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
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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

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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
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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
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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
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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
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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
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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
>
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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

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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

 

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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
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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
>
>
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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
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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
>
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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
>
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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
>
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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
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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
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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 :)

--
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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
>
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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 :)
>
>
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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.
>
>
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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
>


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





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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
>
>
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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




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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
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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

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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
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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
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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
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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
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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
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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



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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
>
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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.



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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



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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
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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
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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
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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
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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
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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

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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

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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
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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)

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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)
> 

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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
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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
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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
>
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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
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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
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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
>
>
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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
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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
>
>
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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
>
>
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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
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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
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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
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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
>
>
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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
>
>
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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
>
>
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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
>
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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
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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
>
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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
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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
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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

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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...
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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

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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
>
>
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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




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

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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
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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
>
>
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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.
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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

 

 

 

 

 

 

 

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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
>
>
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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
>
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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

 

 

 

 

 

 

 

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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
>
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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

 

 

 

 

 

 

 

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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
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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
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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
>
>
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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
>
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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
>
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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.
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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

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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

 

 

 

 

 

 

 

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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



----

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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
>
>
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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
>
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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

 

 

 

 

 

 

 

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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

 

 

 

 

 

 

 

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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
>
>
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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
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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
>
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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
>
>
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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
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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
>>
>
>
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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
>
>
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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
>
>
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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
>
>
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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
>
>
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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
>
>
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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
>
>
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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
>
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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>

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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
>
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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
>>
>>
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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
>
>
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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
>>
>
>
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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
>
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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
>
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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
>
>
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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
>
>
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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
>
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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
>


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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
>>>
>>
>>
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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
>
>
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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
>>
>>
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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
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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



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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
>>
>>
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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.


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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
>
>
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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/>


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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
>
>
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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
>
>
>
> ----
>
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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 

--
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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?

--
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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.
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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

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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

--
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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
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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
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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
>
>
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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.
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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
>
>
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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
>
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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
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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
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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
>
>
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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
>
>
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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
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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
>
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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
>
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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
>
>
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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
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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
>
>
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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?

--
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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"/>


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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
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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
>
>
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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
>
>
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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
>
>
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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
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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
>
>
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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
>
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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
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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.


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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
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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
>
>
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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
>
>
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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
>
>
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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.
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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
>
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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

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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

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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
>
>
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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

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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?

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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
>
>
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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

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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
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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 ...
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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 ...
>
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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/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$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->015261786.0d] continue=false
Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [015261786.0d] destination_number(77) =~ /^0(\d+)$/ break=on-false
Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->Recordings] continue=false
2011-09-11 21:42:23.848092 [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 21:42:23.848092 [DEBUG] switch_core_state_machine.c:128 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_ROUTING -> CS_EXECUTE
2011-09-11 21:42:23.848092 [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_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING 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_EXECUTE
2011-09-11 21:42:23.848092 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE
2011-09-11 21:42:23.848092 [DEBUG] mod_sofia.c:241 sofia/external/0476058096 at voip.belgacom.be SOFIA EXECUTE
2011-09-11 21:42:23.848092 [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/192.168.1.101-spymap/0476058096/2bcd9716-dcae-11e0-99fc-f177d4ea3eea)
EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.101-last_dial/0476058096/77)
EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.101-last_dial/global/2bcd9716-dcae-11e0-99fc-f177d4ea3eea)
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 enum)
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 enum[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 [DEBUG] mod_enum.c:541 ENUM Lookup on 77

2011-09-11 21:42:53.748066 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 21:42:53.748066 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 21:42:53.748066 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:128 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_ROUTING -> CS_EXECUTE
2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING going to sleep
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_EXECUTE
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE
2011-09-11 21:42:53.848076 [DEBUG] mod_sofia.c:241 sofia/external/0476058096 at voip.belgacom.be SOFIA EXECUTE
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:166 sofia/external/0476058096 at voip.belgacom.be Standard EXECUTE
2011-09-11 21:42:53.848076 [DEBUG] sofia.c:5156 Channel sofia/external/0476058096 at voip.belgacom.be entering state [terminated][487]
2011-09-11 21:42:53.848076 [DEBUG] switch_channel.c:2797 (sofia/external/0476058096 at voip.belgacom.be) Callstate Change RINGING -> HANGUP
2011-09-11 21:42:53.848076 [NOTICE] sofia.c:5881 Hangup sofia/external/0476058096 at voip.belgacom.be [CS_EXECUTE] [ORIGINATOR_CANCEL]
2011-09-11 21:42:53.848076 [DEBUG] switch_channel.c:2813 Send signal sofia/external/0476058096 at voip.belgacom.be [KILL]
2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE going to sleep
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_HANGUP
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:576 (sofia/external/0476058096 at voip.belgacom.be) State HANGUP
2011-09-11 21:42:53.848076 [DEBUG] mod_sofia.c:452 sofia/external/0476058096 at voip.belgacom.be Overriding SIP cause 487 with 487 from the other leg
2011-09-11 21:42:53.848076 [DEBUG] mod_sofia.c:458 Channel sofia/external/0476058096 at voip.belgacom.be hanging up, cause: ORIGINATOR_CANCEL
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:46 sofia/external/0476058096 at voip.belgacom.be Standard HANGUP, cause: ORIGINATOR_CANCEL
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:576 (sofia/external/0476058096 at voip.belgacom.be) State HANGUP going to sleep
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:367 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_HANGUP -> CS_REPORTING
2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_REPORTING
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:636 (sofia/external/0476058096 at voip.belgacom.be) State REPORTING
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:53 sofia/external/0476058096 at voip.belgacom.be Standard REPORTING, cause: ORIGINATOR_CANCEL
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:636 (sofia/external/0476058096 at voip.belgacom.be) State REPORTING going to sleep
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:361 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_REPORTING -> CS_DESTROY
2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1349 Session 4 (sofia/external/0476058096 at voip.belgacom.be) Locked, Waiting on external entities
2011-09-11 21:42:53.848076 [NOTICE] switch_core_session.c:1367 Session 4 (sofia/external/0476058096 at voip.belgacom.be) Ended
2011-09-11 21:42:53.848076 [NOTICE] switch_core_session.c:1369 Close Channel sofia/external/0476058096 at voip.belgacom.be [CS_DESTROY]
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:465 (sofia/external/0476058096 at voip.belgacom.be) Callstate Change HANGUP -> DOWN
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:468 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_DESTROY
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:478 (sofia/external/0476058096 at voip.belgacom.be) State DESTROY
2011-09-11 21:42:53.848076 [DEBUG] mod_sofia.c:363 sofia/external/0476058096 at voip.belgacom.be SOFIA DESTROY
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:60 sofia/external/0476058096 at voip.belgacom.be Standard DESTROY
2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:478 (sofia/external/0476058096 at voip.belgacom.be) State DESTROY going to sleep

On 11.09.2011, at 17:15, curriegrad2004 wrote:

> 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
>> 
> 
> 
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> http://www.freeswitch.org




From gcd at i.ph  Mon Sep 12 04:47:27 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Mon, 12 Sep 2011 08:47:27 +0800
Subject: [Freeswitch-users] external sip profile
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper>
References: <D9A75749-55E5-4DC5-A401-5F887ABEDE1A@lyonl.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper>
Message-ID: <CAHKqMb4fp1jBxkjGAFVifBb6-MJxM9p01oiKzyqo8kaTQ7v2Vw@mail.gmail.com>

in addition to peter's advise, take a look at the SIP port 5070. FS is using
port 5080 for the external SIP profile. modify the port number at
"external.xml" then delete the port numbers in your proxy settings.

On Sun, Sep 11, 2011 at 6:36 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> 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!
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gcd at i.ph  Mon Sep 12 05:36:28 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Mon, 12 Sep 2011 09:36:28 +0800
Subject: [Freeswitch-users] freeswitch as a gateway with cdr lookup
In-Reply-To: <20110907170333.154010@gmx.net>
References: <20110907170333.154010@gmx.net>
Message-ID: <CAHKqMb6vsYC5=UC1z9TcL0xp0a7JOW0GUY8Lu5w4A3=mkgEjuA@mail.gmail.com>

hi thomas,

is your gateway a SIP-GSM? or FXO(analog)-GSM?

-nandy

On Thu, Sep 8, 2011 at 1:03 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
>
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From curriegrad2004 at gmail.com  Mon Sep 12 06:04:25 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 11 Sep 2011 19:04:25 -0700
Subject: [Freeswitch-users] Help needed on basic dialing plan: no calls
 are routed
In-Reply-To: <23AF52E8-D5CE-4760-A07C-EC8217F66482@aharm.de>
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>
	<23AF52E8-D5CE-4760-A07C-EC8217F66482@aharm.de>
Message-ID: <CAMWiMy9ppx83Td_tcp5A5YJcyRrS_QDHZ2jCAV7vpxjmjfvZ+g@mail.gmail.com>

Okay, I assumed you've probably set up the dialplans for the internal
extensions, but the extensions 77 and 88 aren't even defined at all in
the dialplan. You'll need to create a dialplan that defines those
extensions. Have a look at the default.xml dialplan file and find
extensions 1000-1019 and have a look at that config. This example
might give you a good idea on what you're trying to get accomplished.

http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_9:_Routing_DID_to_an_extension

On Sun, Sep 11, 2011 at 12:52 PM, Alexander Harm <contact at aharm.de> wrote:
> 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/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$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->015261786.0d] continue=false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [015261786.0d] destination_number(77) =~ /^0(\d+)$/ break=on-false
> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->Recordings] continue=false
> 2011-09-11 21:42:23.848092 [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 21:42:23.848092 [DEBUG] switch_core_state_machine.c:128 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_ROUTING -> CS_EXECUTE
> 2011-09-11 21:42:23.848092 [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_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING 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_EXECUTE
> 2011-09-11 21:42:23.848092 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE
> 2011-09-11 21:42:23.848092 [DEBUG] mod_sofia.c:241 sofia/external/0476058096 at voip.belgacom.be SOFIA EXECUTE
> 2011-09-11 21:42:23.848092 [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/192.168.1.101-spymap/0476058096/2bcd9716-dcae-11e0-99fc-f177d4ea3eea)
> EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.101-last_dial/0476058096/77)
> EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.101-last_dial/global/2bcd9716-dcae-11e0-99fc-f177d4ea3eea)
> 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 enum)
> 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 enum[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 [DEBUG] mod_enum.c:541 ENUM Lookup on 77
>
> 2011-09-11 21:42:53.748066 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
> 2011-09-11 21:42:53.748066 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
> 2011-09-11 21:42:53.748066 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:128 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_ROUTING -> CS_EXECUTE
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING going to sleep
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_EXECUTE
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE
> 2011-09-11 21:42:53.848076 [DEBUG] mod_sofia.c:241 sofia/external/0476058096 at voip.belgacom.be SOFIA EXECUTE
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:166 sofia/external/0476058096 at voip.belgacom.be Standard EXECUTE
> 2011-09-11 21:42:53.848076 [DEBUG] sofia.c:5156 Channel sofia/external/0476058096 at voip.belgacom.be entering state [terminated][487]
> 2011-09-11 21:42:53.848076 [DEBUG] switch_channel.c:2797 (sofia/external/0476058096 at voip.belgacom.be) Callstate Change RINGING -> HANGUP
> 2011-09-11 21:42:53.848076 [NOTICE] sofia.c:5881 Hangup sofia/external/0476058096 at voip.belgacom.be [CS_EXECUTE] [ORIGINATOR_CANCEL]
> 2011-09-11 21:42:53.848076 [DEBUG] switch_channel.c:2813 Send signal sofia/external/0476058096 at voip.belgacom.be [KILL]
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE going to sleep
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_HANGUP
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:576 (sofia/external/0476058096 at voip.belgacom.be) State HANGUP
> 2011-09-11 21:42:53.848076 [DEBUG] mod_sofia.c:452 sofia/external/0476058096 at voip.belgacom.be Overriding SIP cause 487 with 487 from the other leg
> 2011-09-11 21:42:53.848076 [DEBUG] mod_sofia.c:458 Channel sofia/external/0476058096 at voip.belgacom.be hanging up, cause: ORIGINATOR_CANCEL
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:46 sofia/external/0476058096 at voip.belgacom.be Standard HANGUP, cause: ORIGINATOR_CANCEL
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:576 (sofia/external/0476058096 at voip.belgacom.be) State HANGUP going to sleep
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:367 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_HANGUP -> CS_REPORTING
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_REPORTING
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:636 (sofia/external/0476058096 at voip.belgacom.be) State REPORTING
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:53 sofia/external/0476058096 at voip.belgacom.be Standard REPORTING, cause: ORIGINATOR_CANCEL
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:636 (sofia/external/0476058096 at voip.belgacom.be) State REPORTING going to sleep
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:361 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_REPORTING -> CS_DESTROY
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1349 Session 4 (sofia/external/0476058096 at voip.belgacom.be) Locked, Waiting on external entities
> 2011-09-11 21:42:53.848076 [NOTICE] switch_core_session.c:1367 Session 4 (sofia/external/0476058096 at voip.belgacom.be) Ended
> 2011-09-11 21:42:53.848076 [NOTICE] switch_core_session.c:1369 Close Channel sofia/external/0476058096 at voip.belgacom.be [CS_DESTROY]
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:465 (sofia/external/0476058096 at voip.belgacom.be) Callstate Change HANGUP -> DOWN
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:468 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_DESTROY
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:478 (sofia/external/0476058096 at voip.belgacom.be) State DESTROY
> 2011-09-11 21:42:53.848076 [DEBUG] mod_sofia.c:363 sofia/external/0476058096 at voip.belgacom.be SOFIA DESTROY
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:60 sofia/external/0476058096 at voip.belgacom.be Standard DESTROY
> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:478 (sofia/external/0476058096 at voip.belgacom.be) State DESTROY going to sleep
>
> On 11.09.2011, at 17:15, curriegrad2004 wrote:
>
>> 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
>>>>>
>>>>
>>>>
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>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
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>>>
>>
>>
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>
>
>
>
> FreeSWITCH-users mailing list
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From lakindia89 at gmail.com  Mon Sep 12 07:57:56 2011
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 12 Sep 2011 09:27:56 +0530
Subject: [Freeswitch-users] PRI Signalling Status down!
In-Reply-To: <201109102126.40444.nyilmaz@cybetech.com>
References: <201109102126.40444.nyilmaz@cybetech.com>
Message-ID: <CAMx6C7c5FHybKTy_-exzacxbSAwuewA-Gimir8uhWi0HkZoWFw@mail.gmail.com>

I suspect 2 things,
1) From your configuration it seems, that you are the NET. But when you
connect to TELCO, you should always be CPE.
2) Check whether it is related to CRC4 or not.


On Sat, Sep 10, 2011 at 11:56 PM, Nihat Yilmaz <nyilmaz at cybetech.com> wrote:

> **
>
> 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 <http://www.voip-info.org/boards/NOTICE> mod_freetdm.c:1819
> Alarm raised on channel 1:1
> 2011-09-10 16:47:27.948506 WARNING<http://www.voip-info.org/boards/WARNING> ftmod_libpri.c:1419
> channel 1:1 (1:1) has alarms! BLUE <http://www.voip-info.org/boards/BLUE>
> 2011-09-10 16:47:27.948506 NOTICE <http://www.voip-info.org/boards/NOTICE> mod_freetdm.c:1819
> Alarm raised on channel 1:2
> 2011-09-10 16:47:27.948506 WARNING<http://www.voip-info.org/boards/WARNING> ftmod_libpri.c:1419
> channel 1:2 (1:2) has alarms! BLUE <http://www.voip-info.org/boards/BLUE>
> .
> .
> .
> 2011-09-10 16:47:27.948506 NOTICE <http://www.voip-info.org/boards/NOTICE> mod_freetdm.c:1819
> Alarm raised on channel 1:30
> 2011-09-10 16:47:27.948506 WARNING<http://www.voip-info.org/boards/WARNING> ftmod_libpri.c:1419
> channel 1:30 (1:30) has alarms! BLUE<http://www.voip-info.org/boards/BLUE>
> 2011-09-10 16:47:27.948506 NOTICE <http://www.voip-info.org/boards/NOTICE> mod_freetdm.c:1819
> Alarm raised on channel 1:31
> 2011-09-10 16:47:27.948506 WARNING<http://www.voip-info.org/boards/WARNING> ftmod_libpri.c:1419
> channel 1:31 (1:31) has alarms! BLUE<http://www.voip-info.org/boards/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 <http://www.voip-info.org/boards/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<http://www.voip-info.org/boards/span%20zt%20GSMChannelBank>
> 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.
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From contact at aharm.de  Mon Sep 12 09:52:17 2011
From: contact at aharm.de (Alexander Harm)
Date: Mon, 12 Sep 2011 07:52:17 +0200
Subject: [Freeswitch-users] Help needed on basic dialing plan: no calls
	are routed
In-Reply-To: <CAMWiMy9ppx83Td_tcp5A5YJcyRrS_QDHZ2jCAV7vpxjmjfvZ+g@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>
	<23AF52E8-D5CE-4760-A07C-EC8217F66482@aharm.de>
	<CAMWiMy9ppx83Td_tcp5A5YJcyRrS_QDHZ2jCAV7vpxjmjfvZ+g@mail.gmail.com>
Message-ID: <13C62CB3-0841-467B-A4E0-8AF5079C9AB6@aharm.de>

wow, i thought i missed something basic. thanks a lot. one more thing: if i call the call gets disconnected after 10 seconds or so. i specified a call timeout of 120 seconds. what could be the reason?

    <extension name="RLRL_Extension">
      <condition field="destination_number" expression="^([1-9]{2})$">
	<action application="set" data="dialed_extension=$1"/>
	<action application="export" data="dialed_extension=$1"/>
	<!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
	<action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
	<action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
	<action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
	<action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/>
	<action application="set" data="ringback=${us-ring}"/>
	<action application="set" data="transfer_ringback=$${hold_music}"/>
	<action application="set" data="call_timeout=120"/>
	<!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> -->
	<action application="set" data="hangup_after_bridge=true"/>
	<!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
	<action application="set" data="continue_on_fail=true"/>
	<action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
	<action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
	<action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
	<action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
	<action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
	<!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
	<action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
	<action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
	<action application="answer"/>
	<action application="sleep" data="1000"/>
	<action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>


On 12.09.2011, at 04:04, curriegrad2004 wrote:

> Okay, I assumed you've probably set up the dialplans for the internal
> extensions, but the extensions 77 and 88 aren't even defined at all in
> the dialplan. You'll need to create a dialplan that defines those
> extensions. Have a look at the default.xml dialplan file and find
> extensions 1000-1019 and have a look at that config. This example
> might give you a good idea on what you're trying to get accomplished.
> 
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_9:_Routing_DID_to_an_extension
> 
> On Sun, Sep 11, 2011 at 12:52 PM, Alexander Harm <contact at aharm.de> wrote:
>> 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/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$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->015261786.0d] continue=false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [015261786.0d] destination_number(77) =~ /^0(\d+)$/ break=on-false
>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->Recordings] continue=false
>> 2011-09-11 21:42:23.848092 [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 21:42:23.848092 [DEBUG] switch_core_state_machine.c:128 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_ROUTING -> CS_EXECUTE
>> 2011-09-11 21:42:23.848092 [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_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING 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_EXECUTE
>> 2011-09-11 21:42:23.848092 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE
>> 2011-09-11 21:42:23.848092 [DEBUG] mod_sofia.c:241 sofia/external/0476058096 at voip.belgacom.be SOFIA EXECUTE
>> 2011-09-11 21:42:23.848092 [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/192.168.1.101-spymap/0476058096/2bcd9716-dcae-11e0-99fc-f177d4ea3eea)
>> EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.101-last_dial/0476058096/77)
>> EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.101-last_dial/global/2bcd9716-dcae-11e0-99fc-f177d4ea3eea)
>> 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 enum)
>> 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 enum[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 [DEBUG] mod_enum.c:541 ENUM Lookup on 77
>> 
>> 2011-09-11 21:42:53.748066 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 2011-09-11 21:42:53.748066 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 2011-09-11 21:42:53.748066 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:128 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_ROUTING -> CS_EXECUTE
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING going to sleep
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_EXECUTE
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE
>> 2011-09-11 21:42:53.848076 [DEBUG] mod_sofia.c:241 sofia/external/0476058096 at voip.belgacom.be SOFIA EXECUTE
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:166 sofia/external/0476058096 at voip.belgacom.be Standard EXECUTE
>> 2011-09-11 21:42:53.848076 [DEBUG] sofia.c:5156 Channel sofia/external/0476058096 at voip.belgacom.be entering state [terminated][487]
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_channel.c:2797 (sofia/external/0476058096 at voip.belgacom.be) Callstate Change RINGING -> HANGUP
>> 2011-09-11 21:42:53.848076 [NOTICE] sofia.c:5881 Hangup sofia/external/0476058096 at voip.belgacom.be [CS_EXECUTE] [ORIGINATOR_CANCEL]
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_channel.c:2813 Send signal sofia/external/0476058096 at voip.belgacom.be [KILL]
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE going to sleep
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_HANGUP
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:576 (sofia/external/0476058096 at voip.belgacom.be) State HANGUP
>> 2011-09-11 21:42:53.848076 [DEBUG] mod_sofia.c:452 sofia/external/0476058096 at voip.belgacom.be Overriding SIP cause 487 with 487 from the other leg
>> 2011-09-11 21:42:53.848076 [DEBUG] mod_sofia.c:458 Channel sofia/external/0476058096 at voip.belgacom.be hanging up, cause: ORIGINATOR_CANCEL
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:46 sofia/external/0476058096 at voip.belgacom.be Standard HANGUP, cause: ORIGINATOR_CANCEL
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:576 (sofia/external/0476058096 at voip.belgacom.be) State HANGUP going to sleep
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:367 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_HANGUP -> CS_REPORTING
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_REPORTING
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:636 (sofia/external/0476058096 at voip.belgacom.be) State REPORTING
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:53 sofia/external/0476058096 at voip.belgacom.be Standard REPORTING, cause: ORIGINATOR_CANCEL
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:636 (sofia/external/0476058096 at voip.belgacom.be) State REPORTING going to sleep
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:361 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_REPORTING -> CS_DESTROY
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1349 Session 4 (sofia/external/0476058096 at voip.belgacom.be) Locked, Waiting on external entities
>> 2011-09-11 21:42:53.848076 [NOTICE] switch_core_session.c:1367 Session 4 (sofia/external/0476058096 at voip.belgacom.be) Ended
>> 2011-09-11 21:42:53.848076 [NOTICE] switch_core_session.c:1369 Close Channel sofia/external/0476058096 at voip.belgacom.be [CS_DESTROY]
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:465 (sofia/external/0476058096 at voip.belgacom.be) Callstate Change HANGUP -> DOWN
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:468 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_DESTROY
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:478 (sofia/external/0476058096 at voip.belgacom.be) State DESTROY
>> 2011-09-11 21:42:53.848076 [DEBUG] mod_sofia.c:363 sofia/external/0476058096 at voip.belgacom.be SOFIA DESTROY
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:60 sofia/external/0476058096 at voip.belgacom.be Standard DESTROY
>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:478 (sofia/external/0476058096 at voip.belgacom.be) State DESTROY going to sleep
>> 
>> On 11.09.2011, at 17:15, curriegrad2004 wrote:
>> 
>>> 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
>>>> 
>>> 
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Glen.Ganderton at premier.com.au  Mon Sep 12 10:26:34 2011
From: Glen.Ganderton at premier.com.au (Glen Ganderton)
Date: Mon, 12 Sep 2011 16:26:34 +1000
Subject: [Freeswitch-users] Lua XML parser
Message-ID: <CD5A1499C8DEF34FAA3509722B00393E74029E2ECA@m-mail03.office.premiertech>

Hey Guys,

I'm currently trying to create a basic IVR with voice recognition (just a simple YES or NO response). I have successfully configured the freeswitch MRCP client to connect to my nuance MRCP server and I successfully get a responses from the server. Now what I need to do is process the response in my freeswitch lua script. The nuance box returns the following:

----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
Completion-Cause: 000 success
Content-Type: application/nlsml+xml
Content-Length: 178

<?xml version='1.0'?><result><interpretation grammar="session:nuance5-mrcp2" confidence="0.99"><input mode="speech">yes</input><instance>true</instance></interpretation></result>

Now what I want to be able to do is parse the XML in my lua script so that I can simply have the XML stored in some variables:

Eg.

