streamFile(prompt)
> > >> >
> > >> > -- Record record file
> > >> >
> > session:streamFile("phrase:voicemail_record_message")
> > >> >
> > >> > -- Play a ""bong"" tone prior to recording
> > >> >
> > >>
> >
> session:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
> > >> 0, 640)")
> > >> >
> > >> > -- record a message
> > >> > filename = session:getVariable('sounds_dir')
> > ..
> > >> pathsep .. "123.wav"
> > >> > session:recordFile(filename,300,100,10)
> > >> >
> > >> > -- play back the recorded msg
> > >> > session:streamFile(filename)
> > >> >
> > >> > -- Hangup
> > >> > session:hangup()
> > >> > }
> > >> >
> > >> > -- [xml dialplan for extension 8887]:
> > >> > {
> > >> >        <extension name="Simple Lua
> > Test">
> > >> >                <condition
> > >> field="destination_number"
> > expression="^(8887)$">
> > >> >
> >  <action
> > >> application="set"
> > data="record_waste_resources=true"/>
> > >> >
> >  <action
> > >> application="lua" data="test1.lua"/>
> > >> >                </condition>
> > >> >        </extension>
> > >> > }
> > >> >
> > >> >
> > >>
> > ======================================================
> > >> >
> > >> > For all the above testing cases,
> > session:recordFile(-)
> > >> always creates an empty wav file for each of
> > outbound IVR
> > >> calls, however, if I make an inbound IVR call to
> > run Lua
> > >> script "test1.lua", session:recordFile(-) always
> > works
> > >> perfect to generate a normal wav file.
> > >> >
> > >> > So, what's wrong with [session:recordFile(-)]
> > during
> > >> an outbound IVR call?
> > >> >
> > >> > x.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
> > >> >
> > >>
> > >>
> > >>
> > >> --
> > >> 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
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/50db6313/attachment-0001.html 

From michel.daggelinckx at gmail.com  Thu Sep  1 19:50:58 2011
From: michel.daggelinckx at gmail.com (Michel Daggelinckx)
Date: Thu, 1 Sep 2011 17:50:58 +0200
Subject: [Freeswitch-users] FXO/FXS card advice
In-Reply-To: <4E5FA3B0.9030606@the800group.com>
References: <4E5FA3B0.9030606@the800group.com>
Message-ID: <CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com>

Digium and sangoma cards are high quality and work greast with FS

Michel


On Thu, Sep 1, 2011 at 5:24 PM, ocset <ocset at the800group.com> wrote:

> Hi
>
> As a new member of the forum, I am curious to know your experience with
> FXO/FXS cards.
>
> I have a SPA3102 and have configured it to work with FS, but it feels a
> bit like I am trying to make a square peg fit a round hole. I am hoping
> to implement FS at a business which currently has 4 POTS lines and would
> prefer to use an internal IDE card for the job of integrating these
> phone lines into FreeSwitch.
>
> Has anyone got some advice on which cards I should be looking at that
> just "work" with FS? What about echo cancellation - is that something I
> should just cater for or does it depend on the client situation? What
> about software cancellation?
>
> This seems like one of those times when too much choice is a bad thing
> and I need some guidance on what has worked for you.
>
> All advice would be greatly appreciated.
>
> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/baa12004/attachment.html 

From anthony.minessale at gmail.com  Thu Sep  1 19:51:09 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 1 Sep 2011 10:51:09 -0500
Subject: [Freeswitch-users] session:recordFile(-) always creates empty
 wav file during outbound IVR call
In-Reply-To: <1314870492.70429.YahooMailClassic@web39708.mail.mud.yahoo.com>
References: <CAKbxfG_Bi1P1Ykbf2v2XyLdSdFgugz+LDqrejzQvbLOXmF6itg@mail.gmail.com>
	<1314870492.70429.YahooMailClassic@web39708.mail.mud.yahoo.com>
Message-ID: <CAKbxfG8vwn72DLzdwaTadAu0gzNoVgwRcham7vW49Z1kZRNcSA@mail.gmail.com>

now try this:

{record_waste_resources=true,ignore_early_media=true}sofia/gateway/mygateway/1726011

My guess is that the other end of your call does not support
asynchronous RTP and since we do not send media while recording its
not either.




On Thu, Sep 1, 2011 at 4:48 AM, king2kin <xing2kin at yahoo.com> wrote:
> Hi Anthony,
>
> Actually FreeSwitch application "record" doesn't work either for outbound IVR call (see below), created an empty wav file with its size 68 bytes, the wav file doesn't contain any samples of audio data;
>
> <action application="record" data="C:/c4dev/freeswitch/Debug/sounds/test.wav 20 200"/>
>
> earlier I reported that "session:recordFile(-)" doesn't work inside Lua Script for any outbound IVR call.
>
> - My CLI commands to make an outbound IVR call:
>
> originate {ignore_early_media=true}sofia/gateway/mygateway/1726011 ?8884
> or
> originate user/1005 ? 8884
>
> - xml diaplan for extension 8884:
> {
> ? ? ? ?<extension name="Play and Record Sound File">
> ? ? ? ? ? ? ? ?<condition field="destination_number" expression="^(8884)$">
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="answer"/>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="sleep" data="1000"/>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="playback" data="ivr/ivr-thank_you.wav"/>
>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="record" data="C:/c4dev/freeswitch/Debug/sounds/test.wav 20 200"/>
>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="sleep" data="1000"/>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="playback" data="voicemail/vm-goodbye.wav"/>
>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="sleep" data="1000"/>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="hangup"/>
> ? ? ? ? ? ? ? ?</condition>
> ? ? ? ?</extension>
> }
>
> --- On Wed, 8/31/11, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> Subject: Re: [Freeswitch-users] session:recordFile(-) always creates empty wav file during outbound IVR call
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Date: Wednesday, August 31, 2011, 7:46 PM
>> Does the empty file contain silence
>> that corresponds to the duration
>> of the time it's recording?
>> Are you producing the audio yourself into the recording and
>> can you
>> verify with a pcap that there is actually any audio to
>> record?
>>
>>
>> On Wed, Aug 31, 2011 at 9:21 PM, king2kin <xing2kin at yahoo.com>
>> wrote:
>> > Anthony,
>> >
>> > Thank you for help. I tried outbound IVR call in
>> multiple ways again based on your advice,
>> ?session:recordFile(-) still doesn't work normally, still
>> created empty wav file .
>> >
>> > Could anyone please give me a hand on FreeSwitch
>> Record? It always fails to record audio file during any
>> outbound IVR call (auto dialer) although it works well
>> during any inbound ivr call.
>> >
>> > Session:streamFile(-) works well to play back prompt
>> files, dtmf keypress also works, ... during outbound ivr
>> call.
>> >
>> > x.k.
>> >
>> > --- On Wed, 8/31/11, Anthony Minessale <anthony.minessale at gmail.com>
>> wrote:
>> >
>> >> From: Anthony Minessale <anthony.minessale at gmail.com>
>> >> Subject: Re: [Freeswitch-users]
>> session:recordFile(-) always creates empty wav file during
>> outbound IVR call
>> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> >> Date: Wednesday, August 31, 2011, 12:17 PM
>> >> try this dial string instead
>> >>
>> >>
>> {ignore_early_media=true}sofia/gateway/sip.tpad.com/1726011
>> >>
>> >> On Wed, Aug 31, 2011 at 12:52 PM, king2kin <xing2kin at yahoo.com>
>> >> wrote:
>> >> > Hi folks,
>> >> >
>> >> > With Lua script and/or originate command, I
>> have tried
>> >> recording a message file during outbound IVR call
>> over and
>> >> over, session:recordFile(-) inside Lua script does
>> create a
>> >> wav file during each of my testings but the
>> recorded audio
>> >> file is always empty.
>> >> >
>> >> > However, session:recordFile(-) works well for
>> inbound
>> >> IVR call.
>> >> >
>> >> > I tried the session:recordFile(-) via Lua
>> script in
>> >> three ways:
>> >> >
>> >> > 1. run lua script "test_outcall_ivr.lua" at
>> freeswitch
>> >> command-line:
>> >> >
>> >> > luarun test_outcall_ivr.lua
>> >> >
>> >> >
>> >> > -- [test_outcall_ivr.lua]
>> >> > {
>> >> > local sessionx =
>> >>
>> freeswitch.Session("sofia/gateway/sip.tpad.com/1726011",
>> >> session)
>> >> >
>> >> > -- Set the path separator
>> >> > pathsep = '/'
>> >> >
>> >> > -- Windows users do this instead:
>> >> > -- pathsep = '\'
>> >> >
>> >> > -- Answer the call
>> >> > -- sessionx:answer()
>> >> >
>> >> > --Create a string with path and filename of a
>> sound
>> >> file
>> >> > prompt = "ivr" .. pathsep ..
>> >> "ivr-welcome_to_freeswitch.wav"
>> >> >
>> >> > -- Print a log message
>> >> > freeswitch.consoleLog("INFO","Prompt file is
>> '" ..
>> >> prompt .. "'\n")
>> >> >
>> >> > --Play the prompt
>> >> > sessionx:streamFile(prompt)
>> >> >
>> >> > -- Record record file
>> >> >
>> >>
>> sessionx:streamFile("phrase:voicemail_record_message")
>> >> >
>> >> > -- Play a ""bong"" tone prior to recording
>> >> >
>> >>
>> sessionx:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
>> >> 0, 640)")
>> >> >
>> >> > -- record a message
>> >> > filename = sessionx:getVariable('sounds_dir')
>> ..
>> >> pathsep .. "123.wav"
>> >> > sessionx:recordFile(filename,300,100,10)
>> >> >
>> >> > -- play back the recorded msg
>> >> > sessionx:streamFile(filename)
>> >> >
>> >> > -- Hangup
>> >> > sessionx:hangup()
>> >> >
>> >> > }
>> >> >
>> >> > 2. I also tried it differently by submitting
>> the
>> >> following commands at the FreeSwitch command-line
>> >> interface:
>> >> >
>> >> > originate user/1005 &transfer(8887 xml
>> default)
>> >> >
>> >> > originate user/1005 &lua('test1.lua')
>> >> >
>> >> > originate sofia/gateway/sip.tpad.com/1726011
>> >> &lua('test1.lua')
>> >> >
>> >> >
>> >> > -- [test1.lua]
>> >> > {
>> >> > -- Set the path separator
>> >> > pathsep = '/'
>> >> >
>> >> > -- Windows users do this instead:
>> >> > -- pathsep = '\'
>> >> >
>> >> > --Answer the call
>> >> > session:answer()
>> >> >
>> >> > --Create a string with path and filename of a
>> sound
>> >> file
>> >> > prompt = "ivr" .. pathsep ..
>> >> "ivr-welcome_to_freeswitch.wav"
>> >> >
>> >> > -- Print a log message
>> >> > freeswitch.consoleLog("INFO","Prompt file is
>> '" ..
>> >> prompt .. "'\n")
>> >> >
>> >> > --Play the prompt
>> >> > session:streamFile(prompt)
>> >> >
>> >> > -- Record record file
>> >> >
>> session:streamFile("phrase:voicemail_record_message")
>> >> >
>> >> > -- Play a ""bong"" tone prior to recording
>> >> >
>> >>
>> session:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
>> >> 0, 640)")
>> >> >
>> >> > -- record a message
>> >> > filename = session:getVariable('sounds_dir')
>> ..
>> >> pathsep .. "123.wav"
>> >> > session:recordFile(filename,300,100,10)
>> >> >
>> >> > -- play back the recorded msg
>> >> > session:streamFile(filename)
>> >> >
>> >> > -- Hangup
>> >> > session:hangup()
>> >> > }
>> >> >
>> >> > -- [xml dialplan for extension 8887]:
>> >> > {
>> >> > ? ? ? ?<extension name="Simple Lua
>> Test">
>> >> > ? ? ? ? ? ? ? ?<condition
>> >> field="destination_number"
>> expression="^(8887)$">
>> >> >
>> ?<action
>> >> application="set"
>> data="record_waste_resources=true"/>
>> >> >
>> ?<action
>> >> application="lua" data="test1.lua"/>
>> >> > ? ? ? ? ? ? ? ?</condition>
>> >> > ? ? ? ?</extension>
>> >> > }
>> >> >
>> >> >
>> >>
>> ======================================================
>> >> >
>> >> > For all the above testing cases,
>> session:recordFile(-)
>> >> always creates an empty wav file for each of
>> outbound IVR
>> >> calls, however, if I make an inbound IVR?call to
>> run Lua
>> >> script "test1.lua", session:recordFile(-) always
>> works
>> >> perfect to generate a normal wav file.
>> >> >
>> >> > So, what's wrong with [session:recordFile(-)]
>> during
>> >> an outbound IVR call?
>> >> >
>> >> > x.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
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> 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
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Sep  1 19:56:54 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 1 Sep 2011 08:56:54 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] Investigating build testing
 for FreeSWITCH
In-Reply-To: <1314844040696-6748773.post@n2.nabble.com>
References: <CAKzWOxX5w7E4JFRRgKwmwXH-6YsdeHANO4aZha91V1_Y+79SyQ@mail.gmail.com>
	<CABH8QX3QFdP6eqXdxWecMMW8GWJhS+foz0hL9LYAMYq6jw-cig@mail.gmail.com>
	<1314844040696-6748773.post@n2.nabble.com>
Message-ID: <CAKzWOxVS1RaNC2VLyQ2SdJgz+k3xuv-eoOw_jjkFbp-67ByQUg@mail.gmail.com>

Okay, so we have two different integration servers, one for Linux/Unix and
one for Windows. That sounds like a pretty decent setup to me. The only
improvement I could think of is if we could have a "unified" system that
collected the tests for all platforms that FreeSWITCH supports and put the
results online so we could review them. Is that asking too much? :)

-MC

On Wed, Aug 31, 2011 at 7:27 PM, Jeff Lenk <jeff at jefflenk.com> wrote:

> cypromis runs a continous integration server (hudson) which is a very
> popular
> framework written in Java. He has done some cool things with it such as IRC
> notifications of build problems to the dev channel and lots more. Maybe he
> will pipe in here. I also run a continous integration server for the
> windows
> builds (CruiseControl.Net) of which I receive email notifications of build
> related problems if they occur for windows.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Investigating-build-testing-for-FreeSWITCH-tp6748651p6748773.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/96436a28/attachment.html 

From jeff at jefflenk.com  Thu Sep  1 20:31:05 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 1 Sep 2011 09:31:05 -0700 (PDT)
Subject: [Freeswitch-users] [Freeswitch-dev] Investigating build testing
	for FreeSWITCH
In-Reply-To: <CAKzWOxVS1RaNC2VLyQ2SdJgz+k3xuv-eoOw_jjkFbp-67ByQUg@mail.gmail.com>
References: <CAKzWOxX5w7E4JFRRgKwmwXH-6YsdeHANO4aZha91V1_Y+79SyQ@mail.gmail.com>
	<CABH8QX3QFdP6eqXdxWecMMW8GWJhS+foz0hL9LYAMYq6jw-cig@mail.gmail.com>
	<1314844040696-6748773.post@n2.nabble.com>
	<CAKzWOxVS1RaNC2VLyQ2SdJgz+k3xuv-eoOw_jjkFbp-67ByQUg@mail.gmail.com>
Message-ID: <1314894665780-6750588.post@n2.nabble.com>

Nope Its a good idea. I have been meaning to look into the hudson extension
for windows visual studio but just have never gotten around to it.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Investigating-build-testing-for-FreeSWITCH-tp6748651p6750588.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From all.eforums at gmail.com  Thu Sep  1 20:44:18 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Thu, 1 Sep 2011 12:44:18 -0400
Subject: [Freeswitch-users] Reorder tone immediately on Cisco 7960 Line 1
	upon off-hook
Message-ID: <CADzY+JhcxtCLZ4AH6tAyCBb0uboTCX+TZyHe3ZvBzc=f9Qyi9g@mail.gmail.com>

Hi Guys,

I'm pretty sure this is a Cisco firmware problem (v 8.9) problem but just
wanted to check if anyone else has seen this.

With Line 1 on the Cisco 7960 registered to FS, when the line buttons is
pressed or the "Speaker" button is pressed, it immediately gets a re-order
tone instead of a dial-tone. Watching the console on FS, I see an Invite
going to "%14" which obv doesn't exist.

   INVITE sip:%14 at 192.168.3.80 SIP/2.0
   Via: SIP/2.0/UDP 69.201.146.xxx:5062;branch=z9hG4bK121399ed
   From: "user1" <sip:user1 at 192.168.3.80
>;tag=003094c3e08b02705d7d5000-560be9da
   To: <sip:%14 at 192.168.3.80>
   Call-ID: 003094c3-e08b000a-43fa29c5-5d1840e9 at 69.201.146.xxx
   Max-Forwards: 70
   Date: Thu, 01 Sep 2011 16:39:30 GMT
   CSeq: 101 INVITE
   User-Agent: Cisco-CP7960G/8.0
   Contact: <sip:user1 at 69.201.146.xxx:5062;transport=udp>
   Expires: 180
   Accept: application/sdp
   Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
   Supported: replaces,join,norefersub
   Content-Length: 278
   Content-Type: application/sdp
   Content-Disposition: session;handling=optional

   v=0
   o=Cisco-SIPUA 12375 0 IN IP4 69.201.146.xxx
   s=SIP Call
   t=0 0
   m=audio 19982 RTP/AVP 0 8 18 101
   c=IN IP4 69.201.146.xxx
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:18 G729/8000
   a=fmtp:18 annexb=no
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=sendrecv

When this same account is configured on any other line, pressing the line
button brings up a dial-tone.

Anyone have any idea why it does that or how to make it not do it?

Cheers
aeg
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/1438ad3c/attachment-0001.html 

From jeff at jefflenk.com  Thu Sep  1 21:02:26 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 1 Sep 2011 10:02:26 -0700 (PDT)
Subject: [Freeswitch-users] recordFile max_len not obeyed
In-Reply-To: <CAAcoRq1_NFPDpUgXMpwazN165pO7tKBP+y8X03bBjVi+pyY-CQ@mail.gmail.com>
References: <CAAcoRq1_NFPDpUgXMpwazN165pO7tKBP+y8X03bBjVi+pyY-CQ@mail.gmail.com>
Message-ID: <1314896546108-6750754.post@n2.nabble.com>

try 10 instead that value is in seconds :)

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/recordFile-max-len-not-obeyed-tp6749371p6750754.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From curriegrad2004 at gmail.com  Thu Sep  1 21:04:53 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 1 Sep 2011 10:04:53 -0700
Subject: [Freeswitch-users] freeswitch-1.0.7 Conference set auto outcall
	bug?
In-Reply-To: <007c01cc6845$e0d3c9b0$a27b5d10$@gmail.com>
References: <007c01cc6845$e0d3c9b0$a27b5d10$@gmail.com>
Message-ID: <CAMWiMy_9utyCkRwCvh+-3QVqr_+iJTYqHQS9a-iOsFH2Sju_3Q@mail.gmail.com>

Did you run a make moh-install after you installed FS?

On Wed, Aug 31, 2011 at 6:24 PM, qiaoqiao7036 <qiaoqiao7036 at gmail.com> wrote:
> Hi,
>
> ? I have used the function of conference set auto outcall in freeswitch
> 1.0.6, it is very good,
>
> several days ago, I installed a new freeswitch with git, the version is
> 1.0.7, I wrote a very simple
>
> dialplan to test the conference auto outcall function, and the problem
> happened:
>
>
>
> This is my dialplan, I have tested it in my freeswitch 1.0.6 successfully:
>
>
>
> ?? <extension name="Demonstrate conference_set_auto_outcall">
>
> ???? <condition field="destination_number" expression="^(12345)$">
>
>
>
> ?????? <action application="answer"/>
>
>
>
> ? ?????<action application="set" data="conference_auto_outcall_timeout=5"/>
>
> ?????? <action application="set" data="conference_auto_outcall_flags=none"/>
>
> ?????? <action application="set"
> data="conference_auto_outcall_caller_id_name=$${ caller_id_name}"/>
>
> ?????? <action application="set"
> data="conference_auto_outcall_caller_id_number=$${ caller_id_number}"/>
>
> ?????? <action application="set"
> data="conference_auto_outcall_profile=default"/>
>
>
>
> ?????? <action application="conference_set_auto_outcall"
> data="[call_timeout=15]user/1001@$${domain}"/>
>
> ?????? <action application="conference_set_auto_outcall"
> data="[call_timeout=15]user/1002@$${domain}"/>
>
>
>
> ?????? <action application="conference" data="$1 at default"/>
>
>
>
> ???? </condition>
>
> ?? </extension>
>
>
>
> Problem 1:? When I use a softphone to dial number 12345, in my old
> freeswitch I can hear the moh music when the phone 1001 and 1002 is ring,
>
> But in freeswitch 1.0.7 I can?t hear anything, and I found the debug
> informations in freeswitch cli:
>
>
>
> 2011-08-31 18:59:14.520108 [DEBUG] switch_core_session.c:1156 Send signal
> sofia/internal/sip:1001 at 192.168.2.226:5060 [BREAK]
>
> 2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:369
> (sofia/internal/sip:1001 at 192.168.2.226:5060) State INIT going to sleep
>
> 2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:330
> (sofia/internal/sip:1001 at 192.168.2.226:5060) Running State Change CS_ROUTING
>
> 2011-08-31 18:59:14.520108 [DEBUG] switch_channel.c:1828
> (sofia/internal/sip:1001 at 192.168.2.226:5060) Callstate Change DOWN ->
> RINGING
>
> 2011-08-31 18:59:14.520108 [DEBUG] switch_core_session.c:857 Send signal
> sofia/internal/sip:1001 at 192.168.2.226:5060 [BREAK]
>
> 2011-08-31 18:59:14.520108 [DEBUG] sofia.c:5128 Channel
> sofia/internal/sip:1001 at 192.168.2.226:5060 entering state [calling][0]
>
> 2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:378
> (sofia/internal/sip:1001 at 192.168.2.226:5060) State ROUTING
>
> 2011-08-31 18:59:14.520108 [DEBUG] mod_sofia.c:148
> sofia/internal/sip:1001 at 192.168.2.226:5060 SOFIA ROUTING
>
> 2011-08-31 18:59:14.520108 [DEBUG] switch_ivr_originate.c:66
> (sofia/internal/sip:1001 at 192.168.2.226:5060) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
>
> 2011-08-31 18:59:14.520108 [DEBUG] switch_core_session.c:1156 Send signal
> sofia/internal/sip:1001 at 192.168.2.226:5060 [BREAK]
>
> 2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:378
> (sofia/internal/sip:1001 at 192.168.2.226:5060) State ROUTING going to sleep
>
> 2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:330
> (sofia/internal/sip:1001 at 192.168.2.226:5060) Running State Change
> CS_CONSUME_MEDIA
>
> 2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:397
> (sofia/internal/sip:1001 at 192.168.2.226:5060) State CONSUME_MEDIA
>
> 2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:397
> (sofia/internal/sip:1001 at 192.168.2.226:5060) State CONSUME_MEDIA going to
> sleep
>
> 2011-08-31 18:59:14.540080 [DEBUG] mod_local_stream.c:421 Opening Stream
> [moh/8000] 8000hz
>
> 2011-08-31 18:59:14.540080 [DEBUG] switch_core_session.c:857 Send signal
> sofia/internal/1000 at 192.168.2.242 [BREAK]
>
> 2011-08-31 18:59:14.540080 [DEBUG] switch_core_session.c:857 Send signal
> sofia/internal/1000 at 192.168.2.242 [BREAK]
>
> 2011-08-31 18:59:14.540080 [DEBUG] switch_core_session.c:857 Send signal
> sofia/internal/1000 at 192.168.2.242 [BREAK]
>
> 2011-08-31 18:59:14.559978 [DEBUG] sofia.c:5128 Channel
> sofia/internal/1000 at 192.168.2.242 entering state [ready][200]
>
>
>
> ?????? That mean the freeswitch is sending the moh music to the caller 1000,
> is that right? But I can?t hear anything.
>
>
>
> ?????? Problem 2: when just one phone pickup, like phone 1001, and phone
> 1002 don?t, (I use phone 1000 to call number 12345) I can?t hear anything
> from phone 1001 when 1002 is ringing except phone 1002 pick up or when the
> phone 1002 is timeout to ring, it?s very stranger!!! I tested this situation
> in freeswitch 1.0.6, when just only person pickup the phone the conference
> will work, 1000 will hear the phone 1001?s voice, don?t need the 1002 to
> pickup or timeout.
>
>
>
> By the way: I have tested this with the newest version: FreeSWITCH Version
> 1.0.head (git-6a5f6e5 2011-08-30 15-00-07 -0500)
>
> The problem is still exist.
>
> Thank you for advising me how to solve this problem.
>
>
>
>
>
> Regards!
>
> Dennis
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Sep  1 21:09:12 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 1 Sep 2011 10:09:12 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] Investigating build testing
 for FreeSWITCH
In-Reply-To: <1314894665780-6750588.post@n2.nabble.com>
References: <CAKzWOxX5w7E4JFRRgKwmwXH-6YsdeHANO4aZha91V1_Y+79SyQ@mail.gmail.com>
	<CABH8QX3QFdP6eqXdxWecMMW8GWJhS+foz0hL9LYAMYq6jw-cig@mail.gmail.com>
	<1314844040696-6748773.post@n2.nabble.com>
	<CAKzWOxVS1RaNC2VLyQ2SdJgz+k3xuv-eoOw_jjkFbp-67ByQUg@mail.gmail.com>
	<1314894665780-6750588.post@n2.nabble.com>
Message-ID: <CAKzWOxUgUi5M4mMX-bK5+8=UpDDPK=R79Owc2z8Q8PNk8ggCaw@mail.gmail.com>

No worries. If people whine about Windows "not being a fully supported
platform" just remind them that we have limited resources and that they are
welcome to donate time/money/energy/components. :)

-MC

On Thu, Sep 1, 2011 at 9:31 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

> Nope Its a good idea. I have been meaning to look into the hudson extension
> for windows visual studio but just have never gotten around to it.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Investigating-build-testing-for-FreeSWITCH-tp6748651p6750588.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/d0359fce/attachment.html 

From msc at freeswitch.org  Thu Sep  1 21:11:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 1 Sep 2011 10:11:38 -0700
Subject: [Freeswitch-users] recordFile max_len not obeyed
In-Reply-To: <1314896546108-6750754.post@n2.nabble.com>
References: <CAAcoRq1_NFPDpUgXMpwazN165pO7tKBP+y8X03bBjVi+pyY-CQ@mail.gmail.com>
	<1314896546108-6750754.post@n2.nabble.com>
Message-ID: <CAKzWOxXuyF56j_1bE+OxG3tc492gGRGJg7jtst9SsTd7eS+QXQ@mail.gmail.com>

Good catch! I don't think he wanted a time limit of 2 hours and 46 minutes
on that recording. ;)
-MC

On Thu, Sep 1, 2011 at 10:02 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

> try 10 instead that value is in seconds :)
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/8554e55b/attachment.html 

From jeff at jefflenk.com  Thu Sep  1 21:25:51 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 1 Sep 2011 10:25:51 -0700 (PDT)
Subject: [Freeswitch-users] [Freeswitch-dev] Investigating build testing
	for FreeSWITCH
In-Reply-To: <CAKzWOxUgUi5M4mMX-bK5+8=UpDDPK=R79Owc2z8Q8PNk8ggCaw@mail.gmail.com>
References: <CAKzWOxX5w7E4JFRRgKwmwXH-6YsdeHANO4aZha91V1_Y+79SyQ@mail.gmail.com>
	<CABH8QX3QFdP6eqXdxWecMMW8GWJhS+foz0hL9LYAMYq6jw-cig@mail.gmail.com>
	<1314844040696-6748773.post@n2.nabble.com>
	<CAKzWOxVS1RaNC2VLyQ2SdJgz+k3xuv-eoOw_jjkFbp-67ByQUg@mail.gmail.com>
	<1314894665780-6750588.post@n2.nabble.com>
	<CAKzWOxUgUi5M4mMX-bK5+8=UpDDPK=R79Owc2z8Q8PNk8ggCaw@mail.gmail.com>
Message-ID: <1314897951581-6750871.post@n2.nabble.com>

If there was a hosted virtual machine with windows 2008 r2 available I
certainly could make available the build status of the windows stuff as it
is today. I dont have the resources available to host this myself right now.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Investigating-build-testing-for-FreeSWITCH-tp6748651p6750871.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Thu Sep  1 21:30:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 1 Sep 2011 10:30:58 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] Investigating build testing
 for FreeSWITCH
In-Reply-To: <1314897951581-6750871.post@n2.nabble.com>
References: <CAKzWOxX5w7E4JFRRgKwmwXH-6YsdeHANO4aZha91V1_Y+79SyQ@mail.gmail.com>
	<CABH8QX3QFdP6eqXdxWecMMW8GWJhS+foz0hL9LYAMYq6jw-cig@mail.gmail.com>
	<1314844040696-6748773.post@n2.nabble.com>
	<CAKzWOxVS1RaNC2VLyQ2SdJgz+k3xuv-eoOw_jjkFbp-67ByQUg@mail.gmail.com>
	<1314894665780-6750588.post@n2.nabble.com>
	<CAKzWOxUgUi5M4mMX-bK5+8=UpDDPK=R79Owc2z8Q8PNk8ggCaw@mail.gmail.com>
	<1314897951581-6750871.post@n2.nabble.com>
Message-ID: <CAKzWOxWY-CZBfUJwroGGoxRJSa1Kw=DztXSMFq2X7xfQ+aaJdQ@mail.gmail.com>

Understood. On next week's conf call we can discuss options.
-MC

On Thu, Sep 1, 2011 at 10:25 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

> If there was a hosted virtual machine with windows 2008 r2 available I
> certainly could make available the build status of the windows stuff as it
> is today. I dont have the resources available to host this myself right
> now.
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/7904f59a/attachment.html 

From xing2kin at yahoo.com  Thu Sep  1 21:38:47 2011
From: xing2kin at yahoo.com (king2kin)
Date: Thu, 1 Sep 2011 10:38:47 -0700 (PDT)
Subject: [Freeswitch-users] session:recordFile(-) always creates empty
	wav file during outbound IVR call
In-Reply-To: <CAKzWOxVB8s+UpVQ58_7gcOA2sZqfPtf3e6_r+a2eu2KBva+fwQ@mail.gmail.com>
Message-ID: <1314898727.83719.YahooMailClassic@web39702.mail.mud.yahoo.com>

I tried it again in the following three cases of outbound ivr calls according to you and anthony's advice, I get the same result (i.e. record empty wav file with its size 68 bytes),?nothing seems to be improved.
For inbound ivr call, application 'record' never has a problem!
?
originate {record_waste_resources=true, ignore_early_media=true}sofia/gateway/mygateway/1726011? 8884
?
originate {record_waste_resources=true}sofia/gateway/mygateway/1726011? 8884

originate {ignore_early_media=true}sofia/gateway/mygateway/1726011? 8884
?
-- Here?is the?xml dialplan for extension 8884:
?
?<extension name="Play and Record Sound File">
??<condition field="destination_number" expression="^(8884)$">
???<action application="answer"/>
???<action application="sleep" data="1000"/>
???<action application="playback" data="ivr/ivr-thank_you.wav"/>?
???
???<action application="record" data="C:/c4dev/freeswitch/Debug/sounds/test.wav 20 200"/>
???
???<action application="sleep" data="1000"/>
???<action application="playback" data="voicemail/vm-goodbye.wav"/>?
???
???<action application="sleep" data="1000"/>??
???<action application="hangup"/>???
??</condition>
?</extension>

--- On Thu, 9/1/11, Michael Collins <msc at freeswitch.org> wrote:


From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] session:recordFile(-) always creates empty wav file during outbound IVR call
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Thursday, September 1, 2011, 8:49 AM


Tony mentioned something that might be an issue for you. Try setting this on the channel:
http://wiki.freeswitch.org/wiki/Channel_Variables#record_waste_resources


Let us know if that makes a difference.


-MC


On Thu, Sep 1, 2011 at 2:48 AM, king2kin <xing2kin at yahoo.com> wrote:

Hi Anthony,

Actually FreeSwitch application "record" doesn't work either for outbound IVR call (see below), created an empty wav file with its size 68 bytes, the wav file doesn't contain any samples of audio data;

<action application="record" data="C:/c4dev/freeswitch/Debug/sounds/test.wav 20 200"/>

earlier I reported that "session:recordFile(-)" doesn't work inside Lua Script for any outbound IVR call.

- My CLI commands to make an outbound IVR call:

originate {ignore_early_media=true}sofia/gateway/mygateway/1726011 ?8884
or
originate user/1005 ? 8884

- xml diaplan for extension 8884:
{
? ? ? ?<extension name="Play and Record Sound File">
? ? ? ? ? ? ? ?<condition field="destination_number" expression="^(8884)$">
? ? ? ? ? ? ? ? ? ? ? ?<action application="answer"/>
? ? ? ? ? ? ? ? ? ? ? ?<action application="sleep" data="1000"/>
? ? ? ? ? ? ? ? ? ? ? ?<action application="playback" data="ivr/ivr-thank_you.wav"/>

? ? ? ? ? ? ? ? ? ? ? ?<action application="record" data="C:/c4dev/freeswitch/Debug/sounds/test.wav 20 200"/>

? ? ? ? ? ? ? ? ? ? ? ?<action application="sleep" data="1000"/>
? ? ? ? ? ? ? ? ? ? ? ?<action application="playback" data="voicemail/vm-goodbye.wav"/>

? ? ? ? ? ? ? ? ? ? ? ?<action application="sleep" data="1000"/>
? ? ? ? ? ? ? ? ? ? ? ?<action application="hangup"/>
? ? ? ? ? ? ? ?</condition>
? ? ? ?</extension>

}

--- On Wed, 8/31/11, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] session:recordFile(-) always creates empty wav file during outbound IVR call
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Wednesday, August 31, 2011, 7:46 PM



> Does the empty file contain silence
> that corresponds to the duration
> of the time it's recording?
> Are you producing the audio yourself into the recording and
> can you
> verify with a pcap that there is actually any audio to
> record?
>
>
> On Wed, Aug 31, 2011 at 9:21 PM, king2kin <xing2kin at yahoo.com>
> wrote:
> > Anthony,
> >
> > Thank you for help. I tried outbound IVR call in
> multiple ways again based on your advice,
> ?session:recordFile(-) still doesn't work normally, still
> created empty wav file .
> >
> > Could anyone please give me a hand on FreeSwitch
> Record? It always fails to record audio file during any
> outbound IVR call (auto dialer) although it works well
> during any inbound ivr call.
> >
> > Session:streamFile(-) works well to play back prompt
> files, dtmf keypress also works, ... during outbound ivr
> call.
> >
> > x.k.
> >
> > --- On Wed, 8/31/11, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
> >
> >> From: Anthony Minessale <anthony.minessale at gmail.com>
> >> Subject: Re: [Freeswitch-users]
> session:recordFile(-) always creates empty wav file during
> outbound IVR call
> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> >> Date: Wednesday, August 31, 2011, 12:17 PM
> >> try this dial string instead
> >>
> >>
> {ignore_early_media=true}sofia/gateway/sip.tpad.com/1726011
> >>
> >> On Wed, Aug 31, 2011 at 12:52 PM, king2kin <xing2kin at yahoo.com>
> >> wrote:
> >> > Hi folks,
> >> >
> >> > With Lua script and/or originate command, I
> have tried
> >> recording a message file during outbound IVR call
> over and
> >> over, session:recordFile(-) inside Lua script does
> create a
> >> wav file during each of my testings but the
> recorded audio
> >> file is always empty.
> >> >
> >> > However, session:recordFile(-) works well for
> inbound
> >> IVR call.
> >> >
> >> > I tried the session:recordFile(-) via Lua
> script in
> >> three ways:
> >> >
> >> > 1. run lua script "test_outcall_ivr.lua" at
> freeswitch
> >> command-line:
> >> >
> >> > luarun test_outcall_ivr.lua
> >> >
> >> >
> >> > -- [test_outcall_ivr.lua]
> >> > {
> >> > local sessionx =
> >>
> freeswitch.Session("sofia/gateway/sip.tpad.com/1726011",
> >> session)
> >> >
> >> > -- Set the path separator
> >> > pathsep = '/'
> >> >
> >> > -- Windows users do this instead:
> >> > -- pathsep = '\'
> >> >
> >> > -- Answer the call
> >> > -- sessionx:answer()
> >> >
> >> > --Create a string with path and filename of a
> sound
> >> file
> >> > prompt = "ivr" .. pathsep ..
> >> "ivr-welcome_to_freeswitch.wav"
> >> >
> >> > -- Print a log message
> >> > freeswitch.consoleLog("INFO","Prompt file is
> '" ..
> >> prompt .. "'\n")
> >> >
> >> > --Play the prompt
> >> > sessionx:streamFile(prompt)
> >> >
> >> > -- Record record file
> >> >
> >>
> sessionx:streamFile("phrase:voicemail_record_message")
> >> >
> >> > -- Play a ""bong"" tone prior to recording
> >> >
> >>
> sessionx:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
> >> 0, 640)")
> >> >
> >> > -- record a message
> >> > filename = sessionx:getVariable('sounds_dir')
> ..
> >> pathsep .. "123.wav"
> >> > sessionx:recordFile(filename,300,100,10)
> >> >
> >> > -- play back the recorded msg
> >> > sessionx:streamFile(filename)
> >> >
> >> > -- Hangup
> >> > sessionx:hangup()
> >> >
> >> > }
> >> >
> >> > 2. I also tried it differently by submitting
> the
> >> following commands at the FreeSwitch command-line
> >> interface:
> >> >
> >> > originate user/1005 &transfer(8887 xml
> default)
> >> >
> >> > originate user/1005 &lua('test1.lua')
> >> >
> >> > originate sofia/gateway/sip.tpad.com/1726011
> >> &lua('test1.lua')
> >> >
> >> >
> >> > -- [test1.lua]
> >> > {
> >> > -- Set the path separator
> >> > pathsep = '/'
> >> >
> >> > -- Windows users do this instead:
> >> > -- pathsep = '\'
> >> >
> >> > --Answer the call
> >> > session:answer()
> >> >
> >> > --Create a string with path and filename of a
> sound
> >> file
> >> > prompt = "ivr" .. pathsep ..
> >> "ivr-welcome_to_freeswitch.wav"
> >> >
> >> > -- Print a log message
> >> > freeswitch.consoleLog("INFO","Prompt file is
> '" ..
> >> prompt .. "'\n")
> >> >
> >> > --Play the prompt
> >> > session:streamFile(prompt)
> >> >
> >> > -- Record record file
> >> >
> session:streamFile("phrase:voicemail_record_message")
> >> >
> >> > -- Play a ""bong"" tone prior to recording
> >> >
> >>
> session:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
> >> 0, 640)")
> >> >
> >> > -- record a message
> >> > filename = session:getVariable('sounds_dir')
> ..
> >> pathsep .. "123.wav"
> >> > session:recordFile(filename,300,100,10)
> >> >
> >> > -- play back the recorded msg
> >> > session:streamFile(filename)
> >> >
> >> > -- Hangup
> >> > session:hangup()
> >> > }
> >> >
> >> > -- [xml dialplan for extension 8887]:
> >> > {
> >> > ? ? ? ?<extension name="Simple Lua
> Test">
> >> > ? ? ? ? ? ? ? ?<condition
> >> field="destination_number"
> expression="^(8887)$">
> >> > ? ? ? ? ? ? ? ? ? ? ?
> ?<action
> >> application="set"
> data="record_waste_resources=true"/>
> >> > ? ? ? ? ? ? ? ? ? ? ?
> ?<action
> >> application="lua" data="test1.lua"/>
> >> > ? ? ? ? ? ? ? ?</condition>
> >> > ? ? ? ?</extension>
> >> > }
> >> >
> >> >
> >>
> ======================================================
> >> >
> >> > For all the above testing cases,
> session:recordFile(-)
> >> always creates an empty wav file for each of
> outbound IVR
> >> calls, however, if I make an inbound IVR?call to
> run Lua
> >> script "test1.lua", session:recordFile(-) always
> works
> >> perfect to generate a normal wav file.
> >> >
> >> > So, what's wrong with [session:recordFile(-)]
> during
> >> an outbound IVR call?
> >> >
> >> > x.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
> >> >
> >>
> >>
> >>
> >> --
> >> 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
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


FreeSWITCH-users mailing list
FreeSWITCH-users at 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
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/e4d528f6/attachment-0001.html 

From xing2kin at yahoo.com  Thu Sep  1 21:45:17 2011
From: xing2kin at yahoo.com (king2kin)
Date: Thu, 1 Sep 2011 10:45:17 -0700 (PDT)
Subject: [Freeswitch-users] session:recordFile(-) always creates empty
	wav file during outbound IVR call
In-Reply-To: <CAKbxfG8vwn72DLzdwaTadAu0gzNoVgwRcham7vW49Z1kZRNcSA@mail.gmail.com>
Message-ID: <1314899117.16552.YahooMailClassic@web39707.mail.mud.yahoo.com>

I tried it again like you advise, but there is no difference. It's really a problem.

By the way, my dev system is windows 2003 server, SIP client is X-Lite,
and FS versions are GIT 1.0-head-2011-08-31 or GIT 1.0-head-2011-05-23.


--- On Thu, 9/1/11, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] session:recordFile(-) always creates empty wav file during outbound IVR call
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Thursday, September 1, 2011, 8:51 AM
> now try this:
> 
> {record_waste_resources=true,ignore_early_media=true}sofia/gateway/mygateway/1726011
> 
> My guess is that the other end of your call does not
> support
> asynchronous RTP and since we do not send media while
> recording its
> not either.
> 
> 
> 
> 
> On Thu, Sep 1, 2011 at 4:48 AM, king2kin <xing2kin at yahoo.com>
> wrote:
> > Hi Anthony,
> >
> > Actually FreeSwitch application "record" doesn't work
> either for outbound IVR call (see below), created an empty
> wav file with its size 68 bytes, the wav file doesn't
> contain any samples of audio data;
> >
> > <action application="record"
> data="C:/c4dev/freeswitch/Debug/sounds/test.wav 20
> 200"/>
> >
> > earlier I reported that "session:recordFile(-)"
> doesn't work inside Lua Script for any outbound IVR call.
> >
> > - My CLI commands to make an outbound IVR call:
> >
> > originate
> {ignore_early_media=true}sofia/gateway/mygateway/1726011
> ?8884
> > or
> > originate user/1005 ? 8884
> >
> > - xml diaplan for extension 8884:
> > {
> > ? ? ? ?<extension name="Play and Record Sound
> File">
> > ? ? ? ? ? ? ? ?<condition
> field="destination_number" expression="^(8884)$">
> > ? ? ? ? ? ? ? ? ? ? ? ?<action
> application="answer"/>
> > ? ? ? ? ? ? ? ? ? ? ? ?<action
> application="sleep" data="1000"/>
> > ? ? ? ? ? ? ? ? ? ? ? ?<action
> application="playback" data="ivr/ivr-thank_you.wav"/>
> >
> > ? ? ? ? ? ? ? ? ? ? ? ?<action
> application="record"
> data="C:/c4dev/freeswitch/Debug/sounds/test.wav 20
> 200"/>
> >
> > ? ? ? ? ? ? ? ? ? ? ? ?<action
> application="sleep" data="1000"/>
> > ? ? ? ? ? ? ? ? ? ? ? ?<action
> application="playback" data="voicemail/vm-goodbye.wav"/>
> >
> > ? ? ? ? ? ? ? ? ? ? ? ?<action
> application="sleep" data="1000"/>
> > ? ? ? ? ? ? ? ? ? ? ? ?<action
> application="hangup"/>
> > ? ? ? ? ? ? ? ?</condition>
> > ? ? ? ?</extension>
> > }
> >
> > --- On Wed, 8/31/11, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
> >
> >> From: Anthony Minessale <anthony.minessale at gmail.com>
> >> Subject: Re: [Freeswitch-users]
> session:recordFile(-) always creates empty wav file during
> outbound IVR call
> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> >> Date: Wednesday, August 31, 2011, 7:46 PM
> >> Does the empty file contain silence
> >> that corresponds to the duration
> >> of the time it's recording?
> >> Are you producing the audio yourself into the
> recording and
> >> can you
> >> verify with a pcap that there is actually any
> audio to
> >> record?
> >>
> >>
> >> On Wed, Aug 31, 2011 at 9:21 PM, king2kin <xing2kin at yahoo.com>
> >> wrote:
> >> > Anthony,
> >> >
> >> > Thank you for help. I tried outbound IVR call
> in
> >> multiple ways again based on your advice,
> >> ?session:recordFile(-) still doesn't work
> normally, still
> >> created empty wav file .
> >> >
> >> > Could anyone please give me a hand on
> FreeSwitch
> >> Record? It always fails to record audio file
> during any
> >> outbound IVR call (auto dialer) although it works
> well
> >> during any inbound ivr call.
> >> >
> >> > Session:streamFile(-) works well to play back
> prompt
> >> files, dtmf keypress also works, ... during
> outbound ivr
> >> call.
> >> >
> >> > x.k.
> >> >
> >> > --- On Wed, 8/31/11, Anthony Minessale <anthony.minessale at gmail.com>
> >> wrote:
> >> >
> >> >> From: Anthony Minessale <anthony.minessale at gmail.com>
> >> >> Subject: Re: [Freeswitch-users]
> >> session:recordFile(-) always creates empty wav
> file during
> >> outbound IVR call
> >> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> >> >> Date: Wednesday, August 31, 2011, 12:17
> PM
> >> >> try this dial string instead
> >> >>
> >> >>
> >>
> {ignore_early_media=true}sofia/gateway/sip.tpad.com/1726011
> >> >>
> >> >> On Wed, Aug 31, 2011 at 12:52 PM,
> king2kin <xing2kin at yahoo.com>
> >> >> wrote:
> >> >> > Hi folks,
> >> >> >
> >> >> > With Lua script and/or originate
> command, I
> >> have tried
> >> >> recording a message file during outbound
> IVR call
> >> over and
> >> >> over, session:recordFile(-) inside Lua
> script does
> >> create a
> >> >> wav file during each of my testings but
> the
> >> recorded audio
> >> >> file is always empty.
> >> >> >
> >> >> > However, session:recordFile(-) works
> well for
> >> inbound
> >> >> IVR call.
> >> >> >
> >> >> > I tried the session:recordFile(-)
> via Lua
> >> script in
> >> >> three ways:
> >> >> >
> >> >> > 1. run lua script
> "test_outcall_ivr.lua" at
> >> freeswitch
> >> >> command-line:
> >> >> >
> >> >> > luarun test_outcall_ivr.lua
> >> >> >
> >> >> >
> >> >> > -- [test_outcall_ivr.lua]
> >> >> > {
> >> >> > local sessionx =
> >> >>
> >>
> freeswitch.Session("sofia/gateway/sip.tpad.com/1726011",
> >> >> session)
> >> >> >
> >> >> > -- Set the path separator
> >> >> > pathsep = '/'
> >> >> >
> >> >> > -- Windows users do this instead:
> >> >> > -- pathsep = '\'
> >> >> >
> >> >> > -- Answer the call
> >> >> > -- sessionx:answer()
> >> >> >
> >> >> > --Create a string with path and
> filename of a
> >> sound
> >> >> file
> >> >> > prompt = "ivr" .. pathsep ..
> >> >> "ivr-welcome_to_freeswitch.wav"
> >> >> >
> >> >> > -- Print a log message
> >> >> > freeswitch.consoleLog("INFO","Prompt
> file is
> >> '" ..
> >> >> prompt .. "'\n")
> >> >> >
> >> >> > --Play the prompt
> >> >> > sessionx:streamFile(prompt)
> >> >> >
> >> >> > -- Record record file
> >> >> >
> >> >>
> >>
> sessionx:streamFile("phrase:voicemail_record_message")
> >> >> >
> >> >> > -- Play a ""bong"" tone prior to
> recording
> >> >> >
> >> >>
> >>
> sessionx:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
> >> >> 0, 640)")
> >> >> >
> >> >> > -- record a message
> >> >> > filename =
> sessionx:getVariable('sounds_dir')
> >> ..
> >> >> pathsep .. "123.wav"
> >> >> >
> sessionx:recordFile(filename,300,100,10)
> >> >> >
> >> >> > -- play back the recorded msg
> >> >> > sessionx:streamFile(filename)
> >> >> >
> >> >> > -- Hangup
> >> >> > sessionx:hangup()
> >> >> >
> >> >> > }
> >> >> >
> >> >> > 2. I also tried it differently by
> submitting
> >> the
> >> >> following commands at the FreeSwitch
> command-line
> >> >> interface:
> >> >> >
> >> >> > originate user/1005
> &transfer(8887 xml
> >> default)
> >> >> >
> >> >> > originate user/1005
> &lua('test1.lua')
> >> >> >
> >> >> > originate
> sofia/gateway/sip.tpad.com/1726011
> >> >> &lua('test1.lua')
> >> >> >
> >> >> >
> >> >> > -- [test1.lua]
> >> >> > {
> >> >> > -- Set the path separator
> >> >> > pathsep = '/'
> >> >> >
> >> >> > -- Windows users do this instead:
> >> >> > -- pathsep = '\'
> >> >> >
> >> >> > --Answer the call
> >> >> > session:answer()
> >> >> >
> >> >> > --Create a string with path and
> filename of a
> >> sound
> >> >> file
> >> >> > prompt = "ivr" .. pathsep ..
> >> >> "ivr-welcome_to_freeswitch.wav"
> >> >> >
> >> >> > -- Print a log message
> >> >> > freeswitch.consoleLog("INFO","Prompt
> file is
> >> '" ..
> >> >> prompt .. "'\n")
> >> >> >
> >> >> > --Play the prompt
> >> >> > session:streamFile(prompt)
> >> >> >
> >> >> > -- Record record file
> >> >> >
> >>
> session:streamFile("phrase:voicemail_record_message")
> >> >> >
> >> >> > -- Play a ""bong"" tone prior to
> recording
> >> >> >
> >> >>
> >>
> session:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
> >> >> 0, 640)")
> >> >> >
> >> >> > -- record a message
> >> >> > filename =
> session:getVariable('sounds_dir')
> >> ..
> >> >> pathsep .. "123.wav"
> >> >> >
> session:recordFile(filename,300,100,10)
> >> >> >
> >> >> > -- play back the recorded msg
> >> >> > session:streamFile(filename)
> >> >> >
> >> >> > -- Hangup
> >> >> > session:hangup()
> >> >> > }
> >> >> >
> >> >> > -- [xml dialplan for extension
> 8887]:
> >> >> > {
> >> >> > ? ? ? ?<extension
> name="Simple Lua
> >> Test">
> >> >> > ? ? ? ? ? ? ?
> ?<condition
> >> >> field="destination_number"
> >> expression="^(8887)$">
> >> >> >
> >> ?<action
> >> >> application="set"
> >> data="record_waste_resources=true"/>
> >> >> >
> >> ?<action
> >> >> application="lua" data="test1.lua"/>
> >> >> > ? ? ? ? ? ? ?
> ?</condition>
> >> >> > ? ? ? ?</extension>
> >> >> > }
> >> >> >
> >> >> >
> >> >>
> >>
> ======================================================
> >> >> >
> >> >> > For all the above testing cases,
> >> session:recordFile(-)
> >> >> always creates an empty wav file for each
> of
> >> outbound IVR
> >> >> calls, however, if I make an inbound
> IVR?call to
> >> run Lua
> >> >> script "test1.lua", session:recordFile(-)
> always
> >> works
> >> >> perfect to generate a normal wav file.
> >> >> >
> >> >> > So, what's wrong with
> [session:recordFile(-)]
> >> during
> >> >> an outbound IVR call?
> >> >> >
> >> >> > x.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
> >> >> >
> >> >>
> >> >>
> >> >>
> >> >> --
> >> >> 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
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 anthony.minessale at gmail.com  Thu Sep  1 21:55:27 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 1 Sep 2011 12:55:27 -0500
Subject: [Freeswitch-users] FreeSwitch and core dumps
In-Reply-To: <CAFv7ub0crGLj13WSu8s0zEgP759zGP=bFFaVjwMtx0X=oKZ30g@mail.gmail.com>
References: <CAFv7ub1ciPKA9ceWcJrUx3fQROXrAxsdDyc0Q355gRB-ZtU52g@mail.gmail.com>
	<1314884710492-6750074.post@n2.nabble.com>
	<CAFv7ub0crGLj13WSu8s0zEgP759zGP=bFFaVjwMtx0X=oKZ30g@mail.gmail.com>
Message-ID: <CAKbxfG-v+KUxvNDnjGrDbY3uOri29Vn14VObiGfOXZmwBek98Q@mail.gmail.com>

try a top level configure using the --without-libcurl argument so it
uses our libcurl in tree.



On Thu, Sep 1, 2011 at 8:57 AM, Tomasz Hyziak <tomasz at hyziak.pl> wrote:
> Hi Jeef.
>
> Yes, that's FreeSWITCH compiled from git version: 1.0.head
> (git-cd31633 2011-08-17 19-34-22 -0500).
>
> I use mod_xml_cdr for saving cdrs to database via webservice.
>
> What could I do with it ? Recompiling mod_xml_cdr or curl ?
>
> --
> pozdrawiam - Tomasz Hyziak
>
>
>
> 2011/9/1 Jeff Lenk <jeff at jefflenk.com>:
>> Is this with Git Head? It appears that something is wrong with curl.
>>
>> --
>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-and-core-dumps-tp6750013p6750074.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
>



-- 
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  Thu Sep  1 21:57:34 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 1 Sep 2011 12:57:34 -0500
Subject: [Freeswitch-users] session:recordFile(-) always creates empty
 wav file during outbound IVR call
In-Reply-To: <1314899117.16552.YahooMailClassic@web39707.mail.mud.yahoo.com>
References: <CAKbxfG8vwn72DLzdwaTadAu0gzNoVgwRcham7vW49Z1kZRNcSA@mail.gmail.com>
	<1314899117.16552.YahooMailClassic@web39707.mail.mud.yahoo.com>
Message-ID: <CAKbxfG9tytOXeyAKEEOzB-VsAyF-82zPDONfFk5qzj3OVgXTZw@mail.gmail.com>

Did you verify audio is going to the channel with a packet capture as I asked?
Most likely you are not getting any media into the channel due to some
issue so there is nothing to record.



On Thu, Sep 1, 2011 at 12:45 PM, king2kin <xing2kin at yahoo.com> wrote:
> I tried it again like you advise, but there is no difference. It's really a problem.
>
> By the way, my dev system is windows 2003 server, SIP client is X-Lite,
> and FS versions are GIT 1.0-head-2011-08-31 or GIT 1.0-head-2011-05-23.
>
>
> --- On Thu, 9/1/11, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> Subject: Re: [Freeswitch-users] session:recordFile(-) always creates empty wav file during outbound IVR call
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Date: Thursday, September 1, 2011, 8:51 AM
>> now try this:
>>
>> {record_waste_resources=true,ignore_early_media=true}sofia/gateway/mygateway/1726011
>>
>> My guess is that the other end of your call does not
>> support
>> asynchronous RTP and since we do not send media while
>> recording its
>> not either.
>>
>>
>>
>>
>> On Thu, Sep 1, 2011 at 4:48 AM, king2kin <xing2kin at yahoo.com>
>> wrote:
>> > Hi Anthony,
>> >
>> > Actually FreeSwitch application "record" doesn't work
>> either for outbound IVR call (see below), created an empty
>> wav file with its size 68 bytes, the wav file doesn't
>> contain any samples of audio data;
>> >
>> > <action application="record"
>> data="C:/c4dev/freeswitch/Debug/sounds/test.wav 20
>> 200"/>
>> >
>> > earlier I reported that "session:recordFile(-)"
>> doesn't work inside Lua Script for any outbound IVR call.
>> >
>> > - My CLI commands to make an outbound IVR call:
>> >
>> > originate
>> {ignore_early_media=true}sofia/gateway/mygateway/1726011
>> ?8884
>> > or
>> > originate user/1005 ? 8884
>> >
>> > - xml diaplan for extension 8884:
>> > {
>> > ? ? ? ?<extension name="Play and Record Sound
>> File">
>> > ? ? ? ? ? ? ? ?<condition
>> field="destination_number" expression="^(8884)$">
>> > ? ? ? ? ? ? ? ? ? ? ? ?<action
>> application="answer"/>
>> > ? ? ? ? ? ? ? ? ? ? ? ?<action
>> application="sleep" data="1000"/>
>> > ? ? ? ? ? ? ? ? ? ? ? ?<action
>> application="playback" data="ivr/ivr-thank_you.wav"/>
>> >
>> > ? ? ? ? ? ? ? ? ? ? ? ?<action
>> application="record"
>> data="C:/c4dev/freeswitch/Debug/sounds/test.wav 20
>> 200"/>
>> >
>> > ? ? ? ? ? ? ? ? ? ? ? ?<action
>> application="sleep" data="1000"/>
>> > ? ? ? ? ? ? ? ? ? ? ? ?<action
>> application="playback" data="voicemail/vm-goodbye.wav"/>
>> >
>> > ? ? ? ? ? ? ? ? ? ? ? ?<action
>> application="sleep" data="1000"/>
>> > ? ? ? ? ? ? ? ? ? ? ? ?<action
>> application="hangup"/>
>> > ? ? ? ? ? ? ? ?</condition>
>> > ? ? ? ?</extension>
>> > }
>> >
>> > --- On Wed, 8/31/11, Anthony Minessale <anthony.minessale at gmail.com>
>> wrote:
>> >
>> >> From: Anthony Minessale <anthony.minessale at gmail.com>
>> >> Subject: Re: [Freeswitch-users]
>> session:recordFile(-) always creates empty wav file during
>> outbound IVR call
>> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> >> Date: Wednesday, August 31, 2011, 7:46 PM
>> >> Does the empty file contain silence
>> >> that corresponds to the duration
>> >> of the time it's recording?
>> >> Are you producing the audio yourself into the
>> recording and
>> >> can you
>> >> verify with a pcap that there is actually any
>> audio to
>> >> record?
>> >>
>> >>
>> >> On Wed, Aug 31, 2011 at 9:21 PM, king2kin <xing2kin at yahoo.com>
>> >> wrote:
>> >> > Anthony,
>> >> >
>> >> > Thank you for help. I tried outbound IVR call
>> in
>> >> multiple ways again based on your advice,
>> >> ?session:recordFile(-) still doesn't work
>> normally, still
>> >> created empty wav file .
>> >> >
>> >> > Could anyone please give me a hand on
>> FreeSwitch
>> >> Record? It always fails to record audio file
>> during any
>> >> outbound IVR call (auto dialer) although it works
>> well
>> >> during any inbound ivr call.
>> >> >
>> >> > Session:streamFile(-) works well to play back
>> prompt
>> >> files, dtmf keypress also works, ... during
>> outbound ivr
>> >> call.
>> >> >
>> >> > x.k.
>> >> >
>> >> > --- On Wed, 8/31/11, Anthony Minessale <anthony.minessale at gmail.com>
>> >> wrote:
>> >> >
>> >> >> From: Anthony Minessale <anthony.minessale at gmail.com>
>> >> >> Subject: Re: [Freeswitch-users]
>> >> session:recordFile(-) always creates empty wav
>> file during
>> >> outbound IVR call
>> >> >> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> >> >> Date: Wednesday, August 31, 2011, 12:17
>> PM
>> >> >> try this dial string instead
>> >> >>
>> >> >>
>> >>
>> {ignore_early_media=true}sofia/gateway/sip.tpad.com/1726011
>> >> >>
>> >> >> On Wed, Aug 31, 2011 at 12:52 PM,
>> king2kin <xing2kin at yahoo.com>
>> >> >> wrote:
>> >> >> > Hi folks,
>> >> >> >
>> >> >> > With Lua script and/or originate
>> command, I
>> >> have tried
>> >> >> recording a message file during outbound
>> IVR call
>> >> over and
>> >> >> over, session:recordFile(-) inside Lua
>> script does
>> >> create a
>> >> >> wav file during each of my testings but
>> the
>> >> recorded audio
>> >> >> file is always empty.
>> >> >> >
>> >> >> > However, session:recordFile(-) works
>> well for
>> >> inbound
>> >> >> IVR call.
>> >> >> >
>> >> >> > I tried the session:recordFile(-)
>> via Lua
>> >> script in
>> >> >> three ways:
>> >> >> >
>> >> >> > 1. run lua script
>> "test_outcall_ivr.lua" at
>> >> freeswitch
>> >> >> command-line:
>> >> >> >
>> >> >> > luarun test_outcall_ivr.lua
>> >> >> >
>> >> >> >
>> >> >> > -- [test_outcall_ivr.lua]
>> >> >> > {
>> >> >> > local sessionx =
>> >> >>
>> >>
>> freeswitch.Session("sofia/gateway/sip.tpad.com/1726011",
>> >> >> session)
>> >> >> >
>> >> >> > -- Set the path separator
>> >> >> > pathsep = '/'
>> >> >> >
>> >> >> > -- Windows users do this instead:
>> >> >> > -- pathsep = '\'
>> >> >> >
>> >> >> > -- Answer the call
>> >> >> > -- sessionx:answer()
>> >> >> >
>> >> >> > --Create a string with path and
>> filename of a
>> >> sound
>> >> >> file
>> >> >> > prompt = "ivr" .. pathsep ..
>> >> >> "ivr-welcome_to_freeswitch.wav"
>> >> >> >
>> >> >> > -- Print a log message
>> >> >> > freeswitch.consoleLog("INFO","Prompt
>> file is
>> >> '" ..
>> >> >> prompt .. "'\n")
>> >> >> >
>> >> >> > --Play the prompt
>> >> >> > sessionx:streamFile(prompt)
>> >> >> >
>> >> >> > -- Record record file
>> >> >> >
>> >> >>
>> >>
>> sessionx:streamFile("phrase:voicemail_record_message")
>> >> >> >
>> >> >> > -- Play a ""bong"" tone prior to
>> recording
>> >> >> >
>> >> >>
>> >>
>> sessionx:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
>> >> >> 0, 640)")
>> >> >> >
>> >> >> > -- record a message
>> >> >> > filename =
>> sessionx:getVariable('sounds_dir')
>> >> ..
>> >> >> pathsep .. "123.wav"
>> >> >> >
>> sessionx:recordFile(filename,300,100,10)
>> >> >> >
>> >> >> > -- play back the recorded msg
>> >> >> > sessionx:streamFile(filename)
>> >> >> >
>> >> >> > -- Hangup
>> >> >> > sessionx:hangup()
>> >> >> >
>> >> >> > }
>> >> >> >
>> >> >> > 2. I also tried it differently by
>> submitting
>> >> the
>> >> >> following commands at the FreeSwitch
>> command-line
>> >> >> interface:
>> >> >> >
>> >> >> > originate user/1005
>> &transfer(8887 xml
>> >> default)
>> >> >> >
>> >> >> > originate user/1005
>> &lua('test1.lua')
>> >> >> >
>> >> >> > originate
>> sofia/gateway/sip.tpad.com/1726011
>> >> >> &lua('test1.lua')
>> >> >> >
>> >> >> >
>> >> >> > -- [test1.lua]
>> >> >> > {
>> >> >> > -- Set the path separator
>> >> >> > pathsep = '/'
>> >> >> >
>> >> >> > -- Windows users do this instead:
>> >> >> > -- pathsep = '\'
>> >> >> >
>> >> >> > --Answer the call
>> >> >> > session:answer()
>> >> >> >
>> >> >> > --Create a string with path and
>> filename of a
>> >> sound
>> >> >> file
>> >> >> > prompt = "ivr" .. pathsep ..
>> >> >> "ivr-welcome_to_freeswitch.wav"
>> >> >> >
>> >> >> > -- Print a log message
>> >> >> > freeswitch.consoleLog("INFO","Prompt
>> file is
>> >> '" ..
>> >> >> prompt .. "'\n")
>> >> >> >
>> >> >> > --Play the prompt
>> >> >> > session:streamFile(prompt)
>> >> >> >
>> >> >> > -- Record record file
>> >> >> >
>> >>
>> session:streamFile("phrase:voicemail_record_message")
>> >> >> >
>> >> >> > -- Play a ""bong"" tone prior to
>> recording
>> >> >> >
>> >> >>
>> >>
>> session:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000,
>> >> >> 0, 640)")
>> >> >> >
>> >> >> > -- record a message
>> >> >> > filename =
>> session:getVariable('sounds_dir')
>> >> ..
>> >> >> pathsep .. "123.wav"
>> >> >> >
>> session:recordFile(filename,300,100,10)
>> >> >> >
>> >> >> > -- play back the recorded msg
>> >> >> > session:streamFile(filename)
>> >> >> >
>> >> >> > -- Hangup
>> >> >> > session:hangup()
>> >> >> > }
>> >> >> >
>> >> >> > -- [xml dialplan for extension
>> 8887]:
>> >> >> > {
>> >> >> > ? ? ? ?<extension
>> name="Simple Lua
>> >> Test">
>> >> >> >
>> ?<condition
>> >> >> field="destination_number"
>> >> expression="^(8887)$">
>> >> >> >
>> >> ?<action
>> >> >> application="set"
>> >> data="record_waste_resources=true"/>
>> >> >> >
>> >> ?<action
>> >> >> application="lua" data="test1.lua"/>
>> >> >> >
>> ?</condition>
>> >> >> > ? ? ? ?</extension>
>> >> >> > }
>> >> >> >
>> >> >> >
>> >> >>
>> >>
>> ======================================================
>> >> >> >
>> >> >> > For all the above testing cases,
>> >> session:recordFile(-)
>> >> >> always creates an empty wav file for each
>> of
>> >> outbound IVR
>> >> >> calls, however, if I make an inbound
>> IVR?call to
>> >> run Lua
>> >> >> script "test1.lua", session:recordFile(-)
>> always
>> >> works
>> >> >> perfect to generate a normal wav file.
>> >> >> >
>> >> >> > So, what's wrong with
>> [session:recordFile(-)]
>> >> during
>> >> >> an outbound IVR call?
>> >> >> >
>> >> >> > x.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
>> >> >> >
>> >> >>
>> >> >>
>> >> >>
>> >> >> --
>> >> >> 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
>> >>
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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 tomasz at hyziak.pl  Thu Sep  1 23:36:19 2011
From: tomasz at hyziak.pl (Tomasz Hyziak)
Date: Thu, 1 Sep 2011 21:36:19 +0200
Subject: [Freeswitch-users] FreeSwitch and core dumps
In-Reply-To: <CAKbxfG-v+KUxvNDnjGrDbY3uOri29Vn14VObiGfOXZmwBek98Q@mail.gmail.com>
References: <CAFv7ub1ciPKA9ceWcJrUx3fQROXrAxsdDyc0Q355gRB-ZtU52g@mail.gmail.com>
	<1314884710492-6750074.post@n2.nabble.com>
	<CAFv7ub0crGLj13WSu8s0zEgP759zGP=bFFaVjwMtx0X=oKZ30g@mail.gmail.com>
	<CAKbxfG-v+KUxvNDnjGrDbY3uOri29Vn14VObiGfOXZmwBek98Q@mail.gmail.com>
Message-ID: <CAFv7ub013xcpxyXWaGW-_5qmCAZ1DBQ5xXdZ5R-tRRBocu1i0w@mail.gmail.com>

Thanks Anthony for the info.

I'll run configure --without-libcurl and recompile FS... tomorrow will
be judgement day :)

I'll let you know.

--
pozdrawiam - Tomasz Hyziak



2011/9/1 Anthony Minessale <anthony.minessale at gmail.com>:
> try a top level configure using the --without-libcurl argument so it
> uses our libcurl in tree.
>
>
>
> On Thu, Sep 1, 2011 at 8:57 AM, Tomasz Hyziak <tomasz at hyziak.pl> wrote:
>> Hi Jeef.
>>
>> Yes, that's FreeSWITCH compiled from git version: 1.0.head
>> (git-cd31633 2011-08-17 19-34-22 -0500).
>>
>> I use mod_xml_cdr for saving cdrs to database via webservice.
>>
>> What could I do with it ? Recompiling mod_xml_cdr or curl ?
>>
>> --
>> pozdrawiam - Tomasz Hyziak
>>
>>
>>
>> 2011/9/1 Jeff Lenk <jeff at jefflenk.com>:
>>> Is this with Git Head? It appears that something is wrong with curl.
>>>
>>> --
>>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-and-core-dumps-tp6750013p6750074.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
>>
>
>
>
> --
> 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 msc at freeswitch.org  Fri Sep  2 03:09:57 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 1 Sep 2011 16:09:57 -0700
Subject: [Freeswitch-users] session:recordFile(-) always creates empty
 wav file during outbound IVR call
In-Reply-To: <CAKbxfG9tytOXeyAKEEOzB-VsAyF-82zPDONfFk5qzj3OVgXTZw@mail.gmail.com>
References: <CAKbxfG8vwn72DLzdwaTadAu0gzNoVgwRcham7vW49Z1kZRNcSA@mail.gmail.com>
	<1314899117.16552.YahooMailClassic@web39707.mail.mud.yahoo.com>
	<CAKbxfG9tytOXeyAKEEOzB-VsAyF-82zPDONfFk5qzj3OVgXTZw@mail.gmail.com>
Message-ID: <CAKzWOxXBCum7LqbCJ1-w9TsKKw_1EEB+RQyHAiZRY0ip7uZKfA@mail.gmail.com>

On Thu, Sep 1, 2011 at 10:57 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Did you verify audio is going to the channel with a packet capture as I
> asked?
> Most likely you are not getting any media into the channel due to some
> issue so there is nothing to record.
>
>
FYI, if you're new to the whole packet capture thing then check this out:

www.wireshark.org

Wireshark for Windows is really easy to use once you get the hang of it. The
trick will be learning how to filter out the stuff you don't want to
capture. The easiest thing to do there is use an IP address filter and plug
in the IP addr of the x-lite phone. Start the capture, make the test call,
hangup, then end the capture. Wireshark has some nice analysis tools as
well. If you have any questions on how to use it with VoIP captures then
google around or come join us in #freeswitch on irc.freenode.net.

Thanks,
MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/97b6fa58/attachment.html 

From neilp at cs.stanford.edu  Fri Sep  2 05:09:19 2011
From: neilp at cs.stanford.edu (Neil Patel)
Date: Fri, 2 Sep 2011 06:39:19 +0530
Subject: [Freeswitch-users] recordFile max_len not obeyed
In-Reply-To: <CAKzWOxXuyF56j_1bE+OxG3tc492gGRGJg7jtst9SsTd7eS+QXQ@mail.gmail.com>
References: <CAAcoRq1_NFPDpUgXMpwazN165pO7tKBP+y8X03bBjVi+pyY-CQ@mail.gmail.com>
	<1314896546108-6750754.post@n2.nabble.com>
	<CAKzWOxXuyF56j_1bE+OxG3tc492gGRGJg7jtst9SsTd7eS+QXQ@mail.gmail.com>
Message-ID: <CAAcoRq3k8oVeu4REgPH1SO-sceGB3kYS+=VEe8Tsk0CUieFoGA@mail.gmail.com>

Thanks! I've updated the documentation to make this more obvious.

-Neil

On Thu, Sep 1, 2011 at 10:41 PM, Michael Collins <msc at freeswitch.org> wrote:

> Good catch! I don't think he wanted a time limit of 2 hours and 46 minutes
> on that recording. ;)
> -MC
>
>
> On Thu, Sep 1, 2011 at 10:02 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>
>> try 10 instead that value is in seconds :)
>>
>>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110902/b6079df2/attachment.html 

From curriegrad2004 at gmail.com  Fri Sep  2 06:29:22 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 1 Sep 2011 19:29:22 -0700
Subject: [Freeswitch-users] FXO/FXS card advice
In-Reply-To: <CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com>
References: <4E5FA3B0.9030606@the800group.com>
	<CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com>
Message-ID: <CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com>

In the following order, most people usually recommend:

1. Sangoma
2. Diginum/OpenVox
3. Building your own Tormenta Zapatel Card (Only for serious
engineering types of people) from COTS

On Thu, Sep 1, 2011 at 8:50 AM, Michel Daggelinckx
<michel.daggelinckx at gmail.com> wrote:
> Digium and sangoma cards are high quality and work greast with FS
>
> Michel
>
>
> On Thu, Sep 1, 2011 at 5:24 PM, ocset <ocset at the800group.com> wrote:
>>
>> Hi
>>
>> As a new member of the forum, I am curious to know your experience with
>> FXO/FXS cards.
>>
>> I have a SPA3102 and have configured it to work with FS, but it feels a
>> bit like I am trying to make a square peg fit a round hole. I am hoping
>> to implement FS at a business which currently has 4 POTS lines and would
>> prefer to use an internal IDE card for the job of integrating these
>> phone lines into FreeSwitch.
>>
>> Has anyone got some advice on which cards I should be looking at that
>> just "work" with FS? What about echo cancellation - is that something I
>> should just cater for or does it depend on the client situation? What
>> about software cancellation?
>>
>> This seems like one of those times when too much choice is a bad thing
>> and I need some guidance on what has worked for you.
>>
>> All advice would be greatly appreciated.
>>
>> 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
>
>


From rupa at rupa.com  Fri Sep  2 06:35:43 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 1 Sep 2011 21:35:43 -0500
Subject: [Freeswitch-users] FreeSwitch and core dumps
In-Reply-To: <CAKbxfG-v+KUxvNDnjGrDbY3uOri29Vn14VObiGfOXZmwBek98Q@mail.gmail.com>
References: <CAFv7ub1ciPKA9ceWcJrUx3fQROXrAxsdDyc0Q355gRB-ZtU52g@mail.gmail.com>
	<1314884710492-6750074.post@n2.nabble.com>
	<CAFv7ub0crGLj13WSu8s0zEgP759zGP=bFFaVjwMtx0X=oKZ30g@mail.gmail.com>
	<CAKbxfG-v+KUxvNDnjGrDbY3uOri29Vn14VObiGfOXZmwBek98Q@mail.gmail.com>
Message-ID: <CAJw9H-MQKFrZ4N-mbfupCm-NiUAODbq4UQfeCTY70iZ4Bp6RGA@mail.gmail.com>

Be careful doing this on distributions that are pulling out sslv2.
Debian (testing) is has done this and so there are linkage errors at
module load time when libcurl pulls in ssl.

Also, while minor, the in-tree libcurl cannot timeout with a sub 1s
granularity.  Depending on your use case this may be an issue.

On Thu, Sep 1, 2011 at 12:55 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> try a top level configure using the --without-libcurl argument so it
> uses our libcurl in tree.
>
>
>
> On Thu, Sep 1, 2011 at 8:57 AM, Tomasz Hyziak <tomasz at hyziak.pl> wrote:
>> Hi Jeef.
>>
>> Yes, that's FreeSWITCH compiled from git version: 1.0.head
>> (git-cd31633 2011-08-17 19-34-22 -0500).
>>
>> I use mod_xml_cdr for saving cdrs to database via webservice.
>>
>> What could I do with it ? Recompiling mod_xml_cdr or curl ?
>>
>> --
>> pozdrawiam - Tomasz Hyziak
>>
>>
>>
>> 2011/9/1 Jeff Lenk <jeff at jefflenk.com>:
>>> Is this with Git Head? It appears that something is wrong with curl.
>>>
>>> --
>>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-and-core-dumps-tp6750013p6750074.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
>>
>
>
>
> --
> 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
>



-- 
-Rupa


From xing2kin at yahoo.com  Fri Sep  2 08:25:41 2011
From: xing2kin at yahoo.com (king2kin)
Date: Thu, 1 Sep 2011 21:25:41 -0700 (PDT)
Subject: [Freeswitch-users] session:recordFile(-) always creates empty
	wav file during outbound IVR call
In-Reply-To: <CAKzWOxXBCum7LqbCJ1-w9TsKKw_1EEB+RQyHAiZRY0ip7uZKfA@mail.gmail.com>
Message-ID: <1314937541.78565.YahooMailClassic@web39708.mail.mud.yahoo.com>

Michael and Anthony,
?
Thank you for your advice. I installed and run?the pcap tool 'wireshark' on the win 2003 where FS is running to capture packets during outbound IVR call to SIP client X-Lite (external or internal user); according to pcap packet info, it seems that X-Lite sent audio packets to FS during recording. Now I don't know how to fix this FS recording problem.
?
I post a bug report to 
http://jira.freeswitch.org/browse/FS-3536?
?
where I also uploaded/attached my pcap result data file. 
?
Please go to take a look at the details of the pcap result to see what's problem on FS recording.
?
Thanks again.
?
x.k.

--- On Thu, 9/1/11, Michael Collins <msc at freeswitch.org> wrote:


From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] session:recordFile(-) always creates empty wav file during outbound IVR call
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Thursday, September 1, 2011, 4:09 PM





On Thu, Sep 1, 2011 at 10:57 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

Did you verify audio is going to the channel with a packet capture as I asked?
Most likely you are not getting any media into the channel due to some
issue so there is nothing to record.








FYI, if you're new to the whole packet capture thing then check this out:


www.wireshark.org


Wireshark for Windows is really easy to use once you get the hang of it. The trick will be learning how to filter out the stuff you don't want to capture. The easiest thing to do there is use an IP address filter and plug in the IP addr of the x-lite phone. Start the capture, make the test call, hangup, then end the capture. Wireshark has some nice analysis tools as well. If you have any questions on how to use it with VoIP captures then google around or come join us in #freeswitch on irc.freenode.net.

Thanks,
MC?
-----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
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110901/6b5821e1/attachment-0001.html 

From codecomplete at free.fr  Fri Sep  2 14:30:06 2011
From: codecomplete at free.fr (GillesToo)
Date: Fri, 2 Sep 2011 03:30:06 -0700 (PDT)
Subject: [Freeswitch-users] Compiling Freeswitch for Android?
Message-ID: <1314959406122-6753409.post@n2.nabble.com>

Hello

Searching the archives only seems to return information on how to use
SIPDroid as an SIP client to connect to a Freeswitch server.

Since Android is built on Linux, and Freeswitch has been compiled for
compact devices, I was wondering if someone had tried compiling it for
Android? I could use a small IVR on a smartphone to handle incoming calls.

Thank you.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Compiling-Freeswitch-for-Android-tp6753409p6753409.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mrene_lists at avgs.ca  Fri Sep  2 14:55:52 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 2 Sep 2011 12:55:52 +0200
Subject: [Freeswitch-users] Compiling Freeswitch for Android?
In-Reply-To: <1314959406122-6753409.post@n2.nabble.com>
References: <1314959406122-6753409.post@n2.nabble.com>
Message-ID: <BD97C6DD-E80A-4CE8-8599-C5E851150454@avgs.ca>

Hi,

I am planning on doing an Android port but I'm currently busy on the iOS one. There will be some challenges since Android uses its own libc, so we will have to tweak a few things to make it work.

I think the best way to do this is to provide freeswitch as a native library so that it can be integrated in different projects, then have a softphone UI in plain java.

Let me know if you would be interested in helping out.

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




On 2011-09-02, at 12:30 PM, GillesToo wrote:

> Hello
> 
> Searching the archives only seems to return information on how to use
> SIPDroid as an SIP client to connect to a Freeswitch server.
> 
> Since Android is built on Linux, and Freeswitch has been compiled for
> compact devices, I was wondering if someone had tried compiling it for
> Android? I could use a small IVR on a smartphone to handle incoming calls.
> 
> Thank you.
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Compiling-Freeswitch-for-Android-tp6753409p6753409.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 codecomplete at free.fr  Fri Sep  2 15:28:33 2011
From: codecomplete at free.fr (GillesToo)
Date: Fri, 2 Sep 2011 04:28:33 -0700 (PDT)
Subject: [Freeswitch-users] Compiling Freeswitch for Android?
In-Reply-To: <BD97C6DD-E80A-4CE8-8599-C5E851150454@avgs.ca>
References: <1314959406122-6753409.post@n2.nabble.com>
	<BD97C6DD-E80A-4CE8-8599-C5E851150454@avgs.ca>
Message-ID: <1314962913219-6753523.post@n2.nabble.com>

Great news :-) Unfortunately, I don't have the technical skills to port
Freeswitch. Did you put up a web site so that we can keep an eye on the
project?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Compiling-Freeswitch-for-Android-tp6753409p6753523.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jmesquita at freeswitch.org  Fri Sep  2 15:54:48 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 2 Sep 2011 08:54:48 -0300
Subject: [Freeswitch-users] FXO/FXS card advice
In-Reply-To: <CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com>
References: <4E5FA3B0.9030606@the800group.com>
	<CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com>
	<CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com>
Message-ID: <CABH8QX0a-xGZnUmXRVJBrWfqirc0AMPfVpz3Jiw7HEp3304XUQ@mail.gmail.com>

Just don't forget about Khomp even tho there is no support for T1 or J1.

www.khomp.com

Regards,

JM

On Thursday, September 1, 2011, curriegrad2004 wrote:

> In the following order, most people usually recommend:
>
> 1. Sangoma
> 2. Diginum/OpenVox
> 3. Building your own Tormenta Zapatel Card (Only for serious
> engineering types of people) from COTS
>
> On Thu, Sep 1, 2011 at 8:50 AM, Michel Daggelinckx
> <michel.daggelinckx at gmail.com <javascript:;>> wrote:
> > Digium and sangoma cards are high quality and work greast with FS
> >
> > Michel
> >
> >
> > On Thu, Sep 1, 2011 at 5:24 PM, ocset <ocset at the800group.com<javascript:;>>
> wrote:
> >>
> >> Hi
> >>
> >> As a new member of the forum, I am curious to know your experience with
> >> FXO/FXS cards.
> >>
> >> I have a SPA3102 and have configured it to work with FS, but it feels a
> >> bit like I am trying to make a square peg fit a round hole. I am hoping
> >> to implement FS at a business which currently has 4 POTS lines and would
> >> prefer to use an internal IDE card for the job of integrating these
> >> phone lines into FreeSwitch.
> >>
> >> Has anyone got some advice on which cards I should be looking at that
> >> just "work" with FS? What about echo cancellation - is that something I
> >> should just cater for or does it depend on the client situation? What
> >> about software cancellation?
> >>
> >> This seems like one of those times when too much choice is a bad thing
> >> and I need some guidance on what has worked for you.
> >>
> >> All advice would be greatly appreciated.
> >>
> >> Regards
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


-- 
Jo?o Mesquita
FreeSWITCH? Solutions
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110902/21a95023/attachment.html 

From mrene_lists at avgs.ca  Fri Sep  2 17:23:58 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 2 Sep 2011 15:23:58 +0200
Subject: [Freeswitch-users] Compiling Freeswitch for Android?
In-Reply-To: <1314962913219-6753523.post@n2.nabble.com>
References: <1314959406122-6753409.post@n2.nabble.com>
	<BD97C6DD-E80A-4CE8-8599-C5E851150454@avgs.ca>
	<1314962913219-6753523.post@n2.nabble.com>
Message-ID: <296C71FF-C5F6-41E8-A08C-5DE3AE61300F@avgs.ca>

It'll be released with the main distribution :)

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




On 2011-09-02, at 1:28 PM, GillesToo wrote:

> Great news :-) Unfortunately, I don't have the technical skills to port
> Freeswitch. Did you put up a web site so that we can keep an eye on the
> project?
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Compiling-Freeswitch-for-Android-tp6753409p6753523.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 dgarcia at anew.com.ve  Fri Sep  2 17:39:28 2011
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Fri, 02 Sep 2011 09:09:28 -0430
Subject: [Freeswitch-users] FXO/FXS card advice
In-Reply-To: <CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com>
References: <4E5FA3B0.9030606@the800group.com>
	<CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com>
	<CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com>
Message-ID: <4E60DC90.9030201@anew.com.ve>

One think you should consider also, is the FXO/FXS board that you will 
buy, you have to check compatibility with your server/pc.

For example openvox models A800P and A1200P has an issue with 
motherboard based on chipset H55 for example.


On 9/1/2011 9:59 PM, curriegrad2004 wrote:
> In the following order, most people usually recommend:
>
> 1. Sangoma
> 2. Diginum/OpenVox
> 3. Building your own Tormenta Zapatel Card (Only for serious
> engineering types of people) from COTS
>
> On Thu, Sep 1, 2011 at 8:50 AM, Michel Daggelinckx
> <michel.daggelinckx at gmail.com>  wrote:
>> Digium and sangoma cards are high quality and work greast with FS
>>
>> Michel
>>
>>
>> On Thu, Sep 1, 2011 at 5:24 PM, ocset<ocset at the800group.com>  wrote:
>>> Hi
>>>
>>> As a new member of the forum, I am curious to know your experience with
>>> FXO/FXS cards.
>>>
>>> I have a SPA3102 and have configured it to work with FS, but it feels a
>>> bit like I am trying to make a square peg fit a round hole. I am hoping
>>> to implement FS at a business which currently has 4 POTS lines and would
>>> prefer to use an internal IDE card for the job of integrating these
>>> phone lines into FreeSwitch.
>>>
>>> Has anyone got some advice on which cards I should be looking at that
>>> just "work" with FS? What about echo cancellation - is that something I
>>> should just cater for or does it depend on the client situation? What
>>> about software cancellation?
>>>
>>> This seems like one of those times when too much choice is a bad thing
>>> and I need some guidance on what has worked for you.
>>>
>>> All advice would be greatly appreciated.
>>>
>>> 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
>
>
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 10.0.1392 / Virus Database: 1520/3872 - Release Date: 09/02/11
>
>


-- 
Atentamente,
*Dario Garc?a*
Consultor.

CCCT, Nivel C2, Sector Yarey, Mz,
Ofc. MZ03a.
Caracas-Venezuela.
Tel?fono: +58 212 9081842
Cel: +58 412 2221515
dgarcia at anew.com.ve
http://www.anew.com.ve
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110902/435603ff/attachment.html 

From erin.omeara at salmonbaytechnology.com  Fri Sep  2 21:59:01 2011
From: erin.omeara at salmonbaytechnology.com (Erin O'Meara)
Date: Fri, 2 Sep 2011 10:59:01 -0700
Subject: [Freeswitch-users] Strange one-way audio problem
Message-ID: <CAB4Oy0XSxfdJiwoB3i2nkaUHfvD=zC8cF=Xvi6+uE=cPihE0FQ@mail.gmail.com>

I have a Freeswitch server in the cloud with three SIP providers, DIDForSale
and IPKall for incoming and CallWithUs for outgoing.

When someone calls me, thru either incoming provider I hear everything clear
but i'm choppy to the other end.
When I make an outgoing call, audio is great in and out, also if I do an
echo test my audio is perfect.

I have check the freeswitch.log and its not transcoding  (grep transcod
freeswitch.log) which was a suggestion from #freeswitch

also added

verbose_sdp=true
<param name="vad" value="disable"/>
<param name="suppress-cng" value="true"/>

with no success, any thoughts?


Regards,

206.905.9520
http://salmonbaytechnology.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110902/0ff945b1/attachment-0001.html 

From msc at freeswitch.org  Fri Sep  2 22:07:04 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Sep 2011 11:07:04 -0700
Subject: [Freeswitch-users] recordFile max_len not obeyed
In-Reply-To: <CAAcoRq3k8oVeu4REgPH1SO-sceGB3kYS+=VEe8Tsk0CUieFoGA@mail.gmail.com>
References: <CAAcoRq1_NFPDpUgXMpwazN165pO7tKBP+y8X03bBjVi+pyY-CQ@mail.gmail.com>
	<1314896546108-6750754.post@n2.nabble.com>
	<CAKzWOxXuyF56j_1bE+OxG3tc492gGRGJg7jtst9SsTd7eS+QXQ@mail.gmail.com>
	<CAAcoRq3k8oVeu4REgPH1SO-sceGB3kYS+=VEe8Tsk0CUieFoGA@mail.gmail.com>
Message-ID: <CAKzWOxXCBwUq=jOpgMLjg32LfxMc4OFLQNoBgW3puMGXeAONeA@mail.gmail.com>

High five for improving the documentation.
-MC

On Thu, Sep 1, 2011 at 6:09 PM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Thanks! I've updated the documentation to make this more obvious.
>
> -Neil
>
> On Thu, Sep 1, 2011 at 10:41 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Good catch! I don't think he wanted a time limit of 2 hours and 46 minutes
>> on that recording. ;)
>> -MC
>>
>>
>> On Thu, Sep 1, 2011 at 10:02 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>>
>>> try 10 instead that value is in seconds :)
>>>
>>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110902/0c1bef95/attachment.html 

From lfurrea at gmail.com  Fri Sep  2 22:32:11 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Fri, 2 Sep 2011 12:32:11 -0600
Subject: [Freeswitch-users] Caller-Privacy-Hide-Name: [true]
Message-ID: <CAB2KLyd05_n6219mUNxXzzRgoXgSaq+GSZw0TYZAYkxMogJh3w@mail.gmail.com>

Hi all,

Any one can shed some light on why this channel variable

Caller-Privacy-Hide-Name: [true]

suddenly got set to true for a single phone ?

Is this something that can be set by the phone through the SIP INVITE? or
maybe in the directory?

This got turned on on a Snom 370 which is the receptionist phone and
threfore all calls from this phone appear as anonymous.

Nothing was changed on the dial plan or even directory.

This is a way old build of FS just FYI.

Your help is greatly appreciated.

Regards,

Luis
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110902/356bb764/attachment.html 

From potxoka at gmail.com  Sat Sep  3 01:44:28 2011
From: potxoka at gmail.com (Anto)
Date: Fri, 2 Sep 2011 23:44:28 +0200
Subject: [Freeswitch-users] External/internal profile
Message-ID: <CA+Ecp7Tgn6ObPkzNzQLQi418uQd5OJFTTX1B5cG6OF7cPnSY-g@mail.gmail.com>

Hello

Today I can not understand the internal and external profiles. I've
used other times asterisk setup and both the carriers and customers
share the same ports and same ips. I do not quite understand because
FreeSWITCH must use different ports (do not know if you can configure
the same ports. In all the how-to I've seen that change the profiles
and use different ports). Could anyone guide me for this?.

If for example I have a server with two public IPs, you could
configure each profile with an ip and both with the same ports (5060
and 5061)?. Perhaps it is an obvious question, but do not quite
understand the issue of profiling, I have doubts. Thank you very much.

Best regards.


From brad at tech21.com  Sat Sep  3 01:48:41 2011
From: brad at tech21.com (Brad Mina)
Date: Fri, 2 Sep 2011 14:48:41 -0700
Subject: [Freeswitch-users] External/internal profile
In-Reply-To: <CA+Ecp7Tgn6ObPkzNzQLQi418uQd5OJFTTX1B5cG6OF7cPnSY-g@mail.gmail.com>
References: <CA+Ecp7Tgn6ObPkzNzQLQi418uQd5OJFTTX1B5cG6OF7cPnSY-g@mail.gmail.com>
Message-ID: <CAGRr05-5F_B990V9MMFb4sQZkp2-c9R+Au1GWrP8Lyq2mvh+Cw@mail.gmail.com>

All profile listen on a single IP.

Default external profile port is 5080
Default internal profile port is 5060

Yes you can configure two different IP addresses to use the same port.