Confidence = 0.99
Input = yes
Instance =  true
....etc.

I am just starting to learn lua and am unable to do this myself. I have seen a few scripts online to parse XML for lua but none seem to work correctly (maybe they are meant for different format..not sure).

If anybody could point me in the right direction or write some sample code for what I need to do that would be a great help.

Thanks in advance.

-Glen
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From avi at avimarcus.net  Mon Sep 12 10:37:05 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 12 Sep 2011 09:37:05 +0300
Subject: [Freeswitch-users] Lua XML parser
In-Reply-To: <CD5A1499C8DEF34FAA3509722B00393E74029E2ECA@m-mail03.office.premiertech>
References: <CD5A1499C8DEF34FAA3509722B00393E74029E2ECA@m-mail03.office.premiertech>
Message-ID: <CALNZuw=k=0S6DruBs3omM7Tvzk_Lo4tarFL35uxettzWn-A3rg@mail.gmail.com>

I tried a whole bunch from http://lua-users.org/wiki/LuaXml in order to
query mod_lcr and then custom-use the results but I wasn't able to get
"keyed" results, only numbered ones. I'm also interested in this..

-Avi


On Mon, Sep 12, 2011 at 9:26 AM, Glen Ganderton <
Glen.Ganderton at premier.com.au> wrote:

> Hey Guys,****
>
> ** **
>
> I?m currently trying to create a basic IVR with voice recognition (just a
> simple YES or NO response). I have successfully configured the freeswitch
> MRCP client to connect to my nuance MRCP server and I successfully get a
> responses from the server. Now what I need to do is process the response in
> my freeswitch lua script. The nuance box returns the following:****
>
> ** **
>
>
> ----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
> ****
>
> Completion-Cause: 000 success****
>
> Content-Type: application/nlsml+xml****
>
> Content-Length: 178****
>
> ** **
>
> <?xml version='1.0'?><result><interpretation
> grammar="session:nuance5-mrcp2" confidence="0.99"><input
> mode="speech">yes</input><instance>true</instance></interpretation></result>
> ****
>
> ** **
>
> Now what I want to be able to do is parse the XML in my lua script so that
> I can simply have the XML stored in some variables:****
>
> ** **
>
> Eg.****
>
> ** **
>
> Confidence = 0.99****
>
> Input = yes****
>
> Instance =  true****
>
> ?.etc.****
>
> ** **
>
> I am just starting to learn lua and am unable to do this myself. I have
> seen a few scripts online to parse XML for lua but none seem to work
> correctly (maybe they are meant for different format..not sure).****
>
> ** **
>
> If anybody could point me in the right direction or write some sample code
> for what I need to do that would be a great help.****
>
> ** **
>
> Thanks 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
>
>
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From peter.schrock at gmail.com  Mon Sep 12 10:49:07 2011
From: peter.schrock at gmail.com (Peter Schrock)
Date: Sun, 11 Sep 2011 23:49:07 -0700
Subject: [Freeswitch-users] No Rule to make target 'tport/libtport.la'
Message-ID: <CA+PZoDS4iJhrS=6Z91y5NCeeQ4a1q1+H+hQBNK-DC97X+BZCgQ@mail.gmail.com>

So, I figured out how to fix the 'swab' problem using a powerpc. Now, I am
getting this error message:

LINK libnua.la
make[8]: *** No rule to make target 'tport/libtport.la', needed by '
libsofia-sip-us.la'. Stop.

Does any one have any ideas?

Peter
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From cvogel at lyonl.com  Mon Sep 12 06:00:51 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Mon, 12 Sep 2011 02:00:51 +0000
Subject: [Freeswitch-users] external sip profile
In-Reply-To: <CAHKqMb4fp1jBxkjGAFVifBb6-MJxM9p01oiKzyqo8kaTQ7v2Vw@mail.gmail.com>
References: <D9A75749-55E5-4DC5-A401-5F887ABEDE1A@lyonl.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper>
	<CAHKqMb4fp1jBxkjGAFVifBb6-MJxM9p01oiKzyqo8kaTQ7v2Vw@mail.gmail.com>
Message-ID: <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com>

Level 3 uses port 5070 for their sip server but requires port 5060 to be used on our side; I made changes to the vars.xml config file to support this. I was just able to get the server to answer a call by changing the register value to false and adding an entry in the ACL config for the Level 3 server however now it seems that the audio isn't working correctly. I'm using a simple dial plan to echo the audio back but all I get is dead air.

    <extension name="level3">
      <condition field="destination_number" expression="^13127567000$">
        <action application="answer"/>
        <action application="echo" />
      </condition>
    </extension>



On Sep 11, 2011, at 7:47 PM, Nandy Dagondon wrote:

in addition to peter's advise, take a look at the SIP port 5070. FS is using port 5080 for the external SIP profile. modify the port number at "external.xml" then delete the port numbers in your proxy settings.

On Sun, Sep 11, 2011 at 6:36 PM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
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<mailto:freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] f&#246;r Chad Vogel [cvogel at lyonl.com<mailto:cvogel at lyonl.com>]
Skickat: den 11 september 2011 03:28
Till: freeswitch-users at lists.freeswitch.org<mailto: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<http://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!


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
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From acrow at integrafin.co.uk  Mon Sep 12 13:25:35 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 12 Sep 2011 10:25:35 +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: <4E6DD00F.5050303@integrafin.co.uk>

Brian,

Snom think that this flashing on hold is incorrect - I had this on my 
snom forum post:

No! I've never seen such behaviour - Neither the LED is supposed to
blink when the monitored party is on hold, nor it should be possible to
steal the call by hitting the function key. That indeed is extremely
confusing and is definitely not intended.


I don't have much (if at all) experience with Freeswitch but from what I
  understand, extension monitoring is being configured through dial plans
  (and?/) or themod_snom module  <http://wiki.freeswitch.org/wiki/Mod_snom>.

I guess something is not configured properly which results in the problem you are encountering.


It should not make any difference whether the monitored extension is set
  on hold or not. Can you post a SIP trace which shows such a scenario?


I replied with:

Extension monitoring is being done with bog-standard subscriptions, not with mod_snom or anything extra in the dialplan.


The Polycoms flash the button while a monitored line is held, but you can't pick it up.


I will try to get a trace from the phone, however someone re-used my
test FS box for a windows desktop while I was away, so I have to build a
  new one.


Any more ideas as to why this is happening?

Cheers

Alex

On 05/09/11 23:15, Brian West wrote:
> 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
>
>
> -- 
> This message has been scanned for viruses and
> dangerous content by *MailScanner* <http://www.mailscanner.info/>, and is
> believed to be clean.
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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-- 
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
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From fs-list at communicatefreely.net  Mon Sep 12 19:49:11 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Mon, 12 Sep 2011 11:49:11 -0400
Subject: [Freeswitch-users] TOS / DSCP marking in FreeBSD?
Message-ID: <4E6E29F7.9050004@communicatefreely.net>

Hello,

Does anyone know how to make Freeswitch mark outgoing SIP and RTP 
packets with appropriate TOS or DSCP values?  Most other software does 
this when it creates the packets but Freeswitch sends everything as 0x0  
I found some great examples on the Wiki on doing this with IPTABLES in 
Linux, but that doesn't apply to FreeBSD  We can use IPFW or PF, but I 
can't find any information there on re-marking.  Everything in those 
packages seems to relate to using the marks, but not setting them.

Thanks!

-Tim


From fs-list at communicatefreely.net  Mon Sep 12 19:58:03 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Mon, 12 Sep 2011 11:58:03 -0400
Subject: [Freeswitch-users] Follow Me and Call Forwarding
In-Reply-To: <1B608BE2-2D12-48B9-ACAF-C14E220C861C@5ninesolutions.com>
References: <DB319D80-A364-41A8-B972-75A1713B7C7C@5ninesolutions.com>	<CALNZuwmXfw23LZRSD=JCN5QOwN4OHAGXtgiy_kGOWcY1mo+nxQ@mail.gmail.com>
	<1B608BE2-2D12-48B9-ACAF-C14E220C861C@5ninesolutions.com>
Message-ID: <4E6E2C0B.1080400@communicatefreely.net>

You could replace the dial string in the directory.

Instead of doing forwarding in the dialplan, you could specify an 
alternate bridge string.

The default bridge string is normally 
${sofia_contact(${dialed_user}@${dialed_domain})}

but you could specify something else on a per-user basis for in the 
directory.  If you did
${sofia_contact(5243@${dialed_domain})} any calls bridged to this user 
(using the user/extension at domain syntax) would call extension 5243 
instead.  If you want this to be an external destination, you would 
replace the sofia_contact() with a meaningful gateway dial string.

The user/exten at domain syntax is just a macro to look up the user in the 
directory.  You can return whatever you want, it doesn't have to be the 
sofia contact of that user, it could be anything.

Hope that makes sense.

-Tim

Spencer Thomason wrote:
> The problem is that it doesn't work from a hunt group because the 
> extensions are dialed with a bridge to user/EXTENSION@${domain_name} 
> instead of fro example transfer EXTENSION XML default since the follow 
> me or forward is inserted as another extension with the same number 
> just earlier in the dialplan.
>
>  
> On Sep 10, 2011, at 10:22 PM, Avi Marcus wrote:
>
>> It sounds like you already have it doing what you want it to do. How 
>> do you want it to work differently..?
>> -Avi
>>
>>
>>
>> On Sun, Sep 11, 2011 at 7:19 AM, Spencer Thomason 
>> <spencer at 5ninesolutions.com <mailto:spencer at 5ninesolutions.com>> wrote:
>>
>>     Hello all,
>>     I'd like to implement a graphical call forwarding and follow me
>>     feature that works from hunt groups as well as a direct dial to
>>     an extension.  Currently when a user enables follow me for their
>>     extension a new dialplan entry is created with the same
>>     destination number as the extension before the local user
>>     extension which dials the extension with a bridge to
>>     user/${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
>>     <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
>
> ------------------------------------------------------------------------
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 cogs66 at gmail.com  Mon Sep 12 13:22:30 2011
From: cogs66 at gmail.com (cogs66)
Date: Mon, 12 Sep 2011 02:22:30 -0700 (PDT)
Subject: [Freeswitch-users] User Agent Displayed by sipsak
Message-ID: <1315819350559-6782746.post@n2.nabble.com>

Hi

Very much learning here and working on security at present.

I have used sipsak to see what is displayed to the outside world. Whilst I
am proud to be using FreeSWITCH I am concerned this is too much info?

User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-db88cd3 2011-08-25 13-27-11
-0500 

I also notice SIP/2.0 200 OK. When I run on a provider I see SIP/2.0 501
Method not allowed. Maybe this is because they use a SIP proxy and I am not
here.

Maybe this is nothing to worry about but I would rather be sure and would
appreciate any feedback here.

Cheers

A

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/User-Agent-Displayed-by-sipsak-tp6782746p6782746.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From faisal.rehman22 at hotmail.com  Mon Sep 12 15:16:35 2011
From: faisal.rehman22 at hotmail.com (Faisal Rehman)
Date: Mon, 12 Sep 2011 17:16:35 +0600
Subject: [Freeswitch-users] Codec Mis Match Issue
Message-ID: <BAY167-W107FFB0EB4D315435F6BE0CEC020@phx.gbl>


Dear All,
I have a problem of some codec mis-match issue while making calls with freeswitch, the calls coming are all with only one codec G711 and I am amazed with the strange behaviour of Free-Switchthat some times it responds the calls with the same codec and making them successful but numerous times it responds with G729 & hence calls got failed due to this. I have disabled the G729 codec in all my configuration but I don't know where is it coming from.

Any help regarding it would be highly appreicated.


Thanks 

Faisal Rehman 		 	   		  
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From brtantunes at gmail.com  Mon Sep 12 18:30:48 2011
From: brtantunes at gmail.com (Brtantunes)
Date: Mon, 12 Sep 2011 07:30:48 -0700 (PDT)
Subject: [Freeswitch-users] Timeout Message or something similar
Message-ID: <1315837848697-6783735.post@n2.nabble.com>

Hello,

I want to create a bridge between two users, something like this:

originate {ringback=c:/teste.wav}sofia/DOMAIN/1001
&bridge(sofia/DOMAIN/1002)

This works fine. But now i want to add a timeout message if the second user
don't answer (a sound played to the first user cause of the no_answer of se
second user). Is that possible?

tks for your help

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From msc at freeswitch.org  Mon Sep 12 20:46:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Sep 2011 09:46:10 -0700
Subject: [Freeswitch-users] Cut PCAP file?
In-Reply-To: <CALNZuwmA1CDzRT3BDNc01iO0Kq8k+z9AqsmK3_dUDQpQPp28bg@mail.gmail.com>
References: <CALNZuw==7kXb7zZgJYDjb8ey+47J0jK9dVHLTMX2Q7w2H+8ROw@mail.gmail.com>
	<CABHdwCr9+kieEKOfoG3akFAMV5+ZJfzWTBS5V71PgvODMhkO7w@mail.gmail.com>
	<CALNZuwmA1CDzRT3BDNc01iO0Kq8k+z9AqsmK3_dUDQpQPp28bg@mail.gmail.com>
Message-ID: <CAKzWOxXK5EdrCjRGHyyecMXPnHDdC7hS3Y4OZ82eKxjNrP3Jjw@mail.gmail.com>

And you were so proud of finding this that you put it on the wiki also? ;)
-MC

On Sun, Sep 11, 2011 at 9:11 AM, Avi Marcus <avi at avimarcus.net> wrote:

> 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
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian.wiese.freeswitch at gmail.com  Mon Sep 12 22:36:06 2011
From: brian.wiese.freeswitch at gmail.com (Brian Wiese)
Date: Mon, 12 Sep 2011 13:36:06 -0500
Subject: [Freeswitch-users] Multiple DTMF on Single Keypress
In-Reply-To: <CALNZuwm+vi61Q=w-qjS73mCFcrk4ra8dr+_80+w39_Yrpw-F2w@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>
	<CALNZuwm+vi61Q=w-qjS73mCFcrk4ra8dr+_80+w39_Yrpw-F2w@mail.gmail.com>
Message-ID: <CADEQS5xOBjAgfvFhF8AZ2RivRE787Q0iRbJHCUyS+K=hEVQXQQ@mail.gmail.com>

Avi:

Thank you for taking the time to review my issue and for your help.  I will
JIRA the issue.

~Brian

On Sat, Sep 10, 2011 at 12:13 PM, Avi Marcus <avi at avimarcus.net> wrote:

> 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
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Sep 12 22:45:24 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Sep 2011 11:45:24 -0700
Subject: [Freeswitch-users] [TRAINING] OpenSIPS EBootcamp
In-Reply-To: <4E6E4533.6080500@opensips.org>
References: <4E6E4533.6080500@opensips.org>
Message-ID: <CAKzWOxXfz67vwaZAX1gTVvQwudKT2v4=tygxW5R13NELj1NnNw@mail.gmail.com>

Dear FreeSWITCH Community,

For those of you who may not be aware, Bogdan-Andrei Iancu is the lead
author of the OpenSIPS SIP proxy and is a regular at ClueCon. If you are
looking to gain more exposure and get some "official" OpenSIPS training
please read the following message from Bogdan. He has made a number of
training options available.

Thanks!
-Michael

>From Bogdan:

**
 Hi all,

A New OpenSIPS eBootcamp training session is about to start next week.
    *19th of September 2011*

The training will touch several topics of inters for the FreeSwitch users,
like Load Balancing, PSTN connectivity, Edge functionality (NAT traversal,
security), Advanced SIP call flows.

This new eBootcamp session targets OpenSIPS 1.7.0 with the following
content:
    SIP introduction
    OpenSIPS introduction
    SQL support (authentication, aliases, domains)
    OpenSIPS Control Panel web interface
    PSTN connectivity (dialplan, ACLs, drouting, failover)
    Advanced SIP Call Flows (parallel and serial forking, call forwarding,
call transfer)
    SIP presence
    SIP Dialog Awareness (profiling, topology hiding, security)
    Load Balancing with OpenSIPS
    OpenSIPS High Availability
    SIP NAT traversal
    OpenSIPS accounting and billing
    SIP security (floods, auth, DNS poisoning, register attacks, TLS and
SRTP)
    OpenSIPS B2BUA ( topology hiding )

More details on:
   http://www.opensips.org/Training/EBootcamp

Registration at:
   http://ebootcamp.opensips.org/

Questions:
   bootcamp at opensips.org

Best regards,
Bogdan

-- 
Bogdan-Andrei Iancu
OpenSIPS eBootcamp - 19th of September 2011
OpenSIPS solutions and "know-how"
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From aakashviswam at gmail.com  Mon Sep 12 22:54:29 2011
From: aakashviswam at gmail.com (Aakash)
Date: Mon, 12 Sep 2011 11:54:29 -0700 (PDT)
Subject: [Freeswitch-users] how to block a particular number
Message-ID: <1315853669503-6784851.post@n2.nabble.com>

Hi,

Is that possible to block the particular incoming number to the extension ?


Thanks,
Aakash

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/how-to-block-a-particular-number-tp6784851p6784851.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From avi at avimarcus.net  Mon Sep 12 23:20:12 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 12 Sep 2011 22:20:12 +0300
Subject: [Freeswitch-users] how to block a particular number
In-Reply-To: <1315853669503-6784851.post@n2.nabble.com>
References: <1315853669503-6784851.post@n2.nabble.com>
Message-ID: <CALNZuwkmrRrEw6mcEJ_6N3Yz_VwDOsV3OvnJban8-k0ZV5WAXQ@mail.gmail.com>

You can set up an extension in the dialplan to route a certain number to
hangup, or sip:lenny at sip.itslenny.com:5060  or the like, before it tries to
ring the extension.

-Avi


On Mon, Sep 12, 2011 at 9:54 PM, Aakash <aakashviswam at gmail.com> wrote:

> Hi,
>
> Is that possible to block the particular incoming number to the extension ?
>
>
> Thanks,
> Aakash
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/how-to-block-a-particular-number-tp6784851p6784851.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From curriegrad2004 at gmail.com  Mon Sep 12 23:26:28 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 12 Sep 2011 12:26:28 -0700
Subject: [Freeswitch-users] how to block a particular number
In-Reply-To: <CALNZuwkmrRrEw6mcEJ_6N3Yz_VwDOsV3OvnJban8-k0ZV5WAXQ@mail.gmail.com>
References: <1315853669503-6784851.post@n2.nabble.com>
	<CALNZuwkmrRrEw6mcEJ_6N3Yz_VwDOsV3OvnJban8-k0ZV5WAXQ@mail.gmail.com>
Message-ID: <CAMWiMy8s4NwHBEVMZQP0u1eAO07+Wea=zJH5Vbd29KaCRZEYFA@mail.gmail.com>

Also, be sure to check up on itslenny.com from time to time. You'll
hear some very interesting calls there.

On Mon, Sep 12, 2011 at 12:20 PM, Avi Marcus <avi at avimarcus.net> wrote:
> You can set up an extension in the dialplan to route a certain number to
> hangup, or?sip:lenny at sip.itslenny.com:5060 ?or the like, before it tries to
> ring the extension.
> -Avi
>
> On Mon, Sep 12, 2011 at 9:54 PM, Aakash <aakashviswam at gmail.com> wrote:
>>
>> Hi,
>>
>> Is that possible to block the particular incoming number to the extension
>> ?
>>
>>
>> Thanks,
>> Aakash
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/how-to-block-a-particular-number-tp6784851p6784851.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From curriegrad2004 at gmail.com  Mon Sep 12 23:29:46 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 12 Sep 2011 12:29:46 -0700
Subject: [Freeswitch-users] Help needed on basic dialing plan: no calls
 are routed
In-Reply-To: <13C62CB3-0841-467B-A4E0-8AF5079C9AB6@aharm.de>
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>
	<23AF52E8-D5CE-4760-A07C-EC8217F66482@aharm.de>
	<CAMWiMy9ppx83Td_tcp5A5YJcyRrS_QDHZ2jCAV7vpxjmjfvZ+g@mail.gmail.com>
	<13C62CB3-0841-467B-A4E0-8AF5079C9AB6@aharm.de>
Message-ID: <CAMWiMy_BV6CMDpgXt1zcn0qV5YVebOkwnrME3WsFaGfQBSxgZw@mail.gmail.com>

This would be something with your provider. You may want to talk to
them about this or you can always make it answer first by appending
this:

<action application="answer" />

Before the dialplan does anything inside the condition tag. However be
warned, the remote caller will have to pay for any airtime/charges
once you force it to answer first, so choose carefully.