On Fri, Sep 2, 2011 at 2:44 PM, Anto <potxoka at gmail.com> wrote:

> Hello
>
> Today I can not understand the internal and external profiles. I've
> used other times asterisk setup and both the carriers and customers
> share the same ports and same ips. I do not quite understand because
> FreeSWITCH must use different ports (do not know if you can configure
> the same ports. In all the how-to I've seen that change the profiles
> and use different ports). Could anyone guide me for this?.
>
> If for example I have a server with two public IPs, you could
> configure each profile with an ip and both with the same ports (5060
> and 5061)?. Perhaps it is an obvious question, but do not quite
> understand the issue of profiling, I have doubts. Thank you very much.
>
> Best 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110902/7250d93d/attachment.html 

From anthony.minessale at gmail.com  Sat Sep  3 01:54:59 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 2 Sep 2011 16:54:59 -0500
Subject: [Freeswitch-users] FXO/FXS card advice
In-Reply-To: <4E60DC90.9030201@anew.com.ve>
References: <4E5FA3B0.9030606@the800group.com>
	<CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com>
	<CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com>
	<4E60DC90.9030201@anew.com.ve>
Message-ID: <CAKbxfG9LnStDCsZ7gopLfMq=37PTdeTrN-EwNrsa1tVvdq+T0g@mail.gmail.com>

If you are doing FXO and you already have an ATA working that might be
as good as you'll get.
Its nice to not have to deal with TDM if you can help it and most
analog ATA are cheap and effective and much less painful to deal with
than analog cards.



On Fri, Sep 2, 2011 at 8:39 AM, Saugort Dario Garcia Tovar
<dgarcia at anew.com.ve> wrote:
> One think you should consider also, is the FXO/FXS board that you will buy,
> you have to check compatibility with your server/pc.
>
> For example openvox models A800P and A1200P has an issue with motherboard
> based on chipset H55 for example.
>
>
> On 9/1/2011 9:59 PM, curriegrad2004 wrote:
>
> In the following order, most people usually recommend:
>
> 1. Sangoma
> 2. Diginum/OpenVox
> 3. Building your own Tormenta Zapatel Card (Only for serious
> engineering types of people) from COTS
>
> On Thu, Sep 1, 2011 at 8:50 AM, Michel Daggelinckx
> <michel.daggelinckx at gmail.com> wrote:
>
> Digium and sangoma cards are high quality and work greast with FS
>
> Michel
>
>
> On Thu, Sep 1, 2011 at 5:24 PM, ocset <ocset at the800group.com> wrote:
>
> Hi
>
> As a new member of the forum, I am curious to know your experience with
> FXO/FXS cards.
>
> I have a SPA3102 and have configured it to work with FS, but it feels a
> bit like I am trying to make a square peg fit a round hole. I am hoping
> to implement FS at a business which currently has 4 POTS lines and would
> prefer to use an internal IDE card for the job of integrating these
> phone lines into FreeSwitch.
>
> Has anyone got some advice on which cards I should be looking at that
> just "work" with FS? What about echo cancellation - is that something I
> should just cater for or does it depend on the client situation? What
> about software cancellation?
>
> This seems like one of those times when too much choice is a bad thing
> and I need some guidance on what has worked for you.
>
> All advice would be greatly appreciated.
>
> 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
>
>
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 10.0.1392 / Virus Database: 1520/3872 - Release Date: 09/02/11
>
>
>
>
> --
> Atentamente,
> Dario Garc?a
> Consultor.
>
> CCCT, Nivel C2, Sector Yarey, Mz,
> Ofc. MZ03a.
> Caracas-Venezuela.
> Tel?fono: +58 212 9081842
> Cel: +58 412 2221515
> dgarcia at anew.com.ve
> http://www.anew.com.ve
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 tomasz at hyziak.pl  Fri Sep  2 22:59:01 2011
From: tomasz at hyziak.pl (Tomasz Hyziak)
Date: Fri, 2 Sep 2011 20:59:01 +0200
Subject: [Freeswitch-users] FreeSwitch and core dumps
In-Reply-To: <CAFv7ub013xcpxyXWaGW-_5qmCAZ1DBQ5xXdZ5R-tRRBocu1i0w@mail.gmail.com>
References: <CAFv7ub1ciPKA9ceWcJrUx3fQROXrAxsdDyc0Q355gRB-ZtU52g@mail.gmail.com>
	<1314884710492-6750074.post@n2.nabble.com>
	<CAFv7ub0crGLj13WSu8s0zEgP759zGP=bFFaVjwMtx0X=oKZ30g@mail.gmail.com>
	<CAKbxfG-v+KUxvNDnjGrDbY3uOri29Vn14VObiGfOXZmwBek98Q@mail.gmail.com>
	<CAFv7ub013xcpxyXWaGW-_5qmCAZ1DBQ5xXdZ5R-tRRBocu1i0w@mail.gmail.com>
Message-ID: <CAFv7ub3oeNtMtgtkUd2sfY73AQpy+GbMZWMR4bWCWAjuZ5W9Hw@mail.gmail.com>

Hi again.

24 hours without problems. I hope that there will not be any more problems.

Thanks for your help :)

--
pozdrawiam - Tomasz Hyziak



2011/9/1 Tomasz Hyziak <tomasz at hyziak.pl>:
> Thanks Anthony for the info.
>
> I'll run configure --without-libcurl and recompile FS... tomorrow will
> be judgement day :)
>
> I'll let you know.
>
> --
> pozdrawiam - Tomasz Hyziak
>
>
>
> 2011/9/1 Anthony Minessale <anthony.minessale at gmail.com>:
>> try a top level configure using the --without-libcurl argument so it
>> uses our libcurl in tree.
>>
>>
>>
>> On Thu, Sep 1, 2011 at 8:57 AM, Tomasz Hyziak <tomasz at hyziak.pl> wrote:
>>> Hi Jeef.
>>>
>>> Yes, that's FreeSWITCH compiled from git version: 1.0.head
>>> (git-cd31633 2011-08-17 19-34-22 -0500).
>>>
>>> I use mod_xml_cdr for saving cdrs to database via webservice.
>>>
>>> What could I do with it ? Recompiling mod_xml_cdr or curl ?
>>>
>>> --
>>> pozdrawiam - Tomasz Hyziak
>>>
>>>
>>>
>>> 2011/9/1 Jeff Lenk <jeff at jefflenk.com>:
>>>> Is this with Git Head? It appears that something is wrong with curl.
>>>>
>>>> --
>>>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-and-core-dumps-tp6750013p6750074.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
>>>
>>
>>
>>
>> --
>> 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 contact at aharm.de  Sat Sep  3 00:59:29 2011
From: contact at aharm.de (Alexander Harm)
Date: Fri, 2 Sep 2011 22:59:29 +0200
Subject: [Freeswitch-users] Need help cc Gateway (SIP trunk) setup for
	provider using Auth ID
Message-ID: <4777C308-2C0C-44A8-B713-E0D318546E6B@aharm.de>

My SIP provider uses
- User ID (same as Caller ID Number)
- Password
- Auth ID (different from User ID)
for registration. I have to admit that I'm completely at loss on how to configure freeSWITCH using Auth ID. I tried all combinations I could think off but I just keep getting 403 error messages.
Help is very much appreciated.



From michal.bielicki at seventhsignal.de  Sat Sep  3 03:46:01 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Sat, 3 Sep 2011 01:46:01 +0200
Subject: [Freeswitch-users] Need help cc Gateway (SIP trunk) setup for
	provider using Auth ID
In-Reply-To: <4777C308-2C0C-44A8-B713-E0D318546E6B@aharm.de>
References: <4777C308-2C0C-44A8-B713-E0D318546E6B@aharm.de>
Message-ID: <14C7CA65-D77E-47C6-B196-28DE75069208@seventhsignal.de>

Which provider ?
I can probably help you with most german ones.

Am 02.09.2011 um 22:59 schrieb Alexander Harm:

> My SIP provider uses
> - User ID (same as Caller ID Number)
> - Password
> - Auth ID (different from User ID)
> for registration. I have to admit that I'm completely at loss on how to configure freeSWITCH using Auth ID. I tried all combinations I could think off but I just keep getting 403 error messages.
> Help is very much appreciated.
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht 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





----

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110903/ac597d54/attachment-0001.html 

From djbinter at gmail.com  Sat Sep  3 03:48:59 2011
From: djbinter at gmail.com (DJB International)
Date: Fri, 2 Sep 2011 16:48:59 -0700
Subject: [Freeswitch-users] Need help cc Gateway (SIP trunk) setup for
 provider using Auth ID
In-Reply-To: <14C7CA65-D77E-47C6-B196-28DE75069208@seventhsignal.de>
References: <4777C308-2C0C-44A8-B713-E0D318546E6B@aharm.de>
	<14C7CA65-D77E-47C6-B196-28DE75069208@seventhsignal.de>
Message-ID: <CAAB3jWuHUt65n7T94Ko_NnarntiVyd-T6RXPNEGx-nL6nfj0hw@mail.gmail.com>

Maybe something similar to this:
http://wiki.freeswitch.org/wiki/Provider_Configuration:_Phonzo

-djbinter

On Fri, Sep 2, 2011 at 4:46 PM, Michal Bielicki <
michal.bielicki at seventhsignal.de> wrote:

> Which provider ?
> I can probably help you with most german ones.
>
> Am 02.09.2011 um 22:59 schrieb Alexander Harm:
>
> My SIP provider uses
> - User ID (same as Caller ID Number)
> - Password
> - Auth ID (different from User ID)
> for registration. I have to admit that I'm completely at loss on how to
> configure freeSWITCH using Auth ID. I tried all combinations I could think
> off but I just keep getting 403 error messages.
> Help is very much appreciated.
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> Michal Bielicki
> Gesch?ftsf?hrer / CEO
>
> *Seventh Signal Ltd. & Co. KG*
> Weigandufer 45, B?ro 115, D-12059 Berlin
> Voice: +49 30 60988730
>
> Amtsgericht 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
>
>
>
>
> ----
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110902/fcd48638/attachment.html 

From xing2kin at yahoo.com  Sat Sep  3 08:52:31 2011
From: xing2kin at yahoo.com (king2kin)
Date: Fri, 2 Sep 2011 21:52:31 -0700 (PDT)
Subject: [Freeswitch-users] How to specify language for phrase macro inside
	session:playAndGetDigits(-)
Message-ID: <1315025551.80678.YahooMailClassic@web39705.mail.mud.yahoo.com>

Hi all,

{
session:playAndGetDigits(1, 1, 1, 3000, "#", "phrase:xk_confirm_userid:" .. uid, invalid, "[0,1,9]")
} 
always uses default language (e.g. 'en') to pick up phrase marco definition,
but now I would like to specify another language instead of FS default language to play back this phrase macro inside session:playAndGetDigits(-).

Could anyone please tell me how I can specify non-default language for playing back my phrase macro inside session:playAndGetDigits(-)?

Thanks

x.k.



From ocset at the800group.com  Sat Sep  3 09:38:10 2011
From: ocset at the800group.com (ocset)
Date: Sat, 03 Sep 2011 13:38:10 +0800
Subject: [Freeswitch-users] FXO/FXS card advice
In-Reply-To: <CAKbxfG9LnStDCsZ7gopLfMq=37PTdeTrN-EwNrsa1tVvdq+T0g@mail.gmail.com>
References: <4E5FA3B0.9030606@the800group.com>	<CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com>	<CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com>	<4E60DC90.9030201@anew.com.ve>
	<CAKbxfG9LnStDCsZ7gopLfMq=37PTdeTrN-EwNrsa1tVvdq+T0g@mail.gmail.com>
Message-ID: <4E61BD42.9090109@the800group.com>

Thank you all for your replies.

Anthony, you reply has left me uncertain again. All of the ATA's listed 
here (wiki.freeswitch.org/wiki/Interop_List) seem to have some 
limitations and bugs which have to worked around.

Your reply suggests that buying a OpenVox A400P04 card (currently only 
$219 for 4 FXO's) would be more difficult to install and maintain that 4 
ATA's. I would have thought that the FXS/FXO cards were engineered to 
work with Freeswitch/Asterisk etc. without limitations like caller-id 
not working or call-transfer not working as expected.

I can see the benefit of an ATA since I can add them on as needed with 
little or no hassle (except for the large power board to plug them all 
into:-)

I have only just got the SPA3102 receiving incoming calls and I am no 
expert in either solution but would have thought that a 4 port FXO card 
would be easier as it is made to purpose?

ps. the PAP2T seems to be one of the better Linksys ATA's from the list 
of features that work?

Thanks again for all your help and suggestions.

On 09/03/2011 05:54 AM, Anthony Minessale wrote:
> If you are doing FXO and you already have an ATA working that might be
> as good as you'll get.
> Its nice to not have to deal with TDM if you can help it and most
> analog ATA are cheap and effective and much less painful to deal with
> than analog cards.
>
>
>
> On Fri, Sep 2, 2011 at 8:39 AM, Saugort Dario Garcia Tovar
> <dgarcia at anew.com.ve>  wrote:
>> One think you should consider also, is the FXO/FXS board that you will buy,
>> you have to check compatibility with your server/pc.
>>
>> For example openvox models A800P and A1200P has an issue with motherboard
>> based on chipset H55 for example.
>>
>>
>> On 9/1/2011 9:59 PM, curriegrad2004 wrote:
>>
>> In the following order, most people usually recommend:
>>
>> 1. Sangoma
>> 2. Diginum/OpenVox
>> 3. Building your own Tormenta Zapatel Card (Only for serious
>> engineering types of people) from COTS
>>
>> On Thu, Sep 1, 2011 at 8:50 AM, Michel Daggelinckx
>> <michel.daggelinckx at gmail.com>  wrote:
>>
>> Digium and sangoma cards are high quality and work greast with FS
>>
>> Michel
>>
>>
>> On Thu, Sep 1, 2011 at 5:24 PM, ocset<ocset at the800group.com>  wrote:
>>
>> Hi
>>
>> As a new member of the forum, I am curious to know your experience with
>> FXO/FXS cards.
>>
>> I have a SPA3102 and have configured it to work with FS, but it feels a
>> bit like I am trying to make a square peg fit a round hole. I am hoping
>> to implement FS at a business which currently has 4 POTS lines and would
>> prefer to use an internal IDE card for the job of integrating these
>> phone lines into FreeSwitch.
>>
>> Has anyone got some advice on which cards I should be looking at that
>> just "work" with FS? What about echo cancellation - is that something I
>> should just cater for or does it depend on the client situation? What
>> about software cancellation?
>>
>> This seems like one of those times when too much choice is a bad thing
>> and I need some guidance on what has worked for you.
>>
>> All advice would be greatly appreciated.
>>
>> 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
>>
>>
>> -----
>> No virus found in this message.
>> Checked by AVG - www.avg.com
>> Version: 10.0.1392 / Virus Database: 1520/3872 - Release Date: 09/02/11
>>
>>
>>
>>
>> --
>> Atentamente,
>> Dario Garc?a
>> Consultor.
>>
>> CCCT, Nivel C2, Sector Yarey, Mz,
>> Ofc. MZ03a.
>> Caracas-Venezuela.
>> Tel?fono: +58 212 9081842
>> Cel: +58 412 2221515
>> dgarcia at anew.com.ve
>> http://www.anew.com.ve
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>


From darcy at primrose.ws  Sat Sep  3 14:30:39 2011
From: darcy at primrose.ws (Darcy)
Date: Sat, 3 Sep 2011 06:30:39 -0400
Subject: [Freeswitch-users] FXO/FXS card advice
In-Reply-To: <4E61BD42.9090109@the800group.com>
References: <4E5FA3B0.9030606@the800group.com>	<CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com>	<CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com>	<4E60DC90.9030201@anew.com.ve><CAKbxfG9LnStDCsZ7gopLfMq=37PTdeTrN-EwNrsa1tVvdq+T0g@mail.gmail.com>
	<4E61BD42.9090109@the800group.com>
Message-ID: <6F8ACCA987D84F068F0B1A5D4AC515A4@DWP>

Hi, for fxo's with the freeswitch I currently use:

Mfr's SKU: GXW4104
Brand: Grandstream
4 x FXO Gateway, 2 x LAN, 1 or 2 stage dialing, G.168 echo cancellation

They work exceptionally well, easy to configure and can be auto configured 
if you
are into that, we are.  You can also get an 8 port version.  I pay 220 for 
the 4 port
and 300 for the 8 port.  I have quite a few of them deployed.

Darcy


-----Original Message----- 
From: ocset
Sent: Saturday, September 03, 2011 1:38 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FXO/FXS card advice

Thank you all for your replies.

Anthony, you reply has left me uncertain again. All of the ATA's listed
here (wiki.freeswitch.org/wiki/Interop_List) seem to have some
limitations and bugs which have to worked around.

Your reply suggests that buying a OpenVox A400P04 card (currently only
$219 for 4 FXO's) would be more difficult to install and maintain that 4
ATA's. I would have thought that the FXS/FXO cards were engineered to
work with Freeswitch/Asterisk etc. without limitations like caller-id
not working or call-transfer not working as expected.

I can see the benefit of an ATA since I can add them on as needed with
little or no hassle (except for the large power board to plug them all
into:-)

I have only just got the SPA3102 receiving incoming calls and I am no
expert in either solution but would have thought that a 4 port FXO card
would be easier as it is made to purpose?

ps. the PAP2T seems to be one of the better Linksys ATA's from the list
of features that work?

Thanks again for all your help and suggestions.

On 09/03/2011 05:54 AM, Anthony Minessale wrote:
> If you are doing FXO and you already have an ATA working that might be
> as good as you'll get.
> Its nice to not have to deal with TDM if you can help it and most
> analog ATA are cheap and effective and much less painful to deal with
> than analog cards.
>
>
>
> On Fri, Sep 2, 2011 at 8:39 AM, Saugort Dario Garcia Tovar
> <dgarcia at anew.com.ve>  wrote:
>> One think you should consider also, is the FXO/FXS board that you will 
>> buy,
>> you have to check compatibility with your server/pc.
>>
>> For example openvox models A800P and A1200P has an issue with motherboard
>> based on chipset H55 for example.
>>
>>
>> On 9/1/2011 9:59 PM, curriegrad2004 wrote:
>>
>> In the following order, most people usually recommend:
>>
>> 1. Sangoma
>> 2. Diginum/OpenVox
>> 3. Building your own Tormenta Zapatel Card (Only for serious
>> engineering types of people) from COTS
>>
>> On Thu, Sep 1, 2011 at 8:50 AM, Michel Daggelinckx
>> <michel.daggelinckx at gmail.com>  wrote:
>>
>> Digium and sangoma cards are high quality and work greast with FS
>>
>> Michel
>>
>>
>> On Thu, Sep 1, 2011 at 5:24 PM, ocset<ocset at the800group.com>  wrote:
>>
>> Hi
>>
>> As a new member of the forum, I am curious to know your experience with
>> FXO/FXS cards.
>>
>> I have a SPA3102 and have configured it to work with FS, but it feels a
>> bit like I am trying to make a square peg fit a round hole. I am hoping
>> to implement FS at a business which currently has 4 POTS lines and would
>> prefer to use an internal IDE card for the job of integrating these
>> phone lines into FreeSwitch.
>>
>> Has anyone got some advice on which cards I should be looking at that
>> just "work" with FS? What about echo cancellation - is that something I
>> should just cater for or does it depend on the client situation? What
>> about software cancellation?
>>
>> This seems like one of those times when too much choice is a bad thing
>> and I need some guidance on what has worked for you.
>>
>> All advice would be greatly appreciated.
>>
>> 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
>>
>>
>> -----
>> No virus found in this message.
>> Checked by AVG - www.avg.com
>> Version: 10.0.1392 / Virus Database: 1520/3872 - Release Date: 09/02/11
>>
>>
>>
>>
>> --
>> Atentamente,
>> Dario Garc?a
>> Consultor.
>>
>> CCCT, Nivel C2, Sector Yarey, Mz,
>> Ofc. MZ03a.
>> Caracas-Venezuela.
>> Tel?fono: +58 212 9081842
>> Cel: +58 412 2221515
>> dgarcia at anew.com.ve
>> http://www.anew.com.ve
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>


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



From covici at ccs.covici.com  Sat Sep  3 15:06:13 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sat, 03 Sep 2011 07:06:13 -0400
Subject: [Freeswitch-users] FXO/FXS card advice
In-Reply-To: <6F8ACCA987D84F068F0B1A5D4AC515A4@DWP>
References: <4E5FA3B0.9030606@the800group.com>
	<CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com>
	<CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com>
	<4E60DC90.9030201@anew.com.ve><CAKbxfG9LnStDCsZ7gopLfMq=37PTdeTrN-EwNrsa1tVvdq+T0g@mail.gmail.com>
	<4E61BD42.9090109@the800group.com>
	<6F8ACCA987D84F068F0B1A5D4AC515A4@DWP>
Message-ID: <22926.1315047973@ccs.covici.com>

Does this ata support picking up call waiting while in a call from the
fxo?  If it were to do that, I would get one right away.  The Digium FXO
card I have has problems doing this with freetdm.

Darcy <darcy at primrose.ws> wrote:

> Hi, for fxo's with the freeswitch I currently use:
> 
> Mfr's SKU: GXW4104
> Brand: Grandstream
> 4 x FXO Gateway, 2 x LAN, 1 or 2 stage dialing, G.168 echo cancellation
> 
> They work exceptionally well, easy to configure and can be auto configured 
> if you
> are into that, we are.  You can also get an 8 port version.  I pay 220 for 
> the 4 port
> and 300 for the 8 port.  I have quite a few of them deployed.
> 
> Darcy
> 
> 
> -----Original Message----- 
> From: ocset
> Sent: Saturday, September 03, 2011 1:38 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FXO/FXS card advice
> 
> Thank you all for your replies.
> 
> Anthony, you reply has left me uncertain again. All of the ATA's listed
> here (wiki.freeswitch.org/wiki/Interop_List) seem to have some
> limitations and bugs which have to worked around.
> 
> Your reply suggests that buying a OpenVox A400P04 card (currently only
> $219 for 4 FXO's) would be more difficult to install and maintain that 4
> ATA's. I would have thought that the FXS/FXO cards were engineered to
> work with Freeswitch/Asterisk etc. without limitations like caller-id
> not working or call-transfer not working as expected.
> 
> I can see the benefit of an ATA since I can add them on as needed with
> little or no hassle (except for the large power board to plug them all
> into:-)
> 
> I have only just got the SPA3102 receiving incoming calls and I am no
> expert in either solution but would have thought that a 4 port FXO card
> would be easier as it is made to purpose?
> 
> ps. the PAP2T seems to be one of the better Linksys ATA's from the list
> of features that work?
> 
> Thanks again for all your help and suggestions.
> 
> On 09/03/2011 05:54 AM, Anthony Minessale wrote:
> > If you are doing FXO and you already have an ATA working that might be
> > as good as you'll get.
> > Its nice to not have to deal with TDM if you can help it and most
> > analog ATA are cheap and effective and much less painful to deal with
> > than analog cards.
> >
> >
> >
> > On Fri, Sep 2, 2011 at 8:39 AM, Saugort Dario Garcia Tovar
> > <dgarcia at anew.com.ve>  wrote:
> >> One think you should consider also, is the FXO/FXS board that you will 
> >> buy,
> >> you have to check compatibility with your server/pc.
> >>
> >> For example openvox models A800P and A1200P has an issue with motherboard
> >> based on chipset H55 for example.
> >>
> >>
> >> On 9/1/2011 9:59 PM, curriegrad2004 wrote:
> >>
> >> In the following order, most people usually recommend:
> >>
> >> 1. Sangoma
> >> 2. Diginum/OpenVox
> >> 3. Building your own Tormenta Zapatel Card (Only for serious
> >> engineering types of people) from COTS
> >>
> >> On Thu, Sep 1, 2011 at 8:50 AM, Michel Daggelinckx
> >> <michel.daggelinckx at gmail.com>  wrote:
> >>
> >> Digium and sangoma cards are high quality and work greast with FS
> >>
> >> Michel
> >>
> >>
> >> On Thu, Sep 1, 2011 at 5:24 PM, ocset<ocset at the800group.com>  wrote:
> >>
> >> Hi
> >>
> >> As a new member of the forum, I am curious to know your experience with
> >> FXO/FXS cards.
> >>
> >> I have a SPA3102 and have configured it to work with FS, but it feels a
> >> bit like I am trying to make a square peg fit a round hole. I am hoping
> >> to implement FS at a business which currently has 4 POTS lines and would
> >> prefer to use an internal IDE card for the job of integrating these
> >> phone lines into FreeSwitch.
> >>
> >> Has anyone got some advice on which cards I should be looking at that
> >> just "work" with FS? What about echo cancellation - is that something I
> >> should just cater for or does it depend on the client situation? What
> >> about software cancellation?
> >>
> >> This seems like one of those times when too much choice is a bad thing
> >> and I need some guidance on what has worked for you.
> >>
> >> All advice would be greatly appreciated.
> >>
> >> 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
> >>
> >>
> >> -----
> >> No virus found in this message.
> >> Checked by AVG - www.avg.com
> >> Version: 10.0.1392 / Virus Database: 1520/3872 - Release Date: 09/02/11
> >>
> >>
> >>
> >>
> >> --
> >> Atentamente,
> >> Dario Garc?a
> >> Consultor.
> >>
> >> CCCT, Nivel C2, Sector Yarey, Mz,
> >> Ofc. MZ03a.
> >> Caracas-Venezuela.
> >> Tel?fono: +58 212 9081842
> >> Cel: +58 412 2221515
> >> dgarcia at anew.com.ve
> >> http://www.anew.com.ve
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 codecomplete at free.fr  Sat Sep  3 16:07:46 2011
From: codecomplete at free.fr (GillesToo)
Date: Sat, 3 Sep 2011 05:07:46 -0700 (PDT)
Subject: [Freeswitch-users] Compiling Freeswitch for Android?
In-Reply-To: <296C71FF-C5F6-41E8-A08C-5DE3AE61300F@avgs.ca>
References: <1314959406122-6753409.post@n2.nabble.com>
	<BD97C6DD-E80A-4CE8-8599-C5E851150454@avgs.ca>
	<1314962913219-6753523.post@n2.nabble.com>
	<296C71FF-C5F6-41E8-A08C-5DE3AE61300F@avgs.ca>
Message-ID: <1315051666071-6756501.post@n2.nabble.com>

Thanks :)

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Compiling-Freeswitch-for-Android-tp6753409p6756501.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From darcy at primrose.ws  Sat Sep  3 16:35:58 2011
From: darcy at primrose.ws (Darcy)
Date: Sat, 3 Sep 2011 08:35:58 -0400
Subject: [Freeswitch-users] FXO/FXS card advice
In-Reply-To: <22926.1315047973@ccs.covici.com>
References: <4E5FA3B0.9030606@the800group.com><CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com><CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com><4E60DC90.9030201@anew.com.ve><CAKbxfG9LnStDCsZ7gopLfMq=37PTdeTrN-EwNrsa1tVvdq+T0g@mail.gmail.com><4E61BD42.9090109@the800group.com><6F8ACCA987D84F068F0B1A5D4AC515A4@DWP>
	<22926.1315047973@ccs.covici.com>
Message-ID: <8DF93FC7EB194882863E75A2B19AEFA6@DWP>

The grandstream has hookflash timers, but I have never been able
to get it to work and info back from grandstream indicates is is not a
pending implementation.
For cases where I require a hook flash I use the audio codes fxo gateways.
I never tested it on the freeswitch, we have own ap we wrote to handle it,
we were doing this before we discovered freeswitch.  I know it works and it
is quite reliable. If I get time later today, I will give it a try on a 
freeswitch.

Brand: AudioCodes
Mfr's SKU: MP114/4O/SIP
VoIP SIP Gateway, 4x FXO, 1x WAN, LBR Codecs,

Here are the options you have for using it.  they have a variety of 
configurations, fxo/fxs combos
a 4 port is in the 375 range.

Hook Flash Option
[HookFlashOption]
Supported hook-flash Transport Type (method by which hook-flash is sent and
received).
Valid options include:
0 = Hook-Flash indication isn?t sent (default)
1 = Send proprietary INFO message with Hook-Flash indication
4 = RFC 2833

Darcy

-----Original Message----- 
From: covici at ccs.covici.com
Sent: Saturday, September 03, 2011 7:06 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FXO/FXS card advice

Does this ata support picking up call waiting while in a call from the
fxo?  If it were to do that, I would get one right away.  The Digium FXO
card I have has problems doing this with freetdm.

Darcy <darcy at primrose.ws> wrote:

> Hi, for fxo's with the freeswitch I currently use:
>
> Mfr's SKU: GXW4104
> Brand: Grandstream
> 4 x FXO Gateway, 2 x LAN, 1 or 2 stage dialing, G.168 echo cancellation
>
> They work exceptionally well, easy to configure and can be auto configured
> if you
> are into that, we are.  You can also get an 8 port version.  I pay 220 for
> the 4 port
> and 300 for the 8 port.  I have quite a few of them deployed.
>
> Darcy
>
>
> -----Original Message----- 
> From: ocset
> Sent: Saturday, September 03, 2011 1:38 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FXO/FXS card advice
>
> Thank you all for your replies.
>
> Anthony, you reply has left me uncertain again. All of the ATA's listed
> here (wiki.freeswitch.org/wiki/Interop_List) seem to have some
> limitations and bugs which have to worked around.
>
> Your reply suggests that buying a OpenVox A400P04 card (currently only
> $219 for 4 FXO's) would be more difficult to install and maintain that 4
> ATA's. I would have thought that the FXS/FXO cards were engineered to
> work with Freeswitch/Asterisk etc. without limitations like caller-id
> not working or call-transfer not working as expected.
>
> I can see the benefit of an ATA since I can add them on as needed with
> little or no hassle (except for the large power board to plug them all
> into:-)
>
> I have only just got the SPA3102 receiving incoming calls and I am no
> expert in either solution but would have thought that a 4 port FXO card
> would be easier as it is made to purpose?
>
> ps. the PAP2T seems to be one of the better Linksys ATA's from the list
> of features that work?
>
> Thanks again for all your help and suggestions.
>
> On 09/03/2011 05:54 AM, Anthony Minessale wrote:
> > If you are doing FXO and you already have an ATA working that might be
> > as good as you'll get.
> > Its nice to not have to deal with TDM if you can help it and most
> > analog ATA are cheap and effective and much less painful to deal with
> > than analog cards.
> >
> >
> >
> > On Fri, Sep 2, 2011 at 8:39 AM, Saugort Dario Garcia Tovar
> > <dgarcia at anew.com.ve>  wrote:
> >> One think you should consider also, is the FXO/FXS board that you will
> >> buy,
> >> you have to check compatibility with your server/pc.
> >>
> >> For example openvox models A800P and A1200P has an issue with 
> >> motherboard
> >> based on chipset H55 for example.
> >>
> >>
> >> On 9/1/2011 9:59 PM, curriegrad2004 wrote:
> >>
> >> In the following order, most people usually recommend:
> >>
> >> 1. Sangoma
> >> 2. Diginum/OpenVox
> >> 3. Building your own Tormenta Zapatel Card (Only for serious
> >> engineering types of people) from COTS
> >>
> >> On Thu, Sep 1, 2011 at 8:50 AM, Michel Daggelinckx
> >> <michel.daggelinckx at gmail.com>  wrote:
> >>
> >> Digium and sangoma cards are high quality and work greast with FS
> >>
> >> Michel
> >>
> >>
> >> On Thu, Sep 1, 2011 at 5:24 PM, ocset<ocset at the800group.com>  wrote:
> >>
> >> Hi
> >>
> >> As a new member of the forum, I am curious to know your experience with
> >> FXO/FXS cards.
> >>
> >> I have a SPA3102 and have configured it to work with FS, but it feels a
> >> bit like I am trying to make a square peg fit a round hole. I am hoping
> >> to implement FS at a business which currently has 4 POTS lines and 
> >> would
> >> prefer to use an internal IDE card for the job of integrating these
> >> phone lines into FreeSwitch.
> >>
> >> Has anyone got some advice on which cards I should be looking at that
> >> just "work" with FS? What about echo cancellation - is that something I
> >> should just cater for or does it depend on the client situation? What
> >> about software cancellation?
> >>
> >> This seems like one of those times when too much choice is a bad thing
> >> and I need some guidance on what has worked for you.
> >>
> >> All advice would be greatly appreciated.
> >>
> >> 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
> >>
> >>
> >> -----
> >> No virus found in this message.
> >> Checked by AVG - www.avg.com
> >> Version: 10.0.1392 / Virus Database: 1520/3872 - Release Date: 09/02/11
> >>
> >>
> >>
> >>
> >> --
> >> Atentamente,
> >> Dario Garc?a
> >> Consultor.
> >>
> >> CCCT, Nivel C2, Sector Yarey, Mz,
> >> Ofc. MZ03a.
> >> Caracas-Venezuela.
> >> Tel?fono: +58 212 9081842
> >> Cel: +58 412 2221515
> >> dgarcia at anew.com.ve
> >> http://www.anew.com.ve
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 



From ocset at the800group.com  Sat Sep  3 17:21:54 2011
From: ocset at the800group.com (ocset)
Date: Sat, 03 Sep 2011 21:21:54 +0800
Subject: [Freeswitch-users] FXO/FXS card advice
In-Reply-To: <8DF93FC7EB194882863E75A2B19AEFA6@DWP>
References: <4E5FA3B0.9030606@the800group.com><CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com><CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com><4E60DC90.9030201@anew.com.ve><CAKbxfG9LnStDCsZ7gopLfMq=37PTdeTrN-EwNrsa1tVvdq+T0g@mail.gmail.com><4E61BD42.9090109@the800group.com><6F8ACCA987D84F068F0B1A5D4AC515A4@DWP>	<22926.1315047973@ccs.covici.com>
	<8DF93FC7EB194882863E75A2B19AEFA6@DWP>
Message-ID: <4E6229F2.8090000@the800group.com>

Darcy

This info is very useful, especially since you have this setup running 
in production.

One question  - the two ATA devices (Grandstream & AudioCodes) are 
around the same price so could you give us some more info on why you 
would prefer the one over the other (besides the hook flash feature).

Do they handle echo cancellation or do you deploy a separate solution 
for that?

Thanks again for your help!

On 09/03/2011 08:35 PM, Darcy wrote:
> The grandstream has hookflash timers, but I have never been able
> to get it to work and info back from grandstream indicates is is not a
> pending implementation.
> For cases where I require a hook flash I use the audio codes fxo gateways.
> I never tested it on the freeswitch, we have own ap we wrote to handle it,
> we were doing this before we discovered freeswitch.  I know it works and it
> is quite reliable. If I get time later today, I will give it a try on a
> freeswitch.
>
> Brand: AudioCodes
> Mfr's SKU: MP114/4O/SIP
> VoIP SIP Gateway, 4x FXO, 1x WAN, LBR Codecs,
>
> Here are the options you have for using it.  they have a variety of
> configurations, fxo/fxs combos
> a 4 port is in the 375 range.
>
> Hook Flash Option
> [HookFlashOption]
> Supported hook-flash Transport Type (method by which hook-flash is sent and
> received).
> Valid options include:
> 0 = Hook-Flash indication isn?t sent (default)
> 1 = Send proprietary INFO message with Hook-Flash indication
> 4 = RFC 2833
>
> Darcy
>
> -----Original Message-----
> From: covici at ccs.covici.com
> Sent: Saturday, September 03, 2011 7:06 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] FXO/FXS card advice
>
> Does this ata support picking up call waiting while in a call from the
> fxo?  If it were to do that, I would get one right away.  The Digium FXO
> card I have has problems doing this with freetdm.
>
> Darcy<darcy at primrose.ws>  wrote:
>
>> Hi, for fxo's with the freeswitch I currently use:
>>
>> Mfr's SKU: GXW4104
>> Brand: Grandstream
>> 4 x FXO Gateway, 2 x LAN, 1 or 2 stage dialing, G.168 echo cancellation
>>
>> They work exceptionally well, easy to configure and can be auto configured
>> if you
>> are into that, we are.  You can also get an 8 port version.  I pay 220 for
>> the 4 port
>> and 300 for the 8 port.  I have quite a few of them deployed.
>>
>> Darcy
>>
>>
>> -----Original Message-----
>> From: ocset
>> Sent: Saturday, September 03, 2011 1:38 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] FXO/FXS card advice
>>
>> Thank you all for your replies.
>>
>> Anthony, you reply has left me uncertain again. All of the ATA's listed
>> here (wiki.freeswitch.org/wiki/Interop_List) seem to have some
>> limitations and bugs which have to worked around.
>>
>> Your reply suggests that buying a OpenVox A400P04 card (currently only
>> $219 for 4 FXO's) would be more difficult to install and maintain that 4
>> ATA's. I would have thought that the FXS/FXO cards were engineered to
>> work with Freeswitch/Asterisk etc. without limitations like caller-id
>> not working or call-transfer not working as expected.
>>
>> I can see the benefit of an ATA since I can add them on as needed with
>> little or no hassle (except for the large power board to plug them all
>> into:-)
>>
>> I have only just got the SPA3102 receiving incoming calls and I am no
>> expert in either solution but would have thought that a 4 port FXO card
>> would be easier as it is made to purpose?
>>
>> ps. the PAP2T seems to be one of the better Linksys ATA's from the list
>> of features that work?
>>
>> Thanks again for all your help and suggestions.
>>
>> On 09/03/2011 05:54 AM, Anthony Minessale wrote:
>>> If you are doing FXO and you already have an ATA working that might be
>>> as good as you'll get.
>>> Its nice to not have to deal with TDM if you can help it and most
>>> analog ATA are cheap and effective and much less painful to deal with
>>> than analog cards.
>>>
>>>
>>>
>>> On Fri, Sep 2, 2011 at 8:39 AM, Saugort Dario Garcia Tovar
>>> <dgarcia at anew.com.ve>   wrote:
>>>> One think you should consider also, is the FXO/FXS board that you will
>>>> buy,
>>>> you have to check compatibility with your server/pc.
>>>>
>>>> For example openvox models A800P and A1200P has an issue with
>>>> motherboard
>>>> based on chipset H55 for example.
>>>>
>>>>
>>>> On 9/1/2011 9:59 PM, curriegrad2004 wrote:
>>>>
>>>> In the following order, most people usually recommend:
>>>>
>>>> 1. Sangoma
>>>> 2. Diginum/OpenVox
>>>> 3. Building your own Tormenta Zapatel Card (Only for serious
>>>> engineering types of people) from COTS
>>>>
>>>> On Thu, Sep 1, 2011 at 8:50 AM, Michel Daggelinckx
>>>> <michel.daggelinckx at gmail.com>   wrote:
>>>>
>>>> Digium and sangoma cards are high quality and work greast with FS
>>>>
>>>> Michel
>>>>
>>>>
>>>> On Thu, Sep 1, 2011 at 5:24 PM, ocset<ocset at the800group.com>   wrote:
>>>>
>>>> Hi
>>>>
>>>> As a new member of the forum, I am curious to know your experience with
>>>> FXO/FXS cards.
>>>>
>>>> I have a SPA3102 and have configured it to work with FS, but it feels a
>>>> bit like I am trying to make a square peg fit a round hole. I am hoping
>>>> to implement FS at a business which currently has 4 POTS lines and
>>>> would
>>>> prefer to use an internal IDE card for the job of integrating these
>>>> phone lines into FreeSwitch.
>>>>
>>>> Has anyone got some advice on which cards I should be looking at that
>>>> just "work" with FS? What about echo cancellation - is that something I
>>>> should just cater for or does it depend on the client situation? What
>>>> about software cancellation?
>>>>
>>>> This seems like one of those times when too much choice is a bad thing
>>>> and I need some guidance on what has worked for you.
>>>>
>>>> All advice would be greatly appreciated.
>>>>
>>>> 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
>>>>
>>>>
>>>> -----
>>>> No virus found in this message.
>>>> Checked by AVG - www.avg.com
>>>> Version: 10.0.1392 / Virus Database: 1520/3872 - Release Date: 09/02/11
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Atentamente,
>>>> Dario Garc?a
>>>> Consultor.
>>>>
>>>> CCCT, Nivel C2, Sector Yarey, Mz,
>>>> Ofc. MZ03a.
>>>> Caracas-Venezuela.
>>>> Tel?fono: +58 212 9081842
>>>> Cel: +58 412 2221515
>>>> dgarcia at anew.com.ve
>>>> http://www.anew.com.ve
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org


From darcy at primrose.ws  Sat Sep  3 18:21:48 2011
From: darcy at primrose.ws (Darcy)
Date: Sat, 3 Sep 2011 10:21:48 -0400
Subject: [Freeswitch-users] FXO/FXS card advice
In-Reply-To: <4E6229F2.8090000@the800group.com>
References: <4E5FA3B0.9030606@the800group.com><CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com><CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com><4E60DC90.9030201@anew.com.ve><CAKbxfG9LnStDCsZ7gopLfMq=37PTdeTrN-EwNrsa1tVvdq+T0g@mail.gmail.com><4E61BD42.9090109@the800group.com><6F8ACCA987D84F068F0B1A5D4AC515A4@DWP>	<22926.1315047973@ccs.covici.com><8DF93FC7EB194882863E75A2B19AEFA6@DWP>
	<4E6229F2.8090000@the800group.com>
Message-ID: <31E98B785AFA401EB9F3B4005EF2544E@DWP>

they both handle echo suppression.  The grandstream has a program that auto 
detects this
and sets it up.
The audio codes is primarily programmed using bootp while we have an auto 
provisioning
system for the grandstream, we only use the audiocodes where we require 
hookflash.

Darcy

-----Original Message----- 
From: ocset
Sent: Saturday, September 03, 2011 9:21 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FXO/FXS card advice

Darcy

This info is very useful, especially since you have this setup running
in production.

One question  - the two ATA devices (Grandstream & AudioCodes) are
around the same price so could you give us some more info on why you
would prefer the one over the other (besides the hook flash feature).

Do they handle echo cancellation or do you deploy a separate solution
for that?

Thanks again for your help!

On 09/03/2011 08:35 PM, Darcy wrote:
> The grandstream has hookflash timers, but I have never been able
> to get it to work and info back from grandstream indicates is is not a
> pending implementation.
> For cases where I require a hook flash I use the audio codes fxo gateways.
> I never tested it on the freeswitch, we have own ap we wrote to handle it,
> we were doing this before we discovered freeswitch.  I know it works and 
> it
> is quite reliable. If I get time later today, I will give it a try on a
> freeswitch.
>
> Brand: AudioCodes
> Mfr's SKU: MP114/4O/SIP
> VoIP SIP Gateway, 4x FXO, 1x WAN, LBR Codecs,
>
> Here are the options you have for using it.  they have a variety of
> configurations, fxo/fxs combos
> a 4 port is in the 375 range.
>
> Hook Flash Option
> [HookFlashOption]
> Supported hook-flash Transport Type (method by which hook-flash is sent 
> and
> received).
> Valid options include:
> 0 = Hook-Flash indication isn?t sent (default)
> 1 = Send proprietary INFO message with Hook-Flash indication
> 4 = RFC 2833
>
> Darcy
>
> -----Original Message-----
> From: covici at ccs.covici.com
> Sent: Saturday, September 03, 2011 7:06 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] FXO/FXS card advice
>
> Does this ata support picking up call waiting while in a call from the
> fxo?  If it were to do that, I would get one right away.  The Digium FXO
> card I have has problems doing this with freetdm.
>
> Darcy<darcy at primrose.ws>  wrote:
>
>> Hi, for fxo's with the freeswitch I currently use:
>>
>> Mfr's SKU: GXW4104
>> Brand: Grandstream
>> 4 x FXO Gateway, 2 x LAN, 1 or 2 stage dialing, G.168 echo cancellation
>>
>> They work exceptionally well, easy to configure and can be auto 
>> configured
>> if you
>> are into that, we are.  You can also get an 8 port version.  I pay 220 
>> for
>> the 4 port
>> and 300 for the 8 port.  I have quite a few of them deployed.
>>
>> Darcy
>>
>>
>> -----Original Message-----
>> From: ocset
>> Sent: Saturday, September 03, 2011 1:38 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] FXO/FXS card advice
>>
>> Thank you all for your replies.
>>
>> Anthony, you reply has left me uncertain again. All of the ATA's listed
>> here (wiki.freeswitch.org/wiki/Interop_List) seem to have some
>> limitations and bugs which have to worked around.
>>
>> Your reply suggests that buying a OpenVox A400P04 card (currently only
>> $219 for 4 FXO's) would be more difficult to install and maintain that 4
>> ATA's. I would have thought that the FXS/FXO cards were engineered to
>> work with Freeswitch/Asterisk etc. without limitations like caller-id
>> not working or call-transfer not working as expected.
>>
>> I can see the benefit of an ATA since I can add them on as needed with
>> little or no hassle (except for the large power board to plug them all
>> into:-)
>>
>> I have only just got the SPA3102 receiving incoming calls and I am no
>> expert in either solution but would have thought that a 4 port FXO card
>> would be easier as it is made to purpose?
>>
>> ps. the PAP2T seems to be one of the better Linksys ATA's from the list
>> of features that work?
>>
>> Thanks again for all your help and suggestions.
>>
>> On 09/03/2011 05:54 AM, Anthony Minessale wrote:
>>> If you are doing FXO and you already have an ATA working that might be
>>> as good as you'll get.
>>> Its nice to not have to deal with TDM if you can help it and most
>>> analog ATA are cheap and effective and much less painful to deal with
>>> than analog cards.
>>>
>>>
>>>
>>> On Fri, Sep 2, 2011 at 8:39 AM, Saugort Dario Garcia Tovar
>>> <dgarcia at anew.com.ve>   wrote:
>>>> One think you should consider also, is the FXO/FXS board that you will
>>>> buy,
>>>> you have to check compatibility with your server/pc.
>>>>
>>>> For example openvox models A800P and A1200P has an issue with
>>>> motherboard
>>>> based on chipset H55 for example.
>>>>
>>>>
>>>> On 9/1/2011 9:59 PM, curriegrad2004 wrote:
>>>>
>>>> In the following order, most people usually recommend:
>>>>
>>>> 1. Sangoma
>>>> 2. Diginum/OpenVox
>>>> 3. Building your own Tormenta Zapatel Card (Only for serious
>>>> engineering types of people) from COTS
>>>>
>>>> On Thu, Sep 1, 2011 at 8:50 AM, Michel Daggelinckx
>>>> <michel.daggelinckx at gmail.com>   wrote:
>>>>
>>>> Digium and sangoma cards are high quality and work greast with FS
>>>>
>>>> Michel
>>>>
>>>>
>>>> On Thu, Sep 1, 2011 at 5:24 PM, ocset<ocset at the800group.com>   wrote:
>>>>
>>>> Hi
>>>>
>>>> As a new member of the forum, I am curious to know your experience with
>>>> FXO/FXS cards.
>>>>
>>>> I have a SPA3102 and have configured it to work with FS, but it feels a
>>>> bit like I am trying to make a square peg fit a round hole. I am hoping
>>>> to implement FS at a business which currently has 4 POTS lines and
>>>> would
>>>> prefer to use an internal IDE card for the job of integrating these
>>>> phone lines into FreeSwitch.
>>>>
>>>> Has anyone got some advice on which cards I should be looking at that
>>>> just "work" with FS? What about echo cancellation - is that something I
>>>> should just cater for or does it depend on the client situation? What
>>>> about software cancellation?
>>>>
>>>> This seems like one of those times when too much choice is a bad thing
>>>> and I need some guidance on what has worked for you.
>>>>
>>>> All advice would be greatly appreciated.
>>>>
>>>> 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
>>>>
>>>>
>>>> -----
>>>> No virus found in this message.
>>>> Checked by AVG - www.avg.com
>>>> Version: 10.0.1392 / Virus Database: 1520/3872 - Release Date: 09/02/11
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Atentamente,
>>>> Dario Garc?a
>>>> Consultor.
>>>>
>>>> CCCT, Nivel C2, Sector Yarey, Mz,
>>>> Ofc. MZ03a.
>>>> Caracas-Venezuela.
>>>> Tel?fono: +58 212 9081842
>>>> Cel: +58 412 2221515
>>>> dgarcia at anew.com.ve
>>>> http://www.anew.com.ve
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org


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



From covici at ccs.covici.com  Sat Sep  3 18:37:26 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sat, 03 Sep 2011 10:37:26 -0400
Subject: [Freeswitch-users] FXO/FXS card advice
In-Reply-To: <31E98B785AFA401EB9F3B4005EF2544E@DWP>
References: <4E5FA3B0.9030606@the800group.com><CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com><CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com><4E60DC90.9030201@anew.com.ve><CAKbxfG9LnStDCsZ7gopLfMq=37PTdeTrN-EwNrsa1tVvdq+T0g@mail.gmail.com><4E61BD42.9090109@the800group.com><6F8ACCA987D84F068F0B1A5D4AC515A4@DWP>
	<22926.1315047973@ccs.covici.com><8DF93FC7EB194882863E75A2B19AEFA6@DWP>
	<4E6229F2.8090000@the800group.com>
	<31E98B785AFA401EB9F3B4005EF2544E@DWP>
Message-ID: <18213.1315060646@ccs.covici.com>

And how do you  flash the hook --is it some kind of built in feature
code?

Darcy <darcy at primrose.ws> wrote:

> they both handle echo suppression.  The grandstream has a program that auto 
> detects this
> and sets it up.
> The audio codes is primarily programmed using bootp while we have an auto 
> provisioning
> system for the grandstream, we only use the audiocodes where we require 
> hookflash.
> 
> Darcy
> 
> -----Original Message----- 
> From: ocset
> Sent: Saturday, September 03, 2011 9:21 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FXO/FXS card advice
> 
> Darcy
> 
> This info is very useful, especially since you have this setup running
> in production.
> 
> One question  - the two ATA devices (Grandstream & AudioCodes) are
> around the same price so could you give us some more info on why you
> would prefer the one over the other (besides the hook flash feature).
> 
> Do they handle echo cancellation or do you deploy a separate solution
> for that?
> 
> Thanks again for your help!
> 
> On 09/03/2011 08:35 PM, Darcy wrote:
> > The grandstream has hookflash timers, but I have never been able
> > to get it to work and info back from grandstream indicates is is not a
> > pending implementation.
> > For cases where I require a hook flash I use the audio codes fxo gateways.
> > I never tested it on the freeswitch, we have own ap we wrote to handle it,
> > we were doing this before we discovered freeswitch.  I know it works and 
> > it
> > is quite reliable. If I get time later today, I will give it a try on a
> > freeswitch.
> >
> > Brand: AudioCodes
> > Mfr's SKU: MP114/4O/SIP
> > VoIP SIP Gateway, 4x FXO, 1x WAN, LBR Codecs,
> >
> > Here are the options you have for using it.  they have a variety of
> > configurations, fxo/fxs combos
> > a 4 port is in the 375 range.
> >
> > Hook Flash Option
> > [HookFlashOption]
> > Supported hook-flash Transport Type (method by which hook-flash is sent 
> > and
> > received).
> > Valid options include:
> > 0 = Hook-Flash indication isn?t sent (default)
> > 1 = Send proprietary INFO message with Hook-Flash indication
> > 4 = RFC 2833
> >
> > Darcy
> >
> > -----Original Message-----
> > From: covici at ccs.covici.com
> > Sent: Saturday, September 03, 2011 7:06 AM
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] FXO/FXS card advice
> >
> > Does this ata support picking up call waiting while in a call from the
> > fxo?  If it were to do that, I would get one right away.  The Digium FXO
> > card I have has problems doing this with freetdm.
> >
> > Darcy<darcy at primrose.ws>  wrote:
> >
> >> Hi, for fxo's with the freeswitch I currently use:
> >>
> >> Mfr's SKU: GXW4104
> >> Brand: Grandstream
> >> 4 x FXO Gateway, 2 x LAN, 1 or 2 stage dialing, G.168 echo cancellation
> >>
> >> They work exceptionally well, easy to configure and can be auto 
> >> configured
> >> if you
> >> are into that, we are.  You can also get an 8 port version.  I pay 220 
> >> for
> >> the 4 port
> >> and 300 for the 8 port.  I have quite a few of them deployed.
> >>
> >> Darcy
> >>
> >>
> >> -----Original Message-----
> >> From: ocset
> >> Sent: Saturday, September 03, 2011 1:38 AM
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] FXO/FXS card advice
> >>
> >> Thank you all for your replies.
> >>
> >> Anthony, you reply has left me uncertain again. All of the ATA's listed
> >> here (wiki.freeswitch.org/wiki/Interop_List) seem to have some
> >> limitations and bugs which have to worked around.
> >>
> >> Your reply suggests that buying a OpenVox A400P04 card (currently only
> >> $219 for 4 FXO's) would be more difficult to install and maintain that 4
> >> ATA's. I would have thought that the FXS/FXO cards were engineered to
> >> work with Freeswitch/Asterisk etc. without limitations like caller-id
> >> not working or call-transfer not working as expected.
> >>
> >> I can see the benefit of an ATA since I can add them on as needed with
> >> little or no hassle (except for the large power board to plug them all
> >> into:-)
> >>
> >> I have only just got the SPA3102 receiving incoming calls and I am no
> >> expert in either solution but would have thought that a 4 port FXO card
> >> would be easier as it is made to purpose?
> >>
> >> ps. the PAP2T seems to be one of the better Linksys ATA's from the list
> >> of features that work?
> >>
> >> Thanks again for all your help and suggestions.
> >>
> >> On 09/03/2011 05:54 AM, Anthony Minessale wrote:
> >>> If you are doing FXO and you already have an ATA working that might be
> >>> as good as you'll get.
> >>> Its nice to not have to deal with TDM if you can help it and most
> >>> analog ATA are cheap and effective and much less painful to deal with
> >>> than analog cards.
> >>>
> >>>
> >>>
> >>> On Fri, Sep 2, 2011 at 8:39 AM, Saugort Dario Garcia Tovar
> >>> <dgarcia at anew.com.ve>   wrote:
> >>>> One think you should consider also, is the FXO/FXS board that you will
> >>>> buy,
> >>>> you have to check compatibility with your server/pc.
> >>>>
> >>>> For example openvox models A800P and A1200P has an issue with
> >>>> motherboard
> >>>> based on chipset H55 for example.
> >>>>
> >>>>
> >>>> On 9/1/2011 9:59 PM, curriegrad2004 wrote:
> >>>>
> >>>> In the following order, most people usually recommend:
> >>>>
> >>>> 1. Sangoma
> >>>> 2. Diginum/OpenVox
> >>>> 3. Building your own Tormenta Zapatel Card (Only for serious
> >>>> engineering types of people) from COTS
> >>>>
> >>>> On Thu, Sep 1, 2011 at 8:50 AM, Michel Daggelinckx
> >>>> <michel.daggelinckx at gmail.com>   wrote:
> >>>>
> >>>> Digium and sangoma cards are high quality and work greast with FS
> >>>>
> >>>> Michel
> >>>>
> >>>>
> >>>> On Thu, Sep 1, 2011 at 5:24 PM, ocset<ocset at the800group.com>   wrote:
> >>>>
> >>>> Hi
> >>>>
> >>>> As a new member of the forum, I am curious to know your experience with
> >>>> FXO/FXS cards.
> >>>>
> >>>> I have a SPA3102 and have configured it to work with FS, but it feels a
> >>>> bit like I am trying to make a square peg fit a round hole. I am hoping
> >>>> to implement FS at a business which currently has 4 POTS lines and
> >>>> would
> >>>> prefer to use an internal IDE card for the job of integrating these
> >>>> phone lines into FreeSwitch.
> >>>>
> >>>> Has anyone got some advice on which cards I should be looking at that
> >>>> just "work" with FS? What about echo cancellation - is that something I
> >>>> should just cater for or does it depend on the client situation? What
> >>>> about software cancellation?
> >>>>
> >>>> This seems like one of those times when too much choice is a bad thing
> >>>> and I need some guidance on what has worked for you.
> >>>>
> >>>> All advice would be greatly appreciated.
> >>>>
> >>>> 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
> >>>>
> >>>>
> >>>> -----
> >>>> No virus found in this message.
> >>>> Checked by AVG - www.avg.com
> >>>> Version: 10.0.1392 / Virus Database: 1520/3872 - Release Date: 09/02/11
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> --
> >>>> Atentamente,
> >>>> Dario Garc?a
> >>>> Consultor.
> >>>>
> >>>> CCCT, Nivel C2, Sector Yarey, Mz,
> >>>> Ofc. MZ03a.
> >>>> Caracas-Venezuela.
> >>>> Tel?fono: +58 212 9081842
> >>>> Cel: +58 412 2221515
> >>>> dgarcia at anew.com.ve
> >>>> http://www.anew.com.ve
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 contact at aharm.de  Sat Sep  3 23:49:43 2011
From: contact at aharm.de (Alexander Harm)
Date: Sat, 3 Sep 2011 21:49:43 +0200
Subject: [Freeswitch-users] Need help cc Gateway (SIP trunk) setup for
	provider using Auth ID
In-Reply-To: <14C7CA65-D77E-47C6-B196-28DE75069208@seventhsignal.de>
References: <4777C308-2C0C-44A8-B713-E0D318546E6B@aharm.de>
	<14C7CA65-D77E-47C6-B196-28DE75069208@seventhsignal.de>
Message-ID: <3ADDEFE5-FA10-452D-86A4-816F5F505C0F@aharm.de>

The provider is Belgacom, a Belgian carrier.

Thanks.

On 03.09.2011, at 01:46, Michal Bielicki wrote:

> Which provider ?
> I can probably help you with most german ones.
> 
> Am 02.09.2011 um 22:59 schrieb Alexander Harm:
> 
>> My SIP provider uses
>> - User ID (same as Caller ID Number)
>> - Password
>> - Auth ID (different from User ID)
>> for registration. I have to admit that I'm completely at loss on how to configure freeSWITCH using Auth ID. I tried all combinations I could think off but I just keep getting 403 error messages.
>> Help is very much appreciated.
>> 
>> 
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> Michal Bielicki
> Gesch?ftsf?hrer / CEO
> 
> Seventh Signal Ltd. & Co. KG
> Weigandufer 45, B?ro 115, D-12059 Berlin
> Voice: +49 30 60988730
> 
> Amtsgericht 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
> 
> 
> 
> 
> ---- 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110903/5438df7b/attachment.html 

From jmesquita at freeswitch.org  Sun Sep  4 04:35:05 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 3 Sep 2011 19:35:05 -0500
Subject: [Freeswitch-users] FXO/FXS card advice
In-Reply-To: <18213.1315060646@ccs.covici.com>
References: <4E5FA3B0.9030606@the800group.com>
	<CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com>
	<CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com>
	<4E60DC90.9030201@anew.com.ve>
	<CAKbxfG9LnStDCsZ7gopLfMq=37PTdeTrN-EwNrsa1tVvdq+T0g@mail.gmail.com>
	<4E61BD42.9090109@the800group.com>
	<6F8ACCA987D84F068F0B1A5D4AC515A4@DWP>
	<22926.1315047973@ccs.covici.com>
	<8DF93FC7EB194882863E75A2B19AEFA6@DWP>
	<4E6229F2.8090000@the800group.com>
	<31E98B785AFA401EB9F3B4005EF2544E@DWP>
	<18213.1315060646@ccs.covici.com>
Message-ID: <CABH8QX1jpzjzTgS7BvSv7K9_gsm1WXy99P2coe6S3oih5Zxgug@mail.gmail.com>

Khomp does it. As a matter of fact it registers an app to ease the pain of
doinf flash based transfers, including when used with QSIG, EL7, E1LC and
LineSide if you use those...

JM
On Sep 3, 2011 9:37 AM, <covici at ccs.covici.com> wrote:
> And how do you flash the hook --is it some kind of built in feature
> code?
>
> Darcy <darcy at primrose.ws> wrote:
>
>> they both handle echo suppression. The grandstream has a program that
auto
>> detects this
>> and sets it up.
>> The audio codes is primarily programmed using bootp while we have an auto

>> provisioning
>> system for the grandstream, we only use the audiocodes where we require
>> hookflash.
>>
>> Darcy
>>
>> -----Original Message-----
>> From: ocset
>> Sent: Saturday, September 03, 2011 9:21 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] FXO/FXS card advice
>>
>> Darcy
>>
>> This info is very useful, especially since you have this setup running
>> in production.
>>
>> One question - the two ATA devices (Grandstream & AudioCodes) are
>> around the same price so could you give us some more info on why you
>> would prefer the one over the other (besides the hook flash feature).
>>
>> Do they handle echo cancellation or do you deploy a separate solution
>> for that?
>>
>> Thanks again for your help!
>>
>> On 09/03/2011 08:35 PM, Darcy wrote:
>> > The grandstream has hookflash timers, but I have never been able
>> > to get it to work and info back from grandstream indicates is is not a
>> > pending implementation.
>> > For cases where I require a hook flash I use the audio codes fxo
gateways.
>> > I never tested it on the freeswitch, we have own ap we wrote to handle
it,
>> > we were doing this before we discovered freeswitch. I know it works and

>> > it
>> > is quite reliable. If I get time later today, I will give it a try on a
>> > freeswitch.
>> >
>> > Brand: AudioCodes
>> > Mfr's SKU: MP114/4O/SIP
>> > VoIP SIP Gateway, 4x FXO, 1x WAN, LBR Codecs,
>> >
>> > Here are the options you have for using it. they have a variety of
>> > configurations, fxo/fxs combos
>> > a 4 port is in the 375 range.
>> >
>> > Hook Flash Option
>> > [HookFlashOption]
>> > Supported hook-flash Transport Type (method by which hook-flash is sent

>> > and
>> > received).
>> > Valid options include:
>> > 0 = Hook-Flash indication isn?t sent (default)
>> > 1 = Send proprietary INFO message with Hook-Flash indication
>> > 4 = RFC 2833
>> >
>> > Darcy
>> >
>> > -----Original Message-----
>> > From: covici at ccs.covici.com
>> > Sent: Saturday, September 03, 2011 7:06 AM
>> > To: FreeSWITCH Users Help
>> > Subject: Re: [Freeswitch-users] FXO/FXS card advice
>> >
>> > Does this ata support picking up call waiting while in a call from the
>> > fxo? If it were to do that, I would get one right away. The Digium FXO
>> > card I have has problems doing this with freetdm.
>> >
>> > Darcy<darcy at primrose.ws> wrote:
>> >
>> >> Hi, for fxo's with the freeswitch I currently use:
>> >>
>> >> Mfr's SKU: GXW4104
>> >> Brand: Grandstream
>> >> 4 x FXO Gateway, 2 x LAN, 1 or 2 stage dialing, G.168 echo
cancellation
>> >>
>> >> They work exceptionally well, easy to configure and can be auto
>> >> configured
>> >> if you
>> >> are into that, we are. You can also get an 8 port version. I pay 220
>> >> for
>> >> the 4 port
>> >> and 300 for the 8 port. I have quite a few of them deployed.
>> >>
>> >> Darcy
>> >>
>> >>
>> >> -----Original Message-----
>> >> From: ocset
>> >> Sent: Saturday, September 03, 2011 1:38 AM
>> >> To: freeswitch-users at lists.freeswitch.org
>> >> Subject: Re: [Freeswitch-users] FXO/FXS card advice
>> >>
>> >> Thank you all for your replies.
>> >>
>> >> Anthony, you reply has left me uncertain again. All of the ATA's
listed
>> >> here (wiki.freeswitch.org/wiki/Interop_List) seem to have some
>> >> limitations and bugs which have to worked around.
>> >>
>> >> Your reply suggests that buying a OpenVox A400P04 card (currently only
>> >> $219 for 4 FXO's) would be more difficult to install and maintain that
4
>> >> ATA's. I would have thought that the FXS/FXO cards were engineered to
>> >> work with Freeswitch/Asterisk etc. without limitations like caller-id
>> >> not working or call-transfer not working as expected.
>> >>
>> >> I can see the benefit of an ATA since I can add them on as needed with
>> >> little or no hassle (except for the large power board to plug them all
>> >> into:-)
>> >>
>> >> I have only just got the SPA3102 receiving incoming calls and I am no
>> >> expert in either solution but would have thought that a 4 port FXO
card
>> >> would be easier as it is made to purpose?
>> >>
>> >> ps. the PAP2T seems to be one of the better Linksys ATA's from the
list
>> >> of features that work?
>> >>
>> >> Thanks again for all your help and suggestions.
>> >>
>> >> On 09/03/2011 05:54 AM, Anthony Minessale wrote:
>> >>> If you are doing FXO and you already have an ATA working that might
be
>> >>> as good as you'll get.
>> >>> Its nice to not have to deal with TDM if you can help it and most
>> >>> analog ATA are cheap and effective and much less painful to deal with
>> >>> than analog cards.
>> >>>
>> >>>
>> >>>
>> >>> On Fri, Sep 2, 2011 at 8:39 AM, Saugort Dario Garcia Tovar
>> >>> <dgarcia at anew.com.ve> wrote:
>> >>>> One think you should consider also, is the FXO/FXS board that you
will
>> >>>> buy,
>> >>>> you have to check compatibility with your server/pc.
>> >>>>
>> >>>> For example openvox models A800P and A1200P has an issue with
>> >>>> motherboard
>> >>>> based on chipset H55 for example.
>> >>>>
>> >>>>
>> >>>> On 9/1/2011 9:59 PM, curriegrad2004 wrote:
>> >>>>
>> >>>> In the following order, most people usually recommend:
>> >>>>
>> >>>> 1. Sangoma
>> >>>> 2. Diginum/OpenVox
>> >>>> 3. Building your own Tormenta Zapatel Card (Only for serious
>> >>>> engineering types of people) from COTS
>> >>>>
>> >>>> On Thu, Sep 1, 2011 at 8:50 AM, Michel Daggelinckx
>> >>>> <michel.daggelinckx at gmail.com> wrote:
>> >>>>
>> >>>> Digium and sangoma cards are high quality and work greast with FS
>> >>>>
>> >>>> Michel
>> >>>>
>> >>>>
>> >>>> On Thu, Sep 1, 2011 at 5:24 PM, ocset<ocset at the800group.com> wrote:
>> >>>>
>> >>>> Hi
>> >>>>
>> >>>> As a new member of the forum, I am curious to know your experience
with
>> >>>> FXO/FXS cards.
>> >>>>
>> >>>> I have a SPA3102 and have configured it to work with FS, but it
feels a
>> >>>> bit like I am trying to make a square peg fit a round hole. I am
hoping
>> >>>> to implement FS at a business which currently has 4 POTS lines and
>> >>>> would
>> >>>> prefer to use an internal IDE card for the job of integrating these
>> >>>> phone lines into FreeSwitch.
>> >>>>
>> >>>> Has anyone got some advice on which cards I should be looking at
that
>> >>>> just "work" with FS? What about echo cancellation - is that
something I
>> >>>> should just cater for or does it depend on the client situation?
What
>> >>>> about software cancellation?
>> >>>>
>> >>>> This seems like one of those times when too much choice is a bad
thing
>> >>>> and I need some guidance on what has worked for you.
>> >>>>
>> >>>> All advice would be greatly appreciated.
>> >>>>
>> >>>> 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
>> >>>>
>> >>>>
>> >>>> -----
>> >>>> No virus found in this message.
>> >>>> Checked by AVG - www.avg.com
>> >>>> Version: 10.0.1392 / Virus Database: 1520/3872 - Release Date:
09/02/11
>> >>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>> --
>> >>>> Atentamente,
>> >>>> Dario Garc?a
>> >>>> Consultor.
>> >>>>
>> >>>> CCCT, Nivel C2, Sector Yarey, Mz,
>> >>>> Ofc. MZ03a.
>> >>>> Caracas-Venezuela.
>> >>>> Tel?fono: +58 212 9081842
>> >>>> Cel: +58 412 2221515
>> >>>> dgarcia at anew.com.ve
>> >>>> http://www.anew.com.ve
>> >>>>
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>>
>> >>>
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110903/eafa54e6/attachment-0001.html 

From covici at ccs.covici.com  Sun Sep  4 06:45:35 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sat, 03 Sep 2011 22:45:35 -0400
Subject: [Freeswitch-users] FXO/FXS card advice
In-Reply-To: <CABH8QX1jpzjzTgS7BvSv7K9_gsm1WXy99P2coe6S3oih5Zxgug@mail.gmail.com>
References: <4E5FA3B0.9030606@the800group.com>
	<CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com>
	<CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com>
	<4E60DC90.9030201@anew.com.ve>
	<CAKbxfG9LnStDCsZ7gopLfMq=37PTdeTrN-EwNrsa1tVvdq+T0g@mail.gmail.com>
	<4E61BD42.9090109@the800group.com>
	<6F8ACCA987D84F068F0B1A5D4AC515A4@DWP>
	<22926.1315047973@ccs.covici.com>
	<8DF93FC7EB194882863E75A2B19AEFA6@DWP>
	<4E6229F2.8090000@the800group.com>
	<31E98B785AFA401EB9F3B4005EF2544E@DWP>
	<18213.1315060646@ccs.covici.com>
	<CABH8QX1jpzjzTgS7BvSv7K9_gsm1WXy99P2coe6S3oih5Zxgug@mail.gmail.com>
Message-ID: <18847.1315104335@ccs.covici.com>

What is Khomp exactly -- a freeswitch dialplan app or what?  Sorry to be
ignorant.

Jo?o Mesquita <jmesquita at freeswitch.org> wrote:

> Khomp does it. As a matter of fact it registers an app to ease the pain of
> doinf flash based transfers, including when used with QSIG, EL7, E1LC and
> LineSide if you use those...
> 
> JM
> On Sep 3, 2011 9:37 AM, <covici at ccs.covici.com> wrote:
> > And how do you flash the hook --is it some kind of built in feature
> > code?
> >
> > Darcy <darcy at primrose.ws> wrote:
> >
> >> they both handle echo suppression. The grandstream has a program that
> auto
> >> detects this
> >> and sets it up.
> >> The audio codes is primarily programmed using bootp while we have an auto
> 
> >> provisioning
> >> system for the grandstream, we only use the audiocodes where we require
> >> hookflash.
> >>
> >> Darcy
> >>
> >> -----Original Message-----
> >> From: ocset
> >> Sent: Saturday, September 03, 2011 9:21 AM
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] FXO/FXS card advice
> >>
> >> Darcy
> >>
> >> This info is very useful, especially since you have this setup running
> >> in production.
> >>
> >> One question - the two ATA devices (Grandstream & AudioCodes) are
> >> around the same price so could you give us some more info on why you
> >> would prefer the one over the other (besides the hook flash feature).
> >>
> >> Do they handle echo cancellation or do you deploy a separate solution
> >> for that?
> >>
> >> Thanks again for your help!
> >>
> >> On 09/03/2011 08:35 PM, Darcy wrote:
> >> > The grandstream has hookflash timers, but I have never been able
> >> > to get it to work and info back from grandstream indicates is is not a
> >> > pending implementation.
> >> > For cases where I require a hook flash I use the audio codes fxo
> gateways.
> >> > I never tested it on the freeswitch, we have own ap we wrote to handle
> it,
> >> > we were doing this before we discovered freeswitch. I know it works and
> 
> >> > it
> >> > is quite reliable. If I get time later today, I will give it a try on a
> >> > freeswitch.
> >> >
> >> > Brand: AudioCodes
> >> > Mfr's SKU: MP114/4O/SIP
> >> > VoIP SIP Gateway, 4x FXO, 1x WAN, LBR Codecs,
> >> >
> >> > Here are the options you have for using it. they have a variety of
> >> > configurations, fxo/fxs combos
> >> > a 4 port is in the 375 range.
> >> >
> >> > Hook Flash Option
> >> > [HookFlashOption]
> >> > Supported hook-flash Transport Type (method by which hook-flash is sent
> 
> >> > and
> >> > received).
> >> > Valid options include:
> >> > 0 = Hook-Flash indication isn?t sent (default)
> >> > 1 = Send proprietary INFO message with Hook-Flash indication
> >> > 4 = RFC 2833
> >> >
> >> > Darcy
> >> >
> >> > -----Original Message-----
> >> > From: covici at ccs.covici.com
> >> > Sent: Saturday, September 03, 2011 7:06 AM
> >> > To: FreeSWITCH Users Help
> >> > Subject: Re: [Freeswitch-users] FXO/FXS card advice
> >> >
> >> > Does this ata support picking up call waiting while in a call from the
> >> > fxo? If it were to do that, I would get one right away. The Digium FXO
> >> > card I have has problems doing this with freetdm.
> >> >
> >> > Darcy<darcy at primrose.ws> wrote:
> >> >
> >> >> Hi, for fxo's with the freeswitch I currently use:
> >> >>
> >> >> Mfr's SKU: GXW4104
> >> >> Brand: Grandstream
> >> >> 4 x FXO Gateway, 2 x LAN, 1 or 2 stage dialing, G.168 echo
> cancellation
> >> >>
> >> >> They work exceptionally well, easy to configure and can be auto
> >> >> configured
> >> >> if you
> >> >> are into that, we are. You can also get an 8 port version. I pay 220
> >> >> for
> >> >> the 4 port
> >> >> and 300 for the 8 port. I have quite a few of them deployed.
> >> >>
> >> >> Darcy
> >> >>
> >> >>
> >> >> -----Original Message-----
> >> >> From: ocset
> >> >> Sent: Saturday, September 03, 2011 1:38 AM
> >> >> To: freeswitch-users at lists.freeswitch.org
> >> >> Subject: Re: [Freeswitch-users] FXO/FXS card advice
> >> >>
> >> >> Thank you all for your replies.
> >> >>
> >> >> Anthony, you reply has left me uncertain again. All of the ATA's
> listed
> >> >> here (wiki.freeswitch.org/wiki/Interop_List) seem to have some
> >> >> limitations and bugs which have to worked around.
> >> >>
> >> >> Your reply suggests that buying a OpenVox A400P04 card (currently only
> >> >> $219 for 4 FXO's) would be more difficult to install and maintain that
> 4
> >> >> ATA's. I would have thought that the FXS/FXO cards were engineered to
> >> >> work with Freeswitch/Asterisk etc. without limitations like caller-id
> >> >> not working or call-transfer not working as expected.
> >> >>
> >> >> I can see the benefit of an ATA since I can add them on as needed with
> >> >> little or no hassle (except for the large power board to plug them all
> >> >> into:-)
> >> >>
> >> >> I have only just got the SPA3102 receiving incoming calls and I am no
> >> >> expert in either solution but would have thought that a 4 port FXO
> card
> >> >> would be easier as it is made to purpose?
> >> >>
> >> >> ps. the PAP2T seems to be one of the better Linksys ATA's from the
> list
> >> >> of features that work?
> >> >>
> >> >> Thanks again for all your help and suggestions.
> >> >>
> >> >> On 09/03/2011 05:54 AM, Anthony Minessale wrote:
> >> >>> If you are doing FXO and you already have an ATA working that might
> be
> >> >>> as good as you'll get.
> >> >>> Its nice to not have to deal with TDM if you can help it and most
> >> >>> analog ATA are cheap and effective and much less painful to deal with
> >> >>> than analog cards.
> >> >>>
> >> >>>
> >> >>>
> >> >>> On Fri, Sep 2, 2011 at 8:39 AM, Saugort Dario Garcia Tovar
> >> >>> <dgarcia at anew.com.ve> wrote:
> >> >>>> One think you should consider also, is the FXO/FXS board that you
> will
> >> >>>> buy,
> >> >>>> you have to check compatibility with your server/pc.
> >> >>>>
> >> >>>> For example openvox models A800P and A1200P has an issue with
> >> >>>> motherboard
> >> >>>> based on chipset H55 for example.
> >> >>>>
> >> >>>>
> >> >>>> On 9/1/2011 9:59 PM, curriegrad2004 wrote:
> >> >>>>
> >> >>>> In the following order, most people usually recommend:
> >> >>>>
> >> >>>> 1. Sangoma
> >> >>>> 2. Diginum/OpenVox
> >> >>>> 3. Building your own Tormenta Zapatel Card (Only for serious
> >> >>>> engineering types of people) from COTS
> >> >>>>
> >> >>>> On Thu, Sep 1, 2011 at 8:50 AM, Michel Daggelinckx
> >> >>>> <michel.daggelinckx at gmail.com> wrote:
> >> >>>>
> >> >>>> Digium and sangoma cards are high quality and work greast with FS
> >> >>>>
> >> >>>> Michel
> >> >>>>
> >> >>>>
> >> >>>> On Thu, Sep 1, 2011 at 5:24 PM, ocset<ocset at the800group.com> wrote:
> >> >>>>
> >> >>>> Hi
> >> >>>>
> >> >>>> As a new member of the forum, I am curious to know your experience
> with
> >> >>>> FXO/FXS cards.
> >> >>>>
> >> >>>> I have a SPA3102 and have configured it to work with FS, but it
> feels a
> >> >>>> bit like I am trying to make a square peg fit a round hole. I am
> hoping
> >> >>>> to implement FS at a business which currently has 4 POTS lines and
> >> >>>> would
> >> >>>> prefer to use an internal IDE card for the job of integrating these
> >> >>>> phone lines into FreeSwitch.
> >> >>>>
> >> >>>> Has anyone got some advice on which cards I should be looking at
> that
> >> >>>> just "work" with FS? What about echo cancellation - is that
> something I
> >> >>>> should just cater for or does it depend on the client situation?
> What
> >> >>>> about software cancellation?
> >> >>>>
> >> >>>> This seems like one of those times when too much choice is a bad
> thing
> >> >>>> and I need some guidance on what has worked for you.
> >> >>>>
> >> >>>> All advice would be greatly appreciated.
> >> >>>>
> >> >>>> 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
> >> >>>>
> >> >>>>
> >> >>>> -----
> >> >>>> No virus found in this message.
> >> >>>> Checked by AVG - www.avg.com
> >> >>>> Version: 10.0.1392 / Virus Database: 1520/3872 - Release Date:
> 09/02/11
> >> >>>>
> >> >>>>
> >> >>>>
> >> >>>>
> >> >>>> --
> >> >>>> Atentamente,
> >> >>>> Dario Garc?a
> >> >>>> Consultor.
> >> >>>>
> >> >>>> CCCT, Nivel C2, Sector Yarey, Mz,
> >> >>>> Ofc. MZ03a.
> >> >>>> Caracas-Venezuela.
> >> >>>> Tel?fono: +58 212 9081842
> >> >>>> Cel: +58 412 2221515
> >> >>>> dgarcia at anew.com.ve
> >> >>>> http://www.anew.com.ve
> >> >>>>
> >> >>>> FreeSWITCH-users mailing list
> >> >>>> FreeSWITCH-users at lists.freeswitch.org
> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>>> http://www.freeswitch.org
> >> >>>>
> >> >>>>
> >> >>>
> >> >>
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >>
> >> >>
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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 contact at aharm.de  Sun Sep  4 09:33:16 2011
From: contact at aharm.de (Alexander Harm)
Date: Sun, 4 Sep 2011 07:33:16 +0200
Subject: [Freeswitch-users] Need help cc Gateway (SIP trunk) setup for
	provider using Auth ID
In-Reply-To: <14C7CA65-D77E-47C6-B196-28DE75069208@seventhsignal.de>
References: <4777C308-2C0C-44A8-B713-E0D318546E6B@aharm.de>
	<14C7CA65-D77E-47C6-B196-28DE75069208@seventhsignal.de>
Message-ID: <B7A8010F-E61E-4A12-8F74-CF4F161D0611@aharm.de>

I found this website outlining the setup for Belgacom for trixbox. Will try to relate it to freeSWITCH:

http://www.fonality.com/trixbox/forums/trixbox-forums/sip-and-iax-trunks-and-providers/belgacom-i-talk

not sure where to edit the register string in freeSWITCH.

regards, alexander

On 03.09.2011, at 01:46, Michal Bielicki wrote:

> Which provider ?
> I can probably help you with most german ones.
> 
> Am 02.09.2011 um 22:59 schrieb Alexander Harm:
> 
>> My SIP provider uses
>> - User ID (same as Caller ID Number)
>> - Password
>> - Auth ID (different from User ID)
>> for registration. I have to admit that I'm completely at loss on how to configure freeSWITCH using Auth ID. I tried all combinations I could think off but I just keep getting 403 error messages.
>> Help is very much appreciated.
>> 
>> 
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> Michal Bielicki
> Gesch?ftsf?hrer / CEO
> 
> Seventh Signal Ltd. & Co. KG
> Weigandufer 45, B?ro 115, D-12059 Berlin
> Voice: +49 30 60988730
> 
> Amtsgericht 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
> 
> 
> 
> 
> ---- 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110904/9f0a9b58/attachment.html 

From steveu at coppice.org  Sun Sep  4 09:53:10 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 04 Sep 2011 13:53:10 +0800
Subject: [Freeswitch-users] FXO/FXS card advice
In-Reply-To: <CABH8QX1jpzjzTgS7BvSv7K9_gsm1WXy99P2coe6S3oih5Zxgug@mail.gmail.com>
References: <4E5FA3B0.9030606@the800group.com>	<CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com>	<CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com>	<4E60DC90.9030201@anew.com.ve>	<CAKbxfG9LnStDCsZ7gopLfMq=37PTdeTrN-EwNrsa1tVvdq+T0g@mail.gmail.com>	<4E61BD42.9090109@the800group.com>	<6F8ACCA987D84F068F0B1A5D4AC515A4@DWP>	<22926.1315047973@ccs.covici.com>	<8DF93FC7EB194882863E75A2B19AEFA6@DWP>	<4E6229F2.8090000@the800group.com>	<31E98B785AFA401EB9F3B4005EF2544E@DWP>	<18213.1315060646@ccs.covici.com>
	<CABH8QX1jpzjzTgS7BvSv7K9_gsm1WXy99P2coe6S3oih5Zxgug@mail.gmail.com>
Message-ID: <4E631246.60808@coppice.org>

On 09/04/2011 08:35 AM, Jo?o Mesquita wrote:
>
> Khomp does it. As a matter of fact it registers an app to ease the 
> pain of doinf flash based transfers, including when used with QSIG, 
> EL7, E1LC and LineSide if you use those...
>
> JM
>
>
Does Khomp implement those protocols, or are you just saying hey 
interwork well with them?

Steve



From sunwood360 at gmail.com  Sun Sep  4 19:26:31 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Sun, 4 Sep 2011 08:26:31 -0700
Subject: [Freeswitch-users] Compiling Freeswitch for Android?
In-Reply-To: <296C71FF-C5F6-41E8-A08C-5DE3AE61300F@avgs.ca>
References: <1314959406122-6753409.post@n2.nabble.com>
	<BD97C6DD-E80A-4CE8-8599-C5E851150454@avgs.ca>
	<1314962913219-6753523.post@n2.nabble.com>
	<296C71FF-C5F6-41E8-A08C-5DE3AE61300F@avgs.ca>
Message-ID: <CAKq5mS6fXXjJafcyyL=oj+fbhESnvOL_j=n1s7ZV56Uw4nvaAw@mail.gmail.com>

I am also very interested in your progress.

Thanks
On Sep 2, 2011 6:29 AM, "Mathieu Rene" <mrene_lists at avgs.ca> wrote:
> It'll be released with the main distribution :)
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2011-09-02, at 1:28 PM, GillesToo wrote:
>
>> Great news :-) Unfortunately, I don't have the technical skills to port
>> Freeswitch. Did you put up a web site so that we can keep an eye on the
>> project?
>>
>> --
>> View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Compiling-Freeswitch-for-Android-tp6753409p6753523.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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110904/6172aaf0/attachment-0001.html 

From peter.schrock at gmail.com  Sun Sep  4 20:07:12 2011
From: peter.schrock at gmail.com (Peter Schrock)
Date: Sun, 4 Sep 2011 09:07:12 -0700
Subject: [Freeswitch-users] Compiling Freeswitch for Android?
In-Reply-To: <CAKq5mS6fXXjJafcyyL=oj+fbhESnvOL_j=n1s7ZV56Uw4nvaAw@mail.gmail.com>
References: <1314959406122-6753409.post@n2.nabble.com>
	<BD97C6DD-E80A-4CE8-8599-C5E851150454@avgs.ca>
	<1314962913219-6753523.post@n2.nabble.com>
	<296C71FF-C5F6-41E8-A08C-5DE3AE61300F@avgs.ca>
	<CAKq5mS6fXXjJafcyyL=oj+fbhESnvOL_j=n1s7ZV56Uw4nvaAw@mail.gmail.com>
Message-ID: <6918529636939195413@unknownmsgid>

What about iPhone since it's Unix based?

Peter

On Sep 4, 2011, at 8:30 AM, envelopes envelopes <sunwood360 at gmail.com>
wrote:

I am also very interested in your progress.

Thanks
On Sep 2, 2011 6:29 AM, "Mathieu Rene" <mrene_lists at avgs.ca> wrote:
> It'll be released with the main distribution :)
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2011-09-02, at 1:28 PM, GillesToo wrote:
>
>> Great news :-) Unfortunately, I don't have the technical skills to port
>> Freeswitch. Did you put up a web site so that we can keep an eye on the
>> project?
>>
>> --
>> View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Compiling-Freeswitch-for-Android-tp6753409p6753523.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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110904/3aedf2cd/attachment.html 

From jmesquita at freeswitch.org  Sun Sep  4 21:22:20 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 4 Sep 2011 12:22:20 -0500
Subject: [Freeswitch-users] FXO/FXS card advice
In-Reply-To: <4E631246.60808@coppice.org>
References: <4E5FA3B0.9030606@the800group.com>
	<CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com>
	<CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com>
	<4E60DC90.9030201@anew.com.ve>
	<CAKbxfG9LnStDCsZ7gopLfMq=37PTdeTrN-EwNrsa1tVvdq+T0g@mail.gmail.com>
	<4E61BD42.9090109@the800group.com>
	<6F8ACCA987D84F068F0B1A5D4AC515A4@DWP>
	<22926.1315047973@ccs.covici.com>
	<8DF93FC7EB194882863E75A2B19AEFA6@DWP>
	<4E6229F2.8090000@the800group.com>
	<31E98B785AFA401EB9F3B4005EF2544E@DWP>
	<18213.1315060646@ccs.covici.com>
	<CABH8QX1jpzjzTgS7BvSv7K9_gsm1WXy99P2coe6S3oih5Zxgug@mail.gmail.com>
	<4E631246.60808@coppice.org>
Message-ID: <CABH8QX3o2cv1z0ezk+tr7JzMyukHJbi15r+mRH9uhX2f=pjm0Q@mail.gmail.com>

They do implement those protocols either on the DSP level such as R2 based
or via their low level API called K3L. The API is what talks to mod_khomp.

JM
On Sep 4, 2011 2:54 AM, "Steve Underwood" <steveu at coppice.org> wrote:
> On 09/04/2011 08:35 AM, Jo?o Mesquita wrote:
>>
>> Khomp does it. As a matter of fact it registers an app to ease the
>> pain of doinf flash based transfers, including when used with QSIG,
>> EL7, E1LC and LineSide if you use those...
>>
>> JM
>>
>>
> Does Khomp implement those protocols, or are you just saying hey
> interwork well with them?
>
> 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110904/5d8d3bb2/attachment.html 

From rhuddleston at gmail.com  Sun Sep  4 21:22:30 2011
From: rhuddleston at gmail.com (Robert-iPhone)
Date: Sun, 4 Sep 2011 13:22:30 -0400
Subject: [Freeswitch-users] Compiling Freeswitch for Android?
In-Reply-To: <6918529636939195413@unknownmsgid>
References: <1314959406122-6753409.post@n2.nabble.com>
	<BD97C6DD-E80A-4CE8-8599-C5E851150454@avgs.ca>
	<1314962913219-6753523.post@n2.nabble.com>
	<296C71FF-C5F6-41E8-A08C-5DE3AE61300F@avgs.ca>
	<CAKq5mS6fXXjJafcyyL=oj+fbhESnvOL_j=n1s7ZV56Uw4nvaAw@mail.gmail.com>
	<6918529636939195413@unknownmsgid>
Message-ID: <779CE20F-64B3-4586-B712-9E55561C4794@gmail.com>

I dont understand all the interest here. Nobody is going to provide low level access to radio or phone to do anything of value with Freeswitch or Asterisk (someone posted there too).

If you really want to do something crazy and are trying to use cell network - get a gsm modem or similar and yank the sim card.

Are iphone and android powerfull enough - YES but again Apple and Google arent going to just give away access to low level functions at will.



Sent from my iPhone

On Sep 4, 2011, at 12:07 PM, Peter Schrock <peter.schrock at gmail.com> wrote:

> What about iPhone since it's Unix based?
> 
> Peter
> 
> On Sep 4, 2011, at 8:30 AM, envelopes envelopes <sunwood360 at gmail.com> wrote:
> 
>> I am also very interested in your progress.
>> 
>> Thanks
>> 
>> On Sep 2, 2011 6:29 AM, "Mathieu Rene" <mrene_lists at avgs.ca> wrote:
>> > It'll be released with the main distribution :)
>> > 
>> > Mathieu Rene
>> > Avant-Garde Solutions Inc
>> > Office: + 1 (514) 664-1044 x100
>> > Cell: +1 (514) 664-1044 x200
>> > mrene at avgs.ca
>> > 
>> > 
>> > 
>> > 
>> > On 2011-09-02, at 1:28 PM, GillesToo wrote:
>> > 
>> >> Great news :-) Unfortunately, I don't have the technical skills to port
>> >> Freeswitch. Did you put up a web site so that we can keep an eye on the
>> >> project?
>> >> 
>> >> --
>> >> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Compiling-Freeswitch-for-Android-tp6753409p6753523.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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110904/07229708/attachment.html 

From jmesquita at freeswitch.org  Sun Sep  4 21:24:17 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 4 Sep 2011 12:24:17 -0500
Subject: [Freeswitch-users] FXO/FXS card advice
In-Reply-To: <18847.1315104335@ccs.covici.com>
References: <4E5FA3B0.9030606@the800group.com>
	<CAFYZzbuXmywGBe8bLtOMjWWiO2q-W0Z1RmqrusPrQ2oGc+SM9A@mail.gmail.com>
	<CAMWiMy-9h=Y8_o9_DrgX=L4LQ1qsJvwJDat5wc2iZXkb0wW1qg@mail.gmail.com>
	<4E60DC90.9030201@anew.com.ve>
	<CAKbxfG9LnStDCsZ7gopLfMq=37PTdeTrN-EwNrsa1tVvdq+T0g@mail.gmail.com>
	<4E61BD42.9090109@the800group.com>
	<6F8ACCA987D84F068F0B1A5D4AC515A4@DWP>
	<22926.1315047973@ccs.covici.com>
	<8DF93FC7EB194882863E75A2B19AEFA6@DWP>
	<4E6229F2.8090000@the800group.com>
	<31E98B785AFA401EB9F3B4005EF2544E@DWP>
	<18213.1315060646@ccs.covici.com>
	<CABH8QX1jpzjzTgS7BvSv7K9_gsm1WXy99P2coe6S3oih5Zxgug@mail.gmail.com>
	<18847.1315104335@ccs.covici.com>
Message-ID: <CABH8QX0UXaFVtXZB4ZWn2Vz5yKQnpcrxJrkV-oLpxGp1u=7SxQ@mail.gmail.com>

Khomp is.a brazilian hardcard manufacturer. http://www.khomp.com.br

It is the only one besides sangoma that officially supports freeswitch using
their own development.
On Sep 3, 2011 11:46 PM, <covici at ccs.covici.com> wrote:
> What is Khomp exactly -- a freeswitch dialplan app or what? Sorry to be
> ignorant.
>
> Jo?o Mesquita <jmesquita at freeswitch.org> wrote:
>
>> Khomp does it. As a matter of fact it registers an app to ease the pain
of
>> doinf flash based transfers, including when used with QSIG, EL7, E1LC and
>> LineSide if you use those...
>>
>> JM
>> On Sep 3, 2011 9:37 AM, <covici at ccs.covici.com> wrote:
>> > And how do you flash the hook --is it some kind of built in feature
>> > code?
>> >
>> > Darcy <darcy at primrose.ws> wrote:
>> >
>> >> they both handle echo suppression. The grandstream has a program that
>> auto
>> >> detects this
>> >> and sets it up.
>> >> The audio codes is primarily programmed using bootp while we have an
auto
>>
>> >> provisioning
>> >> system for the grandstream, we only use the audiocodes where we
require
>> >> hookflash.
>> >>
>> >> Darcy
>> >>
>> >> -----Original Message-----
>> >> From: ocset
>> >> Sent: Saturday, September 03, 2011 9:21 AM
>> >> To: freeswitch-users at lists.freeswitch.org
>> >> Subject: Re: [Freeswitch-users] FXO/FXS card advice
>> >>
>> >> Darcy
>> >>
>> >> This info is very useful, especially since you have this setup running
>> >> in production.
>> >>
>> >> One question - the two ATA devices (Grandstream & AudioCodes) are
>> >> around the same price so could you give us some more info on why you
>> >> would prefer the one over the other (besides the hook flash feature).
>> >>
>> >> Do they handle echo cancellation or do you deploy a separate solution
>> >> for that?
>> >>
>> >> Thanks again for your help!
>> >>
>> >> On 09/03/2011 08:35 PM, Darcy wrote:
>> >> > The grandstream has hookflash timers, but I have never been able
>> >> > to get it to work and info back from grandstream indicates is is not
a
>> >> > pending implementation.
>> >> > For cases where I require a hook flash I use the audio codes fxo
>> gateways.
>> >> > I never tested it on the freeswitch, we have own ap we wrote to
handle
>> it,
>> >> > we were doing this before we discovered freeswitch. I know it works
and
>>
>> >> > it
>> >> > is quite reliable. If I get time later today, I will give it a try
on a
>> >> > freeswitch.
>> >> >
>> >> > Brand: AudioCodes
>> >> > Mfr's SKU: MP114/4O/SIP
>> >> > VoIP SIP Gateway, 4x FXO, 1x WAN, LBR Codecs,
>> >> >
>> >> > Here are the options you have for using it. they have a variety of
>> >> > configurations, fxo/fxs combos
>> >> > a 4 port is in the 375 range.
>> >> >
>> >> > Hook Flash Option
>> >> > [HookFlashOption]
>> >> > Supported hook-flash Transport Type (method by which hook-flash is
sent
>>
>> >> > and
>> >> > received).
>> >> > Valid options include:
>> >> > 0 = Hook-Flash indication isn?t sent (default)
>> >> > 1 = Send proprietary INFO message with Hook-Flash indication
>> >> > 4 = RFC 2833
>> >> >
>> >> > Darcy
>> >> >
>> >> > -----Original Message-----
>> >> > From: covici at ccs.covici.com
>> >> > Sent: Saturday, September 03, 2011 7:06 AM
>> >> > To: FreeSWITCH Users Help
>> >> > Subject: Re: [Freeswitch-users] FXO/FXS card advice
>> >> >
>> >> > Does this ata support picking up call waiting while in a call from
the
>> >> > fxo? If it were to do that, I would get one right away. The Digium
FXO
>> >> > card I have has problems doing this with freetdm.
>> >> >
>> >> > Darcy<darcy at primrose.ws> wrote:
>> >> >
>> >> >> Hi, for fxo's with the freeswitch I currently use:
>> >> >>
>> >> >> Mfr's SKU: GXW4104
>> >> >> Brand: Grandstream
>> >> >> 4 x FXO Gateway, 2 x LAN, 1 or 2 stage dialing, G.168 echo
>> cancellation
>> >> >>
>> >> >> They work exceptionally well, easy to configure and can be auto
>> >> >> configured
>> >> >> if you
>> >> >> are into that, we are. You can also get an 8 port version. I pay
220
>> >> >> for
>> >> >> the 4 port
>> >> >> and 300 for the 8 port. I have quite a few of them deployed.
>> >> >>
>> >> >> Darcy
>> >> >>
>> >> >>
>> >> >> -----Original Message-----
>> >> >> From: ocset
>> >> >> Sent: Saturday, September 03, 2011 1:38 AM
>> >> >> To: freeswitch-users at lists.freeswitch.org
>> >> >> Subject: Re: [Freeswitch-users] FXO/FXS card advice
>> >> >>
>> >> >> Thank you all for your replies.
>> >> >>
>> >> >> Anthony, you reply has left me uncertain again. All of the ATA's
>> listed
>> >> >> here (wiki.freeswitch.org/wiki/Interop_List) seem to have some
>> >> >> limitations and bugs which have to worked around.
>> >> >>
>> >> >> Your reply suggests that buying a OpenVox A400P04 card (currently
only
>> >> >> $219 for 4 FXO's) would be more difficult to install and maintain
that
>> 4
>> >> >> ATA's. I would have thought that the FXS/FXO cards were engineered
to
>> >> >> work with Freeswitch/Asterisk etc. without limitations like
caller-id
>> >> >> not working or call-transfer not working as expected.
>> >> >>
>> >> >> I can see the benefit of an ATA since I can add them on as needed
with
>> >> >> little or no hassle (except for the large power board to plug them
all
>> >> >> into:-)
>> >> >>
>> >> >> I have only just got the SPA3102 receiving incoming calls and I am
no
>> >> >> expert in either solution but would have thought that a 4 port FXO
>> card
>> >> >> would be easier as it is made to purpose?
>> >> >>
>> >> >> ps. the PAP2T seems to be one of the better Linksys ATA's from the
>> list
>> >> >> of features that work?
>> >> >>
>> >> >> Thanks again for all your help and suggestions.
>> >> >>
>> >> >> On 09/03/2011 05:54 AM, Anthony Minessale wrote:
>> >> >>> If you are doing FXO and you already have an ATA working that
might
>> be
>> >> >>> as good as you'll get.
>> >> >>> Its nice to not have to deal with TDM if you can help it and most
>> >> >>> analog ATA are cheap and effective and much less painful to deal
with
>> >> >>> than analog cards.
>> >> >>>
>> >> >>>
>> >> >>>
>> >> >>> On Fri, Sep 2, 2011 at 8:39 AM, Saugort Dario Garcia Tovar
>> >> >>> <dgarcia at anew.com.ve> wrote:
>> >> >>>> One think you should consider also, is the FXO/FXS board that you
>> will
>> >> >>>> buy,
>> >> >>>> you have to check compatibility with your server/pc.
>> >> >>>>
>> >> >>>> For example openvox models A800P and A1200P has an issue with
>> >> >>>> motherboard
>> >> >>>> based on chipset H55 for example.
>> >> >>>>
>> >> >>>>
>> >> >>>> On 9/1/2011 9:59 PM, curriegrad2004 wrote:
>> >> >>>>
>> >> >>>> In the following order, most people usually recommend:
>> >> >>>>
>> >> >>>> 1. Sangoma
>> >> >>>> 2. Diginum/OpenVox
>> >> >>>> 3. Building your own Tormenta Zapatel Card (Only for serious
>> >> >>>> engineering types of people) from COTS
>> >> >>>>
>> >> >>>> On Thu, Sep 1, 2011 at 8:50 AM, Michel Daggelinckx
>> >> >>>> <michel.daggelinckx at gmail.com> wrote:
>> >> >>>>
>> >> >>>> Digium and sangoma cards are high quality and work greast with FS
>> >> >>>>
>> >> >>>> Michel
>> >> >>>>
>> >> >>>>
>> >> >>>> On Thu, Sep 1, 2011 at 5:24 PM, ocset<ocset at the800group.com>
wrote:
>> >> >>>>
>> >> >>>> Hi
>> >> >>>>
>> >> >>>> As a new member of the forum, I am curious to know your
experience
>> with
>> >> >>>> FXO/FXS cards.
>> >> >>>>
>> >> >>>> I have a SPA3102 and have configured it to work with FS, but it
>> feels a
>> >> >>>> bit like I am trying to make a square peg fit a round hole. I am
>> hoping
>> >> >>>> to implement FS at a business which currently has 4 POTS lines
and
>> >> >>>> would
>> >> >>>> prefer to use an internal IDE card for the job of integrating
these
>> >> >>>> phone lines into FreeSwitch.
>> >> >>>>
>> >> >>>> Has anyone got some advice on which cards I should be looking at
>> that
>> >> >>>> just "work" with FS? What about echo cancellation - is that
>> something I
>> >> >>>> should just cater for or does it depend on the client situation?
>> What
>> >> >>>> about software cancellation?
>> >> >>>>
>> >> >>>> This seems like one of those times when too much choice is a bad
>> thing
>> >> >>>> and I need some guidance on what has worked for you.
>> >> >>>>
>> >> >>>> All advice would be greatly appreciated.
>> >> >>>>
>> >> >>>> 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
>> >> >>>>
>> >> >>>>
>> >> >>>> -----
>> >> >>>> No virus found in this message.
>> >> >>>> Checked by AVG - www.avg.com
>> >> >>>> Version: 10.0.1392 / Virus Database: 1520/3872 - Release Date:
>> 09/02/11
>> >> >>>>
>> >> >>>>
>> >> >>>>
>> >> >>>>
>> >> >>>> --
>> >> >>>> Atentamente,
>> >> >>>> Dario Garc?a
>> >> >>>> Consultor.
>> >> >>>>
>> >> >>>> CCCT, Nivel C2, Sector Yarey, Mz,
>> >> >>>> Ofc. MZ03a.
>> >> >>>> Caracas-Venezuela.
>> >> >>>> Tel?fono: +58 212 9081842
>> >> >>>> Cel: +58 412 2221515
>> >> >>>> dgarcia at anew.com.ve
>> >> >>>> http://www.anew.com.ve
>> >> >>>>
>> >> >>>> FreeSWITCH-users mailing list
>> >> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>>> http://www.freeswitch.org
>> >> >>>>
>> >> >>>>
>> >> >>>
>> >> >>
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >>
>> >> >>
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >>
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at 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
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110904/b822c6f8/attachment-0001.html 

From edpimentl at gmail.com  Sun Sep  4 22:31:57 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Sun, 4 Sep 2011 14:31:57 -0400
Subject: [Freeswitch-users] Compiling Freeswitch for Android?
In-Reply-To: <779CE20F-64B3-4586-B712-9E55561C4794@gmail.com>
References: <1314959406122-6753409.post@n2.nabble.com>
	<BD97C6DD-E80A-4CE8-8599-C5E851150454@avgs.ca>
	<1314962913219-6753523.post@n2.nabble.com>
	<296C71FF-C5F6-41E8-A08C-5DE3AE61300F@avgs.ca>
	<CAKq5mS6fXXjJafcyyL=oj+fbhESnvOL_j=n1s7ZV56Uw4nvaAw@mail.gmail.com>
	<6918529636939195413@unknownmsgid>
	<779CE20F-64B3-4586-B712-9E55561C4794@gmail.com>
Message-ID: <CANMSB1YEe+eFmo7y+YCgfSvjUE=MqVntEAE2OKb4R32xs--f0A@mail.gmail.com>

A  source for multifunction radios....
http://www.marvell.com/platforms/smartphones.jsp


From darcy at thevoiphighway.com  Sun Sep  4 22:33:16 2011
From: darcy at thevoiphighway.com (Darcy Primrose)
Date: Sun, 04 Sep 2011 14:33:16 -0400
Subject: [Freeswitch-users] sending ani using originate with fs_cli
Message-ID: <4E63C46C.40308@thevoiphighway.com>

I have tried various variables using the originate command in fs_cli and 
I cannot get the ani and name to be sent, it populates the cid_num with 
'0000000000'.  Anybody have any ideas that could help me with this.

originate 
{caller-ANI='6134545851'}sofia/internal/16149753370 at 208.185.148.43 
&txfax(/tmp/FAX.tif)

Darcy



From peter.olsson at visionutveckling.se  Sun Sep  4 23:11:44 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 4 Sep 2011 21:11:44 +0200
Subject: [Freeswitch-users] sending ani using originate with fs_cli
In-Reply-To: <4E63C46C.40308@thevoiphighway.com>
References: <4E63C46C.40308@thevoiphighway.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0E0@cooper>

Use variable origination_caller_id_number. Check out this example: http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_number

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Darcy Primrose [darcy at thevoiphighway.com]
Skickat: den 4 september 2011 20:33
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] sending ani using originate with fs_cli

I have tried various variables using the originate command in fs_cli and
I cannot get the ani and name to be sent, it populates the cid_num with
'0000000000'.  Anybody have any ideas that could help me with this.

originate
{caller-ANI='6134545851'}sofia/internal/16149753370 at 208.185.148.43
&txfax(/tmp/FAX.tif)

Darcy



FreeSWITCH-users mailing list
FreeSWITCH-users at 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:4e63c49632761772313601!



From darcy at primrose.ws  Mon Sep  5 01:33:25 2011
From: darcy at primrose.ws (Darcy)
Date: Sun, 4 Sep 2011 17:33:25 -0400
Subject: [Freeswitch-users] sending ani using originate with fs_cli
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0E0@cooper>
References: <4E63C46C.40308@thevoiphighway.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0E0@cooper>
Message-ID: <3B356CC8D84842259FEB17424A5497B3@DWP>

Thank you, it ended up being that I needed to use a gateway versus 
/sofia/external/xxxx at url, this format will dial a call, but not put in clid 
etc.
So I tried your format with the gateway and it works perfectly.

Darcy

-----Original Message----- 
From: Peter Olsson
Sent: Sunday, September 04, 2011 3:11 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sending ani using originate with fs_cli

Use variable origination_caller_id_number. Check out this example: 
http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_number

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org 
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Darcy Primrose 
[darcy at thevoiphighway.com]
Skickat: den 4 september 2011 20:33
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] sending ani using originate with fs_cli

I have tried various variables using the originate command in fs_cli and
I cannot get the ani and name to be sent, it populates the cid_num with
'0000000000'.  Anybody have any ideas that could help me with this.

originate
{caller-ANI='6134545851'}sofia/internal/16149753370 at 208.185.148.43
&txfax(/tmp/FAX.tif)

Darcy



FreeSWITCH-users mailing list
FreeSWITCH-users at 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:4e63c49632761772313601!



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



From ljjimenez at gmail.com  Mon Sep  5 05:40:18 2011
From: ljjimenez at gmail.com (Luis Jimenez)
Date: Sun, 4 Sep 2011 21:40:18 -0400
Subject: [Freeswitch-users] How to specify language for phrase macro
 inside session:playAndGetDigits(-)
In-Reply-To: <1315025551.80678.YahooMailClassic@web39705.mail.mud.yahoo.com>
References: <1315025551.80678.YahooMailClassic@web39705.mail.mud.yahoo.com>
Message-ID: <CAKUOwoEC1UCYhfwf02WR8Twm=WkU2zukCg9gXzp23L5dKvwxVA@mail.gmail.com>

You can set the variable default_language before or during your script.

http://wiki.freeswitch.org/wiki/Variable_default_language



On Sat, Sep 3, 2011 at 12:52 AM, king2kin <xing2kin at yahoo.com> wrote:

> Hi all,
>
> {
> session:playAndGetDigits(1, 1, 1, 3000, "#", "phrase:xk_confirm_userid:" ..
> uid, invalid, "[0,1,9]")
> }
> always uses default language (e.g. 'en') to pick up phrase marco
> definition,
> but now I would like to specify another language instead of FS default
> language to play back this phrase macro inside session:playAndGetDigits(-).
>
> Could anyone please tell me how I can specify non-default language for
> playing back my phrase macro inside session:playAndGetDigits(-)?
>
> Thanks
>
> x.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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110904/90c8f790/attachment.html 

From xing2kin at yahoo.com  Mon Sep  5 08:11:55 2011
From: xing2kin at yahoo.com (king2kin)
Date: Sun, 4 Sep 2011 21:11:55 -0700 (PDT)
Subject: [Freeswitch-users] How to specify language for phrase macro
	inside session:playAndGetDigits(-)
In-Reply-To: <CAKUOwoEC1UCYhfwf02WR8Twm=WkU2zukCg9gXzp23L5dKvwxVA@mail.gmail.com>
Message-ID: <1315195915.7643.YahooMailClassic@web39703.mail.mud.yahoo.com>

Thanks! but this method for?playAndGetDigits(-) seems not as convenient as app 'say' where we may directly specify current language (e.g. 'fr') other than default one (e.g. 'en') at any time.

--- On Sun, 9/4/11, Luis Jimenez <ljjimenez at gmail.com> wrote:


From: Luis Jimenez <ljjimenez at gmail.com>
Subject: Re: [Freeswitch-users] How to specify language for phrase macro inside session:playAndGetDigits(-)
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Sunday, September 4, 2011, 6:40 PM


You can set the variable default_language before or during your script.

http://wiki.freeswitch.org/wiki/Variable_default_language




On Sat, Sep 3, 2011 at 12:52 AM, king2kin <xing2kin at yahoo.com> wrote:

Hi all,

{
session:playAndGetDigits(1, 1, 1, 3000, "#", "phrase:xk_confirm_userid:" .. uid, invalid, "[0,1,9]")
}
always uses default language (e.g. 'en') to pick up phrase marco definition,
but now I would like to specify another language instead of FS default language to play back this phrase macro inside session:playAndGetDigits(-).

Could anyone please tell me how I can specify non-default language for playing back my phrase macro inside session:playAndGetDigits(-)?

Thanks

x.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


-----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
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110904/04c21ad9/attachment.html 

From peter.olsson at visionutveckling.se  Mon Sep  5 09:47:23 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 5 Sep 2011 07:47:23 +0200
Subject: [Freeswitch-users] sending ani using originate with fs_cli
Message-ID: <D9A899E9-C0D7-473D-8B5F-66FF85A8FA60@visionutveckling.se>

If you're using a gateway or not have no effect on how clid is sent from FS, it will work in both cases.

/Peter

----- Reply message -----
Fr?n: "Darcy" <darcy at primrose.ws>
Datum: s?n, sep 4, 2011 23:44
Rubrik: [Freeswitch-users] sending ani using originate with fs_cli
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

Thank you, it ended up being that I needed to use a gateway versus
/sofia/external/xxxx at url, this format will dial a call, but not put in clid
etc.
So I tried your format with the gateway and it works perfectly.

Darcy

-----Original Message-----
From: Peter Olsson
Sent: Sunday, September 04, 2011 3:11 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sending ani using originate with fs_cli

Use variable origination_caller_id_number. Check out this example:
http://wiki.freeswitch.org/wiki/Variable_origination_caller_id_number

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Darcy Primrose
[darcy at thevoiphighway.com]
Skickat: den 4 september 2011 20:33
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] sending ani using originate with fs_cli

I have tried various variables using the originate command in fs_cli and
I cannot get the ani and name to be sent, it populates the cid_num with
'0000000000'.  Anybody have any ideas that could help me with this.

originate
{caller-ANI='6134545851'}sofia/internal/16149753370 at 208.185.148.43
&txfax(/tmp/FAX.tif)

Darcy



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





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



FreeSWITCH-users mailing list
FreeSWITCH-users at 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:4e63efc032764450382824!



From wagnerspi at gmail.com  Sat Sep  3 17:04:50 2011
From: wagnerspi at gmail.com (Wagner)
Date: Sat, 3 Sep 2011 10:04:50 -0300
Subject: [Freeswitch-users]  pt_br sound
Message-ID: <CAO0inon2kQkNW4GFJejQQFEVHwrYT3h7p=41bu4x6yyZ_VZMnQ@mail.gmail.com>

Hello,

does anyone has the pt_br sounds for freeswitch?

Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110903/b40f9c8c/attachment-0001.html 

From zyryznet at gmail.com  Sun Sep  4 08:44:25 2011
From: zyryznet at gmail.com (Allan Piske)
Date: Sun, 4 Sep 2011 01:44:25 -0300
Subject: [Freeswitch-users] hangup error missing remote port
Message-ID: <CAK=jKZJ3cvdya8i-2A-ygRx12VK3d8ZSbLkxLDxPsA3uUxj2gg@mail.gmail.com>

Hi,

I'm having some trouble discovering the source and solution to one problem.

I setup a FS box for receiving faxes, and a already have another FS box as
SBC.


SPA3102   ---> opensips ---> FS_SBC --> opensips --> FS_FAX
this works ok, call gets answered, i hear the fax tone, the SPA3102 changes
to T38 and everyone is happy.


Nortel CS2000   ------> FS_SBC -------> opensips(regitrar,rtpproxy,etc) --->
FS_FAX
but his doen't.

This little devil CS2000 sends every invite with m=image header along with
m=audio. With aparently FS doesn't like.


like that:

freeswitch at internal> recv 1215 bytes from udp/[10.143.82.250]:5060 at
08:26:08.226858:
   ------------------------------------------------------------------------
   INVITE sip:4730365302 at 10.143.82.253:5060;transport=UDP;user=phone SIP/2.0
   Record-Route:
<sip:10.143.82.250;lr=on;ftag=-45026-10d9cdb-2668a317-10d9cdb>
   f: <sip:4784064435 at 10.150.65.16:5060
;user=phone>;tag=-45026-10d9cdb-2668a317-10d9cdb
   t: <sip:4730365302 at 10.143.82.250:5060;user=phone>
   i: 416cf3e01041960a13c410d9cdb1d2cba42b82409c10f14bb8-0322-4481
   CSeq: 1 INVITE
   User-agent: CS2000_NGSS/9.0
   P-Asserted-Identity: <sip:4784064435 at 10.150.65.16;user=phone>
   Allow:
ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK,UPDATE
   Via: SIP/2.0/UDP 10.143.82.250;branch=z9hG4bK1d78.5262ccd1.0
   v: SIP/2.0/UDP
PAE1CS2K:5060;maddr=10.150.65.16;branch=z9hG4bK-10d9cdb-1d2cba43-57ebb21d
   Max-Forwards: 69
   m: <sip:10.150.65.16:5060;transport=UDP>
   k: 100rel
   c: application/sdp
   l: 420

   v=0
   o=PVG 1315110636830 1315110636830 IN IP4 10.152.204.202
   s=-
   p=+1 6135555555
   c=IN IP4 10.152.204.202
   t=0 0
   m=audio 55920 RTP/AVP 18 8 101
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:20
   a=fmtp:18 annexb=no
   m=image 64112 udptl t38
   a=T38FaxVersion:0
   a=T38FaxMaxBuffer:1100
   a=T38FaxMaxDatagram:612
   a=T38MaxBitRate:14400
   a=T38FaxRateManagement:transferredTCF
   a=T38FaxUdpEC:t38UDPRedundancy

But then we have a problem ...




My beloved FS_SBC changes the SDP before sending it to the next hop .. as it
should becase they are on diferent subnets and set to proxy_media

2011-09-04 01:26:08.231102 [DEBUG] sofia_glue.c:1759 sofia/tpa/4730365302
Patched SDP
---
v=0
o=PVG 1315110636830 1315110636830 IN IP4 10.152.204.202
s=-
p=+1 6135555555
c=IN IP4 10.152.204.202
t=0 0
m=audio 55920 RTP/AVP 18 8 101
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=image 64112 udptl t38
a=T38FaxVersion:0
a=T38FaxMaxBuffer:1100
a=T38FaxMaxDatagram:612
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

+++
v=0
o=FreeSWITCH 2718891449 2718891450 IN IP4 A.B.C.D
s=FreeSWITCH
p=+1 6135555555
c=IN IP4 A.B.C.D
t=0 0
m=audio 19262 RTP/AVP 18 8 101
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=image 19262 udptl t38
a=T38FaxVersion:0
a=T38FaxMaxBuffer:1100
a=T38FaxMaxDatagram:612
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy


so we have now  2  m=   fields with the same port !!!!!

then when it gets answered on the FS_FAX box .. we got disconnection on the
FS_SBC box


2011-09-04 01:26:08.257064 [DEBUG] sofia.c:4761 Channel sofia/tpa/4730365302
entering state [completing][200]
2011-09-04 01:26:08.257064 [DEBUG] sofia.c:4772 Remote SDP:
v=0
o=FreeSWITCH 1315079202 1315079203 IN IP4 189.45.192.19
s=FreeSWITCH
c=IN IP4 189.45.192.19
t=0 0
m=audio 31436 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
m=image 0 udptl 19

2011-09-04 01:26:08.258409 [DEBUG] sofia.c:4761 Channel sofia/tpa/4730365302
entering state [ready][200]
2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2830
(sofia/tpa/4730365302) Callstate Change RINGING -> ACTIVE
2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2842 Send signal
sofia/voxip/4784064435 at 10.150.65.16:5060 [BREAK]
2011-09-04 01:26:08.258409 [NOTICE] sofia.c:5318 Channel
[sofia/tpa/4730365302] has been answered
2011-09-04 01:26:08.258409 [DEBUG] sofia_glue.c:2774 Set Codec
sofia/tpa/4730365302 PROXY/8000 20 ms 160 samples 0 bits
2011-09-04 01:26:08.258409 [DEBUG] sofia_glue.c:3079 PROXY AUDIO RTP
[sofia/tpa/4730365302] A.B.C.D:19262->W.X.Y.Z:0 codec: 0 ms: 20
2011-09-04 01:26:08.258409 [ERR] sofia_glue.c:3512 AUDIO RTP REPORTS ERROR:
[Missing remote port]
2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2563
(sofia/tpa/4730365302) Callstate Change ACTIVE -> HANGUP
2011-09-04 01:26:08.258409 [NOTICE] sofia_glue.c:3513 Hangup
sofia/tpa/4730365302 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER]
2011-09-04 01:26:08.258409 [DEBUG] switch_channel.c:2579 Send signal
sofia/tpa/4730365302 [KILL]
2011-09-04 01:26:08.258409 [DEBUG] switch_core_session.c:1116 Send signal
sofia/tpa/4730365302 [BREAK]
2011-09-04 01:26:08.259605 [DEBUG] switch_ivr_originate.c:3299 Originate
Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2011-09-04 01:26:08.259605 [INFO] mod_dptools.c:2647 Originate Failed.
 Cause: DESTINATION_OUT_OF_ORDER


port = 0 !!!

is that maybe a fault in opensips in the middle because the both m= fields
with the same port ? , or one of the FS boxes ?
something is broken here, sadly :(

Appreciate any help.

Allan
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110904/f5dac53f/attachment-0001.html 

From igor.kuvaldin at studia52.ru  Sun Sep  4 20:23:29 2011
From: igor.kuvaldin at studia52.ru (=?utf-8?B?0JjQs9C+0YDRjCDQmtGD0LLQsNC70LTQuNC9?=)
Date: Sun, 04 Sep 2011 20:23:29 +0400 (MSD)
Subject: [Freeswitch-users] mod_callcenter questions
In-Reply-To: <108ffc3b-47f5-408d-838a-3fe1544c2c08@main.studia52>
Message-ID: <eabd443b-731c-4e6e-b9ab-0fa7a7f54348@main.studia52>

Hi all! 
Sorry for my bad english. 
My question about of the application mod_callcenter. Is there an analog variable hangup_after_conference but for call-center? I would like to continue in the dialplan to check the value of CC-Cause and CC-Cause-Reason and depending on it to redirect the call to the IVR, perform another extension, or simply execution another extension.

Because now exit the application terminating the current call in mod_callcenter.


From n43w79 at gmail.com  Mon Sep  5 00:40:12 2011
From: n43w79 at gmail.com (n43w79)
Date: Sun, 04 Sep 2011 16:40:12 -0400
Subject: [Freeswitch-users] pfSense 2.0 with FreeSWITCH and FusionPBX?
Message-ID: <j40nnc$e8l$1@dough.gmane.org>

Hi,
Just wondering is there a FS package for pfSense 2.0?
I am trying to test with PC Engines MB alix.2d3.
Perhaps, like the Android port, may be FS can be installed in pfSense as 
a native library as well?
Btw, like GillesToo, I don't have the technical skills to port FS as well.
Thank you.



From wagnerspi at gmail.com  Mon Sep  5 04:52:04 2011
From: wagnerspi at gmail.com (Wagner)
Date: Sun, 4 Sep 2011 21:52:04 -0300
Subject: [Freeswitch-users]  Calls over 3G
Message-ID: <CAO0ino=uZbveU24WufHEvnH=AKa0TP5Xy1sBbcJEdv7NbW8iWg@mail.gmail.com>

Hello,

What's the best way to make calls over 3G networks with less delay and more
quality?

any kind of compression, or different codec, any ideas?

Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110904/b12c5b2b/attachment-0001.html 

From Glen.Ganderton at premier.com.au  Mon Sep  5 08:07:54 2011
From: Glen.Ganderton at premier.com.au (Glen Ganderton)
Date: Mon, 5 Sep 2011 14:07:54 +1000
Subject: [Freeswitch-users] FreeSWITCH, Nuance, ASR: IVR Integration
Message-ID: <CD5A1499C8DEF34FAA3509722B00393E740262C99D@m-mail03.office.premiertech>

Hi Guys,

What i am trying to do is configure an IVR using freeswitch and Nuance 5.1.5 to perform speech recognition. I have installed the latest
release of freeswitch on my CentOS system, I have also enabled and configured the unimrcp module. I could find lots of information to
configure the unimrcp module but there is no documentation on how to create IVR menu's using this module with Nuance. I have used the
demo pizza IVR with the pocketsphinx module and I keep getting refered to this but it's no help as I want to use Nuance for ASR. Any
help on how to configure this would be great, specifically:

* How to I call the IVR script from the dialplan
* What language/s can I code the IVR in and can you please provide a basic sample of how this is done with Nuance.

Thank you in advance

--Glen
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/4b765f7c/attachment-0001.html 

From Glen.Ganderton at premier.com.au  Mon Sep  5 11:02:49 2011
From: Glen.Ganderton at premier.com.au (Glen Ganderton)
Date: Mon, 5 Sep 2011 17:02:49 +1000
Subject: [Freeswitch-users] FreeSWITCH, Nuance, ASR: IVR Integration
Message-ID: <CD5A1499C8DEF34FAA3509722B00393E740262CA5F@m-mail03.office.premiertech>

Hi Guys,

What i am trying to do is configure an IVR using freeswitch and Nuance 5.1.5 to perform speech recognition. I have installed the latest
release of freeswitch on my CentOS system, I have also enabled and configured the unimrcp module. I could find lots of information to
configure the unimrcp module but there is no documentation on how to create IVR menu's using this module with Nuance. I have used the
demo pizza IVR with the pocketsphinx module and I keep getting refered to this but it's no help as I want to use Nuance for ASR. Any
help on how to configure this would be great, specifically:

* How to I call the IVR script from the dialplan
* What language/s can I code the IVR in and can you please provide a basic sample of how this is done with Nuance.

Thank you in advance

--Glen
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/3902c743/attachment-0001.html 

From gmaruzz at gmail.com  Mon Sep  5 11:19:12 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 5 Sep 2011 09:19:12 +0200
Subject: [Freeswitch-users] Calls over 3G
In-Reply-To: <CAO0ino=uZbveU24WufHEvnH=AKa0TP5Xy1sBbcJEdv7NbW8iWg@mail.gmail.com>
References: <CAO0ino=uZbveU24WufHEvnH=AKa0TP5Xy1sBbcJEdv7NbW8iWg@mail.gmail.com>
Message-ID: <CALXCt0q8+6bPYNVU49QHcZ576C_m-JnNeKu7pu7gGfc9CUTDqw@mail.gmail.com>

If I recall correctly, best is use a long ptime (less overhead on network).

Look in the mailing list archives for messages to/from naif (fabio
pietrosanti), he has done lot of work on this.

-giovanni

On Mon, Sep 5, 2011 at 2:52 AM, Wagner <wagnerspi at gmail.com> wrote:
> Hello,
> What's the best way to make calls over 3G networks with less delay and more
> quality?
> any kind of compression, or different codec, 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
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From devel at omninet.eu  Mon Sep  5 11:34:09 2011
From: devel at omninet.eu (Anestis Mavro)
Date: Mon, 5 Sep 2011 10:34:09 +0300
Subject: [Freeswitch-users] Calls over 3G
In-Reply-To: <CAO0ino=uZbveU24WufHEvnH=AKa0TP5Xy1sBbcJEdv7NbW8iWg@mail.gmail.com>
References: <CAO0ino=uZbveU24WufHEvnH=AKa0TP5Xy1sBbcJEdv7NbW8iWg@mail.gmail.com>
Message-ID: <512047ED5AE74A819B2D13E467806500@omni1.local>

 

 

Hi,

 

I have tested G729 and iLBC and both are working even on GPRS Edge networks.

 

Don't forget, that many mobile operators are blocking VoIP. With one
operator we had issues on some cells. In many areas his network supported
VoIP in some other areas even the registration was not working.

And of course don't forget, that the bandwidth is shared (and limited): in
one cell you might get very good quality and in another one (or at another
time) you might get very poor quality phone calls.

 

Good luck

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wagner
Sent: Monday, September 05, 2011 3:52 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Calls over 3G

 

Hello,

 

What's the best way to make calls over 3G networks with less delay and more
quality?

 

any kind of compression, or different codec, any ideas?

 

Thanks

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/7c164e54/attachment.html 

From mrene_lists at avgs.ca  Mon Sep  5 13:06:26 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 5 Sep 2011 11:06:26 +0200
Subject: [Freeswitch-users] Compiling Freeswitch for Android?
In-Reply-To: <779CE20F-64B3-4586-B712-9E55561C4794@gmail.com>
References: <1314959406122-6753409.post@n2.nabble.com>
	<BD97C6DD-E80A-4CE8-8599-C5E851150454@avgs.ca>
	<1314962913219-6753523.post@n2.nabble.com>
	<296C71FF-C5F6-41E8-A08C-5DE3AE61300F@avgs.ca>
	<CAKq5mS6fXXjJafcyyL=oj+fbhESnvOL_j=n1s7ZV56Uw4nvaAw@mail.gmail.com>
	<6918529636939195413@unknownmsgid>
	<779CE20F-64B3-4586-B712-9E55561C4794@gmail.com>
Message-ID: <A4D1E411-4089-4573-8C30-9A97FA674439@avgs.ca>

The point is to use freeswitch as a software phone, since it supports most voip protocols pretty well.

For iOS, it is based on darwin, so porting wasn't too much of a pain. 

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




On 2011-09-04, at 7:22 PM, Robert-iPhone wrote:

> I dont understand all the interest here. Nobody is going to provide low level access to radio or phone to do anything of value with Freeswitch or Asterisk (someone posted there too).
> 
> If you really want to do something crazy and are trying to use cell network - get a gsm modem or similar and yank the sim card.
> 
> Are iphone and android powerfull enough - YES but again Apple and Google arent going to just give away access to low level functions at will.
> 
> 
> 
> Sent from my iPhone
> 
> On Sep 4, 2011, at 12:07 PM, Peter Schrock <peter.schrock at gmail.com> wrote:
> 
>> What about iPhone since it's Unix based?
>> 
>> Peter
>> 
>> On Sep 4, 2011, at 8:30 AM, envelopes envelopes <sunwood360 at gmail.com> wrote:
>> 
>>> I am also very interested in your progress.
>>> 
>>> Thanks
>>> 
>>> On Sep 2, 2011 6:29 AM, "Mathieu Rene" <mrene_lists at avgs.ca> wrote:
>>> > It'll be released with the main distribution :)
>>> > 
>>> > Mathieu Rene
>>> > Avant-Garde Solutions Inc
>>> > Office: + 1 (514) 664-1044 x100
>>> > Cell: +1 (514) 664-1044 x200
>>> > mrene at avgs.ca
>>> > 
>>> > 
>>> > 
>>> > 
>>> > On 2011-09-02, at 1:28 PM, GillesToo wrote:
>>> > 
>>> >> Great news :-) Unfortunately, I don't have the technical skills to port
>>> >> Freeswitch. Did you put up a web site so that we can keep an eye on the
>>> >> project?
>>> >> 
>>> >> --
>>> >> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Compiling-Freeswitch-for-Android-tp6753409p6753523.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
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/9bde2b82/attachment.html 

From asilva at wirelessmundi.com  Mon Sep  5 13:31:39 2011
From: asilva at wirelessmundi.com (Antonio)
Date: Mon, 05 Sep 2011 11:31:39 +0200
Subject: [Freeswitch-users] pt_br sound
In-Reply-To: <CAO0inon2kQkNW4GFJejQQFEVHwrYT3h7p=41bu4x6yyZ_VZMnQ@mail.gmail.com>
References: <CAO0inon2kQkNW4GFJejQQFEVHwrYT3h7p=41bu4x6yyZ_VZMnQ@mail.gmail.com>
Message-ID: <1315215099.3761.37.camel@marces.madrid.commsmundi.com>

Hi,

Yes, you can download it from
http://www.wirelessmundi.com/en/specialoffersprompts.html




On Sat, 2011-09-03 at 10:04 -0300, Wagner wrote:

> Hello, 
> 
> does anyone has the pt_br sounds for freeswitch?
> 
> 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


-- 

Un cordial saludo / Best regards, 

 _________________________

Ant?nio Silva

E-mail:asilva at wirelessmundi.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/6ddffc9d/attachment-0001.html 

From dujinfang at gmail.com  Mon Sep  5 13:45:31 2011
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 5 Sep 2011 17:45:31 +0800
Subject: [Freeswitch-users] mod_say_zh the Chinese way discussion
Message-ID: <77711D0F24454766B5AD79DB41B382F8@gmail.com>

Hi all,

First sorry I send both to users and devs as I want to talk in both groups.

I wonder how many of you actually use mod_say_zh

I had submitted a patch but not been accepted because it was argued that it might breaks zh in other locations than China mainland.

http://jira.freeswitch.org/browse/FS-2809 

How ever, I doubt is it useful actually. Besides the above example, think of the following


when say time, EN is Sept 1, 2011 and mod_say_zh says 

9 yue di 15, 11 

Actually in China we say 

2011 nian(year) 9 yue (month) 15 ri(day)

Again, I'd like to know if anyone is actually using the current way.


So I need a PRC way, questions are:

1) Is it safe to change the default behavior?

2) If not how could I change? add a channel variable? or

3) create another dialect mod_say_zh_cn ?

Thanks.

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

Sent with Sparrow (http://www.sparrowmailapp.com)

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/ab010d4e/attachment.html 

From dujinfang at gmail.com  Mon Sep  5 14:10:04 2011
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 5 Sep 2011 18:10:04 +0800
Subject: [Freeswitch-users] mod_say_zh the Chinese way discussion
In-Reply-To: <77711D0F24454766B5AD79DB41B382F8@gmail.com>
References: <77711D0F24454766B5AD79DB41B382F8@gmail.com>
Message-ID: <51B6F146553F48D9B42E8AEEC41C1F6F@gmail.com>

 Ah, sorry, please ignore this for now, I loaded the wrong(english) module for test.



On Monday, September 5, 2011 at 5:45 PM, Seven Du wrote:

> Hi all,
> 
> First sorry I send both to users and devs as I want to talk in both groups.
> 
> I wonder how many of you actually use mod_say_zh
> 
> I had submitted a patch but not been accepted because it was argued that it might breaks zh in other locations than China mainland.
> 
> http://jira.freeswitch.org/browse/FS-2809 
> 
> How ever, I doubt is it useful actually. Besides the above example, think of the following
> 
> 
> when say time, EN is Sept 1, 2011 and mod_say_zh says 
> 
> 9 yue di 15, 11 
> 
> Actually in China we say 
> 
> 2011 nian(year) 9 yue (month) 15 ri(day)
> 
> Again, I'd like to know if anyone is actually using the current way.
> 
> 
> So I need a PRC way, questions are:
> 
> 1) Is it safe to change the default behavior?
> 
> 2) If not how could I change? add a channel variable? or
> 
> 3) create another dialect mod_say_zh_cn ?
> 
> Thanks.
> 
> -- 
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj: http://www.freeswitch.org.cn
> 
> Sent with Sparrow (http://www.sparrowmailapp.com)
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/57be231f/attachment.html 

From jalsot at gmail.com  Mon Sep  5 15:03:51 2011
From: jalsot at gmail.com (Tamas Jalsovszky)
Date: Mon, 5 Sep 2011 13:03:51 +0200
Subject: [Freeswitch-users] Sofia DNS resolver fail-over
Message-ID: <CAHCLHGt9FTVyOHqh9WTJra01BBbtTYOGrwYuYCdUt9NUMp3WFA@mail.gmail.com>

Hello,

Is there any way to let quickly fail-over the DNS resolver in sofia when the
nameserver is unreacheable?
What I mean is, when the 1st nameserver given in resolver.conf does not
respond, lets try the next NS server.

Br,
   Tamas
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/1de83c02/attachment.html 

From nazim.aghabayov at gmail.com  Mon Sep  5 15:35:59 2011
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Mon, 05 Sep 2011 16:35:59 +0500
Subject: [Freeswitch-users] Compiling Freeswitch for Android?
In-Reply-To: <A4D1E411-4089-4573-8C30-9A97FA674439@avgs.ca>
References: <1314959406122-6753409.post@n2.nabble.com>
	<BD97C6DD-E80A-4CE8-8599-C5E851150454@avgs.ca>
	<1314962913219-6753523.post@n2.nabble.com>
	<296C71FF-C5F6-41E8-A08C-5DE3AE61300F@avgs.ca>
	<CAKq5mS6fXXjJafcyyL=oj+fbhESnvOL_j=n1s7ZV56Uw4nvaAw@mail.gmail.com>
	<6918529636939195413@unknownmsgid>
	<779CE20F-64B3-4586-B712-9E55561C4794@gmail.com>
	<A4D1E411-4089-4573-8C30-9A97FA674439@avgs.ca>
Message-ID: <4E64B41F.9090203@gmail.com>

Hello Mathieu,

I'm interested in porting FS to android. Going to get an android sdk 
today and start looking into it.

Best Regards,
Nazim


On 09/05/2011 02:06 PM, Mathieu Rene wrote:
> The point is to use freeswitch as a software phone, since it supports 
> most voip protocols pretty well.
>
> For iOS, it is based on darwin, so porting wasn't too much of a pain.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca <mailto:mrene at avgs.ca>
>
>
>
>



From ijurado at econcept.es  Mon Sep  5 16:25:32 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Mon, 5 Sep 2011 14:25:32 +0200
Subject: [Freeswitch-users] _sofia_replaces_ from hangup_hook
Message-ID: <CABj_V-=EtZJda4f=cyLueCJsHma0r6WmpS9wqgzFKLAnBVuO0g@mail.gmail.com>

Hi,

Given the following scenario:

    X calls Y  (channels A0 inbound and B0 outbound)
    Y calls Z  (channels A1 inbound and B1 outbound)

Then, if Y performs an attended transfer, B0 terminates with an
ATTENDED_TRANSFER hangup_cause (or endpoint_disposition).  However, A1
does not show any special sign of beign part of the transfer operation
except a CDR variable named "_sofia_replaces_".

The problem with that variable is that it doesn't appear as part of the
"env" object in Lua hangup hooks.  Nevertheless, it appears in the XML
CDR.

What's happening here?  Am I doing something wrong?

Cheers.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From asilva at wirelessmundi.com  Mon Sep  5 16:46:26 2011
From: asilva at wirelessmundi.com (Antonio)
Date: Mon, 05 Sep 2011 14:46:26 +0200
Subject: [Freeswitch-users] Problem receiving fax
In-Reply-To: <1314087013.29574.59.camel@marces.madrid.commsmundi.com>
References: <1313493347.30552.80.camel@marces.madrid.commsmundi.com>
	<CAAMQ8bROOOci2ocWZkcn_qxh-vc8kY1Tg+3deUJMs3Lvf0PMpw@mail.gmail.com>
	<1313496873.30552.82.camel@marces.madrid.commsmundi.com>
	<1314087013.29574.59.camel@marces.madrid.commsmundi.com>
Message-ID: <1315226786.3761.48.camel@marces.madrid.commsmundi.com>

Hi, 

I'm still fighting this problem.

This now happen to me in another machine with the different hardware. 

I has receiving faxes with no problem and at some point or reason (i
Can't catch IT!!) i is just stops. 
The only solution that i have for now is reboot the server. 


Is there anyone with the same problem as me?
Can you point some way to have more debug/logs from spandsp or
freeswitch so i can try to find out the problem?

I'm not sure that is a bug but this behavior is pretty weird!! 

Btw: i did some audio captures and nothing wrong....

Thanks,

Ant?nio




On Tue, 2011-08-23 at 10:10 +0200, Antonio wrote: 

> I tried with the last version, and the same occurred.
> 
> After restarting the server i can receive faxes and them it stops
> receiving it. 
> 
> I can't find out what is the cause... Can anyone help me how to find out
> where could be the problem.
> 
> 
> I'm think replacing the hardware, just to be sure that is not an
> hardware problem.
> 
> 
> 
> Thanks,
> 
> Ant?nio
> 
> 
> 
> 
> 
> 
> On Tue, 2011-08-16 at 14:14 +0200, Antonio wrote: 
> > I'm using libpri-1.4.11 and freeswitch head. I'm going to try with the
> > latest libpri-1.4.12.
> > 
> > And post the results.
> > 
> > Thanks,
> > Ant?nio
> > 
> > 
> > On Tue, 2011-08-16 at 13:46 +0200, Christian Benke wrote: 
> > > On 16 August 2011 13:15, Antonio <asilva at wirelessmundi.com> wrote:
> > > > I'm having problems receiving fax in a pri E1 line.
> > > > The log can be found at http://pastebin.freeswitch.org/17047
> > > 
> > > Hi!
> > > 
> > > I had the same issue a few days ago("FLOW T.30 Bad HDLC CRC received").
> > > Recompiling&Reinstalling libpri&FreeSWITCH helped.
> > > 
> > > hthu2
> > > Christian
> > > 
> > > 
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > 
> 


-- 

Un cordial saludo / Best regards, 

 _________________________

Ant?nio Silva

E-mail:asilva at wirelessmundi.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/50608a60/attachment.html 

From mays.david at gmail.com  Mon Sep  5 17:56:16 2011
From: mays.david at gmail.com (dma)
Date: Mon, 5 Sep 2011 06:56:16 -0700 (PDT)
Subject: [Freeswitch-users] Will bypass-media work when a call is bridged to
 multiple SIP destination
Message-ID: <1315230976599-6760883.post@n2.nabble.com>

Hi Support, 

This is a simple question but I don't find an answer from the web. 

I hope to know whether bypass-media work when a call is bridged to multiple
SIP destination?

Thanks,
D.Ma

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Will-bypass-media-work-when-a-call-is-bridged-to-multiple-SIP-destination-tp6760883p6760883.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From avi at avimarcus.net  Mon Sep  5 18:05:41 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 5 Sep 2011 17:05:41 +0300
Subject: [Freeswitch-users] Will bypass-media work when a call is
 bridged to multiple SIP destination
In-Reply-To: <1315230976599-6760883.post@n2.nabble.com>
References: <1315230976599-6760883.post@n2.nabble.com>
Message-ID: <CALNZuw=gQ6nzEufq2NmRZ81P+EdsGfPjH2u2fpHDgno8D3LQxQ@mail.gmail.com>

You mean you fork the dialing so it fails over or tries different places at
once?
Sure. You can even set the bypass_media variable differently on each leg.

-Avi


On Mon, Sep 5, 2011 at 4:56 PM, dma <mays.david at gmail.com> wrote:

> Hi Support,
>
> This is a simple question but I don't find an answer from the web.
>
> I hope to know whether bypass-media work when a call is bridged to multiple
> SIP destination?
>
> Thanks,
> D.Ma
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Will-bypass-media-work-when-a-call-is-bridged-to-multiple-SIP-destination-tp6760883p6760883.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/a2fafc09/attachment-0001.html 

From nbhatti at gmail.com  Mon Sep  5 18:45:47 2011
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Mon, 5 Sep 2011 17:45:47 +0300
Subject: [Freeswitch-users] vBilling Beta Program!!
Message-ID: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>

Hello everyone,
As promised we have opened beta testing program for *vBilling*. An open
source billing platform for FreeSWITCH. You are invited to login, take a
look and play with it. Email us with your comments, let us know what
improvements can be made. Following are the details for the program:

========================================
vBilling User Panel: *http://demo.vbilling.org/*
User Login: *demouser*
User Password: *P at ssw0rd*

vBilling Admin Panel: *http://demo.vbilling.org/admin/*
Admin Login: *admin*
Admin Password: *P at ssw0rd*

We have configured a LIVE gateway for you. You can send your *SIP* calls to
*demo.digitallinx.com* port *5060* and call any toll free (800 and 888 only)
number in US. Make calls, login to *vBilling* and see how the billing works.
========================================

Most of the features are working. Some of the them mentioned on the site are
still is development so be patient :) For product features and more details,
visit our website at *http://www.vbilling.org/*
<http://www.vbilling.org/>and us know what do you think about it.


Regards,
Muhammad Naseer
CEO vBilling/Digital Linx
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/bf588136/attachment.html 

From ce at kapper.net  Mon Sep  5 18:55:23 2011
From: ce at kapper.net (Clemens Ebentheuer)
Date: Mon, 5 Sep 2011 16:55:23 +0200
Subject: [Freeswitch-users] vBilling Beta Program!!
In-Reply-To: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>
References: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>
Message-ID: <1B19ABD72889C245AE8EEE08AC24103A63A769F311@exmachina.office.kapper.net>

Hi,

got Invalid Email/Password  here - both user and admin. Too fast?

Thx,

Clemens

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Naseer Bhatti
Sent: Monday, September 05, 2011 4:46 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] vBilling Beta Program!!


Hello everyone,
As promised we have opened beta testing program for vBilling. An open source billing platform for FreeSWITCH. You are invited to login, take a look and play with it. Email us with your comments, let us know what improvements can be made. Following are the details for the program:

========================================
vBilling User Panel: http://demo.vbilling.org/
User Login: demouser
User Password: P at ssw0rd

vBilling Admin Panel: http://demo.vbilling.org/admin/
Admin Login: admin
Admin Password: P at ssw0rd

We have configured a LIVE gateway for you. You can send your SIP calls to demo.digitallinx.com<http://demo.digitallinx.com> port 5060 and call any toll free (800 and 888 only) number in US. Make calls, login to vBilling and see how the billing works.
========================================

Most of the features are working. Some of the them mentioned on the site are still is development so be patient :) For product features and more details, visit our website at http://www.vbilling.org/ and us know what do you think about it.


Regards,
Muhammad Naseer
CEO vBilling/Digital Linx
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/d0244815/attachment.html 

From avi at avimarcus.net  Mon Sep  5 19:05:54 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 5 Sep 2011 18:05:54 +0300
Subject: [Freeswitch-users] vBilling Beta Program!!
In-Reply-To: <1B19ABD72889C245AE8EEE08AC24103A63A769F311@exmachina.office.kapper.net>
References: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A63A769F311@exmachina.office.kapper.net>
Message-ID: <CALNZuw=Pd7vpDZpw_vaodrGzKrCg5vRiEeAbO6OJCz75kKgf3g@mail.gmail.com>

Only user worked for me, maybe someone was testing out the change password
feature?
-Avi

On Mon, Sep 5, 2011 at 5:55 PM, Clemens Ebentheuer <ce at kapper.net> wrote:

> Hi,****
>
> ** **
>
> got Invalid Email/Password  here ? both user and admin. Too fast?****
>
> ** **
>
> Thx,****
>
> ** **
>
> Clemens****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Muhammad
> Naseer Bhatti
> *Sent:* Monday, September 05, 2011 4:46 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] vBilling Beta Program!!****
>
> ** **
>
>
> Hello everyone,
> As promised we have opened beta testing program for *vBilling*. An open
> source billing platform for FreeSWITCH. You are invited to login, take a
> look and play with it. Email us with your comments, let us know what
> improvements can be made. Following are the details for the program:
>
> ========================================
> vBilling User Panel: *http://demo.vbilling.org/*
> User Login: *demouser*
> User Password: *P at ssw0rd*
>
> vBilling Admin Panel: *http://demo.vbilling.org/admin/*
> Admin Login: *admin*
> Admin Password: *P at ssw0rd*
>
> We have configured a LIVE gateway for you. You can send your *SIP* calls
> to *demo.digitallinx.com* port *5060* and call any toll free (800 and 888
> only) number in US. Make calls, login to *vBilling* and see how the
> billing works.
> ========================================
>
> Most of the features are working. Some of the them mentioned on the site
> are still is development so be patient :) For product features and more
> details, visit our website at *http://www.vbilling.org/*<http://www.vbilling.org/>and us know what do you think about it.
>
>
> Regards,
> Muhammad Naseer
> CEO vBilling/Digital Linx****
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/1511116c/attachment.html 

From a.afzali2003 at gmail.com  Mon Sep  5 19:07:21 2011
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Mon, 5 Sep 2011 19:37:21 +0430
Subject: [Freeswitch-users] Finding User's Current Session
Message-ID: <CABK0fC7GLpqneudT753L3m2W72jHTVW82t3BZNmCe843O7cZCA@mail.gmail.com>

Hi Guys,

Is there a API to be use to get current session / uuid of a user?

BEST,
-- afshin
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/9c471bc2/attachment-0001.html 

From nbhatti at gmail.com  Mon Sep  5 19:11:42 2011
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Mon, 5 Sep 2011 18:11:42 +0300
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 63, Issue 23
In-Reply-To: <mailman.2540.1315235272.29749.freeswitch-users@lists.freeswitch.org>
References: <mailman.2540.1315235272.29749.freeswitch-users@lists.freeswitch.org>
Message-ID: <CACak_ywjY-NUUpma9Ho2hfXYfcHDN7eLt15RwJ-_h9FWKtuPhQ@mail.gmail.com>

I'll disable the password change for now. Try again please :)

On Mon, Sep 5, 2011 at 6:07 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. vBilling Beta Program!! (Muhammad Naseer Bhatti)
>   2. Re: vBilling Beta Program!! (Clemens Ebentheuer)
>   3. Re: vBilling Beta Program!! (Avi Marcus)
>   4. Finding User's Current Session (afshin afzali)
>
>
> ---------- Forwarded message ----------
> From: Muhammad Naseer Bhatti <nbhatti at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 5 Sep 2011 17:45:47 +0300
> Subject: [Freeswitch-users] vBilling Beta Program!!
>
> Hello everyone,
> As promised we have opened beta testing program for *vBilling*. An open
> source billing platform for FreeSWITCH. You are invited to login, take a
> look and play with it. Email us with your comments, let us know what
> improvements can be made. Following are the details for the program:
>
> ========================================
> vBilling User Panel: *http://demo.vbilling.org/*
> User Login: *demouser*
> User Password: *P at ssw0rd*
>
> vBilling Admin Panel: *http://demo.vbilling.org/admin/*
> Admin Login: *admin*
> Admin Password: *P at ssw0rd*
>
> We have configured a LIVE gateway for you. You can send your *SIP* calls
> to *demo.digitallinx.com* port *5060* and call any toll free (800 and 888
> only) number in US. Make calls, login to *vBilling* and see how the
> billing works.
> ========================================
>
> Most of the features are working. Some of the them mentioned on the site
> are still is development so be patient :) For product features and more
> details, visit our website at *http://www.vbilling.org/*<http://www.vbilling.org/>and us know what do you think about it.
>
>
> Regards,
> Muhammad Naseer
> CEO vBilling/Digital Linx
>
>
> ---------- Forwarded message ----------
> From: Clemens Ebentheuer <ce at kapper.net>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 5 Sep 2011 16:55:23 +0200
> Subject: Re: [Freeswitch-users] vBilling Beta Program!!
>
> Hi,****
>
> ** **
>
> got Invalid Email/Password  here ? both user and admin. Too fast?****
>
> ** **
>
> Thx,****
>
> ** **
>
> Clemens****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Muhammad
> Naseer Bhatti
> *Sent:* Monday, September 05, 2011 4:46 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] vBilling Beta Program!!****
>
> ** **
>
>
> Hello everyone,
> As promised we have opened beta testing program for *vBilling*. An open
> source billing platform for FreeSWITCH. You are invited to login, take a
> look and play with it. Email us with your comments, let us know what
> improvements can be made. Following are the details for the program:
>
> ========================================
> vBilling User Panel: *http://demo.vbilling.org/*
> User Login: *demouser*
> User Password: *P at ssw0rd*
>
> vBilling Admin Panel: *http://demo.vbilling.org/admin/*
> Admin Login: *admin*
> Admin Password: *P at ssw0rd*
>
> We have configured a LIVE gateway for you. You can send your *SIP* calls
> to *demo.digitallinx.com* port *5060* and call any toll free (800 and 888
> only) number in US. Make calls, login to *vBilling* and see how the
> billing works.
> ========================================
>
> Most of the features are working. Some of the them mentioned on the site
> are still is development so be patient :) For product features and more
> details, visit our website at *http://www.vbilling.org/*<http://www.vbilling.org/>and us know what do you think about it.
>
>
> Regards,
> Muhammad Naseer
> CEO vBilling/Digital Linx****
>
>
> ---------- Forwarded message ----------
> From: Avi Marcus <avi at avimarcus.net>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 5 Sep 2011 18:05:54 +0300
> Subject: Re: [Freeswitch-users] vBilling Beta Program!!
> Only user worked for me, maybe someone was testing out the change password
> feature?
> -Avi
>
> On Mon, Sep 5, 2011 at 5:55 PM, Clemens Ebentheuer <ce at kapper.net> wrote:
>
>> Hi,****
>>
>> ** **
>>
>> got Invalid Email/Password  here ? both user and admin. Too fast?****
>>
>> ** **
>>
>> Thx,****
>>
>> ** **
>>
>> Clemens****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Muhammad
>> Naseer Bhatti
>> *Sent:* Monday, September 05, 2011 4:46 PM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* [Freeswitch-users] vBilling Beta Program!!****
>>
>> ** **
>>
>>
>> Hello everyone,
>> As promised we have opened beta testing program for *vBilling*. An open
>> source billing platform for FreeSWITCH. You are invited to login, take a
>> look and play with it. Email us with your comments, let us know what
>> improvements can be made. Following are the details for the program:
>>
>> ========================================
>> vBilling User Panel: *http://demo.vbilling.org/*
>> User Login: *demouser*
>> User Password: *P at ssw0rd*
>>
>> vBilling Admin Panel: *http://demo.vbilling.org/admin/*
>> Admin Login: *admin*
>> Admin Password: *P at ssw0rd*
>>
>> We have configured a LIVE gateway for you. You can send your *SIP* calls
>> to *demo.digitallinx.com* port *5060* and call any toll free (800 and 888
>> only) number in US. Make calls, login to *vBilling* and see how the
>> billing works.
>> ========================================
>>
>> Most of the features are working. Some of the them mentioned on the site
>> are still is development so be patient :) For product features and more
>> details, visit our website at *http://www.vbilling.org/*<http://www.vbilling.org/>and us know what do you think about it.
>>
>>
>> Regards,
>> Muhammad Naseer
>> CEO vBilling/Digital Linx****
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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: afshin afzali <a.afzali2003 at gmail.com>
> To: freeswitch-users <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 5 Sep 2011 19:37:21 +0430
> Subject: [Freeswitch-users] Finding User's Current Session
> Hi Guys,
>
> Is there a API to be use to get current session / uuid of a user?
>
> 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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/e8f96e57/attachment.html 

From nbhatti at gmail.com  Mon Sep  5 19:15:27 2011
From: nbhatti at gmail.com (nbhatti)
Date: Mon, 5 Sep 2011 08:15:27 -0700 (PDT)
Subject: [Freeswitch-users] vBilling Beta Program!!
In-Reply-To: <CALNZuw=Pd7vpDZpw_vaodrGzKrCg5vRiEeAbO6OJCz75kKgf3g@mail.gmail.com>
References: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A63A769F311@exmachina.office.kapper.net>
	<CALNZuw=Pd7vpDZpw_vaodrGzKrCg5vRiEeAbO6OJCz75kKgf3g@mail.gmail.com>
Message-ID: <1315235727102-6761088.post@n2.nabble.com>

I have disabled the admin password change for now :)

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p6761088.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From cesar.bermudez at gmail.com  Mon Sep  5 19:23:24 2011
From: cesar.bermudez at gmail.com (Cesar Bermudez)
Date: Mon, 5 Sep 2011 09:23:24 -0600
Subject: [Freeswitch-users] vBilling Beta Program!!
In-Reply-To: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>
References: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>
Message-ID: <CADrcvV85dWd=BvssSkHkp2d=geU9DDzVdhF0JJe5Fmo_JppoNQ@mail.gmail.com>

Nice !!!!

On Mon, Sep 5, 2011 at 8:45 AM, Muhammad Naseer Bhatti <nbhatti at gmail.com>wrote:

>
> Hello everyone,
> As promised we have opened beta testing program for *vBilling*. An open
> source billing platform for FreeSWITCH. You are invited to login, take a
> look and play with it. Email us with your comments, let us know what
> improvements can be made. Following are the details for the program:
>
> ========================================
> vBilling User Panel: *http://demo.vbilling.org/*
> User Login: *demouser*
> User Password: *P at ssw0rd*
>
> vBilling Admin Panel: *http://demo.vbilling.org/admin/*
> Admin Login: *admin*
> Admin Password: *P at ssw0rd*
>
> We have configured a LIVE gateway for you. You can send your *SIP* calls
> to *demo.digitallinx.com* port *5060* and call any toll free (800 and 888
> only) number in US. Make calls, login to *vBilling* and see how the
> billing works.
> ========================================
>
> Most of the features are working. Some of the them mentioned on the site
> are still is development so be patient :) For product features and more
> details, visit our website at *http://www.vbilling.org/*<http://www.vbilling.org/>and us know what do you think about it.
>
>
> Regards,
> Muhammad Naseer
> CEO vBilling/Digital Linx
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/eac425af/attachment-0001.html 

From infos at madovsky.org  Mon Sep  5 19:24:38 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 5 Sep 2011 11:24:38 -0400
Subject: [Freeswitch-users] Sofia DNS resolver fail-over
References: <CAHCLHGt9FTVyOHqh9WTJra01BBbtTYOGrwYuYCdUt9NUMp3WFA@mail.gmail.com>
Message-ID: <375D55C8E83A48F689719997DBB16090@e1705>

try DNS NAPTR SRV
  ----- Original Message ----- 
  From: Tamas Jalsovszky 
  To: FreeSWITCH Users Help 
  Sent: Monday, September 05, 2011 7:03 AM
  Subject: [Freeswitch-users] Sofia DNS resolver fail-over


  Hello,

  Is there any way to let quickly fail-over the DNS resolver in sofia when the nameserver is unreacheable?
  What I mean is, when the 1st nameserver given in resolver.conf does not respond, lets try the next NS server.

  Br,
     Tamas



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



  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/8a427697/attachment.html 

From curriegrad2004 at gmail.com  Mon Sep  5 20:19:30 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 5 Sep 2011 09:19:30 -0700
Subject: [Freeswitch-users] Calls over 3G
In-Reply-To: <512047ED5AE74A819B2D13E467806500@omni1.local>
References: <CAO0ino=uZbveU24WufHEvnH=AKa0TP5Xy1sBbcJEdv7NbW8iWg@mail.gmail.com>
	<512047ED5AE74A819B2D13E467806500@omni1.local>
Message-ID: <CAMWiMy-vvfYGRREx9_MtJjntwdJ0b8roZKtvfdeijBpAEQ1R=Q@mail.gmail.com>

There's always the option of using OpenVPN over a 3G network, that way
it should bypass most of the restrictions placed by your mobile
carrier. Not to mention that NAT-T over IPSec works well too

On Mon, Sep 5, 2011 at 12:34 AM, Anestis Mavro <devel at omninet.eu> wrote:
>
>
>
>
> Hi,
>
>
>
> I have tested G729 and iLBC and both are working even on GPRS Edge networks.
>
>
>
> Don?t forget, that many mobile operators are blocking VoIP. With one
> operator we had issues on some cells. In many areas his network supported
> VoIP in some other areas even the registration was not working.
>
> And of course don?t forget, that the bandwidth is shared (and limited): in
> one cell you might get very good quality and in another one (or at another
> time) you might get very poor quality phone calls.
>
>
>
> Good luck
>
>
>
> ________________________________
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wagner
> Sent: Monday, September 05, 2011 3:52 AM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Calls over 3G
>
>
>
> Hello,
>
>
>
> What's the best way to make calls over 3G networks with less delay and more
> quality?
>
>
>
> any kind of compression, or different codec, any ideas?
>
>
>
> Thanks
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature
> database 5054 (20100423) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.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 nbhatti at gmail.com  Mon Sep  5 20:30:42 2011
From: nbhatti at gmail.com (nbhatti)
Date: Mon, 5 Sep 2011 09:30:42 -0700 (PDT)
Subject: [Freeswitch-users] Finding User's Current Session
In-Reply-To: <CABK0fC7GLpqneudT753L3m2W72jHTVW82t3BZNmCe843O7cZCA@mail.gmail.com>
References: <CABK0fC7GLpqneudT753L3m2W72jHTVW82t3BZNmCe843O7cZCA@mail.gmail.com>
Message-ID: <1315240242664-6761305.post@n2.nabble.com>


Try "show calls" at FS console.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Finding-User-s-Current-Session-tp6761060p6761305.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From michaelt at callall.co.za  Mon Sep  5 12:39:46 2011
From: michaelt at callall.co.za (Michael Toop)
Date: Mon, 5 Sep 2011 10:39:46 +0200
Subject: [Freeswitch-users] Bug or Not? "longjmp causes uninitialized stack
	frame"
Message-ID: <CAFmdODjqu2dVBXoCZbJmn+WftypQPqJCw=nS4adG4QMeg17-sA@mail.gmail.com>

Hi,

 Sorry not sure what to do with this, if this is a bug or another problem,
but getting this message in the bt when running FS as it does a core dump:
" longjmp causes uninitialized stack frame".

 Running on: 2.6.38-11-server #48-Ubuntu, Ubuntu 11.04 on the latest GIT
release.

Thanks,

Michael

2011-09-04 15:34:59.149089 [INFO] mod_native_file.c:94 Opening File
[/usr/local/freeswitch/sounds/custom/mwc-enter-destination.G729] 8000hz
2011-09-04 15:34:59.169011 [INFO] sofia.c:755 sofia/cellc/
763070366 at 172.103.0.36 Update Callee ID to "763070366" <763070366>
2011-09-04 15:34:59.169011 [NOTICE] switch_core_session.c:1367 Session 946
(sofia/cellc/790938072 at 172.103.0.36) Ended
2011-09-04 15:34:59.169011 [NOTICE] switch_core_session.c:1369 Close Channel
sofia/cellc/790938072 at 172.103.0.36 [CS_DESTROY]
*** longjmp causes uninitialized stack frame ***: freeswitch terminated
======= Backtrace: =========
/lib/x86_64-linux-gnu/libc.so.6(__fortify_fail+0x37)[0x7f1157fbe1d7]
/lib/x86_64-linux-gnu/libc.so.6(+0xfe169)[0x7f1157fbe169]
/lib/x86_64-linux-gnu/libc.so.6(__longjmp_chk+0x33)[0x7f1157fbe0d3]
/usr/lib/libcurl.so.4(+0xd165)[0x7f1156457165]
/lib/x86_64-linux-gnu/libpthread.so.0(+0xfc60)[0x7f1158a89c60]
/lib/x86_64-linux-gnu/libc.so.6(__select+0x33)[0x7f1157f9e143]
/usr/local/freeswitch/lib/libfreeswitch.so.1(apr_sleep+0x45)[0x7f11593d16c5]
/usr/local/freeswitch/lib/libfreeswitch.so.1(+0xcab5c)[0x7f11593a7b5c]
/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_console_loop+0x74a)[0x7f1159332e9a]
freeswitch[0x402b22]
/lib/x86_64-linux-gnu/libc.so.6(__libc_start_main+0xff)[0x7f1157edeeff]
freeswitch[0x401769]
======= Memory map: ========
00400000-00404000 r-xp 00000000 08:01 1845708
 /usr/local/freeswitch/bin/freeswitch
00603000-00604000 r--p 00003000 08:01 1845708
 /usr/local/freeswitch/bin/freeswitch
00604000-00605000 rw-p 00004000 08:01 1845708
 /usr/local/freeswitch/bin/freeswitch
017b0000-026c9000 rw-p 00000000 00:00 0
 [heap]
7f114004d000-7f114004e000 ---p 00000000 00:00 0
7f114004e000-7f1140089000 rw-p 00000000 00:00 0
7f1140089000-7f114008a000 ---p 00000000 00:00 0
7f114008a000-7f11400c5000 rw-p 00000000 00:00 0
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/f34d0eac/attachment-0001.html 

From g.kocjan at systemycallcenter.pl  Mon Sep  5 13:02:01 2011
From: g.kocjan at systemycallcenter.pl (kkocyk)
Date: Mon, 5 Sep 2011 02:02:01 -0700 (PDT)
Subject: [Freeswitch-users] Can't make mod_rtmp
In-Reply-To: <1308388959641-6490318.post@n2.nabble.com>
References: <1308334343913-6488229.post@n2.nabble.com>
	<BANLkTimx1MnAhfhHVmVds3qYpqNpaC9JKQ@mail.gmail.com>
	<1308388959641-6490318.post@n2.nabble.com>
Message-ID: <1315213321297-6760239.post@n2.nabble.com>


peely wrote:
> 
> I'm not sure if I need to do anything else, I just naively added
> endpoints/mod_rtmp to modules.conf and did a make clean install?
> 

I'm trying to make it same way (my first install was without mod_rtmp), but
after starting FS it seams that I still don't have rtmp. When I enter "rtmp
status" it says that there is no such command.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6760239.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From vikram.agrawal at gmail.com  Mon Sep  5 17:47:12 2011
From: vikram.agrawal at gmail.com (vikram)
Date: Mon, 5 Sep 2011 06:47:12 -0700 (PDT)
Subject: [Freeswitch-users] Can't make mod_rtmp
In-Reply-To: <1315213321297-6760239.post@n2.nabble.com>
References: <1308334343913-6488229.post@n2.nabble.com>
	<BANLkTimx1MnAhfhHVmVds3qYpqNpaC9JKQ@mail.gmail.com>
	<1308388959641-6490318.post@n2.nabble.com>
	<1315213321297-6760239.post@n2.nabble.com>
Message-ID: <1315230432566-6760858.post@n2.nabble.com>

I am facing the same issue.  I have made a fresh installation with mod_rtmp. 
 
unknown command: rtmp


kkocyk wrote:
> 
> 
> peely wrote:
>> 
>> I'm not sure if I need to do anything else, I just naively added
>> endpoints/mod_rtmp to modules.conf and did a make clean install?
>> 
> 
> I'm trying to make it same way (my first install was without mod_rtmp),
> but after starting FS it seams that I still don't have rtmp. When I enter
> "rtmp status" it says that there is no such command.
> 


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6760858.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kashif at kashifbukhari.com  Mon Sep  5 18:49:05 2011
From: kashif at kashifbukhari.com (Kashif Ali)
Date: Mon, 5 Sep 2011 19:49:05 +0500
Subject: [Freeswitch-users] vBilling Beta Program!!
In-Reply-To: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>
References: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>
Message-ID: <CANwDMSXYxNkGuUx=t56xOLYCaZ3Zxe3_t=ESFuNvSaG2p4GdiQ@mail.gmail.com>

admin password is not working


On Mon, Sep 5, 2011 at 7:45 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com>wrote:

>
> Hello everyone,
> As promised we have opened beta testing program for *vBilling*. An open
> source billing platform for FreeSWITCH. You are invited to login, take a
> look and play with it. Email us with your comments, let us know what
> improvements can be made. Following are the details for the program:
>
> ========================================
> vBilling User Panel: *http://demo.vbilling.org/*
> User Login: *demouser*
> User Password: *P at ssw0rd*
>
> vBilling Admin Panel: *http://demo.vbilling.org/admin/*
> Admin Login: *admin*
> Admin Password: *P at ssw0rd*
>
> We have configured a LIVE gateway for you. You can send your *SIP* calls
> to *demo.digitallinx.com* port *5060* and call any toll free (800 and 888
> only) number in US. Make calls, login to *vBilling* and see how the
> billing works.
> ========================================
>
> Most of the features are working. Some of the them mentioned on the site
> are still is development so be patient :) For product features and more
> details, visit our website at *http://www.vbilling.org/*<http://www.vbilling.org/>and us know what do you think about it.
>
>
> Regards,
> Muhammad Naseer
> CEO vBilling/Digital Linx
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/adec02dc/attachment-0001.html 

From kashif at kashifbukhari.com  Mon Sep  5 19:07:45 2011
From: kashif at kashifbukhari.com (Kashif Ali)
Date: Mon, 5 Sep 2011 20:07:45 +0500
Subject: [Freeswitch-users] vBilling Beta Program!!
In-Reply-To: <1B19ABD72889C245AE8EEE08AC24103A63A769F311@exmachina.office.kapper.net>
References: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A63A769F311@exmachina.office.kapper.net>
Message-ID: <CANwDMSWKA-MOmEFn_mOueS+DkMFZg2WTTpFRkiDetNVVeX564A@mail.gmail.com>

i was able to login to the user interface, it seems some one changed the
passwords.

*you should disable the change password feature in demo.
*
On Mon, Sep 5, 2011 at 7:55 PM, Clemens Ebentheuer <ce at kapper.net> wrote:

> Hi,****
>
> ** **
>
> got Invalid Email/Password  here ? both user and admin. Too fast?****
>
> ** **
>
> Thx,****
>
> ** **
>
> Clemens****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Muhammad
> Naseer Bhatti
> *Sent:* Monday, September 05, 2011 4:46 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] vBilling Beta Program!!****
>
> ** **
>
>
> Hello everyone,
> As promised we have opened beta testing program for *vBilling*. An open
> source billing platform for FreeSWITCH. You are invited to login, take a
> look and play with it. Email us with your comments, let us know what
> improvements can be made. Following are the details for the program:
>
> ========================================
> vBilling User Panel: *http://demo.vbilling.org/*
> User Login: *demouser*
> User Password: *P at ssw0rd*
>
> vBilling Admin Panel: *http://demo.vbilling.org/admin/*
> Admin Login: *admin*
> Admin Password: *P at ssw0rd*
>
> We have configured a LIVE gateway for you. You can send your *SIP* calls
> to *demo.digitallinx.com* port *5060* and call any toll free (800 and 888
> only) number in US. Make calls, login to *vBilling* and see how the
> billing works.
> ========================================
>
> Most of the features are working. Some of the them mentioned on the site
> are still is development so be patient :) For product features and more
> details, visit our website at *http://www.vbilling.org/*<http://www.vbilling.org/>and us know what do you think about it.
>
>
> Regards,
> Muhammad Naseer
> CEO vBilling/Digital Linx****
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/1bee58fe/attachment-0001.html 

From kashif at kashifbukhari.com  Mon Sep  5 19:13:14 2011
From: kashif at kashifbukhari.com (Kashif Ali)
Date: Mon, 5 Sep 2011 20:13:14 +0500
Subject: [Freeswitch-users] vBilling Beta Program!!
In-Reply-To: <CALNZuw=Pd7vpDZpw_vaodrGzKrCg5vRiEeAbO6OJCz75kKgf3g@mail.gmail.com>
References: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A63A769F311@exmachina.office.kapper.net>
	<CALNZuw=Pd7vpDZpw_vaodrGzKrCg5vRiEeAbO6OJCz75kKgf3g@mail.gmail.com>
Message-ID: <CANwDMSXg1ctK2cfKJygJSeyv1YVPS94nTf025H9_3+m_o+d_BQ@mail.gmail.com>

it looks like test is successful now its time to change back to default
password:P
WHO EVER CHANGED IT, i request him/her please change back to
its original status so others can test the demo.

On Mon, Sep 5, 2011 at 8:05 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Only user worked for me, maybe someone was testing out the change password
> feature?
> -Avi
>
> On Mon, Sep 5, 2011 at 5:55 PM, Clemens Ebentheuer <ce at kapper.net> wrote:
>
>> Hi,****
>>
>> ** **
>>
>> got Invalid Email/Password  here ? both user and admin. Too fast?****
>>
>> ** **
>>
>> Thx,****
>>
>> ** **
>>
>> Clemens****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Muhammad
>> Naseer Bhatti
>> *Sent:* Monday, September 05, 2011 4:46 PM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* [Freeswitch-users] vBilling Beta Program!!****
>>
>> ** **
>>
>>
>> Hello everyone,
>> As promised we have opened beta testing program for *vBilling*. An open
>> source billing platform for FreeSWITCH. You are invited to login, take a
>> look and play with it. Email us with your comments, let us know what
>> improvements can be made. Following are the details for the program:
>>
>> ========================================
>> vBilling User Panel: *http://demo.vbilling.org/*
>> User Login: *demouser*
>> User Password: *P at ssw0rd*
>>
>> vBilling Admin Panel: *http://demo.vbilling.org/admin/*
>> Admin Login: *admin*
>> Admin Password: *P at ssw0rd*
>>
>> We have configured a LIVE gateway for you. You can send your *SIP* calls
>> to *demo.digitallinx.com* port *5060* and call any toll free (800 and 888
>> only) number in US. Make calls, login to *vBilling* and see how the
>> billing works.
>> ========================================
>>
>> Most of the features are working. Some of the them mentioned on the site
>> are still is development so be patient :) For product features and more
>> details, visit our website at *http://www.vbilling.org/*<http://www.vbilling.org/>and us know what do you think about it.
>>
>>
>> Regards,
>> Muhammad Naseer
>> CEO vBilling/Digital Linx****
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/668d80b2/attachment.html 

From kashif at kashifbukhari.com  Mon Sep  5 19:22:16 2011
From: kashif at kashifbukhari.com (Kashif Ali)
Date: Mon, 5 Sep 2011 20:22:16 +0500
Subject: [Freeswitch-users] vBilling Beta Program!!
In-Reply-To: <1315235727102-6761088.post@n2.nabble.com>
References: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A63A769F311@exmachina.office.kapper.net>
	<CALNZuw=Pd7vpDZpw_vaodrGzKrCg5vRiEeAbO6OJCz75kKgf3g@mail.gmail.com>
	<1315235727102-6761088.post@n2.nabble.com>
Message-ID: <CANwDMSUfi5_-X28e2itCoK1+bdcWWH1tP6xS-AymS3YTrj5+uA@mail.gmail.com>

is it full featured or beta version?

On Mon, Sep 5, 2011 at 8:15 PM, nbhatti <nbhatti at gmail.com> wrote:

> I have disabled the admin password change for now :)
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p6761088.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/f4f6118d/attachment.html 

From brian.wiese.freeswitch at gmail.com  Mon Sep  5 21:41:39 2011
From: brian.wiese.freeswitch at gmail.com (Brian Wiese FreeSWITCH List)
Date: Mon, 5 Sep 2011 12:41:39 -0500
Subject: [Freeswitch-users] Multiple DTMF on Single Keypress
Message-ID: <CADEQS5zp8bh3epB3-ya9uR2HvNjnt8qf7u4rS=a11tNNGGo7hA@mail.gmail.com>

Hello everyone!

I'm getting multiple RTP DTMF from random keypresses and I can't figure out
why.  I've PB'ed the packet capture and FS log for a call.  As you can see
from the FS log, the 9,8,7,6 numbers weren't repeated, but 5 was (again,
though, different calls lead to different numbers being repeated).
Log:  http://pastebin.freeswitch.org/17280
Capture:  http://pastebin.freeswitch.org/17282

I appreciate any ideas as to what I might have wrong here.

Thanks.

~Brian
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/70781f70/attachment.html 

From a.afzali2003 at gmail.com  Mon Sep  5 22:46:38 2011
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Mon, 5 Sep 2011 23:16:38 +0430
Subject: [Freeswitch-users] Finding User's Current Session
In-Reply-To: <1315240242664-6761305.post@n2.nabble.com>
References: <CABK0fC7GLpqneudT753L3m2W72jHTVW82t3BZNmCe843O7cZCA@mail.gmail.com>
	<1315240242664-6761305.post@n2.nabble.com>
Message-ID: <CABK0fC6w9z2rb-yLQQpCbwLT1xfidQVj_NBFXpAgkB9Pz-Bn1w@mail.gmail.com>

I hoped an API which accept user name as input param !
Thanks

On Mon, Sep 5, 2011 at 9:00 PM, nbhatti <nbhatti at gmail.com> wrote:

>
> Try "show calls" at FS console.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Finding-User-s-Current-Session-tp6761060p6761305.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/78cdcd55/attachment.html 

From nbhatti at gmail.com  Mon Sep  5 22:49:04 2011
From: nbhatti at gmail.com (nbhatti)
Date: Mon, 5 Sep 2011 11:49:04 -0700 (PDT)
Subject: [Freeswitch-users] vBilling Beta Program!!
In-Reply-To: <CANwDMSWKA-MOmEFn_mOueS+DkMFZg2WTTpFRkiDetNVVeX564A@mail.gmail.com>
References: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A63A769F311@exmachina.office.kapper.net>
	<CANwDMSWKA-MOmEFn_mOueS+DkMFZg2WTTpFRkiDetNVVeX564A@mail.gmail.com>
Message-ID: <CACak_yybu30gKmkDHQV6vvogxLtyAj9q3mODJD48qdshsrhXNw@mail.gmail.com>

The password change is already disabled and the old password is still valid.
Please be case sEnsaT1ve. Password is "P at ssw0rd" (Without quotes)

On Mon, Sep 5, 2011 at 8:54 PM, Kashif Ali [via freeswitch-users] <
ml-node+6761527-1375959898-297835 at n2.nabble.com> wrote:

> i was able to login to the user interface, it seems some one changed the
> passwords.
>
> *you should disable the change password feature in demo.
> *
> On Mon, Sep 5, 2011 at 7:55 PM, Clemens Ebentheuer <[hidden email]<http://user/SendEmail.jtp?type=node&node=6761527&i=0>
> > wrote:
>
>> Hi,****
>>
>> ** **
>>
>> got Invalid Email/Password  here ? both user and admin. Too fast?****
>>
>> ** **
>>
>> Thx,****
>>
>> ** **
>>
>> Clemens****
>>
>> ** **
>>
>> *From:* [hidden email]<http://user/SendEmail.jtp?type=node&node=6761527&i=1>[mailto:[hidden
>> email] <http://user/SendEmail.jtp?type=node&node=6761527&i=2>] *On Behalf
>> Of *Muhammad Naseer Bhatti
>>
>> *Sent:* Monday, September 05, 2011 4:46 PM
>> *To:* [hidden email]<http://user/SendEmail.jtp?type=node&node=6761527&i=3>
>>
>> *Subject:* [Freeswitch-users] vBilling Beta Program!!****
>>
>> ** **
>>
>>
>> Hello everyone,
>> As promised we have opened beta testing program for *vBilling*. An open
>> source billing platform for FreeSWITCH. You are invited to login, take a
>> look and play with it. Email us with your comments, let us know what
>> improvements can be made. Following are the details for the program:
>>
>> ========================================
>> vBilling User Panel: *http://demo.vbilling.org/*
>> User Login: *demouser*
>> User Password: *P at ssw0rd*
>>
>> vBilling Admin Panel: *http://demo.vbilling.org/admin/*
>> Admin Login: *admin*
>> Admin Password: *P at ssw0rd*
>>
>> We have configured a LIVE gateway for you. You can send your *SIP* calls
>> to *demo.digitallinx.com* port *5060* and call any toll free (800 and 888
>> only) number in US. Make calls, login to *vBilling* and see how the
>> billing works.
>> ========================================
>>
>> Most of the features are working. Some of the them mentioned on the site
>> are still is development so be patient :) For product features and more
>> details, visit our website at *http://www.vbilling.org/*<http://www.vbilling.org/>and us know what do you think about it.
>>
>>
>> Regards,
>> Muhammad Naseer
>> CEO vBilling/Digital Linx****
>>
>>
>>
>> FreeSWITCH-users mailing list
>> [hidden email] <http://user/SendEmail.jtp?type=node&node=6761527&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=6761527&i=5>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
>  If you reply to this email, your message will be added to the discussion
> below:
>
> http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p6761527.html
>  To unsubscribe from vBilling Beta Program!!, click here<http://freeswitch-users.2379917.n2.nabble.com/template/NamlServlet.jtp?macro=unsubscribe_by_code&node=6760999&code=bmJoYXR0aUBnbWFpbC5jb218Njc2MDk5OXwxMDEwNzIwMzM1>.
>
>


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p6761673.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/24d5e4de/attachment-0001.html 

From magnus.kelly at gmail.com  Mon Sep  5 23:12:03 2011
From: magnus.kelly at gmail.com (Magnus.Kelly)
Date: Mon, 5 Sep 2011 20:12:03 +0100
Subject: [Freeswitch-users] vBilling Beta Program!!
In-Reply-To: <CACak_yybu30gKmkDHQV6vvogxLtyAj9q3mODJD48qdshsrhXNw@mail.gmail.com>
References: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A63A769F311@exmachina.office.kapper.net>
	<CANwDMSWKA-MOmEFn_mOueS+DkMFZg2WTTpFRkiDetNVVeX564A@mail.gmail.com>
	<CACak_yybu30gKmkDHQV6vvogxLtyAj9q3mODJD48qdshsrhXNw@mail.gmail.com>
Message-ID: <F50D8803-06B5-4E09-A825-A55270B03876@gmail.com>

Looks interesting, but should the add reseller function work in this beta?

Magnus

On 5 Sep 2011, at 19:49, nbhatti <nbhatti at gmail.com> wrote:

> 
> The password change is already disabled and the old password is still valid. Please be case sEnsaT1ve. Password is "P at ssw0rd" (Without quotes)
> 
> On Mon, Sep 5, 2011 at 8:54 PM, Kashif Ali [via freeswitch-users] <[hidden email]> wrote:
> i was able to login to the user interface, it seems some one changed the passwords.
> 
> you should disable the change password feature in demo. 
> 
> On Mon, Sep 5, 2011 at 7:55 PM, Clemens Ebentheuer <[hidden email]> wrote:
> Hi,
> 
>  
> 
> got Invalid Email/Password  here ? both user and admin. Too fast?
> 
>  
> 
> Thx,
> 
>  
> 
> Clemens
> 
>  
> 
> From: [hidden email] [mailto:[hidden email]] On Behalf Of Muhammad Naseer Bhatti
> 
> 
> Sent: Monday, September 05, 2011 4:46 PM
> To: [hidden email]
> 
> Subject: [Freeswitch-users] vBilling Beta Program!!
>  
> 
> 
> 
> Hello everyone,
> As promised we have opened beta testing program for vBilling. An open source billing platform for FreeSWITCH. You are invited to login, take a look and play with it. Email us with your comments, let us know what improvements can be made. Following are the details for the program:
> 
> ========================================
> vBilling User Panel: http://demo.vbilling.org/
> User Login: demouser
> User Password: P at ssw0rd
> 
> vBilling Admin Panel: http://demo.vbilling.org/admin/
> Admin Login: admin
> Admin Password: P at ssw0rd
> 
> We have configured a LIVE gateway for you. You can send your SIP calls to demo.digitallinx.com port 5060 and call any toll free (800 and 888 only) number in US. Make calls, login to vBilling and see how the billing works.
> ========================================
> 
> Most of the features are working. Some of the them mentioned on the site are still is development so be patient :) For product features and more details, visit our website at http://www.vbilling.org/ and us know what do you think about it.
> 
> 
> Regards,
> Muhammad Naseer
> CEO vBilling/Digital Linx
> 
> 
> 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
> 
> 
> If you reply to this email, your message will be added to the discussion below:
> http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p6761527.html
> To unsubscribe from vBilling Beta Program!!, click here.
> 
> 
> View this message in context: Re: vBilling Beta Program!!
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/c453b735/attachment.html 

From acrow at integrafin.co.uk  Tue Sep  6 00:24:09 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 05 Sep 2011 21:24:09 +0100
Subject: [Freeswitch-users] Hold and BLF - disable flashing or pickup for
 held call?
Message-ID: <4E652FE9.5010600@integrafin.co.uk>

Hi,

Anyone know if there is a way to change notify behaviour in FS to avoid 
the below confusion (eg on Snom, you can intercept a call that someone 
you monitor has held), up to and including disabling NOTIFY for a held 
endpoint but not for a ringing one?

Thanks,

Alex

*******

Hi,

I have some snom 370s, and I noticed that when a monitored extension has 
held a call then the corresponding BLF lamp flashes exactly as if the 
monitored extension is ringing, and it is then possible to steal the 
call from the "holder" by pressing the button.

Ideally I'd like neither of these things to happen. Is it possible to 
disable one or both for a held call?

I have a polycom IP 650 with productivity licence and it shows the 
status of the holding extension as ringing (with the musical notes 
animation on the status display - not great) but at least it doesn't 
steal the call.

Cheers

Alex

-- 
This message is intended only for the addressee and may contain
confidential information.  Unless you are that person, you may not
disclose its contents or use it in any way and are requested to delete
the message along with any attachments and notify us immediately.

"Transact" is operated by Integrated Financial Arrangements plc
Domain House, 5-7 Singer Street, London  EC2A 4BQ
Tel: (020) 7608 4900 Fax: (020) 7608 5300
(Registered office: as above; Registered in England and Wales under 
number: 3727592)
Authorised and regulated by the Financial Services Authority (entered on 
the FSA Register; number: 190856)



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

-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.



From avi at avimarcus.net  Tue Sep  6 02:08:05 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 6 Sep 2011 01:08:05 +0300
Subject: [Freeswitch-users] Opposite of flush_dtmf - grab dtmf from before
	the IVR + debug rfc2833
Message-ID: <CALNZuwmvEvmzOYCjq7tumZWJ2bznZWU1zQhdrYqbGYTe8TdMUA@mail.gmail.com>

I have a calling card IVR and upon an error, it announces the error, then
goes back to the calling card IVR. How can I use the digits they started
dialing during the pre-IVR announcement as part of the IVR?

e.g. http://pastebin.freeswitch.org/17283
During the announcement of the "wrong" dialed number, I started dialing and
FS got the digits. But it didn't start "listening" until... just before (??)
the calling_card_short IVR started.
How can I get it to listen to everything that got queued up?

Thanks,
-Avi

p.s. is there somelinke like ngrep/tcpdump on regular SIP packets to
live-view the rfc-2833 that come in, before they even hit FS? I know
wireshark can understand them in a pcap, but I couldn't figure out how to
filter live...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/f92ef711/attachment.html 

From brian at freeswitch.org  Tue Sep  6 02:15:32 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Sep 2011 17:15:32 -0500
Subject: [Freeswitch-users] Hold and BLF - disable flashing or pickup
	for held call?
In-Reply-To: <4E652FE9.5010600@integrafin.co.uk>
References: <4E652FE9.5010600@integrafin.co.uk>
Message-ID: <EC5966E8-12BE-4C8A-B36E-5E0D0E0120D2@freeswitch.org>

Are you on the 8.x firmware?