On Sun, Sep 11, 2011 at 10:52 PM, Alexander Harm <contact at aharm.de> wrote:
> wow, i thought i missed something basic. thanks a lot. one more thing: if i call the call gets disconnected after 10 seconds or so. i specified a call timeout of 120 seconds. what could be the reason?
>
> ? ?<extension name="RLRL_Extension">
> ? ? ?<condition field="destination_number" expression="^([1-9]{2})$">
> ? ? ? ?<action application="set" data="dialed_extension=$1"/>
> ? ? ? ?<action application="export" data="dialed_extension=$1"/>
> ? ? ? ?<!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
> ? ? ? ?<action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
> ? ? ? ?<action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
> ? ? ? ?<action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
> ? ? ? ?<action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/>
> ? ? ? ?<action application="set" data="ringback=${us-ring}"/>
> ? ? ? ?<action application="set" data="transfer_ringback=$${hold_music}"/>
> ? ? ? ?<action application="set" data="call_timeout=120"/>
> ? ? ? ?<!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> -->
> ? ? ? ?<action application="set" data="hangup_after_bridge=true"/>
> ? ? ? ?<!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
> ? ? ? ?<action application="set" data="continue_on_fail=true"/>
> ? ? ? ?<action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
> ? ? ? ?<action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
> ? ? ? ?<action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
> ? ? ? ?<action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
> ? ? ? ?<action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
> ? ? ? ?<!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
> ? ? ? ?<action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
> ? ? ? ?<action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
> ? ? ? ?<action application="answer"/>
> ? ? ? ?<action application="sleep" data="1000"/>
> ? ? ? ?<action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
> ? ? ?</condition>
> ? ?</extension>
>
>
> On 12.09.2011, at 04:04, curriegrad2004 wrote:
>
>> Okay, I assumed you've probably set up the dialplans for the internal
>> extensions, but the extensions 77 and 88 aren't even defined at all in
>> the dialplan. You'll need to create a dialplan that defines those
>> extensions. Have a look at the default.xml dialplan file and find
>> extensions 1000-1019 and have a look at that config. This example
>> might give you a good idea on what you're trying to get accomplished.
>>
>> http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_9:_Routing_DID_to_an_extension
>>
>> On Sun, Sep 11, 2011 at 12:52 PM, Alexander Harm <contact at aharm.de> wrote:
>>> 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/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$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->015261786.0d] continue=false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be Regex (FAIL) [015261786.0d] destination_number(77) =~ /^0(\d+)$/ break=on-false
>>> Dialplan: sofia/external/0476058096 at voip.belgacom.be parsing [default->Recordings] continue=false
>>> 2011-09-11 21:42:23.848092 [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 21:42:23.848092 [DEBUG] switch_core_state_machine.c:128 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_ROUTING -> CS_EXECUTE
>>> 2011-09-11 21:42:23.848092 [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_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING 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_EXECUTE
>>> 2011-09-11 21:42:23.848092 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE
>>> 2011-09-11 21:42:23.848092 [DEBUG] mod_sofia.c:241 sofia/external/0476058096 at voip.belgacom.be SOFIA EXECUTE
>>> 2011-09-11 21:42:23.848092 [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/192.168.1.101-spymap/0476058096/2bcd9716-dcae-11e0-99fc-f177d4ea3eea)
>>> EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.101-last_dial/0476058096/77)
>>> EXECUTE sofia/external/0476058096 at voip.belgacom.be hash(insert/192.168.1.101-last_dial/global/2bcd9716-dcae-11e0-99fc-f177d4ea3eea)
>>> 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 enum)
>>> 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 enum[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 [DEBUG] mod_enum.c:541 ENUM Lookup on 77
>>>
>>> 2011-09-11 21:42:53.748066 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>> 2011-09-11 21:42:53.748066 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>> 2011-09-11 21:42:53.748066 [DEBUG] switch_core_session.c:870 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:128 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_ROUTING -> CS_EXECUTE
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:384 (sofia/external/0476058096 at voip.belgacom.be) State ROUTING going to sleep
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_EXECUTE
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE
>>> 2011-09-11 21:42:53.848076 [DEBUG] mod_sofia.c:241 sofia/external/0476058096 at voip.belgacom.be SOFIA EXECUTE
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:166 sofia/external/0476058096 at voip.belgacom.be Standard EXECUTE
>>> 2011-09-11 21:42:53.848076 [DEBUG] sofia.c:5156 Channel sofia/external/0476058096 at voip.belgacom.be entering state [terminated][487]
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_channel.c:2797 (sofia/external/0476058096 at voip.belgacom.be) Callstate Change RINGING -> HANGUP
>>> 2011-09-11 21:42:53.848076 [NOTICE] sofia.c:5881 Hangup sofia/external/0476058096 at voip.belgacom.be [CS_EXECUTE] [ORIGINATOR_CANCEL]
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_channel.c:2813 Send signal sofia/external/0476058096 at voip.belgacom.be [KILL]
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:391 (sofia/external/0476058096 at voip.belgacom.be) State EXECUTE going to sleep
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_HANGUP
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:576 (sofia/external/0476058096 at voip.belgacom.be) State HANGUP
>>> 2011-09-11 21:42:53.848076 [DEBUG] mod_sofia.c:452 sofia/external/0476058096 at voip.belgacom.be Overriding SIP cause 487 with 487 from the other leg
>>> 2011-09-11 21:42:53.848076 [DEBUG] mod_sofia.c:458 Channel sofia/external/0476058096 at voip.belgacom.be hanging up, cause: ORIGINATOR_CANCEL
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:46 sofia/external/0476058096 at voip.belgacom.be Standard HANGUP, cause: ORIGINATOR_CANCEL
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:576 (sofia/external/0476058096 at voip.belgacom.be) State HANGUP going to sleep
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:367 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_HANGUP -> CS_REPORTING
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:336 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_REPORTING
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:636 (sofia/external/0476058096 at voip.belgacom.be) State REPORTING
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:53 sofia/external/0476058096 at voip.belgacom.be Standard REPORTING, cause: ORIGINATOR_CANCEL
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:636 (sofia/external/0476058096 at voip.belgacom.be) State REPORTING going to sleep
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:361 (sofia/external/0476058096 at voip.belgacom.be) State Change CS_REPORTING -> CS_DESTROY
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/0476058096 at voip.belgacom.be [BREAK]
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_session.c:1349 Session 4 (sofia/external/0476058096 at voip.belgacom.be) Locked, Waiting on external entities
>>> 2011-09-11 21:42:53.848076 [NOTICE] switch_core_session.c:1367 Session 4 (sofia/external/0476058096 at voip.belgacom.be) Ended
>>> 2011-09-11 21:42:53.848076 [NOTICE] switch_core_session.c:1369 Close Channel sofia/external/0476058096 at voip.belgacom.be [CS_DESTROY]
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:465 (sofia/external/0476058096 at voip.belgacom.be) Callstate Change HANGUP -> DOWN
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:468 (sofia/external/0476058096 at voip.belgacom.be) Running State Change CS_DESTROY
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:478 (sofia/external/0476058096 at voip.belgacom.be) State DESTROY
>>> 2011-09-11 21:42:53.848076 [DEBUG] mod_sofia.c:363 sofia/external/0476058096 at voip.belgacom.be SOFIA DESTROY
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:60 sofia/external/0476058096 at voip.belgacom.be Standard DESTROY
>>> 2011-09-11 21:42:53.848076 [DEBUG] switch_core_state_machine.c:478 (sofia/external/0476058096 at voip.belgacom.be) State DESTROY going to sleep
>>>
>>> On 11.09.2011, at 17:15, curriegrad2004 wrote:
>>>
>>>> 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
>>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>


From cvogel at lyonl.com  Mon Sep 12 23:57:36 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Mon, 12 Sep 2011 19:57:36 +0000
Subject: [Freeswitch-users] User Agent Displayed by sipsak
In-Reply-To: <1315819350559-6782746.post@n2.nabble.com>
References: <1315819350559-6782746.post@n2.nabble.com>
Message-ID: <2B146FED-757D-45E2-BB9E-F18FE6F370CE@lyonl.com>

in the external.xml config file use:

<param name="user-agent-string" value="your agent here"/>


On Sep 12, 2011, at 4:22 AM, cogs66 wrote:

Hi

Very much learning here and working on security at present.

I have used sipsak to see what is displayed to the outside world. Whilst I
am proud to be using FreeSWITCH I am concerned this is too much info?

User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-db88cd3 2011-08-25 13-27-11
-0500

I also notice SIP/2.0 200 OK. When I run on a provider I see SIP/2.0 501
Method not allowed. Maybe this is because they use a SIP proxy and I am not
here.

Maybe this is nothing to worry about but I would rather be sure and would
appreciate any feedback here.

Cheers

A

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/User-Agent-Displayed-by-sipsak-tp6782746p6782746.html
Sent from the freeswitch-users mailing list archive at Nabble.com<http://Nabble.com>.


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From gavin.henry at gmail.com  Tue Sep 13 00:36:27 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 12 Sep 2011 21:36:27 +0100
Subject: [Freeswitch-users] One Way Audio - Auto Change RTP port?
In-Reply-To: <CAAuJ8MDdz-EoaLRzR2es+aH0tURL9kN0K+6bQcKEpP_RJ0BEGw@mail.gmail.com>
References: <CAAuJ8MBnFKYzEjPnCNeNmiU-T=jRu3M1vcrR49-p2LPyRU=+uQ@mail.gmail.com>
	<CAKbxfG8vU1pe-TU5_PHT8BcabYPceLem9QiyysCDxnf+gKSBYQ@mail.gmail.com>
	<CAAuJ8MDdz-EoaLRzR2es+aH0tURL9kN0K+6bQcKEpP_RJ0BEGw@mail.gmail.com>
Message-ID: <CAA8_NKC1Fgtw9vBEKVffYG9NHO9toOMEkcA0qto03Ttn1UJ7JA@mail.gmail.com>

Hi Dan,

We're currently debugging this also with Snom 3XX/8XX on Draytek 2820 routers:

http://jira.freeswitch.org/browse/FS-3552

On 9 September 2011 00:17, Dan Lan <danlanweb at gmail.com> wrote:
> 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
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk


From brad at tech21.com  Tue Sep 13 00:44:33 2011
From: brad at tech21.com (Brad Mina)
Date: Mon, 12 Sep 2011 13:44:33 -0700
Subject: [Freeswitch-users] Cut PCAP file?
In-Reply-To: <CAKzWOxXK5EdrCjRGHyyecMXPnHDdC7hS3Y4OZ82eKxjNrP3Jjw@mail.gmail.com>
References: <CALNZuw==7kXb7zZgJYDjb8ey+47J0jK9dVHLTMX2Q7w2H+8ROw@mail.gmail.com>
	<CABHdwCr9+kieEKOfoG3akFAMV5+ZJfzWTBS5V71PgvODMhkO7w@mail.gmail.com>
	<CALNZuwmA1CDzRT3BDNc01iO0Kq8k+z9AqsmK3_dUDQpQPp28bg@mail.gmail.com>
	<CAKzWOxXK5EdrCjRGHyyecMXPnHDdC7hS3Y4OZ82eKxjNrP3Jjw@mail.gmail.com>
Message-ID: <CAGRr058HeK16YLAgjw1kZp6g+8L-e_eoyeEMkO25muxik7M7nQ@mail.gmail.com>

+1 !

On Mon, Sep 12, 2011 at 9:46 AM, Michael Collins <msc at freeswitch.org> wrote:

> And you were so proud of finding this that you put it on the wiki also? ;)
> -MC
>
>
> On Sun, Sep 11, 2011 at 9:11 AM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> 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
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From buscom123+fs at gmail.com  Tue Sep 13 02:53:25 2011
From: buscom123+fs at gmail.com (R H)
Date: Mon, 12 Sep 2011 16:53:25 -0600
Subject: [Freeswitch-users] problem using mod_managed under linux
In-Reply-To: <20320.1314825366@ccs.covici.com>
References: <rjxwrb84m9olrbg6m7ghyb97.1314822061391@email.android.com>
	<20320.1314825366@ccs.covici.com>
Message-ID: <CAAtQP55W982E1NvDYYZ5_moKk9aGdddU2j4pDdR-Ncy5JodYTg@mail.gmail.com>

Hey Guys,

I just noticed this thread in the email queue so this may be coming too
late. I have the same problem and sent an email about it to this group with
the subject line: *Mod_Managed Mono is not following the system LD Path.*
*
*
That said,
I have mod_managed working but it is a bit of a hack to do so. I simply CD
into the mod directory and then start the server from there.

EG.

cd /usr/local/freeswitch/mod
../bin/freeswitch

This makes that particular problem go away. I have tried changing the LD
Path, configured mono per the instructions on the wiki, etc? None of that
has worked for me but cd'ing into the mod directory and then starting the
server at least gets me running.

This might be useful to everyone while we wait for this to be resolved. I
have yet to add this as a bug in Jira. I was attempting to get some response
here before doing so.

Ryan

On Wed, Aug 31, 2011 at 3:16 PM, <covici at ccs.covici.com> wrote:

> 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.
>
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From nyilmaz at cybetech.com  Tue Sep 13 01:24:04 2011
From: nyilmaz at cybetech.com (Nihat Yilmaz)
Date: Tue, 13 Sep 2011 00:24:04 +0300
Subject: [Freeswitch-users] PRI Signalling Status down! (lakshmanan
	ganapathy)
In-Reply-To: <mailman.3143.1315806742.29749.freeswitch-users@lists.freeswitch.org>
References: <mailman.3143.1315806742.29749.freeswitch-users@lists.freeswitch.org>
Message-ID: <201109130024.04460.nyilmaz@cybetech.com>

Hi,

Thanks a lot for your suggestions. I tried CRC4. If it doesn't match the port 
is also down. I have also tried both cpe and net and they both give the same 
result. 

I would appreciate any other suggestions.

Regards,

Nihat


On Monday, September 12, 2011 08:52:22 AM 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..."
> 


From msc at freeswitch.org  Tue Sep 13 04:18:45 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 12 Sep 2011 17:18:45 -0700
Subject: [Freeswitch-users] external sip profile
In-Reply-To: <05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com>
References: <D9A75749-55E5-4DC5-A401-5F887ABEDE1A@lyonl.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper>
	<CAHKqMb4fp1jBxkjGAFVifBb6-MJxM9p01oiKzyqo8kaTQ7v2Vw@mail.gmail.com>
	<05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com>
Message-ID: <CAKzWOxX5f_Cmfm8FRX6N2q-GQhBgPDpn3RYeJTu7F_1zRofkog@mail.gmail.com>

Can you confirm if you have one-way audio? That is, an echo test won't tell
you if you have one-way audio. A simple way to do this is to use the record
app. It will play a file (whatever you choose, like "record at the tone..."
- there are tons of sound files you can try) and then record the audio from
the caller. Another test to do is to get a pcap of that call so you can
analyze it in wireshark. If you have RTP going in both directions to/from
the FS box then that indicates a NAT issue...

-MC

On Sun, Sep 11, 2011 at 7:00 PM, Chad Vogel <cvogel at lyonl.com> wrote:

>  Level 3 uses port 5070 for their sip server but requires port 5060 to be
> used on our side; I made changes to the vars.xml config file to support
> this. I was just able to get the server to answer a call by changing the
> register value to false and adding an entry in the ACL config for the Level
> 3 server however now it seems that the audio isn't working correctly. I'm
> using a simple dial plan to echo the audio back but all I get is dead air.
>
>       <extension name="level3">
>        <condition field="destination_number" expression="^13127567000$">
>          <action application="answer"/>
>          <action application="echo" />
>        </condition>
>      </extension>
>
>
>
>  On Sep 11, 2011, at 7:47 PM, Nandy Dagondon wrote:
>
> in addition to peter's advise, take a look at the SIP port 5070. FS is
> using port 5080 for the external SIP profile. modify the port number at
> "external.xml" then delete the port numbers in your proxy settings.
>
> On Sun, Sep 11, 2011 at 6:36 PM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
>> 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!
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From covici at ccs.covici.com  Tue Sep 13 04:26:01 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 12 Sep 2011 20:26:01 -0400
Subject: [Freeswitch-users] problem using mod_managed under linux
In-Reply-To: <CAAtQP55W982E1NvDYYZ5_moKk9aGdddU2j4pDdR-Ncy5JodYTg@mail.gmail.com>
References: <rjxwrb84m9olrbg6m7ghyb97.1314822061391@email.android.com>
	<20320.1314825366@ccs.covici.com>
	<CAAtQP55W982E1NvDYYZ5_moKk9aGdddU2j4pDdR-Ncy5JodYTg@mail.gmail.com>
Message-ID: <15565.1315873561@ccs.covici.com>

Well, that made it work -- after a make reswig and a reinstall.  I have
run into a problem where the fact that I have the libc.so link in my
/lib64, causes some programs which have nothing to do with freeswitch
not to compile and the only solution is to remove the link before
compiling them.

Thanks.

R H <buscom123+fs at gmail.com> wrote:

> Hey Guys,
> 
> I just noticed this thread in the email queue so this may be coming too
> late. I have the same problem and sent an email about it to this group with
> the subject line: *Mod_Managed Mono is not following the system LD Path.*
> *
> *
> That said,
> I have mod_managed working but it is a bit of a hack to do so. I simply CD
> into the mod directory and then start the server from there.
> 
> EG.
> 
> cd /usr/local/freeswitch/mod
> ../bin/freeswitch
> 
> This makes that particular problem go away. I have tried changing the LD
> Path, configured mono per the instructions on the wiki, etc? None of that
> has worked for me but cd'ing into the mod directory and then starting the
> server at least gets me running.
> 
> This might be useful to everyone while we wait for this to be resolved. I
> have yet to add this as a bug in Jira. I was attempting to get some response
> here before doing so.
> 
> Ryan
> 
> On Wed, Aug 31, 2011 at 3:16 PM, <covici at ccs.covici.com> wrote:
> 
> > 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.
> >
> 
> ----------------------------------------------------
> 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 gabe at gundy.org  Tue Sep 13 05:29:47 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 12 Sep 2011 19:29:47 -0600
Subject: [Freeswitch-users] Timeout Message or something similar
In-Reply-To: <1315837848697-6783735.post@n2.nabble.com>
References: <1315837848697-6783735.post@n2.nabble.com>
Message-ID: <CAMwZ-tK6GT7EdtCFdZqNTahfGp1BHt45jBUXt68eGfbKv7TBig@mail.gmail.com>

On Mon, Sep 12, 2011 at 8:30 AM, Brtantunes <brtantunes at gmail.com> wrote:
> I want to create a bridge between two users, something like this:
>
> originate {ringback=c:/teste.wav}sofia/DOMAIN/1001
> &bridge(sofia/DOMAIN/1002)
>
> This works fine. But now i want to add a timeout message if the second user
> don't answer (a sound played to the first user cause of the no_answer of se
> second user). Is that possible?

http://wiki.freeswitch.org/wiki/Mod_commands#originate

Note the timeout option. The args are positional, so because
timeout_sec is last, you have to provide all of the other options.

Then have a look at something like failover:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_7:_Action_failover_on_failed_action

I don't know if that gets you all the way there, but it should get you
close (or at least get you pointed in the right direction).


Gabe


From gabe at gundy.org  Tue Sep 13 05:36:35 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 12 Sep 2011 19:36:35 -0600
Subject: [Freeswitch-users] Codec Mis Match Issue
In-Reply-To: <BAY167-W107FFB0EB4D315435F6BE0CEC020@phx.gbl>
References: <BAY167-W107FFB0EB4D315435F6BE0CEC020@phx.gbl>
Message-ID: <CAMwZ-tLDLys3-JbiFEM=mM3nPabuizyUENmK66crrCKt6tk0ZQ@mail.gmail.com>

On Mon, Sep 12, 2011 at 5:16 AM, Faisal Rehman
<faisal.rehman22 at hotmail.com> wrote:
> I have a problem of some codec mis-match issue while making calls with
> freeswitch, the calls coming are all with only one codec G711 and I am
> amazed with the strange behaviour of Free-Switch
> that some times it responds the calls with the same codec and making them
> successful but numerous times it responds with G729 & hence calls got failed
> due to this. I have disabled the G729 codec in
> all my configuration but I don't know where is it coming from.

Have you read this yet?
http://wiki.freeswitch.org/wiki/Codec_negotiation

That's a good start.  If we're going to help you beyond this, we're
going to need more information about what you're actually seeing.

Gabe


From mstockton at harqen.com  Tue Sep 13 05:52:57 2011
From: mstockton at harqen.com (Matt Stockton)
Date: Mon, 12 Sep 2011 20:52:57 -0500
Subject: [Freeswitch-users] FreeSWITCH failover/HA
In-Reply-To: <CAC-LwPOenhYwEr7Gs9VYYKNgnjm8PQHyYCqD11WxDY-707qW=Q@mail.gmail.com>
References: <CAJWEF2ZPcrVirCFKCtHOugntRZ5sETDpDPCZyTRNso8OKMaRJA@mail.gmail.com>
	<CAHMdK5nhYOixxkh2DSrf0qQGmc8LLhFbZsTuq_pRGgWGgM-t8w@mail.gmail.com>
	<CAC-LwPOenhYwEr7Gs9VYYKNgnjm8PQHyYCqD11WxDY-707qW=Q@mail.gmail.com>
Message-ID: <CAHMdK5kFb88=EwEernGAngDKu5MA1u-EPrfcKYGsw3kDkmJBmQ@mail.gmail.com>

Sorry, this was confusing, I should have clarified. The way I use freeswitch
is via the event socket, so I don't have any UAs, just applications that
make calls via ESL.

On Fri, Sep 9, 2011 at 7:23 AM, vip killa <vipkilla at gmail.com> wrote:

> - 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
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From covici at ccs.covici.com  Tue Sep 13 06:01:13 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 12 Sep 2011 22:01:13 -0400
Subject: [Freeswitch-users] problem using mod_managed under linux
In-Reply-To: <CAAtQP55W982E1NvDYYZ5_moKk9aGdddU2j4pDdR-Ncy5JodYTg@mail.gmail.com>
References: <rjxwrb84m9olrbg6m7ghyb97.1314822061391@email.android.com>
	<20320.1314825366@ccs.covici.com>
	<CAAtQP55W982E1NvDYYZ5_moKk9aGdddU2j4pDdR-Ncy5JodYTg@mail.gmail.com>
Message-ID: <30786.1315879273@ccs.covici.com>

Well, I spoke too soon.  It works, but if I make a change and try to
reload the .dll, it kills fs altogether -- makes it not too usable.