/b

On Sep 5, 2011, at 3:24 PM, Alex Crow wrote:

> Hi,
> 
> Anyone know if there is a way to change notify behaviour in FS to avoid 
> the below confusion (eg on Snom, you can intercept a call that someone 
> you monitor has held), up to and including disabling NOTIFY for a held 
> endpoint but not for a ringing one?
> 
> Thanks,
> 
> Alex
> 
> *******
> 
> Hi,
> 
> I have some snom 370s, and I noticed that when a monitored extension has 
> held a call then the corresponding BLF lamp flashes exactly as if the 
> monitored extension is ringing, and it is then possible to steal the 
> call from the "holder" by pressing the button.
> 
> Ideally I'd like neither of these things to happen. Is it possible to 
> disable one or both for a held call?
> 
> I have a polycom IP 650 with productivity licence and it shows the 
> status of the holding extension as ringing (with the musical notes 
> animation on the status display - not great) but at least it doesn't 
> steal the call.
> 
> Cheers
> 
> Alex

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/5c2e4586/attachment-0001.html 

From mays.david at gmail.com  Tue Sep  6 06:01:24 2011
From: mays.david at gmail.com (David Ma)
Date: Tue, 6 Sep 2011 10:01:24 +0800
Subject: [Freeswitch-users] Will bypass-media work when a call is
 bridged to multiple SIP destination
In-Reply-To: <CALNZuw=gQ6nzEufq2NmRZ81P+EdsGfPjH2u2fpHDgno8D3LQxQ@mail.gmail.com>
References: <1315230976599-6760883.post@n2.nabble.com>
	<CALNZuw=gQ6nzEufq2NmRZ81P+EdsGfPjH2u2fpHDgno8D3LQxQ@mail.gmail.com>
Message-ID: <CAMLBmR+14mhi5huYFVJRai0zqWwFv2A3yodphzxfxD8LVV7gCQ@mail.gmail.com>

Hi Avi,

Thanks for your prompt response. My objective is to bridge the incoming call
to multiple contacts with a single "bridge" so that 3 or more parties are
involved in the call. I know that FreeSwitch support such a bridge like

<action application="bridge" data="sofia/internal/1010 at sip.yourprovider.com
,sofia/sip/1011 at sip.yourprovider.com"/>

I just wish bypass-media works in such a scenario so that the RTP packets
(media) doesn't flow through my FreeSwitch box when 3 or more parties are
talking.

Thanks a lot!
D.Ma

On Mon, Sep 5, 2011 at 10:05 PM, Avi Marcus <avi at avimarcus.net> wrote:

> You mean you fork the dialing so it fails over or tries different places at
> once?
> Sure. You can even set the bypass_media variable differently on each leg.
>
> -Avi
>
>
> On Mon, Sep 5, 2011 at 4:56 PM, dma <mays.david at gmail.com> wrote:
>
>> Hi Support,
>>
>> This is a simple question but I don't find an answer from the web.
>>
>> I hope to know whether bypass-media work when a call is bridged to
>> multiple
>> SIP destination?
>>
>> Thanks,
>> D.Ma
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Will-bypass-media-work-when-a-call-is-bridged-to-multiple-SIP-destination-tp6760883p6760883.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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/592d0074/attachment.html 

From jack at livecall.com  Tue Sep  6 06:39:54 2011
From: jack at livecall.com (Jack)
Date: Mon, 05 Sep 2011 19:39:54 -0700
Subject: [Freeswitch-users] Can't make mod_rtmp
In-Reply-To: <1315230432566-6760858.post@n2.nabble.com>
References: <1308334343913-6488229.post@n2.nabble.com>
	<BANLkTimx1MnAhfhHVmVds3qYpqNpaC9JKQ@mail.gmail.com>
	<1308388959641-6490318.post@n2.nabble.com>
	<1315213321297-6760239.post@n2.nabble.com>
	<1315230432566-6760858.post@n2.nabble.com>
Message-ID: <4E6587FA.2040307@livecall.com>

There is also a rtmp.conf.xml in the autoload_configs folder.
It sounds like you need to include the rtmp source in your build.
src\mod\endpoints\mod_rtmp\mod_rtmp.c
I do not think it builds by default.
Jack

On 9/5/2011 6:47 AM, vikram wrote:
> I am facing the same issue.  I have made a fresh installation with mod_rtmp.
>
> unknown command: rtmp
>
>
> kkocyk wrote:
>>
>> peely wrote:
>>> I'm not sure if I need to do anything else, I just naively added
>>> endpoints/mod_rtmp to modules.conf and did a make clean install?
>>>
>> I'm trying to make it same way (my first install was without mod_rtmp),
>> but after starting FS it seams that I still don't have rtmp. When I enter
>> "rtmp status" it says that there is no such command.
>>
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6760858.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 xing2kin at yahoo.com  Tue Sep  6 08:43:29 2011
From: xing2kin at yahoo.com (king2kin)
Date: Mon, 5 Sep 2011 21:43:29 -0700 (PDT)
Subject: [Freeswitch-users] mod_say_zh the Chinese way discussion
In-Reply-To: <51B6F146553F48D9B42E8AEEC41C1F6F@gmail.com>
Message-ID: <1315284209.85229.YahooMailClassic@web39706.mail.mud.yahoo.com>

Hi Seven,
?
I ever tried mod_say_zh, but only tested those phrase and?play-file/streamFile/playback. 
Did you fix mod_say_zh and check in your patches in GIT head version?
?
x.k.

--- On Mon, 9/5/11, Seven Du <dujinfang at gmail.com> wrote:


From: Seven Du <dujinfang at gmail.com>
Subject: Re: [Freeswitch-users] mod_say_zh the Chinese way discussion
To: "freeswitch-users" <freeswitch-users at lists.freeswitch.org>
Cc: "freeswitch-dev" <freeswitch-dev at lists.freeswitch.org>
Date: Monday, September 5, 2011, 3:10 AM



Ah, sorry, please ignore this for now, I loaded the wrong(english) module for test.




On Monday, September 5, 2011 at 5:45 PM, Seven Du wrote:



Hi all,


First sorry I send both to users and devs as I want to talk in both groups.


I wonder how many of you actually use mod_say_zh


I had submitted a patch but not been accepted because it was argued that it might breaks ? ?zh in other locations than China mainland.


http://jira.freeswitch.org/browse/FS-2809 


How ever, I doubt is it useful actually. Besides the above example, think of the following




when say time, EN is Sept 1, 2011 and mod_say_zh says?


9 yue di 15, 11?


Actually in China we say?


2011 nian(year) 9 yue (month) 15 ri(day)


Again, I'd like to know if anyone is actually using the current way.




So I need a PRC way, questions are:


1) Is it safe to change the default behavior?


2) If not how could I change? ?add a channel variable? or


3) create another dialect mod_say_zh_cn ?


Thanks.


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

Sent with Sparrow




-----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
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110905/d04252f3/attachment.html 

From avi at avimarcus.net  Tue Sep  6 09:20:41 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 6 Sep 2011 08:20:41 +0300
Subject: [Freeswitch-users] Will bypass-media work when a call is
 bridged to multiple SIP destination
In-Reply-To: <CAMLBmR+14mhi5huYFVJRai0zqWwFv2A3yodphzxfxD8LVV7gCQ@mail.gmail.com>
References: <1315230976599-6760883.post@n2.nabble.com>
	<CALNZuw=gQ6nzEufq2NmRZ81P+EdsGfPjH2u2fpHDgno8D3LQxQ@mail.gmail.com>
	<CAMLBmR+14mhi5huYFVJRai0zqWwFv2A3yodphzxfxD8LVV7gCQ@mail.gmail.com>
Message-ID: <CALNZuwmkLRmW_nsgb8qXpg6Oht7Vv56QBvW5QZ0rqTh_m5-jEw@mail.gmail.com>

When you use , to bridge to multiple locations, that's simul-ring.
FreeSWITCH only connects the call to the FIRST one that picks up.
If you want to connect more than one source and one endpoint, you will need
to use a conference to mix the media locally, and you can't bypass_media on
that...

-Avi

On Tue, Sep 6, 2011 at 5:01 AM, David Ma <mays.david at gmail.com> wrote:

>
> Hi Avi,
>
> Thanks for your prompt response. My objective is to bridge the incoming
> call to multiple contacts with a single "bridge" so that 3 or more parties
> are involved in the call. I know that FreeSwitch support such a bridge like
>
> <action application="bridge" data="sofia/internal/
> 1010 at sip.yourprovider.com,sofia/sip/1011 at sip.yourprovider.com"/>
>
> I just wish bypass-media works in such a scenario so that the RTP packets
> (media) doesn't flow through my FreeSwitch box when 3 or more parties are
> talking.
>
> Thanks a lot!
> D.Ma
>
>
> On Mon, Sep 5, 2011 at 10:05 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> You mean you fork the dialing so it fails over or tries different places
>> at once?
>> Sure. You can even set the bypass_media variable differently on each leg.
>>
>> -Avi
>>
>>
>> On Mon, Sep 5, 2011 at 4:56 PM, dma <mays.david at gmail.com> wrote:
>>
>>> Hi Support,
>>>
>>> This is a simple question but I don't find an answer from the web.
>>>
>>> I hope to know whether bypass-media work when a call is bridged to
>>> multiple
>>> SIP destination?
>>>
>>> Thanks,
>>> D.Ma
>>>
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/Will-bypass-media-work-when-a-call-is-bridged-to-multiple-SIP-destination-tp6760883p6760883.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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/136a32c2/attachment-0001.html 

From kbdfck at gmail.com  Tue Sep  6 09:36:06 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 6 Sep 2011 09:36:06 +0400
Subject: [Freeswitch-users] Multiple DTMF on Single Keypress
In-Reply-To: <CADEQS5zp8bh3epB3-ya9uR2HvNjnt8qf7u4rS=a11tNNGGo7hA@mail.gmail.com>
References: <CADEQS5zp8bh3epB3-ya9uR2HvNjnt8qf7u4rS=a11tNNGGo7hA@mail.gmail.com>
Message-ID: <CA+Gmxbbm46vCtW70+Qt7LurJEybMHwDA6BA+tNa8OnL4U0uKfw@mail.gmail.com>

See the same behaviour with inband DTMF detector sometimes.

2011/9/5 Brian Wiese FreeSWITCH List <brian.wiese.freeswitch at gmail.com>

> Hello everyone!
>
> I'm getting multiple RTP DTMF from random keypresses and I can't figure out
> why.  I've PB'ed the packet capture and FS log for a call.  As you can see
> from the FS log, the 9,8,7,6 numbers weren't repeated, but 5 was (again,
> though, different calls lead to different numbers being repeated).
> Log:  http://pastebin.freeswitch.org/17280
> Capture:  http://pastebin.freeswitch.org/17282
>
> I appreciate any ideas as to what I might have wrong here.
>
> 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
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/f3f2c536/attachment.html 

From acrow at integrafin.co.uk  Tue Sep  6 12:10:38 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Tue, 06 Sep 2011 09:10:38 +0100
Subject: [Freeswitch-users] Hold and BLF - disable flashing or pickup
 for held call?
In-Reply-To: <EC5966E8-12BE-4C8A-B36E-5E0D0E0120D2@freeswitch.org>
References: <4E652FE9.5010600@integrafin.co.uk>
	<EC5966E8-12BE-4C8A-B36E-5E0D0E0120D2@freeswitch.org>
Message-ID: <4E65D57E.4090105@integrafin.co.uk>

On 05/09/11 23:15, Brian West wrote:
> Are you on the 8.x firmware?
>
> /b
>

This is on 8.x, I think it's also the same on 7.3.30

Cheers

Alex



From basit.engg at gmail.com  Tue Sep  6 12:26:32 2011
From: basit.engg at gmail.com (Abdul Basit)
Date: Tue, 6 Sep 2011 13:26:32 +0500
Subject: [Freeswitch-users] vBilling Beta Program!!
In-Reply-To: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>
References: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>
Message-ID: <CAGQG5Mbskb7e24igA_AeBpHWVmXKJ_E-fkj3+SN3wH39ioJLzg@mail.gmail.com>

Interesting...

Any max call limits? what is cps?
We will appreciate stress test results if anyone can share.

-- 
Regards,

Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445



On Mon, Sep 5, 2011 at 7:45 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com>wrote:

>
> Hello everyone,
> As promised we have opened beta testing program for *vBilling*. An open
> source billing platform for FreeSWITCH. You are invited to login, take a
> look and play with it. Email us with your comments, let us know what
> improvements can be made. Following are the details for the program:
>
> ========================================
> vBilling User Panel: *http://demo.vbilling.org/*
> User Login: *demouser*
> User Password: *P at ssw0rd*
>
> vBilling Admin Panel: *http://demo.vbilling.org/admin/*
> Admin Login: *admin*
> Admin Password: *P at ssw0rd*
>
> We have configured a LIVE gateway for you. You can send your *SIP* calls
> to *demo.digitallinx.com* port *5060* and call any toll free (800 and 888
> only) number in US. Make calls, login to *vBilling* and see how the
> billing works.
> ========================================
>
> Most of the features are working. Some of the them mentioned on the site
> are still is development so be patient :) For product features and more
> details, visit our website at *http://www.vbilling.org/*<http://www.vbilling.org/>and us know what do you think about it.
>
>
> Regards,
> Muhammad Naseer
> CEO vBilling/Digital Linx
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/2b55ef1c/attachment.html 

From nbhatti at gmail.com  Tue Sep  6 12:35:52 2011
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Tue, 6 Sep 2011 11:35:52 +0300
Subject: [Freeswitch-users] vBilling Beta Program!!
In-Reply-To: <CAGQG5Mbskb7e24igA_AeBpHWVmXKJ_E-fkj3+SN3wH39ioJLzg@mail.gmail.com>
References: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>
	<CAGQG5Mbskb7e24igA_AeBpHWVmXKJ_E-fkj3+SN3wH39ioJLzg@mail.gmail.com>
Message-ID: <CACak_yxUrzNGL2-bgb_ekEXTbBivS9e7f6-SM49+z8cyx_rr3A@mail.gmail.com>

Our last test showed around 500 concurrent calls. Since we support
distributed setups, so in case you need more numbers, simply add a new
machine running FreeSWITCH and you are done. Billing interface will be the
same running on 1 single node. We are going to publish some benchmarks in a
few days. It is actually undergoing some real bad test by one of our
customers :) Stay tuned.

On Tue, Sep 6, 2011 at 11:26 AM, Abdul Basit <basit.engg at gmail.com> wrote:

> Interesting...
>
> Any max call limits? what is cps?
> We will appreciate stress test results if anyone can share.
>
> --
> Regards,
>
> Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445
>
>
>
> On Mon, Sep 5, 2011 at 7:45 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com>wrote:
>
>>
>> Hello everyone,
>> As promised we have opened beta testing program for *vBilling*. An open
>> source billing platform for FreeSWITCH. You are invited to login, take a
>> look and play with it. Email us with your comments, let us know what
>> improvements can be made. Following are the details for the program:
>>
>> ========================================
>> vBilling User Panel: *http://demo.vbilling.org/*
>> User Login: *demouser*
>> User Password: *P at ssw0rd*
>>
>> vBilling Admin Panel: *http://demo.vbilling.org/admin/*
>> Admin Login: *admin*
>> Admin Password: *P at ssw0rd*
>>
>> We have configured a LIVE gateway for you. You can send your *SIP* calls
>> to *demo.digitallinx.com* port *5060* and call any toll free (800 and 888
>> only) number in US. Make calls, login to *vBilling* and see how the
>> billing works.
>> ========================================
>>
>> Most of the features are working. Some of the them mentioned on the site
>> are still is development so be patient :) For product features and more
>> details, visit our website at *http://www.vbilling.org/*<http://www.vbilling.org/>and us know what do you think about it.
>>
>>
>> Regards,
>> Muhammad Naseer
>> CEO vBilling/Digital Linx
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/ff7acf0c/attachment.html 

From asilva at wirelessmundi.com  Tue Sep  6 13:25:05 2011
From: asilva at wirelessmundi.com (Antonio)
Date: Tue, 06 Sep 2011 11:25:05 +0200
Subject: [Freeswitch-users] Multiple Tone Detect
In-Reply-To: <CAKbxfG-UHXsRstwp0Ut3zz-BQDxumyfZXZSCLWoRBNqAt=-qvQ@mail.gmail.com>
References: <201108281613.18871.justlikeef@gmail.com>
	<CAKzWOxXGmU2G4Eqz4FX2tV223Cb4dW1zSg_-qnnaLA4kEcUngA@mail.gmail.com>
	<CAKbxfG-UHXsRstwp0Ut3zz-BQDxumyfZXZSCLWoRBNqAt=-qvQ@mail.gmail.com>
Message-ID: <1315301105.3761.138.camel@marces.madrid.commsmundi.com>

Hi 

Using spandsp_start_fax_detect instead tone_detect doesn't work.

I have the latest git head.

My dialplan works whit tone_detect.


<extension name="2103">
<condition field="destination_number" expression="^2103$">
<!--
<action application="answer"/>
<action application="tone_detect" data="fax 1100 r +3000 transfer
2103_rxfax XML default"/>
<action application="sleep" data="3000"/>
-->
<action application="pre_answer"/>
<action application="spandsp_start_fax_detect" data="transfer
'2103_rxfax XML default' 3"/>
<action application="sleep" data="3000"/>
<action application="set" data="ringback=%(1500,3000,425.0)"/>
<action application="bridge"
data="[leg_timeout=60]user/2103 at domain.com"/>
</condition>
</extension>


<extension name="2103_rxfax">
<condition field="destination_number" expression="^2103_rxfax$">
<action application="answer"/>
<action application="playback" data="silence_stream://3000"/>
<action application="lua" data="rxfax.lua ${user_data(2103 at domain.com
var faxbox)} domain.com"/>
</condition>
</extension>






On Mon, 2011-08-29 at 13:13 -0500, Anthony Minessale wrote: 

> there is a specific app for fax detecting inside mod spandsp called
> spandsp_start_fax_detect
> 
> 
> it takes about the same input args namely <app> [<arg>][ <timeout>]
> It uses the official fax identification code inside spandsp and works
> much better than the single tone detect way.
> 
> 
> 
> On Mon, Aug 29, 2011 at 12:59 PM, Michael Collins <msc at freeswitch.org> wrote:
> > If you need multiple tone detects then definitely use the first method. I've
> > done as many as 6 tone_detects on a single call and it works well.
> > -MC
> >
> > On Sun, Aug 28, 2011 at 1:13 PM, Rob Hutton <justlikeef at gmail.com> wrote:
> >>
> >> I am trying to get tone detection set up so that a single number can be
> >> used for both voice and fax.  Tone detection needs to run concurrent to the
> >> normal call processing, so that a call proceeds normally unless the fax
> >> tones are heard.
> >>
> >> On this wiki page
> >> (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect) it gives
> >> the format:
> >>
> >> <action application="tone_detect" data="fax 1100 r +5000 transfer fax XML
> >> default" />
> >>
> >> and an associated dialplan entry for the call to be transfered to that
> >> actually handles the fax reception.
> >>
> >> On this wiki page
> >> (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect) it says
> >> that the format from the first page is not the preferred method, and that
> >> the format should be something like:
> >>
> >> <action application="set" data="execute_on_tone_detect=transfer fax XML
> >> default"/>
> >> <action application="set" data="tone_detect_hits=1"/>
> >> <action application="set" data="execute_on_media=tone_detect mytone 1100 r
> >> +5000"/>
> >>
> >> My question is, using the first format, I could set up multiple
> >> detections, eg. one for fax tones that sends/receives a fax, one for
> >> answering machines tone that hangs up, etc.
> >>
> >> Is this possible with the preferred syntax?
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 


-- 

Un cordial saludo / Best regards, 

 _________________________

Ant?nio Silva

E-mail:asilva at wirelessmundi.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/52c161d3/attachment-0001.html 

From sescher_ml at omeco.de  Tue Sep  6 14:03:56 2011
From: sescher_ml at omeco.de (Silvio Escher)
Date: Tue, 06 Sep 2011 12:03:56 +0200
Subject: [Freeswitch-users] Variables in Dialplan - Problem with getting
 Variables from User Directory
In-Reply-To: <4E5D50BD.8010208@omeco.de>
References: <4E5D50BD.8010208@omeco.de>
Message-ID: <4E65F00C.8060909@omeco.de>

Ok - finally i'll answer my own questions maybe this helps some others... - thanks to Michal & 
Mathieu for this nice gdb Session yesterday ;-)

Am 30.08.11 23:06, schrieb Silvio Escher:
>
> I've changed our Freeswitch Version some Weeks ago ( cannot remember exactly but proably from 
> 1.0.6 ) to the git Head. ).
> Since this Change ( or better since the followed Config File Adaptions ;-) ) i noticed that i 
> cannot use User Variables/Params inside the XML-Dialplan anymore.
thats related to "auth" - during the calls my phones arent regonized as "auth by user" atm - just by 
our ip subnet over the internal acl - thats why (as Mathieu told me) my User Vars/Params arent 
available in the DP (just with user_data or set_user)
related docs .. http://wiki.freeswitch.org/wiki/Variable_sip_auth_username or 
http://wiki.freeswitch.org/wiki/Variable_sip_authorized

i just disabled <param name="apply-inbound-acl" value="domains"/> ( commented it out ) and 
everything works fine ( btw i havent changed this parameter since months .. so maybe the default 
behavior
  ( acl auth overrule user auth ) changed in one of the "last" versions ?!

> So i've still Issues with mod_voicemail ( Mailto is undef )
>
> 2011-08-30 22:02:32.904211 [DEBUG] switch_utils.c:709 Emailed data to [(null)]
> 2011-08-30 22:02:32.904211 [DEBUG] mod_voicemail.c:2809 Sending notify message to (null)

on the one side an (now solved) bug 
(http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=8974f9d62e0346cd0763c508378e740fc0f2ce70) - 
on the other side
i havent used the vm-notify-mailto param directly (as the wiki told me that vm-notify-mailto 
defaults to vm-mailto)



-- 
Silvio Escher
omeco GmbH

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/5f48b77a/attachment.html 

From sescher_ml at omeco.de  Tue Sep  6 14:11:26 2011
From: sescher_ml at omeco.de (Silvio Escher)
Date: Tue, 06 Sep 2011 12:11:26 +0200
Subject: [Freeswitch-users] mod_fifo to Voicemail if there's no "agents"
Message-ID: <4E65F1CE.7070707@omeco.de>

Hi there,

ive following in my public dp:

<extension name="zentrale_fifo">
<condition field="destination_number" expression="^zentrale_fifo$">
<action application="set" data="fifo_music=intro_und_schleife.wav"/>
<action application="answer"/>
<action application="set" data="fifo_orbit_exten=zentrale_voicemail:30"/>
<action application="fifo" data="zentrale_fifo in"/>
</condition>
</extension>

<extension name="Zentrale">
<condition field="${sip_to_user}" expression="^0631680350$" break="on-false"/>
<condition wday="2-6" hour="9-16"> # zeit geht bis 59
<action application="transfer" data="zentrale_fifo XML public"/>
<anti-action application="answer"/>
<anti-action application="sleep" data="1500"/>
<anti-action application="playback" data="niemand_da_zeit.wav"/>
<anti-action application="sleep" data="1000"/>
<anti-action application="hangup" data="NO_ANSWER"/>
</condition>
</extension>

and iam adding Members to the fifo at runtime ( during *extension or an daily reset by cron )

./fs_cli -x "fifo_member add zentrale_fifo user/26"
or
./fs_cli -x "fifo_member del zentrale_fifo user/26"

Everything works fine beside the little issue that if theres no Member/Agent in the fifo - the Call 
is just also waiting 30 seconds till transfered to voicemail.

I noticed that with fifo list zentrale_fifo iam able to get an Memberlist ( fifo count just shows me 
0 Members - dunno why - bug ? ) - but iam unsure how to process further.

Whats the ideal Solution to get an Caller directly to the Voicemail when no Member/Agent is in the 
called Fifo ?

Best Regards,
Silvio

-- 
Silvio Escher
omeco GmbH



From avi at avimarcus.net  Tue Sep  6 17:22:46 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 6 Sep 2011 16:22:46 +0300
Subject: [Freeswitch-users] bridge after IVR delayed/no media in git-2689081
	2011-09-01 ?
Message-ID: <CALNZuwmS=rfNc9KXK_tYPiWRYQecSvfjo2Qyu_Sj7+oCbnc1QA@mail.gmail.com>

Soon after updating to GIT FreeSWITCH Version 1.0.head (git-2689081
2011-09-01 21-16-22 -0500) I started getting some complaints on my calling
card that they wouldn't hear anything. I see the calls in my CDR are marked
as the other party picking up, across several carriers.
I have a recording where after the bridge, there seems to be a long delay on
the A leg media to start to show up in the recording, and A couldn't hear B,
either (who was in the recording).
I WOULD upgrade to the latest GIT and test, but it doesn't happen all the
time. In fact, since I turned on call recording, this delay is the only
issue I've seen yet.

Did something change around that date that might have caused this?

I did several normal SIP calls (no IVR involved) and didn't have any issues
with hearing the B leg...

-Avi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/e4af8466/attachment.html 

From mays.david at gmail.com  Tue Sep  6 18:31:25 2011
From: mays.david at gmail.com (David Ma)
Date: Tue, 6 Sep 2011 22:31:25 +0800
Subject: [Freeswitch-users] Will bypass-media work when a call is
 bridged to multiple SIP destination
In-Reply-To: <CALNZuwmkLRmW_nsgb8qXpg6Oht7Vv56QBvW5QZ0rqTh_m5-jEw@mail.gmail.com>
References: <1315230976599-6760883.post@n2.nabble.com>
	<CALNZuw=gQ6nzEufq2NmRZ81P+EdsGfPjH2u2fpHDgno8D3LQxQ@mail.gmail.com>
	<CAMLBmR+14mhi5huYFVJRai0zqWwFv2A3yodphzxfxD8LVV7gCQ@mail.gmail.com>
	<CALNZuwmkLRmW_nsgb8qXpg6Oht7Vv56QBvW5QZ0rqTh_m5-jEw@mail.gmail.com>
Message-ID: <CAMLBmRKocF_zeq-C3_kDi_Z5YCqj_Kixs=0Tzjxie65XJdxOQA@mail.gmail.com>

Hello Avi,

Thanks a lot for the detailed explanation!! Such an answer is exactly what I
need.

The case can be closed.

Best regards,
David Ma

On Tue, Sep 6, 2011 at 1:20 PM, Avi Marcus <avi at avimarcus.net> wrote:

> When you use , to bridge to multiple locations, that's simul-ring.
> FreeSWITCH only connects the call to the FIRST one that picks up.
> If you want to connect more than one source and one endpoint, you will need
> to use a conference to mix the media locally, and you can't bypass_media on
> that...
>
> -Avi
>
> On Tue, Sep 6, 2011 at 5:01 AM, David Ma <mays.david at gmail.com> wrote:
>
>>
>> Hi Avi,
>>
>> Thanks for your prompt response. My objective is to bridge the incoming
>> call to multiple contacts with a single "bridge" so that 3 or more parties
>> are involved in the call. I know that FreeSwitch support such a bridge like
>>
>> <action application="bridge" data="sofia/internal/
>> 1010 at sip.yourprovider.com,sofia/sip/1011 at sip.yourprovider.com"/>
>>
>> I just wish bypass-media works in such a scenario so that the RTP packets
>> (media) doesn't flow through my FreeSwitch box when 3 or more parties are
>> talking.
>>
>> Thanks a lot!
>> D.Ma
>>
>>
>> On Mon, Sep 5, 2011 at 10:05 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> You mean you fork the dialing so it fails over or tries different places
>>> at once?
>>> Sure. You can even set the bypass_media variable differently on each leg.
>>>
>>> -Avi
>>>
>>>
>>> On Mon, Sep 5, 2011 at 4:56 PM, dma <mays.david at gmail.com> wrote:
>>>
>>>> Hi Support,
>>>>
>>>> This is a simple question but I don't find an answer from the web.
>>>>
>>>> I hope to know whether bypass-media work when a call is bridged to
>>>> multiple
>>>> SIP destination?
>>>>
>>>> Thanks,
>>>> D.Ma
>>>>
>>>> --
>>>> View this message in context:
>>>> http://freeswitch-users.2379917.n2.nabble.com/Will-bypass-media-work-when-a-call-is-bridged-to-multiple-SIP-destination-tp6760883p6760883.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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/280d1dfb/attachment.html 

From yungwei at resolvity.com  Tue Sep  6 18:46:53 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Tue, 6 Sep 2011 10:46:53 -0400
Subject: [Freeswitch-users] Is there a SIP client that is capable of
 displaying arbitrary SIP headers?
Message-ID: <33095823FD21DF429B481B5163264B79511882D287@VMBX102.ihostexchange.net>

Hi,

I'm wondering if there's a SIP client that can display arbibrary SIP headers on its UI. Thanks.


From potxoka at gmail.com  Tue Sep  6 18:57:00 2011
From: potxoka at gmail.com (Anto)
Date: Tue, 6 Sep 2011 16:57:00 +0200
Subject: [Freeswitch-users] External/internal profile
In-Reply-To: <CAGRr05-5F_B990V9MMFb4sQZkp2-c9R+Au1GWrP8Lyq2mvh+Cw@mail.gmail.com>
References: <CA+Ecp7Tgn6ObPkzNzQLQi418uQd5OJFTTX1B5cG6OF7cPnSY-g@mail.gmail.com>
	<CAGRr05-5F_B990V9MMFb4sQZkp2-c9R+Au1GWrP8Lyq2mvh+Cw@mail.gmail.com>
Message-ID: <CA+Ecp7R3GyOiiL3F_Qv4C1viMKnO2QyaNr4w_etXa3XE8V0oPg@mail.gmail.com>

Thanks very much :-)

Regards

2011/9/2 Brad Mina <brad at tech21.com>:
> All profile listen on a single IP.
>
> Default external profile port is 5080
> Default internal profile port is 5060
> Yes you can configure two different IP addresses to use the same port.
> On Fri, Sep 2, 2011 at 2:44 PM, Anto <potxoka at gmail.com> wrote:
>>
>> Hello
>>
>> Today I can not understand the internal and external profiles. I've
>> used other times asterisk setup and both the carriers and customers
>> share the same ports and same ips. I do not quite understand because
>> FreeSWITCH must use different ports (do not know if you can configure
>> the same ports. In all the how-to I've seen that change the profiles
>> and use different ports). Could anyone guide me for this?.
>>
>> If for example I have a server with two public IPs, you could
>> configure each profile with an ip and both with the same ports (5060
>> and 5061)?. Perhaps it is an obvious question, but do not quite
>> understand the issue of profiling, I have doubts. Thank you very much.
>>
>> Best 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
>
>


From vikram.agrawal at gmail.com  Tue Sep  6 16:20:04 2011
From: vikram.agrawal at gmail.com (vikram)
Date: Tue, 6 Sep 2011 05:20:04 -0700 (PDT)
Subject: [Freeswitch-users] Can't make mod_rtmp
In-Reply-To: <4E6587FA.2040307@livecall.com>
References: <1308334343913-6488229.post@n2.nabble.com>
	<BANLkTimx1MnAhfhHVmVds3qYpqNpaC9JKQ@mail.gmail.com>
	<1308388959641-6490318.post@n2.nabble.com>
	<1315213321297-6760239.post@n2.nabble.com>
	<1315230432566-6760858.post@n2.nabble.com>
	<4E6587FA.2040307@livecall.com>
Message-ID: <CAOrstdPsLoT2oQu7Zwj94HE8zZg1e3DzrUc-LLMUJcidjPXLdA@mail.gmail.com>

mod_rtmp is building by default but it is not loaded when I start
freeswitch.

However if I type "load mod_rtmp" in freeswitch, it loads the module.

On Tue, Sep 6, 2011 at 8:12 AM, Jack [via freeswitch-users] <
ml-node+6762531-1355026059-354831 at n2.nabble.com> wrote:

> There is also a rtmp.conf.xml in the autoload_configs folder.
> It sounds like you need to include the rtmp source in your build.
> src\mod\endpoints\mod_rtmp\mod_rtmp.c
> I do not think it builds by default.
> Jack
>
> On 9/5/2011 6:47 AM, vikram wrote:
>
> > I am facing the same issue.  I have made a fresh installation with
> mod_rtmp.
> >
> > unknown command: rtmp
> >
> >
> > kkocyk wrote:
> >>
> >> peely wrote:
> >>> I'm not sure if I need to do anything else, I just naively added
> >>> endpoints/mod_rtmp to modules.conf and did a make clean install?
> >>>
> >> I'm trying to make it same way (my first install was without mod_rtmp),
> >> but after starting FS it seams that I still don't have rtmp. When I
> enter
> >> "rtmp status" it says that there is no such command.
> >>
> >
> > --
> > View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6760858.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> >
> > FreeSWITCH-users mailing list
> > [hidden email] <http://user/SendEmail.jtp?type=node&node=6762531&i=0>
> > http://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=6762531&i=1>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
>  If you reply to this email, your message will be added to the discussion
> below:
>
> http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6762531.html
>  To unsubscribe from Can't make mod_rtmp, click here<http://freeswitch-users.2379917.n2.nabble.com/template/NamlServlet.jtp?macro=unsubscribe_by_code&node=6488229&code=dmlrcmFtLmFncmF3YWxAZ21haWwuY29tfDY0ODgyMjl8LTEyMjk2MjEwNg==>.
>
>



-- 
Vikram Agrawal
Director - Samuday Web Technologies


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6763678.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/7e1c6bc3/attachment.html 

From simon0922 at gmail.com  Tue Sep  6 20:42:18 2011
From: simon0922 at gmail.com (Simon Leck)
Date: Wed, 7 Sep 2011 00:42:18 +0800
Subject: [Freeswitch-users] - freeswitch bye issue
Message-ID: <009001cc6cb3$f24b51f0$d6e1f5d0$@gmail.com>

Hi Everyone, kindly help if you can 

 

At the moment I did encounter a issue, A leg user called a b leg user and
when b leg user hanged up (Bye is not send by freeswitch to the A leg user)
the only way I could hanged up this call  to hanged up the A leg user
manually? Would be great if someone could enlighten me on how this can be
achieved?

 

Thanks in advanced to anyone for helping me out

 

Thanks again

Simon Leck

 

 

 

 

 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/8ab8bf20/attachment.html 

From brian.wiese.freeswitch at gmail.com  Tue Sep  6 20:59:47 2011
From: brian.wiese.freeswitch at gmail.com (Brian Wiese)
Date: Tue, 6 Sep 2011 11:59:47 -0500
Subject: [Freeswitch-users] Multiple DTMF on Single Keypress
In-Reply-To: <CA+Gmxbbm46vCtW70+Qt7LurJEybMHwDA6BA+tNa8OnL4U0uKfw@mail.gmail.com>
References: <CADEQS5zp8bh3epB3-ya9uR2HvNjnt8qf7u4rS=a11tNNGGo7hA@mail.gmail.com>
	<CA+Gmxbbm46vCtW70+Qt7LurJEybMHwDA6BA+tNa8OnL4U0uKfw@mail.gmail.com>
Message-ID: <CADEQS5z9wV90SVYOwy30o3YA4vLqp3a6yN3MYeVj_eGM8bHHiA@mail.gmail.com>

Dmitry:

I actually thought it was my provider because the call is coming from the
PSTN to them...  but I ruled that out when I was able to route the call to
Asterisk and don't experience the problem.

~Brian

On Tue, Sep 6, 2011 at 12:36 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> See the same behaviour with inband DTMF detector sometimes.
>
>   2011/9/5 Brian Wiese FreeSWITCH List <brian.wiese.freeswitch at gmail.com>
>
>>   Hello everyone!
>>
>> I'm getting multiple RTP DTMF from random keypresses and I can't figure
>> out why.  I've PB'ed the packet capture and FS log for a call.  As you can
>> see from the FS log, the 9,8,7,6 numbers weren't repeated, but 5 was (again,
>> though, different calls lead to different numbers being repeated).
>> Log:  http://pastebin.freeswitch.org/17280
>> Capture:  http://pastebin.freeswitch.org/17282
>>
>> I appreciate any ideas as to what I might have wrong here.
>>
>> 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
>>
>>
>
>
> --
> 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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/b633fd66/attachment.html 

From anthony.minessale at gmail.com  Tue Sep  6 21:11:07 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 6 Sep 2011 12:11:07 -0500
Subject: [Freeswitch-users] bridge after IVR delayed/no media in
 git-2689081 2011-09-01 ?
In-Reply-To: <CALNZuwmS=rfNc9KXK_tYPiWRYQecSvfjo2Qyu_Sj7+oCbnc1QA@mail.gmail.com>
References: <CALNZuwmS=rfNc9KXK_tYPiWRYQecSvfjo2Qyu_Sj7+oCbnc1QA@mail.gmail.com>
Message-ID: <CAKbxfG9b44q9dT3TAPM72v-1KxZQ=yf1SPhBuMrqPAhdtPCjQg@mail.gmail.com>

> Did something change around that date that might have caused this?
> I did several normal SIP calls (no IVR involved) and didn't have any issues
> with hearing the B leg...
> -Avi

That would depend on what revision you were on before that.
You would  need a little more evidence than that to confirm but you
may want to try HEAD again for good measure.

There was one issue on sept 2 that *could* be your fix.



-- 
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 avimarcus.net  Tue Sep  6 21:17:52 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 6 Sep 2011 20:17:52 +0300
Subject: [Freeswitch-users] bridge after IVR delayed/no media in
 git-2689081 2011-09-01 ?
In-Reply-To: <CAKbxfG9b44q9dT3TAPM72v-1KxZQ=yf1SPhBuMrqPAhdtPCjQg@mail.gmail.com>
References: <CALNZuwmS=rfNc9KXK_tYPiWRYQecSvfjo2Qyu_Sj7+oCbnc1QA@mail.gmail.com>
	<CAKbxfG9b44q9dT3TAPM72v-1KxZQ=yf1SPhBuMrqPAhdtPCjQg@mail.gmail.com>
Message-ID: <CALNZuwnSFD+z5RbqQdZTrThUnpF5hjYs29uxyO51WDfr50Ak5Q@mail.gmail.com>

Right, so I tried HEAD and then I recorded 2x calls where the
bridge occurred, and I lost A leg audio (who was interacting with the IVR)
and B leg was recorded, and he couldn't hear A either.

It's not consistent, it doesn't happen all the time or I'd have a nice error
log / recordings for you.

-Avi

On Tue, Sep 6, 2011 at 8:11 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> > Did something change around that date that might have caused this?
> > I did several normal SIP calls (no IVR involved) and didn't have any
> issues
> > with hearing the B leg...
> > -Avi
>
> That would depend on what revision you were on before that.
> You would  need a little more evidence than that to confirm but you
> may want to try HEAD again for good measure.
>
> There was one issue on sept 2 that *could* be your fix.
>
>
>
> --
> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/051daa5a/attachment-0001.html 

From anthony.minessale at gmail.com  Tue Sep  6 21:25:24 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 6 Sep 2011 12:25:24 -0500
Subject: [Freeswitch-users] bridge after IVR delayed/no media in
 git-2689081 2011-09-01 ?
In-Reply-To: <CALNZuwnSFD+z5RbqQdZTrThUnpF5hjYs29uxyO51WDfr50Ak5Q@mail.gmail.com>
References: <CALNZuwmS=rfNc9KXK_tYPiWRYQecSvfjo2Qyu_Sj7+oCbnc1QA@mail.gmail.com>
	<CAKbxfG9b44q9dT3TAPM72v-1KxZQ=yf1SPhBuMrqPAhdtPCjQg@mail.gmail.com>
	<CALNZuwnSFD+z5RbqQdZTrThUnpF5hjYs29uxyO51WDfr50Ak5Q@mail.gmail.com>
Message-ID: <CAKbxfG8Lmv0JM_e2f6qyguftHdn8XZ_bzrCjJMOz8uuNYs5tZQ@mail.gmail.com>

can you at least supply *some* details

Every time you say HEAD please include revision number.
Describe the callflow, if you use any execute_on_* vars or other dp features...

And still, what was your old version...


On Tue, Sep 6, 2011 at 12:17 PM, Avi Marcus <avi at avimarcus.net> wrote:
> Right, so I tried HEAD and then I recorded 2x calls where the
> bridge?occurred, and I lost A leg audio (who was interacting with the IVR)
> and B leg was recorded, and he couldn't hear A either.
> It's not consistent, it doesn't happen all the time or I'd have a nice error
> log / recordings for you.
> -Avi
> On Tue, Sep 6, 2011 at 8:11 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> > Did something change around that date that might have caused this?
>> > I did several normal SIP calls (no IVR involved) and didn't have any
>> > issues
>> > with hearing the B leg...
>> > -Avi
>>
>> That would depend on what revision you were on before that.
>> You would ?need a little more evidence than that to confirm but you
>> may want to try HEAD again for good measure.
>>
>> There was one issue on sept 2 that *could* be your fix.
>>
>>
>>
>> --
>> 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 peder at networkoblivion.com  Tue Sep  6 21:57:03 2011
From: peder at networkoblivion.com (Peder)
Date: Tue, 6 Sep 2011 12:57:03 -0500
Subject: [Freeswitch-users] - freeswitch bye issue
In-Reply-To: <009001cc6cb3$f24b51f0$d6e1f5d0$@gmail.com>
References: <009001cc6cb3$f24b51f0$d6e1f5d0$@gmail.com>
Message-ID: <0aae01cc6cbe$633ab680$29b02380$@com>

It should happen automatically.  Is A behind NAT?  Sounds like a
firewall/NAT may be dropping the bye.  Also, have you done a debug on FS to
see if it actually sends a bye or not to A?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Simon
Leck
Sent: Tuesday, September 06, 2011 11:42 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] - freeswitch bye issue

 

Hi Everyone, kindly help if you can 

 

At the moment I did encounter a issue, A leg user called a b leg user and
when b leg user hanged up (Bye is not send by freeswitch to the A leg user)
the only way I could hanged up this call  to hanged up the A leg user
manually? Would be great if someone could enlighten me on how this can be
achieved?

 

Thanks in advanced to anyone for helping me out

 

Thanks again

Simon Leck

 

 

 

 

 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/70cc2c54/attachment.html 

From jonyoung111 at gmail.com  Wed Sep  7 00:36:04 2011
From: jonyoung111 at gmail.com (Jon Young)
Date: Tue, 6 Sep 2011 13:36:04 -0700
Subject: [Freeswitch-users] Multiple DTMF on Single Keypress
In-Reply-To: <CA+Gmxbbm46vCtW70+Qt7LurJEybMHwDA6BA+tNa8OnL4U0uKfw@mail.gmail.com>
References: <CADEQS5zp8bh3epB3-ya9uR2HvNjnt8qf7u4rS=a11tNNGGo7hA@mail.gmail.com>
	<CA+Gmxbbm46vCtW70+Qt7LurJEybMHwDA6BA+tNa8OnL4U0uKfw@mail.gmail.com>
Message-ID: <CAFKNP3-Z6Zhas8DZobHqw4fxampyYjwqMU=guzG1jvZvHuWWFA@mail.gmail.com>

Is it possible you are receiving 2833 and Inband DTMF?

On Mon, Sep 5, 2011 at 10:36 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> See the same behaviour with inband DTMF detector sometimes.
>
> 2011/9/5 Brian Wiese FreeSWITCH List <brian.wiese.freeswitch at gmail.com>
>>
>> Hello everyone!
>>
>> I'm getting multiple RTP DTMF from random keypresses and I can't figure
>> out why.? I've PB'ed the packet capture and FS log for a call.? As you can
>> see from the FS log, the 9,8,7,6 numbers weren't repeated, but 5 was (again,
>> though, different calls lead to different numbers being repeated).
>> Log:? http://pastebin.freeswitch.org/17280
>> Capture:? http://pastebin.freeswitch.org/17282
>>
>> I appreciate any ideas as to what I might have wrong here.
>>
>> 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
>>
>
>
>
> --
> 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
>
>


From avi at avimarcus.net  Wed Sep  7 00:48:19 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 6 Sep 2011 23:48:19 +0300
Subject: [Freeswitch-users] Multiple DTMF on Single Keypress
In-Reply-To: <CAFKNP3-Z6Zhas8DZobHqw4fxampyYjwqMU=guzG1jvZvHuWWFA@mail.gmail.com>
References: <CADEQS5zp8bh3epB3-ya9uR2HvNjnt8qf7u4rS=a11tNNGGo7hA@mail.gmail.com>
	<CA+Gmxbbm46vCtW70+Qt7LurJEybMHwDA6BA+tNa8OnL4U0uKfw@mail.gmail.com>
	<CAFKNP3-Z6Zhas8DZobHqw4fxampyYjwqMU=guzG1jvZvHuWWFA@mail.gmail.com>
Message-ID: <CALNZuwk372OUr1cNdqGrPqmBdFPEf6EUSBuPzhqBmBu1Bczvxg@mail.gmail.com>

line 389
2011-09-04?16:20:14.390350?[DEBUG]?switch_rtp.c:3317?RTP RECV DTMF?5:1440
2011-09-04?16:20:14.390350?[DEBUG]?switch_ivr_bridge.c:391?Send signal
sofia/internal/sip:20005 at 172.31.6.253?[BREAK]
2011-09-04?16:20:14.410350?[DEBUG]?switch_rtp.c:2343?Send start packet
for?[5]?ts=97600?dur=160/160/1440?seq=29252
2011-09-04?16:20:14.410350?[DEBUG]?switch_rtp.c:3317?RTP RECV DTMF?5:1440

It's coming as RTP RECV, meaning rfc2833. Otherwise, it would say "DETECTED".

-Avi


On Tue, Sep 6, 2011 at 11:36 PM, Jon Young <jonyoung111 at gmail.com> wrote:
>
> Is it possible you are receiving 2833 and Inband DTMF?
>
> On Mon, Sep 5, 2011 at 10:36 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> > See the same behaviour with inband DTMF detector sometimes.
> >
> > 2011/9/5 Brian Wiese FreeSWITCH List <brian.wiese.freeswitch at gmail.com>
> >>
> >> Hello everyone!
> >>
> >> I'm getting multiple RTP DTMF from random keypresses and I can't figure
> >> out why.? I've PB'ed the packet capture and FS log for a call.? As you can
> >> see from the FS log, the 9,8,7,6 numbers weren't repeated, but 5 was (again,
> >> though, different calls lead to different numbers being repeated).
> >> Log:? http://pastebin.freeswitch.org/17280
> >> Capture:? http://pastebin.freeswitch.org/17282
> >>
> >> I appreciate any ideas as to what I might have wrong here.
> >>
> >> 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
> >>
> >
> >
> >
> > --
> > 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


From avi at avimarcus.net  Wed Sep  7 02:18:42 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 7 Sep 2011 01:18:42 +0300
Subject: [Freeswitch-users] Opposite of flush_dtmf - grab dtmf from
	before the IVR + debug rfc2833
In-Reply-To: <CALNZuwmvEvmzOYCjq7tumZWJ2bznZWU1zQhdrYqbGYTe8TdMUA@mail.gmail.com>
References: <CALNZuwmvEvmzOYCjq7tumZWJ2bznZWU1zQhdrYqbGYTe8TdMUA@mail.gmail.com>
Message-ID: <CALNZuwnDyhkg8vfHz86DBgP1t=+UhK0WVkrwPnWhU5S3e+iBDg@mail.gmail.com>

Thanks to Anthony's help in the channel...  I'm not sure where to put
this on the wiki, but thought I should at least report back.
You don't need to SAVE The dtmf digits, you just need to make sure
they don't get eaten.
If you use lua to do session:execute('playback',file) or 'say', it
eats the dtmf digits. If you do session:streamfile or session:say it
won't eat the digits and they will be queued up for whatever is next.
execute('log',stuff) won't eat any digits.

-Avi



On Tue, Sep 6, 2011 at 1:08 AM, Avi Marcus <avi at avimarcus.net> wrote:
> I have a calling card IVR and upon an error, it announces the error, then
> goes back to the calling card IVR. How can I use the digits they started
> dialing during the pre-IVR announcement as part of the IVR?
> e.g.?http://pastebin.freeswitch.org/17283
> During the announcement of the "wrong" dialed number, I started dialing and
> FS got the digits. But it didn't start "listening" until... just before (??)
> the calling_card_short IVR started.
> How can I get it to listen to everything that got queued up?
> Thanks,
> -Avi
> p.s. is there somelinke like ngrep/tcpdump on regular SIP packets to
> live-view the rfc-2833 that come in, before they even hit FS? I know
> wireshark can understand them in a pcap, but I couldn't figure out how to
> filter live...


From lfurrea at gmail.com  Wed Sep  7 05:43:39 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Tue, 6 Sep 2011 19:43:39 -0600
Subject: [Freeswitch-users] Caller-Privacy-Hide-Name: [true]
In-Reply-To: <CAB2KLyd05_n6219mUNxXzzRgoXgSaq+GSZw0TYZAYkxMogJh3w@mail.gmail.com>
References: <CAB2KLyd05_n6219mUNxXzzRgoXgSaq+GSZw0TYZAYkxMogJh3w@mail.gmail.com>
Message-ID: <CAB2KLyf0S5Mg1qpzExfS-OG5zxoT4p8T1cs9apB8CqnrQxnJtw@mail.gmail.com>

Just for reference in case google ever yields this thread for a desperate
soul.

Resetting the phone to factory defaults fixed the issue!

On Fri, Sep 2, 2011 at 12:32 PM, Luis F Urrea <lfurrea at gmail.com> wrote:

> Hi all,
>
> Any one can shed some light on why this channel variable
>
> Caller-Privacy-Hide-Name: [true]
>
> suddenly got set to true for a single phone ?
>
> Is this something that can be set by the phone through the SIP INVITE? or
> maybe in the directory?
>
> This got turned on on a Snom 370 which is the receptionist phone and
> threfore all calls from this phone appear as anonymous.
>
> Nothing was changed on the dial plan or even directory.
>
> This is a way old build of FS just FYI.
>
> Your help is greatly appreciated.
>
> Regards,
>
> Luis
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/36554895/attachment-0001.html 

From simon0922 at gmail.com  Wed Sep  7 06:46:59 2011
From: simon0922 at gmail.com (Simon Leck)
Date: Wed, 7 Sep 2011 10:46:59 +0800
Subject: [Freeswitch-users] - freeswitch bye issue
In-Reply-To: <0aae01cc6cbe$633ab680$29b02380$@com>
References: <009001cc6cb3$f24b51f0$d6e1f5d0$@gmail.com>
	<0aae01cc6cbe$633ab680$29b02380$@com>
Message-ID: <011c01cc6d08$6badcce0$430966a0$@gmail.com>

Hi Pedar, 

 

Thanks for your reply. At first I thought it was NAT issue so to isolate the
problem I used a Public IP but I still get the same result? I have used
wireshark to do a trace but I don't see FS sending a bye to A?

 

Kindly advice me on how else could I solved this issue. On the NAT Part, I
have enable RTP keep alive as well. Under RTCP, I have the keep alive turned
on as well?

 

Which other place are there setting I can configure with?

 

Thanks in advanced Pedar for your kindest help

 

Many thanks

Simon Leck

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peder
Sent: Wednesday, September 07, 2011 1:57 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] - freeswitch bye issue

 

It should happen automatically.  Is A behind NAT?  Sounds like a
firewall/NAT may be dropping the bye.  Also, have you done a debug on FS to
see if it actually sends a bye or not to A?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Simon
Leck
Sent: Tuesday, September 06, 2011 11:42 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] - freeswitch bye issue

 

Hi Everyone, kindly help if you can 

 

At the moment I did encounter a issue, A leg user called a b leg user and
when b leg user hanged up (Bye is not send by freeswitch to the A leg user)
the only way I could hanged up this call  to hanged up the A leg user
manually? Would be great if someone could enlighten me on how this can be
achieved?

 

Thanks in advanced to anyone for helping me out

 

Thanks again

Simon Leck

 

 

 

 

 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/492ab309/attachment.html 

From joe.jflemmings at gmail.com  Wed Sep  7 07:21:59 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Tue, 6 Sep 2011 20:21:59 -0700
Subject: [Freeswitch-users] Sophia authentication without using editing
	acl.conf.xml
Message-ID: <CALH5ZrZPtg+3iaR+=s+oNaumo=wD27MQn7pY4UyJEQ-5t0QiUA@mail.gmail.com>

I use xml_curl to authenticate sip devices and was wondering if there is a
way to do IP authentication without having to edit and reaload acl.conf.xml
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/a9e59b9a/attachment.html 

From john_platts at hotmail.com  Wed Sep  7 08:01:19 2011
From: john_platts at hotmail.com (John Platts)
Date: Tue, 6 Sep 2011 23:01:19 -0500
Subject: [Freeswitch-users] One-way audio issue with attended transfer
Message-ID: <SNT116-W212CCAFBD521CA3F341979D1F0@phx.gbl>


We have been experiencing a one-way audio issue with attended transfer. The problem is that after an call is transferred through an attended transfer, the caller cannot be heard, but the caller can hear the voice of the person that the call is transferred to.

Steps to reproduce the problem:
1. Dial into FreeSWITCH from an external phone.
2. Answer the call. One-way audio is not experienced here.
3. Press the Transfer button on the IP phone to initiate an attended transfer.
4. Dial the extension that the call is to be transferred to.
5. Allow the extension that the call is to be transferred to pick up the call. One-way audio is not experienced here.
6. Press the Transfer button to complete the transfer. The caller does hear audio from the person that the call is transferred to, but the person that the call is transferred to cannot hear the caller. The one way audio issue is occurring here.

The attended transfer was performed from an extension on one SPA303 phone to an extension on another SPA303 phone. FreeSWITCH actually detects that the transfer is an attended transfer.

Here are attachments that I have uploaded to pastebin:
conf/dialplan/features.xml file: http://pastebin.freeswitch.org/17291
FreeSWITCH log, including the attended transfer: http://pastebin.freeswitch.org/17286

How do we get this one-way audio issue fixed?
 		 	   		  

From brad at tech21.com  Wed Sep  7 09:04:08 2011
From: brad at tech21.com (Brad Mina)
Date: Tue, 6 Sep 2011 22:04:08 -0700
Subject: [Freeswitch-users] Sophia authentication without using editing
	acl.conf.xml
In-Reply-To: <CALH5ZrZPtg+3iaR+=s+oNaumo=wD27MQn7pY4UyJEQ-5t0QiUA@mail.gmail.com>
References: <CALH5ZrZPtg+3iaR+=s+oNaumo=wD27MQn7pY4UyJEQ-5t0QiUA@mail.gmail.com>
Message-ID: <B5F566F1-4244-475E-94D6-52963B5DCC05@tech21.com>

I believe you can add an acl param in a user's XML.

Sent from my iPhone

On Sep 6, 2011, at 8:21 PM, Joe Flemmings <joe.jflemmings at gmail.com>  
wrote:

>
> I use xml_curl to authenticate sip devices and was wondering if  
> there is a way to do IP authentication without having to edit and  
> reaload acl.conf.xml
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/f0754baf/attachment.html 

From joe.jflemmings at gmail.com  Wed Sep  7 09:20:24 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Tue, 6 Sep 2011 22:20:24 -0700
Subject: [Freeswitch-users] Sophia authentication without using editing
	acl.conf.xml
In-Reply-To: <B5F566F1-4244-475E-94D6-52963B5DCC05@tech21.com>
References: <CALH5ZrZPtg+3iaR+=s+oNaumo=wD27MQn7pY4UyJEQ-5t0QiUA@mail.gmail.com>
	<B5F566F1-4244-475E-94D6-52963B5DCC05@tech21.com>
Message-ID: <CALH5ZrbVN5hZ6DTo0PHM0NPMBSv8JtObrXoF7fZjC4KeOW1zWg@mail.gmail.com>

I tried that but it seams the acl has to be already defined in acl.conf.xml

On Tue, Sep 6, 2011 at 10:04 PM, Brad Mina <brad at tech21.com> wrote:

> I believe you can add an acl param in a user's XML.
>
> Sent from my iPhone
>
> On Sep 6, 2011, at 8:21 PM, Joe Flemmings <joe.jflemmings at gmail.com>
> wrote:
>
>
> I use xml_curl to authenticate sip devices and was wondering if there is a
> way to do IP authentication without having to edit and reaload acl.conf.xml
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/c7b82f94/attachment.html 

From brian.wiese.freeswitch at gmail.com  Wed Sep  7 03:13:35 2011
From: brian.wiese.freeswitch at gmail.com (Brian Wiese)
Date: Tue, 6 Sep 2011 18:13:35 -0500
Subject: [Freeswitch-users] Multiple DTMF on Single Keypress
In-Reply-To: <CALNZuwk372OUr1cNdqGrPqmBdFPEf6EUSBuPzhqBmBu1Bczvxg@mail.gmail.com>
References: <CADEQS5zp8bh3epB3-ya9uR2HvNjnt8qf7u4rS=a11tNNGGo7hA@mail.gmail.com>
	<CA+Gmxbbm46vCtW70+Qt7LurJEybMHwDA6BA+tNa8OnL4U0uKfw@mail.gmail.com>
	<CAFKNP3-Z6Zhas8DZobHqw4fxampyYjwqMU=guzG1jvZvHuWWFA@mail.gmail.com>
	<CALNZuwk372OUr1cNdqGrPqmBdFPEf6EUSBuPzhqBmBu1Bczvxg@mail.gmail.com>
Message-ID: <CADEQS5xsRouLa1fxGu9+=sUJdagO4cW6q0AfsoPKwDghKMvSCw@mail.gmail.com>

Avi:

I had thought it was inband, but I couldn't find anything that supported it,
like you mentioned.

Is there anything else I can provide that would help solve this problem?

Thanks.

~Brian

On Tue, Sep 6, 2011 at 3:48 PM, Avi Marcus <avi at avimarcus.net> wrote:

> line 389
> 2011-09-04 16:20:14.390350 [DEBUG] switch_rtp.c:3317 RTP RECV DTMF 5:1440
> 2011-09-04 16:20:14.390350 [DEBUG] switch_ivr_bridge.c:391 Send signal
> sofia/internal/sip:20005 at 172.31.6.253 [BREAK]
> 2011-09-04 16:20:14.410350 [DEBUG] switch_rtp.c:2343 Send start packet
> for [5] ts=97600 dur=160/160/1440 seq=29252
> 2011-09-04 16:20:14.410350 [DEBUG] switch_rtp.c:3317 RTP RECV DTMF 5:1440
>
> It's coming as RTP RECV, meaning rfc2833. Otherwise, it would say
> "DETECTED".
>
> -Avi
>
>
> On Tue, Sep 6, 2011 at 11:36 PM, Jon Young <jonyoung111 at gmail.com> wrote:
> >
> > Is it possible you are receiving 2833 and Inband DTMF?
> >
> > On Mon, Sep 5, 2011 at 10:36 PM, Dmitry Sytchev <kbdfck at gmail.com>
> wrote:
> > > See the same behaviour with inband DTMF detector sometimes.
> > >
> > > 2011/9/5 Brian Wiese FreeSWITCH List <brian.wiese.freeswitch at gmail.com
> >
> > >>
> > >> Hello everyone!
> > >>
> > >> I'm getting multiple RTP DTMF from random keypresses and I can't
> figure
> > >> out why.  I've PB'ed the packet capture and FS log for a call.  As you
> can
> > >> see from the FS log, the 9,8,7,6 numbers weren't repeated, but 5 was
> (again,
> > >> though, different calls lead to different numbers being repeated).
> > >> Log:  http://pastebin.freeswitch.org/17280
> > >> Capture:  http://pastebin.freeswitch.org/17282
> > >>
> > >> I appreciate any ideas as to what I might have wrong here.
> > >>
> > >> 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
> > >>
> > >
> > >
> > >
> > > --
> > > 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
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110906/2b2846c5/attachment-0001.html 

From jcgpoza at gmail.com  Wed Sep  7 12:10:14 2011
From: jcgpoza at gmail.com (jcgpoza gonzalez)
Date: Wed, 7 Sep 2011 10:10:14 +0200
Subject: [Freeswitch-users] Is there a SIP client that is capable of
 displaying arbitrary SIP headers?
In-Reply-To: <33095823FD21DF429B481B5163264B79511882D287@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B79511882D287@VMBX102.ihostexchange.net>
Message-ID: <CAMDVdeitxqqT-JXTsS=dJYxgW81jDqX1pPQvEta=XoGD+F9=vQ@mail.gmail.com>

Why don't you use SIPp?

http://sipp.sourceforge.net/

really useful tool.

2011/9/6 Yungwei Chen <yungwei at resolvity.com>

> Hi,
>
> I'm wondering if there's a SIP client that can display arbibrary SIP
> headers on its UI. Thanks.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/21f707e7/attachment.html 

From vikram.agrawal at gmail.com  Wed Sep  7 14:38:38 2011
From: vikram.agrawal at gmail.com (vikram)
Date: Wed, 7 Sep 2011 03:38:38 -0700 (PDT)
Subject: [Freeswitch-users] Can't make mod_rtmp
In-Reply-To: <1315391201025-6767247.post@n2.nabble.com>
References: <1308334343913-6488229.post@n2.nabble.com>
	<BANLkTimx1MnAhfhHVmVds3qYpqNpaC9JKQ@mail.gmail.com>
	<1308388959641-6490318.post@n2.nabble.com>
	<1315213321297-6760239.post@n2.nabble.com>
	<1315230432566-6760858.post@n2.nabble.com>
	<4E6587FA.2040307@livecall.com>
	<CAOrstdPsLoT2oQu7Zwj94HE8zZg1e3DzrUc-LLMUJcidjPXLdA@mail.gmail.com>
	<1315391201025-6767247.post@n2.nabble.com>
Message-ID: <CAOrstdOygDF_xWs57EY6iMafe-Sv_G-PyrG1G9yXXDdBGJ9mnw@mail.gmail.com>

In that case, there are issues in building of mod_rtmp.

when you do "make" , do you see any error in building mod_rtmp?  ideally it
should compile the rtmp code and create mod_rtmp.la

On Wed, Sep 7, 2011 at 3:56 PM, kkocyk [via freeswitch-users] <
ml-node+6767247-498842209-354831 at n2.nabble.com> wrote:

> When I try to load module there is error:
>
> "-ERR [module load file routine returned an error]
>
> 2011-09-07 12:25:09.101871 [CRIT] switch_loadable_module.c:969 Error
> Loading module /usr/local/freeswitch/mod/mod_rtmp.so
> **/usr/local/freeswitch/mod/mod_rtmp.so: cannot open shared object file: No
> such file or directory**"
>
> vikram wrote:
> mod_rtmp is building by default but it is not loaded when I start
> freeswitch.
>
> However if I type "load mod_rtmp" in freeswitch, it loads the module.
>
> On Tue, Sep 6, 2011 at 8:12 AM, Jack [via freeswitch-users] <
> [hidden email] <http://user/SendEmail.jtp?type=node&node=6767247&i=0>>
> wrote:
>
> > There is also a rtmp.conf.xml in the autoload_configs folder.
> > It sounds like you need to include the rtmp source in your build.
> > src\mod\endpoints\mod_rtmp\mod_rtmp.c
> > I do not think it builds by default.
> > Jack
> >
> > On 9/5/2011 6:47 AM, vikram wrote:
> >
> > > I am facing the same issue.  I have made a fresh installation with
> > mod_rtmp.
> > >
> > > unknown command: rtmp
> > >
> > >
> > > kkocyk wrote:
> > >>
> > >> peely wrote:
> > >>> I'm not sure if I need to do anything else, I just naively added
> > >>> endpoints/mod_rtmp to modules.conf and did a make clean install?
> > >>>
> > >> I'm trying to make it same way (my first install was without
> mod_rtmp),
> > >> but after starting FS it seams that I still don't have rtmp. When I
> > enter
> > >> "rtmp status" it says that there is no such command.
> > >>
> > >
> > > --
> > > View this message in context:
> >
> http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6760858.html
> > > Sent from the freeswitch-users mailing list archive at Nabble.com.
> > >
> > >
> > > FreeSWITCH-users mailing list
> > > [hidden email] <http://user/SendEmail.jtp?type=node&node=6762531&i=0>
>
> > > http://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=6762531&i=1>
>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > ------------------------------
> >  If you reply to this email, your message will be added to the discussion
>
> > below:
> >
> >
> http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6762531.html
> >  To unsubscribe from Can't make mod_rtmp, click here<
> http://freeswitch-users.2379917.n2.nabble.com/template/NamlServlet.jtp?macro=unsubscribe_by_code&node=6488229&code=dmlrcmFtLmFncmF3YWxAZ21haWwuY29tfDY0ODgyMjl8LTEyMjk2MjEwNg==>.
>
> >
> >
>
>
>
> --
> Vikram Agrawal
> Director - Samuday Web Technologies
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6763678.html
>
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> FreeSWITCH-users mailing list
> [hidden email] <http://user/SendEmail.jtp?type=node&node=6767247&i=1>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
>  If you reply to this email, your message will be added to the discussion
> below:
>
> http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6767247.html
>  To unsubscribe from Can't make mod_rtmp, click here<http://freeswitch-users.2379917.n2.nabble.com/template/NamlServlet.jtp?macro=unsubscribe_by_code&node=6488229&code=dmlrcmFtLmFncmF3YWxAZ21haWwuY29tfDY0ODgyMjl8LTEyMjk2MjEwNg==>.
>
>



-- 
Vikram Agrawal
Director - Samuday Web Technologies


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6767269.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/f60c1c06/attachment.html 

From avi at avimarcus.net  Wed Sep  7 15:18:21 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 7 Sep 2011 14:18:21 +0300
Subject: [Freeswitch-users] Multiple DTMF on Single Keypress
In-Reply-To: <CADEQS5xsRouLa1fxGu9+=sUJdagO4cW6q0AfsoPKwDghKMvSCw@mail.gmail.com>
References: <CADEQS5zp8bh3epB3-ya9uR2HvNjnt8qf7u4rS=a11tNNGGo7hA@mail.gmail.com>
	<CA+Gmxbbm46vCtW70+Qt7LurJEybMHwDA6BA+tNa8OnL4U0uKfw@mail.gmail.com>
	<CAFKNP3-Z6Zhas8DZobHqw4fxampyYjwqMU=guzG1jvZvHuWWFA@mail.gmail.com>
	<CALNZuwk372OUr1cNdqGrPqmBdFPEf6EUSBuPzhqBmBu1Bczvxg@mail.gmail.com>
	<CADEQS5xsRouLa1fxGu9+=sUJdagO4cW6q0AfsoPKwDghKMvSCw@mail.gmail.com>
Message-ID: <CALNZuw=q2Up_2EpAkQ-5Pvfu_zZGWUsHUeK=iimhK+ueyyfSjg@mail.gmail.com>

Can you get a normal PCAP of the SIP/RTP with something here?
http://wiki.freeswitch.org/wiki/Packet_Capture

e.g. pcapsipdump is quite nice. (Just make sure the folder exists
before running the command.)

-Avi


On Wed, Sep 7, 2011 at 2:13 AM, Brian Wiese
<brian.wiese.freeswitch at gmail.com> wrote:
> Avi:
>
> I had thought it was inband, but I couldn't find anything that supported it,
> like you mentioned.
>
> Is there anything else I can provide that would help solve this problem?
>
> Thanks.
>
> ~Brian
>
> On Tue, Sep 6, 2011 at 3:48 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>> line 389
>> 2011-09-04?16:20:14.390350?[DEBUG]?switch_rtp.c:3317?RTP RECV DTMF?5:1440
>> 2011-09-04?16:20:14.390350?[DEBUG]?switch_ivr_bridge.c:391?Send signal
>> sofia/internal/sip:20005 at 172.31.6.253?[BREAK]
>> 2011-09-04?16:20:14.410350?[DEBUG]?switch_rtp.c:2343?Send start packet
>> for?[5]?ts=97600?dur=160/160/1440?seq=29252
>> 2011-09-04?16:20:14.410350?[DEBUG]?switch_rtp.c:3317?RTP RECV DTMF?5:1440
>>
>> It's coming as RTP RECV, meaning rfc2833. Otherwise, it would say
>> "DETECTED".
>>
>> -Avi
>>
>>
>> On Tue, Sep 6, 2011 at 11:36 PM, Jon Young <jonyoung111 at gmail.com> wrote:
>> >
>> > Is it possible you are receiving 2833 and Inband DTMF?
>> >
>> > On Mon, Sep 5, 2011 at 10:36 PM, Dmitry Sytchev <kbdfck at gmail.com>
>> > wrote:
>> > > See the same behaviour with inband DTMF detector sometimes.
>> > >
>> > > 2011/9/5 Brian Wiese FreeSWITCH List
>> > > <brian.wiese.freeswitch at gmail.com>
>> > >>
>> > >> Hello everyone!
>> > >>
>> > >> I'm getting multiple RTP DTMF from random keypresses and I can't
>> > >> figure
>> > >> out why.? I've PB'ed the packet capture and FS log for a call.? As
>> > >> you can
>> > >> see from the FS log, the 9,8,7,6 numbers weren't repeated, but 5 was
>> > >> (again,
>> > >> though, different calls lead to different numbers being repeated).
>> > >> Log:? http://pastebin.freeswitch.org/17280
>> > >> Capture:? http://pastebin.freeswitch.org/17282
>> > >>
>> > >> I appreciate any ideas as to what I might have wrong here.
>> > >>
>> > >> 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
>> > >>
>> > >
>> > >
>> > >
>> > > --
>> > > 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
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Sep  7 15:56:59 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 7 Sep 2011 13:56:59 +0200
Subject: [Freeswitch-users] Can't make mod_rtmp
In-Reply-To: <CAOrstdOygDF_xWs57EY6iMafe-Sv_G-PyrG1G9yXXDdBGJ9mnw@mail.gmail.com>
References: <1308334343913-6488229.post@n2.nabble.com>
	<BANLkTimx1MnAhfhHVmVds3qYpqNpaC9JKQ@mail.gmail.com>
	<1308388959641-6490318.post@n2.nabble.com>
	<1315213321297-6760239.post@n2.nabble.com>
	<1315230432566-6760858.post@n2.nabble.com>
	<4E6587FA.2040307@livecall.com>
	<CAOrstdPsLoT2oQu7Zwj94HE8zZg1e3DzrUc-LLMUJcidjPXLdA@mail.gmail.com>
	<1315391201025-6767247.post@n2.nabble.com>
	<CAOrstdOygDF_xWs57EY6iMafe-Sv_G-PyrG1G9yXXDdBGJ9mnw@mail.gmail.com>
Message-ID: <4D68DBA7-4FC8-4257-BC61-20531A07C67F@avgs.ca>

No, it means it is not built. You can add it to modules.conf and do "make mod_rtmp-install" from the source directory (you could also simply do "make", but the former will only build and install mod_rtmp)

Once it is installed, you can add it to autoload_configs/modules.conf.xml so it automatically loads when freeswitch starts.

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




On 2011-09-07, at 12:38 PM, vikram wrote:

> In that case, there are issues in building of mod_rtmp. 
> 
> when you do "make" , do you see any error in building mod_rtmp?  ideally it should compile the rtmp code and create mod_rtmp.la 
> 
> On Wed, Sep 7, 2011 at 3:56 PM, kkocyk [via freeswitch-users] <[hidden email]> wrote:
> When I try to load module there is error: 
> 
> "-ERR [module load file routine returned an error] 
> 
> 2011-09-07 12:25:09.101871 [CRIT] switch_loadable_module.c:969 Error Loading module /usr/local/freeswitch/mod/mod_rtmp.so 
> **/usr/local/freeswitch/mod/mod_rtmp.so: cannot open shared object file: No such file or directory**" 
> 
> vikram wrote:
> mod_rtmp is building by default but it is not loaded when I start 
> freeswitch. 
> 
> However if I type "load mod_rtmp" in freeswitch, it loads the module. 
> 
> On Tue, Sep 6, 2011 at 8:12 AM, Jack [via freeswitch-users] < 
> [hidden email]> wrote: 
> 
> > There is also a rtmp.conf.xml in the autoload_configs folder. 
> > It sounds like you need to include the rtmp source in your build. 
> > src\mod\endpoints\mod_rtmp\mod_rtmp.c 
> > I do not think it builds by default. 
> > Jack 
> > 
> > On 9/5/2011 6:47 AM, vikram wrote: 
> > 
> > > I am facing the same issue.  I have made a fresh installation with 
> > mod_rtmp. 
> > > 
> > > unknown command: rtmp 
> > > 
> > > 
> > > kkocyk wrote: 
> > >> 
> > >> peely wrote: 
> > >>> I'm not sure if I need to do anything else, I just naively added 
> > >>> endpoints/mod_rtmp to modules.conf and did a make clean install? 
> > >>> 
> > >> I'm trying to make it same way (my first install was without mod_rtmp), 
> > >> but after starting FS it seams that I still don't have rtmp. When I 
> > enter 
> > >> "rtmp status" it says that there is no such command. 
> > >> 
> > > 
> > > -- 
> > > View this message in context: 
> > http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6760858.html
> > > Sent from the freeswitch-users mailing list archive at Nabble.com. 
> > > 
> > > 
> > > FreeSWITCH-users mailing list 
> > > [hidden email] <http://user/SendEmail.jtp?type=node&node=6762531&i=0>
> 
> > > http://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=6762531&i=1>
> 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> > 
> > ------------------------------ 
> >  If you reply to this email, your message will be added to the discussion 
> > below: 
> > 
> > http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6762531.html
> >  To unsubscribe from Can't make mod_rtmp, click here< 
> > 
> > 
> 
> 
> 
> -- 
> Vikram Agrawal 
> Director - Samuday Web Technologies 
> 
> 
> -- 
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6763678.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
> 
> 
> If you reply to this email, your message will be added to the discussion below:
> http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6767247.html
> To unsubscribe from Can't make mod_rtmp, click here.
> 
> 
> 
> -- 
> Vikram Agrawal
> Director - Samuday Web Technologies
> 
> 
> View this message in context: Re: Can't make mod_rtmp
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/b60edef2/attachment-0001.html 

From peder at networkoblivion.com  Wed Sep  7 16:42:18 2011
From: peder at networkoblivion.com (Peder)
Date: Wed, 7 Sep 2011 07:42:18 -0500
Subject: [Freeswitch-users] - freeswitch bye issue
In-Reply-To: <011c01cc6d08$6badcce0$430966a0$@gmail.com>
References: <009001cc6cb3$f24b51f0$d6e1f5d0$@gmail.com>	<0aae01cc6cbe$633ab680$29b02380$@com>
	<011c01cc6d08$6badcce0$430966a0$@gmail.com>
Message-ID: <006601cc6d5b$94dd89e0$be989da0$@com>

Public IP on what?  The phone, or FS?  Did you do a packet capture at the FS
box, or at the phone?  If you did it at the phone, it still doesn't prove
that FS didn't send a bye.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Simon
Leck
Sent: Tuesday, September 06, 2011 9:47 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] - freeswitch bye issue

 

Hi Pedar, 

 

Thanks for your reply. At first I thought it was NAT issue so to isolate the
problem I used a Public IP but I still get the same result? I have used
wireshark to do a trace but I don't see FS sending a bye to A?

 

Kindly advice me on how else could I solved this issue. On the NAT Part, I
have enable RTP keep alive as well. Under RTCP, I have the keep alive turned
on as well?

 

Which other place are there setting I can configure with?

 

Thanks in advanced Pedar for your kindest help

 

Many thanks

Simon Leck

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peder
Sent: Wednesday, September 07, 2011 1:57 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] - freeswitch bye issue

 

It should happen automatically.  Is A behind NAT?  Sounds like a
firewall/NAT may be dropping the bye.  Also, have you done a debug on FS to
see if it actually sends a bye or not to A?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Simon
Leck
Sent: Tuesday, September 06, 2011 11:42 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] - freeswitch bye issue

 

Hi Everyone, kindly help if you can 

 

At the moment I did encounter a issue, A leg user called a b leg user and
when b leg user hanged up (Bye is not send by freeswitch to the A leg user)
the only way I could hanged up this call  to hanged up the A leg user
manually? Would be great if someone could enlighten me on how this can be
achieved?

 

Thanks in advanced to anyone for helping me out

 

Thanks again

Simon Leck

 

 

 

 

 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/eed80950/attachment.html 

From michal.bielicki at seventhsignal.de  Wed Sep  7 17:05:32 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Wed, 7 Sep 2011 15:05:32 +0200
Subject: [Freeswitch-users] Is there a SIP client that is capable of
	displaying arbitrary SIP headers?
In-Reply-To: <CAMDVdeitxqqT-JXTsS=dJYxgW81jDqX1pPQvEta=XoGD+F9=vQ@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B79511882D287@VMBX102.ihostexchange.net>
	<CAMDVdeitxqqT-JXTsS=dJYxgW81jDqX1pPQvEta=XoGD+F9=vQ@mail.gmail.com>
Message-ID: <B0D2659B-552F-4EA3-AE70-8E267BCD9EAA@seventhsignal.de>

Wouldn't wireshak plus ANY UA be the answer ? 
Am 07.09.2011 um 10:10 schrieb jcgpoza gonzalez:

> Why don't you use SIPp?
> 
> http://sipp.sourceforge.net/
> 
> really useful tool.
> 
> 2011/9/6 Yungwei Chen <yungwei at resolvity.com>
> Hi,
> 
> I'm wondering if there's a SIP client that can display arbibrary SIP headers on its UI. 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

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht 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



----

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/c5bb1893/attachment.html 

From govoiper at gmail.com  Wed Sep  7 17:16:50 2011
From: govoiper at gmail.com (Sam Govind)
Date: Wed, 7 Sep 2011 18:16:50 +0500
Subject: [Freeswitch-users] Is there a SIP client that is capable of
 displaying arbitrary SIP headers?
In-Reply-To: <B0D2659B-552F-4EA3-AE70-8E267BCD9EAA@seventhsignal.de>
References: <33095823FD21DF429B481B5163264B79511882D287@VMBX102.ihostexchange.net>
	<CAMDVdeitxqqT-JXTsS=dJYxgW81jDqX1pPQvEta=XoGD+F9=vQ@mail.gmail.com>
	<B0D2659B-552F-4EA3-AE70-8E267BCD9EAA@seventhsignal.de>
Message-ID: <CAJUJwtiyVz2a68nLeR1We64YCc=GJLPdKBmE=1o0183o_1_0nQ@mail.gmail.com>

1- SIPp is a tool not a user friendly application to take calls and display
info happily on UI
2- Imagine how the world would look like with end users finding their
required header from wireshark while taking calls from their ANY UA

I think the requirement is to send some informatory SIP Header to an end
point (Softphone) and that soft-phone display that custom SIP header some
where on screen. Like an agent taking calls with some call specific info on
screen.


On Wed, Sep 7, 2011 at 6:05 PM, Michal Bielicki <
michal.bielicki at seventhsignal.de> wrote:

> Wouldn't wireshak plus ANY UA be the answer ?
> Am 07.09.2011 um 10:10 schrieb jcgpoza gonzalez:
>
> Why don't you use SIPp?
>
> http://sipp.sourceforge.net/
>
> really useful tool.
>
> 2011/9/6 Yungwei Chen <yungwei at resolvity.com>
>
>> Hi,
>>
>> I'm wondering if there's a SIP client that can display arbibrary SIP
>> headers on its UI. 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
>
>
> *Michal Bielicki*
> Gesch?ftsf?hrer / CEO
>
> *Seventh Signal Ltd. & Co. KG*
> Weigandufer 45, B?ro 115, D-12059 Berlin
> Voice: +49 30 60988730
>
> Amtsgericht 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
>
>
> ----
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/835a4e92/attachment-0001.html 

From iwansss at gmail.com  Wed Sep  7 14:17:05 2011
From: iwansss at gmail.com (e1s)
Date: Wed, 7 Sep 2011 03:17:05 -0700 (PDT)
Subject: [Freeswitch-users] [mod_xml_curl] Keep receiving 403 forbidden
Message-ID: <1315390625471-6767218.post@n2.nabble.com>

I keep on receiving 403 Forbidden when try to register using mod_curl_xml,

here's the log lines : 
http://pastebin.freeswitch.org/17293

my curl return : 

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="directory" description="User Directory">
<domain name="10.1.1.47">"
   <user id="6597293279">
         <params>
                  
         </params>
    </user>
</domain>
</section>
</document>

I couldn't find what's wrong..


-----
/e1.s
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-xml-curl-Keep-receiving-403-forbidden-tp6767218p6767218.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From g.kocjan at systemycallcenter.pl  Wed Sep  7 14:26:41 2011
From: g.kocjan at systemycallcenter.pl (kkocyk)
Date: Wed, 7 Sep 2011 03:26:41 -0700 (PDT)
Subject: [Freeswitch-users] Can't make mod_rtmp
In-Reply-To: <CAOrstdPsLoT2oQu7Zwj94HE8zZg1e3DzrUc-LLMUJcidjPXLdA@mail.gmail.com>
References: <1308334343913-6488229.post@n2.nabble.com>
	<BANLkTimx1MnAhfhHVmVds3qYpqNpaC9JKQ@mail.gmail.com>
	<1308388959641-6490318.post@n2.nabble.com>
	<1315213321297-6760239.post@n2.nabble.com>
	<1315230432566-6760858.post@n2.nabble.com>
	<4E6587FA.2040307@livecall.com>
	<CAOrstdPsLoT2oQu7Zwj94HE8zZg1e3DzrUc-LLMUJcidjPXLdA@mail.gmail.com>
Message-ID: <1315391201025-6767247.post@n2.nabble.com>

When I try to load module there is error:

"-ERR [module load file routine returned an error]

2011-09-07 12:25:09.101871 [CRIT] switch_loadable_module.c:969 Error Loading
module /usr/local/freeswitch/mod/mod_rtmp.so
**/usr/local/freeswitch/mod/mod_rtmp.so: cannot open shared object file: No
such file or directory**"


vikram wrote:
> 
> mod_rtmp is building by default but it is not loaded when I start
> freeswitch.
> 
> However if I type "load mod_rtmp" in freeswitch, it loads the module.
> 
> On Tue, Sep 6, 2011 at 8:12 AM, Jack [via freeswitch-users] <
> ml-node+6762531-1355026059-354831 at n2.nabble.com> wrote:
> 
>> There is also a rtmp.conf.xml in the autoload_configs folder.
>> It sounds like you need to include the rtmp source in your build.
>> src\mod\endpoints\mod_rtmp\mod_rtmp.c
>> I do not think it builds by default.
>> Jack
>>
>> On 9/5/2011 6:47 AM, vikram wrote:
>>
>> > I am facing the same issue.  I have made a fresh installation with
>> mod_rtmp.
>> >
>> > unknown command: rtmp
>> >
>> >
>> > kkocyk wrote:
>> >>
>> >> peely wrote:
>> >>> I'm not sure if I need to do anything else, I just naively added
>> >>> endpoints/mod_rtmp to modules.conf and did a make clean install?
>> >>>
>> >> I'm trying to make it same way (my first install was without
>> mod_rtmp),
>> >> but after starting FS it seams that I still don't have rtmp. When I
>> enter
>> >> "rtmp status" it says that there is no such command.
>> >>
>> >
>> > --
>> > View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6760858.html
>> > Sent from the freeswitch-users mailing list archive at Nabble.com.
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > [hidden email]
>> &lt;http://user/SendEmail.jtp?type=node&amp;node=6762531&amp;i=0&gt;
>> > http://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]
>> &lt;http://user/SendEmail.jtp?type=node&amp;node=6762531&amp;i=1&gt;
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> ------------------------------
>>  If you reply to this email, your message will be added to the discussion
>> below:
>>
>> http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6762531.html
>>  To unsubscribe from Can't make mod_rtmp, click
>> here&lt;http://freeswitch-users.2379917.n2.nabble.com/template/NamlServlet.jtp?macro=unsubscribe_by_code&amp;node=6488229&amp;code=dmlrcmFtLmFncmF3YWxAZ21haWwuY29tfDY0ODgyMjl8LTEyMjk2MjEwNg==&gt;.
>>
>>
> 
> 
> 
> -- 
> Vikram Agrawal
> Director - Samuday Web Technologies
> 
> 
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6763678.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
> 


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6767247.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From wagnerspi at gmail.com  Wed Sep  7 16:15:28 2011
From: wagnerspi at gmail.com (Wagner)
Date: Wed, 7 Sep 2011 09:15:28 -0300
Subject: [Freeswitch-users] Calls over 3G
In-Reply-To: <CAMWiMy-vvfYGRREx9_MtJjntwdJ0b8roZKtvfdeijBpAEQ1R=Q@mail.gmail.com>
References: <CAO0ino=uZbveU24WufHEvnH=AKa0TP5Xy1sBbcJEdv7NbW8iWg@mail.gmail.com>
	<512047ED5AE74A819B2D13E467806500@omni1.local>
	<CAMWiMy-vvfYGRREx9_MtJjntwdJ0b8roZKtvfdeijBpAEQ1R=Q@mail.gmail.com>
Message-ID: <CAO0ino=RO=LSteqan-TaML8nszJJc4WtK-wgNOtvUpGY+o=wSA@mail.gmail.com>

I just changed the codec to iLBC and it got really better, is working real
fine over 3G :)

thanks for the help guys

2011/9/5 curriegrad2004 <curriegrad2004 at gmail.com>

> There's always the option of using OpenVPN over a 3G network, that way
> it should bypass most of the restrictions placed by your mobile
> carrier. Not to mention that NAT-T over IPSec works well too
>
> On Mon, Sep 5, 2011 at 12:34 AM, Anestis Mavro <devel at omninet.eu> wrote:
> >
> >
> >
> >
> > Hi,
> >
> >
> >
> > I have tested G729 and iLBC and both are working even on GPRS Edge
> networks.
> >
> >
> >
> > Don?t forget, that many mobile operators are blocking VoIP. With one
> > operator we had issues on some cells. In many areas his network supported
> > VoIP in some other areas even the registration was not working.
> >
> > And of course don?t forget, that the bandwidth is shared (and limited):
> in
> > one cell you might get very good quality and in another one (or at
> another
> > time) you might get very poor quality phone calls.
> >
> >
> >
> > Good luck
> >
> >
> >
> > ________________________________
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Wagner
> > Sent: Monday, September 05, 2011 3:52 AM
> > To: FreeSWITCH Users Help
> > Subject: [Freeswitch-users] Calls over 3G
> >
> >
> >
> > Hello,
> >
> >
> >
> > What's the best way to make calls over 3G networks with less delay and
> more
> > quality?
> >
> >
> >
> > any kind of compression, or different codec, any ideas?
> >
> >
> >
> > Thanks
> >
> > __________ Information from ESET NOD32 Antivirus, version of virus
> signature
> > database 5054 (20100423) __________
> >
> > The message was checked by ESET NOD32 Antivirus.
> >
> > http://www.eset.com
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/b463eb31/attachment.html 

From simon0922 at gmail.com  Wed Sep  7 17:50:12 2011
From: simon0922 at gmail.com (Simon Leck)
Date: Wed, 7 Sep 2011 21:50:12 +0800
Subject: [Freeswitch-users] - freeswitch bye issue
In-Reply-To: <006601cc6d5b$94dd89e0$be989da0$@com>
References: <009001cc6cb3$f24b51f0$d6e1f5d0$@gmail.com>	<0aae01cc6cbe$633ab680$29b02380$@com>	<011c01cc6d08$6badcce0$430966a0$@gmail.com>
	<006601cc6d5b$94dd89e0$be989da0$@com>
Message-ID: <031b01cc6d65$13fd65c0$3bf83140$@gmail.com>

Hi Peder, 

 

Thanks again for your swift reply. FS is running on a Public IP address. The
UA is also running on a Public IP address. Yes, I did used wireshark to do a
sip trace on FS and see that the bye was not present. 

 

One reason I could think of could be the on-hook and off-hook time interval
or frequency for off-net calling is different that's why it could not hang
up? Or what else could be affecting.

 

Thanks in advanced for helping me out and I look forward to reply. 

 

Btw, are you on IRC and if you are can I chat with you?

 

Many thanks 

Simon Leck

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peder
Sent: Wednesday, September 07, 2011 8:42 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] - freeswitch bye issue

 

Public IP on what?  The phone, or FS?  Did you do a packet capture at the FS
box, or at the phone?  If you did it at the phone, it still doesn't prove
that FS didn't send a bye.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Simon
Leck
Sent: Tuesday, September 06, 2011 9:47 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] - freeswitch bye issue

 

Hi Pedar, 

 

Thanks for your reply. At first I thought it was NAT issue so to isolate the
problem I used a Public IP but I still get the same result? I have used
wireshark to do a trace but I don't see FS sending a bye to A?

 

Kindly advice me on how else could I solved this issue. On the NAT Part, I
have enable RTP keep alive as well. Under RTCP, I have the keep alive turned
on as well?

 

Which other place are there setting I can configure with?

 

Thanks in advanced Pedar for your kindest help

 

Many thanks

Simon Leck

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peder
Sent: Wednesday, September 07, 2011 1:57 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] - freeswitch bye issue

 

It should happen automatically.  Is A behind NAT?  Sounds like a
firewall/NAT may be dropping the bye.  Also, have you done a debug on FS to
see if it actually sends a bye or not to A?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Simon
Leck
Sent: Tuesday, September 06, 2011 11:42 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] - freeswitch bye issue

 

Hi Everyone, kindly help if you can 

 

At the moment I did encounter a issue, A leg user called a b leg user and
when b leg user hanged up (Bye is not send by freeswitch to the A leg user)
the only way I could hanged up this call  to hanged up the A leg user
manually? Would be great if someone could enlighten me on how this can be
achieved?

 

Thanks in advanced to anyone for helping me out

 

Thanks again

Simon Leck

 

 

 

 

 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/10eba325/attachment-0001.html 

From peder at networkoblivion.com  Wed Sep  7 17:55:34 2011
From: peder at networkoblivion.com (Peder)
Date: Wed, 7 Sep 2011 08:55:34 -0500
Subject: [Freeswitch-users] - freeswitch bye issue
In-Reply-To: <031b01cc6d65$13fd65c0$3bf83140$@gmail.com>
References: <009001cc6cb3$f24b51f0$d6e1f5d0$@gmail.com>	<0aae01cc6cbe$633ab680$29b02380$@com>	<011c01cc6d08$6badcce0$430966a0$@gmail.com>	<006601cc6d5b$94dd89e0$be989da0$@com>
	<031b01cc6d65$13fd65c0$3bf83140$@gmail.com>
Message-ID: <00c801cc6d65$d0ce5790$726b06b0$@com>

Are these users SIP, or thru an analog line, or?  I assumed they were both
SIP, but based on your comments about on-hook and off-hook, I am thinking
maybe analog for one.  No, I am not on IRC.  Lots of other people are
though.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Simon
Leck
Sent: Wednesday, September 07, 2011 8:50 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] - freeswitch bye issue

 

Hi Peder, 

 

Thanks again for your swift reply. FS is running on a Public IP address. The
UA is also running on a Public IP address. Yes, I did used wireshark to do a
sip trace on FS and see that the bye was not present. 