R H <buscom123+fs at gmail.com> wrote:

> Hey Guys,
> 
> I just noticed this thread in the email queue so this may be coming too
> late. I have the same problem and sent an email about it to this group with
> the subject line: *Mod_Managed Mono is not following the system LD Path.*
> *
> *
> That said,
> I have mod_managed working but it is a bit of a hack to do so. I simply CD
> into the mod directory and then start the server from there.
> 
> EG.
> 
> cd /usr/local/freeswitch/mod
> ../bin/freeswitch
> 
> This makes that particular problem go away. I have tried changing the LD
> Path, configured mono per the instructions on the wiki, etc? None of that
> has worked for me but cd'ing into the mod directory and then starting the
> server at least gets me running.
> 
> This might be useful to everyone while we wait for this to be resolved. I
> have yet to add this as a bug in Jira. I was attempting to get some response
> here before doing so.
> 
> Ryan
> 
> On Wed, Aug 31, 2011 at 3:16 PM, <covici at ccs.covici.com> wrote:
> 
> > 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.
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From lakindia89 at gmail.com  Tue Sep 13 08:06:27 2011
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 13 Sep 2011 09:36:27 +0530
Subject: [Freeswitch-users] PRI Signalling Status down! (lakshmanan
	ganapathy)
In-Reply-To: <201109130024.04460.nyilmaz@cybetech.com>
References: <mailman.3143.1315806742.29749.freeswitch-users@lists.freeswitch.org>
	<201109130024.04460.nyilmaz@cybetech.com>
Message-ID: <CAMx6C7cXpwSynB_1vLSef08S1KQvVqSGsTyD7y+FM1uvU=jwWg@mail.gmail.com>

Try the following configuation
system.conf

span=1,1,0,ccs,hdb3
# termtype: te
bchan=1-15,17-31
dchan=16

fretdm.conf.xml
<configuration name="freetdm.conf" description="Freetdm Configuration">
  <settings>
    <param name="debug" value="0"/>
    <param name="hold-music" value="$${hold_music}"/>
  </settings>
 <libpri_spans>
 <span name="wp1">
     <param name="node" value="cpe"/>
     <param name="switch" value="euroisdn"/>
     <param name="dialplan" value="XML"/>
     <param name="context" value="default"/>
   </span>
</libpri_spans>
</configuration>


On Tue, Sep 13, 2011 at 2:54 AM, Nihat Yilmaz <nyilmaz at cybetech.com> wrote:

> Hi,
>
> Thanks a lot for your suggestions. I tried CRC4. If it doesn't match the
> port
> is also down. I have also tried both cpe and net and they both give the
> same
> result.
>
> I would appreciate any other suggestions.
>
> Regards,
>
> Nihat
>
>
> On Monday, September 12, 2011 08:52:22 AM 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..."
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From wasim at convergence.pk  Tue Sep 13 08:12:36 2011
From: wasim at convergence.pk (Wasim Baig)
Date: Tue, 13 Sep 2011 09:12:36 +0500
Subject: [Freeswitch-users] Cut PCAP file?
In-Reply-To: <CAKzWOxXK5EdrCjRGHyyecMXPnHDdC7hS3Y4OZ82eKxjNrP3Jjw@mail.gmail.com>
References: <CALNZuw==7kXb7zZgJYDjb8ey+47J0jK9dVHLTMX2Q7w2H+8ROw@mail.gmail.com>
	<CABHdwCr9+kieEKOfoG3akFAMV5+ZJfzWTBS5V71PgvODMhkO7w@mail.gmail.com>
	<CALNZuwmA1CDzRT3BDNc01iO0Kq8k+z9AqsmK3_dUDQpQPp28bg@mail.gmail.com>
	<CAKzWOxXK5EdrCjRGHyyecMXPnHDdC7hS3Y4OZ82eKxjNrP3Jjw@mail.gmail.com>
Message-ID: <CABHdwCprY2cGDoOH4ONVqai6r2b8NKdBJtLrO+fbHrH1tdEPqQ@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Packet_Capture#Edit_.2F_Cut_a_large_pcap_file_into_smaller_chunks

On Mon, Sep 12, 2011 at 21:46, Michael Collins <msc at freeswitch.org> wrote:

> And you were so proud of finding this that you put it on the wiki also? ;)
> -MC
>
>
> On Sun, Sep 11, 2011 at 9:11 AM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> 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
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ...
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From Glen.Ganderton at premier.com.au  Tue Sep 13 09:29:37 2011
From: Glen.Ganderton at premier.com.au (Glen Ganderton)
Date: Tue, 13 Sep 2011 15:29:37 +1000
Subject: [Freeswitch-users] Speech Input Timeout issue
Message-ID: <CD5A1499C8DEF34FAA3509722B00393E74029E3090@m-mail03.office.premiertech>

Hi Guys,

I am in the process of configuring an speech recognition IVR using freeswitch and nuance, with the module unimrcp to link them.  This seems to be working fine, I have create a basic IVR script that works well however  around 50% - 60% of calls fail with some sort of input timeout issue.

Here is a few relevant lines of the log file for a successful call:

2011-09-11 09:07:18.017882 [DEBUG] mrcp_stream.c:382 Parsed MRCP Message [70]
2011-09-11 09:07:18.017882 [DEBUG] mrcp_client.c:1104 Signal Connection Task Message
2011-09-11 09:07:18.017882 [DEBUG] mrcp_client.c:974 Receive Connection Task Message [3]
2011-09-11 09:07:18.017882 [INFO] mrcp_client_session.c:504 Raise App MRCP Response 0xb7252778 <160>
2011-09-11 09:07:18.017882 [DEBUG] mod_unimrcp.c:3094 (ASR-16) RECOGNIZE IN PROGRESS
2011-09-11 09:07:18.017882 [DEBUG] mod_unimrcp.c:1488 (ASR-16) READY ==> PROCESSING
2011-09-11 09:07:18.017882 [DEBUG] apt_consumer_task.c:90 Wait for Task Messages [MRCP Client]
2011-09-11 09:07:18.017882 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/1000 at 10.3.5.80
2011-09-11 09:07:18.690084 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed 0xb7252d18 [1000]
2011-09-11 09:07:18.690084 [DEBUG] mpf_timer_manager.c:111 Set Timer 0xb7252d18 [2000]
2011-09-11 09:07:19.446976 [INFO] mrcp_client_connection.c:525 Receive MRCPv2 Stream 10.3.5.80:49844 <-> 10.3.15.180:6075 [136 bytes]
MRCP/2.0 136 START-OF-INPUT 2 IN-PROGRESS
Channel-Identifier: 160 at speechrecog
Proxy-Sync-Id: 0-160 at speechrecog
Input-Type: speech


Here are a few lines for an unsuccessful call:
2011-09-11 09:07:44.605658 [DEBUG] mrcp_stream.c:382 Parsed MRCP Message [70]
2011-09-11 09:07:44.605658 [DEBUG] mrcp_client.c:1104 Signal Connection Task Message
2011-09-11 09:07:44.606901 [DEBUG] mrcp_client.c:974 Receive Connection Task Message [3]
2011-09-11 09:07:44.606901 [INFO] mrcp_client_session.c:504 Raise App MRCP Response 0xb72d8228 <161>
2011-09-11 09:07:44.606901 [DEBUG] mod_unimrcp.c:3094 (ASR-17) RECOGNIZE IN PROGRESS
2011-09-11 09:07:44.606901 [DEBUG] mod_unimrcp.c:1488 (ASR-17) READY ==> PROCESSING
2011-09-11 09:07:44.606901 [DEBUG] apt_consumer_task.c:90 Wait for Task Messages [MRCP Client]
2011-09-11 09:07:44.606901 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/1000 at 10.3.5.80
2011-09-11 09:07:44.814845 [INFO] mrcp_client_connection.c:525 Receive MRCPv2 Stream 10.3.5.80:49845 <-> 10.3.15.180:6075 [141 bytes]
MRCP/2.0 141 RECOGNITION-COMPLETE 2 COMPLETE
Channel-Identifier: 161 at speechrecog
Waveform-URI:
Completion-Cause: 002 no-input-timeout

It seems to me the main difference is that in the successful call there is a timer that is set. Has anybody seen this issue or have any idea's on how to resolve it?


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From Glen.Ganderton at premier.com.au  Tue Sep 13 09:48:54 2011
From: Glen.Ganderton at premier.com.au (Glen Ganderton)
Date: Tue, 13 Sep 2011 15:48:54 +1000
Subject: [Freeswitch-users] Speech Input Timeout issue
In-Reply-To: <CD5A1499C8DEF34FAA3509722B00393E74029E3090@m-mail03.office.premiertech>
References: <CD5A1499C8DEF34FAA3509722B00393E74029E3090@m-mail03.office.premiertech>
Message-ID: <CD5A1499C8DEF34FAA3509722B00393E74029E309E@m-mail03.office.premiertech>

Just to add one more thing, when I was simply using the XML dialplan I didn't seem to have this issue. It may be an issue with my lua ivr script. Im just learning to program in lua and have hacked something together. So if you want to have a look maybe you will see an issue with it.

http://pastebin.com/Bagn8ESS



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Glen Ganderton
Sent: Tuesday, 13 September 2011 3:30 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Speech Input Timeout issue

Hi Guys,

I am in the process of configuring an speech recognition IVR using freeswitch and nuance, with the module unimrcp to link them.  This seems to be working fine, I have create a basic IVR script that works well however  around 50% - 60% of calls fail with some sort of input timeout issue.

Here is a few relevant lines of the log file for a successful call:

2011-09-11 09:07:18.017882 [DEBUG] mrcp_stream.c:382 Parsed MRCP Message [70]
2011-09-11 09:07:18.017882 [DEBUG] mrcp_client.c:1104 Signal Connection Task Message
2011-09-11 09:07:18.017882 [DEBUG] mrcp_client.c:974 Receive Connection Task Message [3]
2011-09-11 09:07:18.017882 [INFO] mrcp_client_session.c:504 Raise App MRCP Response 0xb7252778 <160>
2011-09-11 09:07:18.017882 [DEBUG] mod_unimrcp.c:3094 (ASR-16) RECOGNIZE IN PROGRESS
2011-09-11 09:07:18.017882 [DEBUG] mod_unimrcp.c:1488 (ASR-16) READY ==> PROCESSING
2011-09-11 09:07:18.017882 [DEBUG] apt_consumer_task.c:90 Wait for Task Messages [MRCP Client]
2011-09-11 09:07:18.017882 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/1000 at 10.3.5.80<mailto:sofia/internal/1000 at 10.3.5.80>
2011-09-11 09:07:18.690084 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed 0xb7252d18 [1000]
2011-09-11 09:07:18.690084 [DEBUG] mpf_timer_manager.c:111 Set Timer 0xb7252d18 [2000]
2011-09-11 09:07:19.446976 [INFO] mrcp_client_connection.c:525 Receive MRCPv2 Stream 10.3.5.80:49844 <-> 10.3.15.180:6075 [136 bytes]
MRCP/2.0 136 START-OF-INPUT 2 IN-PROGRESS
Channel-Identifier: 160 at speechrecog
Proxy-Sync-Id: 0-160 at speechrecog
Input-Type: speech


Here are a few lines for an unsuccessful call:
2011-09-11 09:07:44.605658 [DEBUG] mrcp_stream.c:382 Parsed MRCP Message [70]
2011-09-11 09:07:44.605658 [DEBUG] mrcp_client.c:1104 Signal Connection Task Message
2011-09-11 09:07:44.606901 [DEBUG] mrcp_client.c:974 Receive Connection Task Message [3]
2011-09-11 09:07:44.606901 [INFO] mrcp_client_session.c:504 Raise App MRCP Response 0xb72d8228 <161>
2011-09-11 09:07:44.606901 [DEBUG] mod_unimrcp.c:3094 (ASR-17) RECOGNIZE IN PROGRESS
2011-09-11 09:07:44.606901 [DEBUG] mod_unimrcp.c:1488 (ASR-17) READY ==> PROCESSING
2011-09-11 09:07:44.606901 [DEBUG] apt_consumer_task.c:90 Wait for Task Messages [MRCP Client]
2011-09-11 09:07:44.606901 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/1000 at 10.3.5.80<mailto:sofia/internal/1000 at 10.3.5.80>
2011-09-11 09:07:44.814845 [INFO] mrcp_client_connection.c:525 Receive MRCPv2 Stream 10.3.5.80:49845 <-> 10.3.15.180:6075 [141 bytes]
MRCP/2.0 141 RECOGNITION-COMPLETE 2 COMPLETE
Channel-Identifier: 161 at speechrecog
Waveform-URI:
Completion-Cause: 002 no-input-timeout

It seems to me the main difference is that in the successful call there is a timer that is set. Has anybody seen this issue or have any idea's on how to resolve it?


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From avi at avimarcus.net  Tue Sep 13 10:14:20 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 13 Sep 2011 09:14:20 +0300
Subject: [Freeswitch-users] Cut PCAP file?
In-Reply-To: <CABHdwCprY2cGDoOH4ONVqai6r2b8NKdBJtLrO+fbHrH1tdEPqQ@mail.gmail.com>
References: <CALNZuw==7kXb7zZgJYDjb8ey+47J0jK9dVHLTMX2Q7w2H+8ROw@mail.gmail.com>
	<CABHdwCr9+kieEKOfoG3akFAMV5+ZJfzWTBS5V71PgvODMhkO7w@mail.gmail.com>
	<CALNZuwmA1CDzRT3BDNc01iO0Kq8k+z9AqsmK3_dUDQpQPp28bg@mail.gmail.com>
	<CAKzWOxXK5EdrCjRGHyyecMXPnHDdC7hS3Y4OZ82eKxjNrP3Jjw@mail.gmail.com>
	<CABHdwCprY2cGDoOH4ONVqai6r2b8NKdBJtLrO+fbHrH1tdEPqQ@mail.gmail.com>
Message-ID: <CALNZuw=v0HhkRuwin6zmhA2RaU0dhekS1zNNgwfGpZrtRTEu1Q@mail.gmail.com>

Heh. I already added it 7 hours before.. merging both.
-Avi


On Tue, Sep 13, 2011 at 7:12 AM, Wasim Baig <wasim at convergence.pk> wrote:

>
> http://wiki.freeswitch.org/wiki/Packet_Capture#Edit_.2F_Cut_a_large_pcap_file_into_smaller_chunks
>
>
> On Mon, Sep 12, 2011 at 21:46, Michael Collins <msc at freeswitch.org> wrote:
>
>> And you were so proud of finding this that you put it on the wiki also? ;)
>> -MC
>>
>>
>> On Sun, Sep 11, 2011 at 9:11 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> 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
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>>>>
>>>
>>>
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>
>
> --
> 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
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From cvogel at lyonl.com  Tue Sep 13 05:31:01 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Tue, 13 Sep 2011 01:31:01 +0000
Subject: [Freeswitch-users] external sip profile
In-Reply-To: <CAKzWOxX5f_Cmfm8FRX6N2q-GQhBgPDpn3RYeJTu7F_1zRofkog@mail.gmail.com>
References: <D9A75749-55E5-4DC5-A401-5F887ABEDE1A@lyonl.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper>
	<CAHKqMb4fp1jBxkjGAFVifBb6-MJxM9p01oiKzyqo8kaTQ7v2Vw@mail.gmail.com>
	<05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com>
	<CAKzWOxX5f_Cmfm8FRX6N2q-GQhBgPDpn3RYeJTu7F_1zRofkog@mail.gmail.com>
Message-ID: <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com>

I was able to confirm that i have one way and two way audio now (I can see it in wireshark), it seems like the echo application doesn't work correctly, it will not send out audio. when i use <action application="delay_echo" data="1000"/> in the dialplan it does seems to send out rtc traffic however the audio is static noise. when i use <action application="echo" />  it doest seem to work. any thoughts on why cant I echo test and why delay_echo sends out static noise. I tried to use the record app it creates a wav file of 44 KB but doesn't grow beyond.


On Sep 12, 2011, at 7:18 PM, Michael Collins wrote:

Can you confirm if you have one-way audio? That is, an echo test won't tell you if you have one-way audio. A simple way to do this is to use the record app. It will play a file (whatever you choose, like "record at the tone..." - there are tons of sound files you can try) and then record the audio from the caller. Another test to do is to get a pcap of that call so you can analyze it in wireshark. If you have RTP going in both directions to/from the FS box then that indicates a NAT issue...

-MC

On Sun, Sep 11, 2011 at 7:00 PM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:
Level 3 uses port 5070 for their sip server but requires port 5060 to be used on our side; I made changes to the vars.xml config file to support this. I was just able to get the server to answer a call by changing the register value to false and adding an entry in the ACL config for the Level 3 server however now it seems that the audio isn't working correctly. I'm using a simple dial plan to echo the audio back but all I get is dead air.

    <extension name="level3">
      <condition field="destination_number" expression="^13127567000$">
        <action application="answer"/>
        <action application="echo" />
      </condition>
    </extension>



On Sep 11, 2011, at 7:47 PM, Nandy Dagondon wrote:

in addition to peter's advise, take a look at the SIP port 5070. FS is using port 5080 for the external SIP profile. modify the port number at "external.xml" then delete the port numbers in your proxy settings.

On Sun, Sep 11, 2011 at 6:36 PM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
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<mailto:freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] f&#246;r Chad Vogel [cvogel at lyonl.com<mailto:cvogel at lyonl.com>]
Skickat: den 11 september 2011 03:28
Till: freeswitch-users at lists.freeswitch.org<mailto: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<http://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!


FreeSWITCH-users mailing list
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From ocset at the800group.com  Tue Sep 13 10:49:30 2011
From: ocset at the800group.com (ocset)
Date: Tue, 13 Sep 2011 14:49:30 +0800
Subject: [Freeswitch-users] GXW4104 gateway setup for outgoing calls
Message-ID: <4E6EFCFA.6000803@the800group.com>

Hi

I have recently bought a Grandstream GXW4104 (4 FXO ports) and need some 
help setting up a gateway to call out using the GXW4104. I am really out 
of my depth here and may be looking at this the wrong way so please bear 
with me.

I followed the advice on this 
website "http://www.timhunt.net/wiki/FreeSwitch:GXW4104" and incoming 
calls from a PSTN line are working great. Now I need to setup a dialplan 
so that outgoing calls are routed through the same PSTN line on the 
GXW4104. I will eventually have 4 PSTN lines with a dialplan to use the 
first available line (if that is possible).

According to the FreeSWITCH 1.0.6 book (and many online posts) I need to 
create a gateway and a dialplan but all the gateway examples are for SIP 
accounts.

So, the gateway definition seems to need a username and password but the 
GXW4104 does not have that capability. I found this gateway definition 
in the  freeswitch.xml.fsxml file but am not sure how many of these 
variables are required.

<gateways>
<gateway name="example.com">
<param name="username" value="joeuser"/>
<param name="password" value="password"/>
<param name="from-user" value="joeuser"/>
<param name="from-domain" value="example.com"/>
<param name="expire-seconds" value="600"/>
<param name="register" value="false"/>
<param name="retry-seconds" value="30"/>
<param name="extension" value="5000"/>
<param name="context" value="public"/>
</gateway>
</gateways>


If I define a gateway called "gxw4104", then this is what I think a 
simple dialplan should look like but I'm not sure of the gateway details 
in the "bridge" section of the definition.

<extension name="gxw4104_out">
<condition field="destination_number" expression="^(\d{10})$">
*<action application="bridge" 
data="sofia/gateway/gxw4104/........"/>        (what should this be???)*
</condition>
</extension>

Am I moving in the right direction and can someone fill in the blanks 
for me please

Thanks in advance!


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From Glen.Ganderton at premier.com.au  Tue Sep 13 11:14:21 2011
From: Glen.Ganderton at premier.com.au (Glen Ganderton)
Date: Tue, 13 Sep 2011 17:14:21 +1000
Subject: [Freeswitch-users] Speech Input Timeout issue
In-Reply-To: <CD5A1499C8DEF34FAA3509722B00393E74029E309E@m-mail03.office.premiertech>
References: <CD5A1499C8DEF34FAA3509722B00393E74029E3090@m-mail03.office.premiertech>
	<CD5A1499C8DEF34FAA3509722B00393E74029E309E@m-mail03.office.premiertech>
Message-ID: <CD5A1499C8DEF34FAA3509722B00393E74029E30CE@m-mail03.office.premiertech>

Just to add to this again, I also get this random error "RECOGNIZER channel error!" and the only way for ASR to start operating again is to restart freeswitch.

2011-09-11 11:45:42.931526 [DEBUG] apt_consumer_task.c:90 Wait for Task Messages [MRCP Client]
2011-09-11 11:45:42.931526 [INFO] mrcp_sofiasip_client_agent.c:513 Receive SIP Event [nua_i_state] Status 0 INVITE sent
2011-09-11 11:45:42.931526 [NOTICE] mrcp_sofiasip_client_agent.c:453 SIP Call State 0xb7154118 [calling]
2011-09-11 11:45:43.134239 [INFO] mrcp_sofiasip_client_agent.c:513 Receive SIP Event [nua_r_invite] Status 503 Service Unavailable
2011-09-11 11:45:43.134239 [INFO] mrcp_sofiasip_client_agent.c:513 Receive SIP Event [nua_i_state] Status 503 Service Unavailable
2011-09-11 11:45:43.134239 [NOTICE] mrcp_sofiasip_client_agent.c:453 SIP Call State 0xb7154118 [terminated]
2011-09-11 11:45:43.134239 [DEBUG] mrcp_client.c:1075 Signal Signaling Task Message
2011-09-11 11:45:43.134239 [DEBUG] mrcp_client.c:949 Receive Signaling Task Message [4]
2011-09-11 11:45:43.134239 [INFO] mrcp_client_session.c:535 Raise App Response 0xb7154118 <new> [2] FAILURE [2]
2011-09-11 11:45:43.134239 [ERR] mod_unimrcp.c:1795 (ASR-3) RECOGNIZER channel error!
2011-09-11 11:45:43.134239 [DEBUG] mod_unimrcp.c:1801 Terminating MRCP session
2011-09-11 11:45:43.134239 [DEBUG] mod_unimrcp.c:1488 (ASR-3) CLOSED ==> ERROR

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Glen Ganderton
Sent: Tuesday, 13 September 2011 3:49 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Speech Input Timeout issue

Just to add one more thing, when I was simply using the XML dialplan I didn't seem to have this issue. It may be an issue with my lua ivr script. Im just learning to program in lua and have hacked something together. So if you want to have a look maybe you will see an issue with it.

http://pastebin.com/Bagn8ESS



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Glen Ganderton
Sent: Tuesday, 13 September 2011 3:30 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Speech Input Timeout issue

Hi Guys,

I am in the process of configuring an speech recognition IVR using freeswitch and nuance, with the module unimrcp to link them.  This seems to be working fine, I have create a basic IVR script that works well however  around 50% - 60% of calls fail with some sort of input timeout issue.