 

One reason I could think of could be the on-hook and off-hook time interval
or frequency for off-net calling is different that's why it could not hang
up? Or what else could be affecting.

 

Thanks in advanced for helping me out and I look forward to reply. 

 

Btw, are you on IRC and if you are can I chat with you?

 

Many thanks 

Simon Leck

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peder
Sent: Wednesday, September 07, 2011 8:42 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] - freeswitch bye issue

 

Public IP on what?  The phone, or FS?  Did you do a packet capture at the FS
box, or at the phone?  If you did it at the phone, it still doesn't prove
that FS didn't send a bye.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Simon
Leck
Sent: Tuesday, September 06, 2011 9:47 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] - freeswitch bye issue

 

Hi Pedar, 

 

Thanks for your reply. At first I thought it was NAT issue so to isolate the
problem I used a Public IP but I still get the same result? I have used
wireshark to do a trace but I don't see FS sending a bye to A?

 

Kindly advice me on how else could I solved this issue. On the NAT Part, I
have enable RTP keep alive as well. Under RTCP, I have the keep alive turned
on as well?

 

Which other place are there setting I can configure with?

 

Thanks in advanced Pedar for your kindest help

 

Many thanks

Simon Leck

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peder
Sent: Wednesday, September 07, 2011 1:57 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] - freeswitch bye issue

 

It should happen automatically.  Is A behind NAT?  Sounds like a
firewall/NAT may be dropping the bye.  Also, have you done a debug on FS to
see if it actually sends a bye or not to A?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Simon
Leck
Sent: Tuesday, September 06, 2011 11:42 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] - freeswitch bye issue

 

Hi Everyone, kindly help if you can 

 

At the moment I did encounter a issue, A leg user called a b leg user and
when b leg user hanged up (Bye is not send by freeswitch to the A leg user)
the only way I could hanged up this call  to hanged up the A leg user
manually? Would be great if someone could enlighten me on how this can be
achieved?

 

Thanks in advanced to anyone for helping me out

 

Thanks again

Simon Leck

 

 

 

 

 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/311597d1/attachment.html 

From jcgpoza at gmail.com  Wed Sep  7 18:28:24 2011
From: jcgpoza at gmail.com (jcgpoza gonzalez)
Date: Wed, 7 Sep 2011 16:28:24 +0200
Subject: [Freeswitch-users] Is there a SIP client that is capable of
 displaying arbitrary SIP headers?
In-Reply-To: <CAJUJwtiyVz2a68nLeR1We64YCc=GJLPdKBmE=1o0183o_1_0nQ@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B79511882D287@VMBX102.ihostexchange.net>
	<CAMDVdeitxqqT-JXTsS=dJYxgW81jDqX1pPQvEta=XoGD+F9=vQ@mail.gmail.com>
	<B0D2659B-552F-4EA3-AE70-8E267BCD9EAA@seventhsignal.de>
	<CAJUJwtiyVz2a68nLeR1We64YCc=GJLPdKBmE=1o0183o_1_0nQ@mail.gmail.com>
Message-ID: <CAMDVdejjij2aKChVed=DQtgO9TNH19VQ8yZOdrJfHdeVpv4s-Q@mail.gmail.com>

Yes, I probably got it wrong, somehow I thought this was a
testing/performance-related question.

- If you just want to verify you are getting these headersI would use this
java applet -> http://www.mizu-voip.com/Download.aspx you can enable a SIP
trace window.
- If what you want to to "do something" with those specific headers I would
probably modify an open source solution just like Jitsi (also in Java).

Hope this helps.


2011/9/7 Sam Govind <govoiper at gmail.com>

> 1- SIPp is a tool not a user friendly application to take calls and display
> info happily on UI
> 2- Imagine how the world would look like with end users finding their
> required header from wireshark while taking calls from their ANY UA
>
> I think the requirement is to send some informatory SIP Header to an end
> point (Softphone) and that soft-phone display that custom SIP header some
> where on screen. Like an agent taking calls with some call specific info on
> screen.
>
>
> On Wed, Sep 7, 2011 at 6:05 PM, Michal Bielicki <
> michal.bielicki at seventhsignal.de> wrote:
>
>> Wouldn't wireshak plus ANY UA be the answer ?
>> Am 07.09.2011 um 10:10 schrieb jcgpoza gonzalez:
>>
>> Why don't you use SIPp?
>>
>> http://sipp.sourceforge.net/
>>
>> really useful tool.
>>
>> 2011/9/6 Yungwei Chen <yungwei at resolvity.com>
>>
>>> Hi,
>>>
>>> I'm wondering if there's a SIP client that can display arbibrary SIP
>>> headers on its UI. 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
>>
>>
>>  *Michal Bielicki*
>> Gesch?ftsf?hrer / CEO
>>
>> *Seventh Signal Ltd. & Co. KG*
>> Weigandufer 45, B?ro 115, D-12059 Berlin
>> Voice: +49 30 60988730
>>
>> Amtsgericht 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
>>
>>
>> ----
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/1a04ff2c/attachment-0001.html 

From jmesquita at freeswitch.org  Wed Sep  7 18:33:02 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 7 Sep 2011 11:33:02 -0300
Subject: [Freeswitch-users] Is there a SIP client that is capable of
 displaying arbitrary SIP headers?
In-Reply-To: <CAMDVdejjij2aKChVed=DQtgO9TNH19VQ8yZOdrJfHdeVpv4s-Q@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B79511882D287@VMBX102.ihostexchange.net>
	<CAMDVdeitxqqT-JXTsS=dJYxgW81jDqX1pPQvEta=XoGD+F9=vQ@mail.gmail.com>
	<B0D2659B-552F-4EA3-AE70-8E267BCD9EAA@seventhsignal.de>
	<CAJUJwtiyVz2a68nLeR1We64YCc=GJLPdKBmE=1o0183o_1_0nQ@mail.gmail.com>
	<CAMDVdejjij2aKChVed=DQtgO9TNH19VQ8yZOdrJfHdeVpv4s-Q@mail.gmail.com>
Message-ID: <CABH8QX1c8cHisbpPiadtGY8vavUzfA0CNmrkABNPiFCpF0ZH_Q@mail.gmail.com>

Just a wild idea... Sponsoring FSComm? The project is kinda set aside
because of my professional priorities but I would looove to find a good
reason to get back to it..

Regards,
JM

On Wednesday, September 7, 2011, jcgpoza gonzalez wrote:

> Yes, I probably got it wrong, somehow I thought this was a
> testing/performance-related question.
>
> - If you just want to verify you are getting these headersI would use this
> java applet -> http://www.mizu-voip.com/Download.aspx you can enable a SIP
> trace window.
> - If what you want to to "do something" with those specific headers I would
> probably modify an open source solution just like Jitsi (also in Java).
>
> Hope this helps.
>
>
> 2011/9/7 Sam Govind <govoiper at gmail.com>
>
> 1- SIPp is a tool not a user friendly application to take calls and display
> info happily on UI
> 2- Imagine how the world would look like with end users finding their
> required header from wireshark while taking calls from their ANY UA
>
> I think the requirement is to send some informatory SIP Header to an end
> point (Softphone) and that soft-phone display that custom SIP header some
> where on screen. Like an agent taking calls with some call specific info on
> screen.
>
>
> On Wed, Sep 7, 2011 at 6:05 PM, Michal Bielicki <
> michal.bielicki at seventhsignal.de> wrote:
>
> Wouldn't wireshak plus ANY UA be the answer ?
> Am 07.09.2011 um 10:10 schrieb jcgpoza gonzalez:
>
> Why don't you use SIPp?
>
> http://sipp.sourceforge.net/
>
> really useful tool.
>
> 2011/9/6 Yungwei Chen <yungwei at resolvity.com>
>
> Hi,
>
> I'm wondering if there's a SIP client that can display arbibrary SIP
> headers on its UI. 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
>
>
>  *Michal Bielicki*
> Gesch?ftsf?hrer / CEO
>
> *Seventh Signal Ltd. & Co. KG*
> Weigandufer 45, B?ro 115, D-12059 Berlin
> Voice: +49 30 60988730
>
> Amtsgericht 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
>
>
> ----
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>
>

-- 
Jo?o Mesquita
FreeSWITCH? Solutions
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/263b9e22/attachment.html 

From yungwei at resolvity.com  Wed Sep  7 18:39:07 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 7 Sep 2011 10:39:07 -0400
Subject: [Freeswitch-users] Connecting to FS from outside
Message-ID: <33095823FD21DF429B481B5163264B79511882D4BE@VMBX102.ihostexchange.net>

Hi,

I am trying to connect to FS from outside using XLite.
In XLite settings, I set the domain to <the external IP address of FS>:5080 in order to use the external profile.
The problem is that XLite is unable to register with FS due to the following error: 

2011-09-06 09:32:52.030921 [WARNING] sofia_reg.c:2183 Can't find user [1000 at 71.197.220.29]
You must define a domain called '71.197.220.29' in your directory and add a user with the id="1000" attribute
and you must configure your device to use the proper domain in it's authentication credentials.

In conf/vavrs.xml, domain is set to the local IP address of FS.
How can I allow user 1000 to register with FS from outside? Thanks.


From michal.bielicki at seventhsignal.de  Wed Sep  7 18:56:04 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Wed, 7 Sep 2011 16:56:04 +0200
Subject: [Freeswitch-users] Can't make mod_rtmp
In-Reply-To: <1315391201025-6767247.post@n2.nabble.com>
References: <1308334343913-6488229.post@n2.nabble.com>
	<BANLkTimx1MnAhfhHVmVds3qYpqNpaC9JKQ@mail.gmail.com>
	<1308388959641-6490318.post@n2.nabble.com>
	<1315213321297-6760239.post@n2.nabble.com>
	<1315230432566-6760858.post@n2.nabble.com>
	<4E6587FA.2040307@livecall.com>
	<CAOrstdPsLoT2oQu7Zwj94HE8zZg1e3DzrUc-LLMUJcidjPXLdA@mail.gmail.com>
	<1315391201025-6767247.post@n2.nabble.com>
Message-ID: <0F89265D-2490-4D99-8633-5668E807154E@seventhsignal.de>

You did not build the module, so it cannot load it.

Am 07.09.2011 um 12:26 schrieb kkocyk:

> When I try to load module there is error:
> 
> "-ERR [module load file routine returned an error]
> 
> 2011-09-07 12:25:09.101871 [CRIT] switch_loadable_module.c:969 Error Loading
> module /usr/local/freeswitch/mod/mod_rtmp.so
> **/usr/local/freeswitch/mod/mod_rtmp.so: cannot open shared object file: No
> such file or directory**"
> 
> 
> vikram wrote:
>> 
>> mod_rtmp is building by default but it is not loaded when I start
>> freeswitch.
>> 
>> However if I type "load mod_rtmp" in freeswitch, it loads the module.
>> 
>> On Tue, Sep 6, 2011 at 8:12 AM, Jack [via freeswitch-users] <
>> ml-node+6762531-1355026059-354831 at n2.nabble.com> wrote:
>> 
>>> There is also a rtmp.conf.xml in the autoload_configs folder.
>>> It sounds like you need to include the rtmp source in your build.
>>> src\mod\endpoints\mod_rtmp\mod_rtmp.c
>>> I do not think it builds by default.
>>> Jack
>>> 
>>> On 9/5/2011 6:47 AM, vikram wrote:
>>> 
>>>> I am facing the same issue.  I have made a fresh installation with
>>> mod_rtmp.
>>>> 
>>>> unknown command: rtmp
>>>> 
>>>> 
>>>> kkocyk wrote:
>>>>> 
>>>>> peely wrote:
>>>>>> I'm not sure if I need to do anything else, I just naively added
>>>>>> endpoints/mod_rtmp to modules.conf and did a make clean install?
>>>>>> 
>>>>> I'm trying to make it same way (my first install was without
>>> mod_rtmp),
>>>>> but after starting FS it seams that I still don't have rtmp. When I
>>> enter
>>>>> "rtmp status" it says that there is no such command.
>>>>> 
>>>> 
>>>> --
>>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6760858.html
>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>> 
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> [hidden email]
>>> &lt;http://user/SendEmail.jtp?type=node&amp;node=6762531&amp;i=0&gt;
>>>> http://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]
>>> &lt;http://user/SendEmail.jtp?type=node&amp;node=6762531&amp;i=1&gt;
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> ------------------------------
>>> If you reply to this email, your message will be added to the discussion
>>> below:
>>> 
>>> http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6762531.html
>>> To unsubscribe from Can't make mod_rtmp, click
>>> here&lt;http://freeswitch-users.2379917.n2.nabble.com/template/NamlServlet.jtp?macro=unsubscribe_by_code&amp;node=6488229&amp;code=dmlrcmFtLmFncmF3YWxAZ21haWwuY29tfDY0ODgyMjl8LTEyMjk2MjEwNg==&gt;.
>>> 
>>> 
>> 
>> 
>> 
>> -- 
>> Vikram Agrawal
>> Director - Samuday Web Technologies
>> 
>> 
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6763678.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
>> 
> 
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229p6767247.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

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht 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


----


From jcgpoza at gmail.com  Wed Sep  7 19:00:46 2011
From: jcgpoza at gmail.com (jcgpoza gonzalez)
Date: Wed, 7 Sep 2011 17:00:46 +0200
Subject: [Freeswitch-users] Connecting to FS from outside
In-Reply-To: <33095823FD21DF429B481B5163264B79511882D4BE@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B79511882D4BE@VMBX102.ihostexchange.net>
Message-ID: <CAMDVdeg_dcumXfNMb=5OXKZk16rTydqAP2bX00-hcW4UniG6BA@mail.gmail.com>

Why don't you use the internal profile instead. port 5060

internal is for registered users
external is for non registered users (external.xml auth-calls false)

Remember to set the default(1234) password in your X-lite.

Regards.




2011/9/7 Yungwei Chen <yungwei at resolvity.com>

> Hi,
>
> I am trying to connect to FS from outside using XLite.
> In XLite settings, I set the domain to <the external IP address of FS>:5080
> in order to use the external profile.
> The problem is that XLite is unable to register with FS due to the
> following error:
>
> 2011-09-06 09:32:52.030921 [WARNING] sofia_reg.c:2183 Can't find user [
> 1000 at 71.197.220.29]
> You must define a domain called '71.197.220.29' in your directory and add a
> user with the id="1000" attribute
> and you must configure your device to use the proper domain in it's
> authentication credentials.
>
> In conf/vavrs.xml, domain is set to the local IP address of FS.
> How can I allow user 1000 to register with FS from outside? Thanks.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/1281e931/attachment.html 

From steveayre at gmail.com  Wed Sep  7 19:43:31 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 7 Sep 2011 16:43:31 +0100
Subject: [Freeswitch-users] - freeswitch bye issue
In-Reply-To: <031b01cc6d65$13fd65c0$3bf83140$@gmail.com>
References: <009001cc6cb3$f24b51f0$d6e1f5d0$@gmail.com>
	<0aae01cc6cbe$633ab680$29b02380$@com>
	<011c01cc6d08$6badcce0$430966a0$@gmail.com>
	<006601cc6d5b$94dd89e0$be989da0$@com>
	<031b01cc6d65$13fd65c0$3bf83140$@gmail.com>
Message-ID: <CAFiqYumrxRkiZVgH0az9pATP7OJXd0Ae1iRY=VUjfLOHd0DkeQ@mail.gmail.com>

What does the Contact header in the INVITE look like?

The BYE is actually a separate SIP dialog, and the SIP server has to send it
to the address in the Contact header not the address the INVITE came from.

NAT or a misbehaving SIP client could potentially screw that up by sending a
bad Contact header, which might mean you're not seeing it in Wireshark
because it's not being sent where you expect.

-Steve



On 7 September 2011 14:50, Simon Leck <simon0922 at gmail.com> wrote:

> Hi Peder, ****
>
> ** **
>
> Thanks again for your swift reply. FS is running on a Public IP address.
> The UA is also running on a Public IP address. Yes, I did used wireshark to
> do a sip trace on FS and see that the bye was not present. ****
>
> ** **
>
> One reason I could think of could be the on-hook and off-hook time interval
> or frequency for off-net calling is different that?s why it could not hang
> up? Or what else could be affecting.****
>
> ** **
>
> Thanks in advanced for helping me out and I look forward to reply. ****
>
> ** **
>
> Btw, are you on IRC and if you are can I chat with you?****
>
> ** **
>
> Many thanks ****
>
> Simon Leck****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Peder
> *Sent:* Wednesday, September 07, 2011 8:42 PM
>
> *To:* 'FreeSWITCH Users Help'
> *Subject:* Re: [Freeswitch-users] - freeswitch bye issue****
>
> ** **
>
> Public IP on what?  The phone, or FS?  Did you do a packet capture at the
> FS box, or at the phone?  If you did it at the phone, it still doesn?t prove
> that FS didn?t send a bye.****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Simon Leck
> *Sent:* Tuesday, September 06, 2011 9:47 PM
> *To:* 'FreeSWITCH Users Help'
> *Subject:* Re: [Freeswitch-users] - freeswitch bye issue****
>
> ** **
>
> Hi Pedar, ****
>
> ** **
>
> Thanks for your reply. At first I thought it was NAT issue so to isolate
> the problem I used a Public IP but I still get the same result? I have used
> wireshark to do a trace but I don?t see FS sending a bye to A?****
>
> ** **
>
> Kindly advice me on how else could I solved this issue. On the NAT Part, I
> have enable RTP keep alive as well. Under RTCP, I have the keep alive turned
> on as well?****
>
> ** **
>
> Which other place are there setting I can configure with?****
>
> ** **
>
> Thanks in advanced Pedar for your kindest help****
>
> ** **
>
> Many thanks****
>
> Simon Leck****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Peder
> *Sent:* Wednesday, September 07, 2011 1:57 AM
> *To:* 'FreeSWITCH Users Help'
> *Subject:* Re: [Freeswitch-users] - freeswitch bye issue****
>
> ** **
>
> It should happen automatically.  Is A behind NAT?  Sounds like a
> firewall/NAT may be dropping the bye.  Also, have you done a debug on FS to
> see if it actually sends a bye or not to A?****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Simon Leck
> *Sent:* Tuesday, September 06, 2011 11:42 AM
> *To:* 'FreeSWITCH Users Help'
> *Subject:* Re: [Freeswitch-users] - freeswitch bye issue****
>
> ** **
>
> Hi Everyone, kindly help if you can ****
>
> ** **
>
> At the moment I did encounter a issue, A leg user called a b leg user and
> when b leg user hanged up (Bye is not send by freeswitch to the A leg user)
>  the only way I could hanged up this call  to hanged up the A leg user
> manually? Would be great if someone could enlighten me on how this can be
> achieved?****
>
> ** **
>
> Thanks in advanced to anyone for helping me out****
>
> ** **
>
> Thanks again****
>
> Simon Leck****
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/6e97d22a/attachment-0001.html 

From msc at freeswitch.org  Wed Sep  7 20:17:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 7 Sep 2011 09:17:36 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxU=J8GUdQ4FVOdp1BTm6SrgVyyeojNX=HU7UC2y22K2_w@mail.gmail.com>

Hello all!

We have a few items go over on today's conference call. I've been able to
get caught up on the ChangeLog and, as usual, there are some new items that
you may be interested in hearing about. Also, we could use some assistance
in getting caught up in documenting all these things. The agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_09_07

Looking forward to talking to you soon!

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/3efc8bc5/attachment.html 

From stviper at gmail.com  Wed Sep  7 18:16:47 2011
From: stviper at gmail.com (=?ISO-8859-2?B?qXRlZmFuIMh1ZGFp?=)
Date: Wed, 7 Sep 2011 16:16:47 +0200
Subject: [Freeswitch-users] Overlap dialing and not presented 'sending
	complete' flag
In-Reply-To: <mailman.2759.1315404783.29749.freeswitch-users@lists.freeswitch.org>
References: <mailman.2759.1315404783.29749.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAP3+0r1kLTxJ_8nSMyFX4Tnv91UKGF+ksbYKScEcvn3hsMfRCw@mail.gmail.com>

Dear Friends,

my questions are about FreeTDM and his module sangoma_isdn. I have to
use overlap dialing because of customer requirements.

First question:
In file ftmod_sangoma_isdn_stack_hndl.c on line [474] is following condition:

? ? ? ?if (cnStEvnt->sndCmplt.eh.pres || num_digits >= min_digits) {
? ? ? ? ? ? ? ?ftdm_set_state(ftdmchan, FTDM_CHANNEL_STATE_RING);
? ? ? ?} else {
? ? ? ? ? ? ? ?ftdm_log_chan(ftdmchan, FTDM_LOG_DEBUG, "received %d of %d
digits\n", num_digits, min_digits);
? ? ? ?}

I guess it should be && and no || because when you have set min_digits = 8
channel goes immediately to RING state after INFO message with eighth
digit is received. Than it isn't possible receive numbers longer then 8 digits!

Is it BUG???


Second question:
According to "ITU-T Recommendation Q.931" in chapter "5.1.3 Overlap
sending" is written:
---------------------------------------------------------------------------------------------------
The call information in the message which completes the information
sending may contain a sending complete indication, (e.g. the # character
or, as a network option, the sending complete information element)
appropriate to the dialling plan being used. The network shall restart
timer T302 on the receipt of every INFORMATION message not
containing a sending complete indication.
--------------------------------------------------------------------------------------------------

If I understand this flag "cnStEvnt->sndCmplt.eh.pres" mentioned in
the code above is used to find out if INFO message has 'sending
complete indication'. But what about situation when FreeTDM doesn't get it???
In that case FreeTDM received after some time (in my case 15s)
STATUS CONFIRM message with reason:

Recovery on timer expire (call_state:25 channel-state:COLLECT
cause:102) (suId:1 suInstId:2 spInstId:2)

Can you tell me if implementation of sangoma_isdn module also solves
this situation when 'sending complete' is not set?

Thanks.

Regards,
Stefan


From wagnerspi at gmail.com  Wed Sep  7 20:00:53 2011
From: wagnerspi at gmail.com (Wagner)
Date: Wed, 7 Sep 2011 13:00:53 -0300
Subject: [Freeswitch-users] Calls over 3G
In-Reply-To: <CAO0ino=RO=LSteqan-TaML8nszJJc4WtK-wgNOtvUpGY+o=wSA@mail.gmail.com>
References: <CAO0ino=uZbveU24WufHEvnH=AKa0TP5Xy1sBbcJEdv7NbW8iWg@mail.gmail.com>
	<512047ED5AE74A819B2D13E467806500@omni1.local>
	<CAMWiMy-vvfYGRREx9_MtJjntwdJ0b8roZKtvfdeijBpAEQ1R=Q@mail.gmail.com>
	<CAO0ino=RO=LSteqan-TaML8nszJJc4WtK-wgNOtvUpGY+o=wSA@mail.gmail.com>
Message-ID: <CAO0inomiBtcvOW+vp2s4e6XP773Stk6jqSdPuzcQi0sjwuaEXg@mail.gmail.com>

By any chance, does anyone knows a android voip client (free) that supports
iLBC?

2011/9/7 Wagner <wagnerspi at gmail.com>

> I just changed the codec to iLBC and it got really better, is working real
> fine over 3G :)
>
> thanks for the help guys
>
>
> 2011/9/5 curriegrad2004 <curriegrad2004 at gmail.com>
>
>> There's always the option of using OpenVPN over a 3G network, that way
>> it should bypass most of the restrictions placed by your mobile
>> carrier. Not to mention that NAT-T over IPSec works well too
>>
>> On Mon, Sep 5, 2011 at 12:34 AM, Anestis Mavro <devel at omninet.eu> wrote:
>> >
>> >
>> >
>> >
>> > Hi,
>> >
>> >
>> >
>> > I have tested G729 and iLBC and both are working even on GPRS Edge
>> networks.
>> >
>> >
>> >
>> > Don?t forget, that many mobile operators are blocking VoIP. With one
>> > operator we had issues on some cells. In many areas his network
>> supported
>> > VoIP in some other areas even the registration was not working.
>> >
>> > And of course don?t forget, that the bandwidth is shared (and limited):
>> in
>> > one cell you might get very good quality and in another one (or at
>> another
>> > time) you might get very poor quality phone calls.
>> >
>> >
>> >
>> > Good luck
>> >
>> >
>> >
>> > ________________________________
>> >
>> > From: freeswitch-users-bounces at lists.freeswitch.org
>> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Wagner
>> > Sent: Monday, September 05, 2011 3:52 AM
>> > To: FreeSWITCH Users Help
>> > Subject: [Freeswitch-users] Calls over 3G
>> >
>> >
>> >
>> > Hello,
>> >
>> >
>> >
>> > What's the best way to make calls over 3G networks with less delay and
>> more
>> > quality?
>> >
>> >
>> >
>> > any kind of compression, or different codec, any ideas?
>> >
>> >
>> >
>> > Thanks
>> >
>> > __________ Information from ESET NOD32 Antivirus, version of virus
>> signature
>> > database 5054 (20100423) __________
>> >
>> > The message was checked by ESET NOD32 Antivirus.
>> >
>> > http://www.eset.com
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/e2c4e856/attachment.html 

From jcgpoza at gmail.com  Wed Sep  7 20:28:29 2011
From: jcgpoza at gmail.com (jcgpoza gonzalez)
Date: Wed, 7 Sep 2011 18:28:29 +0200
Subject: [Freeswitch-users] Calls over 3G
In-Reply-To: <CAO0inomiBtcvOW+vp2s4e6XP773Stk6jqSdPuzcQi0sjwuaEXg@mail.gmail.com>
References: <CAO0ino=uZbveU24WufHEvnH=AKa0TP5Xy1sBbcJEdv7NbW8iWg@mail.gmail.com>
	<512047ED5AE74A819B2D13E467806500@omni1.local>
	<CAMWiMy-vvfYGRREx9_MtJjntwdJ0b8roZKtvfdeijBpAEQ1R=Q@mail.gmail.com>
	<CAO0ino=RO=LSteqan-TaML8nszJJc4WtK-wgNOtvUpGY+o=wSA@mail.gmail.com>
	<CAO0inomiBtcvOW+vp2s4e6XP773Stk6jqSdPuzcQi0sjwuaEXg@mail.gmail.com>
Message-ID: <CAMDVdehvWoiz4ZheDUtTNcMuiMbAs3+J8Qujcq0pA1GmFyFV4w@mail.gmail.com>

csipsimple,

By the way I use the GSM codec over G or 3G and It works well


2011/9/7 Wagner <wagnerspi at gmail.com>

> By any chance, does anyone knows a android voip client (free) that supports
> iLBC?
>
>
> 2011/9/7 Wagner <wagnerspi at gmail.com>
>
>> I just changed the codec to iLBC and it got really better, is working real
>> fine over 3G :)
>>
>> thanks for the help guys
>>
>>
>> 2011/9/5 curriegrad2004 <curriegrad2004 at gmail.com>
>>
>>> There's always the option of using OpenVPN over a 3G network, that way
>>> it should bypass most of the restrictions placed by your mobile
>>> carrier. Not to mention that NAT-T over IPSec works well too
>>>
>>> On Mon, Sep 5, 2011 at 12:34 AM, Anestis Mavro <devel at omninet.eu> wrote:
>>> >
>>> >
>>> >
>>> >
>>> > Hi,
>>> >
>>> >
>>> >
>>> > I have tested G729 and iLBC and both are working even on GPRS Edge
>>> networks.
>>> >
>>> >
>>> >
>>> > Don?t forget, that many mobile operators are blocking VoIP. With one
>>> > operator we had issues on some cells. In many areas his network
>>> supported
>>> > VoIP in some other areas even the registration was not working.
>>> >
>>> > And of course don?t forget, that the bandwidth is shared (and limited):
>>> in
>>> > one cell you might get very good quality and in another one (or at
>>> another
>>> > time) you might get very poor quality phone calls.
>>> >
>>> >
>>> >
>>> > Good luck
>>> >
>>> >
>>> >
>>> > ________________________________
>>> >
>>> > From: freeswitch-users-bounces at lists.freeswitch.org
>>> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>> Wagner
>>> > Sent: Monday, September 05, 2011 3:52 AM
>>> > To: FreeSWITCH Users Help
>>> > Subject: [Freeswitch-users] Calls over 3G
>>> >
>>> >
>>> >
>>> > Hello,
>>> >
>>> >
>>> >
>>> > What's the best way to make calls over 3G networks with less delay and
>>> more
>>> > quality?
>>> >
>>> >
>>> >
>>> > any kind of compression, or different codec, any ideas?
>>> >
>>> >
>>> >
>>> > Thanks
>>> >
>>> > __________ Information from ESET NOD32 Antivirus, version of virus
>>> signature
>>> > database 5054 (20100423) __________
>>> >
>>> > The message was checked by ESET NOD32 Antivirus.
>>> >
>>> > http://www.eset.com
>>> >
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/da07b35c/attachment.html 

From jcgpoza at gmail.com  Wed Sep  7 20:29:06 2011
From: jcgpoza at gmail.com (jcgpoza gonzalez)
Date: Wed, 7 Sep 2011 18:29:06 +0200
Subject: [Freeswitch-users] Calls over 3G
In-Reply-To: <CAO0inomiBtcvOW+vp2s4e6XP773Stk6jqSdPuzcQi0sjwuaEXg@mail.gmail.com>
References: <CAO0ino=uZbveU24WufHEvnH=AKa0TP5Xy1sBbcJEdv7NbW8iWg@mail.gmail.com>
	<512047ED5AE74A819B2D13E467806500@omni1.local>
	<CAMWiMy-vvfYGRREx9_MtJjntwdJ0b8roZKtvfdeijBpAEQ1R=Q@mail.gmail.com>
	<CAO0ino=RO=LSteqan-TaML8nszJJc4WtK-wgNOtvUpGY+o=wSA@mail.gmail.com>
	<CAO0inomiBtcvOW+vp2s4e6XP773Stk6jqSdPuzcQi0sjwuaEXg@mail.gmail.com>
Message-ID: <CAMDVdehmq1SpueuFcbSJ9xSisV38=LZAMOSaDB7O6H+zWJYohQ@mail.gmail.com>

csipsimple,

By the way I use the GSM codec over G or 3G and It works well


2011/9/7 Wagner <wagnerspi at gmail.com>

> By any chance, does anyone knows a android voip client (free) that supports
> iLBC?
>
>
> 2011/9/7 Wagner <wagnerspi at gmail.com>
>
>> I just changed the codec to iLBC and it got really better, is working real
>> fine over 3G :)
>>
>> thanks for the help guys
>>
>>
>> 2011/9/5 curriegrad2004 <curriegrad2004 at gmail.com>
>>
>>> There's always the option of using OpenVPN over a 3G network, that way
>>> it should bypass most of the restrictions placed by your mobile
>>> carrier. Not to mention that NAT-T over IPSec works well too
>>>
>>> On Mon, Sep 5, 2011 at 12:34 AM, Anestis Mavro <devel at omninet.eu> wrote:
>>> >
>>> >
>>> >
>>> >
>>> > Hi,
>>> >
>>> >
>>> >
>>> > I have tested G729 and iLBC and both are working even on GPRS Edge
>>> networks.
>>> >
>>> >
>>> >
>>> > Don?t forget, that many mobile operators are blocking VoIP. With one
>>> > operator we had issues on some cells. In many areas his network
>>> supported
>>> > VoIP in some other areas even the registration was not working.
>>> >
>>> > And of course don?t forget, that the bandwidth is shared (and limited):
>>> in
>>> > one cell you might get very good quality and in another one (or at
>>> another
>>> > time) you might get very poor quality phone calls.
>>> >
>>> >
>>> >
>>> > Good luck
>>> >
>>> >
>>> >
>>> > ________________________________
>>> >
>>> > From: freeswitch-users-bounces at lists.freeswitch.org
>>> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>> Wagner
>>> > Sent: Monday, September 05, 2011 3:52 AM
>>> > To: FreeSWITCH Users Help
>>> > Subject: [Freeswitch-users] Calls over 3G
>>> >
>>> >
>>> >
>>> > Hello,
>>> >
>>> >
>>> >
>>> > What's the best way to make calls over 3G networks with less delay and
>>> more
>>> > quality?
>>> >
>>> >
>>> >
>>> > any kind of compression, or different codec, any ideas?
>>> >
>>> >
>>> >
>>> > Thanks
>>> >
>>> > __________ Information from ESET NOD32 Antivirus, version of virus
>>> signature
>>> > database 5054 (20100423) __________
>>> >
>>> > The message was checked by ESET NOD32 Antivirus.
>>> >
>>> > http://www.eset.com
>>> >
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/cc3bda2f/attachment-0001.html 

From msc at freeswitch.org  Wed Sep  7 20:53:07 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 7 Sep 2011 09:53:07 -0700
Subject: [Freeswitch-users] DTMF from Nortel BCM 400 to Freeswitch
In-Reply-To: <CANDXbUzevDVxbo2ZHzVr1b5vtiVyM38m++WeYKvs8+8MGSKdTA@mail.gmail.com>
References: <CANDXbUyv9Q9i9FuajPs8rG7KaQagC3JG2Ji7zeJS9uZHYsxv+w@mail.gmail.com>
	<CANDXbUz8S-i21HcDDyuHJ5oe1HAgzi-8pc-MhNzE_nmkA+JSmA@mail.gmail.com>
	<CAKzWOxV2u9XoU596c=H67Z2RbF5oEw3_Ze6mRp1V5-MkUdT0FA@mail.gmail.com>
	<CANDXbUzevDVxbo2ZHzVr1b5vtiVyM38m++WeYKvs8+8MGSKdTA@mail.gmail.com>
Message-ID: <CAKzWOxW9AqAc5C12K-NYqMOejmiaSiLHYzYqLUSH=3AVvnfGFw@mail.gmail.com>

Sorry for the late followup... did you make any progress on this? If not,
get a pcap of the actual call w/ media to go along with a debug log +
siptrace. Put the latter on pb and put the pcap on a web server somewhere
that we can download it and have a look.

Thanks,
MC

On Wed, Aug 31, 2011 at 10:11 AM, Ray Pang <rpang88 at gmail.com> wrote:

> Yes.  Freeswitch is receiving SIP INFO messages (not sure if BCM is
> converting or sending it native).  I've tried on Freeswitch using suggested
> options with no avail.
>
> Example: <param name="dtmf-type" value="info"/> or <variable
> direction="inbound|outbound|both" name="dtmf_type" value="info"> and any
> other variations I that i was able to find.
> Thanks.
> RP
>
>
> On Wed, Aug 31, 2011 at 9:29 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Tue, Aug 30, 2011 at 8:47 PM, Ray Pang <rpang88 at gmail.com> wrote:
>>
>>> I?ve been unable to get DTMF to work from BCM 400 to Freeswitch.  I've
>>>> tried all DTMF settings with no luck.
>>>>
>>>>
>>>>
>>>
>> When you say that you've tried all DTMF settings, what does that mean?
>> Also, is the BCM converting in-band DTMFs into SIP INFO messages?
>>
>> -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
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/b9fd842d/attachment.html 

From msc at freeswitch.org  Wed Sep  7 21:01:23 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 7 Sep 2011 10:01:23 -0700
Subject: [Freeswitch-users] Problem receiving fax
In-Reply-To: <1315226786.3761.48.camel@marces.madrid.commsmundi.com>
References: <1313493347.30552.80.camel@marces.madrid.commsmundi.com>
	<CAAMQ8bROOOci2ocWZkcn_qxh-vc8kY1Tg+3deUJMs3Lvf0PMpw@mail.gmail.com>
	<1313496873.30552.82.camel@marces.madrid.commsmundi.com>
	<1314087013.29574.59.camel@marces.madrid.commsmundi.com>
	<1315226786.3761.48.camel@marces.madrid.commsmundi.com>
Message-ID: <CAKzWOxXLeN3s9mZ6T-LGoXdN+Y9j=NtBWteYJguvBQt9OCFBHA@mail.gmail.com>

This is probably sufficiently weird/goofy/difficult that you will need
professional assistance. I would suggest consulting at freeswitch.org.

-MC

On Mon, Sep 5, 2011 at 5:46 AM, Antonio <asilva at wirelessmundi.com> wrote:

> **
> Hi,
>
> I'm still fighting this problem.
>
> This now happen to me in another machine with the different hardware.
>
> I has receiving faxes with no problem and at some point or reason (i Can't
> catch IT!!) i is just stops.
> The only solution that i have for now is reboot the server.
>
>
> Is there anyone with the same problem as me?
> Can you point some way to have more debug/logs from spandsp or freeswitch
> so i can try to find out the problem?
>
> I'm not sure that is a bug but this behavior is pretty weird!!
>
> Btw: i did some audio captures and nothing wrong....
>
> Thanks,
>
> Ant?nio
>
>
>
>
> On Tue, 2011-08-23 at 10:10 +0200, Antonio wrote:
>
> I tried with the last version, and the same occurred.
>
> After restarting the server i can receive faxes and them it stops
> receiving it.
>
> I can't find out what is the cause... Can anyone help me how to find out
> where could be the problem.
>
>
> I'm think replacing the hardware, just to be sure that is not an
> hardware problem.
>
>
>
> Thanks,
>
> Ant?nio
>
>
>
>
>
>
> On Tue, 2011-08-16 at 14:14 +0200, Antonio wrote:
> > I'm using libpri-1.4.11 and freeswitch head. I'm going to try with the
> > latest libpri-1.4.12.
> >
> > And post the results.
> >
> > Thanks,
> > Ant?nio
> >
> >
> > On Tue, 2011-08-16 at 13:46 +0200, Christian Benke wrote:
> > > On 16 August 2011 13:15, Antonio <asilva at wirelessmundi.com> wrote:
> > > > I'm having problems receiving fax in a pri E1 line.
> > > > The log can be found at http://pastebin.freeswitch.org/17047
> > >
> > > Hi!
> > >
> > > I had the same issue a few days ago("FLOW T.30 Bad HDLC CRC received").
> > > Recompiling&Reinstalling libpri&FreeSWITCH helped.
> > >
> > > hthu2
> > > Christian
> > >
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
>
>
>
>   --
>
> Un cordial saludo / Best regards,
>
>  _________________________
>
> Ant?nio Silva
>
> E-mail:asilva at wirelessmundi.com
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/ebffd2be/attachment.html 

From thomas.ji at gmx.at  Wed Sep  7 21:03:33 2011
From: thomas.ji at gmx.at (thomas peterseil)
Date: Wed, 07 Sep 2011 19:03:33 +0200
Subject: [Freeswitch-users] freeswitch as a gateway with cdr lookup
Message-ID: <20110907170333.154010@gmx.net>

Hello FreeSWITCH-Users,

I am running a PBX with a GSM Gateway and i have problems with incoming calls on the GSM Gateway. I am wondering, if there is an "easy" possibility to solve it with FreeSwitch. Here is the problem:

Extension 1000 calls the mobile phone 1234, but on the display of the mobile is the number 987 (i can?t modify this number, it?s the number of the GSM Gateway) and nobody picks up the call. later the mobile number 1234 calls back the number 987, but the GSM Gateway has no idea who has called the 1234, so there is no way of routing the call back to the extension 1000.

Is there a possibility to put a FS between my PBX and the GSM Gateway, so when a call from the GSM Gateway comes in, FS makes a lookup in the CDR to check, which extension called this mobile number last time and then FS should route the call to the right extension.

Is it possible to realize that with FS and if yes, is that very difficult?

Thanks in advanced for any help and suggestions.

thomas
-- 
NEU: FreePhone - 0ct/min Handyspartarif mit Geld-zur?ck-Garantie!		
Jetzt informieren: http://www.gmx.net/de/go/freephone


From msc at freeswitch.org  Wed Sep  7 21:04:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 7 Sep 2011 10:04:02 -0700
Subject: [Freeswitch-users] Bug or Not? "longjmp causes uninitialized
 stack frame"
In-Reply-To: <CAFmdODjqu2dVBXoCZbJmn+WftypQPqJCw=nS4adG4QMeg17-sA@mail.gmail.com>
References: <CAFmdODjqu2dVBXoCZbJmn+WftypQPqJCw=nS4adG4QMeg17-sA@mail.gmail.com>
Message-ID: <CAKzWOxVvpDfY=N16BTAtsFqE4aU4sqfvxsRsSmirUCx-HGTLFg@mail.gmail.com>

Definitely put this one on Jira.
-MC

On Mon, Sep 5, 2011 at 1:39 AM, Michael Toop <michaelt at callall.co.za> wrote:

> Hi,
>
>  Sorry not sure what to do with this, if this is a bug or another problem,
> but getting this message in the bt when running FS as it does a core dump:
> " longjmp causes uninitialized stack frame".
>
>  Running on: 2.6.38-11-server #48-Ubuntu, Ubuntu 11.04 on the latest GIT
> release.
>
> Thanks,
>
> Michael
>
> 2011-09-04 15:34:59.149089 [INFO] mod_native_file.c:94 Opening File
> [/usr/local/freeswitch/sounds/custom/mwc-enter-destination.G729] 8000hz
> 2011-09-04 15:34:59.169011 [INFO] sofia.c:755 sofia/cellc/
> 763070366 at 172.103.0.36 Update Callee ID to "763070366" <763070366>
> 2011-09-04 15:34:59.169011 [NOTICE] switch_core_session.c:1367 Session 946
> (sofia/cellc/790938072 at 172.103.0.36) Ended
>  2011-09-04 15:34:59.169011 [NOTICE] switch_core_session.c:1369 Close
> Channel sofia/cellc/790938072 at 172.103.0.36 [CS_DESTROY]
> *** longjmp causes uninitialized stack frame ***: freeswitch terminated
> ======= Backtrace: =========
> /lib/x86_64-linux-gnu/libc.so.6(__fortify_fail+0x37)[0x7f1157fbe1d7]
> /lib/x86_64-linux-gnu/libc.so.6(+0xfe169)[0x7f1157fbe169]
> /lib/x86_64-linux-gnu/libc.so.6(__longjmp_chk+0x33)[0x7f1157fbe0d3]
> /usr/lib/libcurl.so.4(+0xd165)[0x7f1156457165]
> /lib/x86_64-linux-gnu/libpthread.so.0(+0xfc60)[0x7f1158a89c60]
> /lib/x86_64-linux-gnu/libc.so.6(__select+0x33)[0x7f1157f9e143]
>
> /usr/local/freeswitch/lib/libfreeswitch.so.1(apr_sleep+0x45)[0x7f11593d16c5]
> /usr/local/freeswitch/lib/libfreeswitch.so.1(+0xcab5c)[0x7f11593a7b5c]
>
> /usr/local/freeswitch/lib/libfreeswitch.so.1(switch_console_loop+0x74a)[0x7f1159332e9a]
> freeswitch[0x402b22]
> /lib/x86_64-linux-gnu/libc.so.6(__libc_start_main+0xff)[0x7f1157edeeff]
> freeswitch[0x401769]
> ======= Memory map: ========
> 00400000-00404000 r-xp 00000000 08:01 1845708
>  /usr/local/freeswitch/bin/freeswitch
> 00603000-00604000 r--p 00003000 08:01 1845708
>  /usr/local/freeswitch/bin/freeswitch
> 00604000-00605000 rw-p 00004000 08:01 1845708
>  /usr/local/freeswitch/bin/freeswitch
> 017b0000-026c9000 rw-p 00000000 00:00 0
>  [heap]
> 7f114004d000-7f114004e000 ---p 00000000 00:00 0
> 7f114004e000-7f1140089000 rw-p 00000000 00:00 0
> 7f1140089000-7f114008a000 ---p 00000000 00:00 0
> 7f114008a000-7f11400c5000 rw-p 00000000 00:00 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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/8760b6c7/attachment.html 

From anthony.minessale at gmail.com  Wed Sep  7 21:14:40 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 7 Sep 2011 12:14:40 -0500
Subject: [Freeswitch-users] Bug or Not? "longjmp causes uninitialized
 stack frame"
In-Reply-To: <CAKzWOxVvpDfY=N16BTAtsFqE4aU4sqfvxsRsSmirUCx-HGTLFg@mail.gmail.com>
References: <CAFmdODjqu2dVBXoCZbJmn+WftypQPqJCw=nS4adG4QMeg17-sA@mail.gmail.com>
	<CAKzWOxVvpDfY=N16BTAtsFqE4aU4sqfvxsRsSmirUCx-HGTLFg@mail.gmail.com>
Message-ID: <CAKbxfG87gWRZeM+PEmeM=MfoyrgofsnUSDY+Mu7eBnO2Kki1Ow@mail.gmail.com>

the latest libcurl on most systems is buggy, this is why we package
our own version of libs.

reconfigure your build with --without-libcurl to use our version.


On Wed, Sep 7, 2011 at 12:04 PM, Michael Collins <msc at freeswitch.org> wrote:
> Definitely put this one on Jira.
> -MC
>
> On Mon, Sep 5, 2011 at 1:39 AM, Michael Toop <michaelt at callall.co.za> wrote:
>>
>> Hi,
>> ?Sorry not sure what to do with this, if this is a bug or another problem,
>> but getting this message in the bt when running FS as it does a core dump:
>> "?longjmp causes uninitialized stack frame".
>> ?Running on: 2.6.38-11-server #48-Ubuntu, Ubuntu 11.04 on the latest GIT
>> release.
>> Thanks,
>> Michael
>> 2011-09-04 15:34:59.149089 [INFO] mod_native_file.c:94 Opening File
>> [/usr/local/freeswitch/sounds/custom/mwc-enter-destination.G729] 8000hz
>> 2011-09-04 15:34:59.169011 [INFO] sofia.c:755
>> sofia/cellc/763070366 at 172.103.0.36?Update Callee ID to "763070366"
>> <763070366>
>> 2011-09-04 15:34:59.169011 [NOTICE] switch_core_session.c:1367 Session 946
>> (sofia/cellc/790938072 at 172.103.0.36) Ended
>> 2011-09-04 15:34:59.169011 [NOTICE] switch_core_session.c:1369 Close
>> Channel sofia/cellc/790938072 at 172.103.0.36?[CS_DESTROY]
>> *** longjmp causes uninitialized stack frame ***: freeswitch terminated
>> ======= Backtrace: =========
>> /lib/x86_64-linux-gnu/libc.so.6(__fortify_fail+0x37)[0x7f1157fbe1d7]
>> /lib/x86_64-linux-gnu/libc.so.6(+0xfe169)[0x7f1157fbe169]
>> /lib/x86_64-linux-gnu/libc.so.6(__longjmp_chk+0x33)[0x7f1157fbe0d3]
>> /usr/lib/libcurl.so.4(+0xd165)[0x7f1156457165]
>> /lib/x86_64-linux-gnu/libpthread.so.0(+0xfc60)[0x7f1158a89c60]
>> /lib/x86_64-linux-gnu/libc.so.6(__select+0x33)[0x7f1157f9e143]
>>
>> /usr/local/freeswitch/lib/libfreeswitch.so.1(apr_sleep+0x45)[0x7f11593d16c5]
>> /usr/local/freeswitch/lib/libfreeswitch.so.1(+0xcab5c)[0x7f11593a7b5c]
>>
>> /usr/local/freeswitch/lib/libfreeswitch.so.1(switch_console_loop+0x74a)[0x7f1159332e9a]
>> freeswitch[0x402b22]
>> /lib/x86_64-linux-gnu/libc.so.6(__libc_start_main+0xff)[0x7f1157edeeff]
>> freeswitch[0x401769]
>> ======= Memory map: ========
>> 00400000-00404000 r-xp 00000000 08:01 1845708
>> ?/usr/local/freeswitch/bin/freeswitch
>> 00603000-00604000 r--p 00003000 08:01 1845708
>> ?/usr/local/freeswitch/bin/freeswitch
>> 00604000-00605000 rw-p 00004000 08:01 1845708
>> ?/usr/local/freeswitch/bin/freeswitch
>> 017b0000-026c9000 rw-p 00000000 00:00 0
>> ?[heap]
>> 7f114004d000-7f114004e000 ---p 00000000 00:00 0
>> 7f114004e000-7f1140089000 rw-p 00000000 00:00 0
>> 7f1140089000-7f114008a000 ---p 00000000 00:00 0
>> 7f114008a000-7f11400c5000 rw-p 00000000 00:00 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
>
>



-- 
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 cmrienzo at gmail.com  Wed Sep  7 21:30:33 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 7 Sep 2011 13:30:33 -0400
Subject: [Freeswitch-users] FreeSWITCH, Nuance, ASR: IVR Integration
In-Reply-To: <CD5A1499C8DEF34FAA3509722B00393E740262CA5F@m-mail03.office.premiertech>
References: <CD5A1499C8DEF34FAA3509722B00393E740262CA5F@m-mail03.office.premiertech>
Message-ID: <CAD-qRkjebD1YYrsZb7WkWw0ra+TfpjWNSyoTgavN=U_BhyMRXA@mail.gmail.com>

There's not really an easy way to do this out of the box.  The FS core
provides all the low-level functions necessary for doing ASR, but nothing
higher level for interpreting results, handling barge-in, starting
input-timers, etc.

I solved this issue with a custom dialplan APP.  I know of another other dev
that used event socket and the playback/detect_speech dialplan APPs.
Looking at the scripts/javascript/js_modules/SpeechTools.jm source, you can
see that anthm solved it a different way.



On Mon, Sep 5, 2011 at 3:02 AM, Glen Ganderton <
Glen.Ganderton at premier.com.au> wrote:

> Hi Guys, ****
>
> ** **
>
> What i am trying to do is configure an IVR using freeswitch and Nuance
> 5.1.5 to perform speech recognition. I have installed the latest****
>
> release of freeswitch on my CentOS system, I have also enabled and
> configured the unimrcp module. I could find lots of information to****
>
> configure the unimrcp module but there is no documentation on how to create
> IVR menu's using this module with Nuance. I have used the ****
>
> demo pizza IVR with the pocketsphinx module and I keep getting refered to
> this but it's no help as I want to use Nuance for ASR. Any****
>
> help on how to configure this would be great, specifically:****
>
> ** **
>
> * How to I call the IVR script from the dialplan****
>
> * What language/s can I code the IVR in and can you please provide a basic
> sample of how this is done with Nuance.****
>
> ** **
>
> Thank you in advance****
>
> ** **
>
> --Glen****
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/c7600aa5/attachment.html 

From msc at freeswitch.org  Wed Sep  7 21:37:34 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 7 Sep 2011 10:37:34 -0700
Subject: [Freeswitch-users] mod_fifo to Voicemail if there's no "agents"
In-Reply-To: <4E65F1CE.7070707@omeco.de>
References: <4E65F1CE.7070707@omeco.de>
Message-ID: <CAKzWOxW6EdcV11ncdpKPYD1wBoV1UYUxA5H0kaAF0k4G2aN7pQ@mail.gmail.com>

You can check what's happening in the fifo with the "fifo count <fifo_name>"
http://wiki.freeswitch.org/wiki/Mod_fifo#count

-MC

On Tue, Sep 6, 2011 at 3:11 AM, Silvio Escher <sescher_ml at omeco.de> wrote:

> Hi there,
>
> ive following in my public dp:
>
> <extension name="zentrale_fifo">
> <condition field="destination_number" expression="^zentrale_fifo$">
> <action application="set" data="fifo_music=intro_und_schleife.wav"/>
> <action application="answer"/>
> <action application="set" data="fifo_orbit_exten=zentrale_voicemail:30"/>
> <action application="fifo" data="zentrale_fifo in"/>
> </condition>
> </extension>
>
> <extension name="Zentrale">
> <condition field="${sip_to_user}" expression="^0631680350$"
> break="on-false"/>
> <condition wday="2-6" hour="9-16"> # zeit geht bis 59
> <action application="transfer" data="zentrale_fifo XML public"/>
> <anti-action application="answer"/>
> <anti-action application="sleep" data="1500"/>
> <anti-action application="playback" data="niemand_da_zeit.wav"/>
> <anti-action application="sleep" data="1000"/>
> <anti-action application="hangup" data="NO_ANSWER"/>
> </condition>
> </extension>
>
> and iam adding Members to the fifo at runtime ( during *extension or an
> daily reset by cron )
>
> ./fs_cli -x "fifo_member add zentrale_fifo user/26"
> or
> ./fs_cli -x "fifo_member del zentrale_fifo user/26"
>
> Everything works fine beside the little issue that if theres no
> Member/Agent in the fifo - the Call
> is just also waiting 30 seconds till transfered to voicemail.
>
> I noticed that with fifo list zentrale_fifo iam able to get an Memberlist (
> fifo count just shows me
> 0 Members - dunno why - bug ? ) - but iam unsure how to process further.
>
> Whats the ideal Solution to get an Caller directly to the Voicemail when no
> Member/Agent is in the
> called Fifo ?
>
> Best Regards,
> Silvio
>
> --
> Silvio Escher
> omeco GmbH
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/b5bfdef4/attachment.html 

From john at mobex.biz  Wed Sep  7 20:35:48 2011
From: john at mobex.biz (john at mobex.biz)
Date: Wed, 07 Sep 2011 09:35:48 -0700
Subject: [Freeswitch-users] Calls over 3G
Message-ID: <20110907093548.c4df8f4d94a0e5285fca477124561e40.cb1f77a7c1.wbe@email17.secureserver.net>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/2bcaa2eb/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: sigimg1
Type: image/gif
Size: 1105 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/2bcaa2eb/attachment-0001.gif 

From brian.wiese.freeswitch at gmail.com  Wed Sep  7 21:50:51 2011
From: brian.wiese.freeswitch at gmail.com (Brian Wiese)
Date: Wed, 7 Sep 2011 12:50:51 -0500
Subject: [Freeswitch-users] One-way audio issue with attended transfer
In-Reply-To: <SNT116-W212CCAFBD521CA3F341979D1F0@phx.gbl>
References: <SNT116-W212CCAFBD521CA3F341979D1F0@phx.gbl>
Message-ID: <CADEQS5zejaoLBmyM3UreaTZKbJLHRjjSxiPJrr9zfWzN8wMhfA@mail.gmail.com>

John:

While I don't have the answer, I did want to write to say that I'm
experiencing the same problem.  We just built this server about a week ago,
and I'm running this version:  FreeSWITCH Version 1.0.head (git-a966c2b
2011-09-04 11-09-23 -0500)

If I can figure out what's going on I'll certainly post back here!

~Brian

On Tue, Sep 6, 2011 at 11:01 PM, John Platts <john_platts at hotmail.com>wrote:

>
> We have been experiencing a one-way audio issue with attended transfer. The
> problem is that after an call is transferred through an attended transfer,
> the caller cannot be heard, but the caller can hear the voice of the person
> that the call is transferred to.
>
> Steps to reproduce the problem:
> 1. Dial into FreeSWITCH from an external phone.
> 2. Answer the call. One-way audio is not experienced here.
> 3. Press the Transfer button on the IP phone to initiate an attended
> transfer.
> 4. Dial the extension that the call is to be transferred to.
> 5. Allow the extension that the call is to be transferred to pick up the
> call. One-way audio is not experienced here.
> 6. Press the Transfer button to complete the transfer. The caller does hear
> audio from the person that the call is transferred to, but the person that
> the call is transferred to cannot hear the caller. The one way audio issue
> is occurring here.
>
> The attended transfer was performed from an extension on one SPA303 phone
> to an extension on another SPA303 phone. FreeSWITCH actually detects that
> the transfer is an attended transfer.
>
> Here are attachments that I have uploaded to pastebin:
> conf/dialplan/features.xml file: http://pastebin.freeswitch.org/17291
> FreeSWITCH log, including the attended transfer:
> http://pastebin.freeswitch.org/17286
>
> How do we get this one-way audio issue fixed?
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/df51592c/attachment-0001.html 

From claudio at clickfono.com  Wed Sep  7 22:10:13 2011
From: claudio at clickfono.com (=?iso-8859-1?Q?Claudio_=C1lvarez?=)
Date: Wed, 7 Sep 2011 15:10:13 -0300
Subject: [Freeswitch-users] STUN usage with Freeswitch
Message-ID: <2A0564E0-B521-4607-8BCB-1E417BBE6761@clickfono.com>

Hi all,

I currently operate a publicly accessible stun server, which one of my FreeSWITCH instances operating on Amazon EC2 queries for proper SIP and RTP configuration.

When I query the stun server from FS-EC2, I get the following:

freeswitch at internal> stun 74.xx.xx.xx
184.xx.xx.xx:35087

Can anyone clarify what does the 35087 number port stand for?

Thanks.

Regards,

--
Claudio Alvarez
claudio at clickfono.com

From sunwood360 at gmail.com  Wed Sep  7 22:37:41 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Wed, 7 Sep 2011 11:37:41 -0700
Subject: [Freeswitch-users] FreeSWITCH, Nuance, ASR: IVR Integration
In-Reply-To: <CAD-qRkjebD1YYrsZb7WkWw0ra+TfpjWNSyoTgavN=U_BhyMRXA@mail.gmail.com>
References: <CD5A1499C8DEF34FAA3509722B00393E740262CA5F@m-mail03.office.premiertech>
	<CAD-qRkjebD1YYrsZb7WkWw0ra+TfpjWNSyoTgavN=U_BhyMRXA@mail.gmail.com>
Message-ID: <CAKq5mS4eqivAUw5QWVThUB+Xi6U8QJ8X==UfGi-1Lmmfpde+eA@mail.gmail.com>

Does your app support MRCP protocol? That would make it more generic.
On Sep 7, 2011 10:33 AM, "Christopher Rienzo" <cmrienzo at gmail.com> wrote:
> There's not really an easy way to do this out of the box. The FS core
> provides all the low-level functions necessary for doing ASR, but nothing
> higher level for interpreting results, handling barge-in, starting
> input-timers, etc.
>
> I solved this issue with a custom dialplan APP. I know of another other
dev
> that used event socket and the playback/detect_speech dialplan APPs.
> Looking at the scripts/javascript/js_modules/SpeechTools.jm source, you
can
> see that anthm solved it a different way.
>
>
>
> On Mon, Sep 5, 2011 at 3:02 AM, Glen Ganderton <
> Glen.Ganderton at premier.com.au> wrote:
>
>> Hi Guys, ****
>>
>> ** **
>>
>> What i am trying to do is configure an IVR using freeswitch and Nuance
>> 5.1.5 to perform speech recognition. I have installed the latest****
>>
>> release of freeswitch on my CentOS system, I have also enabled and
>> configured the unimrcp module. I could find lots of information to****
>>
>> configure the unimrcp module but there is no documentation on how to
create
>> IVR menu's using this module with Nuance. I have used the ****
>>
>> demo pizza IVR with the pocketsphinx module and I keep getting refered to
>> this but it's no help as I want to use Nuance for ASR. Any****
>>
>> help on how to configure this would be great, specifically:****
>>
>> ** **
>>
>> * How to I call the IVR script from the dialplan****
>>
>> * What language/s can I code the IVR in and can you please provide a
basic
>> sample of how this is done with Nuance.****
>>
>> ** **
>>
>> Thank you in advance****
>>
>> ** **
>>
>> --Glen****
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/58a50dc2/attachment.html 

From cmrienzo at gmail.com  Wed Sep  7 23:55:27 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 7 Sep 2011 15:55:27 -0400
Subject: [Freeswitch-users] FreeSWITCH, Nuance, ASR: IVR Integration
In-Reply-To: <CAKq5mS4eqivAUw5QWVThUB+Xi6U8QJ8X==UfGi-1Lmmfpde+eA@mail.gmail.com>
References: <CD5A1499C8DEF34FAA3509722B00393E740262CA5F@m-mail03.office.premiertech>
	<CAD-qRkjebD1YYrsZb7WkWw0ra+TfpjWNSyoTgavN=U_BhyMRXA@mail.gmail.com>
	<CAKq5mS4eqivAUw5QWVThUB+Xi6U8QJ8X==UfGi-1Lmmfpde+eA@mail.gmail.com>
Message-ID: <CAD-qRkgTiowDy_8JKpKj1cfTg2PyYvTNEZNnKBXAnER1xwq1Zw@mail.gmail.com>

Yes, this discussion is about doing ASR with Nuance MRCP.  The FreeSWITCH
core does not care about which ASR engine is used, so these solutions are
already pretty generic.  However, the inputs and outputs are dependent on
which ASR engine is used.  Nuance MRCP will use SRGS for grammar definition
and NLSML for the response.


On Wed, Sep 7, 2011 at 2:37 PM, envelopes envelopes <sunwood360 at gmail.com>wrote:

> Does your app support MRCP protocol? That would make it more generic.
> On Sep 7, 2011 10:33 AM, "Christopher Rienzo" <cmrienzo at gmail.com> wrote:
> > There's not really an easy way to do this out of the box. The FS core
> > provides all the low-level functions necessary for doing ASR, but nothing
> > higher level for interpreting results, handling barge-in, starting
> > input-timers, etc.
> >
> > I solved this issue with a custom dialplan APP. I know of another other
> dev
> > that used event socket and the playback/detect_speech dialplan APPs.
> > Looking at the scripts/javascript/js_modules/SpeechTools.jm source, you
> can
> > see that anthm solved it a different way.
> >
> >
> >
> > On Mon, Sep 5, 2011 at 3:02 AM, Glen Ganderton <
> > Glen.Ganderton at premier.com.au> wrote:
> >
> >> Hi Guys, ****
> >>
> >> ** **
> >>
> >> What i am trying to do is configure an IVR using freeswitch and Nuance
> >> 5.1.5 to perform speech recognition. I have installed the latest****
> >>
> >> release of freeswitch on my CentOS system, I have also enabled and
> >> configured the unimrcp module. I could find lots of information to****
> >>
> >> configure the unimrcp module but there is no documentation on how to
> create
> >> IVR menu's using this module with Nuance. I have used the ****
> >>
> >> demo pizza IVR with the pocketsphinx module and I keep getting refered
> to
> >> this but it's no help as I want to use Nuance for ASR. Any****
> >>
> >> help on how to configure this would be great, specifically:****
> >>
> >> ** **
> >>
> >> * How to I call the IVR script from the dialplan****
> >>
> >> * What language/s can I code the IVR in and can you please provide a
> basic
> >> sample of how this is done with Nuance.****
> >>
> >> ** **
> >>
> >> Thank you in advance****
> >>
> >> ** **
> >>
> >> --Glen****
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/2a494d45/attachment.html 

From brian.wiese.freeswitch at gmail.com  Thu Sep  8 00:23:24 2011
From: brian.wiese.freeswitch at gmail.com (Brian Wiese)
Date: Wed, 7 Sep 2011 15:23:24 -0500
Subject: [Freeswitch-users] One-way audio issue with attended transfer
In-Reply-To: <CADEQS5zejaoLBmyM3UreaTZKbJLHRjjSxiPJrr9zfWzN8wMhfA@mail.gmail.com>
References: <SNT116-W212CCAFBD521CA3F341979D1F0@phx.gbl>
	<CADEQS5zejaoLBmyM3UreaTZKbJLHRjjSxiPJrr9zfWzN8wMhfA@mail.gmail.com>
Message-ID: <CADEQS5y1WMD06DNKrEykw+pxtep07Kf-trWF_ZNA7e2bHCwtuQ@mail.gmail.com>

John:

I just tried to transfer a call blindly as well and I had the same problem
as an attended transfer.  Just trying to keep everyone updated here on the
latest...

~Brian

On Wed, Sep 7, 2011 at 12:50 PM, Brian Wiese <
brian.wiese.freeswitch at gmail.com> wrote:

> John:
>
> While I don't have the answer, I did want to write to say that I'm
> experiencing the same problem.  We just built this server about a week ago,
> and I'm running this version:  FreeSWITCH Version 1.0.head (git-a966c2b
> 2011-09-04 11-09-23 -0500)
>
> If I can figure out what's going on I'll certainly post back here!
>
> ~Brian
>
>   On Tue, Sep 6, 2011 at 11:01 PM, John Platts <john_platts at hotmail.com>wrote:
>
>>
>> We have been experiencing a one-way audio issue with attended transfer.
>> The problem is that after an call is transferred through an attended
>> transfer, the caller cannot be heard, but the caller can hear the voice of
>> the person that the call is transferred to.
>>
>> Steps to reproduce the problem:
>> 1. Dial into FreeSWITCH from an external phone.
>> 2. Answer the call. One-way audio is not experienced here.
>> 3. Press the Transfer button on the IP phone to initiate an attended
>> transfer.
>> 4. Dial the extension that the call is to be transferred to.
>> 5. Allow the extension that the call is to be transferred to pick up the
>> call. One-way audio is not experienced here.
>> 6. Press the Transfer button to complete the transfer. The caller does
>> hear audio from the person that the call is transferred to, but the person
>> that the call is transferred to cannot hear the caller. The one way audio
>> issue is occurring here.
>>
>> The attended transfer was performed from an extension on one SPA303 phone
>> to an extension on another SPA303 phone. FreeSWITCH actually detects that
>> the transfer is an attended transfer.
>>
>> Here are attachments that I have uploaded to pastebin:
>> conf/dialplan/features.xml file: http://pastebin.freeswitch.org/17291
>> FreeSWITCH log, including the attended transfer:
>> http://pastebin.freeswitch.org/17286
>>
>> How do we get this one-way audio issue fixed?
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/fefce7f3/attachment.html 

From wayne at hamilton.net  Thu Sep  8 00:47:39 2011
From: wayne at hamilton.net (Wayne)
Date: Wed, 7 Sep 2011 15:47:39 -0500
Subject: [Freeswitch-users] Sending DTMF on B-leg
Message-ID: <2934141FC0D9453B9150F7BD307120F3@ccs.local>

Hello All,

I need to call out on a SIP trunk and when the call is answered send DTMF
tones. 
I need to send the DTMF only on the outbound leg.

Does anyone have a dialplan that will do that? Is it possible. I have only
found one thread on it and did get much out of it.

Thanks
Wayne



From avi at avimarcus.net  Thu Sep  8 00:55:08 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 7 Sep 2011 23:55:08 +0300
Subject: [Freeswitch-users] Sending DTMF on B-leg
In-Reply-To: <2934141FC0D9453B9150F7BD307120F3@ccs.local>
References: <2934141FC0D9453B9150F7BD307120F3@ccs.local>
Message-ID: <CALNZuwkMZoRx-HbRBkHg5FSutwV2xW2GWiQQEz2dsmAh2WoPhA@mail.gmail.com>

I think you looking for queue_dtmf.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_queue_dtmf

-Avi Marcus



On Wed, Sep 7, 2011 at 11:47 PM, Wayne <wayne at hamilton.net> wrote:
> Hello All,
>
> I need to call out on a SIP trunk and when the call is answered send DTMF
> tones.
> I need to send the DTMF only on the outbound leg.
>
> Does anyone have a dialplan that will do that? Is it possible. I have only
> found one thread on it and did get much out of it.
>
> Thanks
> Wayne
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/7bbec25a/attachment-0001.html 

From brian.wiese.freeswitch at gmail.com  Thu Sep  8 01:19:03 2011
From: brian.wiese.freeswitch at gmail.com (Brian Wiese)
Date: Wed, 7 Sep 2011 16:19:03 -0500
Subject: [Freeswitch-users] Multiple DTMF on Single Keypress
In-Reply-To: <CALNZuw=q2Up_2EpAkQ-5Pvfu_zZGWUsHUeK=iimhK+ueyyfSjg@mail.gmail.com>
References: <CADEQS5zp8bh3epB3-ya9uR2HvNjnt8qf7u4rS=a11tNNGGo7hA@mail.gmail.com>
	<CA+Gmxbbm46vCtW70+Qt7LurJEybMHwDA6BA+tNa8OnL4U0uKfw@mail.gmail.com>
	<CAFKNP3-Z6Zhas8DZobHqw4fxampyYjwqMU=guzG1jvZvHuWWFA@mail.gmail.com>
	<CALNZuwk372OUr1cNdqGrPqmBdFPEf6EUSBuPzhqBmBu1Bczvxg@mail.gmail.com>
	<CADEQS5xsRouLa1fxGu9+=sUJdagO4cW6q0AfsoPKwDghKMvSCw@mail.gmail.com>
	<CALNZuw=q2Up_2EpAkQ-5Pvfu_zZGWUsHUeK=iimhK+ueyyfSjg@mail.gmail.com>
Message-ID: <CADEQS5zqoKocYUC=DS_wN_PAEuF4kJofi9+RmSQv5mzd_U+s9A@mail.gmail.com>

Avi:

Thank you for your help on this.

I've captured the traffic as you've requested and another log.  I made it to
the directory (DTMF 9 in the IVR), but then when I tried to dial 94373 you
can see I had some duplicate DTMF.
http://www.netwayaccess.com/pcapsipdump.zip
~Brian
On Wed, Sep 7, 2011 at 6:18 AM, Avi Marcus <avi at avimarcus.net> wrote:

> Can you get a normal PCAP of the SIP/RTP with something here?
> http://wiki.freeswitch.org/wiki/Packet_Capture
>
> e.g. pcapsipdump is quite nice. (Just make sure the folder exists
> before running the command.)
>
> -Avi
>
>
> On Wed, Sep 7, 2011 at 2:13 AM, Brian Wiese
> <brian.wiese.freeswitch at gmail.com> wrote:
> > Avi:
> >
> > I had thought it was inband, but I couldn't find anything that supported
> it,
> > like you mentioned.
> >
> > Is there anything else I can provide that would help solve this problem?
> >
> > Thanks.
> >
> > ~Brian
> >
> > On Tue, Sep 6, 2011 at 3:48 PM, Avi Marcus <avi at avimarcus.net> wrote:
> >>
> >> line 389
> >> 2011-09-04 16:20:14.390350 [DEBUG] switch_rtp.c:3317 RTP RECV
> DTMF 5:1440
> >> 2011-09-04 16:20:14.390350 [DEBUG] switch_ivr_bridge.c:391 Send signal
> >> sofia/internal/sip:20005 at 172.31.6.253 [BREAK]
> >> 2011-09-04 16:20:14.410350 [DEBUG] switch_rtp.c:2343 Send start packet
> >> for [5] ts=97600 dur=160/160/1440 seq=29252
> >> 2011-09-04 16:20:14.410350 [DEBUG] switch_rtp.c:3317 RTP RECV
> DTMF 5:1440
> >>
> >> It's coming as RTP RECV, meaning rfc2833. Otherwise, it would say
> >> "DETECTED".
> >>
> >> -Avi
> >>
> >>
> >> On Tue, Sep 6, 2011 at 11:36 PM, Jon Young <jonyoung111 at gmail.com>
> wrote:
> >> >
> >> > Is it possible you are receiving 2833 and Inband DTMF?
> >> >
> >> > On Mon, Sep 5, 2011 at 10:36 PM, Dmitry Sytchev <kbdfck at gmail.com>
> >> > wrote:
> >> > > See the same behaviour with inband DTMF detector sometimes.
> >> > >
> >> > > 2011/9/5 Brian Wiese FreeSWITCH List
> >> > > <brian.wiese.freeswitch at gmail.com>
> >> > >>
> >> > >> Hello everyone!
> >> > >>
> >> > >> I'm getting multiple RTP DTMF from random keypresses and I can't
> >> > >> figure
> >> > >> out why.  I've PB'ed the packet capture and FS log for a call.  As
> >> > >> you can
> >> > >> see from the FS log, the 9,8,7,6 numbers weren't repeated, but 5
> was
> >> > >> (again,
> >> > >> though, different calls lead to different numbers being repeated).
> >> > >> Log:  http://pastebin.freeswitch.org/17280
> >> > >> Capture:  http://pastebin.freeswitch.org/17282
> >> > >>
> >> > >> I appreciate any ideas as to what I might have wrong here.
> >> > >>
> >> > >> 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
> >> > >>
> >> > >
> >> > >
> >> > >
> >> > > --
> >> > > 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
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/46c9bbab/attachment.html 

From avi at avimarcus.net  Thu Sep  8 01:37:01 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 8 Sep 2011 00:37:01 +0300
Subject: [Freeswitch-users] Multiple DTMF on Single Keypress
In-Reply-To: <CADEQS5zqoKocYUC=DS_wN_PAEuF4kJofi9+RmSQv5mzd_U+s9A@mail.gmail.com>
References: <CADEQS5zp8bh3epB3-ya9uR2HvNjnt8qf7u4rS=a11tNNGGo7hA@mail.gmail.com>
	<CA+Gmxbbm46vCtW70+Qt7LurJEybMHwDA6BA+tNa8OnL4U0uKfw@mail.gmail.com>
	<CAFKNP3-Z6Zhas8DZobHqw4fxampyYjwqMU=guzG1jvZvHuWWFA@mail.gmail.com>
	<CALNZuwk372OUr1cNdqGrPqmBdFPEf6EUSBuPzhqBmBu1Bczvxg@mail.gmail.com>
	<CADEQS5xsRouLa1fxGu9+=sUJdagO4cW6q0AfsoPKwDghKMvSCw@mail.gmail.com>
	<CALNZuw=q2Up_2EpAkQ-5Pvfu_zZGWUsHUeK=iimhK+ueyyfSjg@mail.gmail.com>
	<CADEQS5zqoKocYUC=DS_wN_PAEuF4kJofi9+RmSQv5mzd_U+s9A@mail.gmail.com>
Message-ID: <CALNZuwm3GVLqxTLSquzkrsvZcQoq=gB4btvCBT1uhBH7K75ryQ@mail.gmail.com>

That pcap only shows 9[pause]943 so it's not the whole thing? But anyway,
it's coming in as rfc2833, so it's really unlikely FS is mis-reading it..
Can you get a full pcap? You can open them in wireshark and filter for
"rtpevent" to see the dtmf digits that come in.

-Avi


On Thu, Sep 8, 2011 at 12:19 AM, Brian Wiese <
brian.wiese.freeswitch at gmail.com> wrote:

> Avi:
>
> Thank you for your help on this.
>
> I've captured the traffic as you've requested and another log.  I made it
> to the directory (DTMF 9 in the IVR), but then when I tried to dial 94373
> you can see I had some duplicate DTMF.
> http://www.netwayaccess.com/pcapsipdump.zip
> ~Brian
> On Wed, Sep 7, 2011 at 6:18 AM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> Can you get a normal PCAP of the SIP/RTP with something here?
>> http://wiki.freeswitch.org/wiki/Packet_Capture
>>
>> e.g. pcapsipdump is quite nice. (Just make sure the folder exists
>> before running the command.)
>>
>> -Avi
>>
>>
>> On Wed, Sep 7, 2011 at 2:13 AM, Brian Wiese
>> <brian.wiese.freeswitch at gmail.com> wrote:
>> > Avi:
>> >
>> > I had thought it was inband, but I couldn't find anything that supported
>> it,
>> > like you mentioned.
>> >
>> > Is there anything else I can provide that would help solve this problem?
>> >
>> > Thanks.
>> >
>> > ~Brian
>> >
>> > On Tue, Sep 6, 2011 at 3:48 PM, Avi Marcus <avi at avimarcus.net> wrote:
>> >>
>> >> line 389
>> >> 2011-09-04 16:20:14.390350 [DEBUG] switch_rtp.c:3317 RTP RECV
>> DTMF 5:1440
>> >> 2011-09-04 16:20:14.390350 [DEBUG] switch_ivr_bridge.c:391 Send signal
>> >> sofia/internal/sip:20005 at 172.31.6.253 [BREAK]
>> >> 2011-09-04 16:20:14.410350 [DEBUG] switch_rtp.c:2343 Send start packet
>> >> for [5] ts=97600 dur=160/160/1440 seq=29252
>> >> 2011-09-04 16:20:14.410350 [DEBUG] switch_rtp.c:3317 RTP RECV
>> DTMF 5:1440
>> >>
>> >> It's coming as RTP RECV, meaning rfc2833. Otherwise, it would say
>> >> "DETECTED".
>> >>
>> >> -Avi
>> >>
>> >>
>> >> On Tue, Sep 6, 2011 at 11:36 PM, Jon Young <jonyoung111 at gmail.com>
>> wrote:
>> >> >
>> >> > Is it possible you are receiving 2833 and Inband DTMF?
>> >> >
>> >> > On Mon, Sep 5, 2011 at 10:36 PM, Dmitry Sytchev <kbdfck at gmail.com>
>> >> > wrote:
>> >> > > See the same behaviour with inband DTMF detector sometimes.
>> >> > >
>> >> > > 2011/9/5 Brian Wiese FreeSWITCH List
>> >> > > <brian.wiese.freeswitch at gmail.com>
>> >> > >>
>> >> > >> Hello everyone!
>> >> > >>
>> >> > >> I'm getting multiple RTP DTMF from random keypresses and I can't
>> >> > >> figure
>> >> > >> out why.  I've PB'ed the packet capture and FS log for a call.  As
>> >> > >> you can
>> >> > >> see from the FS log, the 9,8,7,6 numbers weren't repeated, but 5
>> was
>> >> > >> (again,
>> >> > >> though, different calls lead to different numbers being repeated).
>> >> > >> Log:  http://pastebin.freeswitch.org/17280
>> >> > >> Capture:  http://pastebin.freeswitch.org/17282
>> >> > >>
>> >> > >> I appreciate any ideas as to what I might have wrong here.
>> >> > >>
>> >> > >> 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
>> >> > >>
>> >> > >
>> >> > >
>> >> > >
>> >> > > --
>> >> > > 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
>> >>
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/7ea9b0be/attachment-0001.html 

From brian.wiese.freeswitch at gmail.com  Thu Sep  8 01:51:01 2011
From: brian.wiese.freeswitch at gmail.com (Brian Wiese)
Date: Wed, 7 Sep 2011 16:51:01 -0500
Subject: [Freeswitch-users] Multiple DTMF on Single Keypress
In-Reply-To: <CALNZuwm3GVLqxTLSquzkrsvZcQoq=gB4btvCBT1uhBH7K75ryQ@mail.gmail.com>
References: <CADEQS5zp8bh3epB3-ya9uR2HvNjnt8qf7u4rS=a11tNNGGo7hA@mail.gmail.com>
	<CA+Gmxbbm46vCtW70+Qt7LurJEybMHwDA6BA+tNa8OnL4U0uKfw@mail.gmail.com>
	<CAFKNP3-Z6Zhas8DZobHqw4fxampyYjwqMU=guzG1jvZvHuWWFA@mail.gmail.com>
	<CALNZuwk372OUr1cNdqGrPqmBdFPEf6EUSBuPzhqBmBu1Bczvxg@mail.gmail.com>
	<CADEQS5xsRouLa1fxGu9+=sUJdagO4cW6q0AfsoPKwDghKMvSCw@mail.gmail.com>
	<CALNZuw=q2Up_2EpAkQ-5Pvfu_zZGWUsHUeK=iimhK+ueyyfSjg@mail.gmail.com>
	<CADEQS5zqoKocYUC=DS_wN_PAEuF4kJofi9+RmSQv5mzd_U+s9A@mail.gmail.com>
	<CALNZuwm3GVLqxTLSquzkrsvZcQoq=gB4btvCBT1uhBH7K75ryQ@mail.gmail.com>
Message-ID: <CADEQS5yAHOFO=2oP5poWxZBNQE-gi3USJon8Oq+GN-LWD4_CUQ@mail.gmail.com>

Avi:

Here's a new full tcpdump and log:
http://www.netwayaccess.com/newcapture.zip

Like you said, I filtered it by rtpevent and I only see one DTMF 4, but FS
read two.

~Brian

On Wed, Sep 7, 2011 at 4:37 PM, Avi Marcus <avi at avimarcus.net> wrote:

> That pcap only shows 9[pause]943 so it's not the whole thing? But anyway,
> it's coming in as rfc2833, so it's really unlikely FS is mis-reading it..
> Can you get a full pcap? You can open them in wireshark and filter for
> "rtpevent" to see the dtmf digits that come in.
>
> -Avi
>
>
> On Thu, Sep 8, 2011 at 12:19 AM, Brian Wiese <
> brian.wiese.freeswitch at gmail.com> wrote:
>
>> Avi:
>>
>> Thank you for your help on this.
>>
>> I've captured the traffic as you've requested and another log.  I made it
>> to the directory (DTMF 9 in the IVR), but then when I tried to dial 94373
>> you can see I had some duplicate DTMF.
>> http://www.netwayaccess.com/pcapsipdump.zip
>> ~Brian
>>   On Wed, Sep 7, 2011 at 6:18 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> Can you get a normal PCAP of the SIP/RTP with something here?
>>> http://wiki.freeswitch.org/wiki/Packet_Capture
>>>
>>> e.g. pcapsipdump is quite nice. (Just make sure the folder exists
>>> before running the command.)
>>>
>>> -Avi
>>>
>>>
>>> On Wed, Sep 7, 2011 at 2:13 AM, Brian Wiese
>>> <brian.wiese.freeswitch at gmail.com> wrote:
>>> > Avi:
>>> >
>>> > I had thought it was inband, but I couldn't find anything that
>>> supported it,
>>> > like you mentioned.
>>> >
>>> > Is there anything else I can provide that would help solve this
>>> problem?
>>> >
>>> > Thanks.
>>> >
>>> > ~Brian
>>> >
>>> > On Tue, Sep 6, 2011 at 3:48 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>> >>
>>> >> line 389
>>> >> 2011-09-04 16:20:14.390350 [DEBUG] switch_rtp.c:3317 RTP RECV
>>> DTMF 5:1440
>>> >> 2011-09-04 16:20:14.390350 [DEBUG] switch_ivr_bridge.c:391 Send signal
>>> >> sofia/internal/sip:20005 at 172.31.6.253 [BREAK]
>>> >> 2011-09-04 16:20:14.410350 [DEBUG] switch_rtp.c:2343 Send start packet
>>> >> for [5] ts=97600 dur=160/160/1440 seq=29252
>>> >> 2011-09-04 16:20:14.410350 [DEBUG] switch_rtp.c:3317 RTP RECV
>>> DTMF 5:1440
>>> >>
>>> >> It's coming as RTP RECV, meaning rfc2833. Otherwise, it would say
>>> >> "DETECTED".
>>> >>
>>> >> -Avi
>>> >>
>>> >>
>>> >> On Tue, Sep 6, 2011 at 11:36 PM, Jon Young <jonyoung111 at gmail.com>
>>> wrote:
>>> >> >
>>> >> > Is it possible you are receiving 2833 and Inband DTMF?
>>> >> >
>>> >> > On Mon, Sep 5, 2011 at 10:36 PM, Dmitry Sytchev <kbdfck at gmail.com>
>>> >> > wrote:
>>> >> > > See the same behaviour with inband DTMF detector sometimes.
>>> >> > >
>>> >> > > 2011/9/5 Brian Wiese FreeSWITCH List
>>> >> > > <brian.wiese.freeswitch at gmail.com>
>>> >> > >>
>>> >> > >> Hello everyone!
>>> >> > >>
>>> >> > >> I'm getting multiple RTP DTMF from random keypresses and I can't
>>> >> > >> figure
>>> >> > >> out why.  I've PB'ed the packet capture and FS log for a call.
>>> As
>>> >> > >> you can
>>> >> > >> see from the FS log, the 9,8,7,6 numbers weren't repeated, but 5
>>> was
>>> >> > >> (again,
>>> >> > >> though, different calls lead to different numbers being
>>> repeated).
>>> >> > >> Log:  http://pastebin.freeswitch.org/17280
>>> >> > >> Capture:  http://pastebin.freeswitch.org/17282
>>> >> > >>
>>> >> > >> I appreciate any ideas as to what I might have wrong here.
>>> >> > >>
>>> >> > >> 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
>>> >> > >>
>>> >> > >
>>> >> > >
>>> >> > >
>>> >> > > --
>>> >> > > 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
>>> >>
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/2d8f3163/attachment.html 

From michaelt at callall.co.za  Thu Sep  8 00:33:10 2011
From: michaelt at callall.co.za (Michael Toop)
Date: Wed, 7 Sep 2011 22:33:10 +0200
Subject: [Freeswitch-users] Bug or Not? "longjmp causes uninitialized
 stack frame"
In-Reply-To: <CAKbxfG87gWRZeM+PEmeM=MfoyrgofsnUSDY+Mu7eBnO2Kki1Ow@mail.gmail.com>
References: <CAFmdODjqu2dVBXoCZbJmn+WftypQPqJCw=nS4adG4QMeg17-sA@mail.gmail.com>
	<CAKzWOxVvpDfY=N16BTAtsFqE4aU4sqfvxsRsSmirUCx-HGTLFg@mail.gmail.com>
	<CAKbxfG87gWRZeM+PEmeM=MfoyrgofsnUSDY+Mu7eBnO2Kki1Ow@mail.gmail.com>
Message-ID: <CAFmdODhUR2JnuBKA7xM4ppK13LphAxFpJZbetFQoECg91M6TEA@mail.gmail.com>

Hi Anthony,

 Thanks, that fixed it, but wow that is scary, how can such a hairy bug get
into libcurl.  Warning to everyone using Ubuntu 11.04 and libcurl.

 Suppose I should log this with the Ubuntu team.

Thanks again,

Michael

On Wed, Sep 7, 2011 at 7:14 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> the latest libcurl on most systems is buggy, this is why we package
> our own version of libs.
>
> reconfigure your build with --without-libcurl to use our version.
>
>
> On Wed, Sep 7, 2011 at 12:04 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > Definitely put this one on Jira.
> > -MC
> >
> > On Mon, Sep 5, 2011 at 1:39 AM, Michael Toop <michaelt at callall.co.za>
> wrote:
> >>
> >> Hi,
> >>  Sorry not sure what to do with this, if this is a bug or another
> problem,
> >> but getting this message in the bt when running FS as it does a core
> dump:
> >> " longjmp causes uninitialized stack frame".
> >>  Running on: 2.6.38-11-server #48-Ubuntu, Ubuntu 11.04 on the latest GIT
> >> release.
> >> Thanks,
> >> Michael
> >> 2011-09-04 15:34:59.149089 [INFO] mod_native_file.c:94 Opening File
> >> [/usr/local/freeswitch/sounds/custom/mwc-enter-destination.G729] 8000hz
> >> 2011-09-04 15:34:59.169011 [INFO] sofia.c:755
> >> sofia/cellc/763070366 at 172.103.0.36 Update Callee ID to "763070366"
> >> <763070366>
> >> 2011-09-04 15:34:59.169011 [NOTICE] switch_core_session.c:1367 Session
> 946
> >> (sofia/cellc/790938072 at 172.103.0.36) Ended
> >> 2011-09-04 15:34:59.169011 [NOTICE] switch_core_session.c:1369 Close
> >> Channel sofia/cellc/790938072 at 172.103.0.36 [CS_DESTROY]
> >> *** longjmp causes uninitialized stack frame ***: freeswitch terminated
> >> ======= Backtrace: =========
> >> /lib/x86_64-linux-gnu/libc.so.6(__fortify_fail+0x37)[0x7f1157fbe1d7]
> >> /lib/x86_64-linux-gnu/libc.so.6(+0xfe169)[0x7f1157fbe169]
> >> /lib/x86_64-linux-gnu/libc.so.6(__longjmp_chk+0x33)[0x7f1157fbe0d3]
> >> /usr/lib/libcurl.so.4(+0xd165)[0x7f1156457165]
> >> /lib/x86_64-linux-gnu/libpthread.so.0(+0xfc60)[0x7f1158a89c60]
> >> /lib/x86_64-linux-gnu/libc.so.6(__select+0x33)[0x7f1157f9e143]
> >>
> >>
> /usr/local/freeswitch/lib/libfreeswitch.so.1(apr_sleep+0x45)[0x7f11593d16c5]
> >> /usr/local/freeswitch/lib/libfreeswitch.so.1(+0xcab5c)[0x7f11593a7b5c]
> >>
> >>
> /usr/local/freeswitch/lib/libfreeswitch.so.1(switch_console_loop+0x74a)[0x7f1159332e9a]
> >> freeswitch[0x402b22]
> >> /lib/x86_64-linux-gnu/libc.so.6(__libc_start_main+0xff)[0x7f1157edeeff]
> >> freeswitch[0x401769]
> >> ======= Memory map: ========
> >> 00400000-00404000 r-xp 00000000 08:01 1845708
> >>  /usr/local/freeswitch/bin/freeswitch
> >> 00603000-00604000 r--p 00003000 08:01 1845708
> >>  /usr/local/freeswitch/bin/freeswitch
> >> 00604000-00605000 rw-p 00004000 08:01 1845708
> >>  /usr/local/freeswitch/bin/freeswitch
> >> 017b0000-026c9000 rw-p 00000000 00:00 0
> >>  [heap]
> >> 7f114004d000-7f114004e000 ---p 00000000 00:00 0
> >> 7f114004e000-7f1140089000 rw-p 00000000 00:00 0
> >> 7f1140089000-7f114008a000 ---p 00000000 00:00 0
> >> 7f114008a000-7f11400c5000 rw-p 00000000 00:00 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
> >
> >
>
>
>
> --
> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/6fbfaff4/attachment-0001.html 

From wayne at hamilton.net  Thu Sep  8 02:17:04 2011
From: wayne at hamilton.net (Wayne)
Date: Wed, 7 Sep 2011 17:17:04 -0500
Subject: [Freeswitch-users] Sending DTMF on B-leg
In-Reply-To: <CALNZuwkMZoRx-HbRBkHg5FSutwV2xW2GWiQQEz2dsmAh2WoPhA@mail.gmail.com>
References: <2934141FC0D9453B9150F7BD307120F3@ccs.local>
	<CALNZuwkMZoRx-HbRBkHg5FSutwV2xW2GWiQQEz2dsmAh2WoPhA@mail.gmail.com>
Message-ID: <9B18FF40055B495BBAE3EDEEDB7388B0@ccs.local>

I have tried that 
<extension name="test-tone" >
   <condition field="destination_number" expression="(9025556747)" >
       <action application="set" data="call_direction=outbound" />
       <action application="set"
data="effective_caller_id_name=${outbound_caller_id_name}" />
       <action application="set"
data="effective_caller_id_number=${outbound_caller_id_number}" />
       <action application="set" data="continue_on_fail=true" />
       <action application="queue_dtmf" data="0123456789" />
       <action application="bridge" data="freetdm/wp1/a/$1" />
   </condition>
</extension>
 
The debug tells me that this
tail -f freeswitch.log | grep -i dtmf
 
Dialplan:  <mailto:sofia/internal/1333 at 192.168.48.87>
sofia/internal/1333 at 192.168.48.87 Action queue_dtmf(0123456789)
EXECUTE  <mailto:sofia/internal/1333 at 192.168.48.87>
sofia/internal/1333 at 192.168.48.87 queue_dtmf(0123456789)
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
<mailto:sofia/internal/1333 at 192.168.48.87> sofia/internal/1333 at 192.168.48.87
Queue dtmf
2011-09-07 17:13:45.076218 [DEBUG] sofia_glue.c:3355 Set 2833 dtmf send
payload to 101
2011-09-07 17:13:45.076218 [DEBUG] sofia_glue.c:3360 Set 2833 dtmf receive
payload to 101
2011-09-07 17:13:45.116216 [DEBUG] ftdm_io.c:3714 [s1c1][1:1] Generating
DTMF [0123456789]
2011-09-07 17:13:47.876238 [DEBUG] ftmod_wanpipe.c:701 [s1c1][1:1] Enabled
DTMF events
2011-09-07 17:13:47.956228 [DEBUG] ftmod_wanpipe.c:1415 [s1c1][1:1] Queuing
wanpipe DTMF: 9
2011-09-07 17:13:47.956228 [DEBUG] ftdm_io.c:3524 [s1c1][1:1] Queuing DTMF 9
(debug = 0)
2011-09-07 17:13:47.956228 [DEBUG] mod_freetdm.c:733 Queuing DTMF [9] in
channel FreeTDM/1:1/9025556747
2011-09-07 17:13:48.036235 [DEBUG] ftmod_wanpipe.c:1415 [s1c1][1:1] Queuing
wanpipe DTMF: 9
2011-09-07 17:13:48.036235 [DEBUG] ftdm_io.c:3524 [s1c1][1:1] Queuing DTMF 9
(debug = 0)
2011-09-07 17:13:48.036235 [DEBUG] mod_freetdm.c:733 Queuing DTMF [9] in
channel FreeTDM/1:1/9025556747
 
So where would I look next.
 