Here is a few relevant lines of the log file for a successful call:

2011-09-11 09:07:18.017882 [DEBUG] mrcp_stream.c:382 Parsed MRCP Message [70]
2011-09-11 09:07:18.017882 [DEBUG] mrcp_client.c:1104 Signal Connection Task Message
2011-09-11 09:07:18.017882 [DEBUG] mrcp_client.c:974 Receive Connection Task Message [3]
2011-09-11 09:07:18.017882 [INFO] mrcp_client_session.c:504 Raise App MRCP Response 0xb7252778 <160>
2011-09-11 09:07:18.017882 [DEBUG] mod_unimrcp.c:3094 (ASR-16) RECOGNIZE IN PROGRESS
2011-09-11 09:07:18.017882 [DEBUG] mod_unimrcp.c:1488 (ASR-16) READY ==> PROCESSING
2011-09-11 09:07:18.017882 [DEBUG] apt_consumer_task.c:90 Wait for Task Messages [MRCP Client]
2011-09-11 09:07:18.017882 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/1000 at 10.3.5.80<mailto:sofia/internal/1000 at 10.3.5.80>
2011-09-11 09:07:18.690084 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed 0xb7252d18 [1000]
2011-09-11 09:07:18.690084 [DEBUG] mpf_timer_manager.c:111 Set Timer 0xb7252d18 [2000]
2011-09-11 09:07:19.446976 [INFO] mrcp_client_connection.c:525 Receive MRCPv2 Stream 10.3.5.80:49844 <-> 10.3.15.180:6075 [136 bytes]
MRCP/2.0 136 START-OF-INPUT 2 IN-PROGRESS
Channel-Identifier: 160 at speechrecog
Proxy-Sync-Id: 0-160 at speechrecog
Input-Type: speech


Here are a few lines for an unsuccessful call:
2011-09-11 09:07:44.605658 [DEBUG] mrcp_stream.c:382 Parsed MRCP Message [70]
2011-09-11 09:07:44.605658 [DEBUG] mrcp_client.c:1104 Signal Connection Task Message
2011-09-11 09:07:44.606901 [DEBUG] mrcp_client.c:974 Receive Connection Task Message [3]
2011-09-11 09:07:44.606901 [INFO] mrcp_client_session.c:504 Raise App MRCP Response 0xb72d8228 <161>
2011-09-11 09:07:44.606901 [DEBUG] mod_unimrcp.c:3094 (ASR-17) RECOGNIZE IN PROGRESS
2011-09-11 09:07:44.606901 [DEBUG] mod_unimrcp.c:1488 (ASR-17) READY ==> PROCESSING
2011-09-11 09:07:44.606901 [DEBUG] apt_consumer_task.c:90 Wait for Task Messages [MRCP Client]
2011-09-11 09:07:44.606901 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/1000 at 10.3.5.80<mailto:sofia/internal/1000 at 10.3.5.80>
2011-09-11 09:07:44.814845 [INFO] mrcp_client_connection.c:525 Receive MRCPv2 Stream 10.3.5.80:49845 <-> 10.3.15.180:6075 [141 bytes]
MRCP/2.0 141 RECOGNITION-COMPLETE 2 COMPLETE
Channel-Identifier: 161 at speechrecog
Waveform-URI:
Completion-Cause: 002 no-input-timeout

It seems to me the main difference is that in the successful call there is a timer that is set. Has anybody seen this issue or have any idea's on how to resolve it?


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From gcd at i.ph  Tue Sep 13 11:45:10 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 13 Sep 2011 15:45:10 +0800
Subject: [Freeswitch-users] GXW4104 gateway setup for outgoing calls
In-Reply-To: <4E6EFCFA.6000803@the800group.com>
References: <4E6EFCFA.6000803@the800group.com>
Message-ID: <CAHKqMb7vF4ULKjWsHqP20xVRy0oOrDo9A3fpwCBr2NXAjsJ6iw@mail.gmail.com>

hi,

if GWX4104 is in your local network, use the internal profile for the
gateway. register your FXO accounts to receive incoming calls (i think you
did this already).

to dialout the ports, specify the port number 5060~5063 assuming Port1
starts at 5060. to dialout via port4, the bridge data should look like:

7654321 at 192.168.0.9:5063

hope it helps.

-nandy


On Tue, Sep 13, 2011 at 2:49 PM, ocset <ocset at the800group.com> wrote:

>  Hi
>
> I have recently bought a Grandstream GXW4104 (4 FXO ports) and need some
> help setting up a gateway to call out using the GXW4104. I am really out of
> my depth here and may be looking at this the wrong way so please bear with
> me.
>
> I followed the advice on this website
> "http://www.timhunt.net/wiki/FreeSwitch:GXW4104"<http://www.timhunt.net/wiki/FreeSwitch:GXW4104>and incoming calls from a PSTN line are working great. Now I need to setup a
> dialplan so that outgoing calls are routed through the same PSTN line on the
> GXW4104. I will eventually have 4 PSTN lines with a dialplan to use the
> first available line (if that is possible).
>
> According to the FreeSWITCH 1.0.6 book (and many online posts) I need to
> create a gateway and a dialplan but all the gateway examples are for SIP
> accounts.
>
> So, the gateway definition seems to need a username and password but the
> GXW4104 does not have that capability. I found this gateway definition in
> the  freeswitch.xml.fsxml file but am not sure how many of these variables
> are required.
>
> <gateways>
>    <gateway name="example.com">
>       <param name="username" value="joeuser"/>
>       <param name="password" value="password"/>
>       <param name="from-user" value="joeuser"/>
>       <param name="from-domain" value="example.com"/>
>       <param name="expire-seconds" value="600"/>
>       <param name="register" value="false"/>
>       <param name="retry-seconds" value="30"/>
>       <param name="extension" value="5000"/>
>       <param name="context" value="public"/>
>    </gateway>
> </gateways>
>
>
> If I define a gateway called "gxw4104", then this is what I think a simple
> dialplan should look like but I'm not sure of the gateway details in the
> "bridge" section of the definition.
>
> <extension name="gxw4104_out">
>    <condition field="destination_number" expression="^(\d{10})$">
>       *<action application="bridge"
> data="sofia/gateway/gxw4104/........"/>        (what should this be???)*
>    </condition>
> </extension>
>
> Am I moving in the right direction and can someone fill in the blanks for
> me please
>
> Thanks in advance!
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From faisal.rehman22 at hotmail.com  Tue Sep 13 11:54:10 2011
From: faisal.rehman22 at hotmail.com (Faisal Rehman)
Date: Tue, 13 Sep 2011 13:54:10 +0600
Subject: [Freeswitch-users] Codec Mis Match Issue
In-Reply-To: <CAMwZ-tLDLys3-JbiFEM=mM3nPabuizyUENmK66crrCKt6tk0ZQ@mail.gmail.com>
References: <BAY167-W107FFB0EB4D315435F6BE0CEC020@phx.gbl>,
	<CAMwZ-tLDLys3-JbiFEM=mM3nPabuizyUENmK66crrCKt6tk0ZQ@mail.gmail.com>
Message-ID: <BAY167-W78987014339920558F203EC050@phx.gbl>


Hi Gabe,
Thanks for your assistance, yeah I have read that article & also it helped out me a lot & I have got my issue resolved to some extent doing late negotiation. I am still working on it, & if I'll need any help or assistance, I will definitely ask.
Thanks,
Faisal Rehman
> From: gabe at gundy.org
> Date: Mon, 12 Sep 2011 19:36:35 -0600
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Codec Mis Match Issue
> 
> On Mon, Sep 12, 2011 at 5:16 AM, Faisal Rehman
> <faisal.rehman22 at hotmail.com> wrote:
> > I have a problem of some codec mis-match issue while making calls with
> > freeswitch, the calls coming are all with only one codec G711 and I am
> > amazed with the strange behaviour of Free-Switch
> > that some times it responds the calls with the same codec and making them
> > successful but numerous times it responds with G729 & hence calls got failed
> > due to this. I have disabled the G729 codec in
> > all my configuration but I don't know where is it coming from.
> 
> Have you read this yet?
> http://wiki.freeswitch.org/wiki/Codec_negotiation
> 
> That's a good start.  If we're going to help you beyond this, we're
> going to need more information about what you're actually seeing.
> 
> Gabe
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
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From sid at eltc.ru  Tue Sep 13 12:03:33 2011
From: sid at eltc.ru (Sergey Scheglov)
Date: Tue, 13 Sep 2011 15:03:33 +0700
Subject: [Freeswitch-users] GXW4104 gateway setup for outgoing calls
In-Reply-To: <CAHKqMb7vF4ULKjWsHqP20xVRy0oOrDo9A3fpwCBr2NXAjsJ6iw@mail.gmail.com>
References: <4E6EFCFA.6000803@the800group.com>
	<CAHKqMb7vF4ULKjWsHqP20xVRy0oOrDo9A3fpwCBr2NXAjsJ6iw@mail.gmail.com>
Message-ID: <20110913150333.3be80d88@shadow>

? Tue, 13 Sep 2011 15:45:10 +0800
Nandy Dagondon <gcd at i.ph> wrote:

> hi,
> 
> if GWX4104 is in your local network, use the internal profile for the
> gateway. register your FXO accounts to receive incoming calls (i
> think you did this already).
> 
> to dialout the ports, specify the port number 5060~5063 assuming Port1
> starts at 5060. to dialout via port4, the bridge data should look
> like:
> 
> 7654321 at 192.168.0.9:5063
> 
> hope it helps.
> 
> -nandy
> 
> 
> On Tue, Sep 13, 2011 at 2:49 PM, ocset <ocset at the800group.com> wrote:
> 
> >  Hi
> >
> > I have recently bought a Grandstream GXW4104 (4 FXO ports) and need
> > some help setting up a gateway to call out using the GXW4104. I am
> > really out of my depth here and may be looking at this the wrong
> > way so please bear with me.
> >
> > I followed the advice on this website
> > "http://www.timhunt.net/wiki/FreeSwitch:GXW4104"<http://www.timhunt.net/wiki/FreeSwitch:GXW4104>and
> > incoming calls from a PSTN line are working great. Now I need to
> > setup a dialplan so that outgoing calls are routed through the same
> > PSTN line on the GXW4104. I will eventually have 4 PSTN lines with
> > a dialplan to use the first available line (if that is possible).
> >
> > According to the FreeSWITCH 1.0.6 book (and many online posts) I
> > need to create a gateway and a dialplan but all the gateway
> > examples are for SIP accounts.
> >
> > So, the gateway definition seems to need a username and password
> > but the GXW4104 does not have that capability. I found this gateway
> > definition in the  freeswitch.xml.fsxml file but am not sure how
> > many of these variables are required.
> >
> > <gateways>
> >    <gateway name="example.com">
> >       <param name="username" value="joeuser"/>
> >       <param name="password" value="password"/>
> >       <param name="from-user" value="joeuser"/>
> >       <param name="from-domain" value="example.com"/>
> >       <param name="expire-seconds" value="600"/>
> >       <param name="register" value="false"/>
> >       <param name="retry-seconds" value="30"/>
> >       <param name="extension" value="5000"/>
> >       <param name="context" value="public"/>
> >    </gateway>
> > </gateways>
> >
> >
> > If I define a gateway called "gxw4104", then this is what I think a
> > simple dialplan should look like but I'm not sure of the gateway
> > details in the "bridge" section of the definition.
> >
> > <extension name="gxw4104_out">
> >    <condition field="destination_number" expression="^(\d{10})$">
> >       *<action application="bridge"
> > data="sofia/gateway/gxw4104/........"/>        (what should this
> > be???)* </condition>
> > </extension>
> >
> > Am I moving in the right direction and can someone fill in the
> > blanks for me please
> >
> > 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
> >
> >

As far as I know GXW4104 is used by default the following ports: 5060,
5062, 5064, 5066.

-- 
Scheglov Sergey


From gcd at i.ph  Tue Sep 13 12:11:37 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 13 Sep 2011 16:11:37 +0800
Subject: [Freeswitch-users] GXW4104 gateway setup for outgoing calls
In-Reply-To: <20110913150333.3be80d88@shadow>
References: <4E6EFCFA.6000803@the800group.com>
	<CAHKqMb7vF4ULKjWsHqP20xVRy0oOrDo9A3fpwCBr2NXAjsJ6iw@mail.gmail.com>
	<20110913150333.3be80d88@shadow>
Message-ID: <CAHKqMb64CULESLwS5snj42omx+9g2TTHzAQEjbCsj6_enqtXgw@mail.gmail.com>

tks to ur input sergey.

On Tue, Sep 13, 2011 at 4:03 PM, Sergey Scheglov <sid at eltc.ru> wrote:

> ? Tue, 13 Sep 2011 15:45:10 +0800
> Nandy Dagondon <gcd at i.ph> wrote:
>
> > hi,
> >
> > if GWX4104 is in your local network, use the internal profile for the
> > gateway. register your FXO accounts to receive incoming calls (i
> > think you did this already).
> >
> > to dialout the ports, specify the port number 5060~5063 assuming Port1
> > starts at 5060. to dialout via port4, the bridge data should look
> > like:
> >
> > 7654321 at 192.168.0.9:5063
> >
> > hope it helps.
> >
> > -nandy
> >
> >
> > On Tue, Sep 13, 2011 at 2:49 PM, ocset <ocset at the800group.com> wrote:
> >
> > >  Hi
> > >
> > > I have recently bought a Grandstream GXW4104 (4 FXO ports) and need
> > > some help setting up a gateway to call out using the GXW4104. I am
> > > really out of my depth here and may be looking at this the wrong
> > > way so please bear with me.
> > >
> > > I followed the advice on this website
> > > "http://www.timhunt.net/wiki/FreeSwitch:GXW4104"<
> http://www.timhunt.net/wiki/FreeSwitch:GXW4104>and
> > > incoming calls from a PSTN line are working great. Now I need to
> > > setup a dialplan so that outgoing calls are routed through the same
> > > PSTN line on the GXW4104. I will eventually have 4 PSTN lines with
> > > a dialplan to use the first available line (if that is possible).
> > >
> > > According to the FreeSWITCH 1.0.6 book (and many online posts) I
> > > need to create a gateway and a dialplan but all the gateway
> > > examples are for SIP accounts.
> > >
> > > So, the gateway definition seems to need a username and password
> > > but the GXW4104 does not have that capability. I found this gateway
> > > definition in the  freeswitch.xml.fsxml file but am not sure how
> > > many of these variables are required.
> > >
> > > <gateways>
> > >    <gateway name="example.com">
> > >       <param name="username" value="joeuser"/>
> > >       <param name="password" value="password"/>
> > >       <param name="from-user" value="joeuser"/>
> > >       <param name="from-domain" value="example.com"/>
> > >       <param name="expire-seconds" value="600"/>
> > >       <param name="register" value="false"/>
> > >       <param name="retry-seconds" value="30"/>
> > >       <param name="extension" value="5000"/>
> > >       <param name="context" value="public"/>
> > >    </gateway>
> > > </gateways>
> > >
> > >
> > > If I define a gateway called "gxw4104", then this is what I think a
> > > simple dialplan should look like but I'm not sure of the gateway
> > > details in the "bridge" section of the definition.
> > >
> > > <extension name="gxw4104_out">
> > >    <condition field="destination_number" expression="^(\d{10})$">
> > >       *<action application="bridge"
> > > data="sofia/gateway/gxw4104/........"/>        (what should this
> > > be???)* </condition>
> > > </extension>
> > >
> > > Am I moving in the right direction and can someone fill in the
> > > blanks for me please
> > >
> > > 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
> > >
> > >
>
> As far as I know GXW4104 is used by default the following ports: 5060,
> 5062, 5064, 5066.
>
> --
> Scheglov Sergey
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From yehavi.bourvine at gmail.com  Tue Sep 13 12:56:03 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 13 Sep 2011 11:56:03 +0300
Subject: [Freeswitch-users] MySQL high CPU usage after Freeswitch update
Message-ID: <CAAtWKeF4uYbQuxtC9yAdheDaeLfe6kkJqczno0ujNTCX19gocQ@mail.gmail.com>

Hello,

  This mornning I've updated our Freeswitch from a 3 months old GIT to
yesterday's GIT (git-2fa8f11 2011-09-09 14-50-54 -0700). We are using MySQL
via ODBC, and I noticed that since today's mornning MySQL is eating 300% CPU
(I have 16 cores total).

Any idea how to locate the culprit?

BTW, I did "git clone" and then install fresh bin/mod/lib directories (after
deleting the previous content) just to make sure that I have no leftovers
from previous version.

                                  Thanks! __Yehavi:
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From covici at ccs.covici.com  Tue Sep 13 13:27:53 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 13 Sep 2011 05:27:53 -0400
Subject: [Freeswitch-users] GXW4104 gateway setup for outgoing calls
In-Reply-To: <CAHKqMb7vF4ULKjWsHqP20xVRy0oOrDo9A3fpwCBr2NXAjsJ6iw@mail.gmail.com>
References: <4E6EFCFA.6000803@the800group.com>
	<CAHKqMb7vF4ULKjWsHqP20xVRy0oOrDo9A3fpwCBr2NXAjsJ6iw@mail.gmail.com>
Message-ID: <8327.1315906073@ccs.covici.com>

Do you know if you can flash the fxo hook using that gateway?

Nandy Dagondon <gcd at i.ph> wrote:

> hi,
> 
> if GWX4104 is in your local network, use the internal profile for the
> gateway. register your FXO accounts to receive incoming calls (i think you
> did this already).
> 
> to dialout the ports, specify the port number 5060~5063 assuming Port1
> starts at 5060. to dialout via port4, the bridge data should look like:
> 
> 7654321 at 192.168.0.9:5063
> 
> hope it helps.
> 
> -nandy
> 
> 
> On Tue, Sep 13, 2011 at 2:49 PM, ocset <ocset at the800group.com> wrote:
> 
> >  Hi
> >
> > I have recently bought a Grandstream GXW4104 (4 FXO ports) and need some
> > help setting up a gateway to call out using the GXW4104. I am really out of
> > my depth here and may be looking at this the wrong way so please bear with
> > me.
> >
> > I followed the advice on this website
> > "http://www.timhunt.net/wiki/FreeSwitch:GXW4104"<http://www.timhunt.net/wiki/FreeSwitch:GXW4104>and incoming calls from a PSTN line are working great. Now I need to setup a
> > dialplan so that outgoing calls are routed through the same PSTN line on the
> > GXW4104. I will eventually have 4 PSTN lines with a dialplan to use the
> > first available line (if that is possible).
> >
> > According to the FreeSWITCH 1.0.6 book (and many online posts) I need to
> > create a gateway and a dialplan but all the gateway examples are for SIP
> > accounts.
> >
> > So, the gateway definition seems to need a username and password but the
> > GXW4104 does not have that capability. I found this gateway definition in
> > the  freeswitch.xml.fsxml file but am not sure how many of these variables
> > are required.
> >
> > <gateways>
> >    <gateway name="example.com">
> >       <param name="username" value="joeuser"/>
> >       <param name="password" value="password"/>
> >       <param name="from-user" value="joeuser"/>
> >       <param name="from-domain" value="example.com"/>
> >       <param name="expire-seconds" value="600"/>
> >       <param name="register" value="false"/>
> >       <param name="retry-seconds" value="30"/>
> >       <param name="extension" value="5000"/>
> >       <param name="context" value="public"/>
> >    </gateway>
> > </gateways>
> >
> >
> > If I define a gateway called "gxw4104", then this is what I think a simple
> > dialplan should look like but I'm not sure of the gateway details in the
> > "bridge" section of the definition.
> >
> > <extension name="gxw4104_out">
> >    <condition field="destination_number" expression="^(\d{10})$">
> >       *<action application="bridge"
> > data="sofia/gateway/gxw4104/........"/>        (what should this be???)*
> >    </condition>
> > </extension>
> >
> > Am I moving in the right direction and can someone fill in the blanks for
> > me please
> >
> > 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
> >
> >
> 
> ----------------------------------------------------
> 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 yehavi.bourvine at gmail.com  Tue Sep 13 13:31:00 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 13 Sep 2011 12:31:00 +0300
Subject: [Freeswitch-users] MySQL high CPU usage after Freeswitch update
In-Reply-To: <CAAtWKeF4uYbQuxtC9yAdheDaeLfe6kkJqczno0ujNTCX19gocQ@mail.gmail.com>
References: <CAAtWKeF4uYbQuxtC9yAdheDaeLfe6kkJqczno0ujNTCX19gocQ@mail.gmail.com>
Message-ID: <CAAtWKeFuANHnYMdpnq+13G0zN6FsthnbF1eafZjmcYassPvFiQ@mail.gmail.com>

The problem just went away, but while looking after it I've found something
odd: Using TCPDUMP on lo0 I see that there are about 150 MySQL queries per
second with the string "select 1". Searching the source code I found that
this is called from odbc_switch.c to check whether the database is up.
Are150 calls per second ok?

                           Thanks, __Yehavi:

2011/9/13 Yehavi Bourvine <yehavi.bourvine at gmail.com>

> Hello,
>
>   This mornning I've updated our Freeswitch from a 3 months old GIT to
> yesterday's GIT (git-2fa8f11 2011-09-09 14-50-54 -0700). We are using MySQL
> via ODBC, and I noticed that since today's mornning MySQL is eating 300% CPU
> (I have 16 cores total).
>
> Any idea how to locate the culprit?
>
> BTW, I did "git clone" and then install fresh bin/mod/lib
> directories (after deleting the previous content) just to make sure that I
> have no leftovers from previous version.
>
>                                   Thanks! __Yehavi:
>
>
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From chrisbware at interfree.it  Tue Sep 13 13:39:33 2011
From: chrisbware at interfree.it (chrisbware at interfree.it)
Date: 13 Sep 2011 09:39:33 -0000
Subject: [Freeswitch-users] Lua "two steps" bridging
Message-ID: <20110913093933.16714.qmail@community24.interfree.it>


Hi guys,

I've read anything on this argument on Wiki but I can't find a complete answer.