Wayne


  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Wednesday, September 07, 2011 3:55 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Sending DTMF on B-leg


I think you looking for queue_dtmf.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_queue_dtmf

-Avi Marcus



On Wed, Sep 7, 2011 at 11:47 PM, Wayne <wayne at hamilton.net> wrote:
> Hello All,
>
> I need to call out on a SIP trunk and when the call is answered send DTMF
> tones.
> I need to send the DTMF only on the outbound leg.
>
> Does anyone have a dialplan that will do that? Is it possible. I have only
> found one thread on it and did get much out of it.
>
> Thanks
> Wayne
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/8d0b71c3/attachment.html 

From anthony.minessale at gmail.com  Thu Sep  8 02:44:21 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 7 Sep 2011 17:44:21 -0500
Subject: [Freeswitch-users] One-way audio issue with attended transfer
In-Reply-To: <CADEQS5y1WMD06DNKrEykw+pxtep07Kf-trWF_ZNA7e2bHCwtuQ@mail.gmail.com>
References: <SNT116-W212CCAFBD521CA3F341979D1F0@phx.gbl>
	<CADEQS5zejaoLBmyM3UreaTZKbJLHRjjSxiPJrr9zfWzN8wMhfA@mail.gmail.com>
	<CADEQS5y1WMD06DNKrEykw+pxtep07Kf-trWF_ZNA7e2bHCwtuQ@mail.gmail.com>
Message-ID: <CAKbxfG9+picJxyXcBaUydy8x2_cc1JT_jrGaLJxxdXRPicZ0WQ@mail.gmail.com>

This issue should be fixed. Update to latest git head.
On Sep 7, 2011 3:24 PM, "Brian Wiese" <brian.wiese.freeswitch at gmail.com>
wrote:
> John:
>
> I just tried to transfer a call blindly as well and I had the same problem
> as an attended transfer. Just trying to keep everyone updated here on the
> latest...
>
> ~Brian
>
> On Wed, Sep 7, 2011 at 12:50 PM, Brian Wiese <
> brian.wiese.freeswitch at gmail.com> wrote:
>
>> John:
>>
>> While I don't have the answer, I did want to write to say that I'm
>> experiencing the same problem. We just built this server about a week
ago,
>> and I'm running this version: FreeSWITCH Version 1.0.head (git-a966c2b
>> 2011-09-04 11-09-23 -0500)
>>
>> If I can figure out what's going on I'll certainly post back here!
>>
>> ~Brian
>>
>> On Tue, Sep 6, 2011 at 11:01 PM, John Platts <john_platts at hotmail.com
>wrote:
>>
>>>
>>> We have been experiencing a one-way audio issue with attended transfer.
>>> The problem is that after an call is transferred through an attended
>>> transfer, the caller cannot be heard, but the caller can hear the voice
of
>>> the person that the call is transferred to.
>>>
>>> Steps to reproduce the problem:
>>> 1. Dial into FreeSWITCH from an external phone.
>>> 2. Answer the call. One-way audio is not experienced here.
>>> 3. Press the Transfer button on the IP phone to initiate an attended
>>> transfer.
>>> 4. Dial the extension that the call is to be transferred to.
>>> 5. Allow the extension that the call is to be transferred to pick up the
>>> call. One-way audio is not experienced here.
>>> 6. Press the Transfer button to complete the transfer. The caller does
>>> hear audio from the person that the call is transferred to, but the
person
>>> that the call is transferred to cannot hear the caller. The one way
audio
>>> issue is occurring here.
>>>
>>> The attended transfer was performed from an extension on one SPA303
phone
>>> to an extension on another SPA303 phone. FreeSWITCH actually detects
that
>>> the transfer is an attended transfer.
>>>
>>> Here are attachments that I have uploaded to pastebin:
>>> conf/dialplan/features.xml file: http://pastebin.freeswitch.org/17291
>>> FreeSWITCH log, including the attended transfer:
>>> http://pastebin.freeswitch.org/17286
>>>
>>> How do we get this one-way audio issue fixed?
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/ceba5db1/attachment.html 

From brian.wiese.freeswitch at gmail.com  Thu Sep  8 03:48:31 2011
From: brian.wiese.freeswitch at gmail.com (Brian Wiese)
Date: Wed, 7 Sep 2011 18:48:31 -0500
Subject: [Freeswitch-users] One-way audio issue with attended transfer
In-Reply-To: <CAKbxfG9+picJxyXcBaUydy8x2_cc1JT_jrGaLJxxdXRPicZ0WQ@mail.gmail.com>
References: <SNT116-W212CCAFBD521CA3F341979D1F0@phx.gbl>
	<CADEQS5zejaoLBmyM3UreaTZKbJLHRjjSxiPJrr9zfWzN8wMhfA@mail.gmail.com>
	<CADEQS5y1WMD06DNKrEykw+pxtep07Kf-trWF_ZNA7e2bHCwtuQ@mail.gmail.com>
	<CAKbxfG9+picJxyXcBaUydy8x2_cc1JT_jrGaLJxxdXRPicZ0WQ@mail.gmail.com>
Message-ID: <CADEQS5x669R-d7sss_A3DC0mBpVBL=UOChj-io_OfTidvBhw_g@mail.gmail.com>

Anthony:

This issue seems to be resolved!  Thank you!

~Brian

On Wed, Sep 7, 2011 at 5:44 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> This issue should be fixed. Update to latest git head.
>   On Sep 7, 2011 3:24 PM, "Brian Wiese" <brian.wiese.freeswitch at gmail.com>
> wrote:
> > John:
> >
> > I just tried to transfer a call blindly as well and I had the same
> problem
> > as an attended transfer. Just trying to keep everyone updated here on the
> > latest...
> >
> > ~Brian
> >
> > On Wed, Sep 7, 2011 at 12:50 PM, Brian Wiese <
> > brian.wiese.freeswitch at gmail.com> wrote:
> >
> >> John:
> >>
> >> While I don't have the answer, I did want to write to say that I'm
> >> experiencing the same problem. We just built this server about a week
> ago,
> >> and I'm running this version: FreeSWITCH Version 1.0.head (git-a966c2b
> >> 2011-09-04 11-09-23 -0500)
> >>
> >> If I can figure out what's going on I'll certainly post back here!
> >>
> >> ~Brian
> >>
> >> On Tue, Sep 6, 2011 at 11:01 PM, John Platts <john_platts at hotmail.com
> >wrote:
> >>
> >>>
> >>> We have been experiencing a one-way audio issue with attended transfer.
> >>> The problem is that after an call is transferred through an attended
> >>> transfer, the caller cannot be heard, but the caller can hear the voice
> of
> >>> the person that the call is transferred to.
> >>>
> >>> Steps to reproduce the problem:
> >>> 1. Dial into FreeSWITCH from an external phone.
> >>> 2. Answer the call. One-way audio is not experienced here.
> >>> 3. Press the Transfer button on the IP phone to initiate an attended
> >>> transfer.
> >>> 4. Dial the extension that the call is to be transferred to.
> >>> 5. Allow the extension that the call is to be transferred to pick up
> the
> >>> call. One-way audio is not experienced here.
> >>> 6. Press the Transfer button to complete the transfer. The caller does
> >>> hear audio from the person that the call is transferred to, but the
> person
> >>> that the call is transferred to cannot hear the caller. The one way
> audio
> >>> issue is occurring here.
> >>>
> >>> The attended transfer was performed from an extension on one SPA303
> phone
> >>> to an extension on another SPA303 phone. FreeSWITCH actually detects
> that
> >>> the transfer is an attended transfer.
> >>>
> >>> Here are attachments that I have uploaded to pastebin:
> >>> conf/dialplan/features.xml file: http://pastebin.freeswitch.org/17291
> >>> FreeSWITCH log, including the attended transfer:
> >>> http://pastebin.freeswitch.org/17286
> >>>
> >>> How do we get this one-way audio issue fixed?
> >>>
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/8092007c/attachment-0001.html 

From steveu at coppice.org  Thu Sep  8 04:16:07 2011
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 08 Sep 2011 08:16:07 +0800
Subject: [Freeswitch-users] STUN usage with Freeswitch
In-Reply-To: <2A0564E0-B521-4607-8BCB-1E417BBE6761@clickfono.com>
References: <2A0564E0-B521-4607-8BCB-1E417BBE6761@clickfono.com>
Message-ID: <4E680947.4010403@coppice.org>

On 09/08/2011 02:10 AM, Claudio ?lvarez wrote:
> Hi all,
>
> I currently operate a publicly accessible stun server, which one of my FreeSWITCH instances operating on Amazon EC2 queries for proper SIP and RTP configuration.
>
> When I query the stun server from FS-EC2, I get the following:
>
> freeswitch at internal>  stun 74.xx.xx.xx
> 184.xx.xx.xx:35087
>
> Can anyone clarify what does the 35087 number port stand for?
It stands for freedom, and the ability to communicate on the public 
internet. :-\

Any IP communication occurs through a 5-tuple path, between 2 endpoints. 
A source address + a source port (the first 2 elements) forms an 
endpoint. This talks through a protocol (the 3rd element) to a 
destination address + destination port (the last 2 elements), which form 
another endpoint. You asked the STUN server to tell you what your public 
endpoint is. It told you.

Steve



From sunwood360 at gmail.com  Thu Sep  8 05:18:02 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Wed, 7 Sep 2011 18:18:02 -0700
Subject: [Freeswitch-users] FreeSWITCH, Nuance, ASR: IVR Integration
In-Reply-To: <CAD-qRkgTiowDy_8JKpKj1cfTg2PyYvTNEZNnKBXAnER1xwq1Zw@mail.gmail.com>
References: <CD5A1499C8DEF34FAA3509722B00393E740262CA5F@m-mail03.office.premiertech>
	<CAD-qRkjebD1YYrsZb7WkWw0ra+TfpjWNSyoTgavN=U_BhyMRXA@mail.gmail.com>
	<CAKq5mS4eqivAUw5QWVThUB+Xi6U8QJ8X==UfGi-1Lmmfpde+eA@mail.gmail.com>
	<CAD-qRkgTiowDy_8JKpKj1cfTg2PyYvTNEZNnKBXAnER1xwq1Zw@mail.gmail.com>
Message-ID: <CAKq5mS5zx6g324z04Z93S__B-gUJb2jmQM=5PFkMWMLcAYg2Zg@mail.gmail.com>

Do you mind open source it? Thanks.
On Sep 7, 2011 1:02 PM, "Christopher Rienzo" <cmrienzo at gmail.com> wrote:
> Yes, this discussion is about doing ASR with Nuance MRCP. The FreeSWITCH
> core does not care about which ASR engine is used, so these solutions are
> already pretty generic. However, the inputs and outputs are dependent on
> which ASR engine is used. Nuance MRCP will use SRGS for grammar definition
> and NLSML for the response.
>
>
> On Wed, Sep 7, 2011 at 2:37 PM, envelopes envelopes <sunwood360 at gmail.com
>wrote:
>
>> Does your app support MRCP protocol? That would make it more generic.
>> On Sep 7, 2011 10:33 AM, "Christopher Rienzo" <cmrienzo at gmail.com> wrote:
>> > There's not really an easy way to do this out of the box. The FS core
>> > provides all the low-level functions necessary for doing ASR, but
nothing
>> > higher level for interpreting results, handling barge-in, starting
>> > input-timers, etc.
>> >
>> > I solved this issue with a custom dialplan APP. I know of another other
>> dev
>> > that used event socket and the playback/detect_speech dialplan APPs.
>> > Looking at the scripts/javascript/js_modules/SpeechTools.jm source, you
>> can
>> > see that anthm solved it a different way.
>> >
>> >
>> >
>> > On Mon, Sep 5, 2011 at 3:02 AM, Glen Ganderton <
>> > Glen.Ganderton at premier.com.au> wrote:
>> >
>> >> Hi Guys, ****
>> >>
>> >> ** **
>> >>
>> >> What i am trying to do is configure an IVR using freeswitch and Nuance
>> >> 5.1.5 to perform speech recognition. I have installed the latest****
>> >>
>> >> release of freeswitch on my CentOS system, I have also enabled and
>> >> configured the unimrcp module. I could find lots of information to****
>> >>
>> >> configure the unimrcp module but there is no documentation on how to
>> create
>> >> IVR menu's using this module with Nuance. I have used the ****
>> >>
>> >> demo pizza IVR with the pocketsphinx module and I keep getting refered
>> to
>> >> this but it's no help as I want to use Nuance for ASR. Any****
>> >>
>> >> help on how to configure this would be great, specifically:****
>> >>
>> >> ** **
>> >>
>> >> * How to I call the IVR script from the dialplan****
>> >>
>> >> * What language/s can I code the IVR in and can you please provide a
>> basic
>> >> sample of how this is done with Nuance.****
>> >>
>> >> ** **
>> >>
>> >> Thank you in advance****
>> >>
>> >> ** **
>> >>
>> >> --Glen****
>> >>
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110907/c4c11c29/attachment.html 

From avi at avimarcus.net  Thu Sep  8 09:26:28 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 8 Sep 2011 08:26:28 +0300
Subject: [Freeswitch-users] How to use sofia_count_reg ?
Message-ID: <CALNZuw=B3YYD1eWEMKERWtpitQBDv-AJKgw03k4mGCdZDfOQtA@mail.gmail.com>

I'm trying to use the API sofia_count_reg but all I'm getting is an error or
-1.
If I do "sofia_count_reg 1000" - whether there's multiple bindings or the
user doesn't exist, I get -1.
When I try "sofia_count_reg 1000 at domain.com" I get an sql error:

2011-09-08 08:22:29.381355 [ERR] switch_core_sqldb.c:825 ERR: [select
count(*) from sip_registrations where (sip_user='' or dir_user='1102') and
(sip_host='1102' or presence_hosts like '%sip.getbestfone.com%')]
[STATE: 42S22 CODE 1054 ERROR: [unixODBC][MySQL][ODBC 3.51
Driver][mysqld-5.1.41-3ubuntu12.10-log]Unknown column 'dir_user' in 'where
clause'

Thanks!

-Avi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/ee4a3c26/attachment.html 

From michal.bielicki at seventhsignal.de  Thu Sep  8 10:21:03 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Thu, 8 Sep 2011 08:21:03 +0200
Subject: [Freeswitch-users] FreeSWITCH, Nuance, ASR: IVR Integration
In-Reply-To: <CAKq5mS5zx6g324z04Z93S__B-gUJb2jmQM=5PFkMWMLcAYg2Zg@mail.gmail.com>
References: <CD5A1499C8DEF34FAA3509722B00393E740262CA5F@m-mail03.office.premiertech>
	<CAD-qRkjebD1YYrsZb7WkWw0ra+TfpjWNSyoTgavN=U_BhyMRXA@mail.gmail.com>
	<CAKq5mS4eqivAUw5QWVThUB+Xi6U8QJ8X==UfGi-1Lmmfpde+eA@mail.gmail.com>
	<CAD-qRkgTiowDy_8JKpKj1cfTg2PyYvTNEZNnKBXAnER1xwq1Zw@mail.gmail.com>
	<CAKq5mS5zx6g324z04Z93S__B-gUJb2jmQM=5PFkMWMLcAYg2Zg@mail.gmail.com>
Message-ID: <C6447CC0-AB98-4250-85CE-9EA4C7059753@seventhsignal.de>

Open sourcing Nuance ?

Am 08.09.2011 um 03:18 schrieb envelopes envelopes:

> Do you mind open source it? Thanks.
> 
> On Sep 7, 2011 1:02 PM, "Christopher Rienzo" <cmrienzo at gmail.com> wrote:
> > Yes, this discussion is about doing ASR with Nuance MRCP. The FreeSWITCH
> > core does not care about which ASR engine is used, so these solutions are
> > already pretty generic. However, the inputs and outputs are dependent on
> > which ASR engine is used. Nuance MRCP will use SRGS for grammar definition
> > and NLSML for the response.
> > 
> > 
> > On Wed, Sep 7, 2011 at 2:37 PM, envelopes envelopes <sunwood360 at gmail.com>wrote:
> > 
> >> Does your app support MRCP protocol? That would make it more generic.
> >> On Sep 7, 2011 10:33 AM, "Christopher Rienzo" <cmrienzo at gmail.com> wrote:
> >> > There's not really an easy way to do this out of the box. The FS core
> >> > provides all the low-level functions necessary for doing ASR, but nothing
> >> > higher level for interpreting results, handling barge-in, starting
> >> > input-timers, etc.
> >> >
> >> > I solved this issue with a custom dialplan APP. I know of another other
> >> dev
> >> > that used event socket and the playback/detect_speech dialplan APPs.
> >> > Looking at the scripts/javascript/js_modules/SpeechTools.jm source, you
> >> can
> >> > see that anthm solved it a different way.
> >> >
> >> >
> >> >
> >> > On Mon, Sep 5, 2011 at 3:02 AM, Glen Ganderton <
> >> > Glen.Ganderton at premier.com.au> wrote:
> >> >
> >> >> Hi Guys, ****
> >> >>
> >> >> ** **
> >> >>
> >> >> What i am trying to do is configure an IVR using freeswitch and Nuance
> >> >> 5.1.5 to perform speech recognition. I have installed the latest****
> >> >>
> >> >> release of freeswitch on my CentOS system, I have also enabled and
> >> >> configured the unimrcp module. I could find lots of information to****
> >> >>
> >> >> configure the unimrcp module but there is no documentation on how to
> >> create
> >> >> IVR menu's using this module with Nuance. I have used the ****
> >> >>
> >> >> demo pizza IVR with the pocketsphinx module and I keep getting refered
> >> to
> >> >> this but it's no help as I want to use Nuance for ASR. Any****
> >> >>
> >> >> help on how to configure this would be great, specifically:****
> >> >>
> >> >> ** **
> >> >>
> >> >> * How to I call the IVR script from the dialplan****
> >> >>
> >> >> * What language/s can I code the IVR in and can you please provide a
> >> basic
> >> >> sample of how this is done with Nuance.****
> >> >>
> >> >> ** **
> >> >>
> >> >> Thank you in advance****
> >> >>
> >> >> ** **
> >> >>
> >> >> --Glen****
> >> >>
> >> >>
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >>
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 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



----

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/9f5820d0/attachment-0001.html 

From Glen.Ganderton at premier.com.au  Thu Sep  8 06:08:13 2011
From: Glen.Ganderton at premier.com.au (Glen Ganderton)
Date: Thu, 8 Sep 2011 12:08:13 +1000
Subject: [Freeswitch-users] FreeSWITCH, Nuance, ASR: IVR Integration
In-Reply-To: <CAKq5mS5zx6g324z04Z93S__B-gUJb2jmQM=5PFkMWMLcAYg2Zg@mail.gmail.com>
References: <CD5A1499C8DEF34FAA3509722B00393E740262CA5F@m-mail03.office.premiertech>
	<CAD-qRkjebD1YYrsZb7WkWw0ra+TfpjWNSyoTgavN=U_BhyMRXA@mail.gmail.com>
	<CAKq5mS4eqivAUw5QWVThUB+Xi6U8QJ8X==UfGi-1Lmmfpde+eA@mail.gmail.com>
	<CAD-qRkgTiowDy_8JKpKj1cfTg2PyYvTNEZNnKBXAnER1xwq1Zw@mail.gmail.com>
	<CAKq5mS5zx6g324z04Z93S__B-gUJb2jmQM=5PFkMWMLcAYg2Zg@mail.gmail.com>
Message-ID: <CD5A1499C8DEF34FAA3509722B00393E74029E2A16@m-mail03.office.premiertech>

Yes, that would be a great help if you could release your code. Thanks

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of envelopes envelopes
Sent: Thursday, 8 September 2011 11:18 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeSWITCH, Nuance, ASR: IVR Integration


Do you mind open source it? Thanks.
On Sep 7, 2011 1:02 PM, "Christopher Rienzo" <cmrienzo at gmail.com<mailto:cmrienzo at gmail.com>> wrote:
> Yes, this discussion is about doing ASR with Nuance MRCP. The FreeSWITCH
> core does not care about which ASR engine is used, so these solutions are
> already pretty generic. However, the inputs and outputs are dependent on
> which ASR engine is used. Nuance MRCP will use SRGS for grammar definition
> and NLSML for the response.
>
>
> On Wed, Sep 7, 2011 at 2:37 PM, envelopes envelopes <sunwood360 at gmail.com<mailto:sunwood360 at gmail.com>>wrote:
>
>> Does your app support MRCP protocol? That would make it more generic.
>> On Sep 7, 2011 10:33 AM, "Christopher Rienzo" <cmrienzo at gmail.com<mailto:cmrienzo at gmail.com>> wrote:
>> > There's not really an easy way to do this out of the box. The FS core
>> > provides all the low-level functions necessary for doing ASR, but nothing
>> > higher level for interpreting results, handling barge-in, starting
>> > input-timers, etc.
>> >
>> > I solved this issue with a custom dialplan APP. I know of another other
>> dev
>> > that used event socket and the playback/detect_speech dialplan APPs.
>> > Looking at the scripts/javascript/js_modules/SpeechTools.jm source, you
>> can
>> > see that anthm solved it a different way.
>> >
>> >
>> >
>> > On Mon, Sep 5, 2011 at 3:02 AM, Glen Ganderton <
>> > Glen.Ganderton at premier.com.au<mailto:Glen.Ganderton at premier.com.au>> wrote:
>> >
>> >> Hi Guys, ****
>> >>
>> >> ** **
>> >>
>> >> What i am trying to do is configure an IVR using freeswitch and Nuance
>> >> 5.1.5 to perform speech recognition. I have installed the latest****
>> >>
>> >> release of freeswitch on my CentOS system, I have also enabled and
>> >> configured the unimrcp module. I could find lots of information to****
>> >>
>> >> configure the unimrcp module but there is no documentation on how to
>> create
>> >> IVR menu's using this module with Nuance. I have used the ****
>> >>
>> >> demo pizza IVR with the pocketsphinx module and I keep getting refered
>> to
>> >> this but it's no help as I want to use Nuance for ASR. Any****
>> >>
>> >> help on how to configure this would be great, specifically:****
>> >>
>> >> ** **
>> >>
>> >> * How to I call the IVR script from the dialplan****
>> >>
>> >> * What language/s can I code the IVR in and can you please provide a
>> basic
>> >> sample of how this is done with Nuance.****
>> >>
>> >> ** **
>> >>
>> >> Thank you in advance****
>> >>
>> >> ** **
>> >>
>> >> --Glen****
>> >>
>> >>
>> >> 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
>>
>>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/8ff0f4a0/attachment.html 

From Glen.Ganderton at premier.com.au  Thu Sep  8 09:18:34 2011
From: Glen.Ganderton at premier.com.au (Glen Ganderton)
Date: Thu, 8 Sep 2011 15:18:34 +1000
Subject: [Freeswitch-users] What is Event MEDIA_BUG_STOP
Message-ID: <CD5A1499C8DEF34FAA3509722B00393E74029E2ACD@m-mail03.office.premiertech>

Hey Guys,
I am trying to get event information from the event DETECTED_SPEECH however when I telnet into the event console and type "event plain ALL" I see a lot of events but no
DETECTED_SPEECH event. My speech is detected though, this is a sample of the put from the fs_cli:
---------------------------------------------------------------------------------------------------
2011-09-08 14:27:50.512150 [DEBUG] mod_pocketsphinx.c:383 Recognized: YES, Confidence: 100
EXECUTE sofia/sipinterface_1/2001 at 10.3.5.77 detect_speech(stop)
2011-09-08 14:27:57.578391 [INFO] mod_pocketsphinx.c:221 Port Closed.

However in the event system there is no DETECTED_SPEECH and I see another event called MEDIA_BUG_STOP and MEDIA_BUG_STOP these look they they have something
To do with speech but this isn't what im after doesn't give me all the information. Can anybody explain what my issue may be?

Thanks.


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/084408d9/attachment.html 

From steveayre at gmail.com  Thu Sep  8 14:24:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 8 Sep 2011 11:24:01 +0100
Subject: [Freeswitch-users] What is Event MEDIA_BUG_STOP
In-Reply-To: <CD5A1499C8DEF34FAA3509722B00393E74029E2ACD@m-mail03.office.premiertech>
References: <CD5A1499C8DEF34FAA3509722B00393E74029E2ACD@m-mail03.office.premiertech>
Message-ID: <CAFiqYu=qXsrrmHrcHvt+BOUr+JkFiKZUraNyM1o6AKb5iNLEyQ@mail.gmail.com>

A media bug is where an application attaches a callback to the audio channel
so it can listen to the audio data, that's what the detect_speech app is
doing.

That event just means that detect_speech having finished removed the bug
from the channel.

-Steve



On 8 September 2011 06:18, Glen Ganderton <Glen.Ganderton at premier.com.au>wrote:

> Hey Guys,****
> I am trying to get event information from the event DETECTED_SPEECH however
> when I telnet into the event console and type ?event plain ALL? I see a lot
> of events but no**** DETECTED_SPEECH event. My speech is detected though,
> this is a sample of the put from the fs_cli:****
> ---------------------------------------------------------------------------------------------------
> **** 2011-09-08 14:27:50.512150 [DEBUG] mod_pocketsphinx.c:383 Recognized:
> YES, Confidence: 100****EXECUTE sofia/sipinterface_1/2001 at 10.3.5.77detect_speech(stop)
> ****
> 2011-09-08 14:27:57.578391 [INFO] mod_pocketsphinx.c:221 Port Closed.****
> ** **However in the event system there is no DETECTED_SPEECH and I see
> another event called MEDIA_BUG_STOP and MEDIA_BUG_STOP these look they they
> have something**** To do with speech but this isn?t what im after doesn?t
> give me all the information. Can anybody explain what my issue may be?****
> ** ** Thanks.****** **
>
> ** **
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/1ec2ba7b/attachment.html 

From Glen.Ganderton at premier.com.au  Thu Sep  8 11:18:20 2011
From: Glen.Ganderton at premier.com.au (Glen Ganderton)
Date: Thu, 8 Sep 2011 17:18:20 +1000
Subject: [Freeswitch-users] FreeSWITCH, Nuance, ASR: IVR Integration
In-Reply-To: <C6447CC0-AB98-4250-85CE-9EA4C7059753@seventhsignal.de>
References: <CD5A1499C8DEF34FAA3509722B00393E740262CA5F@m-mail03.office.premiertech>
	<CAD-qRkjebD1YYrsZb7WkWw0ra+TfpjWNSyoTgavN=U_BhyMRXA@mail.gmail.com>
	<CAKq5mS4eqivAUw5QWVThUB+Xi6U8QJ8X==UfGi-1Lmmfpde+eA@mail.gmail.com>
	<CAD-qRkgTiowDy_8JKpKj1cfTg2PyYvTNEZNnKBXAnER1xwq1Zw@mail.gmail.com>
	<CAKq5mS5zx6g324z04Z93S__B-gUJb2jmQM=5PFkMWMLcAYg2Zg@mail.gmail.com>
	<C6447CC0-AB98-4250-85CE-9EA4C7059753@seventhsignal.de>
Message-ID: <CD5A1499C8DEF34FAA3509722B00393E74029E2B16@m-mail03.office.premiertech>

He is asking Chris if he wouldn't mind making his custom dialplan app available to us.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michal Bielicki
Sent: Thursday, 8 September 2011 4:21 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeSWITCH, Nuance, ASR: IVR Integration

Open sourcing Nuance ?

Am 08.09.2011 um 03:18 schrieb envelopes envelopes:



Do you mind open source it? Thanks.
On Sep 7, 2011 1:02 PM, "Christopher Rienzo" <cmrienzo at gmail.com<mailto:cmrienzo at gmail.com>> wrote:
> Yes, this discussion is about doing ASR with Nuance MRCP. The FreeSWITCH
> core does not care about which ASR engine is used, so these solutions are
> already pretty generic. However, the inputs and outputs are dependent on
> which ASR engine is used. Nuance MRCP will use SRGS for grammar definition
> and NLSML for the response.
>
>
> On Wed, Sep 7, 2011 at 2:37 PM, envelopes envelopes <sunwood360 at gmail.com<mailto:sunwood360 at gmail.com>>wrote:
>
>> Does your app support MRCP protocol? That would make it more generic.
>> On Sep 7, 2011 10:33 AM, "Christopher Rienzo" <cmrienzo at gmail.com<mailto:cmrienzo at gmail.com>> wrote:
>> > There's not really an easy way to do this out of the box. The FS core
>> > provides all the low-level functions necessary for doing ASR, but nothing
>> > higher level for interpreting results, handling barge-in, starting
>> > input-timers, etc.
>> >
>> > I solved this issue with a custom dialplan APP. I know of another other
>> dev
>> > that used event socket and the playback/detect_speech dialplan APPs.
>> > Looking at the scripts/javascript/js_modules/SpeechTools.jm source, you
>> can
>> > see that anthm solved it a different way.
>> >
>> >
>> >
>> > On Mon, Sep 5, 2011 at 3:02 AM, Glen Ganderton <
>> > Glen.Ganderton at premier.com.au<mailto:Glen.Ganderton at premier.com.au>> wrote:
>> >
>> >> Hi Guys, ****
>> >>
>> >> ** **
>> >>
>> >> What i am trying to do is configure an IVR using freeswitch and Nuance
>> >> 5.1.5 to perform speech recognition. I have installed the latest****
>> >>
>> >> release of freeswitch on my CentOS system, I have also enabled and
>> >> configured the unimrcp module. I could find lots of information to****
>> >>
>> >> configure the unimrcp module but there is no documentation on how to
>> create
>> >> IVR menu's using this module with Nuance. I have used the ****
>> >>
>> >> demo pizza IVR with the pocketsphinx module and I keep getting refered
>> to
>> >> this but it's no help as I want to use Nuance for ASR. Any****
>> >>
>> >> help on how to configure this would be great, specifically:****
>> >>
>> >> ** **
>> >>
>> >> * How to I call the IVR script from the dialplan****
>> >>
>> >> * What language/s can I code the IVR in and can you please provide a
>> basic
>> >> sample of how this is done with Nuance.****
>> >>
>> >> ** **
>> >>
>> >> Thank you in advance****
>> >>
>> >> ** **
>> >>
>> >> --Glen****
>> >>
>> >>
>> >> 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<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<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

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht 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<http://www.seventhsignal.de/>


----
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/a4d0c049/attachment-0001.html 

From glenganderton at gmail.com  Thu Sep  8 15:43:17 2011
From: glenganderton at gmail.com (Glen Ganderton)
Date: Thu, 8 Sep 2011 21:43:17 +1000
Subject: [Freeswitch-users] What is Event MEDIA_BUG_STOP
In-Reply-To: <CAFiqYu=qXsrrmHrcHvt+BOUr+JkFiKZUraNyM1o6AKb5iNLEyQ@mail.gmail.com>
References: <CD5A1499C8DEF34FAA3509722B00393E74029E2ACD@m-mail03.office.premiertech>
	<CAFiqYu=qXsrrmHrcHvt+BOUr+JkFiKZUraNyM1o6AKb5iNLEyQ@mail.gmail.com>
Message-ID: <CAJiHjvf3uyHQA8whgXqsTOJakyfDfwTr2bjRzWECcdOYt5Agrg@mail.gmail.com>

Thanks Steve,

But where is the DETECTED_SPEECH event. This is the event I really need but
I cant see it. Any idea's?

Ive tested with both mod_pocketsphinx and mod_unimrcp and could not see that
event.

-Glen



On Thu, Sep 8, 2011 at 8:24 PM, Steven Ayre <steveayre at gmail.com> wrote:

> A media bug is where an application attaches a callback to the audio
> channel so it can listen to the audio data, that's what the detect_speech
> app is doing.
>
> That event just means that detect_speech having finished removed the bug
> from the channel.
>
> -Steve
>
>
>
> On 8 September 2011 06:18, Glen Ganderton <Glen.Ganderton at premier.com.au>wrote:
>
>> Hey Guys,****
>> I am trying to get event information from the event DETECTED_SPEECH
>> however when I telnet into the event console and type ?event plain ALL? I
>> see a lot of events but no**** DETECTED_SPEECH event. My speech is
>> detected though, this is a sample of the put from the fs_cli:****
>> ---------------------------------------------------------------------------------------------------
>> **** 2011-09-08 14:27:50.512150 [DEBUG] mod_pocketsphinx.c:383
>> Recognized: YES, Confidence: 100****EXECUTE sofia/sipinterface_1/
>> 2001 at 10.3.5.77 detect_speech(stop)****
>> 2011-09-08 14:27:57.578391 [INFO] mod_pocketsphinx.c:221 Port Closed.****
>> ** **However in the event system there is no DETECTED_SPEECH and I see
>> another event called MEDIA_BUG_STOP and MEDIA_BUG_STOP these look they they
>> have something**** To do with speech but this isn?t what im after doesn?t
>> give me all the information. Can anybody explain what my issue may be?***
>> *** ** 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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/c8fb0485/attachment.html 

From sunwood360 at gmail.com  Thu Sep  8 19:28:40 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Thu, 8 Sep 2011 08:28:40 -0700
Subject: [Freeswitch-users] FreeSWITCH, Nuance, ASR: IVR Integration
In-Reply-To: <C6447CC0-AB98-4250-85CE-9EA4C7059753@seventhsignal.de>
References: <CD5A1499C8DEF34FAA3509722B00393E740262CA5F@m-mail03.office.premiertech>
	<CAD-qRkjebD1YYrsZb7WkWw0ra+TfpjWNSyoTgavN=U_BhyMRXA@mail.gmail.com>
	<CAKq5mS4eqivAUw5QWVThUB+Xi6U8QJ8X==UfGi-1Lmmfpde+eA@mail.gmail.com>
	<CAD-qRkgTiowDy_8JKpKj1cfTg2PyYvTNEZNnKBXAnER1xwq1Zw@mail.gmail.com>
	<CAKq5mS5zx6g324z04Z93S__B-gUJb2jmQM=5PFkMWMLcAYg2Zg@mail.gmail.com>
	<C6447CC0-AB98-4250-85CE-9EA4C7059753@seventhsignal.de>
Message-ID: <CAKq5mS4VFgpT+yCobBV938q5Y+FwnMj-z7pi0aAS2RATHerrkw@mail.gmail.com>

My friend, are you sure your calendar's date is April/01?
Kidding....
@_@
On Sep 7, 2011 11:26 PM, "Michal Bielicki" <michal.bielicki at seventhsignal.de>
wrote:
> Open sourcing Nuance ?
>
> Am 08.09.2011 um 03:18 schrieb envelopes envelopes:
>
>> Do you mind open source it? Thanks.
>>
>> On Sep 7, 2011 1:02 PM, "Christopher Rienzo" <cmrienzo at gmail.com> wrote:
>> > Yes, this discussion is about doing ASR with Nuance MRCP. The
FreeSWITCH
>> > core does not care about which ASR engine is used, so these solutions
are
>> > already pretty generic. However, the inputs and outputs are dependent
on
>> > which ASR engine is used. Nuance MRCP will use SRGS for grammar
definition
>> > and NLSML for the response.
>> >
>> >
>> > On Wed, Sep 7, 2011 at 2:37 PM, envelopes envelopes <
sunwood360 at gmail.com>wrote:
>> >
>> >> Does your app support MRCP protocol? That would make it more generic.
>> >> On Sep 7, 2011 10:33 AM, "Christopher Rienzo" <cmrienzo at gmail.com>
wrote:
>> >> > There's not really an easy way to do this out of the box. The FS
core
>> >> > provides all the low-level functions necessary for doing ASR, but
nothing
>> >> > higher level for interpreting results, handling barge-in, starting
>> >> > input-timers, etc.
>> >> >
>> >> > I solved this issue with a custom dialplan APP. I know of another
other
>> >> dev
>> >> > that used event socket and the playback/detect_speech dialplan APPs.
>> >> > Looking at the scripts/javascript/js_modules/SpeechTools.jm source,
you
>> >> can
>> >> > see that anthm solved it a different way.
>> >> >
>> >> >
>> >> >
>> >> > On Mon, Sep 5, 2011 at 3:02 AM, Glen Ganderton <
>> >> > Glen.Ganderton at premier.com.au> wrote:
>> >> >
>> >> >> Hi Guys, ****
>> >> >>
>> >> >> ** **
>> >> >>
>> >> >> What i am trying to do is configure an IVR using freeswitch and
Nuance
>> >> >> 5.1.5 to perform speech recognition. I have installed the
latest****
>> >> >>
>> >> >> release of freeswitch on my CentOS system, I have also enabled and
>> >> >> configured the unimrcp module. I could find lots of information
to****
>> >> >>
>> >> >> configure the unimrcp module but there is no documentation on how
to
>> >> create
>> >> >> IVR menu's using this module with Nuance. I have used the ****
>> >> >>
>> >> >> demo pizza IVR with the pocketsphinx module and I keep getting
refered
>> >> to
>> >> >> this but it's no help as I want to use Nuance for ASR. Any****
>> >> >>
>> >> >> help on how to configure this would be great, specifically:****
>> >> >>
>> >> >> ** **
>> >> >>
>> >> >> * How to I call the IVR script from the dialplan****
>> >> >>
>> >> >> * What language/s can I code the IVR in and can you please provide
a
>> >> basic
>> >> >> sample of how this is done with Nuance.****
>> >> >>
>> >> >> ** **
>> >> >>
>> >> >> Thank you in advance****
>> >> >>
>> >> >> ** **
>> >> >>
>> >> >> --Glen****
>> >> >>
>> >> >>
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> UNSUBSCRIBE:
>> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >>
>> >>
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 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
>
>
>
> ----
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/cfa99119/attachment.html 

From tang.du at hotmail.com  Thu Sep  8 19:38:11 2011
From: tang.du at hotmail.com (tangdu)
Date: Thu, 8 Sep 2011 08:38:11 -0700 (PDT)
Subject: [Freeswitch-users] =?utf-8?q?how_about_config_astpp_for_freeswitc?=
	=?utf-8?b?aO+8nw==?=
Message-ID: <1315496291426-6772321.post@n2.nabble.com>

hello?
i installed astpp-2315 for freeswitch?now i can visit
"http://xxx.xxx.xxx.xxx/astpp-admin/astpp-admin.cgi " and login useing
default passwd?
Promt ?Successful Login! Realtime Database Unavailable! ? it's because of i
couldn't found the file look like realtime.sql ?haven't creat Realtime
Database ?
i found word of menu still is asterisk ?not freeswitch ?when i click any
menu?system come back login interface?
why?who can help me?
thanks 

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/how-about-config-astpp-for-freeswitch-tp6772321p6772321.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From darren at aleph-com.net  Thu Sep  8 19:48:53 2011
From: darren at aleph-com.net (Darren Wiebe)
Date: Thu, 8 Sep 2011 09:48:53 -0600
Subject: [Freeswitch-users]
	=?utf-8?q?how_about_config_astpp_for_freeswitc?=
	=?utf-8?b?aO+8nw==?=
In-Reply-To: <1315496291426-6772321.post@n2.nabble.com>
References: <1315496291426-6772321.post@n2.nabble.com>
Message-ID: <CAGzU91MxgTQyS_WDOF8jA0z4FxnFDmitxbQE_7M9=gQryJA1oQ@mail.gmail.com>

This isn't a freeswitch question.  I see you asked the same question
on the astpp forum a few hours ago.  Somebody will answer you there
but you might have to wait a little longer.

Darren Wiebe
darren at aleph-com.net

On Thu, Sep 8, 2011 at 9:38 AM, tangdu <tang.du at hotmail.com> wrote:
> hello?
> i installed astpp-2315 for freeswitch?now i can visit
> "http://xxx.xxx.xxx.xxx/astpp-admin/astpp-admin.cgi " and login useing
> default passwd?
> Promt ?Successful Login! Realtime Database Unavailable! ? it's because of i
> couldn't found the file look like realtime.sql ?haven't creat Realtime
> Database ?
> i found word of menu still is asterisk ?not freeswitch ?when i click any
> menu?system come back login interface?
> why?who can help me?
> thanks
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/how-about-config-astpp-for-freeswitch-tp6772321p6772321.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 jeff at jefflenk.com  Thu Sep  8 20:03:13 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 8 Sep 2011 09:03:13 -0700 (PDT)
Subject: [Freeswitch-users] What is Event MEDIA_BUG_STOP
In-Reply-To: <CAJiHjvf3uyHQA8whgXqsTOJakyfDfwTr2bjRzWECcdOYt5Agrg@mail.gmail.com>
References: <CD5A1499C8DEF34FAA3509722B00393E74029E2ACD@m-mail03.office.premiertech>
	<CAFiqYu=qXsrrmHrcHvt+BOUr+JkFiKZUraNyM1o6AKb5iNLEyQ@mail.gmail.com>
	<CAJiHjvf3uyHQA8whgXqsTOJakyfDfwTr2bjRzWECcdOYt5Agrg@mail.gmail.com>
Message-ID: <1315497793886-6772422.post@n2.nabble.com>

Do you have the channel variable "fire_asr_events" set?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/What-is-Event-MEDIA-BUG-STOP-tp6770735p6772422.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From dgarcia at anew.com.ve  Thu Sep  8 20:17:38 2011
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Thu, 08 Sep 2011 11:47:38 -0430
Subject: [Freeswitch-users] =?utf-8?q?About_screen_pop_with_freeswitch?=
	=?utf-8?b?77yf?=
In-Reply-To: <CAGzU91MxgTQyS_WDOF8jA0z4FxnFDmitxbQE_7M9=gQryJA1oQ@mail.gmail.com>
References: <1315496291426-6772321.post@n2.nabble.com>
	<CAGzU91MxgTQyS_WDOF8jA0z4FxnFDmitxbQE_7M9=gQryJA1oQ@mail.gmail.com>
Message-ID: <4E68EAA2.5010008@anew.com.ve>

Hi,

I am playing with freeswitch scripting with lua. I want call to number, 
ex 1000, a script lua will process the call ( as an IVR), then transfer 
the call to an extension (or transfer the call to a queue, to get a free 
agent).
Questions:
How I add a pair data (key/value. ex ID/1234) to the call in the lua 
script?
How can check the data attached is avaliable to the end extension?
What you use to make screen pop call data when the call ring in the 
extension?

Thanks



From cesar.bermudez at gmail.com  Thu Sep  8 20:18:21 2011
From: cesar.bermudez at gmail.com (Cesar Bermudez)
Date: Thu, 8 Sep 2011 10:18:21 -0600
Subject: [Freeswitch-users]  What Provider do you guys use?
Message-ID: <CADrcvV86dR63-G6_-_KkpLSVh7UVEosT9gwNOJR-vK-Y37ibNA@mail.gmail.com>

Hi Fs'users

Sorry for this question, but what good providers you recommend?
I need good quality to this destinations:
USA
Nicaragua
Vietnan
China
Indonesia

I want good routes, with cli if possible, and good prices :D

Sorry for this mail again, i dont want to make any flame war or spam ...
only want advice from more experience voip admins.

Best regards.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/d43fb301/attachment.html 

From jeff at jefflenk.com  Thu Sep  8 20:26:14 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 8 Sep 2011 09:26:14 -0700 (PDT)
Subject: [Freeswitch-users] How to use sofia_count_reg ?
In-Reply-To: <CALNZuw=B3YYD1eWEMKERWtpitQBDv-AJKgw03k4mGCdZDfOQtA@mail.gmail.com>
References: <CALNZuw=B3YYD1eWEMKERWtpitQBDv-AJKgw03k4mGCdZDfOQtA@mail.gmail.com>
Message-ID: <1315499174596-6772542.post@n2.nabble.com>

This should be reported to Jira with the missing column

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-use-sofia-count-reg-tp6770571p6772542.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jeff at jefflenk.com  Thu Sep  8 20:26:14 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 8 Sep 2011 09:26:14 -0700 (PDT)
Subject: [Freeswitch-users] How to use sofia_count_reg ?
In-Reply-To: <CALNZuw=B3YYD1eWEMKERWtpitQBDv-AJKgw03k4mGCdZDfOQtA@mail.gmail.com>
References: <CALNZuw=B3YYD1eWEMKERWtpitQBDv-AJKgw03k4mGCdZDfOQtA@mail.gmail.com>
Message-ID: <1315499174093-6772541.post@n2.nabble.com>

This should be reported to Jira with the missing column

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-use-sofia-count-reg-tp6770571p6772541.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gerry at pstn2.net  Thu Sep  8 20:47:34 2011
From: gerry at pstn2.net (Gerry Hull)
Date: Thu, 8 Sep 2011 12:47:34 -0400
Subject: [Freeswitch-users] FreeSwitch Windows ISO Hard Coded Install Path...
Message-ID: <CABWdUKC6S54SydH5P1cYEwxHn3Xq44e2gdYsuBrUeEB4fxGUpQ@mail.gmail.com>

Is there a reason why the ISO has a hard coded install path?

Is it possible to get this changed, as it used to be, so that you can select
the drive/directory where the install should be?

Regards,

Gerry
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/54603590/attachment.html 

From joe.jflemmings at gmail.com  Thu Sep  8 21:03:18 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Thu, 8 Sep 2011 10:03:18 -0700
Subject: [Freeswitch-users] g729 Pass-through
Message-ID: <CALH5ZrZREwaFaTTQKW_yGprUQe9zmHVAPDHLdxnHHG0_Q6N3hA@mail.gmail.com>

Is there a way of having just calls using g711 do media passthrough and NOT
all calls
Because the following parameters in internal.xml apply to all calls


<param name="inbound-bypass-media" value="true"/>

<param name="inbound-proxy-media" value="true"/>


Thanks Joe
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/c91a7f91/attachment.html 

From peter.olsson at visionutveckling.se  Thu Sep  8 21:19:19 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 8 Sep 2011 19:19:19 +0200
Subject: [Freeswitch-users] g729 Pass-through
In-Reply-To: <CALH5ZrZREwaFaTTQKW_yGprUQe9zmHVAPDHLdxnHHG0_Q6N3hA@mail.gmail.com>
References: <CALH5ZrZREwaFaTTQKW_yGprUQe9zmHVAPDHLdxnHHG0_Q6N3hA@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0F5@cooper>

You need to do it in the dialplan then I guess. Just set the bypass_media=true channel variable when you want bypass media.

/Petet
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Joe Flemmings [joe.jflemmings at gmail.com]
Skickat: den 8 september 2011 19:03
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] g729 Pass-through

Is there a way of having just calls using g711 do media passthrough and NOT all calls
Because the following parameters in internal.xml apply to all calls

<param name="inbound-bypass-media" value="true"/>
<param name="inbound-proxy-media" value="true"/>

Thanks Joe
!DSPAM:4e68f59732765335416922!


From juanito1982 at gmail.com  Thu Sep  8 21:47:09 2011
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Thu, 8 Sep 2011 19:47:09 +0200
Subject: [Freeswitch-users] vBilling Beta Program!!
In-Reply-To: <CACak_yxUrzNGL2-bgb_ekEXTbBivS9e7f6-SM49+z8cyx_rr3A@mail.gmail.com>
References: <CACak_ywBJcmp1rGDpWqWiSrahguGUcJ1qQcYqC5yB+mm4LeHrA@mail.gmail.com>
	<CAGQG5Mbskb7e24igA_AeBpHWVmXKJ_E-fkj3+SN3wH39ioJLzg@mail.gmail.com>
	<CACak_yxUrzNGL2-bgb_ekEXTbBivS9e7f6-SM49+z8cyx_rr3A@mail.gmail.com>
Message-ID: <CAHLYDCu5cGU3KFguy28CzuM3RxQ+RYA7nYZnzLmZF4UTOJNH5A@mail.gmail.com>

How do you implement load balancing?

2011/9/6 Muhammad Naseer Bhatti <nbhatti at gmail.com>

>
> Our last test showed around 500 concurrent calls. Since we support
> distributed setups, so in case you need more numbers, simply add a new
> machine running FreeSWITCH and you are done. Billing interface will be the
> same running on 1 single node. We are going to publish some benchmarks in a
> few days. It is actually undergoing some real bad test by one of our
> customers :) Stay tuned.
>
>
> On Tue, Sep 6, 2011 at 11:26 AM, Abdul Basit <basit.engg at gmail.com> wrote:
>
>> Interesting...
>>
>> Any max call limits? what is cps?
>> We will appreciate stress test results if anyone can share.
>>
>> --
>> Regards,
>>
>> Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445
>>
>>
>>
>> On Mon, Sep 5, 2011 at 7:45 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com
>> > wrote:
>>
>>>
>>> Hello everyone,
>>> As promised we have opened beta testing program for *vBilling*. An open
>>> source billing platform for FreeSWITCH. You are invited to login, take a
>>> look and play with it. Email us with your comments, let us know what
>>> improvements can be made. Following are the details for the program:
>>>
>>> ========================================
>>> vBilling User Panel: *http://demo.vbilling.org/*
>>> User Login: *demouser*
>>> User Password: *P at ssw0rd*
>>>
>>> vBilling Admin Panel: *http://demo.vbilling.org/admin/*
>>> Admin Login: *admin*
>>> Admin Password: *P at ssw0rd*
>>>
>>> We have configured a LIVE gateway for you. You can send your *SIP* calls
>>> to *demo.digitallinx.com* port *5060* and call any toll free (800 and
>>> 888 only) number in US. Make calls, login to *vBilling* and see how the
>>> billing works.
>>> ========================================
>>>
>>> Most of the features are working. Some of the them mentioned on the site
>>> are still is development so be patient :) For product features and more
>>> details, visit our website at *http://www.vbilling.org/*<http://www.vbilling.org/>and us know what do you think about it.
>>>
>>>
>>> Regards,
>>> Muhammad Naseer
>>> CEO vBilling/Digital Linx
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/e048980e/attachment.html 

From cabildo at gmail.com  Thu Sep  8 22:34:05 2011
From: cabildo at gmail.com (Julio Saldivar)
Date: Thu, 8 Sep 2011 14:34:05 -0400
Subject: [Freeswitch-users] small problem with the context of an FXO
Message-ID: <CAO2g7M-KTBLUURpwAqFav+enGuVTvqROn23Kb+PQdx9-rc=tFA@mail.gmail.com>

i have a problem with the context of an FXO, i editing the freetdm.conf.xml
with the following configuration:

    <analog_spans>
        <span name="FXO"/>
        <param name="dialplan" value="XML"/>
        <param name="context" value="fxo"/>
        <param name="polarity-delay" value="600" />
        <param name="answer-polarity-reverse" value="true" />
        <param name="hangup-polarity-reverse" value="true" />
    </analog_spans>

after i rebooting freeswitch and i running teh command "ftdm list":

freeswitch at internal> ftdm list
+OK
span: 1 (FXO)
type: analog
physical_status: ok
signaling_status: UP
chan_count: 1
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options: none


and does not change the context

-- 
Si alguna vez
mi voz deja de escucharse
piensen que el bosque
hablar? por m?
con su lenguaje de ra?ces.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/3d59363f/attachment-0001.html 

From steveayre at gmail.com  Thu Sep  8 22:41:46 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 8 Sep 2011 19:41:46 +0100
Subject: [Freeswitch-users] g729 Pass-through
In-Reply-To: <CALH5ZrZREwaFaTTQKW_yGprUQe9zmHVAPDHLdxnHHG0_Q6N3hA@mail.gmail.com>
References: <CALH5ZrZREwaFaTTQKW_yGprUQe9zmHVAPDHLdxnHHG0_Q6N3hA@mail.gmail.com>
Message-ID: <CAFiqYun0MyxOnuBH4gxRGEt5sUtccA9QKGzWp+WsEipZMMQkBg@mail.gmail.com>

They can be set (or unset) from the dialplan:
    <action application="set" data="bypass_media=true" />
    <action application="set" data="bypass_media=false" />
    <action application="set" data="proxy_media=true" />
    <action application="set" data="proxy_media=false" />

Unless you're transcoding or attaching a application that needs to decode
the audio data, all bridged calls will work in passthrough mode by default.
Decoding/encoding of audio data only happens when it's required.

Also note that proxy-media is *not* passthrough. You shouldn't really use it
unless you want to support a codec that's not supported by FS.

-Steve



On 8 September 2011 18:03, Joe Flemmings <joe.jflemmings at gmail.com> wrote:

> Is there a way of having just calls using g711 do media passthrough and NOT
> all calls
> Because the following parameters in internal.xml apply to all calls
>
>
> <param name="inbound-bypass-media" value="true"/>
>
> <param name="inbound-proxy-media" value="true"/>
>
>
> Thanks Joe
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/277cb927/attachment.html 

From joe.jflemmings at gmail.com  Thu Sep  8 22:57:57 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Thu, 8 Sep 2011 11:57:57 -0700
Subject: [Freeswitch-users] g729 Pass-through
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0F5@cooper>
References: <CALH5ZrZREwaFaTTQKW_yGprUQe9zmHVAPDHLdxnHHG0_Q6N3hA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0F5@cooper>
Message-ID: <CALH5ZrYC0hyA4HozdV68kf2MFJqhP60GJZQm5=Zfr72WVq_o=w@mail.gmail.com>

Thank You guys

On Thu, Sep 8, 2011 at 10:19 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> You need to do it in the dialplan then I guess. Just set the
> bypass_media=true channel variable when you want bypass media.
>
> /Petet
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Joe Flemmings [
> joe.jflemmings at gmail.com]
> Skickat: den 8 september 2011 19:03
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] g729 Pass-through
>
> Is there a way of having just calls using g711 do media passthrough and NOT
> all calls
> Because the following parameters in internal.xml apply to all calls
>
> <param name="inbound-bypass-media" value="true"/>
> <param name="inbound-proxy-media" value="true"/>
>
> Thanks Joe
> !DSPAM:4e68f59732765335416922!
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/60af01f9/attachment.html 

From netcentrica at gmail.com  Thu Sep  8 22:59:19 2011
From: netcentrica at gmail.com (Mateusz Bartczak)
Date: Thu, 8 Sep 2011 20:59:19 +0200
Subject: [Freeswitch-users] FreeSWITCH failover/HA
Message-ID: <CAJWEF2ZPcrVirCFKCtHOugntRZ5sETDpDPCZyTRNso8OKMaRJA@mail.gmail.com>

Hi

I would like to implement following scenario:

1. Central SIP proxy with 1 IP address that will redirect all incoming and
outgoing SIP traffic to two SIP application servers (FreeSwitch based).
Proxy will know online/offline status of each box and route calls only to
active one. I need it as a central point because IP address is authorized
with my providers. Also providers route incoming calls to that IP address.
My providers can't automatically reroute traffic to other server, it can be
done manually but it's not fast to do.

2. Mini cluster of two FreeSwitch boxes, box 2 will be exact copy of box 1.
If box 1 fails, central server should automatically route all traffic to box
2.

Do you have any suggestions how to implement this scenario? I think that it
should be easy to do, but have no idea where to start.

Best Regards,
Mateusz
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/88f0059e/attachment.html 

From netcentrica at gmail.com  Thu Sep  8 23:03:38 2011
From: netcentrica at gmail.com (Mateusz Bartczak)
Date: Thu, 8 Sep 2011 21:03:38 +0200
Subject: [Freeswitch-users] FreeSwitch failover/HA
Message-ID: <CAJWEF2a5wuO2W1sp_JgdrgYLkaJs74UBuKD2DRg4og+7Ur0Zmg@mail.gmail.com>

Hi

I would like to implement following scenario:

1. Central SIP proxy with 1 IP address that will redirect all incoming and
outgoing SIP traffic to two SIP application servers (FreeSwitch based).
Proxy will know online/offline status of each box and route calls only to
active one. I need it as a central point because IP address is authorized
with my providers. Also providers route incoming calls to that IP address.
My providers can't automatically reroute traffic to other server, it can be
done manually but it's not fast to do.

2. Mini cluster of two FreeSwitch boxes, box 2 will be exact copy of box 1.
If box 1 fails, central server should automatically route all traffic to box
2.

Do you have any suggestions how to implement this scenario? I think that it
should be easy to do, but have no idea where to start.

Best Regards,
Mateusz
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/281bd0d6/attachment.html 

From brad at tech21.com  Thu Sep  8 23:07:22 2011
From: brad at tech21.com (Brad Mina)
Date: Thu, 8 Sep 2011 12:07:22 -0700
Subject: [Freeswitch-users] FreeSWITCH failover/HA
In-Reply-To: <CAJWEF2ZPcrVirCFKCtHOugntRZ5sETDpDPCZyTRNso8OKMaRJA@mail.gmail.com>
References: <CAJWEF2ZPcrVirCFKCtHOugntRZ5sETDpDPCZyTRNso8OKMaRJA@mail.gmail.com>
Message-ID: <CAGRr058up9tvP0t4p1KOkRzO416Tqxkw_g318HV+T4g94SKRbw@mail.gmail.com>

AviMarcus has an intro page on his wiki user profile.
http://wiki.freeswitch.org/wiki/User:Avi_Marcus

Unfortunately this lacks specifics, so you'll have to read up on the
software.

On Thu, Sep 8, 2011 at 11:59 AM, Mateusz Bartczak <netcentrica at gmail.com>wrote:

> Hi
>
> I would like to implement following scenario:
>
> 1. Central SIP proxy with 1 IP address that will redirect all incoming and
> outgoing SIP traffic to two SIP application servers (FreeSwitch based).
> Proxy will know online/offline status of each box and route calls only to
> active one. I need it as a central point because IP address is authorized
> with my providers. Also providers route incoming calls to that IP address.
> My providers can't automatically reroute traffic to other server, it can be
> done manually but it's not fast to do.
>
> 2. Mini cluster of two FreeSwitch boxes, box 2 will be exact copy of box 1.
> If box 1 fails, central server should automatically route all traffic to box
> 2.
>
> Do you have any suggestions how to implement this scenario? I think that it
> should be easy to do, but have no idea where to start.
>
> Best Regards,
> Mateusz
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/f441fe30/attachment.html 

From mkopacki at gmail.com  Thu Sep  8 23:17:25 2011
From: mkopacki at gmail.com (Michal Kopacki)
Date: Thu, 8 Sep 2011 21:17:25 +0200
Subject: [Freeswitch-users] FreeSwitch failover/HA
In-Reply-To: <CAJWEF2a5wuO2W1sp_JgdrgYLkaJs74UBuKD2DRg4og+7Ur0Zmg@mail.gmail.com>
References: <CAJWEF2a5wuO2W1sp_JgdrgYLkaJs74UBuKD2DRg4og+7Ur0Zmg@mail.gmail.com>
Message-ID: <20110908211725.53a6d071@reapnet.com>

  http://wiki.freeswitch.com/wiki/Enterprise_deployment_OpenSIPS

-- 
Regards,
Michal

On Thu, 8 Sep 2011 21:03:38 +0200
Mateusz Bartczak <netcentrica at gmail.com> wrote:

> Hi
> 
> I would like to implement following scenario:
> 
> 1. Central SIP proxy with 1 IP address that will redirect all
> incoming and outgoing SIP traffic to two SIP application servers
> (FreeSwitch based). Proxy will know online/offline status of each box
> and route calls only to active one. I need it as a central point
> because IP address is authorized with my providers. Also providers
> route incoming calls to that IP address. My providers can't
> automatically reroute traffic to other server, it can be done
> manually but it's not fast to do.
> 
> 2. Mini cluster of two FreeSwitch boxes, box 2 will be exact copy of
> box 1. If box 1 fails, central server should automatically route all
> traffic to box 2.
> 
> Do you have any suggestions how to implement this scenario? I think
> that it should be easy to do, but have no idea where to start.
> 
> Best Regards,
> Mateusz



From anthony.minessale at gmail.com  Fri Sep  9 00:38:37 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 8 Sep 2011 15:38:37 -0500
Subject: [Freeswitch-users] What is Event MEDIA_BUG_STOP
In-Reply-To: <1315497793886-6772422.post@n2.nabble.com>
References: <CD5A1499C8DEF34FAA3509722B00393E74029E2ACD@m-mail03.office.premiertech>
	<CAFiqYu=qXsrrmHrcHvt+BOUr+JkFiKZUraNyM1o6AKb5iNLEyQ@mail.gmail.com>
	<CAJiHjvf3uyHQA8whgXqsTOJakyfDfwTr2bjRzWECcdOYt5Agrg@mail.gmail.com>
	<1315497793886-6772422.post@n2.nabble.com>
Message-ID: <CAKbxfG_LTo2MTqsEgC8UVm6Kg5YOMBgYY+WdkuKxyud7Bfye2A@mail.gmail.com>

also if you use outbound socket you can send the "divert_events"
command to get the asr events over that socket connection.


On Thu, Sep 8, 2011 at 11:03 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
> Do you have the channel variable "fire_asr_events" set?
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/What-is-Event-MEDIA-BUG-STOP-tp6770735p6772422.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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 mstockton at harqen.com  Fri Sep  9 00:41:47 2011
From: mstockton at harqen.com (Matt Stockton)
Date: Thu, 8 Sep 2011 15:41:47 -0500
Subject: [Freeswitch-users] FreeSWITCH failover/HA
In-Reply-To: <CAJWEF2ZPcrVirCFKCtHOugntRZ5sETDpDPCZyTRNso8OKMaRJA@mail.gmail.com>
References: <CAJWEF2ZPcrVirCFKCtHOugntRZ5sETDpDPCZyTRNso8OKMaRJA@mail.gmail.com>
Message-ID: <CAHMdK5nhYOixxkh2DSrf0qQGmc8LLhFbZsTuq_pRGgWGgM-t8w@mail.gmail.com>

Hi Mateusz,

I was just able to recently get this working using OpenSIPS and the
instructions found here:
http://wiki.freeswitch.com/wiki/Enterprise_deployment_OpenSIPS

There were a few modifications I had to make, for example:
- dlg_flag seems to be no longer available in the newest OpenSips, but I do
not think it is needed.
- I removed the dispatcher module because I didn't need to support
registration
- I had to change the INVITE conditional to be dependent on whether or not
the invite was coming from outside (e.g. PSTN) or was coming from one of my
freeswitch servers. For loadbalancing coming in, the load_balancer module
worked fine as configured in the wiki instructions. For outbound, I used the
dynamic routing module:
http://www.unixnews.net/2010/09/dynamic-routing-with-opensips.html

It seems to be working well so far. I will plan to update the wiki with more
information. Feel free to reach out to me if you have any questions.

Thanks,
Matt

On Thu, Sep 8, 2011 at 1:59 PM, Mateusz Bartczak <netcentrica at gmail.com>wrote:

> Hi
>
> I would like to implement following scenario:
>
> 1. Central SIP proxy with 1 IP address that will redirect all incoming and
> outgoing SIP traffic to two SIP application servers (FreeSwitch based).
> Proxy will know online/offline status of each box and route calls only to
> active one. I need it as a central point because IP address is authorized
> with my providers. Also providers route incoming calls to that IP address.
> My providers can't automatically reroute traffic to other server, it can be
> done manually but it's not fast to do.
>
> 2. Mini cluster of two FreeSwitch boxes, box 2 will be exact copy of box 1.
> If box 1 fails, central server should automatically route all traffic to box
> 2.
>
> Do you have any suggestions how to implement this scenario? I think that it
> should be easy to do, but have no idea where to start.
>
> Best Regards,
> Mateusz
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/fef5f74a/attachment-0001.html 

From danlanweb at gmail.com  Fri Sep  9 00:50:22 2011
From: danlanweb at gmail.com (Dan Lan)
Date: Thu, 8 Sep 2011 13:50:22 -0700
Subject: [Freeswitch-users] One Way Audio - Auto Change RTP port?
Message-ID: <CAAuJ8MBnFKYzEjPnCNeNmiU-T=jRu3M1vcrR49-p2LPyRU=+uQ@mail.gmail.com>

Hi,
I run into a weird situation. My media gateay handle voice call with 2
different RTP ports for send & receive

Here is what happened. (ps: both gateway and FS are all on public IP, no NAT
involved)
1. Incoming call INVITE from gateway to FS
Connection Information (c): IN IP4 100.100.100.100  (This is my media
gateway IP address)
Media Description, name and address (m): audio 5294 RTP/AVP 18 0 4
2. FS response with session progress with media information
Connection Information (c): IN IP4 200.200.200.200
Media Description, name and address (m): audio 22428 RTP/AVP 0
3. I start to see some RTP traffic exchange between FS and GW
from FS (22428) --> GW (5294)
from GW (5292) --> FS (22428)
please note: the GW use two DIFFERENT PORT for RTP, one for sending and one
for receiving
4. For a while (about 5 secs, I think)
The RTP flow change on FS side to become, (there is no RTCP packet during
the time)
 from FS (22428) --> GW (5292)
 from GW (5292) --> FS (22428)
In other word, the FS now sending RTP to 5292 instead of 5294 (which was
intended in INVITE SDP message)

And, of course, I cannot hear the voice on GW side after this.

Anyone encounter this before? Are there any paramaters that might involved
in this auto changing RTP port behavior of FS?

Any direction for me is appreciated, I will play around with this, and post
back my result to community.

Dan Lan
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/985bd27e/attachment.html 

From anthony.minessale at gmail.com  Fri Sep  9 01:02:12 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 8 Sep 2011 16:02:12 -0500
Subject: [Freeswitch-users] One Way Audio - Auto Change RTP port?
In-Reply-To: <CAAuJ8MBnFKYzEjPnCNeNmiU-T=jRu3M1vcrR49-p2LPyRU=+uQ@mail.gmail.com>
References: <CAAuJ8MBnFKYzEjPnCNeNmiU-T=jRu3M1vcrR49-p2LPyRU=+uQ@mail.gmail.com>
Message-ID: <CAKbxfG8vU1pe-TU5_PHT8BcabYPceLem9QiyysCDxnf+gKSBYQ@mail.gmail.com>

variables on the leg in question

disable_rtp_auto_adjust=true

and/or (with today or later GIT)

rtp_manual_rtp_bugs=accept_any_packets


On Thu, Sep 8, 2011 at 3:50 PM, Dan Lan <danlanweb at gmail.com> wrote:
> Hi,
> I run into a weird situation. My media gateay handle voice call with 2
> different RTP ports for send & receive
>
> Here is what happened. (ps: both gateway and FS are all on public IP, no NAT
> involved)
> 1. Incoming call INVITE from gateway to FS
> Connection Information (c): IN IP4 100.100.100.100? (This is my media
> gateway IP address)
> Media Description, name and address (m): audio 5294 RTP/AVP 18 0 4
> 2. FS response with session progress with media information
> Connection Information (c): IN IP4 200.200.200.200
> Media Description, name and address (m): audio 22428 RTP/AVP 0
> 3. I start to see some RTP traffic exchange between FS and GW
> from FS (22428) --> GW (5294)
> from GW (5292)?--> FS (22428)
> please note: the GW use two DIFFERENT PORT for RTP, one for sending and one
> for receiving
> 4. For a while (about 5 secs, I think)
> The RTP flow change on FS side to become, (there is no RTCP packet during
> the time)
> from FS (22428) --> GW (5292)
> from GW (5292)?--> FS (22428)
> In other word, the FS now sending RTP to 5292 instead of 5294 (which was
> intended in INVITE SDP message)
>
> And, of course, I cannot hear the voice on GW side after this.
>
> Anyone encounter this before? Are there any paramaters that might involved
> in this auto changing RTP port behavior of FS?
>
> Any direction for me is appreciated, I will play around with this, and post
> back my result to community.
>
> Dan Lan
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jeff at jefflenk.com  Fri Sep  9 01:16:13 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 8 Sep 2011 14:16:13 -0700 (PDT)
Subject: [Freeswitch-users] FreeSwitch Windows ISO Hard Coded Install
	Path...
In-Reply-To: <CABWdUKC6S54SydH5P1cYEwxHn3Xq44e2gdYsuBrUeEB4fxGUpQ@mail.gmail.com>
References: <CABWdUKC6S54SydH5P1cYEwxHn3Xq44e2gdYsuBrUeEB4fxGUpQ@mail.gmail.com>
Message-ID: <1315516573459-6773821.post@n2.nabble.com>

No reason other than simplicity. Your patch would be welcome! Otherwise
perhaps someday I will fix that.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Windows-ISO-Hard-Coded-Install-Path-tp6772680p6773821.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Fri Sep  9 03:07:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Sep 2011 16:07:38 -0700
Subject: [Freeswitch-users] freeswitch as a gateway with cdr lookup
In-Reply-To: <20110907170333.154010@gmx.net>
References: <20110907170333.154010@gmx.net>
Message-ID: <CAKzWOxXXjswhvbZocT6aH7DouSGZBcrsk5SwRVs+mAhJtyNBMA@mail.gmail.com>

I'm sure this is possible, but I think we would like to learn more about
what's involved before we say for sure. What kind of PBX is it? And how
would you connect FS to your PBX and to your GSM gateway? SIP or something
else?

-MC

On Wed, Sep 7, 2011 at 10:03 AM, thomas peterseil <thomas.ji at gmx.at> wrote:

> Hello FreeSWITCH-Users,
>
> I am running a PBX with a GSM Gateway and i have problems with incoming
> calls on the GSM Gateway. I am wondering, if there is an "easy" possibility
> to solve it with FreeSwitch. Here is the problem:
>
> Extension 1000 calls the mobile phone 1234, but on the display of the
> mobile is the number 987 (i can?t modify this number, it?s the number of the
> GSM Gateway) and nobody picks up the call. later the mobile number 1234
> calls back the number 987, but the GSM Gateway has no idea who has called
> the 1234, so there is no way of routing the call back to the extension 1000.
>
> Is there a possibility to put a FS between my PBX and the GSM Gateway, so
> when a call from the GSM Gateway comes in, FS makes a lookup in the CDR to
> check, which extension called this mobile number last time and then FS
> should route the call to the right extension.
>
> Is it possible to realize that with FS and if yes, is that very difficult?
>
> Thanks in advanced for any help and suggestions.
>
> thomas
> --
> NEU: FreePhone - 0ct/min Handyspartarif mit Geld-zur?ck-Garantie!
> Jetzt informieren: http://www.gmx.net/de/go/freephone
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/3f2ffd8e/attachment.html 

From danlanweb at gmail.com  Fri Sep  9 03:17:29 2011
From: danlanweb at gmail.com (Dan Lan)
Date: Thu, 8 Sep 2011 16:17:29 -0700
Subject: [Freeswitch-users] One Way Audio - Auto Change RTP port?
In-Reply-To: <CAKbxfG8vU1pe-TU5_PHT8BcabYPceLem9QiyysCDxnf+gKSBYQ@mail.gmail.com>
References: <CAAuJ8MBnFKYzEjPnCNeNmiU-T=jRu3M1vcrR49-p2LPyRU=+uQ@mail.gmail.com>
	<CAKbxfG8vU1pe-TU5_PHT8BcabYPceLem9QiyysCDxnf+gKSBYQ@mail.gmail.com>
Message-ID: <CAAuJ8MDdz-EoaLRzR2es+aH0tURL9kN0K+6bQcKEpP_RJ0BEGw@mail.gmail.com>

Hi, Anthony:

Thanks for your direction. Could you give me a little bit more info about
where to set the parameters?

What I did was, I put the incoming GW IP into my ACL list, so my FS will
accept call from the GW.
I then create a public dialplan to transfer the incoming DID to a registered
SNOM phone with public IP address.
  <extension name="my_incoming_did">
    <condition field="destination_number" expression="^(999\d{10})$">
    <action application="transfer" data="$1 XML default"/>
    </condition>
  </extension>

After I add
<action application="set"  data="disable_rtp_auto_adjust=true"/>
before action "transfer"

The RTP flow become like this.
GW(5416) -->  FS (31326)
FS (31326) --> GW (5418)
This looks work fine on Leg A now without auto change the port (the incoming
leg)
However, something also change down the road on Leg B.
Now I got
SNOM(52934) --> FS (21464)
NO ANY RTP from FS --> SNOM ...

So now I still got one way voice, but is exact other way around. Before
change, the Leg B is working fine.

My question is where shoud I put
rtp_manual_rtp_bugs=accept_any_packets  ?
Do I have to put togerther this with disable_rtp_auto_adjust?

Did you just fix this problem (because you mentioned using today's git), so
I need to re-compile the most current git to fix this? (I am in window
version)

Thanks again.
Dan Lan
On Thu, Sep 8, 2011 at 2:02 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> variables on the leg in question
>
> disable_rtp_auto_adjust=true
>
> and/or (with today or later GIT)
>
> rtp_manual_rtp_bugs=accept_any_packets
>
>
> On Thu, Sep 8, 2011 at 3:50 PM, Dan Lan <danlanweb at gmail.com> wrote:
> > Hi,
> > I run into a weird situation. My media gateay handle voice call with 2
> > different RTP ports for send & receive
> >
> > Here is what happened. (ps: both gateway and FS are all on public IP, no
> NAT
> > involved)
> > 1. Incoming call INVITE from gateway to FS
> > Connection Information (c): IN IP4 100.100.100.100  (This is my media
> > gateway IP address)
> > Media Description, name and address (m): audio 5294 RTP/AVP 18 0 4
> > 2. FS response with session progress with media information
> > Connection Information (c): IN IP4 200.200.200.200
> > Media Description, name and address (m): audio 22428 RTP/AVP 0
> > 3. I start to see some RTP traffic exchange between FS and GW
> > from FS (22428) --> GW (5294)
> > from GW (5292) --> FS (22428)
> > please note: the GW use two DIFFERENT PORT for RTP, one for sending and
> one
> > for receiving
> > 4. For a while (about 5 secs, I think)
> > The RTP flow change on FS side to become, (there is no RTCP packet during
> > the time)
> > from FS (22428) --> GW (5292)
> > from GW (5292) --> FS (22428)
> > In other word, the FS now sending RTP to 5292 instead of 5294 (which was
> > intended in INVITE SDP message)
> >
> > And, of course, I cannot hear the voice on GW side after this.
> >
> > Anyone encounter this before? Are there any paramaters that might
> involved
> > in this auto changing RTP port behavior of FS?
> >
> > Any direction for me is appreciated, I will play around with this, and
> post
> > back my result to community.
> >
> > Dan Lan
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/a5bdc306/attachment-0001.html 

From msc at freeswitch.org  Fri Sep  9 03:25:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Sep 2011 16:25:02 -0700
Subject: [Freeswitch-users] Sending DTMF on B-leg
In-Reply-To: <9B18FF40055B495BBAE3EDEEDB7388B0@ccs.local>
References: <2934141FC0D9453B9150F7BD307120F3@ccs.local>
	<CALNZuwkMZoRx-HbRBkHg5FSutwV2xW2GWiQQEz2dsmAh2WoPhA@mail.gmail.com>
	<9B18FF40055B495BBAE3EDEEDB7388B0@ccs.local>
Message-ID: <CAKzWOxXVFFhB25Za9xdmkcTWm-76nFfX7hjn5JFU6gfX+Jxyvg@mail.gmail.com>

Are you saying that no DTMFs are being sent out when the bridge is initially
completed?
-MC

On Wed, Sep 7, 2011 at 3:17 PM, Wayne <wayne at hamilton.net> wrote:

> **
> I have tried that
> <extension name="test-tone" >
>    <condition field="destination_number" expression="(9025556747)" >
>        <action application="set" data="call_direction=outbound" />
>        <action application="set"
> data="effective_caller_id_name=${outbound_caller_id_name}" />
>        <action application="set"
> data="effective_caller_id_number=${outbound_caller_id_number}" />
>        <action application="set" data="continue_on_fail=true" />
>        <action application="queue_dtmf" data="0123456789" />
>        <action application="bridge" data="freetdm/wp1/a/$1" />
>    </condition>
> </extension>
>
> The debug tells me that this
> tail -f freeswitch.log | grep -i dtmf
>
> Dialplan: sofia/internal/1333 at 192.168.48.87 Action queue_dtmf(0123456789)
> EXECUTE sofia/internal/1333 at 192.168.48.87 queue_dtmf(0123456789)
> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
> sofia/internal/1333 at 192.168.48.87 Queue dtmf
> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
> sofia/internal/1333 at 192.168.48.87 Queue dtmf
> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
> sofia/internal/1333 at 192.168.48.87 Queue dtmf
> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
> sofia/internal/1333 at 192.168.48.87 Queue dtmf
> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
> sofia/internal/1333 at 192.168.48.87 Queue dtmf
> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
> sofia/internal/1333 at 192.168.48.87 Queue dtmf
> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
> sofia/internal/1333 at 192.168.48.87 Queue dtmf
> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
> sofia/internal/1333 at 192.168.48.87 Queue dtmf
> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
> sofia/internal/1333 at 192.168.48.87 Queue dtmf
> 2011-09-07 17:13:44.676204 [DEBUG] switch_channel.c:465
> sofia/internal/1333 at 192.168.48.87 Queue dtmf
> 2011-09-07 17:13:45.076218 [DEBUG] sofia_glue.c:3355 Set 2833 dtmf send
> payload to 101
> 2011-09-07 17:13:45.076218 [DEBUG] sofia_glue.c:3360 Set 2833 dtmf receive
> payload to 101
> 2011-09-07 17:13:45.116216 [DEBUG] ftdm_io.c:3714 [s1c1][1:1] Generating
> DTMF [0123456789]
> 2011-09-07 17:13:47.876238 [DEBUG] ftmod_wanpipe.c:701 [s1c1][1:1] Enabled
> DTMF events
> 2011-09-07 17:13:47.956228 [DEBUG] ftmod_wanpipe.c:1415 [s1c1][1:1] Queuing
> wanpipe DTMF: 9
> 2011-09-07 17:13:47.956228 [DEBUG] ftdm_io.c:3524 [s1c1][1:1] Queuing DTMF
> 9 (debug = 0)
> 2011-09-07 17:13:47.956228 [DEBUG] mod_freetdm.c:733 Queuing DTMF [9] in
> channel FreeTDM/1:1/9025556747
> 2011-09-07 17:13:48.036235 [DEBUG] ftmod_wanpipe.c:1415 [s1c1][1:1] Queuing
> wanpipe DTMF: 9
> 2011-09-07 17:13:48.036235 [DEBUG] ftdm_io.c:3524 [s1c1][1:1] Queuing DTMF
> 9 (debug = 0)
> 2011-09-07 17:13:48.036235 [DEBUG] mod_freetdm.c:733 Queuing DTMF [9] in
> channel FreeTDM/1:1/9025556747
>
> So where would I look next.
>
> Wayne
>
>
>  ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus
> *Sent:* Wednesday, September 07, 2011 3:55 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Sending DTMF on B-leg
>
>  I think you looking for queue_dtmf.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_queue_dtmf
>
> -Avi Marcus
>
>
>
> On Wed, Sep 7, 2011 at 11:47 PM, Wayne <wayne at hamilton.net> wrote:
> > Hello All,
> >
> > I need to call out on a SIP trunk and when the call is answered send DTMF
> > tones.
> > I need to send the DTMF only on the outbound leg.
> >
> > Does anyone have a dialplan that will do that? Is it possible. I have
> only
> > found one thread on it and did get much out of it.
> >
> > Thanks
> > Wayne
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/29f5a124/attachment.html 

From msc at freeswitch.org  Fri Sep  9 03:26:04 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Sep 2011 16:26:04 -0700
Subject: [Freeswitch-users] Finding User's Current Session
In-Reply-To: <CABK0fC6w9z2rb-yLQQpCbwLT1xfidQVj_NBFXpAgkB9Pz-Bn1w@mail.gmail.com>
References: <CABK0fC7GLpqneudT753L3m2W72jHTVW82t3BZNmCe843O7cZCA@mail.gmail.com>
	<1315240242664-6761305.post@n2.nabble.com>
	<CABK0fC6w9z2rb-yLQQpCbwLT1xfidQVj_NBFXpAgkB9Pz-Bn1w@mail.gmail.com>
Message-ID: <CAKzWOxUgOM=TGkQqsAtUzNn1u0CZutJkT3yC99U9XZ7DZfKM5Q@mail.gmail.com>

On Mon, Sep 5, 2011 at 11:46 AM, afshin afzali <a.afzali2003 at gmail.com>wrote:

> I hoped an API which accept user name as input param !
> Thanks
>
> What if the user has more than one call in progress?

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/9db294cc/attachment.html 

From msc at freeswitch.org  Fri Sep  9 03:37:23 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Sep 2011 16:37:23 -0700
Subject: [Freeswitch-users] Sophia authentication without using editing
	acl.conf.xml
In-Reply-To: <CALH5ZrbVN5hZ6DTo0PHM0NPMBSv8JtObrXoF7fZjC4KeOW1zWg@mail.gmail.com>
References: <CALH5ZrZPtg+3iaR+=s+oNaumo=wD27MQn7pY4UyJEQ-5t0QiUA@mail.gmail.com>
	<B5F566F1-4244-475E-94D6-52963B5DCC05@tech21.com>
	<CALH5ZrbVN5hZ6DTo0PHM0NPMBSv8JtObrXoF7fZjC4KeOW1zWg@mail.gmail.com>
Message-ID: <CAKzWOxUk9C-ZJyt0AcYxuiAzVtYV8-j=yna+Cz7Ms+BiUFaHXA@mail.gmail.com>

Technically you can do the IP auth right in your script if you know what IP
range(s) to add. Look at the request sent to your server from mod_xml_curl
and check the sip_from_host value against your list.

-MC

On Tue, Sep 6, 2011 at 10:20 PM, Joe Flemmings <joe.jflemmings at gmail.com>wrote:

> I tried that but it seams the acl has to be already defined in
> acl.conf.xml
>
>
> On Tue, Sep 6, 2011 at 10:04 PM, Brad Mina <brad at tech21.com> wrote:
>
>> I believe you can add an acl param in a user's XML.
>>
>> Sent from my iPhone
>>
>> On Sep 6, 2011, at 8:21 PM, Joe Flemmings <joe.jflemmings at gmail.com>
>> wrote:
>>
>>
>> I use xml_curl to authenticate sip devices and was wondering if there is a
>> way to do IP authentication without having to edit and reaload acl.conf.xml
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110908/c10ef6cb/attachment.html 

From Glen.Ganderton at premier.com.au  Fri Sep  9 03:52:56 2011
From: Glen.Ganderton at premier.com.au (Glen Ganderton)
Date: Fri, 9 Sep 2011 09:52:56 +1000
Subject: [Freeswitch-users] What is Event MEDIA_BUG_STOP
In-Reply-To: <1315497793886-6772422.post@n2.nabble.com>
References: <CD5A1499C8DEF34FAA3509722B00393E74029E2ACD@m-mail03.office.premiertech>
	<CAFiqYu=qXsrrmHrcHvt+BOUr+JkFiKZUraNyM1o6AKb5iNLEyQ@mail.gmail.com>
	<CAJiHjvf3uyHQA8whgXqsTOJakyfDfwTr2bjRzWECcdOYt5Agrg@mail.gmail.com>
	<1315497793886-6772422.post@n2.nabble.com>
Message-ID: <CD5A1499C8DEF34FAA3509722B00393E74029E2BC7@m-mail03.office.premiertech>

Could you give me an example of how to set this variable from the dialplan. Thanks

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk
Sent: Friday, 9 September 2011 2:03 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] What is Event MEDIA_BUG_STOP

Do you have the channel variable "fire_asr_events" set?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/What-is-Event-MEDIA-BUG-STOP-tp6770735p6772422.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 jeff at jefflenk.com  Fri Sep  9 06:01:40 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 8 Sep 2011 19:01:40 -0700 (PDT)
Subject: [Freeswitch-users] What is Event MEDIA_BUG_STOP
In-Reply-To: <CD5A1499C8DEF34FAA3509722B00393E74029E2BC7@m-mail03.office.premiertech>
References: <CD5A1499C8DEF34FAA3509722B00393E74029E2ACD@m-mail03.office.premiertech>
	<CAFiqYu=qXsrrmHrcHvt+BOUr+JkFiKZUraNyM1o6AKb5iNLEyQ@mail.gmail.com>
	<CAJiHjvf3uyHQA8whgXqsTOJakyfDfwTr2bjRzWECcdOYt5Agrg@mail.gmail.com>
	<1315497793886-6772422.post@n2.nabble.com>
	<CD5A1499C8DEF34FAA3509722B00393E74029E2BC7@m-mail03.office.premiertech>
Message-ID: <1315533700949-6774465.post@n2.nabble.com>

<action application="set" data="fire_asr_events=1"/>


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/What-is-Event-MEDIA-BUG-STOP-tp6770735p6774465.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From wagnerspi at gmail.com  Fri Sep  9 07:22:12 2011
From: wagnerspi at gmail.com (Wagner)
Date: Fri, 9 Sep 2011 00:22:12 -0300
Subject: [Freeswitch-users] [FreeSwitch-users] Calls being dropped
Message-ID: <CAO0ino=Zw2de9-Gx=xg3v=fKL4_CZ3+8LKicuXyUZPSoraXSgw@mail.gmail.com>

Hello Guys,

I'm having some problems with calls, they are being dropped often, this is
the output in fs_cli

2011-09-09 07:13:05.644007 [DEBUG] switch_ivr_play_say.c:1152 Codec
Activated L16 at 8000hz 1 channels 20ms
2011-09-09 07:13:06.182083 [DEBUG] switch_ivr_play_say.c:1444 done playing
file
2011-09-09 07:13:06.184311 [DEBUG] switch_ivr_play_say.c:1152 Codec
Activated L16 at 8000hz 1 channels 20ms
2011-09-09 07:13:06.762395 [DEBUG] switch_ivr_play_say.c:1444 done playing
file
2011-09-09 07:13:06.884005 [WARNING] switch_core_file.c:176 Sample rate
doesn't match
2011-09-09 07:13:06.884005 [DEBUG] switch_ivr_play_say.c:1152 Codec
Activated L16 at 8000hz 1 channels 20ms
2011-09-09 07:13:12.665209 [DEBUG] sofia.c:4153 Channel
sofia/internal/XXX at x.x.x.x entering state [terminating][0]
2011-09-09 07:13:12.665209 [NOTICE] sofia.c:4789 Hangup
sofia/internal/XXX at x.x.x.x [CS_EXECUTE] [NORMAL_UNSPECIFIED]
2011-09-09 07:13:12.665209 [DEBUG] switch_channel.c:2102 Send signal
sofia/internal/XXX at x.x.x.x [KILL]
2011-09-09 07:13:12.665209 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/XXX at x.x.x.x [BREAK]
2011-09-09 07:13:12.682339 [DEBUG] switch_ivr_play_say.c:1444 done playing
file
2011-09-09 07:13:12.687516 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/XXX at x.x.x.x) State EXECUTE going to sleep
2011-09-09 07:13:12.687516 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/XXX at x.x.x.x) Running State Change CS_HANGUP
2011-09-09 07:13:12.688525 [DEBUG] switch_core_media_bug.c:413 Removing BUG
from sofia/internal/XXX at x.x.x.x
2011-09-09 07:13:12.689932 [DEBUG] switch_core_state_machine.c:499
(sofia/internal/XXX at x.x.x.x) State HANGUP
2011-09-09 07:13:12.689932 [DEBUG] mod_sofia.c:414 Channel
sofia/internal/XXX at x.x.x.x hanging up, cause: NORMAL_UNSPECIFIED
2011-09-09 07:13:12.697113 [DEBUG] switch_core_state_machine.c:46
sofia/internal/XXX at x.x.x.x Standard HANGUP, cause: NORMAL_UNSPECIFIED
2011-09-09 07:13:12.698117 [DEBUG] switch_core_state_machine.c:499
(sofia/internal/XXX at x.x.x.x) State HANGUP going to sleep
2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/XXX at x.x.x.x) State Change CS_HANGUP -> CS_REPORTING
2011-09-09 07:13:12.699153 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/XXX at x.x.x.x [BREAK]
2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/XXX at x.x.x.x) Running State Change CS_REPORTING
2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:590
(sofia/internal/XXX at x.x.x.x) State REPORTING
2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:53
sofia/internal/XXX at x.x.x.x Standard REPORTING, cause: NORMAL_UNSPECIFIED
2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:590
(sofia/internal/XXX at x.x.x.x) State REPORTING going to sleep
2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/XXX at x.x.x.x) State Change CS_REPORTING -> CS_DESTROY
2011-09-09 07:13:12.700174 [DEBUG] switch_core_session.c:1021 Send signal
sofia/internal/XXX at x.x.x.x [BREAK]
2011-09-09 07:13:12.700174 [DEBUG] switch_core_session.c:1164 Session 40
(sofia/internal/XXX at x.x.x.x) Locked, Waiting on external entities
2011-09-09 07:13:12.700174 [NOTICE] switch_core_session.c:1182 Session 40
(sofia/internal/XXX at x.x.x.x) Ended
2011-09-09 07:13:12.700174 [NOTICE] switch_core_session.c:1184 Close Channel
sofia/internal/XXX at x.x.x.x [CS_DESTROY]
2011-09-09 07:13:12.701183 [DEBUG] switch_core_state_machine.c:428
(sofia/internal/XXX at x.x.x.x) Running State Change CS_DESTROY
2011-09-09 07:13:12.701183 [DEBUG] switch_core_state_machine.c:439
(sofia/internal/XXX at x.x.x.x) State DESTROY
2011-09-09 07:13:12.701183 [DEBUG] mod_sofia.c:341
sofia/internal/XXX at x.x.x.x SOFIA DESTROY
2011-09-09 07:13:12.701183 [DEBUG] switch_core_state_machine.c:60
sofia/internal/XXX at x.x.x.x Standard DESTROY
2011-09-09 07:13:12.701183 [DEBUG] switch_core_state_machine.c:439
(sofia/internal/XXX at x.x.x.x) State DESTROY going to sleep


what could it be?

Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110909/9c82f4f9/attachment.html 

From govoiper at gmail.com  Fri Sep  9 09:02:16 2011
From: govoiper at gmail.com (Sam Govind)
Date: Fri, 9 Sep 2011 10:02:16 +0500
Subject: [Freeswitch-users] [FreeSwitch-users] Calls being dropped
In-Reply-To: <CAO0ino=Zw2de9-Gx=xg3v=fKL4_CZ3+8LKicuXyUZPSoraXSgw@mail.gmail.com>
References: <CAO0ino=Zw2de9-Gx=xg3v=fKL4_CZ3+8LKicuXyUZPSoraXSgw@mail.gmail.com>
Message-ID: <CAJUJwthkA6QZqq3jF=mYG7cd0wC5ATPAdEs5Eb2R-uokutC0ww@mail.gmail.com>

Check codecs of both legs.Also I suspect the dialplan hitting a dead-end or
something.

On Fri, Sep 9, 2011 at 8:22 AM, Wagner <wagnerspi at gmail.com> wrote:

> Hello Guys,
>
> I'm having some problems with calls, they are being dropped often, this is
> the output in fs_cli
>
> 2011-09-09 07:13:05.644007 [DEBUG] switch_ivr_play_say.c:1152 Codec
> Activated L16 at 8000hz 1 channels 20ms
> 2011-09-09 07:13:06.182083 [DEBUG] switch_ivr_play_say.c:1444 done playing
> file
> 2011-09-09 07:13:06.184311 [DEBUG] switch_ivr_play_say.c:1152 Codec
> Activated L16 at 8000hz 1 channels 20ms
> 2011-09-09 07:13:06.762395 [DEBUG] switch_ivr_play_say.c:1444 done playing
> file
> 2011-09-09 07:13:06.884005 [WARNING] switch_core_file.c:176 Sample rate
> doesn't match
> 2011-09-09 07:13:06.884005 [DEBUG] switch_ivr_play_say.c:1152 Codec
> Activated L16 at 8000hz 1 channels 20ms
> 2011-09-09 07:13:12.665209 [DEBUG] sofia.c:4153 Channel
> sofia/internal/XXX at x.x.x.x entering state [terminating][0]
> 2011-09-09 07:13:12.665209 [NOTICE] sofia.c:4789 Hangup
> sofia/internal/XXX at x.x.x.x [CS_EXECUTE] [NORMAL_UNSPECIFIED]
> 2011-09-09 07:13:12.665209 [DEBUG] switch_channel.c:2102 Send signal
> sofia/internal/XXX at x.x.x.x [KILL]
> 2011-09-09 07:13:12.665209 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/XXX at x.x.x.x [BREAK]
> 2011-09-09 07:13:12.682339 [DEBUG] switch_ivr_play_say.c:1444 done playing
> file
> 2011-09-09 07:13:12.687516 [DEBUG] switch_core_state_machine.c:348
> (sofia/internal/XXX at x.x.x.x) State EXECUTE going to sleep
> 2011-09-09 07:13:12.687516 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/XXX at x.x.x.x) Running State Change CS_HANGUP
> 2011-09-09 07:13:12.688525 [DEBUG] switch_core_media_bug.c:413 Removing BUG
> from sofia/internal/XXX at x.x.x.x
> 2011-09-09 07:13:12.689932 [DEBUG] switch_core_state_machine.c:499
> (sofia/internal/XXX at x.x.x.x) State HANGUP
> 2011-09-09 07:13:12.689932 [DEBUG] mod_sofia.c:414 Channel
> sofia/internal/XXX at x.x.x.x hanging up, cause: NORMAL_UNSPECIFIED
> 2011-09-09 07:13:12.697113 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/XXX at x.x.x.x Standard HANGUP, cause: NORMAL_UNSPECIFIED
> 2011-09-09 07:13:12.698117 [DEBUG] switch_core_state_machine.c:499
> (sofia/internal/XXX at x.x.x.x) State HANGUP going to sleep
> 2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/XXX at x.x.x.x) State Change CS_HANGUP -> CS_REPORTING
> 2011-09-09 07:13:12.699153 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/XXX at x.x.x.x [BREAK]
> 2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/XXX at x.x.x.x) Running State Change CS_REPORTING
> 2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:590
> (sofia/internal/XXX at x.x.x.x) State REPORTING
> 2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/XXX at x.x.x.x Standard REPORTING, cause: NORMAL_UNSPECIFIED
> 2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:590
> (sofia/internal/XXX at x.x.x.x) State REPORTING going to sleep
> 2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/XXX at x.x.x.x) State Change CS_REPORTING -> CS_DESTROY
> 2011-09-09 07:13:12.700174 [DEBUG] switch_core_session.c:1021 Send signal
> sofia/internal/XXX at x.x.x.x [BREAK]
> 2011-09-09 07:13:12.700174 [DEBUG] switch_core_session.c:1164 Session 40
> (sofia/internal/XXX at x.x.x.x) Locked, Waiting on external entities
> 2011-09-09 07:13:12.700174 [NOTICE] switch_core_session.c:1182 Session 40
> (sofia/internal/XXX at x.x.x.x) Ended
> 2011-09-09 07:13:12.700174 [NOTICE] switch_core_session.c:1184 Close
> Channel sofia/internal/XXX at x.x.x.x [CS_DESTROY]
> 2011-09-09 07:13:12.701183 [DEBUG] switch_core_state_machine.c:428
> (sofia/internal/XXX at x.x.x.x) Running State Change CS_DESTROY
> 2011-09-09 07:13:12.701183 [DEBUG] switch_core_state_machine.c:439
> (sofia/internal/XXX at x.x.x.x) State DESTROY
> 2011-09-09 07:13:12.701183 [DEBUG] mod_sofia.c:341
> sofia/internal/XXX at x.x.x.x SOFIA DESTROY
> 2011-09-09 07:13:12.701183 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/XXX at x.x.x.x Standard DESTROY
> 2011-09-09 07:13:12.701183 [DEBUG] switch_core_state_machine.c:439
> (sofia/internal/XXX at x.x.x.x) State DESTROY going to sleep
>
>
> what could it be?
>
> Thanks
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110909/1fc28e2d/attachment.html 

From joe.jflemmings at gmail.com  Fri Sep  9 13:33:04 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Fri, 9 Sep 2011 02:33:04 -0700
Subject: [Freeswitch-users] Sophia authentication without using editing
	acl.conf.xml
In-Reply-To: <CAKzWOxUk9C-ZJyt0AcYxuiAzVtYV8-j=yna+Cz7Ms+BiUFaHXA@mail.gmail.com>
References: <CALH5ZrZPtg+3iaR+=s+oNaumo=wD27MQn7pY4UyJEQ-5t0QiUA@mail.gmail.com>
	<B5F566F1-4244-475E-94D6-52963B5DCC05@tech21.com>
	<CALH5ZrbVN5hZ6DTo0PHM0NPMBSv8JtObrXoF7fZjC4KeOW1zWg@mail.gmail.com>
	<CAKzWOxUk9C-ZJyt0AcYxuiAzVtYV8-j=yna+Cz7Ms+BiUFaHXA@mail.gmail.com>
Message-ID: <CALH5ZraBjmTAiuzHRy=9Nkrp5tdp5kJi2t7S9NigY8kui14w7A@mail.gmail.com>

The request is never sent to mod_xml_curl. This was the first thing i
checked.

On Thu, Sep 8, 2011 at 4:37 PM, Michael Collins <msc at freeswitch.org> wrote:

> Technically you can do the IP auth right in your script if you know what IP
> range(s) to add. Look at the request sent to your server from mod_xml_curl
> and check the sip_from_host value against your list.
>
> -MC
>
> On Tue, Sep 6, 2011 at 10:20 PM, Joe Flemmings <joe.jflemmings at gmail.com>wrote:
>
>> I tried that but it seams the acl has to be already defined in
>> acl.conf.xml
>>
>>
>> On Tue, Sep 6, 2011 at 10:04 PM, Brad Mina <brad at tech21.com> wrote:
>>
>>> I believe you can add an acl param in a user's XML.
>>>
>>> Sent from my iPhone
>>>
>>> On Sep 6, 2011, at 8:21 PM, Joe Flemmings <joe.jflemmings at gmail.com>
>>> wrote:
>>>
>>>
>>> I use xml_curl to authenticate sip devices and was wondering if there is
>>> a way to do IP authentication without having to edit and reaload
>>> acl.conf.xml
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110909/38a64c7f/attachment-0001.html 

From avi at avimarcus.net  Fri Sep  9 14:00:36 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 9 Sep 2011 13:00:36 +0300
Subject: [Freeswitch-users] Finding User's Current Session
In-Reply-To: <CAKzWOxUgOM=TGkQqsAtUzNn1u0CZutJkT3yC99U9XZ7DZfKM5Q@mail.gmail.com>
References: <CABK0fC7GLpqneudT753L3m2W72jHTVW82t3BZNmCe843O7cZCA@mail.gmail.com>
	<1315240242664-6761305.post@n2.nabble.com>
	<CABK0fC6w9z2rb-yLQQpCbwLT1xfidQVj_NBFXpAgkB9Pz-Bn1w@mail.gmail.com>
	<CAKzWOxUgOM=TGkQqsAtUzNn1u0CZutJkT3yC99U9XZ7DZfKM5Q@mail.gmail.com>
Message-ID: <CALNZuwm-2n7dCVkzjmGMDJYTZ10ZNOsrVLZA3u8CQ=M=gnBgwQ@mail.gmail.com>

Well, it would be nifty for the "show calls" to show the
variable_accountcode and/or the variable_user_name from the channels. I'm
not sure if that data is in the same place or it require additional queries
to get, though..
As of now, afaik, you have to query all of them and then get the info on
each channel to find the auth user or account code.

-Avi


On Fri, Sep 9, 2011 at 2:26 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Sep 5, 2011 at 11:46 AM, afshin afzali <a.afzali2003 at gmail.com>wrote:
>
>> I hoped an API which accept user name as input param !
>> Thanks
>>
>> What if the user has more than one call in progress?
>
> -MC
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110909/c52c2e60/attachment.html 

From kbdfck at gmail.com  Fri Sep  9 15:16:19 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Fri, 9 Sep 2011 15:16:19 +0400
Subject: [Freeswitch-users] Sophia authentication without using editing
	acl.conf.xml
In-Reply-To: <CALH5ZraBjmTAiuzHRy=9Nkrp5tdp5kJi2t7S9NigY8kui14w7A@mail.gmail.com>
References: <CALH5ZrZPtg+3iaR+=s+oNaumo=wD27MQn7pY4UyJEQ-5t0QiUA@mail.gmail.com>
	<B5F566F1-4244-475E-94D6-52963B5DCC05@tech21.com>
	<CALH5ZrbVN5hZ6DTo0PHM0NPMBSv8JtObrXoF7fZjC4KeOW1zWg@mail.gmail.com>
	<CAKzWOxUk9C-ZJyt0AcYxuiAzVtYV8-j=yna+Cz7Ms+BiUFaHXA@mail.gmail.com>
	<CALH5ZraBjmTAiuzHRy=9Nkrp5tdp5kJi2t7S9NigY8kui14w7A@mail.gmail.com>
Message-ID: <CA+GmxbYRhaLOZm7ami4zEa9nkMYESE24baK-13vbXdzcH-3PrA@mail.gmail.com>

How can we respond with specific error code to xml_curl request in case we
didn't find user or it was denied by ACL? It would be nice to have ability
to control what FS will respond based on xml_curl result.

2011/9/9 Joe Flemmings <joe.jflemmings at gmail.com>

> The request is never sent to mod_xml_curl. This was the first thing i
> checked.
>
>
> On Thu, Sep 8, 2011 at 4:37 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Technically you can do the IP auth right in your script if you know what
>> IP range(s) to add. Look at the request sent to your server from
>> mod_xml_curl and check the sip_from_host value against your list.
>>
>> -MC
>>
>> On Tue, Sep 6, 2011 at 10:20 PM, Joe Flemmings <joe.jflemmings at gmail.com>wrote:
>>
>>> I tried that but it seams the acl has to be already defined in
>>> acl.conf.xml
>>>
>>>
>>> On Tue, Sep 6, 2011 at 10:04 PM, Brad Mina <brad at tech21.com> wrote:
>>>
>>>> I believe you can add an acl param in a user's XML.
>>>>
>>>> Sent from my iPhone
>>>>
>>>> On Sep 6, 2011, at 8:21 PM, Joe Flemmings <joe.jflemmings at gmail.com>
>>>> wrote:
>>>>
>>>>
>>>> I use xml_curl to authenticate sip devices and was wondering if there is
>>>> a way to do IP authentication without having to edit and reaload
>>>> acl.conf.xml
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110909/f105d5e0/attachment.html 

From vipkilla at gmail.com  Fri Sep  9 16:23:48 2011
From: vipkilla at gmail.com (vip killa)
Date: Fri, 9 Sep 2011 08:23:48 -0400
Subject: [Freeswitch-users] FreeSWITCH failover/HA
In-Reply-To: <CAHMdK5nhYOixxkh2DSrf0qQGmc8LLhFbZsTuq_pRGgWGgM-t8w@mail.gmail.com>
References: <CAJWEF2ZPcrVirCFKCtHOugntRZ5sETDpDPCZyTRNso8OKMaRJA@mail.gmail.com>
	<CAHMdK5nhYOixxkh2DSrf0qQGmc8LLhFbZsTuq_pRGgWGgM-t8w@mail.gmail.com>
Message-ID: <CAC-LwPOenhYwEr7Gs9VYYKNgnjm8PQHyYCqD11WxDY-707qW=Q@mail.gmail.com>

- I removed the dispatcher module because I didn't need to support
registration

Do the UA's register to opensips then? i contributed to that wiki (the
opensips install part) but I'm new to opensips, could you explain how this
works? Thanks.

On Thu, Sep 8, 2011 at 4:41 PM, Matt Stockton <mstockton at harqen.com> wrote:

> Hi Mateusz,
>
> I was just able to recently get this working using OpenSIPS and the
> instructions found here:
> http://wiki.freeswitch.com/wiki/Enterprise_deployment_OpenSIPS
>
> There were a few modifications I had to make, for example:
> - dlg_flag seems to be no longer available in the newest OpenSips, but I do
> not think it is needed.
> - I removed the dispatcher module because I didn't need to support
> registration
> - I had to change the INVITE conditional to be dependent on whether or not
> the invite was coming from outside (e.g. PSTN) or was coming from one of my
> freeswitch servers. For loadbalancing coming in, the load_balancer module
> worked fine as configured in the wiki instructions. For outbound, I used the
> dynamic routing module:
> http://www.unixnews.net/2010/09/dynamic-routing-with-opensips.html
>
> It seems to be working well so far. I will plan to update the wiki with
> more information. Feel free to reach out to me if you have any questions.
>
> Thanks,
> Matt
>
> On Thu, Sep 8, 2011 at 1:59 PM, Mateusz Bartczak <netcentrica at gmail.com>wrote:
>
>> Hi
>>
>> I would like to implement following scenario:
>>
>> 1. Central SIP proxy with 1 IP address that will redirect all incoming and
>> outgoing SIP traffic to two SIP application servers (FreeSwitch based).
>> Proxy will know online/offline status of each box and route calls only to
>> active one. I need it as a central point because IP address is authorized
>> with my providers. Also providers route incoming calls to that IP address.
>> My providers can't automatically reroute traffic to other server, it can be
>> done manually but it's not fast to do.
>>
>> 2. Mini cluster of two FreeSwitch boxes, box 2 will be exact copy of box
>> 1. If box 1 fails, central server should automatically route all traffic to
>> box 2.
>>
>> Do you have any suggestions how to implement this scenario? I think that
>> it should be easy to do, but have no idea where to start.
>>
>> Best Regards,
>> Mateusz
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110909/bf243815/attachment.html 

From dgarcia at anew.com.ve  Fri Sep  9 17:23:55 2011
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Fri, 09 Sep 2011 08:53:55 -0430
Subject: [Freeswitch-users] =?utf-8?q?screen_pop_with_freeswitch=EF=BC=9F?=
Message-ID: <4E6A136B.7070304@anew.com.ve>

Hi,

I am playing with freeswitch scripting with lua. I want call to number,
ex 1000, a script lua will process the call ( as an IVR), then transfer
the call to an extension (or transfer the call to a queue, to get a free
agent).
Questions:
How I add a pair data (key/value. ex ID/1234) to the call in the lua
script?
How can check the data attached is avaliable to the end extension?
What you use to make screen pop call data when the call ring in the
extension?

Thanks



From peter.olsson at visionutveckling.se  Fri Sep  9 17:36:46 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 9 Sep 2011 15:36:46 +0200
Subject: [Freeswitch-users]
 =?utf-8?q?screen_pop_with_freeswitch=EF=BC=9F?=
In-Reply-To: <4E6A136B.7070304@anew.com.ve>
References: <4E6A136B.7070304@anew.com.ve>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F215DAFA@cooper>

I think ESL is the way to go (http://wiki.freeswitch.org/wiki/Event_Socket_Library).

You can add data to the call by setting channel variables, these variables can then be read out in the ESL connection. ESL uses a simple TCP socket, so it can be used on the client's computers.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Saugort Dario Garcia Tovar
Skickat: den 9 september 2011 15:24
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] screen pop with freeswitch?

Hi,

I am playing with freeswitch scripting with lua. I want call to number,
ex 1000, a script lua will process the call ( as an IVR), then transfer
the call to an extension (or transfer the call to a queue, to get a free
agent).
Questions:
How I add a pair data (key/value. ex ID/1234) to the call in the lua
script?
How can check the data attached is avaliable to the end extension?
What you use to make screen pop call data when the call ring in the
extension?

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

!DSPAM:4e6a143432761570634090!


From t.mahe at telemaque.fr  Fri Sep  9 17:40:11 2011
From: t.mahe at telemaque.fr (=?UTF-8?B?VHJpc3RhbiBNYWjDqQ==?=)
Date: Fri, 09 Sep 2011 15:40:11 +0200
Subject: [Freeswitch-users]
	=?utf-8?q?screen_pop_with_freeswitch=EF=BC=9F?=
In-Reply-To: <4E6A136B.7070304@anew.com.ve>
References: <4E6A136B.7070304@anew.com.ve>
Message-ID: <4E6A173B.1020209@telemaque.fr>

Hi,

Some leads in the text, or as Peter told, you can also use ESL that 
maybe 'll scale better...

Regards,

Gled.

Le 09/09/2011 15:23, Saugort Dario Garcia Tovar a ?crit :
> Hi,
>
> I am playing with freeswitch scripting with lua. I want call to number,
> ex 1000, a script lua will process the call ( as an IVR), then transfer
> the call to an extension (or transfer the call to a queue, to get a free
> agent).
> Questions:
> How I add a pair data (key/value. ex ID/1234) to the call in the lua
> script?
Check channel vars ( set / get / export ).
> How can check the data attached is avaliable to the end extension?
Read vars ( getVar ).
> What you use to make screen pop call data when the call ring in the
> extension?
>
use the execute_on_answer or execute_on_ring variables, they allow you 
to fire a script when that condition is met.
> 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

-- 
**********************************
* Tristan Mah?                   *
* TELEMAQUE                      *
* 80 route des Lucioles          *
* 06560 Valbonne                 *
* Tel: +33 4.92.90.99.85         *
* Mob: +33 6.24.16.43.01         *
**********************************



From avi at avimarcus.net  Fri Sep  9 18:11:46 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 9 Sep 2011 17:11:46 +0300
Subject: [Freeswitch-users] How to use sofia_count_reg ?
In-Reply-To: <1315499174093-6772541.post@n2.nabble.com>
References: <CALNZuw=B3YYD1eWEMKERWtpitQBDv-AJKgw03k4mGCdZDfOQtA@mail.gmail.com>
	<1315499174093-6772541.post@n2.nabble.com>
Message-ID: <CALNZuwmi1GKxDc9EWPg7+rmY9mEF1kTEYDP9xgtc8NSajmNVRA@mail.gmail.com>

Even after adding the column, all I can get is a 0 or -1 response regardless
of what I put in.
sofia_count_reg 1000 at domain -> 0
sofia_count_reg 1000@ -> -1

on: git-10d2e80 2011-09-08 22-35-20 -0500

So, broken, or I'm just using it completely wrong?

-Avi


On Thu, Sep 8, 2011 at 7:26 PM, Jeff Lenk <jeff at jefflenk.com> wrote:

> This should be reported to Jira with the missing column
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/How-to-use-sofia-count-reg-tp6770571p6772541.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110909/d5a2af42/attachment.html 

From jeff at jefflenk.com  Fri Sep  9 18:47:49 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 9 Sep 2011 07:47:49 -0700 (PDT)
Subject: [Freeswitch-users] How to use sofia_count_reg ?
In-Reply-To: <CALNZuwmi1GKxDc9EWPg7+rmY9mEF1kTEYDP9xgtc8NSajmNVRA@mail.gmail.com>
References: <CALNZuw=B3YYD1eWEMKERWtpitQBDv-AJKgw03k4mGCdZDfOQtA@mail.gmail.com>
	<1315499174093-6772541.post@n2.nabble.com>
	<CALNZuwmi1GKxDc9EWPg7+rmY9mEF1kTEYDP9xgtc8NSajmNVRA@mail.gmail.com>
Message-ID: <1315579669278-6776183.post@n2.nabble.com>

I looks like that method needs rework.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-use-sofia-count-reg-tp6770571p6776183.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Fri Sep  9 19:48:48 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 9 Sep 2011 10:48:48 -0500
Subject: [Freeswitch-users] Finding User's Current Session
In-Reply-To: <CALNZuwm-2n7dCVkzjmGMDJYTZ10ZNOsrVLZA3u8CQ=M=gnBgwQ@mail.gmail.com>
References: <CABK0fC7GLpqneudT753L3m2W72jHTVW82t3BZNmCe843O7cZCA@mail.gmail.com>
	<1315240242664-6761305.post@n2.nabble.com>
	<CABK0fC6w9z2rb-yLQQpCbwLT1xfidQVj_NBFXpAgkB9Pz-Bn1w@mail.gmail.com>
	<CAKzWOxUgOM=TGkQqsAtUzNn1u0CZutJkT3yC99U9XZ7DZfKM5Q@mail.gmail.com>
	<CALNZuwm-2n7dCVkzjmGMDJYTZ10ZNOsrVLZA3u8CQ=M=gnBgwQ@mail.gmail.com>
Message-ID: <CAKbxfG_o9-=NUX86QbMMua7=NkeCM0YqA1PQeBtO_g7zM5mzNg@mail.gmail.com>

set the variable presence_data with any custom data you want and it
will show up in show channels.


On Fri, Sep 9, 2011 at 5:00 AM, Avi Marcus <avi at avimarcus.net> wrote:
> Well, it would be nifty for the "show calls" to show the
> variable_accountcode and/or the variable_user_name from the channels. I'm
> not sure if that data is in the same place or it require additional queries
> to get, though..
> As of now, afaik, you have to query all of them and then get the info on
> each channel to find the auth user or account code.
>
> -Avi
>
> On Fri, Sep 9, 2011 at 2:26 AM, Michael Collins <msc at freeswitch.org> wrote:
>>
>>
>> On Mon, Sep 5, 2011 at 11:46 AM, afshin afzali <a.afzali2003 at gmail.com>
>> wrote:
>>>
>>> I hoped an API which accept user name as input param !
>>> Thanks
>>
>> What if the user has more than one call in progress?
>> -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
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Sep  9 20:32:39 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 9 Sep 2011 09:32:39 -0700
Subject: [Freeswitch-users] Finding User's Current Session
In-Reply-To: <CAKbxfG_o9-=NUX86QbMMua7=NkeCM0YqA1PQeBtO_g7zM5mzNg@mail.gmail.com>
References: <CABK0fC7GLpqneudT753L3m2W72jHTVW82t3BZNmCe843O7cZCA@mail.gmail.com>
	<1315240242664-6761305.post@n2.nabble.com>
	<CABK0fC6w9z2rb-yLQQpCbwLT1xfidQVj_NBFXpAgkB9Pz-Bn1w@mail.gmail.com>
	<CAKzWOxUgOM=TGkQqsAtUzNn1u0CZutJkT3yC99U9XZ7DZfKM5Q@mail.gmail.com>
	<CALNZuwm-2n7dCVkzjmGMDJYTZ10ZNOsrVLZA3u8CQ=M=gnBgwQ@mail.gmail.com>
	<CAKbxfG_o9-=NUX86QbMMua7=NkeCM0YqA1PQeBtO_g7zM5mzNg@mail.gmail.com>
Message-ID: <CAKzWOxXWR_LK6JxdweXMcOwy2GreEZzEfLObGkhyfz+cVz4VtQ@mail.gmail.com>

On Fri, Sep 9, 2011 at 8:48 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> set the variable presence_data with any custom data you want and it
> will show up in show channels.
>
> I like this trick. I will throw this up on the wiki. Unless there are any
objections I will put it on the show channels API docs section of
mod_commands page and then do a cross-link to the presence_data chan var
page.

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110909/0be420b4/attachment.html 

From chrisbware at interfree.it  Fri Sep  9 20:36:43 2011
From: chrisbware at interfree.it (chrisbware at interfree.it)
Date: 9 Sep 2011 16:36:43 -0000
Subject: [Freeswitch-users] Setting SIP From Header using Lua
Message-ID: <20110909163643.24339.qmail@community29.interfree.it>


Hi,

following Lua wiki example:

session1 = freeswitch.Session("sofia/internal/1001%192.168.1.1");
session2 = freeswitch.Session("sofia/internal/1002%192.168.1.1");
freeswitch.bridge(session1, session2);

I'm trying to bridge two numbers on PSTN. Everything works but my SIP Provider reject call since my SIP From (From: "" <sip:0000000000 at myserverip) is different from the digest username.

I don't have any gateway configured on FS, since auth parameters are passed on Session arguments (["sip_auth_username="..Myuser..",sip_auth_password="..Mypasswd]).

I've tried many arguments in order to chenge from user without luck.

How can I do it? 

Thanks in advance.


-------------------------------------------------------------------------------
Valore legale alle tue mail
InterfreePEC - la tua Posta Elettronica Certificata
http://pec.interfree.it
-------------------------------------------------------------------------------



From tonybecq at yahoo.fr  Fri Sep  9 18:08:12 2011
From: tonybecq at yahoo.fr (obbyone)
Date: Fri, 9 Sep 2011 07:08:12 -0700 (PDT)
Subject: [Freeswitch-users] Making luaSQL or Dbh working on Freeswitch
 installed on a Debian Linux distribution
Message-ID: <1315577292541-6776039.post@n2.nabble.com>

Hye,

I have a server that works on a debian squeeze 6.0.2 Linux distribution.
On that server, since march 2011, is installed Freeswitch 1.0.6

I've just installed Debian packages for "mysql" and created a database that
works fine.
I've also installed Debian packages for "unixodbc" to be able to work on
that database via ODBC in order to access that database from
"Freeswitch.Dbh"

The command : "isql dsn_name" does connect to my database.

I have tried to install luaSQL but I don't know how to edit the "config"
file in the source directory "/usr/src/luasql-2.1.1/" in order to compile
without errors.
I've found that LUA_LIBDIR is not in /usr/local/lib/lua/5.1 but in
/usr/lib/lua/5.1
Same thing for LUA_DIR which is in /usr/share/lua/5.1
(can you confirm ?)

I don't know how to edit the lines about the MySQL Drivers :

DRIVER_LIBS= -L/usr/local/mysql/lib -lmysqlclient -lz
DRIVER_INCS= -I/usr/local/mysql/include

Odbc finds his own drivers in the directory : /usr/lib/odbc

I've heard that it could be better to use Freeswitch.Dbh instead of luaSQL
so this is not crutial but could help me to understand how all those things
work together.

My crutial questions are :

How can I test the connection from Freeswitch.Dbh to the database ?
Now that unixodbc is installed after freeswitch, do I have to re-install
freeswitch from scratch (configure, ..., make, make install, ...) in order
to have it working together?
Is this re-installation operation safe as I have created some new things in
the dialplan and the users directory of freeswitch ?


Thanks a lot.

Obbyone

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Making-luaSQL-or-Dbh-working-on-Freeswitch-installed-on-a-Debian-Linux-distribution-tp6776039p6776039.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From chad at apartmentlines.com  Fri Sep  9 23:51:09 2011
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Fri, 9 Sep 2011 15:51:09 -0400
Subject: [Freeswitch-users] Making luaSQL or Dbh working on Freeswitch
	installed on a Debian Linux distribution
In-Reply-To: <1315577292541-6776039.post@n2.nabble.com>
References: <1315577292541-6776039.post@n2.nabble.com>
Message-ID: <2391DA40-36DD-4AF2-8985-03579F7D6143@apartmentlines.com>

i wouldn't bother with LuaSQL, freeswitch.Dbh is a much better way to go.  you'll need to upgrade your FreeSWITCH installation though, 1.0.6 didn't have it.  you can find usage examples for freeswitch.Dbh on the wiki.

chad

On Sep 9, 2011, at 10:08 AM, obbyone wrote:

> Hye,
> 
> I have a server that works on a debian squeeze 6.0.2 Linux distribution.
> On that server, since march 2011, is installed Freeswitch 1.0.6
> 
> I've just installed Debian packages for "mysql" and created a database that
> works fine.
> I've also installed Debian packages for "unixodbc" to be able to work on
> that database via ODBC in order to access that database from
> "Freeswitch.Dbh"
> 
> The command : "isql dsn_name" does connect to my database.
> 
> I have tried to install luaSQL but I don't know how to edit the "config"
> file in the source directory "/usr/src/luasql-2.1.1/" in order to compile
> without errors.
> I've found that LUA_LIBDIR is not in /usr/local/lib/lua/5.1 but in
> /usr/lib/lua/5.1
> Same thing for LUA_DIR which is in /usr/share/lua/5.1
> (can you confirm ?)
> 
> I don't know how to edit the lines about the MySQL Drivers :
> 
> DRIVER_LIBS= -L/usr/local/mysql/lib -lmysqlclient -lz
> DRIVER_INCS= -I/usr/local/mysql/include
> 
> Odbc finds his own drivers in the directory : /usr/lib/odbc
> 
> I've heard that it could be better to use Freeswitch.Dbh instead of luaSQL
> so this is not crutial but could help me to understand how all those things
> work together.
> 
> My crutial questions are :
> 
> How can I test the connection from Freeswitch.Dbh to the database ?
> Now that unixodbc is installed after freeswitch, do I have to re-install
> freeswitch from scratch (configure, ..., make, make install, ...) in order
> to have it working together?
> Is this re-installation operation safe as I have created some new things in
> the dialplan and the users directory of freeswitch ?
> 
> 
> Thanks a lot.
> 
> Obbyone
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Making-luaSQL-or-Dbh-working-on-Freeswitch-installed-on-a-Debian-Linux-distribution-tp6776039p6776039.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 sunwood360 at gmail.com  Sat Sep 10 01:19:27 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Fri, 9 Sep 2011 14:19:27 -0700
Subject: [Freeswitch-users] FreeSWITCH with external registrar
In-Reply-To: <CAKq5mS7qfy_u2Q6cG6aF1rZPctvXpHBkyzCUHk8feP=NMSv93g@mail.gmail.com>
References: <CAKq5mS7qfy_u2Q6cG6aF1rZPctvXpHBkyzCUHk8feP=NMSv93g@mail.gmail.com>
Message-ID: <CAKq5mS7vSz4B2kxzgUn8vOJ9zPWp2Kwfw=HZEzW_Avd9UR2jHw@mail.gmail.com>

Hi:

Is there any config examples of freeswitch working with external registar?

Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110909/7ef170ed/attachment-0001.html 

From wagnerspi at gmail.com  Sat Sep 10 07:42:29 2011
From: wagnerspi at gmail.com (Wagner)
Date: Sat, 10 Sep 2011 00:42:29 -0300
Subject: [Freeswitch-users] [FreeSwitch-users] Calls being dropped
In-Reply-To: <CAJUJwthkA6QZqq3jF=mYG7cd0wC5ATPAdEs5Eb2R-uokutC0ww@mail.gmail.com>
References: <CAO0ino=Zw2de9-Gx=xg3v=fKL4_CZ3+8LKicuXyUZPSoraXSgw@mail.gmail.com>
	<CAJUJwthkA6QZqq3jF=mYG7cd0wC5ATPAdEs5Eb2R-uokutC0ww@mail.gmail.com>
Message-ID: <CAO0inokMxF1huVRZMAKG=K1L_c+7x_PtAt=8Rm06PuE-gQqfSQ@mail.gmail.com>

The problem is that I'm using the default configuration, as it came after
installed

I've tried to change the codec to use speex, but no luck

I try to call the voicemail, and before it starts playing the message or
sometimes in the miidle of the message it drops but I always can't hear the
full message, when I'm in the same network as the server, it gets a little
better and takes longer to drop but even so, it's before the end of message

Thanks

2011/9/9 Sam Govind <govoiper at gmail.com>

> Check codecs of both legs.Also I suspect the dialplan hitting a dead-end or
> something.
>
> On Fri, Sep 9, 2011 at 8:22 AM, Wagner <wagnerspi at gmail.com> wrote:
>
>> Hello Guys,
>>
>> I'm having some problems with calls, they are being dropped often, this is
>> the output in fs_cli
>>
>> 2011-09-09 07:13:05.644007 [DEBUG] switch_ivr_play_say.c:1152 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>> 2011-09-09 07:13:06.182083 [DEBUG] switch_ivr_play_say.c:1444 done playing
>> file
>> 2011-09-09 07:13:06.184311 [DEBUG] switch_ivr_play_say.c:1152 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>> 2011-09-09 07:13:06.762395 [DEBUG] switch_ivr_play_say.c:1444 done playing
>> file
>> 2011-09-09 07:13:06.884005 [WARNING] switch_core_file.c:176 Sample rate
>> doesn't match
>> 2011-09-09 07:13:06.884005 [DEBUG] switch_ivr_play_say.c:1152 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>> 2011-09-09 07:13:12.665209 [DEBUG] sofia.c:4153 Channel
>> sofia/internal/XXX at x.x.x.x entering state [terminating][0]
>> 2011-09-09 07:13:12.665209 [NOTICE] sofia.c:4789 Hangup
>> sofia/internal/XXX at x.x.x.x [CS_EXECUTE] [NORMAL_UNSPECIFIED]
>> 2011-09-09 07:13:12.665209 [DEBUG] switch_channel.c:2102 Send signal
>> sofia/internal/XXX at x.x.x.x [KILL]
>> 2011-09-09 07:13:12.665209 [DEBUG] switch_core_session.c:1021 Send signal
>> sofia/internal/XXX at x.x.x.x [BREAK]
>> 2011-09-09 07:13:12.682339 [DEBUG] switch_ivr_play_say.c:1444 done playing
>> file
>> 2011-09-09 07:13:12.687516 [DEBUG] switch_core_state_machine.c:348
>> (sofia/internal/XXX at x.x.x.x) State EXECUTE going to sleep
>> 2011-09-09 07:13:12.687516 [DEBUG] switch_core_state_machine.c:314
>> (sofia/internal/XXX at x.x.x.x) Running State Change CS_HANGUP
>> 2011-09-09 07:13:12.688525 [DEBUG] switch_core_media_bug.c:413 Removing
>> BUG from sofia/internal/XXX at x.x.x.x
>> 2011-09-09 07:13:12.689932 [DEBUG] switch_core_state_machine.c:499
>> (sofia/internal/XXX at x.x.x.x) State HANGUP
>> 2011-09-09 07:13:12.689932 [DEBUG] mod_sofia.c:414 Channel
>> sofia/internal/XXX at x.x.x.x hanging up, cause: NORMAL_UNSPECIFIED
>> 2011-09-09 07:13:12.697113 [DEBUG] switch_core_state_machine.c:46
>> sofia/internal/XXX at x.x.x.x Standard HANGUP, cause: NORMAL_UNSPECIFIED
>> 2011-09-09 07:13:12.698117 [DEBUG] switch_core_state_machine.c:499
>> (sofia/internal/XXX at x.x.x.x) State HANGUP going to sleep
>> 2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:333
>> (sofia/internal/XXX at x.x.x.x) State Change CS_HANGUP -> CS_REPORTING
>> 2011-09-09 07:13:12.699153 [DEBUG] switch_core_session.c:1021 Send signal
>> sofia/internal/XXX at x.x.x.x [BREAK]
>> 2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:314
>> (sofia/internal/XXX at x.x.x.x) Running State Change CS_REPORTING
>> 2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:590
>> (sofia/internal/XXX at x.x.x.x) State REPORTING
>> 2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:53
>> sofia/internal/XXX at x.x.x.x Standard REPORTING, cause: NORMAL_UNSPECIFIED
>> 2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:590
>> (sofia/internal/XXX at x.x.x.x) State REPORTING going to sleep
>> 2011-09-09 07:13:12.699153 [DEBUG] switch_core_state_machine.c:327
>> (sofia/internal/XXX at x.x.x.x) State Change CS_REPORTING -> CS_DESTROY
>> 2011-09-09 07:13:12.700174 [DEBUG] switch_core_session.c:1021 Send signal
>> sofia/internal/XXX at x.x.x.x [BREAK]
>> 2011-09-09 07:13:12.700174 [DEBUG] switch_core_session.c:1164 Session 40
>> (sofia/internal/XXX at x.x.x.x) Locked, Waiting on external entities
>> 2011-09-09 07:13:12.700174 [NOTICE] switch_core_session.c:1182 Session 40
>> (sofia/internal/XXX at x.x.x.x) Ended
>> 2011-09-09 07:13:12.700174 [NOTICE] switch_core_session.c:1184 Close
>> Channel sofia/internal/XXX at x.x.x.x [CS_DESTROY]
>> 2011-09-09 07:13:12.701183 [DEBUG] switch_core_state_machine.c:428
>> (sofia/internal/XXX at x.x.x.x) Running State Change CS_DESTROY
>> 2011-09-09 07:13:12.701183 [DEBUG] switch_core_state_machine.c:439
>> (sofia/internal/XXX at x.x.x.x) State DESTROY
>> 2011-09-09 07:13:12.701183 [DEBUG] mod_sofia.c:341
>> sofia/internal/XXX at x.x.x.x SOFIA DESTROY
>> 2011-09-09 07:13:12.701183 [DEBUG] switch_core_state_machine.c:60
>> sofia/internal/XXX at x.x.x.x Standard DESTROY
>> 2011-09-09 07:13:12.701183 [DEBUG] switch_core_state_machine.c:439
>> (sofia/internal/XXX at x.x.x.x) State DESTROY going to sleep
>>
>>
>> what could it be?
>>
>> 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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110910/3b295aea/attachment.html 

From xing2kin at yahoo.com  Sat Sep 10 09:02:37 2011
From: xing2kin at yahoo.com (king2kin)
Date: Fri, 9 Sep 2011 22:02:37 -0700 (PDT)
Subject: [Freeswitch-users] BUG in FS C-Function: switch_separate_string(-)
Message-ID: <1315630957.3007.YahooMailClassic@web39704.mail.mud.yahoo.com>

Hi folks,

I think that switch_separate_string(-) has a bug in splitting a string into array by a token: 

While parsing a char string into an array, [switch_separate_string(-)] always treats some substrings (which start with backslash char '\') as escaped chars, 

for example: even though char '\' has been escaped inside a constant string {"status=10&path=c:\\temp\\log\\&name=tom"}, [switch_separate_string(-)] treats its substring "\t" AS a single tab char; and substring "\&" AS a single char '&'. This result is definitely wrong!

Here is an example that I tried on windows 2003 server:

-- src codes:

int nParams = 0;
char* params[64] = {0};

char mystr[] = "status=10&path=c:\\temp\\log\\&name=tom";

switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "mystr: %s\n", mystr);
nParams = switch_separate_string(mystr, '&', params, (sizeof(params) / sizeof(params[0])));??? 
for (i=0; i < nParams; i++)
{
? ? switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "[%d]: {%s}\n", i, params[i]);
}

-- printed log info:

2011-09-10 12:26:42.566383 [DEBUG] mod_xxx.c:981 mystr: status=10&path=c:\temp\log\&name=tom
2011-09-10 12:26:42.566383 [DEBUG] mod_xxx.c:985 [0]: {status=10}
2011-09-10 12:26:42.566383 [DEBUG] mod_xxx.c:985 [1]: {path=c:??? emp\log&name=tom}

My comment: the above string 'mystr' is supposed to be parsed into a 3-element array {10, path=c:\temp\log\, tom}; however, [switch_separate_string(-)]parsed it into a 2-element array {10, path=c:    emp\log&name=tom}. 


x.k. 




From peter.olsson at visionutveckling.se  Sat Sep 10 14:46:21 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 10 Sep 2011 12:46:21 +0200
Subject: [Freeswitch-users] BUG in FS C-Function:
 switch_separate_string(-)
In-Reply-To: <1315630957.3007.YahooMailClassic@web39704.mail.mud.yahoo.com>
References: <1315630957.3007.YahooMailClassic@web39704.mail.mud.yahoo.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FC@cooper>

Yes, this seems wrong, please add this issue to Jira.

To get around (until it's fixed) it you can use / for the path, for instance C:/test/test.wav - Windows will handle frontslash as well as backslash when entering a path.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r king2kin [xing2kin at yahoo.com]
Skickat: den 10 september 2011 07:02
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] BUG in FS C-Function: switch_separate_string(-)

Hi folks,

I think that switch_separate_string(-) has a bug in splitting a string into array by a token:

While parsing a char string into an array, [switch_separate_string(-)] always treats some substrings (which start with backslash char '\') as escaped chars,

for example: even though char '\' has been escaped inside a constant string {"status=10&path=c:\\temp\\log\\&name=tom"}, [switch_separate_string(-)] treats its substring "\t" AS a single tab char; and substring "\&" AS a single char '&'. This result is definitely wrong!

Here is an example that I tried on windows 2003 server:

-- src codes:

int nParams = 0;
char* params[64] = {0};

char mystr[] = "status=10&path=c:\\temp\\log\\&name=tom";

switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "mystr: %s\n", mystr);
nParams = switch_separate_string(mystr, '&', params, (sizeof(params) / sizeof(params[0])));
for (i=0; i < nParams; i++)
{
    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "[%d]: {%s}\n", i, params[i]);
}

-- printed log info:

2011-09-10 12:26:42.566383 [DEBUG] mod_xxx.c:981 mystr: status=10&path=c:\temp\log\&name=tom
2011-09-10 12:26:42.566383 [DEBUG] mod_xxx.c:985 [0]: {status=10}
2011-09-10 12:26:42.566383 [DEBUG] mod_xxx.c:985 [1]: {path=c:    emp\log&name=tom}

My comment: the above string 'mystr' is supposed to be parsed into a 3-element array {10, path=c:\temp\log\, tom}; however, [switch_separate_string(-)]parsed it into a 2-element array {10, path=c:    emp\log&name=tom}.


x.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

!DSPAM:4e6af00032762055611955!



From avi at avimarcus.net  Sat Sep 10 21:13:33 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 10 Sep 2011 20:13:33 +0300
Subject: [Freeswitch-users] Multiple DTMF on Single Keypress
In-Reply-To: <CADEQS5yAHOFO=2oP5poWxZBNQE-gi3USJon8Oq+GN-LWD4_CUQ@mail.gmail.com>
References: <CADEQS5zp8bh3epB3-ya9uR2HvNjnt8qf7u4rS=a11tNNGGo7hA@mail.gmail.com>
	<CA+Gmxbbm46vCtW70+Qt7LurJEybMHwDA6BA+tNa8OnL4U0uKfw@mail.gmail.com>
	<CAFKNP3-Z6Zhas8DZobHqw4fxampyYjwqMU=guzG1jvZvHuWWFA@mail.gmail.com>
	<CALNZuwk372OUr1cNdqGrPqmBdFPEf6EUSBuPzhqBmBu1Bczvxg@mail.gmail.com>
	<CADEQS5xsRouLa1fxGu9+=sUJdagO4cW6q0AfsoPKwDghKMvSCw@mail.gmail.com>
	<CALNZuw=q2Up_2EpAkQ-5Pvfu_zZGWUsHUeK=iimhK+ueyyfSjg@mail.gmail.com>
	<CADEQS5zqoKocYUC=DS_wN_PAEuF4kJofi9+RmSQv5mzd_U+s9A@mail.gmail.com>
	<CALNZuwm3GVLqxTLSquzkrsvZcQoq=gB4btvCBT1uhBH7K75ryQ@mail.gmail.com>
	<CADEQS5yAHOFO=2oP5poWxZBNQE-gi3USJon8Oq+GN-LWD4_CUQ@mail.gmail.com>
Message-ID: <CALNZuwm+vi61Q=w-qjS73mCFcrk4ra8dr+_80+w39_Yrpw-F2w@mail.gmail.com>

Is it supposed to have 6 "end" packets instead of 3? Maybe this needs to be
jira'd as a compatibility thing.
-Avi


On Thu, Sep 8, 2011 at 12:51 AM, Brian Wiese <
brian.wiese.freeswitch at gmail.com> wrote:

> Avi:
>
> Here's a new full tcpdump and log:
> http://www.netwayaccess.com/newcapture.zip
>
> Like you said, I filtered it by rtpevent and I only see one DTMF 4, but FS
> read two.
>
> ~Brian
>
> On Wed, Sep 7, 2011 at 4:37 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> That pcap only shows 9[pause]943 so it's not the whole thing? But anyway,
>> it's coming in as rfc2833, so it's really unlikely FS is mis-reading it..
>> Can you get a full pcap? You can open them in wireshark and filter for
>> "rtpevent" to see the dtmf digits that come in.
>>
>> -Avi
>>
>>
>> On Thu, Sep 8, 2011 at 12:19 AM, Brian Wiese <
>> brian.wiese.freeswitch at gmail.com> wrote:
>>
>>> Avi:
>>>
>>> Thank you for your help on this.
>>>
>>> I've captured the traffic as you've requested and another log.  I made it
>>> to the directory (DTMF 9 in the IVR), but then when I tried to dial 94373
>>> you can see I had some duplicate DTMF.
>>> http://www.netwayaccess.com/pcapsipdump.zip
>>> ~Brian
>>>   On Wed, Sep 7, 2011 at 6:18 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>>> Can you get a normal PCAP of the SIP/RTP with something here?
>>>> http://wiki.freeswitch.org/wiki/Packet_Capture
>>>>
>>>> e.g. pcapsipdump is quite nice. (Just make sure the folder exists
>>>> before running the command.)
>>>>
>>>> -Avi
>>>>
>>>>
>>>> On Wed, Sep 7, 2011 at 2:13 AM, Brian Wiese
>>>> <brian.wiese.freeswitch at gmail.com> wrote:
>>>> > Avi:
>>>> >
>>>> > I had thought it was inband, but I couldn't find anything that
>>>> supported it,
>>>> > like you mentioned.
>>>> >
>>>> > Is there anything else I can provide that would help solve this
>>>> problem?
>>>> >
>>>> > Thanks.
>>>> >
>>>> > ~Brian
>>>> >
>>>> > On Tue, Sep 6, 2011 at 3:48 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>> >>
>>>> >> line 389
>>>> >> 2011-09-04 16:20:14.390350 [DEBUG] switch_rtp.c:3317 RTP RECV
>>>> DTMF 5:1440
>>>> >> 2011-09-04 16:20:14.390350 [DEBUG] switch_ivr_bridge.c:391 Send
>>>> signal
>>>> >> sofia/internal/sip:20005 at 172.31.6.253 [BREAK]
>>>> >> 2011-09-04 16:20:14.410350 [DEBUG] switch_rtp.c:2343 Send start
>>>> packet
>>>> >> for [5] ts=97600 dur=160/160/1440 seq=29252
>>>> >> 2011-09-04 16:20:14.410350 [DEBUG] switch_rtp.c:3317 RTP RECV
>>>> DTMF 5:1440
>>>> >>
>>>> >> It's coming as RTP RECV, meaning rfc2833. Otherwise, it would say
>>>> >> "DETECTED".
>>>> >>
>>>> >> -Avi
>>>> >>
>>>> >>
>>>> >> On Tue, Sep 6, 2011 at 11:36 PM, Jon Young <jonyoung111 at gmail.com>
>>>> wrote:
>>>> >> >
>>>> >> > Is it possible you are receiving 2833 and Inband DTMF?
>>>> >> >
>>>> >> > On Mon, Sep 5, 2011 at 10:36 PM, Dmitry Sytchev <kbdfck at gmail.com>
>>>> >> > wrote:
>>>> >> > > See the same behaviour with inband DTMF detector sometimes.
>>>> >> > >
>>>> >> > > 2011/9/5 Brian Wiese FreeSWITCH List
>>>> >> > > <brian.wiese.freeswitch at gmail.com>
>>>> >> > >>
>>>> >> > >> Hello everyone!
>>>> >> > >>
>>>> >> > >> I'm getting multiple RTP DTMF from random keypresses and I can't
>>>> >> > >> figure
>>>> >> > >> out why.  I've PB'ed the packet capture and FS log for a call.
>>>> As
>>>> >> > >> you can
>>>> >> > >> see from the FS log, the 9,8,7,6 numbers weren't repeated, but 5
>>>> was
>>>> >> > >> (again,
>>>> >> > >> though, different calls lead to different numbers being
>>>> repeated).
>>>> >> > >> Log:  http://pastebin.freeswitch.org/17280
>>>> >> > >> Capture:  http://pastebin.freeswitch.org/17282
>>>> >> > >>
>>>> >> > >> I appreciate any ideas as to what I might have wrong here.
>>>> >> > >>
>>>> >> > >> 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
>>>> >> > >>
>>>> >> > >
>>>> >> > >
>>>> >> > >
>>>> >> > > --
>>>> >> > > 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
>>>> >>
>>>> >>
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110910/d1a6ad8d/attachment-0001.html 

From tonybecq at yahoo.fr  Sat Sep 10 12:18:56 2011
From: tonybecq at yahoo.fr (obbyone)
Date: Sat, 10 Sep 2011 01:18:56 -0700 (PDT)
Subject: [Freeswitch-users] Making luaSQL or Dbh working on Freeswitch
 installed on a Debian Linux distribution
In-Reply-To: <2391DA40-36DD-4AF2-8985-03579F7D6143@apartmentlines.com>
References: <1315577292541-6776039.post@n2.nabble.com>
	<2391DA40-36DD-4AF2-8985-03579F7D6143@apartmentlines.com>
Message-ID: <1315642736663-6778373.post@n2.nabble.com>

Thanks. 
I've made an upgrade of freeswitch with GIT version of 2011-09-09 and it
works fine.

Can I now use Freeswitch.Dbh on the mysql database through ODBC without
changing anything or do I have to edit and compile something else?

How can I test in a simple way this connection between Freeswitch and MySQL
database ?

Thanks in advance.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Making-luaSQL-or-Dbh-working-on-Freeswitch-installed-on-a-Debian-Linux-distribution-tp6776039p6778373.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From avi at avimarcus.net  Sat Sep 10 22:38:15 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 10 Sep 2011 21:38:15 +0300
Subject: [Freeswitch-users] What Provider do you guys use?
In-Reply-To: <CADrcvV86dR63-G6_-_KkpLSVh7UVEosT9gwNOJR-vK-Y37ibNA@mail.gmail.com>
References: <CADrcvV86dR63-G6_-_KkpLSVh7UVEosT9gwNOJR-vK-Y37ibNA@mail.gmail.com>
Message-ID: <CALNZuwm-RsUL56Y=gqmr3=aMysrq0RsyshUjm9yEGznvNukNXg@mail.gmail.com>

Many folks use flowroute for domestic and also voip.ms - both are flatrate
for USA.
Siproutes pools lots of carriers together, and you pay as npa-nxx with lrn
rates.

High volume opens up your options to other places...

International is a whole 'nother story. If you have *high* volume, places
like xconnect, ovetel, and many other big terminators can help you out.
Otherwise, most USA domestic terminators have pretty horrid a-z rates.

You can message me offlist and we can see if I can help you.

-Avi


On Thu, Sep 8, 2011 at 7:18 PM, Cesar Bermudez <cesar.bermudez at gmail.com>wrote:

> Hi Fs'users
>
> Sorry for this question, but what good providers you recommend?
> I need good quality to this destinations:
> USA
> Nicaragua
> Vietnan
> China
> Indonesia
>
> I want good routes, with cli if possible, and good prices :D
>
> Sorry for this mail again, i dont want to make any flame war or spam ...
> only want advice from more experience voip admins.
>
> Best 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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110910/0208422d/attachment.html 

From nyilmaz at cybetech.com  Sat Sep 10 22:26:40 2011
From: nyilmaz at cybetech.com (Nihat Yilmaz)
Date: Sat, 10 Sep 2011 21:26:40 +0300
Subject: [Freeswitch-users] PRI Signalling Status down!
Message-ID: <201109102126.40444.nyilmaz@cybetech.com>

Dear All,

I am a starter on Freeswitch although I have telecoms experience for a long 
time

I am trying to set-up a Freeswitch server with a Digium TE110P ISDN PrI E1 
card.

I have installed DAHDI Drivres and the card is discovered OK.

I have also installed the libpri library and activated mod_freetdm. When I run 
the freeswitch I can see that my span is defined when I connect disconnect the 
cable I recieve alarms as below

2011-09-10 16:47:27.948506 NOTICE mod_freetdm.c:1819 Alarm raised on channel 
1:1
2011-09-10 16:47:27.948506 WARNING ftmod_libpri.c:1419 channel 1:1 (1:1) has 
alarms! BLUE
2011-09-10 16:47:27.948506 NOTICE mod_freetdm.c:1819 Alarm raised on channel 
1:2
2011-09-10 16:47:27.948506 WARNING ftmod_libpri.c:1419 channel 1:2 (1:2) has 
alarms! BLUE
.
.
.
2011-09-10 16:47:27.948506 NOTICE mod_freetdm.c:1819 Alarm raised on channel 
1:30
2011-09-10 16:47:27.948506 WARNING ftmod_libpri.c:1419 channel 1:30 (1:30) has 
alarms! BLUE
2011-09-10 16:47:27.948506 NOTICE mod_freetdm.c:1819 Alarm raised on channel 
1:31
2011-09-10 16:47:27.948506 WARNING ftmod_libpri.c:1419 channel 1:31 (1:31) has 
alarms! BLUE


The problem I have is no matter what I do signalling_status is always DOWN.


freeswitch at voip.cybetech.com> ftdm list

+OKspan: 1 (GSMChannelBank)
type: isdn
physical_status: ok
signaling_status: DOWN
chan_count: 31
dialplan: XML
context: public
dial_regex:
fail_dial_regex:
hold_music:
analog_options: none


I have changed various settings in the configuration files but there is no 
change. my conf files are as follows:

/etc/dahdi/system.conf:

span=1,0,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
loadzone = us
defaultzone = us

freetdm.conf:


general
cpu_monitor => no
cpu_monitoring_interval => 1000
cpu_set_alarm_threshold => 80
cpu_reset_alarm_threshold => 70
cpu_alarm_action => warn

debugdtmf_directory=/usr/local/freeswitch/log/

span zt GSMChannelBank
trunk_type => E1
b-channel => 1-15
d-channel => 16
b-channel => 17-31

freetdm.conf.xml:


<configuration name="freetdm.conf" description="FreeTDM Configuration">

<settings>
<param name="debug" value="0"/>
</settings>


<libpri_spans>
<span name="GSMChannelBank">
<param name="switch" value="euroisdn"/>
<param name="mode" value="net"/>
<param name="debug" value="none"/>

<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
</span>
</libpri_spans>

</configuration>


I will appreciate if anyone can advise where I shall look for the solution.

Regards,

Nihat Yilmaz
-- 
Cybetech Ltd.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110910/6f906b1d/attachment.html 

From chad at apartmentlines.com  Sun Sep 11 00:48:34 2011
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Sat, 10 Sep 2011 16:48:34 -0400
Subject: [Freeswitch-users] Making luaSQL or Dbh working on Freeswitch
	installed on a Debian Linux distribution
In-Reply-To: <1315642736663-6778373.post@n2.nabble.com>
References: <1315577292541-6776039.post@n2.nabble.com>
	<2391DA40-36DD-4AF2-8985-03579F7D6143@apartmentlines.com>
	<1315642736663-6778373.post@n2.nabble.com>
Message-ID: <A0DE5454-C659-4149-8BF6-EAF4B0B4D908@apartmentlines.com>

RTFM:

http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.Dbh
http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh

chad

On Sep 10, 2011, at 4:18 AM, obbyone wrote:

> Thanks. 
> I've made an upgrade of freeswitch with GIT version of 2011-09-09 and it
> works fine.
> 
> Can I now use Freeswitch.Dbh on the mysql database through ODBC without
> changing anything or do I have to edit and compile something else?
> 
> How can I test in a simple way this connection between Freeswitch and MySQL
> database ?
> 
> Thanks in advance.
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Making-luaSQL-or-Dbh-working-on-Freeswitch-installed-on-a-Debian-Linux-distribution-tp6776039p6778373.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 contact at aharm.de  Sun Sep 11 03:40:04 2011
From: contact at aharm.de (Alexander Harm)
Date: Sun, 11 Sep 2011 01:40:04 +0200
Subject: [Freeswitch-users] Help needed on basic dialing plan: no calls are
	routed
Message-ID: <DB197D09-AC19-49F5-86AF-7951F94F690B@aharm.de>

hello all,

i have a freeswitch installation running and 2 phones connected to it. i setup 1 inbound rule. when i dial the external number i receive the following messages in the log. to me it seems that the routing works correctly but the  transfer to extension 77 fails.

note: further down the error messages i receive when trying to call internally.

any help appreciated.

regards, alexander

LOG EXTERNAL -> INTERNAL

2011-09-11 01:17:17.368068 [NOTICE] switch_channel.c:908 New Channel sofia/external/0476058096 at voip.belgacom.be [068a829e-dc03-11e0-81f7-f9f410ff919e]
2011-09-11 01:17:17.368068 [DEBUG] sofia.c:5156 Channel sofia/external/0476058096 at voip.belgacom.be entering state [received][100]
2011-09-11 01:17:17.368068 [DEBUG] sofia.c:5167 Remote SDP:
v=0
o=- 0 98592702 IN IP4 81.240.251.38
s=IMSS
c=IN IP4 81.240.251.38
t=0 0
m=audio 10874 RTP/AVP 8 18 0 101
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

2011-09-11 01:17:17.368068 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
2011-09-11 01:17:17.368068 [DEBUG] sofia_glue.c:2857 Set Codec sofia/external/0476058096 at voip.belgacom.be PCMA/8000 20 ms 160 samples 64000 bits
2011-09-11 01:17:17.368068 [DEBUG] sofia_glue.c:4859 Set 2833 dtmf send/recv payload to 101
2011-09-11 01:17:17.368068 [DEBUG] sofia.c:5357 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_NEW -> CS_INIT
2011-09-11 01:17:17.368068 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 01:17:17.368068 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_INIT
2011-09-11 01:17:17.368068 [DEBUG] switch_core_state_machine.c:375 (sofia/external/0476058096 at voip.belgacom.be) State INIT
2011-09-11 01:17:17.368068 [DEBUG] mod_sofia.c:85 sofia/external/0476058096 at voip.belgacom.be SOFIA INIT
2011-09-11 01:17:17.368068 [DEBUG] mod_sofia.c:125 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_INIT -> CS_ROUTING
2011-09-11 01:17:17.368068 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 01:17:17.368068 [DEBUG] switch_core_state_machine.c:375 (sofia/external/0476058096 at voip.belgacom.be) State INIT going to sleep
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_ROUTING
2011-09-11 01:17:17.388551 [DEBUG] switch_channel.c:1837 (sofia/external/0476058096 at voip.belgacom.be) Callstate Change DOWN -> RINGING
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING
2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:148 sofia/external/0476058096 at voip.belgacom.be SOFIA ROUTING
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:78 sofia/external/0476058096 at voip.belgacom.be Standard ROUTING
2011-09-11 01:17:17.388551 [INFO] mod_dialplan_xml.c:336 Processing 0476058096 <0476058096>->015261786 in context public
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->unloop] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->outside_call] continue=true
Dialplan: sofia/external/0476058096 at voip.belgacom.be Absolute Condition [outside_call]
Dialplan: sofia/external/0476058096 at voip.belgacom.be Action set(outside_call=true) 
Dialplan: sofia/external/0476058096 at voip.belgacom.be Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->call_debug] continue=true
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->public_extensions] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [public_extensions] destination_number(015261786) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->015261786] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [015261786] context(public) =~ /public/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [015261786] destination_number(015261786) =~ /^015261786$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Action transfer(77 XML default) 
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:128 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_ROUTING -> CS_EXECUTE
2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING going to sleep
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_EXECUTE
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE
2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:241 sofia/external/0476058096 at voip.belgacom.be SOFIA EXECUTE
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:166 sofia/external/0476058096 at voip.belgacom.be Standard EXECUTE
EXECUTE sofia/external/0476058096 at voip.belgacom.be set(outside_call=true)
2011-09-11 01:17:17.388551 [DEBUG] mod_dptools.c:1167 sofia/external/0476058096 at voip.belgacom.be SET [outside_call]=[true]
EXECUTE sofia/external/0476058096 at voip.belgacom.be set(RFC2822_DATE=Sun, 11 Sep 2011 01:17:17 +0200)
2011-09-11 01:17:17.388551 [DEBUG] mod_dptools.c:1167 sofia/external/0476058096 at voip.belgacom.be SET [RFC2822_DATE]=[Sun, 11 Sep 2011 01:17:17 +0200]
EXECUTE sofia/external/0476058096 at voip.belgacom.be transfer(77 XML default)
2011-09-11 01:17:17.388551 [DEBUG] switch_ivr.c:1693 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_EXECUTE -> CS_ROUTING
2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:724 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 01:17:17.388551 [NOTICE] switch_ivr.c:1699 Transfer sofia/external/0476058096 at voip.belgacom.be to XML[77 at default]
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE going to sleep
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_ROUTING
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING
2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:148 sofia/external/0476058096 at voip.belgacom.be SOFIA ROUTING
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:78 sofia/external/0476058096 at voip.belgacom.be Standard ROUTING
2011-09-11 01:17:17.388551 [INFO] mod_dialplan_xml.c:336 Processing 0476058096 <0476058096>->77 in context default
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->unloop] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->tod_example] continue=true
Dialplan: sofia/external/0476058096 at voip.belgacom.be Date/Time Match (FAIL) [tod_example] break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->holiday_example] continue=true
Dialplan: sofia/external/0476058096 at voip.belgacom.be Date/Time Match (FAIL) [holiday_example] break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->global-intercept] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [global-intercept] destination_number(77) =~ /^886$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->group-intercept] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [group-intercept] destination_number(77) =~ /^\*8$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->intercept-ext] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [intercept-ext] destination_number(77) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->redial] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [redial] destination_number(77) =~ /^(redial|870)$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->global] continue=true
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: sofia/external/0476058096 at voip.belgacom.be Absolute Condition [global]
Dialplan: sofia/external/0476058096 at voip.belgacom.be Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) 
Dialplan: sofia/external/0476058096 at voip.belgacom.be Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) 
Dialplan: sofia/external/0476058096 at voip.belgacom.be Action hash(insert/${domain_name}-last_dial/global/${uuid}) 
Dialplan: sofia/external/0476058096 at voip.belgacom.be Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->snom-demo-2] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [snom-demo-2] destination_number(77) =~ /^9001$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->snom-demo-1] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [snom-demo-1] destination_number(77) =~ /^9000$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->eavesdrop] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [eavesdrop] destination_number(77) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->eavesdrop] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [eavesdrop] destination_number(77) =~ /^779$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->call_return] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call_return] destination_number(77) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->del-group] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [del-group] destination_number(77) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->add-group] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [add-group] destination_number(77) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->call-group-simo] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call-group-simo] destination_number(77) =~ /^82(\d{2})$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->call-group-order] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call-group-order] destination_number(77) =~ /^83(\d{2})$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->extension-intercom] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [extension-intercom] destination_number(77) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->Local_Extension] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [Local_Extension] destination_number(77) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->Local_Extension_Skinny] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [Local_Extension_Skinny] destination_number(77) =~ /^(11[01][0-9])$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->group_dial_sales] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [group_dial_sales] destination_number(77) =~ /^2000$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->group_dial_support] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [group_dial_support] destination_number(77) =~ /^2001$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->group_dial_billing] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [group_dial_billing] destination_number(77) =~ /^2002$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->operator] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [operator] destination_number(77) =~ /^(operator|0)$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->vmain] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [vmain] destination_number(77) =~ /^vmain$|^4000$|^\*98$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->sip_uri] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [sip_uri] destination_number(77) =~ /^sip:(.*)$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->nb_conferences] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [nb_conferences] destination_number(77) =~ /^(30\d{2})$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->wb_conferences] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [wb_conferences] destination_number(77) =~ /^(31\d{2})$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->uwb_conferences] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [uwb_conferences] destination_number(77) =~ /^(32\d{2})$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->cdquality_conferences] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [cdquality_conferences] destination_number(77) =~ /^(33\d{2})$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->freeswitch_public_conf_via_sip] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(77) =~ /^9(888|8888|1616|3232)$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->mad_boss_intercom] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [mad_boss_intercom] destination_number(77) =~ /^0911$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->mad_boss_intercom] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [mad_boss_intercom] destination_number(77) =~ /^0912$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->mad_boss] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [mad_boss] destination_number(77) =~ /^0913$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ivr_demo] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ivr_demo] destination_number(77) =~ /^5000$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->dynamic_conference] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [dynamic_conference] destination_number(77) =~ /^5001$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->rtp_multicast_page] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [rtp_multicast_page] destination_number(77) =~ /^pagegroup$|^7243$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->park] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [park] destination_number(77) =~ /^5900$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->unpark] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unpark] destination_number(77) =~ /^5901$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->valet_park] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [valet_park] destination_number(77) =~ /^(6000)$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->valet_park] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [valet_park] destination_number(77) =~ /^(60\d[1-9])$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->park] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [park] destination_number(77) =~ /park\+(\d+)/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->unpark] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unpark] destination_number(77) =~ /^parking$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->park] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [park] destination_number(77) =~ /callpark/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->unpark] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unpark] destination_number(77) =~ /pickup/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->wait] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [wait] destination_number(77) =~ /^wait$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->fax_receive] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [fax_receive] destination_number(77) =~ /^9178$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->fax_transmit] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [fax_transmit] destination_number(77) =~ /^9179$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_180] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_180] destination_number(77) =~ /^9180$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_183_uk_ring] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_183_uk_ring] destination_number(77) =~ /^9181$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_183_music_ring] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_183_music_ring] destination_number(77) =~ /^9182$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_post_answer_uk_ring] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(77) =~ /^9183$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_post_answer_music] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_post_answer_music] destination_number(77) =~ /^9184$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ClueCon] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ClueCon] destination_number(77) =~ /^9191$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->show_info] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [show_info] destination_number(77) =~ /^9192$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->video_record] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [video_record] destination_number(77) =~ /^9193$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->video_playback] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [video_playback] destination_number(77) =~ /^9194$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->delay_echo] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [delay_echo] destination_number(77) =~ /^9195$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->echo] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [echo] destination_number(77) =~ /^9196$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->milliwatt] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [milliwatt] destination_number(77) =~ /^9197$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->tone_stream] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [tone_stream] destination_number(77) =~ /^9198$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->zrtp_enrollement] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [zrtp_enrollement] destination_number(77) =~ /^9787$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->hold_music] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [hold_music] destination_number(77) =~ /^9664$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->Recordings] continue=false
2011-09-11 01:17:17.388551 [ERR] switch_regex.c:101 COMPILE ERROR: 1 [nothing to repeat][^*(732)$]
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [Recordings] destination_number(77) =~ /^*(732)$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->enum] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [enum] ${module_exists(mod_enum)}(true) =~ /true/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [enum] destination_number(77) =~ /^(.*)$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Action transfer(77 enum) 
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:128 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_ROUTING -> CS_EXECUTE
2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING going to sleep
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_EXECUTE
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE
2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:241 sofia/external/0476058096 at voip.belgacom.be SOFIA EXECUTE
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:166 sofia/external/0476058096 at voip.belgacom.be Standard EXECUTE
EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.100-spymap/0476058096/068a829e-dc03-11e0-81f7-f9f410ff919e)
EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.100-last_dial/0476058096/77)
EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.100-last_dial/global/068a829e-dc03-11e0-81f7-f9f410ff919e)
EXECUTE sofia/external/0476058096 at voip.belgacom.be set(RFC2822_DATE=Sun, 11 Sep 2011 01:17:17 +0200)
2011-09-11 01:17:17.388551 [DEBUG] mod_dptools.c:1167 sofia/external/0476058096 at voip.belgacom.be SET [RFC2822_DATE]=[Sun, 11 Sep 2011 01:17:17 +0200]
EXECUTE sofia/external/0476058096 at voip.belgacom.be transfer(77 enum)
2011-09-11 01:17:17.388551 [DEBUG] switch_ivr.c:1693 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_EXECUTE -> CS_ROUTING
2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:724 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 01:17:17.388551 [NOTICE] switch_ivr.c:1699 Transfer sofia/external/0476058096 at voip.belgacom.be to enum[77 at default]
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE going to sleep
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_ROUTING
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING
2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:148 sofia/external/0476058096 at voip.belgacom.be SOFIA ROUTING
2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:78 sofia/external/0476058096 at voip.belgacom.be Standard ROUTING
2011-09-11 01:17:17.388551 [DEBUG] mod_enum.c:541 ENUM Lookup on 77
2011-09-11 01:17:27.188065 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 01:17:27.188065 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 01:17:27.188065 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]


LOG  INTERNAL -> INTERNAL


2011-09-11 01:22:17.268071 [DEBUG] sofia.c:7057 IP 192.168.100.6 Approved by acl "lan[]". Access Granted.
2011-09-11 01:22:17.268071 [NOTICE] switch_channel.c:908 New Channel sofia/internal/88 at 192.168.1.100 [b94b66fa-dc03-11e0-81f9-f9f410ff919e]
2011-09-11 01:22:17.268071 [DEBUG] sofia.c:5156 Channel sofia/internal/88 at 192.168.1.100 entering state [received][100]
2011-09-11 01:22:17.268071 [DEBUG] sofia.c:5167 Remote SDP:
v=0
o=- 1315696937270159 1 IN IP4 192.168.100.6
s=CounterPath X-Lite 4.1
c=IN IP4 192.168.100.6
t=0 0
a=ice-ufrag:de6e17
a=ice-pwd:9e8391bc4955d13bbf67e8eeb2118613
m=audio 63158 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=candidate:1 1 UDP 659136 192.168.100.6 63158 typ host
a=candidate:1 2 UDP 659134 192.168.100.6 63159 typ host

2011-09-11 01:22:17.268071 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_NEW
2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMA:8:8000:20:64000]
2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMU:0:8000:20:64000]/[G729:18:8000:20:8000]
2011-09-11 01:22:17.268071 [DEBUG] switch_core_state_machine.c:354 (sofia/internal/88 at 192.168.1.100) State NEW
2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:2857 Set Codec sofia/internal/88 at 192.168.1.100 PCMA/8000 20 ms 160 samples 64000 bits
2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:4859 Set 2833 dtmf send/recv payload to 101
2011-09-11 01:22:17.268071 [DEBUG] sofia.c:5357 (sofia/internal/88 at 192.168.1.100) State Change CS_NEW -> CS_INIT
2011-09-11 01:22:17.268071 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_INIT
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:375 (sofia/internal/88 at 192.168.1.100) State INIT
2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:85 sofia/internal/88 at 192.168.1.100 SOFIA INIT
2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:125 (sofia/internal/88 at 192.168.1.100) State Change CS_INIT -> CS_ROUTING
2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:375 (sofia/internal/88 at 192.168.1.100) State INIT going to sleep
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_ROUTING
2011-09-11 01:22:17.288082 [DEBUG] switch_channel.c:1837 (sofia/internal/88 at 192.168.1.100) Callstate Change DOWN -> RINGING
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:384 (sofia/internal/88 at 192.168.1.100) State ROUTING
2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:148 sofia/internal/88 at 192.168.1.100 SOFIA ROUTING
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:78 sofia/internal/88 at 192.168.1.100 Standard ROUTING
2011-09-11 01:22:17.288082 [INFO] mod_dialplan_xml.c:336 Processing 88 <88>->77 in context public
Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->unloop] continue=false
Dialplan: sofia/internal/88 at 192.168.1.100 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/88 at 192.168.1.100 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->outside_call] continue=true
Dialplan: sofia/internal/88 at 192.168.1.100 Absolute Condition [outside_call]
Dialplan: sofia/internal/88 at 192.168.1.100 Action set(outside_call=true) 
Dialplan: sofia/internal/88 at 192.168.1.100 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 
Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->call_debug] continue=true
Dialplan: sofia/internal/88 at 192.168.1.100 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->public_extensions] continue=false
Dialplan: sofia/internal/88 at 192.168.1.100 Regex (FAIL) [public_extensions] destination_number(77) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->015261786] continue=false
Dialplan: sofia/internal/88 at 192.168.1.100 Regex (PASS) [015261786] context(public) =~ /public/ break=on-false
Dialplan: sofia/internal/88 at 192.168.1.100 Regex (FAIL) [015261786] destination_number(77) =~ /^015261786$/ break=on-false
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:128 (sofia/internal/88 at 192.168.1.100) State Change CS_ROUTING -> CS_EXECUTE
2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:384 (sofia/internal/88 at 192.168.1.100) State ROUTING going to sleep
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_EXECUTE
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:391 (sofia/internal/88 at 192.168.1.100) State EXECUTE
2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:241 sofia/internal/88 at 192.168.1.100 SOFIA EXECUTE
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:166 sofia/internal/88 at 192.168.1.100 Standard EXECUTE
EXECUTE sofia/internal/88 at 192.168.1.100 set(outside_call=true)
2011-09-11 01:22:17.288082 [DEBUG] mod_dptools.c:1167 sofia/internal/88 at 192.168.1.100 SET [outside_call]=[true]
EXECUTE sofia/internal/88 at 192.168.1.100 set(RFC2822_DATE=Sun, 11 Sep 2011 01:22:17 +0200)
2011-09-11 01:22:17.288082 [DEBUG] mod_dptools.c:1167 sofia/internal/88 at 192.168.1.100 SET [RFC2822_DATE]=[Sun, 11 Sep 2011 01:22:17 +0200]
2011-09-11 01:22:17.288082 [NOTICE] switch_core_state_machine.c:200 sofia/internal/88 at 192.168.1.100 has executed the last dialplan instruction, hanging up.
2011-09-11 01:22:17.288082 [DEBUG] switch_channel.c:2797 (sofia/internal/88 at 192.168.1.100) Callstate Change RINGING -> HANGUP
2011-09-11 01:22:17.288082 [NOTICE] switch_core_state_machine.c:202 Hangup sofia/internal/88 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
2011-09-11 01:22:17.288082 [DEBUG] switch_channel.c:2813 Send signal sofia/internal/88 at 192.168.1.100 [KILL]
2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:391 (sofia/internal/88 at 192.168.1.100) State EXECUTE going to sleep
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_HANGUP
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:576 (sofia/internal/88 at 192.168.1.100) State HANGUP
2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:458 Channel sofia/internal/88 at 192.168.1.100 hanging up, cause: NORMAL_CLEARING
2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:522 Responding to INVITE with: 480
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:46 sofia/internal/88 at 192.168.1.100 Standard HANGUP, cause: NORMAL_CLEARING
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:576 (sofia/internal/88 at 192.168.1.100) State HANGUP going to sleep
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:367 (sofia/internal/88 at 192.168.1.100) State Change CS_HANGUP -> CS_REPORTING
2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_REPORTING
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/88 at 192.168.1.100) State REPORTING
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:53 sofia/internal/88 at 192.168.1.100 Standard REPORTING, cause: NORMAL_CLEARING
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/88 at 192.168.1.100) State REPORTING going to sleep
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/88 at 192.168.1.100) State Change CS_REPORTING -> CS_DESTROY
2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1349 Session 6 (sofia/internal/88 at 192.168.1.100) Locked, Waiting on external entities
2011-09-11 01:22:17.288082 [NOTICE] switch_core_session.c:1367 Session 6 (sofia/internal/88 at 192.168.1.100) Ended
2011-09-11 01:22:17.288082 [NOTICE] switch_core_session.c:1369 Close Channel sofia/internal/88 at 192.168.1.100 [CS_DESTROY]
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:465 (sofia/internal/88 at 192.168.1.100) Callstate Change HANGUP -> DOWN
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:468 (sofia/internal/88 at 192.168.1.100) Running State Change CS_DESTROY
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/88 at 192.168.1.100) State DESTROY
2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:363 sofia/internal/88 at 192.168.1.100 SOFIA DESTROY
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:60 sofia/internal/88 at 192.168.1.100 Standard DESTROY
2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/88 at 192.168.1.100) State DESTROY going to sleep



From gabe at gundy.org  Sun Sep 11 04:24:02 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 10 Sep 2011 18:24:02 -0600
Subject: [Freeswitch-users] FreeSWITCH with external registrar
In-Reply-To: <CAKq5mS7vSz4B2kxzgUn8vOJ9zPWp2Kwfw=HZEzW_Avd9UR2jHw@mail.gmail.com>
References: <CAKq5mS7qfy_u2Q6cG6aF1rZPctvXpHBkyzCUHk8feP=NMSv93g@mail.gmail.com>
	<CAKq5mS7vSz4B2kxzgUn8vOJ9zPWp2Kwfw=HZEzW_Avd9UR2jHw@mail.gmail.com>
Message-ID: <CAMwZ-t+EMF3C=Wwoc=Jhhy4SMg47Jqn4-jYLJDsv-3BnBCEWmw@mail.gmail.com>

On Fri, Sep 9, 2011 at 3:19 PM, envelopes envelopes
<sunwood360 at gmail.com> wrote:
> Is there any config examples of freeswitch working with external registar?

Try this:

http://wiki.freeswitch.org/wiki/SIP_Provider_Examples

Gabe


From gabe at gundy.org  Sun Sep 11 04:56:13 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 10 Sep 2011 18:56:13 -0600
Subject: [Freeswitch-users] freeswitch as a gateway with cdr lookup
In-Reply-To: <20110907170333.154010@gmx.net>
References: <20110907170333.154010@gmx.net>
Message-ID: <CAMwZ-t+Jc0kW79RFj9AFO=6Asev7Py=GKfkAurfxr1u33jaBQA@mail.gmail.com>

On Wed, Sep 7, 2011 at 11:03 AM, thomas peterseil <thomas.ji at gmx.at> wrote:
> Is there a possibility to put a FS between my PBX and the GSM Gateway, so when a call from the GSM Gateway comes in, FS makes a lookup in the CDR to check, which extension called this mobile number last time and then FS should route the call to the right extension.

You might try this... when you you call out, FS notes the caller and
the number they're calling and uses that info (later on) as a hint
when the return call comes back in.  Does that sound like something
that might work?

If so, you don't need to wait for the CDRs, just do it in the
dialplain (both inbound and outbound). Use the db module to persist
and later retrive that data.

http://wiki.freeswitch.org/wiki/Mod_db

Let us know how it goes.

Gabe


From gabe at gundy.org  Sun Sep 11 07:02:43 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 10 Sep 2011 21:02:43 -0600
Subject: [Freeswitch-users] Custom SIP Profile
In-Reply-To: <4E437DF4.2050108@gmail.com>
References: <CAMXUpRy=ft+puH8T9YZabiffPMC=cZSJGV-adKEoa64f0=oBzQ@mail.gmail.com>
	<4E437DF4.2050108@gmail.com>
Message-ID: <CAMwZ-tL2w+6-uV+v-27Vcb3O+yUDmimu+PFDzf4TaWvzkh78Rw@mail.gmail.com>

On Thu, Aug 11, 2011 at 1:00 AM, Nazim Aghabayov
<nazim.aghabayov at gmail.com> wrote:
> There is a nice wiki entry on sofia sip profiles at
> http://wiki.freeswitch.org/wiki/Sofia . It should answer most of your
> questions regarding sip profiles.

Also:

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


Gabe


From gabe at gundy.org  Sun Sep 11 07:10:30 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 10 Sep 2011 21:10:30 -0600
Subject: [Freeswitch-users] FS to a Sonus SIP trunk
In-Reply-To: <4E328433.8070701@hcu-hamburg.de>
References: <4E2ED4AB.1030305@hcu-hamburg.de> <4E31369A.70605@hcu-hamburg.de>
	<4E328433.8070701@hcu-hamburg.de>
Message-ID: <CAMwZ-tL0N0HJzWtNnC8CZmM2sqASY9L3qziKakptiQEMHuBW_A@mail.gmail.com>

On Fri, Jul 29, 2011 at 3:58 AM, michael knop
<michael.knop at hcu-hamburg.de> wrote:
> Hi all!
>
> I'm not sure, if my problem is caused by Sonus or is it's just a problem
> while negotiating the audio params. I put the log into pastebin:
> <http://pastebin.freeswitch.org/16913>
>
> Call starts with good sound quality. After "Duplicate SDP" (row 261)
> sound is choppy.


I know Anthony was recently doing more working to play nice with the
brokenness that is Sonus; you might want to check out the updates and
see if they make things better. I don't recall for sure what it was he
was addressing, but it might be worth a shot.

Gabe


From gabe at gundy.org  Sun Sep 11 07:41:15 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 10 Sep 2011 21:41:15 -0600
Subject: [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR
In-Reply-To: <BANLkTiko++ShiZKb_+bDNPprirrLeXp2RQ@mail.gmail.com>
References: <1QRrkF-0007cj-IK@mail.aastral.net>
	<BANLkTikc5bMokDY6__miju=rOUmsLc7kNg@mail.gmail.com>
	<1QRt7M-0002Zm-GE@mail.aastral.net>
	<BANLkTikXJrxuwUXAd2kYFCiDQOLGQgw9dg@mail.gmail.com>
	<1QS7yM-0005cW-8h@mail.aastral.net>
	<BANLkTi=hv5KZhMUfj1TKbucARH8gzADqpw@mail.gmail.com>
	<1QSAQM-0003GJ-6U@mail.aastral.net>
	<BANLkTink0R3qq96hQez786NJhFM_yAzrhg@mail.gmail.com>
	<1QTeTP-0003ff-Om@mail.aastral.net>
	<BANLkTiko++ShiZKb_+bDNPprirrLeXp2RQ@mail.gmail.com>
Message-ID: <CAMwZ-tKYOQcW+y0ZX2iq+iCYqJfBm_FnncZHpkQQS6YMHRZE6w@mail.gmail.com>

On Mon, Jun 6, 2011 at 3:07 PM, Michael Collins <msc at freeswitch.org> wrote:
> Thanks for the followup email. It's always nice to know how these problems
> eventually get solved...

I can't agree more with this. The least one can do to show their
gratitude for the help they get is to report if / how the issues they
were experiencing were resolved.

I'd sure like to see more of this.

Gabe


From gabe at gundy.org  Sun Sep 11 07:47:35 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 10 Sep 2011 21:47:35 -0600
Subject: [Freeswitch-users] Problem on some of of the calls (not calling
 / no Hangupcompleted event / Not released)
In-Reply-To: <CAGV=Xw1f4p6DP=ufSoHuyGg6Z9VGnEDER_RtawDZvL9y++M1Lg@mail.gmail.com>
References: <CAGV=Xw1f4p6DP=ufSoHuyGg6Z9VGnEDER_RtawDZvL9y++M1Lg@mail.gmail.com>
Message-ID: <CAMwZ-t+ySX+8VFzwgsAo1_KJOfpRK7hJoBRsw+AqG5+fAde5Dw@mail.gmail.com>

On Mon, Aug 8, 2011 at 3:14 PM, Frankie Yiu <frankie.k.yiu at gmail.com> wrote:
> Hi there,
>
> For testing purpose, I have my freeswitch calling itself.
> Once in awhile, I have originated a call, but calls don't seem to go through
> / or get "stuck".? I have code that would create record the whole session,
> and also report the call status by subscribing the HangupCompleted
> event--None of these happens.? Also when I check the status, the session was
> not released/destroyed while most of the calls went through OK.
>
> Could someone please tell me what might go wrong and how to fix this?
>
> My freeswitch is downloaded on 7/27/2011.


I know it's been a while, but if you're still seeing this, you should
update your FS server, and post the changes you made to the dialplan.

Gabe


From curriegrad2004 at gmail.com  Sun Sep 11 08:15:04 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 10 Sep 2011 21:15:04 -0700
Subject: [Freeswitch-users] Help needed on basic dialing plan: no calls
 are routed
In-Reply-To: <DB197D09-AC19-49F5-86AF-7951F94F690B@aharm.de>
References: <DB197D09-AC19-49F5-86AF-7951F94F690B@aharm.de>
Message-ID: <CAMWiMy95Ud-vY358XLDhKhMP=BoZ68SJ9nGgk3Bu57NSLveOWg@mail.gmail.com>

You haven't defined a dialplan for external calls yet. The Regex
didn't even match the number you're trying to dial out. Get a SIP
trunk or a TDM/J1 trunk first, register or make it work with
FreeSWITCH and then create a dialplan to call out.