I need to bridge two calls in Lua, waiting the first to be answered.
My basic script is:

local GwParams = "origination_caller_id_number="..gateway..",sip_auth_username="..gateway..",sip_auth_password="..sip_passwd
local slegA = "[ignore_early_media=false,"..GwParams.."]sofia/external/"..callee.."@"..sip_domain
local slegB = "[ignore_early_media=true,"..GwParams.."]sofia/external/"..called.."@"..sip_domain

legA = freeswitch.Session(slegA);
legB = freeswitch.Session(slegB);
freeswitch.bridge(legA, legB);


It works but legA and legB are called at the same time. Using:

while (legA:answered()== false) do end;

between the two session do the job but called parties can't hear  audio.

I promise to add a script on wiki if you help me ! :)

Thanks in advance.


-------------------------------------------------------------------------------
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From peter.olsson at visionutveckling.se  Tue Sep 13 13:39:55 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 13 Sep 2011 11:39:55 +0200
Subject: [Freeswitch-users] MySQL high CPU usage after Freeswitch update
In-Reply-To: <CAAtWKeFuANHnYMdpnq+13G0zN6FsthnbF1eafZjmcYassPvFiQ@mail.gmail.com>
References: <CAAtWKeF4uYbQuxtC9yAdheDaeLfe6kkJqczno0ujNTCX19gocQ@mail.gmail.com>
	<CAAtWKeFuANHnYMdpnq+13G0zN6FsthnbF1eafZjmcYassPvFiQ@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F2212347@cooper>

I believe it's called every time before the "real" query it wants to execute - just to confirm that the database connection is up. So it probably means you've got lots of traffic, so it need to do lots of SQL updates.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Yehavi Bourvine
Skickat: den 13 september 2011 11:31
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] MySQL high CPU usage after Freeswitch update

The problem just went away, but while looking after it I've found something odd: Using TCPDUMP on lo0 I see that there are about 150 MySQL queries per second with the string "select 1". Searching the source code I found that this is called from odbc_switch.c to check whether the database is up. Are150 calls per second ok?

                           Thanks, __Yehavi:
2011/9/13 Yehavi Bourvine <yehavi.bourvine at gmail.com<mailto:yehavi.bourvine at gmail.com>>
Hello,

  This mornning I've updated our Freeswitch from a 3 months old GIT to yesterday's GIT (git-2fa8f11 2011-09-09 14-50-54 -0700). We are using MySQL via ODBC, and I noticed that since today's mornning MySQL is eating 300% CPU (I have 16 cores total).

Any idea how to locate the culprit?

BTW, I did "git clone" and then install fresh bin/mod/lib directories (after deleting the previous content) just to make sure that I have no leftovers from previous version.

                                  Thanks! __Yehavi:

!DSPAM:4e6f22de32761349872408!
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From brtantunes at gmail.com  Tue Sep 13 13:48:51 2011
From: brtantunes at gmail.com (Brtantunes)
Date: Tue, 13 Sep 2011 02:48:51 -0700 (PDT)
Subject: [Freeswitch-users] Timeout Message or something similar
In-Reply-To: <CAMwZ-tK6GT7EdtCFdZqNTahfGp1BHt45jBUXt68eGfbKv7TBig@mail.gmail.com>
References: <1315837848697-6783735.post@n2.nabble.com>
	<CAMwZ-tK6GT7EdtCFdZqNTahfGp1BHt45jBUXt68eGfbKv7TBig@mail.gmail.com>
Message-ID: <1315907331256-6786830.post@n2.nabble.com>

Hello Gabriel,

Thanks for your help. I'm new with freeswitch and didn't know about the
failover. 

i tried 
/originate
{ringback=c:/teste.wav,fail_on_single_reject=true,ignore_early_media=true,originate_continue_on_timeout=true}sofia/DOMAIN/1001
&bridge(sofia/DOMAIN/1002|sofia/DOMAIN/1003)/ and it works fine. 
Now the problem is on the failover pass the not the bridge to the other
number. I tried /originate
{ringback=c:/teste.wav,fail_on_single_reject=true,ignore_early_media=true,originate_continue_on_timeout=true}sofia/10.101.81.71/1001
&bridge(sofia/10.101.81.71/1002)|&playback(C:/teste2.wav)/ but obviously it
didn't work :(.

Any sugestions?

Tks

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Timeout-Message-or-something-similar-tp6783735p6786830.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From ocset at the800group.com  Tue Sep 13 14:10:15 2011
From: ocset at the800group.com (ocset)
Date: Tue, 13 Sep 2011 18:10:15 +0800
Subject: [Freeswitch-users] GXW4104 gateway setup for outgoing calls
In-Reply-To: <CAHKqMb7vF4ULKjWsHqP20xVRy0oOrDo9A3fpwCBr2NXAjsJ6iw@mail.gmail.com>
References: <4E6EFCFA.6000803@the800group.com>
	<CAHKqMb7vF4ULKjWsHqP20xVRy0oOrDo9A3fpwCBr2NXAjsJ6iw@mail.gmail.com>
Message-ID: <4E6F2C07.80800@the800group.com>

Hi Nandy

Thanks for your reply. I assume 192.168.0.9 in your example is the IP 
address of the GXW4104?

Some more questions

1. When you say port number, is this something I should be setting up on 
the GXW4104 so that it is listening on those 4 port numbers? If yes, 
what would be the setting I am looking for?

2. Does that mean I don't define a new gateway in FreeSWITCH?

3. In your example, you said the bridge data would be 
7654321 at 192.168.0.9:5063. What would the whole line look like in the 
dialplan?*

<action application="bridge" 
data="sofia/gateway/7654321 at 192.168.0.9:5063*<http://7654321 at 192.168.0.9:5063>*"/>*

Still very confused :-)

Thanks

On 09/13/2011 03:45 PM, Nandy Dagondon wrote:
> hi,
>
> if GWX4104 is in your local network, use the internal profile for the 
> gateway. register your FXO accounts to receive incoming calls (i think 
> you did this already).
>
> to dialout the ports, specify the port number 5060~5063 assuming Port1 
> starts at 5060. to dialout via port4, the bridge data should look like:
>
> 7654321 at 192.168.0.9:5063 <http://7654321 at 192.168.0.9:5063>
>
> hope it helps.
>
> -nandy
>
>
> On Tue, Sep 13, 2011 at 2:49 PM, ocset <ocset at the800group.com 
> <mailto:ocset at the800group.com>> wrote:
>
>     Hi
>
>     I have recently bought a Grandstream GXW4104 (4 FXO ports) and
>     need some help setting up a gateway to call out using the GXW4104.
>     I am really out of my depth here and may be looking at this the
>     wrong way so please bear with me.
>
>     I followed the advice on this website
>     "http://www.timhunt.net/wiki/FreeSwitch:GXW4104"
>     <http://www.timhunt.net/wiki/FreeSwitch:GXW4104> and incoming
>     calls from a PSTN line are working great. Now I need to setup a
>     dialplan so that outgoing calls are routed through the same PSTN
>     line on the GXW4104. I will eventually have 4 PSTN lines with a
>     dialplan to use the first available line (if that is possible).
>
>     According to the FreeSWITCH 1.0.6 book (and many online posts) I
>     need to create a gateway and a dialplan but all the gateway
>     examples are for SIP accounts.
>
>     So, the gateway definition seems to need a username and password
>     but the GXW4104 does not have that capability. I found this
>     gateway definition in the  freeswitch.xml.fsxml file but am not
>     sure how many of these variables are required.
>
>     <gateways>
>     <gateway name="example.com <http://example.com>">
>     <param name="username" value="joeuser"/>
>     <param name="password" value="password"/>
>     <param name="from-user" value="joeuser"/>
>     <param name="from-domain" value="example.com <http://example.com>"/>
>     <param name="expire-seconds" value="600"/>
>     <param name="register" value="false"/>
>     <param name="retry-seconds" value="30"/>
>     <param name="extension" value="5000"/>
>     <param name="context" value="public"/>
>     </gateway>
>     </gateways>
>
>
>     If I define a gateway called "gxw4104", then this is what I think
>     a simple dialplan should look like but I'm not sure of the gateway
>     details in the "bridge" section of the definition.
>
>     <extension name="gxw4104_out">
>     <condition field="destination_number" expression="^(\d{10})$">
>     *<action application="bridge"
>     data="sofia/gateway/gxw4104/........"/>        (what should this
>     be???)*
>     </condition>
>     </extension>
>
>     Am I moving in the right direction and can someone fill in the
>     blanks for me please
>
>     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
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From sid at eltc.ru  Tue Sep 13 14:14:59 2011
From: sid at eltc.ru (Sergey Scheglov)
Date: Tue, 13 Sep 2011 17:14:59 +0700
Subject: [Freeswitch-users] Lua "two steps" bridging
In-Reply-To: <20110913093933.16714.qmail@community24.interfree.it>
References: <20110913093933.16714.qmail@community24.interfree.it>
Message-ID: <20110913171459.01b3a79a@shadow>

? 13 Sep 2011 09:39:33 -0000
chrisbware at interfree.it wrote:

> 
> Hi guys,
> 
> I've read anything on this argument on Wiki but I can't find a
> complete answer.
> 
> I need to bridge two calls in Lua, waiting the first to be answered.
> My basic script is:
> 
> local GwParams =
> "origination_caller_id_number="..gateway..",sip_auth_username="..gateway..",sip_auth_password="..sip_passwd
> local slegA =
> "[ignore_early_media=false,"..GwParams.."]sofia/external/"..callee.."@"..sip_domain
> local slegB =
> "[ignore_early_media=true,"..GwParams.."]sofia/external/"..called.."@"..sip_domain
> 
> legA = freeswitch.Session(slegA);
> legB = freeswitch.Session(slegB);
> freeswitch.bridge(legA, legB);
> 
> 
> It works but legA and legB are called at the same time. Using:
> 
> while (legA:answered()== false) do end;
> 
> between the two session do the job but called parties can't hear
> audio.
> 
> I promise to add a script on wiki if you help me ! :)
> 
> Thanks in advance.
> 
> 
> -------------------------------------------------------------------------------
> Valore legale alle tue mail
> InterfreePEC - la tua Posta Elettronica Certificata
> http://pec.interfree.it
> -------------------------------------------------------------------------------
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 

Try:

legA = freeswitch.Session(slegA);
if (legA:ready()) then
  legB = freeswitch.Session(slegB, legA);
  freeswitch.bridge(legA, legB);
end

----

Scheglov Sergey


From ocset at the800group.com  Tue Sep 13 14:59:56 2011
From: ocset at the800group.com (ocset)
Date: Tue, 13 Sep 2011 18:59:56 +0800
Subject: [Freeswitch-users] GXW4104 gateway setup for outgoing calls
In-Reply-To: <4E6F2C07.80800@the800group.com>
References: <4E6EFCFA.6000803@the800group.com>
	<CAHKqMb7vF4ULKjWsHqP20xVRy0oOrDo9A3fpwCBr2NXAjsJ6iw@mail.gmail.com>
	<4E6F2C07.80800@the800group.com>
Message-ID: <4E6F37AC.9030406@the800group.com>

Ok, I have not added any new gateway but added a file called 
02_gxw4104.xml into the dialplan/defaults directory with the following 
content:

<include>
<extension name="To PSTN">
<condition field="destination_number" expression="(.*)">
<action application="bridge" 
data="sofia/internal/${destination_number}@[192.168.0.160]:5060" />
</condition>
</extension>
</include>

When I dial 1234567890, FS sees the extension but then just hangs up 
with the following log output.

2011-09-13 18:51:29.299942 [NOTICE] switch_channel.c:669 New Channel 
sofia/internal/1003 at 192.168.0.23 [9af4207f-2e03-4d6e-8aeb-16b063ecf30a]
2011-09-13 18:51:29.331835 [INFO] mod_dialplan_xml.c:418 Processing 
User1->1234567890 in context default
2011-09-13 18:51:29.343016 [NOTICE] switch_channel.c:669 New Channel 
sofia/internal/1234567890@[192.168.0.160]:5060 
[f9106a7a-c25a-434f-bac1-599a0ad71333]
2011-09-13 18:51:29.792807 [NOTICE] sofia.c:4789 Hangup 
sofia/internal/1234567890@[192.168.0.160]:5060 [CS_CONSUME_MEDIA] 
[NORMAL_TEMPORARY_FAILURE]
2011-09-13 18:51:29.792807 [INFO] mod_dptools.c:2355 Originate Failed.  
Cause: NORMAL_TEMPORARY_FAILURE
2011-09-13 18:51:29.792807 [NOTICE] mod_dptools.c:2418 Hangup 
sofia/internal/1003 at 192.168.0.23 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
2011-09-13 18:51:29.836129 [NOTICE] switch_core_session.c:1182 Session 
29 (sofia/internal/1003 at 192.168.0.23) Ended
2011-09-13 18:51:29.836129 [NOTICE] switch_core_session.c:1184 Close 
Channel sofia/internal/1003 at 192.168.0.23 [CS_DESTROY]
2011-09-13 18:51:29.836129 [NOTICE] switch_core_session.c:1182 Session 
30 (sofia/internal/1234567890@[192.168.0.160]:5060) Ended
2011-09-13 18:51:29.836129 [NOTICE] switch_core_session.c:1184 Close 
Channel sofia/internal/1234567890@[192.168.0.160]:5060 [CS_DESTROY]

What am I missing?
Thanks


On 09/13/2011 06:10 PM, ocset wrote:
> Hi Nandy
>
> Thanks for your reply. I assume 192.168.0.9 in your example is the IP 
> address of the GXW4104?
>
> Some more questions
>
> 1. When you say port number, is this something I should be setting up 
> on the GXW4104 so that it is listening on those 4 port numbers? If 
> yes, what would be the setting I am looking for?
>
> 2. Does that mean I don't define a new gateway in FreeSWITCH?
>
> 3. In your example, you said the bridge data would be 
> 7654321 at 192.168.0.9:5063. What would the whole line look like in the 
> dialplan?*
>
> <action application="bridge" 
> data="sofia/gateway/7654321 at 192.168.0.9:5063**"/>*
>
> Still very confused :-)
>
> Thanks
>
> On 09/13/2011 03:45 PM, Nandy Dagondon wrote:
>> hi,
>>
>> if GWX4104 is in your local network, use the internal profile for the 
>> gateway. register your FXO accounts to receive incoming calls (i 
>> think you did this already).
>>
>> to dialout the ports, specify the port number 5060~5063 assuming 
>> Port1 starts at 5060. to dialout via port4, the bridge data should 
>> look like:
>>
>> 7654321 at 192.168.0.9:5063 <http://7654321 at 192.168.0.9:5063>
>>
>> hope it helps.
>>
>> -nandy
>>
>>
>> On Tue, Sep 13, 2011 at 2:49 PM, ocset <ocset at the800group.com 
>> <mailto:ocset at the800group.com>> wrote:
>>
>>     Hi
>>
>>     I have recently bought a Grandstream GXW4104 (4 FXO ports) and
>>     need some help setting up a gateway to call out using the
>>     GXW4104. I am really out of my depth here and may be looking at
>>     this the wrong way so please bear with me.
>>
>>     I followed the advice on this website
>>     "http://www.timhunt.net/wiki/FreeSwitch:GXW4104"
>>     <http://www.timhunt.net/wiki/FreeSwitch:GXW4104> and incoming
>>     calls from a PSTN line are working great. Now I need to setup a
>>     dialplan so that outgoing calls are routed through the same PSTN
>>     line on the GXW4104. I will eventually have 4 PSTN lines with a
>>     dialplan to use the first available line (if that is possible).
>>
>>     According to the FreeSWITCH 1.0.6 book (and many online posts) I
>>     need to create a gateway and a dialplan but all the gateway
>>     examples are for SIP accounts.
>>
>>     So, the gateway definition seems to need a username and password
>>     but the GXW4104 does not have that capability. I found this
>>     gateway definition in the  freeswitch.xml.fsxml file but am not
>>     sure how many of these variables are required.
>>
>>     <gateways>
>>     <gateway name="example.com <http://example.com>">
>>     <param name="username" value="joeuser"/>
>>     <param name="password" value="password"/>
>>     <param name="from-user" value="joeuser"/>
>>     <param name="from-domain" value="example.com <http://example.com>"/>
>>     <param name="expire-seconds" value="600"/>
>>     <param name="register" value="false"/>
>>     <param name="retry-seconds" value="30"/>
>>     <param name="extension" value="5000"/>
>>     <param name="context" value="public"/>
>>     </gateway>
>>     </gateways>
>>
>>
>>     If I define a gateway called "gxw4104", then this is what I think
>>     a simple dialplan should look like but I'm not sure of the
>>     gateway details in the "bridge" section of the definition.
>>
>>     <extension name="gxw4104_out">
>>     <condition field="destination_number" expression="^(\d{10})$">
>>     *<action application="bridge"
>>     data="sofia/gateway/gxw4104/........"/>        (what should this
>>     be???)*
>>     </condition>
>>     </extension>
>>
>>     Am I moving in the right direction and can someone fill in the
>>     blanks for me please
>>
>>     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
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From sid at eltc.ru  Tue Sep 13 15:15:23 2011
From: sid at eltc.ru (Sergey Scheglov)
Date: Tue, 13 Sep 2011 18:15:23 +0700
Subject: [Freeswitch-users] GXW4104 gateway setup for outgoing calls
In-Reply-To: <4E6F37AC.9030406@the800group.com>
References: <4E6EFCFA.6000803@the800group.com>
	<CAHKqMb7vF4ULKjWsHqP20xVRy0oOrDo9A3fpwCBr2NXAjsJ6iw@mail.gmail.com>
	<4E6F2C07.80800@the800group.com> <4E6F37AC.9030406@the800group.com>
Message-ID: <20110913181523.6ddc2d03@shadow>

? Tue, 13 Sep 2011 18:59:56 +0800
ocset <ocset at the800group.com> wrote:

> <action application="bridge"
> data="sofia/internal/${destination_number}@[192.168.0.160]:5060" />

try:
<action application="bridge"
data="sofia/internal/${destination_number}@192.168.0.160:5060" />
-- 
Scheglov Sergey


From ocset at the800group.com  Tue Sep 13 16:07:58 2011
From: ocset at the800group.com (ocset)
Date: Tue, 13 Sep 2011 20:07:58 +0800
Subject: [Freeswitch-users] GXW4104 gateway setup for outgoing calls
In-Reply-To: <20110913181523.6ddc2d03@shadow>
References: <4E6EFCFA.6000803@the800group.com>
	<CAHKqMb7vF4ULKjWsHqP20xVRy0oOrDo9A3fpwCBr2NXAjsJ6iw@mail.gmail.com>
	<4E6F2C07.80800@the800group.com> <4E6F37AC.9030406@the800group.com>
	<20110913181523.6ddc2d03@shadow>
Message-ID: <4E6F479E.8020406@the800group.com>

Thanks Sergey

Removing the "[]" made FS think for a while and then just hang up. The 
log is slightly different.


2011-09-13 20:05:32.107712 [NOTICE] switch_channel.c:669 New Channel 
sofia/internal/1014 at 192.168.0.23 [af04bba5-91d2-4cd6-b881-aae6afd61cd1]
2011-09-13 20:05:32.137091 [INFO] mod_dialplan_xml.c:418 Processing 
User1->1234567890 in context default
2011-09-13 20:05:32.148271 [NOTICE] switch_channel.c:669 New Channel 
sofia/internal/1234567890 at 192.168.0.160:5060 
[f698d4b1-838d-4160-891a-5ed78405b070]
(here it thinks for a while)
2011-09-13 20:05:42.151745 [NOTICE] sofia.c:4789 Hangup 
sofia/internal/1234567890 at 192.168.0.160:5060 [CS_CONSUME_MEDIA] 
[NORMAL_TEMPORARY_FAILURE]
2011-09-13 20:05:42.151745 [INFO] mod_dptools.c:2355 Originate Failed.  
Cause: NORMAL_TEMPORARY_FAILURE
2011-09-13 20:05:42.151745 [NOTICE] mod_dptools.c:2418 Hangup 
sofia/internal/1014 at 192.168.0.23 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
2011-09-13 20:05:42.151745 [NOTICE] switch_core_session.c:1182 Session 
30 (sofia/internal/1234567890 at 192.168.0.160:5060) Ended
2011-09-13 20:05:42.151745 [NOTICE] switch_core_session.c:1184 Close 
Channel sofia/internal/1234567890 at 192.168.0.160:5060 [CS_DESTROY]
2011-09-13 20:05:42.184766 [NOTICE] switch_core_session.c:1182 Session 
29 (sofia/internal/1014 at 192.168.0.23) Ended
2011-09-13 20:05:42.184766 [NOTICE] switch_core_session.c:1184 Close 
Channel sofia/internal/1014 at 192.168.0.23 [CS_DESTROY]


On 09/13/2011 07:15 PM, Sergey Scheglov wrote:
> ? Tue, 13 Sep 2011 18:59:56 +0800
> ocset<ocset at the800group.com>  wrote:
>
>> <action application="bridge"
>> data="sofia/internal/${destination_number}@[192.168.0.160]:5060" />
> try:
> <action application="bridge"
> data="sofia/internal/${destination_number}@192.168.0.160:5060" />


From chrisbware at interfree.it  Tue Sep 13 16:21:38 2011
From: chrisbware at interfree.it (chrisbware at interfree.it)
Date: 13 Sep 2011 12:21:38 -0000
Subject: [Freeswitch-users] Re2:  Lua "two steps" bridging
Message-ID: <20110913122138.15531.qmail@community17.interfree.it>

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From adam.kelloway at newpace.ca  Tue Sep 13 17:38:18 2011
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Tue, 13 Sep 2011 10:38:18 -0300
Subject: [Freeswitch-users] Detecting silence in recorded file
Message-ID: <4E6F5CCA.3060801@newpace.ca>

Hi there,

Is there way to do any the following when using the record dial plan tool?

- prevent the record tool from writing any samples to the destination 
file if the audio is not above the given silence threshold
- detect whether a recorded file contains any audio above the given 
silence threshold

I would like to treat recordings that are purely "silence" differently 
from those that contain audio above the silence threshold (such as 
speech). I would be interested in knowing if anyone has ever done this, 
and what your approach was.