This might be a good example to start out with:

http://wiki.freeswitch.org/wiki/Dialplan_XML#Dialing_through_gateways

On Sat, Sep 10, 2011 at 4:40 PM, Alexander Harm <contact at aharm.de> wrote:
> hello all,
>
> i have a freeswitch installation running and 2 phones connected to it. i setup 1 inbound rule. when i dial the external number i receive the following messages in the log. to me it seems that the routing works correctly but the ?transfer to extension 77 fails.
>
> note: further down the error messages i receive when trying to call internally.
>
> any help appreciated.
>
> regards, alexander
>
> LOG EXTERNAL -> INTERNAL
>
> 2011-09-11 01:17:17.368068 [NOTICE] switch_channel.c:908 New Channel sofia/external/0476058096 at voip.belgacom.be [068a829e-dc03-11e0-81f7-f9f410ff919e]
> 2011-09-11 01:17:17.368068 [DEBUG] sofia.c:5156 Channel sofia/external/0476058096 at voip.belgacom.be entering state [received][100]
> 2011-09-11 01:17:17.368068 [DEBUG] sofia.c:5167 Remote SDP:
> v=0
> o=- 0 98592702 IN IP4 81.240.251.38
> s=IMSS
> c=IN IP4 81.240.251.38
> t=0 0
> m=audio 10874 RTP/AVP 8 18 0 101
> a=rtpmap:101 telephone-event/8000/1
> a=fmtp:101 0-15
>
> 2011-09-11 01:17:17.368068 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
> 2011-09-11 01:17:17.368068 [DEBUG] sofia_glue.c:2857 Set Codec sofia/external/0476058096 at voip.belgacom.be PCMA/8000 20 ms 160 samples 64000 bits
> 2011-09-11 01:17:17.368068 [DEBUG] sofia_glue.c:4859 Set 2833 dtmf send/recv payload to 101
> 2011-09-11 01:17:17.368068 [DEBUG] sofia.c:5357 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_NEW -> CS_INIT
> 2011-09-11 01:17:17.368068 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
> 2011-09-11 01:17:17.368068 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_INIT
> 2011-09-11 01:17:17.368068 [DEBUG] switch_core_state_machine.c:375 (sofia/external/0476058096 at voip.belgacom.be) State INIT
> 2011-09-11 01:17:17.368068 [DEBUG] mod_sofia.c:85 sofia/external/0476058096 at voip.belgacom.be SOFIA INIT
> 2011-09-11 01:17:17.368068 [DEBUG] mod_sofia.c:125 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_INIT -> CS_ROUTING
> 2011-09-11 01:17:17.368068 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
> 2011-09-11 01:17:17.368068 [DEBUG] switch_core_state_machine.c:375 (sofia/external/0476058096 at voip.belgacom.be) State INIT going to sleep
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_ROUTING
> 2011-09-11 01:17:17.388551 [DEBUG] switch_channel.c:1837 (sofia/external/0476058096 at voip.belgacom.be) Callstate Change DOWN -> RINGING
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING
> 2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:148 sofia/external/0476058096 at voip.belgacom.be SOFIA ROUTING
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:78 sofia/external/0476058096 at voip.belgacom.be Standard ROUTING
> 2011-09-11 01:17:17.388551 [INFO] mod_dialplan_xml.c:336 Processing 0476058096 <0476058096>->015261786 in context public
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->unloop] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->outside_call] continue=true
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Absolute Condition [outside_call]
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action set(outside_call=true)
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->call_debug] continue=true
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->public_extensions] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [public_extensions] destination_number(015261786) =~ /^(10[01][0-9])$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->015261786] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [015261786] context(public) =~ /public/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [015261786] destination_number(015261786) =~ /^015261786$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action transfer(77 XML default)
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:128 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_ROUTING -> CS_EXECUTE
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING going to sleep
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_EXECUTE
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE
> 2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:241 sofia/external/0476058096 at voip.belgacom.be SOFIA EXECUTE
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:166 sofia/external/0476058096 at voip.belgacom.be Standard EXECUTE
> EXECUTE sofia/external/0476058096 at voip.belgacom.be set(outside_call=true)
> 2011-09-11 01:17:17.388551 [DEBUG] mod_dptools.c:1167 sofia/external/0476058096 at voip.belgacom.be SET [outside_call]=[true]
> EXECUTE sofia/external/0476058096 at voip.belgacom.be set(RFC2822_DATE=Sun, 11 Sep 2011 01:17:17 +0200)
> 2011-09-11 01:17:17.388551 [DEBUG] mod_dptools.c:1167 sofia/external/0476058096 at voip.belgacom.be SET [RFC2822_DATE]=[Sun, 11 Sep 2011 01:17:17 +0200]
> EXECUTE sofia/external/0476058096 at voip.belgacom.be transfer(77 XML default)
> 2011-09-11 01:17:17.388551 [DEBUG] switch_ivr.c:1693 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_EXECUTE -> CS_ROUTING
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:724 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
> 2011-09-11 01:17:17.388551 [NOTICE] switch_ivr.c:1699 Transfer sofia/external/0476058096 at voip.belgacom.be to XML[77 at default]
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE going to sleep
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_ROUTING
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING
> 2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:148 sofia/external/0476058096 at voip.belgacom.be SOFIA ROUTING
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:78 sofia/external/0476058096 at voip.belgacom.be Standard ROUTING
> 2011-09-11 01:17:17.388551 [INFO] mod_dialplan_xml.c:336 Processing 0476058096 <0476058096>->77 in context default
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->unloop] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->tod_example] continue=true
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Date/Time Match (FAIL) [tod_example] break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->holiday_example] continue=true
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Date/Time Match (FAIL) [holiday_example] break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->global-intercept] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [global-intercept] destination_number(77) =~ /^886$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->group-intercept] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [group-intercept] destination_number(77) =~ /^\*8$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->intercept-ext] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [intercept-ext] destination_number(77) =~ /^\*\*(\d+)$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->redial] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [redial] destination_number(77) =~ /^(redial|870)$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->global] continue=true
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Absolute Condition [global]
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action hash(insert/${domain_name}-last_dial/global/${uuid})
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->snom-demo-2] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [snom-demo-2] destination_number(77) =~ /^9001$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->snom-demo-1] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [snom-demo-1] destination_number(77) =~ /^9000$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->eavesdrop] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [eavesdrop] destination_number(77) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->eavesdrop] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [eavesdrop] destination_number(77) =~ /^779$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->call_return] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call_return] destination_number(77) =~ /^\*69$|^869$|^lcr$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->del-group] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [del-group] destination_number(77) =~ /^80(\d{2})$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->add-group] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [add-group] destination_number(77) =~ /^81(\d{2})$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->call-group-simo] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call-group-simo] destination_number(77) =~ /^82(\d{2})$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->call-group-order] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call-group-order] destination_number(77) =~ /^83(\d{2})$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->extension-intercom] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [extension-intercom] destination_number(77) =~ /^8(10[01][0-9])$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->Local_Extension] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [Local_Extension] destination_number(77) =~ /^(10[01][0-9])$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->Local_Extension_Skinny] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [Local_Extension_Skinny] destination_number(77) =~ /^(11[01][0-9])$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->group_dial_sales] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [group_dial_sales] destination_number(77) =~ /^2000$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->group_dial_support] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [group_dial_support] destination_number(77) =~ /^2001$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->group_dial_billing] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [group_dial_billing] destination_number(77) =~ /^2002$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->operator] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [operator] destination_number(77) =~ /^(operator|0)$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->vmain] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [vmain] destination_number(77) =~ /^vmain$|^4000$|^\*98$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->sip_uri] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [sip_uri] destination_number(77) =~ /^sip:(.*)$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->nb_conferences] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [nb_conferences] destination_number(77) =~ /^(30\d{2})$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->wb_conferences] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [wb_conferences] destination_number(77) =~ /^(31\d{2})$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->uwb_conferences] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [uwb_conferences] destination_number(77) =~ /^(32\d{2})$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->cdquality_conferences] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [cdquality_conferences] destination_number(77) =~ /^(33\d{2})$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->freeswitch_public_conf_via_sip] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(77) =~ /^9(888|8888|1616|3232)$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->mad_boss_intercom] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [mad_boss_intercom] destination_number(77) =~ /^0911$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->mad_boss_intercom] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [mad_boss_intercom] destination_number(77) =~ /^0912$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->mad_boss] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [mad_boss] destination_number(77) =~ /^0913$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ivr_demo] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ivr_demo] destination_number(77) =~ /^5000$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->dynamic_conference] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [dynamic_conference] destination_number(77) =~ /^5001$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->rtp_multicast_page] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [rtp_multicast_page] destination_number(77) =~ /^pagegroup$|^7243$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->park] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [park] destination_number(77) =~ /^5900$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->unpark] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unpark] destination_number(77) =~ /^5901$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->valet_park] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [valet_park] destination_number(77) =~ /^(6000)$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->valet_park] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [valet_park] destination_number(77) =~ /^(60\d[1-9])$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->park] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [park] destination_number(77) =~ /park\+(\d+)/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->unpark] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unpark] destination_number(77) =~ /^parking$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->park] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [park] destination_number(77) =~ /callpark/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->unpark] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unpark] destination_number(77) =~ /pickup/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->wait] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [wait] destination_number(77) =~ /^wait$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->fax_receive] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [fax_receive] destination_number(77) =~ /^9178$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->fax_transmit] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [fax_transmit] destination_number(77) =~ /^9179$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_180] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_180] destination_number(77) =~ /^9180$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_183_uk_ring] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_183_uk_ring] destination_number(77) =~ /^9181$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_183_music_ring] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_183_music_ring] destination_number(77) =~ /^9182$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_post_answer_uk_ring] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(77) =~ /^9183$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_post_answer_music] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_post_answer_music] destination_number(77) =~ /^9184$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ClueCon] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ClueCon] destination_number(77) =~ /^9191$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->show_info] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [show_info] destination_number(77) =~ /^9192$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->video_record] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [video_record] destination_number(77) =~ /^9193$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->video_playback] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [video_playback] destination_number(77) =~ /^9194$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->delay_echo] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [delay_echo] destination_number(77) =~ /^9195$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->echo] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [echo] destination_number(77) =~ /^9196$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->milliwatt] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [milliwatt] destination_number(77) =~ /^9197$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->tone_stream] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [tone_stream] destination_number(77) =~ /^9198$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->zrtp_enrollement] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [zrtp_enrollement] destination_number(77) =~ /^9787$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->hold_music] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [hold_music] destination_number(77) =~ /^9664$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->Recordings] continue=false
> 2011-09-11 01:17:17.388551 [ERR] switch_regex.c:101 COMPILE ERROR: 1 [nothing to repeat][^*(732)$]
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [Recordings] destination_number(77) =~ /^*(732)$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->enum] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [enum] ${module_exists(mod_enum)}(true) =~ /true/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [enum] destination_number(77) =~ /^(.*)$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action transfer(77 enum)
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:128 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_ROUTING -> CS_EXECUTE
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING going to sleep
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_EXECUTE
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE
> 2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:241 sofia/external/0476058096 at voip.belgacom.be SOFIA EXECUTE
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:166 sofia/external/0476058096 at voip.belgacom.be Standard EXECUTE
> EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.100-spymap/0476058096/068a829e-dc03-11e0-81f7-f9f410ff919e)
> EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.100-last_dial/0476058096/77)
> EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.100-last_dial/global/068a829e-dc03-11e0-81f7-f9f410ff919e)
> EXECUTE sofia/external/0476058096 at voip.belgacom.be set(RFC2822_DATE=Sun, 11 Sep 2011 01:17:17 +0200)
> 2011-09-11 01:17:17.388551 [DEBUG] mod_dptools.c:1167 sofia/external/0476058096 at voip.belgacom.be SET [RFC2822_DATE]=[Sun, 11 Sep 2011 01:17:17 +0200]
> EXECUTE sofia/external/0476058096 at voip.belgacom.be transfer(77 enum)
> 2011-09-11 01:17:17.388551 [DEBUG] switch_ivr.c:1693 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_EXECUTE -> CS_ROUTING
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:724 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
> 2011-09-11 01:17:17.388551 [NOTICE] switch_ivr.c:1699 Transfer sofia/external/0476058096 at voip.belgacom.be to enum[77 at default]
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE going to sleep
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_ROUTING
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING
> 2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:148 sofia/external/0476058096 at voip.belgacom.be SOFIA ROUTING
> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:78 sofia/external/0476058096 at voip.belgacom.be Standard ROUTING
> 2011-09-11 01:17:17.388551 [DEBUG] mod_enum.c:541 ENUM Lookup on 77
> 2011-09-11 01:17:27.188065 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
> 2011-09-11 01:17:27.188065 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
> 2011-09-11 01:17:27.188065 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>
>
> LOG ?INTERNAL -> INTERNAL
>
>
> 2011-09-11 01:22:17.268071 [DEBUG] sofia.c:7057 IP 192.168.100.6 Approved by acl "lan[]". Access Granted.
> 2011-09-11 01:22:17.268071 [NOTICE] switch_channel.c:908 New Channel sofia/internal/88 at 192.168.1.100 [b94b66fa-dc03-11e0-81f9-f9f410ff919e]
> 2011-09-11 01:22:17.268071 [DEBUG] sofia.c:5156 Channel sofia/internal/88 at 192.168.1.100 entering state [received][100]
> 2011-09-11 01:22:17.268071 [DEBUG] sofia.c:5167 Remote SDP:
> v=0
> o=- 1315696937270159 1 IN IP4 192.168.100.6
> s=CounterPath X-Lite 4.1
> c=IN IP4 192.168.100.6
> t=0 0
> a=ice-ufrag:de6e17
> a=ice-pwd:9e8391bc4955d13bbf67e8eeb2118613
> m=audio 63158 RTP/AVP 0 8 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=candidate:1 1 UDP 659136 192.168.100.6 63158 typ host
> a=candidate:1 2 UDP 659134 192.168.100.6 63159 typ host
>
> 2011-09-11 01:22:17.268071 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_NEW
> 2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMA:8:8000:20:64000]
> 2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMU:0:8000:20:64000]/[G729:18:8000:20:8000]
> 2011-09-11 01:22:17.268071 [DEBUG] switch_core_state_machine.c:354 (sofia/internal/88 at 192.168.1.100) State NEW
> 2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
> 2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:2857 Set Codec sofia/internal/88 at 192.168.1.100 PCMA/8000 20 ms 160 samples 64000 bits
> 2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:4859 Set 2833 dtmf send/recv payload to 101
> 2011-09-11 01:22:17.268071 [DEBUG] sofia.c:5357 (sofia/internal/88 at 192.168.1.100) State Change CS_NEW -> CS_INIT
> 2011-09-11 01:22:17.268071 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_INIT
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:375 (sofia/internal/88 at 192.168.1.100) State INIT
> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:85 sofia/internal/88 at 192.168.1.100 SOFIA INIT
> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:125 (sofia/internal/88 at 192.168.1.100) State Change CS_INIT -> CS_ROUTING
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:375 (sofia/internal/88 at 192.168.1.100) State INIT going to sleep
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_ROUTING
> 2011-09-11 01:22:17.288082 [DEBUG] switch_channel.c:1837 (sofia/internal/88 at 192.168.1.100) Callstate Change DOWN -> RINGING
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:384 (sofia/internal/88 at 192.168.1.100) State ROUTING
> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:148 sofia/internal/88 at 192.168.1.100 SOFIA ROUTING
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:78 sofia/internal/88 at 192.168.1.100 Standard ROUTING
> 2011-09-11 01:22:17.288082 [INFO] mod_dialplan_xml.c:336 Processing 88 <88>->77 in context public
> Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->unloop] continue=false
> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->outside_call] continue=true
> Dialplan: sofia/internal/88 at 192.168.1.100 Absolute Condition [outside_call]
> Dialplan: sofia/internal/88 at 192.168.1.100 Action set(outside_call=true)
> Dialplan: sofia/internal/88 at 192.168.1.100 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
> Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->call_debug] continue=true
> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->public_extensions] continue=false
> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (FAIL) [public_extensions] destination_number(77) =~ /^(10[01][0-9])$/ break=on-false
> Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->015261786] continue=false
> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (PASS) [015261786] context(public) =~ /public/ break=on-false
> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (FAIL) [015261786] destination_number(77) =~ /^015261786$/ break=on-false
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:128 (sofia/internal/88 at 192.168.1.100) State Change CS_ROUTING -> CS_EXECUTE
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:384 (sofia/internal/88 at 192.168.1.100) State ROUTING going to sleep
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_EXECUTE
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:391 (sofia/internal/88 at 192.168.1.100) State EXECUTE
> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:241 sofia/internal/88 at 192.168.1.100 SOFIA EXECUTE
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:166 sofia/internal/88 at 192.168.1.100 Standard EXECUTE
> EXECUTE sofia/internal/88 at 192.168.1.100 set(outside_call=true)
> 2011-09-11 01:22:17.288082 [DEBUG] mod_dptools.c:1167 sofia/internal/88 at 192.168.1.100 SET [outside_call]=[true]
> EXECUTE sofia/internal/88 at 192.168.1.100 set(RFC2822_DATE=Sun, 11 Sep 2011 01:22:17 +0200)
> 2011-09-11 01:22:17.288082 [DEBUG] mod_dptools.c:1167 sofia/internal/88 at 192.168.1.100 SET [RFC2822_DATE]=[Sun, 11 Sep 2011 01:22:17 +0200]
> 2011-09-11 01:22:17.288082 [NOTICE] switch_core_state_machine.c:200 sofia/internal/88 at 192.168.1.100 has executed the last dialplan instruction, hanging up.
> 2011-09-11 01:22:17.288082 [DEBUG] switch_channel.c:2797 (sofia/internal/88 at 192.168.1.100) Callstate Change RINGING -> HANGUP
> 2011-09-11 01:22:17.288082 [NOTICE] switch_core_state_machine.c:202 Hangup sofia/internal/88 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-09-11 01:22:17.288082 [DEBUG] switch_channel.c:2813 Send signal sofia/internal/88 at 192.168.1.100 [KILL]
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:391 (sofia/internal/88 at 192.168.1.100) State EXECUTE going to sleep
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_HANGUP
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:576 (sofia/internal/88 at 192.168.1.100) State HANGUP
> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:458 Channel sofia/internal/88 at 192.168.1.100 hanging up, cause: NORMAL_CLEARING
> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:522 Responding to INVITE with: 480
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:46 sofia/internal/88 at 192.168.1.100 Standard HANGUP, cause: NORMAL_CLEARING
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:576 (sofia/internal/88 at 192.168.1.100) State HANGUP going to sleep
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:367 (sofia/internal/88 at 192.168.1.100) State Change CS_HANGUP -> CS_REPORTING
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_REPORTING
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/88 at 192.168.1.100) State REPORTING
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:53 sofia/internal/88 at 192.168.1.100 Standard REPORTING, cause: NORMAL_CLEARING
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/88 at 192.168.1.100) State REPORTING going to sleep
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/88 at 192.168.1.100) State Change CS_REPORTING -> CS_DESTROY
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1349 Session 6 (sofia/internal/88 at 192.168.1.100) Locked, Waiting on external entities
> 2011-09-11 01:22:17.288082 [NOTICE] switch_core_session.c:1367 Session 6 (sofia/internal/88 at 192.168.1.100) Ended
> 2011-09-11 01:22:17.288082 [NOTICE] switch_core_session.c:1369 Close Channel sofia/internal/88 at 192.168.1.100 [CS_DESTROY]
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:465 (sofia/internal/88 at 192.168.1.100) Callstate Change HANGUP -> DOWN
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:468 (sofia/internal/88 at 192.168.1.100) Running State Change CS_DESTROY
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/88 at 192.168.1.100) State DESTROY
> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:363 sofia/internal/88 at 192.168.1.100 SOFIA DESTROY
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:60 sofia/internal/88 at 192.168.1.100 Standard DESTROY
> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/88 at 192.168.1.100) State DESTROY going to sleep
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From spencer at 5ninesolutions.com  Sun Sep 11 08:19:06 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sat, 10 Sep 2011 21:19:06 -0700
Subject: [Freeswitch-users] Follow Me and Call Forwarding
Message-ID: <DB319D80-A364-41A8-B972-75A1713B7C7C@5ninesolutions.com>

Hello all,
I'd like to implement a graphical call forwarding and follow me feature that works from hunt groups as well as a direct dial to an extension.  Currently when a user enables follow me for their extension a new dialplan entry is created with the same destination number as the extension before the local user extension which dials the extension with a bridge to user/${dialed_extension}@${domain_name}.  My hunt groups are setup with a lua script that rings the desired extensions with a long bridge statement to user/EXTENSION@${domain_name}.  Is there any way to call a user directly (i.e. using user/${dialed_extension}@${domain_name} instead of a transfer) and do some sort of call forwarding or follow me feature?


Thanks for your suggestions!

Spencer

From avi at avimarcus.net  Sun Sep 11 09:22:44 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 11 Sep 2011 08:22:44 +0300
Subject: [Freeswitch-users] Follow Me and Call Forwarding
In-Reply-To: <DB319D80-A364-41A8-B972-75A1713B7C7C@5ninesolutions.com>
References: <DB319D80-A364-41A8-B972-75A1713B7C7C@5ninesolutions.com>
Message-ID: <CALNZuwmXfw23LZRSD=JCN5QOwN4OHAGXtgiy_kGOWcY1mo+nxQ@mail.gmail.com>

It sounds like you already have it doing what you want it to do. How do you
want it to work differently..?
-Avi



On Sun, Sep 11, 2011 at 7:19 AM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Hello all,
> I'd like to implement a graphical call forwarding and follow me feature
> that works from hunt groups as well as a direct dial to an extension.
>  Currently when a user enables follow me for their extension a new dialplan
> entry is created with the same destination number as the extension before
> the local user extension which dials the extension with a bridge to
> user/${dialed_extension}@${domain_name}.  My hunt groups are setup with a
> lua script that rings the desired extensions with a long bridge statement to
> user/EXTENSION@${domain_name}.  Is there any way to call a user directly
> (i.e. using user/${dialed_extension}@${domain_name} instead of a transfer)
> and do some sort of call forwarding or follow me feature?
>
>
> Thanks for your suggestions!
>
> Spencer
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110911/1a708519/attachment.html 

From devel at omninet.eu  Sun Sep 11 09:43:57 2011
From: devel at omninet.eu (Anestis Mavro)
Date: Sun, 11 Sep 2011 08:43:57 +0300
Subject: [Freeswitch-users] mod_lcr transfer shows wrong caller id
Message-ID: <0C64150CB61444419B7A1981CABCF183@omni1.local>

Hi,

 

I've been using mod_lcr with nibblebill for a while and everything worked
fine with the bridge application. 

Now I would like to add limit and I have to use it as dialplan like this:

 

<action application="transfer" data="${callednumber} lcr ${lcrprofile}" />

 

I managed to configure the lcr.conf.xml correctly for Limit and everything
seems to be working beside the caller id.

Now the variable origination_caller_id_number gets the accountcode instead
of the correct number.

 

I've tried to explicitly set the variable before transferring and even in
the lcr.conf.xml and export it. no chance.

 

Any idea??

 

Thanks

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110911/1f86f970/attachment.html 

From spencer at 5ninesolutions.com  Sun Sep 11 09:46:56 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sat, 10 Sep 2011 22:46:56 -0700
Subject: [Freeswitch-users] Follow Me and Call Forwarding
In-Reply-To: <CALNZuwmXfw23LZRSD=JCN5QOwN4OHAGXtgiy_kGOWcY1mo+nxQ@mail.gmail.com>
References: <DB319D80-A364-41A8-B972-75A1713B7C7C@5ninesolutions.com>
	<CALNZuwmXfw23LZRSD=JCN5QOwN4OHAGXtgiy_kGOWcY1mo+nxQ@mail.gmail.com>
Message-ID: <1B608BE2-2D12-48B9-ACAF-C14E220C861C@5ninesolutions.com>

The problem is that it doesn't work from a hunt group because the extensions are dialed with a bridge to user/EXTENSION@${domain_name} instead of fro example transfer EXTENSION XML default since the follow me or forward is inserted as another extension with the same number just earlier in the dialplan.

 
On Sep 10, 2011, at 10:22 PM, Avi Marcus wrote:

> It sounds like you already have it doing what you want it to do. How do you want it to work differently..?
> -Avi
> 
> 
> 
> On Sun, Sep 11, 2011 at 7:19 AM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
> Hello all,
> I'd like to implement a graphical call forwarding and follow me feature that works from hunt groups as well as a direct dial to an extension.  Currently when a user enables follow me for their extension a new dialplan entry is created with the same destination number as the extension before the local user extension which dials the extension with a bridge to user/${dialed_extension}@${domain_name}.  My hunt groups are setup with a lua script that rings the desired extensions with a long bridge statement to user/EXTENSION@${domain_name}.  Is there any way to call a user directly (i.e. using user/${dialed_extension}@${domain_name} instead of a transfer) and do some sort of call forwarding or follow me feature?
> 
> 
> Thanks for your suggestions!
> 
> Spencer
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110910/f4c2bca3/attachment.html 

From avi at avimarcus.net  Sun Sep 11 09:47:23 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 11 Sep 2011 08:47:23 +0300
Subject: [Freeswitch-users] mod_lcr transfer shows wrong caller id
In-Reply-To: <0C64150CB61444419B7A1981CABCF183@omni1.local>
References: <0C64150CB61444419B7A1981CABCF183@omni1.local>
Message-ID: <CALNZuw=UOcJ810Qk5BnN+vrDk6jX1N0gjxdJi8-Ry7nWO9htAg@mail.gmail.com>

Did you set effective_caller_id in the A-leg to the value you want?
-Avi


On Sun, Sep 11, 2011 at 8:43 AM, Anestis Mavro <devel at omninet.eu> wrote:

>  Hi,****
>
> ** **
>
> I?ve been using mod_lcr with nibblebill for a while and everything worked
> fine with the bridge application. ****
>
> Now I would like to add limit and I have to use it as dialplan like this:*
> ***
>
> ** **
>
> <action application=?transfer? data=?${callednumber} lcr ${lcrprofile}? />
> ****
>
> ** **
>
> I managed to configure the lcr.conf.xml correctly for Limit and everything
> seems to be working beside the caller id.****
>
> Now the variable origination_caller_id_number gets the accountcode instead
> of the correct number.****
>
> ** **
>
> I?ve tried to explicitly set the variable before transferring and even in
> the lcr.conf.xml and export it? no chance?****
>
> ** **
>
> Any idea??****
>
> ** **
>
> Thanks****
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 5054 (20100423) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110911/2a478dd6/attachment-0001.html 

From devel at omninet.eu  Sun Sep 11 10:56:18 2011
From: devel at omninet.eu (Anestis Mavro)
Date: Sun, 11 Sep 2011 09:56:18 +0300
Subject: [Freeswitch-users] mod_lcr transfer shows wrong caller id
In-Reply-To: <CALNZuw=UOcJ810Qk5BnN+vrDk6jX1N0gjxdJi8-Ry7nWO9htAg@mail.gmail.com>
References: <0C64150CB61444419B7A1981CABCF183@omni1.local>
	<CALNZuw=UOcJ810Qk5BnN+vrDk6jX1N0gjxdJi8-Ry7nWO9htAg@mail.gmail.com>
Message-ID: <FB36DABAE032438AA68C8DC2C1C7877F@omni1.local>

Yes, the same result.

 

I made now a trace and found that in the "From" and "P-Asserted-Identity"
the display name is correct (the number) but the display number is wrong
(the account).

 

Wondering now where to set them to correctly apply.

 

Thanks

Anestis

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Sunday, September 11, 2011 8:47 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_lcr transfer shows wrong caller id

 

Did you set effective_caller_id in the A-leg to the value you want?


-Avi

 

On Sun, Sep 11, 2011 at 8:43 AM, Anestis Mavro <devel at omninet.eu> wrote:

Hi,

 

I've been using mod_lcr with nibblebill for a while and everything worked
fine with the bridge application. 

Now I would like to add limit and I have to use it as dialplan like this:

 

<action application="transfer" data="${callednumber} lcr ${lcrprofile}" />

 

I managed to configure the lcr.conf.xml correctly for Limit and everything
seems to be working beside the caller id.

Now the variable origination_caller_id_number gets the accountcode instead
of the correct number.

 

I've tried to explicitly set the variable before transferring and even in
the lcr.conf.xml and export it. no chance.

 

Any idea??

 

Thanks



__________ Information from ESET NOD32 Antivirus, version of virus signature
database 5054 (20100423) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.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

 

 

__________ Information from ESET NOD32 Antivirus, version of virus signature
database 5054 (20100423) __________

 

The message was checked by ESET NOD32 Antivirus.

 

http://www.eset.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110911/df140a5a/attachment.html 

From contact at aharm.de  Sun Sep 11 11:13:41 2011
From: contact at aharm.de (Alexander Harm)
Date: Sun, 11 Sep 2011 09:13:41 +0200
Subject: [Freeswitch-users] Help needed on basic dialing plan: no calls
	are routed
In-Reply-To: <CAMWiMy95Ud-vY358XLDhKhMP=BoZ68SJ9nGgk3Bu57NSLveOWg@mail.gmail.com>
References: <DB197D09-AC19-49F5-86AF-7951F94F690B@aharm.de>
	<CAMWiMy95Ud-vY358XLDhKhMP=BoZ68SJ9nGgk3Bu57NSLveOWg@mail.gmail.com>
Message-ID: <CF4DF76C-749B-483C-B814-B8B2C41C3455@aharm.de>

hi there,

thanks for the reply. have to admit that i'm not completely sure that i understand you correctly.

my problem:

- i cannot call from one extension to another extension (not sure where the outbound route comes in here, nor the sip trunk)

- my calls from external to an extension are not transferred (sip trunk is registered and the call arrives at the gateway), caller receives no ringtone and internal extension doesn't ring either

to follow your advice i did setup an outbound rule (all number starting with 0 are routed through the gateway). it makes no difference at all, the calls from internal extensions fail.

hope you have some other ideas.

alexander

extension 77

<include>
 <user id="77">
   <params>
     <param name="password" value="xxx"/>
     <param name="vm-password" value="user-choose"/>
     <param name="vm-enabled" value="false"/>
   </params>
   <variables>
     <variable name="toll_allow" value="international"/>
     <variable name="accountcode" value="77"/>
     <variable name="user_context" value="default"/>
     <variable name="effective_caller_id_name" value="77"/>
     <variable name="effective_caller_id_number" value="77"/>
   </variables>
 </user>
</include>

extension 88

<include>
 <user id="88">
   <params>
     <param name="password" value="xxx"/>
     <param name="vm-password" value="user-choose"/>
     <param name="vm-enabled" value="false"/>
   </params>
   <variables>
     <variable name="toll_allow" value="international"/>
     <variable name="accountcode" value="88"/>
     <variable name="user_context" value="default"/>
     <variable name="effective_caller_id_name" value="88"/>
     <variable name="effective_caller_id_number" value="88"/>
   </variables>
 </user>
</include>

my trunk

<include>
   <gateway name="015261786">
     <param name="username" value="u15261786"/>
     <param name="auth-username" value="u15261786"/>
     <param name="password" value="xxx"/>
     <param name="realm" value="voip.belgacom.be"/>
     <param name="from-user" value="015261786"/>
     <param name="from-domain" value="voip.belgacom.be"/>
     <param name="proxy" value="voip.belgacom.be"/>
     <param name="outbound-proxy" value="81.240.251.22"/>
     <param name="expire-seconds" value="1800"/>
     <param name="register" value="true"/>
     <param name="retry-seconds" value="60"/>
     <param name="extension" value="015261786"/>
     <param name="context" value="public"/>
   </gateway>
</include>

my inbound route

<extension name="015261786" >
  <condition field="context" expression="public"/>
  <condition field="destination_number" expression="^015261786$">
      <action application="transfer" data="77 XML default"/>
  </condition>
</extension>

my outbound route

<extension name="015261786.0d" >
  <condition field="destination_number" expression="^0(\d+)$" >
      <action application="set" data="sip_h_X-accountcode=${accountcode}" />
      <action application="set" data="call_direction=outbound" />
      <action application="set" data="hangup_after_bridge=true" />
      <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}" />
      <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}" />
      <action application="set" data="inherit_codec=true" />
      <action application="bridge" data="sofia/gateway/015261786/$1" />
  </condition>
</extension>



On 11.09.2011, at 06:15, curriegrad2004 wrote:

> You haven't defined a dialplan for external calls yet. The Regex
> didn't even match the number you're trying to dial out. Get a SIP
> trunk or a TDM/J1 trunk first, register or make it work with
> FreeSWITCH and then create a dialplan to call out.
> 
> This might be a good example to start out with:
> 
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Dialing_through_gateways
> 
> On Sat, Sep 10, 2011 at 4:40 PM, Alexander Harm <contact at aharm.de> wrote:
>> hello all,
>> 
>> i have a freeswitch installation running and 2 phones connected to it. i setup 1 inbound rule. when i dial the external number i receive the following messages in the log. to me it seems that the routing works correctly but the  transfer to extension 77 fails.
>> 
>> note: further down the error messages i receive when trying to call internally.
>> 
>> any help appreciated.
>> 
>> regards, alexander
>> 
>> LOG EXTERNAL -> INTERNAL
>> 
>> 2011-09-11 01:17:17.368068 [NOTICE] switch_channel.c:908 New Channel sofia/external/0476058096 at voip.belgacom.be [068a829e-dc03-11e0-81f7-f9f410ff919e]
>> 2011-09-11 01:17:17.368068 [DEBUG] sofia.c:5156 Channel sofia/external/0476058096 at voip.belgacom.be entering state [received][100]
>> 2011-09-11 01:17:17.368068 [DEBUG] sofia.c:5167 Remote SDP:
>> v=0
>> o=- 0 98592702 IN IP4 81.240.251.38
>> s=IMSS
>> c=IN IP4 81.240.251.38
>> t=0 0
>> m=audio 10874 RTP/AVP 8 18 0 101
>> a=rtpmap:101 telephone-event/8000/1
>> a=fmtp:101 0-15
>> 
>> 2011-09-11 01:17:17.368068 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
>> 2011-09-11 01:17:17.368068 [DEBUG] sofia_glue.c:2857 Set Codec sofia/external/0476058096 at voip.belgacom.be PCMA/8000 20 ms 160 samples 64000 bits
>> 2011-09-11 01:17:17.368068 [DEBUG] sofia_glue.c:4859 Set 2833 dtmf send/recv payload to 101
>> 2011-09-11 01:17:17.368068 [DEBUG] sofia.c:5357 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_NEW -> CS_INIT
>> 2011-09-11 01:17:17.368068 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 2011-09-11 01:17:17.368068 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_INIT
>> 2011-09-11 01:17:17.368068 [DEBUG] switch_core_state_machine.c:375 (sofia/external/0476058096 at voip.belgacom.be) State INIT
>> 2011-09-11 01:17:17.368068 [DEBUG] mod_sofia.c:85 sofia/external/0476058096 at voip.belgacom.be SOFIA INIT
>> 2011-09-11 01:17:17.368068 [DEBUG] mod_sofia.c:125 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_INIT -> CS_ROUTING
>> 2011-09-11 01:17:17.368068 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 2011-09-11 01:17:17.368068 [DEBUG] switch_core_state_machine.c:375 (sofia/external/0476058096 at voip.belgacom.be) State INIT going to sleep
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_ROUTING
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_channel.c:1837 (sofia/external/0476058096 at voip.belgacom.be) Callstate Change DOWN -> RINGING
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING
>> 2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:148 sofia/external/0476058096 at voip.belgacom.be SOFIA ROUTING
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:78 sofia/external/0476058096 at voip.belgacom.be Standard ROUTING
>> 2011-09-11 01:17:17.388551 [INFO] mod_dialplan_xml.c:336 Processing 0476058096 <0476058096>->015261786 in context public
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->unloop] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->outside_call] continue=true
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Absolute Condition [outside_call]
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action set(outside_call=true)
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->call_debug] continue=true
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->public_extensions] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [public_extensions] destination_number(015261786) =~ /^(10[01][0-9])$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->015261786] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [015261786] context(public) =~ /public/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [015261786] destination_number(015261786) =~ /^015261786$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action transfer(77 XML default)
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:128 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_ROUTING -> CS_EXECUTE
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING going to sleep
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_EXECUTE
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE
>> 2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:241 sofia/external/0476058096 at voip.belgacom.be SOFIA EXECUTE
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:166 sofia/external/0476058096 at voip.belgacom.be Standard EXECUTE
>> EXECUTE sofia/external/0476058096 at voip.belgacom.be set(outside_call=true)
>> 2011-09-11 01:17:17.388551 [DEBUG] mod_dptools.c:1167 sofia/external/0476058096 at voip.belgacom.be SET [outside_call]=[true]
>> EXECUTE sofia/external/0476058096 at voip.belgacom.be set(RFC2822_DATE=Sun, 11 Sep 2011 01:17:17 +0200)
>> 2011-09-11 01:17:17.388551 [DEBUG] mod_dptools.c:1167 sofia/external/0476058096 at voip.belgacom.be SET [RFC2822_DATE]=[Sun, 11 Sep 2011 01:17:17 +0200]
>> EXECUTE sofia/external/0476058096 at voip.belgacom.be transfer(77 XML default)
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_ivr.c:1693 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_EXECUTE -> CS_ROUTING
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:724 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 2011-09-11 01:17:17.388551 [NOTICE] switch_ivr.c:1699 Transfer sofia/external/0476058096 at voip.belgacom.be to XML[77 at default]
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE going to sleep
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_ROUTING
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING
>> 2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:148 sofia/external/0476058096 at voip.belgacom.be SOFIA ROUTING
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:78 sofia/external/0476058096 at voip.belgacom.be Standard ROUTING
>> 2011-09-11 01:17:17.388551 [INFO] mod_dialplan_xml.c:336 Processing 0476058096 <0476058096>->77 in context default
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->unloop] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->tod_example] continue=true
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Date/Time Match (FAIL) [tod_example] break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->holiday_example] continue=true
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Date/Time Match (FAIL) [holiday_example] break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->global-intercept] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [global-intercept] destination_number(77) =~ /^886$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->group-intercept] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [group-intercept] destination_number(77) =~ /^\*8$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->intercept-ext] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [intercept-ext] destination_number(77) =~ /^\*\*(\d+)$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->redial] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [redial] destination_number(77) =~ /^(redial|870)$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->global] continue=true
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Absolute Condition [global]
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action hash(insert/${domain_name}-last_dial/global/${uuid})
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->snom-demo-2] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [snom-demo-2] destination_number(77) =~ /^9001$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->snom-demo-1] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [snom-demo-1] destination_number(77) =~ /^9000$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->eavesdrop] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [eavesdrop] destination_number(77) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->eavesdrop] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [eavesdrop] destination_number(77) =~ /^779$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->call_return] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call_return] destination_number(77) =~ /^\*69$|^869$|^lcr$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->del-group] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [del-group] destination_number(77) =~ /^80(\d{2})$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->add-group] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [add-group] destination_number(77) =~ /^81(\d{2})$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->call-group-simo] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call-group-simo] destination_number(77) =~ /^82(\d{2})$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->call-group-order] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call-group-order] destination_number(77) =~ /^83(\d{2})$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->extension-intercom] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [extension-intercom] destination_number(77) =~ /^8(10[01][0-9])$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->Local_Extension] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [Local_Extension] destination_number(77) =~ /^(10[01][0-9])$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->Local_Extension_Skinny] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [Local_Extension_Skinny] destination_number(77) =~ /^(11[01][0-9])$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->group_dial_sales] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [group_dial_sales] destination_number(77) =~ /^2000$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->group_dial_support] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [group_dial_support] destination_number(77) =~ /^2001$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->group_dial_billing] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [group_dial_billing] destination_number(77) =~ /^2002$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->operator] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [operator] destination_number(77) =~ /^(operator|0)$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->vmain] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [vmain] destination_number(77) =~ /^vmain$|^4000$|^\*98$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->sip_uri] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [sip_uri] destination_number(77) =~ /^sip:(.*)$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->nb_conferences] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [nb_conferences] destination_number(77) =~ /^(30\d{2})$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->wb_conferences] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [wb_conferences] destination_number(77) =~ /^(31\d{2})$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->uwb_conferences] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [uwb_conferences] destination_number(77) =~ /^(32\d{2})$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->cdquality_conferences] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [cdquality_conferences] destination_number(77) =~ /^(33\d{2})$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->freeswitch_public_conf_via_sip] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(77) =~ /^9(888|8888|1616|3232)$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->mad_boss_intercom] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [mad_boss_intercom] destination_number(77) =~ /^0911$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->mad_boss_intercom] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [mad_boss_intercom] destination_number(77) =~ /^0912$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->mad_boss] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [mad_boss] destination_number(77) =~ /^0913$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ivr_demo] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ivr_demo] destination_number(77) =~ /^5000$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->dynamic_conference] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [dynamic_conference] destination_number(77) =~ /^5001$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->rtp_multicast_page] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [rtp_multicast_page] destination_number(77) =~ /^pagegroup$|^7243$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->park] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [park] destination_number(77) =~ /^5900$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->unpark] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unpark] destination_number(77) =~ /^5901$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->valet_park] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [valet_park] destination_number(77) =~ /^(6000)$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->valet_park] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [valet_park] destination_number(77) =~ /^(60\d[1-9])$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->park] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [park] destination_number(77) =~ /park\+(\d+)/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->unpark] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unpark] destination_number(77) =~ /^parking$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->park] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [park] destination_number(77) =~ /callpark/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->unpark] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unpark] destination_number(77) =~ /pickup/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->wait] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [wait] destination_number(77) =~ /^wait$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->fax_receive] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [fax_receive] destination_number(77) =~ /^9178$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->fax_transmit] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [fax_transmit] destination_number(77) =~ /^9179$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_180] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_180] destination_number(77) =~ /^9180$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_183_uk_ring] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_183_uk_ring] destination_number(77) =~ /^9181$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_183_music_ring] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_183_music_ring] destination_number(77) =~ /^9182$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_post_answer_uk_ring] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(77) =~ /^9183$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_post_answer_music] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_post_answer_music] destination_number(77) =~ /^9184$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ClueCon] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ClueCon] destination_number(77) =~ /^9191$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->show_info] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [show_info] destination_number(77) =~ /^9192$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->video_record] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [video_record] destination_number(77) =~ /^9193$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->video_playback] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [video_playback] destination_number(77) =~ /^9194$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->delay_echo] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [delay_echo] destination_number(77) =~ /^9195$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->echo] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [echo] destination_number(77) =~ /^9196$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->milliwatt] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [milliwatt] destination_number(77) =~ /^9197$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->tone_stream] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [tone_stream] destination_number(77) =~ /^9198$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->zrtp_enrollement] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [zrtp_enrollement] destination_number(77) =~ /^9787$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->hold_music] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [hold_music] destination_number(77) =~ /^9664$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->Recordings] continue=false
>> 2011-09-11 01:17:17.388551 [ERR] switch_regex.c:101 COMPILE ERROR: 1 [nothing to repeat][^*(732)$]
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [Recordings] destination_number(77) =~ /^*(732)$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->enum] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [enum] ${module_exists(mod_enum)}(true) =~ /true/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [enum] destination_number(77) =~ /^(.*)$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action transfer(77 enum)
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:128 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_ROUTING -> CS_EXECUTE
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING going to sleep
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_EXECUTE
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE
>> 2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:241 sofia/external/0476058096 at voip.belgacom.be SOFIA EXECUTE
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:166 sofia/external/0476058096 at voip.belgacom.be Standard EXECUTE
>> EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.100-spymap/0476058096/068a829e-dc03-11e0-81f7-f9f410ff919e)
>> EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.100-last_dial/0476058096/77)
>> EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.100-last_dial/global/068a829e-dc03-11e0-81f7-f9f410ff919e)
>> EXECUTE sofia/external/0476058096 at voip.belgacom.be set(RFC2822_DATE=Sun, 11 Sep 2011 01:17:17 +0200)
>> 2011-09-11 01:17:17.388551 [DEBUG] mod_dptools.c:1167 sofia/external/0476058096 at voip.belgacom.be SET [RFC2822_DATE]=[Sun, 11 Sep 2011 01:17:17 +0200]
>> EXECUTE sofia/external/0476058096 at voip.belgacom.be transfer(77 enum)
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_ivr.c:1693 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_EXECUTE -> CS_ROUTING
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:724 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 2011-09-11 01:17:17.388551 [NOTICE] switch_ivr.c:1699 Transfer sofia/external/0476058096 at voip.belgacom.be to enum[77 at default]
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE going to sleep
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_ROUTING
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING
>> 2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:148 sofia/external/0476058096 at voip.belgacom.be SOFIA ROUTING
>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:78 sofia/external/0476058096 at voip.belgacom.be Standard ROUTING
>> 2011-09-11 01:17:17.388551 [DEBUG] mod_enum.c:541 ENUM Lookup on 77
>> 2011-09-11 01:17:27.188065 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 2011-09-11 01:17:27.188065 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 2011-09-11 01:17:27.188065 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 
>> 
>> LOG  INTERNAL -> INTERNAL
>> 
>> 
>> 2011-09-11 01:22:17.268071 [DEBUG] sofia.c:7057 IP 192.168.100.6 Approved by acl "lan[]". Access Granted.
>> 2011-09-11 01:22:17.268071 [NOTICE] switch_channel.c:908 New Channel sofia/internal/88 at 192.168.1.100 [b94b66fa-dc03-11e0-81f9-f9f410ff919e]
>> 2011-09-11 01:22:17.268071 [DEBUG] sofia.c:5156 Channel sofia/internal/88 at 192.168.1.100 entering state [received][100]
>> 2011-09-11 01:22:17.268071 [DEBUG] sofia.c:5167 Remote SDP:
>> v=0
>> o=- 1315696937270159 1 IN IP4 192.168.100.6
>> s=CounterPath X-Lite 4.1
>> c=IN IP4 192.168.100.6
>> t=0 0
>> a=ice-ufrag:de6e17
>> a=ice-pwd:9e8391bc4955d13bbf67e8eeb2118613
>> m=audio 63158 RTP/AVP 0 8 101
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=candidate:1 1 UDP 659136 192.168.100.6 63158 typ host
>> a=candidate:1 2 UDP 659134 192.168.100.6 63159 typ host
>> 
>> 2011-09-11 01:22:17.268071 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_NEW
>> 2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMA:8:8000:20:64000]
>> 2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMU:0:8000:20:64000]/[G729:18:8000:20:8000]
>> 2011-09-11 01:22:17.268071 [DEBUG] switch_core_state_machine.c:354 (sofia/internal/88 at 192.168.1.100) State NEW
>> 2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
>> 2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:2857 Set Codec sofia/internal/88 at 192.168.1.100 PCMA/8000 20 ms 160 samples 64000 bits
>> 2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:4859 Set 2833 dtmf send/recv payload to 101
>> 2011-09-11 01:22:17.268071 [DEBUG] sofia.c:5357 (sofia/internal/88 at 192.168.1.100) State Change CS_NEW -> CS_INIT
>> 2011-09-11 01:22:17.268071 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_INIT
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:375 (sofia/internal/88 at 192.168.1.100) State INIT
>> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:85 sofia/internal/88 at 192.168.1.100 SOFIA INIT
>> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:125 (sofia/internal/88 at 192.168.1.100) State Change CS_INIT -> CS_ROUTING
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:375 (sofia/internal/88 at 192.168.1.100) State INIT going to sleep
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_ROUTING
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_channel.c:1837 (sofia/internal/88 at 192.168.1.100) Callstate Change DOWN -> RINGING
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:384 (sofia/internal/88 at 192.168.1.100) State ROUTING
>> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:148 sofia/internal/88 at 192.168.1.100 SOFIA ROUTING
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:78 sofia/internal/88 at 192.168.1.100 Standard ROUTING
>> 2011-09-11 01:22:17.288082 [INFO] mod_dialplan_xml.c:336 Processing 88 <88>->77 in context public
>> Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->unloop] continue=false
>> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->outside_call] continue=true
>> Dialplan: sofia/internal/88 at 192.168.1.100 Absolute Condition [outside_call]
>> Dialplan: sofia/internal/88 at 192.168.1.100 Action set(outside_call=true)
>> Dialplan: sofia/internal/88 at 192.168.1.100 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>> Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->call_debug] continue=true
>> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
>> Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->public_extensions] continue=false
>> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (FAIL) [public_extensions] destination_number(77) =~ /^(10[01][0-9])$/ break=on-false
>> Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->015261786] continue=false
>> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (PASS) [015261786] context(public) =~ /public/ break=on-false
>> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (FAIL) [015261786] destination_number(77) =~ /^015261786$/ break=on-false
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:128 (sofia/internal/88 at 192.168.1.100) State Change CS_ROUTING -> CS_EXECUTE
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:384 (sofia/internal/88 at 192.168.1.100) State ROUTING going to sleep
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_EXECUTE
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:391 (sofia/internal/88 at 192.168.1.100) State EXECUTE
>> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:241 sofia/internal/88 at 192.168.1.100 SOFIA EXECUTE
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:166 sofia/internal/88 at 192.168.1.100 Standard EXECUTE
>> EXECUTE sofia/internal/88 at 192.168.1.100 set(outside_call=true)
>> 2011-09-11 01:22:17.288082 [DEBUG] mod_dptools.c:1167 sofia/internal/88 at 192.168.1.100 SET [outside_call]=[true]
>> EXECUTE sofia/internal/88 at 192.168.1.100 set(RFC2822_DATE=Sun, 11 Sep 2011 01:22:17 +0200)
>> 2011-09-11 01:22:17.288082 [DEBUG] mod_dptools.c:1167 sofia/internal/88 at 192.168.1.100 SET [RFC2822_DATE]=[Sun, 11 Sep 2011 01:22:17 +0200]
>> 2011-09-11 01:22:17.288082 [NOTICE] switch_core_state_machine.c:200 sofia/internal/88 at 192.168.1.100 has executed the last dialplan instruction, hanging up.
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_channel.c:2797 (sofia/internal/88 at 192.168.1.100) Callstate Change RINGING -> HANGUP
>> 2011-09-11 01:22:17.288082 [NOTICE] switch_core_state_machine.c:202 Hangup sofia/internal/88 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_channel.c:2813 Send signal sofia/internal/88 at 192.168.1.100 [KILL]
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:391 (sofia/internal/88 at 192.168.1.100) State EXECUTE going to sleep
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_HANGUP
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:576 (sofia/internal/88 at 192.168.1.100) State HANGUP
>> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:458 Channel sofia/internal/88 at 192.168.1.100 hanging up, cause: NORMAL_CLEARING
>> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:522 Responding to INVITE with: 480
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:46 sofia/internal/88 at 192.168.1.100 Standard HANGUP, cause: NORMAL_CLEARING
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:576 (sofia/internal/88 at 192.168.1.100) State HANGUP going to sleep
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:367 (sofia/internal/88 at 192.168.1.100) State Change CS_HANGUP -> CS_REPORTING
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_REPORTING
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/88 at 192.168.1.100) State REPORTING
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:53 sofia/internal/88 at 192.168.1.100 Standard REPORTING, cause: NORMAL_CLEARING
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/88 at 192.168.1.100) State REPORTING going to sleep
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/88 at 192.168.1.100) State Change CS_REPORTING -> CS_DESTROY
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1349 Session 6 (sofia/internal/88 at 192.168.1.100) Locked, Waiting on external entities
>> 2011-09-11 01:22:17.288082 [NOTICE] switch_core_session.c:1367 Session 6 (sofia/internal/88 at 192.168.1.100) Ended
>> 2011-09-11 01:22:17.288082 [NOTICE] switch_core_session.c:1369 Close Channel sofia/internal/88 at 192.168.1.100 [CS_DESTROY]
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:465 (sofia/internal/88 at 192.168.1.100) Callstate Change HANGUP -> DOWN
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:468 (sofia/internal/88 at 192.168.1.100) Running State Change CS_DESTROY
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/88 at 192.168.1.100) State DESTROY
>> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:363 sofia/internal/88 at 192.168.1.100 SOFIA DESTROY
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:60 sofia/internal/88 at 192.168.1.100 Standard DESTROY
>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/88 at 192.168.1.100) State DESTROY going to sleep
>> 
>> 
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From cvogel at lyonl.com  Sun Sep 11 05:28:24 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Sun, 11 Sep 2011 01:28:24 +0000
Subject: [Freeswitch-users] external sip profile
Message-ID: <D9A75749-55E5-4DC5-A401-5F887ABEDE1A@lyonl.com>

hello,

I'm trying to switch from asterisk to freeswitch; however i'm wondering how can I create a sip profile because the sip profile i created doesn't seem to function with level 3.

here is the sip profile i created that isn't working:

<include>
  <gateway name="Level3">
    <param name="username" value="USERNAME"/>
    <param name="password" value="PASSWORD"/>
    <param name="proxy" value="4.55.35.60:5070"/>
    <param name="expire-seconds" value="3600"/>
    <param name="register-transport" value="udp"/>
    <param name="register" value="true"/>
  </gateway>
</include>

here is my asterisk profile (it works):

[level3_out]
type=peer
nat=no
host=4.55.35.60
username=***Username***
secret=***Password***
dtmfmode=rfc2833
port=5070

[level3_in]
nat=no
insecure=very
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw&alaw
host=4.55.35.60
type=peer
port=5070

How can I create a sip profile that will function the same in freeswitch?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110911/01d573ab/attachment.html 

From peter.olsson at visionutveckling.se  Sun Sep 11 14:36:50 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 11 Sep 2011 12:36:50 +0200
Subject: [Freeswitch-users] external sip profile
In-Reply-To: <D9A75749-55E5-4DC5-A401-5F887ABEDE1A@lyonl.com>
References: <D9A75749-55E5-4DC5-A401-5F887ABEDE1A@lyonl.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper>

You're not giving us much information here. Please post exactly what doesn't work, and also pastebin the actual logs from FreeSWITCH.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Chad Vogel [cvogel at lyonl.com]
Skickat: den 11 september 2011 03:28
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] external sip profile

hello,

I'm trying to switch from asterisk to freeswitch; however i'm wondering how can I create a sip profile because the sip profile i created doesn't seem to function with level 3.

here is the sip profile i created that isn't working:

<include>
  <gateway name="Level3">
    <param name="username" value="USERNAME"/>
    <param name="password" value="PASSWORD"/>
    <param name="proxy" value="4.55.35.60:5070"/>
    <param name="expire-seconds" value="3600"/>
    <param name="register-transport" value="udp"/>
    <param name="register" value="true"/>
  </gateway>
</include>

here is my asterisk profile (it works):

[level3_out]
type=peer
nat=no
host=4.55.35.60
username=***Username***
secret=***Password***
dtmfmode=rfc2833
port=5070

[level3_in]
nat=no
insecure=very
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw&alaw
host=4.55.35.60
type=peer
port=5070

How can I create a sip profile that will function the same in freeswitch?

!DSPAM:4e6c82af32761635315745!


From avi at avimarcus.net  Sun Sep 11 15:18:55 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 11 Sep 2011 14:18:55 +0300
Subject: [Freeswitch-users] mod_lcr transfer shows wrong caller id
In-Reply-To: <FB36DABAE032438AA68C8DC2C1C7877F@omni1.local>
References: <0C64150CB61444419B7A1981CABCF183@omni1.local>
	<CALNZuw=UOcJ810Qk5BnN+vrDk6jX1N0gjxdJi8-Ry7nWO9htAg@mail.gmail.com>
	<FB36DABAE032438AA68C8DC2C1C7877F@omni1.local>
Message-ID: <CALNZuwkNQau9YCb_7aQoEuASiv=25_ui77PmrHESLtiLBFTS8Q@mail.gmail.com>

"display number" - the effective_caller_id_name ?
If you still have an issue, can you show/pastebin a SIP trace?

-Avi


On Sun, Sep 11, 2011 at 9:56 AM, Anestis Mavro <devel at omninet.eu> wrote:

> **
>
> Yes, the same result.****
>
> ** **
>
> I made now a trace and found that in the ?From? and ?P-Asserted-Identity?
> the display name is correct (the number) but the display number is wrong
> (the account).****
>
> ** **
>
> Wondering now where to set them to correctly apply?****
>
> ** **
>
> Thanks****
>
> Anestis****
>
> ** **
>
> ** **
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus
> *Sent:* Sunday, September 11, 2011 8:47 AM
> *To:* **FreeSWITCH Users Help**
> *Subject:* Re: [Freeswitch-users] mod_lcr transfer shows wrong caller id**
> **
>
> ** **
>
> Did you set effective_caller_id in the A-leg to the value you want?
> ****
>
> -Avi****
>
> ** **
>
> On Sun, Sep 11, 2011 at 8:43 AM, Anestis Mavro <devel at omninet.eu> wrote:**
> **
>
> Hi,****
>
>  ****
>
> I?ve been using mod_lcr with nibblebill for a while and everything worked
> fine with the bridge application. ****
>
> Now I would like to add limit and I have to use it as dialplan like this:*
> ***
>
>  ****
>
> <action application=?transfer? data=?${callednumber} lcr ${lcrprofile}? />
> ****
>
>  ****
>
> I managed to configure the lcr.conf.xml correctly for Limit and everything
> seems to be working beside the caller id.****
>
> Now the variable origination_caller_id_number gets the accountcode instead
> of the correct number.****
>
>  ****
>
> I?ve tried to explicitly set the variable before transferring and even in
> the lcr.conf.xml and export it? no chance?****
>
>  ****
>
> Any idea??****
>
>  ****
>
> Thanks****
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 5054 (20100423) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.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****
>
> ** **
>
> ** **
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 5054 (20100423) __________****
>
> ** **
>
> The message was checked by ESET NOD32 Antivirus.****
>
> ** **
>
> http://www.eset.com****
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 5054 (20100423) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110911/13f95259/attachment-0001.html 

From devel at omninet.eu  Sun Sep 11 16:07:39 2011
From: devel at omninet.eu (Anestis Mavro)
Date: Sun, 11 Sep 2011 15:07:39 +0300
Subject: [Freeswitch-users] mod_lcr transfer shows wrong caller id
In-Reply-To: <CALNZuwkNQau9YCb_7aQoEuASiv=25_ui77PmrHESLtiLBFTS8Q@mail.gmail.com>
References: <0C64150CB61444419B7A1981CABCF183@omni1.local><CALNZuw=UOcJ810Qk5BnN+vrDk6jX1N0gjxdJi8-Ry7nWO9htAg@mail.gmail.com><FB36DABAE032438AA68C8DC2C1C7877F@omni1.local>
	<CALNZuwkNQau9YCb_7aQoEuASiv=25_ui77PmrHESLtiLBFTS8Q@mail.gmail.com>
Message-ID: <72177AFB5FF3491191FBDD584768105A@omni1.local>

 

Here the parts of the correct invite (with bridge): 

From: "1234567890" <sip:1234567890 at 11.22.33.44>

P-Asserted-Identity: "1234567890" <sip: 1234567890 at 11.22.33.44>

 

And here the wrong (with transfer):

From: "1234567890" <sip:1000 at 11.22.33.44>          

P-Asserted-Identity: "1234567890" <sip:1000 at 11.22.33.44>         

 

(where "1234567890" the caller id

1000 the calling account/user

11.22.33.44 the server ip)

 

It seems that LCR overrides the origination_caller_id_number with the
accountcode when it is run as dialplan. 

I have tried all variables like origination., effective. and it was
impossible to change the P-Asserted-Identity. (The P-A-I is added by LCR
automatically with sip_cid_type=pid)

If I try to set it manually in the extension, then I get two P-A-Is and I
have to disable it in the LCR; It much easier to be set by the LCR
dynamically for each user within the query

 

Thanks

Anestis

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Sunday, September 11, 2011 2:19 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_lcr transfer shows wrong caller id

 

"display number" - the effective_caller_id_name ?

If you still have an issue, can you show/pastebin a SIP trace?




-Avi

 

On Sun, Sep 11, 2011 at 9:56 AM, Anestis Mavro <devel at omninet.eu> wrote:

Yes, the same result.

 

I made now a trace and found that in the "From" and "P-Asserted-Identity"
the display name is correct (the number) but the display number is wrong
(the account).

 

Wondering now where to set them to correctly apply.

 

Thanks

Anestis

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Sunday, September 11, 2011 8:47 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_lcr transfer shows wrong caller id

 

Did you set effective_caller_id in the A-leg to the value you want?


-Avi

 

On Sun, Sep 11, 2011 at 8:43 AM, Anestis Mavro <devel at omninet.eu> wrote:

Hi,

 

I've been using mod_lcr with nibblebill for a while and everything worked
fine with the bridge application. 

Now I would like to add limit and I have to use it as dialplan like this:

 

<action application="transfer" data="${callednumber} lcr ${lcrprofile}" />

 

I managed to configure the lcr.conf.xml correctly for Limit and everything
seems to be working beside the caller id.

Now the variable origination_caller_id_number gets the accountcode instead
of the correct number.

 

I've tried to explicitly set the variable before transferring and even in
the lcr.conf.xml and export it. no chance.

 

Any idea??

 

Thanks

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110911/072aeff4/attachment.html 

From avi at avimarcus.net  Sun Sep 11 16:22:48 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 11 Sep 2011 15:22:48 +0300
Subject: [Freeswitch-users] Cut PCAP file?
Message-ID: <CALNZuw==7kXb7zZgJYDjb8ey+47J0jK9dVHLTMX2Q7w2H+8ROw@mail.gmail.com>

I've got a pcap file in wireshark that I want to show to someone... but I
only need the first 1285 frames, not all 103k. How do I cut it..?
Thanks,
-Avi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110911/883ec606/attachment.html 

From wasim at convergence.pk  Sun Sep 11 16:52:40 2011
From: wasim at convergence.pk (Wasim Baig)
Date: Sun, 11 Sep 2011 17:52:40 +0500
Subject: [Freeswitch-users] Cut PCAP file?
In-Reply-To: <CALNZuw==7kXb7zZgJYDjb8ey+47J0jK9dVHLTMX2Q7w2H+8ROw@mail.gmail.com>
References: <CALNZuw==7kXb7zZgJYDjb8ey+47J0jK9dVHLTMX2Q7w2H+8ROw@mail.gmail.com>
Message-ID: <CABHdwCr9+kieEKOfoG3akFAMV5+ZJfzWTBS5V71PgvODMhkO7w@mail.gmail.com>

http://www.wireshark.org/docs/man-pages/editcap.html

-wasim

On Sun, Sep 11, 2011 at 17:22, Avi Marcus <avi at avimarcus.net> wrote:

> I've got a pcap file in wireshark that I want to show to someone... but I
> only need the first 1285 frames, not all 103k. How do I cut it..?
> Thanks,
> -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
>
>


-- 
wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
peace be upon you ...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110911/8dd171c2/attachment-0001.html 

From curriegrad2004 at gmail.com  Sun Sep 11 19:15:44 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 11 Sep 2011 08:15:44 -0700
Subject: [Freeswitch-users] Help needed on basic dialing plan: no calls
 are routed
In-Reply-To: <CF4DF76C-749B-483C-B814-B8B2C41C3455@aharm.de>
References: <DB197D09-AC19-49F5-86AF-7951F94F690B@aharm.de>
	<CAMWiMy95Ud-vY358XLDhKhMP=BoZ68SJ9nGgk3Bu57NSLveOWg@mail.gmail.com>
	<CF4DF76C-749B-483C-B814-B8B2C41C3455@aharm.de>
Message-ID: <CAMWiMy9+qPNfz7U4EimYOdn-nHNXpOjKMGPds39XtceJJEmdFw@mail.gmail.com>

Remove the <condition field="context" expression="public"/> from the
inbound dialplan. That shouldn't be in there in the first place.

On Sun, Sep 11, 2011 at 12:13 AM, Alexander Harm <contact at aharm.de> wrote:
> hi there,
>
> thanks for the reply. have to admit that i'm not completely sure that i understand you correctly.
>
> my problem:
>
> - i cannot call from one extension to another extension (not sure where the outbound route comes in here, nor the sip trunk)
>
> - my calls from external to an extension are not transferred (sip trunk is registered and the call arrives at the gateway), caller receives no ringtone and internal extension doesn't ring either
>
> to follow your advice i did setup an outbound rule (all number starting with 0 are routed through the gateway). it makes no difference at all, the calls from internal extensions fail.
>
> hope you have some other ideas.
>
> alexander
>
> extension 77
>
> <include>
> ?<user id="77">
> ? <params>
> ? ? <param name="password" value="xxx"/>
> ? ? <param name="vm-password" value="user-choose"/>
> ? ? <param name="vm-enabled" value="false"/>
> ? </params>
> ? <variables>
> ? ? <variable name="toll_allow" value="international"/>
> ? ? <variable name="accountcode" value="77"/>
> ? ? <variable name="user_context" value="default"/>
> ? ? <variable name="effective_caller_id_name" value="77"/>
> ? ? <variable name="effective_caller_id_number" value="77"/>
> ? </variables>
> ?</user>
> </include>
>
> extension 88
>
> <include>
> ?<user id="88">
> ? <params>
> ? ? <param name="password" value="xxx"/>
> ? ? <param name="vm-password" value="user-choose"/>
> ? ? <param name="vm-enabled" value="false"/>
> ? </params>
> ? <variables>
> ? ? <variable name="toll_allow" value="international"/>
> ? ? <variable name="accountcode" value="88"/>
> ? ? <variable name="user_context" value="default"/>
> ? ? <variable name="effective_caller_id_name" value="88"/>
> ? ? <variable name="effective_caller_id_number" value="88"/>
> ? </variables>
> ?</user>
> </include>
>
> my trunk
>
> <include>
> ? <gateway name="015261786">
> ? ? <param name="username" value="u15261786"/>
> ? ? <param name="auth-username" value="u15261786"/>
> ? ? <param name="password" value="xxx"/>
> ? ? <param name="realm" value="voip.belgacom.be"/>
> ? ? <param name="from-user" value="015261786"/>
> ? ? <param name="from-domain" value="voip.belgacom.be"/>
> ? ? <param name="proxy" value="voip.belgacom.be"/>
> ? ? <param name="outbound-proxy" value="81.240.251.22"/>
> ? ? <param name="expire-seconds" value="1800"/>
> ? ? <param name="register" value="true"/>
> ? ? <param name="retry-seconds" value="60"/>
> ? ? <param name="extension" value="015261786"/>
> ? ? <param name="context" value="public"/>
> ? </gateway>
> </include>
>
> my inbound route
>
> <extension name="015261786" >
> ?<condition field="context" expression="public"/>
> ?<condition field="destination_number" expression="^015261786$">
> ? ? ?<action application="transfer" data="77 XML default"/>
> ?</condition>
> </extension>
>
> my outbound route
>
> <extension name="015261786.0d" >
> ?<condition field="destination_number" expression="^0(\d+)$" >
> ? ? ?<action application="set" data="sip_h_X-accountcode=${accountcode}" />
> ? ? ?<action application="set" data="call_direction=outbound" />
> ? ? ?<action application="set" data="hangup_after_bridge=true" />
> ? ? ?<action application="set" data="effective_caller_id_name=${outbound_caller_id_name}" />
> ? ? ?<action application="set" data="effective_caller_id_number=${outbound_caller_id_number}" />
> ? ? ?<action application="set" data="inherit_codec=true" />
> ? ? ?<action application="bridge" data="sofia/gateway/015261786/$1" />
> ?</condition>
> </extension>
>
>
>
> On 11.09.2011, at 06:15, curriegrad2004 wrote:
>
>> You haven't defined a dialplan for external calls yet. The Regex
>> didn't even match the number you're trying to dial out. Get a SIP
>> trunk or a TDM/J1 trunk first, register or make it work with
>> FreeSWITCH and then create a dialplan to call out.
>>
>> This might be a good example to start out with:
>>
>> http://wiki.freeswitch.org/wiki/Dialplan_XML#Dialing_through_gateways
>>
>> On Sat, Sep 10, 2011 at 4:40 PM, Alexander Harm <contact at aharm.de> wrote:
>>> hello all,
>>>
>>> i have a freeswitch installation running and 2 phones connected to it. i setup 1 inbound rule. when i dial the external number i receive the following messages in the log. to me it seems that the routing works correctly but the ?transfer to extension 77 fails.
>>>
>>> note: further down the error messages i receive when trying to call internally.
>>>
>>> any help appreciated.
>>>
>>> regards, alexander
>>>
>>> LOG EXTERNAL -> INTERNAL
>>>
>>> 2011-09-11 01:17:17.368068 [NOTICE] switch_channel.c:908 New Channel sofia/external/0476058096 at voip.belgacom.be [068a829e-dc03-11e0-81f7-f9f410ff919e]
>>> 2011-09-11 01:17:17.368068 [DEBUG] sofia.c:5156 Channel sofia/external/0476058096 at voip.belgacom.be entering state [received][100]
>>> 2011-09-11 01:17:17.368068 [DEBUG] sofia.c:5167 Remote SDP:
>>> v=0
>>> o=- 0 98592702 IN IP4 81.240.251.38
>>> s=IMSS
>>> c=IN IP4 81.240.251.38
>>> t=0 0
>>> m=audio 10874 RTP/AVP 8 18 0 101
>>> a=rtpmap:101 telephone-event/8000/1
>>> a=fmtp:101 0-15
>>>
>>> 2011-09-11 01:17:17.368068 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
>>> 2011-09-11 01:17:17.368068 [DEBUG] sofia_glue.c:2857 Set Codec sofia/external/0476058096 at voip.belgacom.be PCMA/8000 20 ms 160 samples 64000 bits
>>> 2011-09-11 01:17:17.368068 [DEBUG] sofia_glue.c:4859 Set 2833 dtmf send/recv payload to 101
>>> 2011-09-11 01:17:17.368068 [DEBUG] sofia.c:5357 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_NEW -> CS_INIT
>>> 2011-09-11 01:17:17.368068 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>> 2011-09-11 01:17:17.368068 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_INIT
>>> 2011-09-11 01:17:17.368068 [DEBUG] switch_core_state_machine.c:375 (sofia/external/0476058096 at voip.belgacom.be) State INIT
>>> 2011-09-11 01:17:17.368068 [DEBUG] mod_sofia.c:85 sofia/external/0476058096 at voip.belgacom.be SOFIA INIT
>>> 2011-09-11 01:17:17.368068 [DEBUG] mod_sofia.c:125 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_INIT -> CS_ROUTING
>>> 2011-09-11 01:17:17.368068 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>> 2011-09-11 01:17:17.368068 [DEBUG] switch_core_state_machine.c:375 (sofia/external/0476058096 at voip.belgacom.be) State INIT going to sleep
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_ROUTING
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_channel.c:1837 (sofia/external/0476058096 at voip.belgacom.be) Callstate Change DOWN -> RINGING
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING
>>> 2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:148 sofia/external/0476058096 at voip.belgacom.be SOFIA ROUTING
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:78 sofia/external/0476058096 at voip.belgacom.be Standard ROUTING
>>> 2011-09-11 01:17:17.388551 [INFO] mod_dialplan_xml.c:336 Processing 0476058096 <0476058096>->015261786 in context public
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->unloop] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->outside_call] continue=true
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Absolute Condition [outside_call]
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action set(outside_call=true)
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->call_debug] continue=true
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->public_extensions] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [public_extensions] destination_number(015261786) =~ /^(10[01][0-9])$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->015261786] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [015261786] context(public) =~ /public/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [015261786] destination_number(015261786) =~ /^015261786$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action transfer(77 XML default)
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:128 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_ROUTING -> CS_EXECUTE
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING going to sleep
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_EXECUTE
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE
>>> 2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:241 sofia/external/0476058096 at voip.belgacom.be SOFIA EXECUTE
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:166 sofia/external/0476058096 at voip.belgacom.be Standard EXECUTE
>>> EXECUTE sofia/external/0476058096 at voip.belgacom.be set(outside_call=true)
>>> 2011-09-11 01:17:17.388551 [DEBUG] mod_dptools.c:1167 sofia/external/0476058096 at voip.belgacom.be SET [outside_call]=[true]
>>> EXECUTE sofia/external/0476058096 at voip.belgacom.be set(RFC2822_DATE=Sun, 11 Sep 2011 01:17:17 +0200)
>>> 2011-09-11 01:17:17.388551 [DEBUG] mod_dptools.c:1167 sofia/external/0476058096 at voip.belgacom.be SET [RFC2822_DATE]=[Sun, 11 Sep 2011 01:17:17 +0200]
>>> EXECUTE sofia/external/0476058096 at voip.belgacom.be transfer(77 XML default)
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_ivr.c:1693 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_EXECUTE -> CS_ROUTING
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:724 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>> 2011-09-11 01:17:17.388551 [NOTICE] switch_ivr.c:1699 Transfer sofia/external/0476058096 at voip.belgacom.be to XML[77 at default]
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE going to sleep
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_ROUTING
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING
>>> 2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:148 sofia/external/0476058096 at voip.belgacom.be SOFIA ROUTING
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:78 sofia/external/0476058096 at voip.belgacom.be Standard ROUTING
>>> 2011-09-11 01:17:17.388551 [INFO] mod_dialplan_xml.c:336 Processing 0476058096 <0476058096>->77 in context default
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->unloop] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->tod_example] continue=true
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Date/Time Match (FAIL) [tod_example] break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->holiday_example] continue=true
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Date/Time Match (FAIL) [holiday_example] break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->global-intercept] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [global-intercept] destination_number(77) =~ /^886$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->group-intercept] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [group-intercept] destination_number(77) =~ /^\*8$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->intercept-ext] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [intercept-ext] destination_number(77) =~ /^\*\*(\d+)$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->redial] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [redial] destination_number(77) =~ /^(redial|870)$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->global] continue=true
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Absolute Condition [global]
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action hash(insert/${domain_name}-last_dial/global/${uuid})
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->snom-demo-2] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [snom-demo-2] destination_number(77) =~ /^9001$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->snom-demo-1] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [snom-demo-1] destination_number(77) =~ /^9000$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->eavesdrop] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [eavesdrop] destination_number(77) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->eavesdrop] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [eavesdrop] destination_number(77) =~ /^779$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->call_return] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call_return] destination_number(77) =~ /^\*69$|^869$|^lcr$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->del-group] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [del-group] destination_number(77) =~ /^80(\d{2})$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->add-group] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [add-group] destination_number(77) =~ /^81(\d{2})$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->call-group-simo] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call-group-simo] destination_number(77) =~ /^82(\d{2})$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->call-group-order] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call-group-order] destination_number(77) =~ /^83(\d{2})$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->extension-intercom] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [extension-intercom] destination_number(77) =~ /^8(10[01][0-9])$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->Local_Extension] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [Local_Extension] destination_number(77) =~ /^(10[01][0-9])$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->Local_Extension_Skinny] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [Local_Extension_Skinny] destination_number(77) =~ /^(11[01][0-9])$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->group_dial_sales] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [group_dial_sales] destination_number(77) =~ /^2000$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->group_dial_support] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [group_dial_support] destination_number(77) =~ /^2001$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->group_dial_billing] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [group_dial_billing] destination_number(77) =~ /^2002$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->operator] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [operator] destination_number(77) =~ /^(operator|0)$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->vmain] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [vmain] destination_number(77) =~ /^vmain$|^4000$|^\*98$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->sip_uri] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [sip_uri] destination_number(77) =~ /^sip:(.*)$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->nb_conferences] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [nb_conferences] destination_number(77) =~ /^(30\d{2})$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->wb_conferences] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [wb_conferences] destination_number(77) =~ /^(31\d{2})$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->uwb_conferences] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [uwb_conferences] destination_number(77) =~ /^(32\d{2})$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->cdquality_conferences] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [cdquality_conferences] destination_number(77) =~ /^(33\d{2})$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->freeswitch_public_conf_via_sip] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(77) =~ /^9(888|8888|1616|3232)$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->mad_boss_intercom] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [mad_boss_intercom] destination_number(77) =~ /^0911$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->mad_boss_intercom] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [mad_boss_intercom] destination_number(77) =~ /^0912$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->mad_boss] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [mad_boss] destination_number(77) =~ /^0913$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ivr_demo] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ivr_demo] destination_number(77) =~ /^5000$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->dynamic_conference] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [dynamic_conference] destination_number(77) =~ /^5001$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->rtp_multicast_page] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [rtp_multicast_page] destination_number(77) =~ /^pagegroup$|^7243$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->park] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [park] destination_number(77) =~ /^5900$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->unpark] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unpark] destination_number(77) =~ /^5901$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->valet_park] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [valet_park] destination_number(77) =~ /^(6000)$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->valet_park] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [valet_park] destination_number(77) =~ /^(60\d[1-9])$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->park] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [park] destination_number(77) =~ /park\+(\d+)/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->unpark] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unpark] destination_number(77) =~ /^parking$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->park] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [park] destination_number(77) =~ /callpark/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->unpark] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unpark] destination_number(77) =~ /pickup/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->wait] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [wait] destination_number(77) =~ /^wait$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->fax_receive] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [fax_receive] destination_number(77) =~ /^9178$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->fax_transmit] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [fax_transmit] destination_number(77) =~ /^9179$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_180] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_180] destination_number(77) =~ /^9180$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_183_uk_ring] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_183_uk_ring] destination_number(77) =~ /^9181$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_183_music_ring] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_183_music_ring] destination_number(77) =~ /^9182$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_post_answer_uk_ring] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(77) =~ /^9183$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ringback_post_answer_music] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ringback_post_answer_music] destination_number(77) =~ /^9184$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->ClueCon] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [ClueCon] destination_number(77) =~ /^9191$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->show_info] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [show_info] destination_number(77) =~ /^9192$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->video_record] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [video_record] destination_number(77) =~ /^9193$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->video_playback] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [video_playback] destination_number(77) =~ /^9194$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->delay_echo] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [delay_echo] destination_number(77) =~ /^9195$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->echo] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [echo] destination_number(77) =~ /^9196$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->milliwatt] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [milliwatt] destination_number(77) =~ /^9197$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->tone_stream] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [tone_stream] destination_number(77) =~ /^9198$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->zrtp_enrollement] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [zrtp_enrollement] destination_number(77) =~ /^9787$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->hold_music] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [hold_music] destination_number(77) =~ /^9664$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->Recordings] continue=false
>>> 2011-09-11 01:17:17.388551 [ERR] switch_regex.c:101 COMPILE ERROR: 1 [nothing to repeat][^*(732)$]
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [Recordings] destination_number(77) =~ /^*(732)$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->enum] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [enum] ${module_exists(mod_enum)}(true) =~ /true/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [enum] destination_number(77) =~ /^(.*)$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Action transfer(77 enum)
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:128 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_ROUTING -> CS_EXECUTE
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING going to sleep
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_EXECUTE
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE
>>> 2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:241 sofia/external/0476058096 at voip.belgacom.be SOFIA EXECUTE
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:166 sofia/external/0476058096 at voip.belgacom.be Standard EXECUTE
>>> EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.100-spymap/0476058096/068a829e-dc03-11e0-81f7-f9f410ff919e)
>>> EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.100-last_dial/0476058096/77)
>>> EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.100-last_dial/global/068a829e-dc03-11e0-81f7-f9f410ff919e)
>>> EXECUTE sofia/external/0476058096 at voip.belgacom.be set(RFC2822_DATE=Sun, 11 Sep 2011 01:17:17 +0200)
>>> 2011-09-11 01:17:17.388551 [DEBUG] mod_dptools.c:1167 sofia/external/0476058096 at voip.belgacom.be SET [RFC2822_DATE]=[Sun, 11 Sep 2011 01:17:17 +0200]
>>> EXECUTE sofia/external/0476058096 at voip.belgacom.be transfer(77 enum)
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_ivr.c:1693 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_EXECUTE -> CS_ROUTING
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_session.c:724 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>> 2011-09-11 01:17:17.388551 [NOTICE] switch_ivr.c:1699 Transfer sofia/external/0476058096 at voip.belgacom.be to enum[77 at default]
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE going to sleep
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_ROUTING
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING
>>> 2011-09-11 01:17:17.388551 [DEBUG] mod_sofia.c:148 sofia/external/0476058096 at voip.belgacom.be SOFIA ROUTING
>>> 2011-09-11 01:17:17.388551 [DEBUG] switch_core_state_machine.c:78 sofia/external/0476058096 at voip.belgacom.be Standard ROUTING
>>> 2011-09-11 01:17:17.388551 [DEBUG] mod_enum.c:541 ENUM Lookup on 77
>>> 2011-09-11 01:17:27.188065 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>> 2011-09-11 01:17:27.188065 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>> 2011-09-11 01:17:27.188065 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>>
>>>
>>> LOG ?INTERNAL -> INTERNAL
>>>
>>>
>>> 2011-09-11 01:22:17.268071 [DEBUG] sofia.c:7057 IP 192.168.100.6 Approved by acl "lan[]". Access Granted.
>>> 2011-09-11 01:22:17.268071 [NOTICE] switch_channel.c:908 New Channel sofia/internal/88 at 192.168.1.100 [b94b66fa-dc03-11e0-81f9-f9f410ff919e]
>>> 2011-09-11 01:22:17.268071 [DEBUG] sofia.c:5156 Channel sofia/internal/88 at 192.168.1.100 entering state [received][100]
>>> 2011-09-11 01:22:17.268071 [DEBUG] sofia.c:5167 Remote SDP:
>>> v=0
>>> o=- 1315696937270159 1 IN IP4 192.168.100.6
>>> s=CounterPath X-Lite 4.1
>>> c=IN IP4 192.168.100.6
>>> t=0 0
>>> a=ice-ufrag:de6e17
>>> a=ice-pwd:9e8391bc4955d13bbf67e8eeb2118613
>>> m=audio 63158 RTP/AVP 0 8 101
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=candidate:1 1 UDP 659136 192.168.100.6 63158 typ host
>>> a=candidate:1 2 UDP 659134 192.168.100.6 63159 typ host
>>>
>>> 2011-09-11 01:22:17.268071 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_NEW
>>> 2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMA:8:8000:20:64000]
>>> 2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMU:0:8000:20:64000]/[G729:18:8000:20:8000]
>>> 2011-09-11 01:22:17.268071 [DEBUG] switch_core_state_machine.c:354 (sofia/internal/88 at 192.168.1.100) State NEW
>>> 2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
>>> 2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:2857 Set Codec sofia/internal/88 at 192.168.1.100 PCMA/8000 20 ms 160 samples 64000 bits
>>> 2011-09-11 01:22:17.268071 [DEBUG] sofia_glue.c:4859 Set 2833 dtmf send/recv payload to 101
>>> 2011-09-11 01:22:17.268071 [DEBUG] sofia.c:5357 (sofia/internal/88 at 192.168.1.100) State Change CS_NEW -> CS_INIT
>>> 2011-09-11 01:22:17.268071 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_INIT
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:375 (sofia/internal/88 at 192.168.1.100) State INIT
>>> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:85 sofia/internal/88 at 192.168.1.100 SOFIA INIT
>>> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:125 (sofia/internal/88 at 192.168.1.100) State Change CS_INIT -> CS_ROUTING
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:375 (sofia/internal/88 at 192.168.1.100) State INIT going to sleep
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_ROUTING
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_channel.c:1837 (sofia/internal/88 at 192.168.1.100) Callstate Change DOWN -> RINGING
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:384 (sofia/internal/88 at 192.168.1.100) State ROUTING
>>> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:148 sofia/internal/88 at 192.168.1.100 SOFIA ROUTING
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:78 sofia/internal/88 at 192.168.1.100 Standard ROUTING
>>> 2011-09-11 01:22:17.288082 [INFO] mod_dialplan_xml.c:336 Processing 88 <88>->77 in context public
>>> Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->unloop] continue=false
>>> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
>>> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
>>> Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->outside_call] continue=true
>>> Dialplan: sofia/internal/88 at 192.168.1.100 Absolute Condition [outside_call]
>>> Dialplan: sofia/internal/88 at 192.168.1.100 Action set(outside_call=true)
>>> Dialplan: sofia/internal/88 at 192.168.1.100 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>>> Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->call_debug] continue=true
>>> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
>>> Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->public_extensions] continue=false
>>> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (FAIL) [public_extensions] destination_number(77) =~ /^(10[01][0-9])$/ break=on-false
>>> Dialplan: sofia/internal/88 at 192.168.1.100 parsing [public->015261786] continue=false
>>> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (PASS) [015261786] context(public) =~ /public/ break=on-false
>>> Dialplan: sofia/internal/88 at 192.168.1.100 Regex (FAIL) [015261786] destination_number(77) =~ /^015261786$/ break=on-false
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:128 (sofia/internal/88 at 192.168.1.100) State Change CS_ROUTING -> CS_EXECUTE
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:384 (sofia/internal/88 at 192.168.1.100) State ROUTING going to sleep
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_EXECUTE
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:391 (sofia/internal/88 at 192.168.1.100) State EXECUTE
>>> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:241 sofia/internal/88 at 192.168.1.100 SOFIA EXECUTE
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:166 sofia/internal/88 at 192.168.1.100 Standard EXECUTE
>>> EXECUTE sofia/internal/88 at 192.168.1.100 set(outside_call=true)
>>> 2011-09-11 01:22:17.288082 [DEBUG] mod_dptools.c:1167 sofia/internal/88 at 192.168.1.100 SET [outside_call]=[true]
>>> EXECUTE sofia/internal/88 at 192.168.1.100 set(RFC2822_DATE=Sun, 11 Sep 2011 01:22:17 +0200)
>>> 2011-09-11 01:22:17.288082 [DEBUG] mod_dptools.c:1167 sofia/internal/88 at 192.168.1.100 SET [RFC2822_DATE]=[Sun, 11 Sep 2011 01:22:17 +0200]
>>> 2011-09-11 01:22:17.288082 [NOTICE] switch_core_state_machine.c:200 sofia/internal/88 at 192.168.1.100 has executed the last dialplan instruction, hanging up.
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_channel.c:2797 (sofia/internal/88 at 192.168.1.100) Callstate Change RINGING -> HANGUP
>>> 2011-09-11 01:22:17.288082 [NOTICE] switch_core_state_machine.c:202 Hangup sofia/internal/88 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_channel.c:2813 Send signal sofia/internal/88 at 192.168.1.100 [KILL]
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:391 (sofia/internal/88 at 192.168.1.100) State EXECUTE going to sleep
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_HANGUP
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:576 (sofia/internal/88 at 192.168.1.100) State HANGUP
>>> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:458 Channel sofia/internal/88 at 192.168.1.100 hanging up, cause: NORMAL_CLEARING
>>> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:522 Responding to INVITE with: 480
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:46 sofia/internal/88 at 192.168.1.100 Standard HANGUP, cause: NORMAL_CLEARING
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:576 (sofia/internal/88 at 192.168.1.100) State HANGUP going to sleep
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:367 (sofia/internal/88 at 192.168.1.100) State Change CS_HANGUP -> CS_REPORTING
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:336 (sofia/internal/88 at 192.168.1.100) Running State Change CS_REPORTING
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/88 at 192.168.1.100) State REPORTING
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:53 sofia/internal/88 at 192.168.1.100 Standard REPORTING, cause: NORMAL_CLEARING
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/88 at 192.168.1.100) State REPORTING going to sleep
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/88 at 192.168.1.100) State Change CS_REPORTING -> CS_DESTROY
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/88 at 192.168.1.100 [BREAK]
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_session.c:1349 Session 6 (sofia/internal/88 at 192.168.1.100) Locked, Waiting on external entities
>>> 2011-09-11 01:22:17.288082 [NOTICE] switch_core_session.c:1367 Session 6 (sofia/internal/88 at 192.168.1.100) Ended
>>> 2011-09-11 01:22:17.288082 [NOTICE] switch_core_session.c:1369 Close Channel sofia/internal/88 at 192.168.1.100 [CS_DESTROY]
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:465 (sofia/internal/88 at 192.168.1.100) Callstate Change HANGUP -> DOWN
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:468 (sofia/internal/88 at 192.168.1.100) Running State Change CS_DESTROY
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/88 at 192.168.1.100) State DESTROY
>>> 2011-09-11 01:22:17.288082 [DEBUG] mod_sofia.c:363 sofia/internal/88 at 192.168.1.100 SOFIA DESTROY
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:60 sofia/internal/88 at 192.168.1.100 Standard DESTROY
>>> 2011-09-11 01:22:17.288082 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/88 at 192.168.1.100) State DESTROY going to sleep
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 avimarcus.net  Sun Sep 11 20:11:05 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 11 Sep 2011 19:11:05 +0300
Subject: [Freeswitch-users] Cut PCAP file?
In-Reply-To: <CABHdwCr9+kieEKOfoG3akFAMV5+ZJfzWTBS5V71PgvODMhkO7w@mail.gmail.com>
References: <CALNZuw==7kXb7zZgJYDjb8ey+47J0jK9dVHLTMX2Q7w2H+8ROw@mail.gmail.com>
	<CABHdwCr9+kieEKOfoG3akFAMV5+ZJfzWTBS5V71PgvODMhkO7w@mail.gmail.com>
Message-ID: <CALNZuwmA1CDzRT3BDNc01iO0Kq8k+z9AqsmK3_dUDQpQPp28bg@mail.gmail.com>

Thanks - it's actually quite easy!


editcap -r $infile $outfile $start_packet#-$last_packet#

-Avi


On Sun, Sep 11, 2011 at 3:52 PM, Wasim Baig <wasim at convergence.pk> wrote:

> http://www.wireshark.org/docs/man-pages/editcap.html
>
> -wasim
>
> On Sun, Sep 11, 2011 at 17:22, Avi Marcus <avi at avimarcus.net> wrote:
>
>> I've got a pcap file in wireshark that I want to show to someone... but I
>> only need the first 1285 frames, not all 103k. How do I cut it..?
>> Thanks,
>> -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
>>
>>
>
>
> --
> wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
> peace be upon you ...
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110911/db5b8ef1/attachment.html 

From contact at aharm.de  Sun Sep 11 23:52:56 2011
From: contact at aharm.de (Alexander Harm)
Date: Sun, 11 Sep 2011 21:52:56 +0200
Subject: [Freeswitch-users] Help needed on basic dialing plan: no calls
	are routed
In-Reply-To: <CAMWiMy9+qPNfz7U4EimYOdn-nHNXpOjKMGPds39XtceJJEmdFw@mail.gmail.com>
References: <DB197D09-AC19-49F5-86AF-7951F94F690B@aharm.de>
	<CAMWiMy95Ud-vY358XLDhKhMP=BoZ68SJ9nGgk3Bu57NSLveOWg@mail.gmail.com>
	<CF4DF76C-749B-483C-B814-B8B2C41C3455@aharm.de>
	<CAMWiMy9+qPNfz7U4EimYOdn-nHNXpOjKMGPds39XtceJJEmdFw@mail.gmail.com>
Message-ID: <23AF52E8-D5CE-4760-A07C-EC8217F66482@aharm.de>

doesn't make a difference.

if i call from my mobile/cell (0476058096) i hear the call get's picked up, but no ring tone. the internal phone 77 doesn't ring either. here is the log:

ps: if i insert a ring_ready i can hear a ring tone. the extension 77 still doesn't ring. i don't know what i'm missing. even calling between extensions doesn't work... there is no firewall active on the server...

alexander

2011-09-11 21:42:23.828066 [NOTICE] switch_channel.c:908 New Channel sofia/external/0476058096 at voip.belgacom.be [2bcd9716-dcae-11e0-99fc-f177d4ea3eea]
2011-09-11 21:42:23.828066 [DEBUG] sofia.c:5156 Channel sofia/external/0476058096 at voip.belgacom.be entering state [received][100]
2011-09-11 21:42:23.828066 [DEBUG] sofia.c:5167 Remote SDP:
v=0
o=- 0 102291732 IN IP4 81.240.251.38
s=IMSS
c=IN IP4 81.240.251.38
t=0 0
m=audio 12492 RTP/AVP 8 18 0 101
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

2011-09-11 21:42:23.828066 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
2011-09-11 21:42:23.828066 [DEBUG] sofia_glue.c:2857 Set Codec sofia/external/0476058096 at voip.belgacom.be PCMA/8000 20 ms 160 samples 64000 bits
2011-09-11 21:42:23.828066 [DEBUG] sofia_glue.c:4859 Set 2833 dtmf send/recv payload to 101
2011-09-11 21:42:23.828066 [DEBUG] sofia.c:5357 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_NEW -> CS_INIT
2011-09-11 21:42:23.828066 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 21:42:23.828066 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_INIT
2011-09-11 21:42:23.828066 [DEBUG] switch_core_state_machine.c:375 (sofia/external/0476058096 at voip.belgacom.be) State INIT
2011-09-11 21:42:23.828066 [DEBUG] mod_sofia.c:85 sofia/external/0476058096 at voip.belgacom.be SOFIA INIT
2011-09-11 21:42:23.828066 [DEBUG] mod_sofia.c:125 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_INIT -> CS_ROUTING
2011-09-11 21:42:23.828066 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 21:42:23.828066 [DEBUG] switch_core_state_machine.c:375 (sofia/external/0476058096 at voip.belgacom.be) State INIT going to sleep
2011-09-11 21:42:23.828066 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_ROUTING
2011-09-11 21:42:23.828066 [DEBUG] switch_channel.c:1837 (sofia/external/0476058096 at voip.belgacom.be) Callstate Change DOWN -> RINGING
2011-09-11 21:42:23.828066 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING
2011-09-11 21:42:23.828066 [DEBUG] mod_sofia.c:148 sofia/external/0476058096 at voip.belgacom.be SOFIA ROUTING
2011-09-11 21:42:23.828066 [DEBUG] switch_core_state_machine.c:78 sofia/external/0476058096 at voip.belgacom.be Standard ROUTING
2011-09-11 21:42:23.828066 [INFO] mod_dialplan_xml.c:336 Processing 0476058096 <0476058096>->015261786 in context public
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->unloop] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->outside_call] continue=true
Dialplan: sofia/external/0476058096 at voip.belgacom.be Absolute Condition [outside_call]
Dialplan: sofia/external/0476058096 at voip.belgacom.be Action set(outside_call=true) 
Dialplan: sofia/external/0476058096 at voip.belgacom.be Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->call_debug] continue=true
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->public_extensions] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [public_extensions] destination_number(015261786) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [public->015261786] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [015261786] context(public) =~ /public/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [015261786] destination_number(015261786) =~ /^015261786$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Action transfer(77 XML default) 
2011-09-11 21:42:23.828066 [DEBUG] switch_core_state_machine.c:128 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_ROUTING -> CS_EXECUTE
2011-09-11 21:42:23.828066 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 21:42:23.828066 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING going to sleep
2011-09-11 21:42:23.828066 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_EXECUTE
2011-09-11 21:42:23.828066 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE
2011-09-11 21:42:23.828066 [DEBUG] mod_sofia.c:241 sofia/external/0476058096 at voip.belgacom.be SOFIA EXECUTE
2011-09-11 21:42:23.828066 [DEBUG] switch_core_state_machine.c:166 sofia/external/0476058096 at voip.belgacom.be Standard EXECUTE
EXECUTE sofia/external/0476058096 at voip.belgacom.be set(outside_call=true)
2011-09-11 21:42:23.848092 [DEBUG] mod_dptools.c:1167 sofia/external/0476058096 at voip.belgacom.be SET [outside_call]=[true]
EXECUTE sofia/external/0476058096 at voip.belgacom.be set(RFC2822_DATE=Sun, 11 Sep 2011 21:42:23 +0200)
2011-09-11 21:42:23.848092 [DEBUG] mod_dptools.c:1167 sofia/external/0476058096 at voip.belgacom.be SET [RFC2822_DATE]=[Sun, 11 Sep 2011 21:42:23 +0200]
EXECUTE sofia/external/0476058096 at voip.belgacom.be transfer(77 XML default)
2011-09-11 21:42:23.848092 [DEBUG] switch_ivr.c:1693 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_EXECUTE -> CS_ROUTING
2011-09-11 21:42:23.848092 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 21:42:23.848092 [DEBUG] switch_core_session.c:724 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 21:42:23.848092 [NOTICE] switch_ivr.c:1699 Transfer sofia/external/0476058096 at voip.belgacom.be to XML[77 at default]
2011-09-11 21:42:23.848092 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE going to sleep
2011-09-11 21:42:23.848092 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_ROUTING
2011-09-11 21:42:23.848092 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING
2011-09-11 21:42:23.848092 [DEBUG] mod_sofia.c:148 sofia/external/0476058096 at voip.belgacom.be SOFIA ROUTING
2011-09-11 21:42:23.848092 [DEBUG] switch_core_state_machine.c:78 sofia/external/0476058096 at voip.belgacom.be Standard ROUTING
2011-09-11 21:42:23.848092 [INFO] mod_dialplan_xml.c:336 Processing 0476058096 <0476058096>->77 in context default
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->unloop] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->tod_example] continue=true
Dialplan: sofia/external/0476058096 at voip.belgacom.be Date/Time Match (FAIL) [tod_example] break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->holiday_example] continue=true
Dialplan: sofia/external/0476058096 at voip.belgacom.be Date/Time Match (FAIL) [holiday_example] break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->global-intercept] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [global-intercept] destination_number(77) =~ /^886$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->group-intercept] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [group-intercept] destination_number(77) =~ /^\*8$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->intercept-ext] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [intercept-ext] destination_number(77) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->redial] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [redial] destination_number(77) =~ /^(redial|870)$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->global] continue=true
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: sofia/external/0476058096 at voip.belgacom.be Absolute Condition [global]
Dialplan: sofia/external/0476058096 at voip.belgacom.be Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) 
Dialplan: sofia/external/0476058096 at voip.belgacom.be Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) 
Dialplan: sofia/external/0476058096 at voip.belgacom.be Action hash(insert/${domain_name}-last_dial/global/${uuid}) 
Dialplan: sofia/external/0476058096 at voip.belgacom.be Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->snom-demo-2] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [snom-demo-2] destination_number(77) =~ /^9001$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->snom-demo-1] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [snom-demo-1] destination_number(77) =~ /^9000$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->eavesdrop] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [eavesdrop] destination_number(77) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->eavesdrop] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [eavesdrop] destination_number(77) =~ /^779$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->call_return] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call_return] destination_number(77) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->del-group] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [del-group] destination_number(77) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->add-group] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [add-group] destination_number(77) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->call-group-simo] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [call-group-simo] destination_number(77) =~ /^82(\d{2})$/ break=on-false
Dialplan: sofia/exte