Thanks,
Adam


From avi at avimarcus.net  Tue Sep 13 17:44:25 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 13 Sep 2011 16:44:25 +0300
Subject: [Freeswitch-users] Cut PCAP file?
In-Reply-To: <CALNZuw=v0HhkRuwin6zmhA2RaU0dhekS1zNNgwfGpZrtRTEu1Q@mail.gmail.com>
References: <CALNZuw==7kXb7zZgJYDjb8ey+47J0jK9dVHLTMX2Q7w2H+8ROw@mail.gmail.com>
	<CABHdwCr9+kieEKOfoG3akFAMV5+ZJfzWTBS5V71PgvODMhkO7w@mail.gmail.com>
	<CALNZuwmA1CDzRT3BDNc01iO0Kq8k+z9AqsmK3_dUDQpQPp28bg@mail.gmail.com>
	<CAKzWOxXK5EdrCjRGHyyecMXPnHDdC7hS3Y4OZ82eKxjNrP3Jjw@mail.gmail.com>
	<CABHdwCprY2cGDoOH4ONVqai6r2b8NKdBJtLrO+fbHrH1tdEPqQ@mail.gmail.com>
	<CALNZuw=v0HhkRuwin6zmhA2RaU0dhekS1zNNgwfGpZrtRTEu1Q@mail.gmail.com>
Message-ID: <CALNZuw=Mn5in8AirwPEAbi3Y66GzEY4zYyZoMR0SdmX_+gb8vQ@mail.gmail.com>

Well. Apparently it was right under my nose - clicking "save as" in the
wireshark gui lets you save all packets, from frame $x to $y, the marked
ones, from the first marked one to the last...
-Avi


On Tue, Sep 13, 2011 at 9:14 AM, Avi Marcus <avi at avimarcus.net> wrote:

> Heh. I already added it 7 hours before.. merging both.
> -Avi
>
>
> On Tue, Sep 13, 2011 at 7:12 AM, Wasim Baig <wasim at convergence.pk> wrote:
>
>>
>> http://wiki.freeswitch.org/wiki/Packet_Capture#Edit_.2F_Cut_a_large_pcap_file_into_smaller_chunks
>>
>>
>> On Mon, Sep 12, 2011 at 21:46, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> And you were so proud of finding this that you put it on the wiki also?
>>> ;)
>>> -MC
>>>
>>>
>>> On Sun, Sep 11, 2011 at 9:11 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>>> 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
>>>>>
>>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>>
>
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From ira at connectmevoice.com  Tue Sep 13 18:36:42 2011
From: ira at connectmevoice.com (Ira Tessler)
Date: Tue, 13 Sep 2011 10:36:42 -0400
Subject: [Freeswitch-users] ESL Send MESSAGE_WAITING Error - Cannot find
	profile
In-Reply-To: <BANLkTikK5D1Cqu8bN1uJxaqGUvM552waJw@mail.gmail.com>
References: <BANLkTimGcbOiDUDgkRxbHZyPYASTidbVjQ@mail.gmail.com>
	<BANLkTikHzwA0KGyBXAXxOwNid1DfMZVMKA@mail.gmail.com>	<BANLkTimYvT-apQ7BZS3FFR8EkpEpht2ohw@mail.gmail.com>
	<BANLkTikK5D1Cqu8bN1uJxaqGUvM552waJw@mail.gmail.com>
Message-ID: <576ebdf94404d6e07128ac55c1eb7e24@mail.gmail.com>

I am the one with the original issue. I have confirmed that the
20064.domain.com domain is aliased to internal profile. Any other ideas?

Ira Tessler
ConnectMe

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Thursday, June 30, 2011 4:14 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] ESL Send MESSAGE_WAITING Error - Cannot
find profile

so that would suggest nobody is in your sip_registrations table with that
host and/or you have not aliased it to your sofia profile

On Thu, Jun 30, 2011 at 2:19 PM, Avi Marcus <avi at avimarcus.net> wrote:
> Anthony,
>
> FreeSWITCH Version 1.0.head (git-3e6cca9 2011-06-28 10-27-00 -0700)
>
> It still produces an error: ?"sofia_presence.c:432 Cannot find profile
> 20064.domain.com"
>
> -Avi
>
> On Tue, Jun 28, 2011 at 8:03 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> can you try this on latest too?
>> The line numbers suggest an older build.
>>
>>
>> On Tue, Jun 28, 2011 at 9:10 AM, Avi Marcus <avi at avimarcus.net> wrote:
>> > I'm trying to send a?MESSAGE_WAITING via ESL on multi-tenant, but
>> > I'm getting this error:
>> > 2011-06-27 10:46:25.443081 [ERR] sofia_presence.c:405 Cannot find
>> > profile?20064.domain.com] the cli lists the domain as an alias to
>> > the internal profile.
>> > what should I do?
>> > I am writing some test code in order to try to send MWI events
>> > using the ESL Manager code DLL. Here is my test code:
>> >
>> >
>> >
>> > static void InboundMode2(Object stateInfo)
>> >
>> > ??? {
>> >
>> > ??????? //Initializes a new instance of ESLconnection, and connects
>> > to the host $host on the port $port, and supplies $password to
>> > freeswitch
>> >
>> > ??????? ESLconnection eslConnection = new ESLconnection(myinfo);
>> >
>> >
>> >
>> > ??????? if (eslConnection.Connected() != ESL_SUCCESS)
>> >
>> > ??????? {
>> >
>> > ??????????? Console.WriteLine("Error connecting to FreeSwitch");
>> >
>> > ??????????? return;
>> >
>> > ??????? }
>> >
>> >
>> >
>> > ??????? //Set log level
>> >
>> > ??????? //ESL.eslSetLogLevel((int)enLogLevel.DEBUG);
>> >
>> >
>> >
>> > ??????? eslConnection.Api("reloadxml", string.Empty);
>> >
>> >
>> >
>> > ??????? // Subscribe to all events
>> >
>> > ????????ESLevent eslEvent2 = eslConnection.SendRecv("event plain
>> > ALL");
>> >
>> >
>> >
>> > ??????? if (eslEvent2 == null)
>> >
>> > ??????? {
>> >
>> > ??????????? Console.WriteLine("Error subscribing to all events");
>> >
>> > ??????????? return;
>> >
>> > ??????? }
>> >
>> > ??????? ESLevent eslEvent = new ESLevent("MESSAGE_WAITING", null);
>> >
>> > ??????? eslEvent.AddHeader("MWI-Messages-Waiting", "yes");
>> >
>> > ??????? eslEvent.AddHeader("MWI-Message-Account",
>> > "103 at 20064.domain.com");
>> >
>> > ???? ???eslEvent.AddHeader("MWI-Voice-Message", "1/1 (1/1)");
>> >
>> >
>> >
>> > ??????? eslEvent = eslConnection.SendEvent(eslEvent);
>> >
>> > ??????? if (eslEvent == null)
>> >
>> > ??????? {
>> >
>> > ??????????? Console.WriteLine("event error");
>> >
>> > ??????????? return;
>> >
>> > ??????? }
>> >
>> >
>> >
>> > ??????? //Turns an event into colon-separated 'name: value' pairs.
>> > The format parameter isn't used
>> >
>> > ??????? Console.WriteLine(eslEvent.Serialize(String.Empty));
>> >
>> >
>> >
>> > ??????? // Grab Events until process is killed
>> >
>> > ??????? while (eslConnection.Connected() == ESL_SUCCESS)
>> >
>> > ??????? {
>> >
>> > ??????????? eslEvent = eslConnection.RecvEvent();
>> >
>> > ??????????? Console.WriteLine(eslEvent.Serialize(String.Empty));
>> >
>> > ??????? }
>> >
>> > ??? }
>> >
>> >
>> >
>> > When I send the event, I am getting a debug message from FS as
follows:
>> >
>> >
>> >
>> > 2011-06-27 10:46:25.443081 [ERR] sofia_presence.c:405 Cannot find
>> > profile 20064.domain.com]
>> >
>> > Thanks!
>> >
>> > -Avi
>> >
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com
> 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>



--
Anthony Minessale II

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

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

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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  Tue Sep 13 19:01:05 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 13 Sep 2011 18:01:05 +0300
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION - did FS not offer a
	codec?
Message-ID: <CALNZuwkvEtVcYXjie4RsBhaKcJiSJsrdBJ_otXijcDHm74EBsA@mail.gmail.com>

I see a failed bridge to my second carrier. The  sip_local_sdp_str doesn't
show PCMU in it - did FS not offer any codecs?
FS log: http://pastebin.freeswitch.org/17316
PCAP: http://ge.tt/95tFji7

Can someone tell me what's going on here?

-Avi
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From kris at kriskinc.com  Tue Sep 13 19:08:57 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 13 Sep 2011 11:08:57 -0400
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION - did FS not offer
	a codec?
In-Reply-To: <CALNZuwkvEtVcYXjie4RsBhaKcJiSJsrdBJ_otXijcDHm74EBsA@mail.gmail.com>
References: <CALNZuwkvEtVcYXjie4RsBhaKcJiSJsrdBJ_otXijcDHm74EBsA@mail.gmail.com>
Message-ID: <CAKDfjgeRN8AZPZNMBT2XqBDcMpAfyM_r2BkMO3EfdsJ87X61og@mail.gmail.com>

Try setting verbose SDP:

http://wiki.freeswitch.org/wiki/Variable_verbose_sdp

On Tue, Sep 13, 2011 at 11:01 AM, Avi Marcus <avi at avimarcus.net> wrote:
> I see a failed bridge to my second carrier.?The??sip_local_sdp_str doesn't
> show PCMU in it - did FS not offer any codecs?
> FS log:?http://pastebin.freeswitch.org/17316
> PCAP:?http://ge.tt/95tFji7
> Can someone tell me what's going on here?
> -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
>
>



-- 
Kristian Kielhofner


From msc at freeswitch.org  Tue Sep 13 19:15:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Sep 2011 08:15:26 -0700
Subject: [Freeswitch-users] external sip profile
In-Reply-To: <174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com>
References: <D9A75749-55E5-4DC5-A401-5F887ABEDE1A@lyonl.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0FE@cooper>
	<CAHKqMb4fp1jBxkjGAFVifBb6-MJxM9p01oiKzyqo8kaTQ7v2Vw@mail.gmail.com>
	<05DF269C-AE48-4AD0-8E34-C31B77A072B4@lyonl.com>
	<CAKzWOxX5f_Cmfm8FRX6N2q-GQhBgPDpn3RYeJTu7F_1zRofkog@mail.gmail.com>
	<174254F1-DA32-4A79-8DD2-6BB5C5A5DDFD@lyonl.com>
Message-ID: <CAKzWOxX2ptnV_TN3dZ+_V4yQi5qKbHmM_WJo5Mh0KU7VGZUiuQ@mail.gmail.com>

Open the pcap file in wireshark and see if the destination port in the SDP
matches the actual port number to which the incoming RTP packets are being
sent. My guess is the router/NAT device is messing with your RTP traffic.

-MC

On Mon, Sep 12, 2011 at 6:31 PM, Chad Vogel <cvogel at lyonl.com> wrote:

>  I was able to confirm that i have one way and two way audio now (I can see
> it in wireshark), it seems like the echo application doesn't work correctly,
> it will not send out audio. when i use <action application="delay_echo"
> data="1000"/> in the dialplan it does seems to send out rtc traffic
> however the audio is static noise. when i use <action application="echo"
>  />  it doest seem to work. any thoughts on why cant I echo test and why
> delay_echo sends out static noise. I tried to use the record app it creates
> a wav file of 44 KB but doesn't grow beyond.
>
>
>  On Sep 12, 2011, at 7:18 PM, Michael Collins wrote:
>
> Can you confirm if you have one-way audio? That is, an echo test won't tell
> you if you have one-way audio. A simple way to do this is to use the record
> app. It will play a file (whatever you choose, like "record at the tone..."
> - there are tons of sound files you can try) and then record the audio from
> the caller. Another test to do is to get a pcap of that call so you can
> analyze it in wireshark. If you have RTP going in both directions to/from
> the FS box then that indicates a NAT issue...
>
>  -MC
>
> On Sun, Sep 11, 2011 at 7:00 PM, Chad Vogel <cvogel at lyonl.com> wrote:
>
>> Level 3 uses port 5070 for their sip server but requires port 5060 to be
>> used on our side; I made changes to the vars.xml config file to support
>> this. I was just able to get the server to answer a call by changing the
>> register value to false and adding an entry in the ACL config for the Level
>> 3 server however now it seems that the audio isn't working correctly. I'm
>> using a simple dial plan to echo the audio back but all I get is dead air.
>>
>>       <extension name="level3">
>>        <condition field="destination_number" expression="^13127567000$">
>>          <action application="answer"/>
>>          <action application="echo" />
>>        </condition>
>>      </extension>
>>
>>
>>
>>  On Sep 11, 2011, at 7:47 PM, Nandy Dagondon wrote:
>>
>> in addition to peter's advise, take a look at the SIP port 5070. FS is
>> using port 5080 for the external SIP profile. modify the port number at
>> "external.xml" then delete the port numbers in your proxy settings.
>>
>> On Sun, Sep 11, 2011 at 6:36 PM, Peter Olsson <
>> peter.olsson at visionutveckling.se> wrote:
>>
>>> 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!
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris at ghosttelecom.com  Tue Sep 13 19:26:19 2011
From: chris at ghosttelecom.com (Chris Martineau)
Date: Tue, 13 Sep 2011 16:26:19 +0100
Subject: [Freeswitch-users] fs_curl
Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF3597F4@SVR01.ghosttelecom.local>

 

Hi,

 

Wanted to download fs_curl to have a play with but cannot seem to
download it as a complete app. Not used fisheye before so cannot see how
to git or svn from it and have tried various ways but the only way I can
see of getting the file contents is to stick the source on fisheye to
raw and copy it file by file!

If someone can provide a git or svn link to get the latest version it
would be greatly appreciated.

 

Many thanks

 

Chris

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From msc at freeswitch.org  Tue Sep 13 19:28:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Sep 2011 08:28:49 -0700
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION - did FS not offer
	a codec?
In-Reply-To: <CALNZuwkvEtVcYXjie4RsBhaKcJiSJsrdBJ_otXijcDHm74EBsA@mail.gmail.com>
References: <CALNZuwkvEtVcYXjie4RsBhaKcJiSJsrdBJ_otXijcDHm74EBsA@mail.gmail.com>
Message-ID: <CAKzWOxUpFV=8UxNhKByiERD3mTO-uxfi0mwfqswfGTzkpFvQWg@mail.gmail.com>

For the record, line #107 contains the codecs (and other stuff) that FS
offered:

m=audio 16480 RTP/AVP 8 0 9 101 13

The "8 0 9 101 13" represent the codecs (and other stuff). What do those
numbers mean? This wiki page has the list:

http://wiki.freeswitch.org/wiki/RTP_payload_list

The SDP spec says that for non-dynamic payload numbers you are NOT required
to have the "a=rtpmap" lines. This makes sense because those payload numbers
don't change meaning until you get up to 96 and higher. The issue is that
some devices are broken and require the "a=rtpmap" lines for the payload
types below number 96. The verbose_sdp chan var mentioned by KK tells FS to
add all those a=rtpmap lines for the non-dynamic payload types. You should
only set this var when absolutely necessary because those verbose SDP's can
cause the packets to be larger than the allowable MTU size.

Dontcha just *love* SIP/SDP interop? :)

-MC

On Tue, Sep 13, 2011 at 8:01 AM, Avi Marcus <avi at avimarcus.net> wrote:

> I see a failed bridge to my second carrier. The  sip_local_sdp_str doesn't
> show PCMU in it - did FS not offer any codecs?
> FS log: http://pastebin.freeswitch.org/17316
> PCAP: http://ge.tt/95tFji7
>
> Can someone tell me what's going on here?
>
> -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
>
>
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From gcd at i.ph  Tue Sep 13 19:42:54 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 13 Sep 2011 23:42:54 +0800
Subject: [Freeswitch-users] GXW4104 gateway setup for outgoing calls
In-Reply-To: <4E6F2C07.80800@the800group.com>
References: <4E6EFCFA.6000803@the800group.com>
	<CAHKqMb7vF4ULKjWsHqP20xVRy0oOrDo9A3fpwCBr2NXAjsJ6iw@mail.gmail.com>
	<4E6F2C07.80800@the800group.com>
Message-ID: <CAHKqMb7BV1oVa5NqvZP+mgS1QSjuaUu-JPpHRFpqmNBs8G8mHQ@mail.gmail.com>

i inserted my answers to your questions below. for point #3), here's an
example how i configured my FXO port of ht503.

included in  sip_profile/internal:
<include>
  <gateway name="ht503-fxo">
  <param name="username" value="1019"/>   <-- it's registered to receive
incoming calls
  <param name="realm" value="192.168.0.9"/>
  <param name="sip-port" value="5062"/>   <-- port 5060 is set to the FXS
port
  <param name="password" value="1234"/>
  <param name="rtp_ip" value="192.168.0.9"/>
  <param name="dtmf-type" value="rfc2833"/>
  <param name="expire-seconds" value="600"/>
  <param name="register" value="false"/>
  <param name="caller-id-in-from" value="false"/>
  </gateway>
</include>

included  in dialplan/default

<include>
  <extension name="ht503-fxo-local">
    <condition field="${toll_allow}" expression="local"/>
    <condition field="destination_number" expression="^9([2-9]\d{6})$">
    <action application="set" data="effective_caller_id_number=0321234567"/>
    <action application="set" data="effective_caller_id_name=ThisIsMy
Company"/>
    <action application="set" data="ignore_early_media=ring_ready"/>
    <action application="set" data="ringback=${us-ring}"/>
      <action application="bridge" data="sofia/gateway/ht503-fxo/$
1 at 192.168.0.9:5062"/>
    </condition>
  </extension>
</include>

it looks you can create 4 internal gateways for the every port, fxo-1 to
fxo-4, w/ the same realm/rtp_ip values but setting different sip-port
values. then  your bridge app would be:

      <action application="bridge"
data="sofia/gateway/fxo-1/$1|sofia/gateway/fxo-2/$1|sofia/gateway/fxo-3/$1sofia/gateway/fxo-4/$1"/>

if u want to dialout any free port.

i haven't tested the above. just try it. i hope it works.

-nandy


On Tue, Sep 13, 2011 at 6:10 PM, ocset <ocset at the800group.com> wrote:

>  Hi Nandy
>
> Thanks for your reply. I assume 192.168.0.9 in your example is the IP
> address of the GXW4104?
>
yes.

>
>
Some more questions
>
> 1. When you say port number, is this something I should be setting up on
> the GXW4104 so that it is listening on those 4 port numbers? If yes, what
> would be the setting I am looking for?
>
not for every port. the gateway has a base port number e.g. 5060 for port#1.
add 2 to the subsequent ports e.g. 5062 for port#2 and so on. this is
pointed out by sergey.

>
> 2. Does that mean I don't define a new gateway in FreeSWITCH?
>
it's an option. but defining a gateway is cleaner.


>
>
3. In your example, you said the bridge data would be
> 7654321 at 192.168.0.9:5063. What would the whole line look like in the
> dialplan?*
>
> <action application="bridge" data="sofia/gateway/7654321 at 192.168.0.9:5063*<http://7654321 at 192.168.0.9:5063>
> *"/>*
>
> Still very confused :-)
>
> Thanks
>
>
> On 09/13/2011 03:45 PM, Nandy Dagondon wrote:
>
> hi,
>
> if GWX4104 is in your local network, use the internal profile for the
> gateway. register your FXO accounts to receive incoming calls (i think you
> did this already).
>
> to dialout the ports, specify the port number 5060~5063 assuming Port1
> starts at 5060. to dialout via port4, the bridge data should look like:
>
> 7654321 at 192.168.0.9:5063
>
> hope it helps.
>
> -nandy
>
>
> On Tue, Sep 13, 2011 at 2:49 PM, ocset <ocset at the800group.com> wrote:
>
>>  Hi
>>
>> I have recently bought a Grandstream GXW4104 (4 FXO ports) and need some
>> help setting up a gateway to call out using the GXW4104. I am really out of
>> my depth here and may be looking at this the wrong way so please bear with
>> me.
>>
>> I followed the advice on this website
>> "http://www.timhunt.net/wiki/FreeSwitch:GXW4104"<http://www.timhunt.net/wiki/FreeSwitch:GXW4104>and incoming calls from a PSTN line are working great. Now I need to setup a
>> dialplan so that outgoing calls are routed through the same PSTN line on the
>> GXW4104. I will eventually have 4 PSTN lines with a dialplan to use the
>> first available line (if that is possible).
>>
>> According to the FreeSWITCH 1.0.6 book (and many online posts) I need to
>> create a gateway and a dialplan but all the gateway examples are for SIP
>> accounts.
>>
>> So, the gateway definition seems to need a username and password but the
>> GXW4104 does not have that capability. I found this gateway definition in
>> the  freeswitch.xml.fsxml file but am not sure how many of these variables
>> are required.
>>
>> <gateways>
>>    <gateway name="example.com">
>>       <param name="username" value="joeuser"/>
>>       <param name="password" value="password"/>
>>       <param name="from-user" value="joeuser"/>
>>       <param name="from-domain" value="example.com"/>
>>       <param name="expire-seconds" value="600"/>
>>       <param name="register" value="false"/>
>>       <param name="retry-seconds" value="30"/>
>>       <param name="extension" value="5000"/>
>>       <param name="context" value="public"/>
>>    </gateway>
>> </gateways>
>>
>>
>> If I define a gateway called "gxw4104", then this is what I think a simple
>> dialplan should look like but I'm not sure of the gateway details in the
>> "bridge" section of the definition.
>>
>> <extension name="gxw4104_out">
>>    <condition field="destination_number" expression="^(\d{10})$">
>>       *<action application="bridge"
>> data="sofia/gateway/gxw4104/........"/>        (what should this be???)*
>>    </condition>
>> </extension>
>>
>> Am I moving in the right direction and can someone fill in the blanks for
>> me please
>>
>> 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
>>
>>
>
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From buscom123+fs at gmail.com  Tue Sep 13 19:51:55 2011
From: buscom123+fs at gmail.com (R H)
Date: Tue, 13 Sep 2011 09:51:55 -0600
Subject: [Freeswitch-users] problem using mod_managed under linux
In-Reply-To: <30786.1315879273@ccs.covici.com>
References: <rjxwrb84m9olrbg6m7ghyb97.1314822061391@email.android.com>
	<20320.1314825366@ccs.covici.com>
	<CAAtQP55W982E1NvDYYZ5_moKk9aGdddU2j4pDdR-Ncy5JodYTg@mail.gmail.com>
	<30786.1315879273@ccs.covici.com>
Message-ID: <CAAtQP54za1xEuT9HzT-sWOjbzzuKcV-FiotHta_ZXEw7y9vFgQ@mail.gmail.com>

Hey Covici,

I currently have a DLL that uses 3 IAppPlugin interfaces, 2 IApiPlugin
interfaces, and 2 ILoadNotificationPlugin interfaces. I build the dll on a
separate machine but using this system I am able to deploy the dll to a
running FreeSwitch and have it unload/reload the dll on the fly without any
problems.Its been working great for a week or two. A few things you may want
to keep in mind:

1) DLL's deployed in the managed directory are not exactly going to work
well if they have dependencies on each-other as you never know which one
will get loaded first. (Well, if you only drop them in 1 at a time it would
probably work)

2) Any crash in FreeSwitch I have seen is because of the above issue, or
because of some uncaught exception thrown from my code. If you look at
your FreeSwitch logs or start FreeSwitch by hand and watch the cli you
should see the exception that killed it.

When I run FreeSwitch as a service even my script in /etc/init.d/ has to cd
into the mod directory and then start the server. Hope this helps!

Ryan


On Mon, Sep 12, 2011 at 8:01 PM, <covici at ccs.covici.com> wrote:

> Well, I spoke too soon.  It works, but if I make a change and try to
> reload the .dll, it kills fs altogether -- makes it not too usable.
>
> R H <buscom123+fs at gmail.com> wrote:
>
> > Hey Guys,
> >
> > I just noticed this thread in the email queue so this may be coming too
> > late. I have the same problem and sent an email about it to this group
> with
> > the subject line: *Mod_Managed Mono is not following the system LD Path.*
> > *
> > *
> > That said,
> > I have mod_managed working but it is a bit of a hack to do so. I simply
> CD
> > into the mod directory and then start the server from there.
> >
> > EG.
> >
> > cd /usr/local/freeswitch/mod
> > ../bin/freeswitch
> >
> > This makes that particular problem go away. I have tried changing the LD
> > Path, configured mono per the instructions on the wiki, etc? None of that
> > has worked for me but cd'ing into the mod directory and then starting the
> > server at least gets me running.
> >
> > This might be useful to everyone while we wait for this to be resolved. I
> > have yet to add this as a bug in Jira. I was attempting to get some
> response
> > here before doing so.
> >
> > Ryan
> >
> > On Wed, Aug 31, 2011 at 3:16 PM, <covici at ccs.covici.com> wrote:
> >
> > > 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.
> > >
> >
> > ----------------------------------------------------
> > Alternatives:
> >
> > ----------------------------------------------------
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From buscom123+fs at gmail.com  Tue Sep 13 19:58:39 2011
From: buscom123+fs at gmail.com (R H)
Date: Tue, 13 Sep 2011 09:58:39 -0600
Subject: [Freeswitch-users] problem using mod_managed under linux
In-Reply-To: <30786.1315879273@ccs.covici.com>
References: <rjxwrb84m9olrbg6m7ghyb97.1314822061391@email.android.com>
	<20320.1314825366@ccs.covici.com>
	<CAAtQP55W982E1NvDYYZ5_moKk9aGdddU2j4pDdR-Ncy5JodYTg@mail.gmail.com>
	<30786.1315879273@ccs.covici.com>
Message-ID: <CAAtQP56+7hAThRmb6RqzA45i6ok83TXS-vu6-k2uHt9DE1OTpg@mail.gmail.com>

I guess I should also mention that I am running the following:

- 2 FreeSwitch Servers (1 Production, 1 Development).
- Both run Suse 11.4 64bit, Mono 2.8.2, FreeSwitch at HEAD.
- I also have to keep a softlink at /lib64/libc.so (although these boxes are
dedicated FreeSwitch only)
- I had to run the 2.8patch on the mod_managed library before build to make
all this work

Aside from this issue mod_managed seems to run fairly well, but I have not
tested it under load. All-in-all it has done well for me so far.

Ryan

On Mon, Sep 12, 2011 at 8:01 PM, <covici at ccs.covici.com> wrote:

> Well, I spoke too soon.  It works, but if I make a change and try to
> reload the .dll, it kills fs altogether -- makes it not too usable.
>
> R H <buscom123+fs at gmail.com> wrote:
>
> > Hey Guys,
> >
> > I just noticed this thread in the email queue so this may be coming too
> > late. I have the same problem and sent an email about it to this group
> with
> > the subject line: *Mod_Managed Mono is not following the system LD Path.*
> > *
> > *
> > That said,
> > I have mod_managed working but it is a bit of a hack to do so. I simply
> CD
> > into the mod directory and then start the server from there.
> >
> > EG.
> >
> > cd /usr/local/freeswitch/mod
> > ../bin/freeswitch
> >
> > This makes that particular problem go away. I have tried changing the LD
> > Path, configured mono per the instructions on the wiki, etc? None of that
> > has worked for me but cd'ing into the mod directory and then starting the
> > server at least gets me running.
> >
> > This might be useful to everyone while we wait for this to be resolved. I
> > have yet to add this as a bug in Jira. I was attempting to get some
> response
> > here before doing so.
> >
> > Ryan
> >
> > On Wed, Aug 31, 2011 at 3:16 PM, <covici at ccs.covici.com> wrote:
> >
> > > 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.
> > >
> >
> > ----------------------------------------------------
> > Alternatives:
> >
> > ----------------------------------------------------
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jeff at jefflenk.com  Tue Sep 13 20:04:43 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 13 Sep 2011 09:04:43 -0700 (PDT)
Subject: [Freeswitch-users] How to use sofia_count_reg ?
In-Reply-To: <1315579669278-6776183.post@n2.nabble.com>
References: <CALNZuw=B3YYD1eWEMKERWtpitQBDv-AJKgw03k4mGCdZDfOQtA@mail.gmail.com>
	<1315499174093-6772541.post@n2.nabble.com>
	<CALNZuwmi1GKxDc9EWPg7+rmY9mEF1kTEYDP9xgtc8NSajmNVRA@mail.gmail.com>
	<1315579669278-6776183.post@n2.nabble.com>
Message-ID: <1315929883123-6788105.post@n2.nabble.com>

Have a look at git head from today and verify the functionality of both :

sofia_reg_count user at domain
sofia_reg_count @domain

should now work.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-use-sofia-count-reg-tp6770571p6788105.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From cogs66 at gmail.com  Tue Sep 13 20:20:53 2011
From: cogs66 at gmail.com (cogs66)
Date: Tue, 13 Sep 2011 09:20:53 -0700 (PDT)
Subject: [Freeswitch-users] User Agent Displayed by sipsak
In-Reply-To: <2B146FED-757D-45E2-BB9E-F18FE6F370CE@lyonl.com>
References: <1315819350559-6782746.post@n2.nabble.com>
	<2B146FED-757D-45E2-BB9E-F18FE6F370CE@lyonl.com>
Message-ID: <1315930853061-6788185.post@n2.nabble.com>

Thanks, you kindly pointed me in the right direction.

I had to also change in internal.xml 

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/User-Agent-Displayed-by-sipsak-tp6782746p6788185.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From avi at avimarcus.net  Tue Sep 13 20:23:50 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 13 Sep 2011 19:23:50 +0300
Subject: [Freeswitch-users] How to use sofia_count_reg ?
In-Reply-To: <1315929883123-6788105.post@n2.nabble.com>
References: <CALNZuw=B3YYD1eWEMKERWtpitQBDv-AJKgw03k4mGCdZDfOQtA@mail.gmail.com>
	<1315499174093-6772541.post@n2.nabble.com>
	<CALNZuwmi1GKxDc9EWPg7+rmY9mEF1kTEYDP9xgtc8NSajmNVRA@mail.gmail.com>
	<1315579669278-6776183.post@n2.nabble.com>
	<1315929883123-6788105.post@n2.nabble.com>
Message-ID: <CALNZuwmqacB1gca-FAtY38rHV4o5t62AYUiMFrKdWkf+MP=C8g@mail.gmail.com>

Ok, updated... Both seem to work now. Great!

It's sofia_count_reg by the way. I meant to jira it but I've been busy..

My point to using this was a better way to route a hunt script - e.g. if my
sofia_count_reg is 2, the hard & softphone is on and I'm in the office.
Otherwise, if only 1, I'm probably not, so ring my hardphone and my cell via
the PSTN.

-Avi


On Tue, Sep 13, 2011 at 7:04 PM, Jeff Lenk <jeff at jefflenk.com> wrote:

> Have a look at git head from today and verify the functionality of both :
>
> sofia_reg_count user at domain
> sofia_reg_count @domain
>
> should now work.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/How-to-use-sofia-count-reg-tp6770571p6788105.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Sep 13 20:27:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Sep 2011 09:27:52 -0700
Subject: [Freeswitch-users] fs_curl
In-Reply-To: <1D10AB188D6CCA46BB4369E3268E36EF3597F4@SVR01.ghosttelecom.local>
References: <1D10AB188D6CCA46BB4369E3268E36EF3597F4@SVR01.ghosttelecom.local>
Message-ID: <CAKzWOxW8_oDL=9jxDxWxvk0MsG2W4nVfp=L2Au6-ipB4gmgfyw@mail.gmail.com>

Are you talking about the fs_curl program that Raymond (IRC: intralanman)
wrote? You need the contrib repo:

git clone git://git.freeswitch.org/freeswitch-contrib.git


Also, that's all doc'd on our git page:

http://wiki.freeswitch.org/wiki/Git_Tips

-MC

On Tue, Sep 13, 2011 at 8:26 AM, Chris Martineau <chris at ghosttelecom.com>wrote:

> ** **
>
> *Hi,*
>
> * *
>
> *Wanted to download fs_curl to have a play with but cannot seem to
> download it as a complete app. Not used fisheye before so cannot see how to
> git or svn from it and have tried various ways but the only way I can see of
> getting the file contents is to stick the source on fisheye to raw and copy
> it file by file!*
>
> *If someone can provide a git or svn link to get the latest version it
> would be greatly appreciated.*
>
> * *
>
> *Many thanks*
>
> * *
>
> *Chris*****
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Sep 13 20:44:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Sep 2011 09:44:52 -0700
Subject: [Freeswitch-users] Re2: Lua "two steps" bridging
In-Reply-To: <20110913122138.15531.qmail@community17.interfree.it>
References: <20110913122138.15531.qmail@community17.interfree.it>
Message-ID: <CAKzWOxW2Dwva26ZBdJ4gy9ChT6coGNsq8BJBvPFvWSG2kpK+BQ@mail.gmail.com>

How are you calling this Lua script to begin with? What is the big picture,
that is, what is the problem you're solving? Some context might help us give
you a better answer.

-MC

On Tue, Sep 13, 2011 at 5:21 AM, <chrisbware at interfree.it> wrote:

> Hi,
>
> thank you for your help, but doesn't work.
> legA:ready() is true when the first session start and not when the call is
> answered.
> So legB is called at the same time.
>
> Any other suggestion?
>
>
>
>
> -----Messaggio originale-----
> Da: Sergey Scheglov <sid at eltc.ru>
> Inviato il: 13 Set 2011 - 17:15
> A: freeswitch-users at lists.freeswitch.org
>
>
>
> ? 13 Sep 2011 09:39:33 -0000
> chrisbware at interfree.it wrote:
>
> >
> > Hi guys,
> >
> > I've read anything on this argument on Wiki but I can't find a
> > complete answer.
> >
> > I need to bridge two calls in Lua, waiting the first to be answered.
> > My basic script is:
> >
> > local GwParams =
> >
> "origination_caller_id_number="..gateway..",sip_auth_username="..gateway..",sip_auth_password="..sip_passwd
> > local slegA =
> >
> "[ignore_early_media=false,"..GwParams.."]sofia/external/"..callee.."@"..sip_domain
> > local slegB =
> >
> "[ignore_early_media=true,"..GwParams.."]sofia/external/"..called.."@"..sip_domain
> >
> > legA = freeswitch.Session(slegA);
> > legB = freeswitch.Session(slegB);
> > freeswitch.bridge(legA, legB);
> >
> >
> > It works but legA and legB are called at the same time. Using:
> >
> > while (legA:answered()== false) do end;
> >
> > between the two session do the job but called parties can't hear
> > audio.
> >
> > I promise to add a script on wiki if you help me ! :)
> >
> > Thanks in advance.
> >
> >
> >
> -------------------------------------------------------------------------------
> > Valore legale alle tue mail
> > InterfreePEC - la tua Posta Elettronica Certificata
> > http://pec.interfree.it
> >
> -------------------------------------------------------------------------------
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> Try:
>
> legA = freeswitch.Session(slegA);
> if (legA:ready()) then
> legB = freeswitch.Session(slegB, legA);
> freeswitch.bridge(legA, legB);
> end
>
> ----
>
> Scheglov Sergey
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> -------------------------------------------------------------------------------
> Valore legale alle tue mail
> InterfreePEC - la tua Posta Elettronica Certificata
> http://pec.interfree.it
>
> -------------------------------------------------------------------------------
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wayne at hamilton.net  Tue Sep 13 22:34:16 2011
From: wayne at hamilton.net (Wayne)
Date: Tue, 13 Sep 2011 13:34:16 -0500
Subject: [Freeswitch-users] Sending DTMF on B-leg
In-Reply-To: <CAKzWOxXVFFhB25Za9xdmkcTWm-76nFfX7hjn5JFU6gfX+Jxyvg@mail.gmail.com>
References: <2934141FC0D9453B9150F7BD307120F3@ccs.local><CALNZuwkMZoRx-HbRBkHg5FSutwV2xW2GWiQQEz2dsmAh2WoPhA@mail.gmail.com><9B18FF40055B495BBAE3EDEEDB7388B0@ccs.local>
	<CAKzWOxXVFFhB25Za9xdmkcTWm-76nFfX7hjn5JFU6gfX+Jxyvg@mail.gmail.com>
Message-ID: <F9A3AD8DC6C3490CAE3F9365B98D1359@ccs.local>

That's true I don't hear any tones when I answer the phone. I don't see any
messages in my log from queue_dtmf. Any help would be great.
 
Wayne
 
 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, September 08, 2011 6:25 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Sending DTMF on B-leg


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
<tel:%289025556747> )" >
       <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 <tel:1%2F9025556747> 
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 <tel:1%2F9025556747> 
 
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




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From jeff at jefflenk.com  Tue Sep 13 22:45:06 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 13 Sep 2011 11:45:06 -0700 (PDT)
Subject: [Freeswitch-users] How to use sofia_count_reg ?
In-Reply-To: <CALNZuwmqacB1gca-FAtY38rHV4o5t62AYUiMFrKdWkf+MP=C8g@mail.gmail.com>
References: <CALNZuw=B3YYD1eWEMKERWtpitQBDv-AJKgw03k4mGCdZDfOQtA@mail.gmail.com>
	<1315499174093-6772541.post@n2.nabble.com>
	<CALNZuwmi1GKxDc9EWPg7+rmY9mEF1kTEYDP9xgtc8NSajmNVRA@mail.gmail.com>
	<1315579669278-6776183.post@n2.nabble.com>
	<1315929883123-6788105.post@n2.nabble.com>
	<CALNZuwmqacB1gca-FAtY38rHV4o5t62AYUiMFrKdWkf+MP=C8g@mail.gmail.com>
Message-ID: <1315939506076-6788796.post@n2.nabble.com>

Thanks for testing!

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-use-sofia-count-reg-tp6770571p6788796.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From aakashviswam at gmail.com  Tue Sep 13 23:09:55 2011
From: aakashviswam at gmail.com (Aakash)
Date: Tue, 13 Sep 2011 12:09:55 -0700 (PDT)
Subject: [Freeswitch-users] call forwarding how to test
Message-ID: <1315940995184-6788875.post@n2.nabble.com>

Hi,

I am trying to call forward from one extension to another extension.I
configured default.xml and features.xml by following below link

http://wiki.freeswitch.org/wiki/Call_Forward_Example

Can anyone help me how do i test or activate the call forwarding in the
extension.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/call-forwarding-how-to-test-tp6788875p6788875.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From chad at apartmentlines.com  Tue Sep 13 23:24:46 2011
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Tue, 13 Sep 2011 15:24:46 -0400
Subject: [Freeswitch-users] Lua API for CouchDB
Message-ID: <C48166F7-EB91-4466-BEDC-B79A42073D1F@apartmentlines.com>

For those interested in connecting to CouchDB (http://couchdb.apache.org) from FreeSWITCH, I've written a Lua API (http://www.lua.org) for it, available here:

https://github.com/thehunmonkgroup/luchia

With Lua being the recommended scripting language for FreeSWITCH, this seems an excellent way to work with Couch.

For those that have the LuaRocks package manager installed, it's a simple 'luarocks install luchia' to install.

I'm interested in exploring CouchDB as the datastore for a distributed, fault tolerant voicemail system -- I think it's attachment feature and bi-directional synchronization are well suited for the task.

Chad

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From msc at freeswitch.org  Tue Sep 13 23:41:06 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Sep 2011 12:41:06 -0700
Subject: [Freeswitch-users] Sending DTMF on B-leg
In-Reply-To: <F9A3AD8DC6C3490CAE3F9365B98D1359@ccs.local>
References: <2934141FC0D9453B9150F7BD307120F3@ccs.local>
	<CALNZuwkMZoRx-HbRBkHg5FSutwV2xW2GWiQQEz2dsmAh2WoPhA@mail.gmail.com>
	<9B18FF40055B495BBAE3EDEEDB7388B0@ccs.local>
	<CAKzWOxXVFFhB25Za9xdmkcTWm-76nFfX7hjn5JFU6gfX+Jxyvg@mail.gmail.com>
	<F9A3AD8DC6C3490CAE3F9365B98D1359@ccs.local>
Message-ID: <CAKzWOxX8tJbPubs=eFaUmajirOKQBYv9WtBp3HNa2vHPKRWvug@mail.gmail.com>

On Tue, Sep 13, 2011 at 11:34 AM, Wayne <wayne at hamilton.net> wrote:

> **
> That's true I don't hear any tones when I answer the phone. I don't see any
> messages in my log from queue_dtmf. Any help would be great.
>
> Wayne
>
Okay, in your original post you said you are making an OB call on a SIP
trunk, yet in your example there is no SIP trunk. You've got an internal
user and a FreeTDM FXO port. Can you please clarify what you actually have?
Who's the caller and who's the callee?
-MC



>
>
>
>  ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Thursday, September 08, 2011 6:25 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Sending DTMF on B-leg
>
> 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
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ijurado at econcept.es  Tue Sep 13 23:46:10 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Tue, 13 Sep 2011 21:46:10 +0200
Subject: [Freeswitch-users] txfax() over loopback
Message-ID: <CABj_V-mrmhZriMe=ibSdqWu1KcaTRHv+thOZqrCgye1LN44YyQ@mail.gmail.com>

Hi,

I've been trying to combine mod_loopback and mod_spandsp very similarly
to the way it is explained here:

    http://jira.freeswitch.org/browse/FS-3494

However, I cannot manage to make any case work, except for the direct
one not using loopback.

I've checked the issue's configuration files and all, and they don't seem
to hide any particular variable.

I'm using a git checkout from may 30th of this year.  The issue has been
fixed in august.  However, I was wondering if between may and august
this scenario was not supposed to work.

Cheers.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From msc at freeswitch.org  Wed Sep 14 00:03:14 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Sep 2011 13:03:14 -0700
Subject: [Freeswitch-users] Timeout Message or something similar
In-Reply-To: <1315907331256-6786830.post@n2.nabble.com>
References: <1315837848697-6783735.post@n2.nabble.com>
	<CAMwZ-tK6GT7EdtCFdZqNTahfGp1BHt45jBUXt68eGfbKv7TBig@mail.gmail.com>
	<1315907331256-6786830.post@n2.nabble.com>
Message-ID: <CAKzWOxUszOu8xGn70+UtaCOtqFC=WvEObORE+vWpdFmXKXSLgA@mail.gmail.com>

What you're trying to do is quite possible, however you need to take a
slightly different approach. The issue is that when you are continuing you
need somewhere to go. You can't have anything but a dialstring inside of
bridge, thus the "&playback" attempt won't work. Try this trick using the
"inline" dialplan:

/originate
}sofia/DOMAIN/1001
bridge:{fail_on_single_reject=true,ignore_early_media=true,originate_continue_on_timeout=true}sofia/DOMAIN/1002,playback:C:/teste2.wav
inline

Note that I moved the necessary parameters over to the b leg since that's
where you want the "continue" to happen. Remember too, that the first
argument to "originate" is a dialstring but the second argument is a
"dialplan" entry. You could just as easily do this:

originate {ringback=c:/teste.wav}sofia/DOMAIN/1001 TRY_1002 XML default

And then add this extension somewhere in default.xml or in
conf/dialplan/default/foo.xml:
<extension name="try to dial an extension">
  <condition field="destination_number" expression="^TRY_(\d+)$">
    <action application="bridge"
data="{fail_on_single_reject=true,ignore_early_media=true,originate_continue_on_timeout=true}sofia/DOMAIN/$1"/>
    <action application="playback" data="C:/teste2.wav"/>
    <action application="hangup"/>
  </condition>
</extension>

Try it out and report back to let us know that you have it all working.
-MC

On Tue, Sep 13, 2011 at 2:48 AM, Brtantunes <brtantunes at gmail.com> wrote:

> Hello Gabriel,
>
> Thanks for your help. I'm new with freeswitch and didn't know about the
> failover.
>
> i tried
> /originate
>
> {ringback=c:/teste.wav,fail_on_single_reject=true,ignore_early_media=true,originate_continue_on_timeout=true}sofia/DOMAIN/1001
> &bridge(sofia/DOMAIN/1002|sofia/DOMAIN/1003)/ and it works fine.
> Now the problem is on the failover pass the not the bridge to the other
> number. I tried /originate
>
> {ringback=c:/teste.wav,fail_on_single_reject=true,ignore_early_media=true,originate_continue_on_timeout=true}sofia/
> 10.101.81.71/1001
> &bridge(sofia/10.101.81.71/1002)|&playback(C:/teste2.wav)/<http://10.101.81.71/1001%0A&bridge%28sofia/10.101.81.71/1002%29%7C&playback%28C:/teste2.wav%29/>but obviously it
> didn't work :(.
>
> Any sugestions?
>
> Tks
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Timeout-Message-or-something-similar-tp6783735p6786830.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Sep 14 00:07:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Sep 2011 13:07:53 -0700
Subject: [Freeswitch-users] txfax() over loopback
In-Reply-To: <CABj_V-mrmhZriMe=ibSdqWu1KcaTRHv+thOZqrCgye1LN44YyQ@mail.gmail.com>
References: <CABj_V-mrmhZriMe=ibSdqWu1KcaTRHv+thOZqrCgye1LN44YyQ@mail.gmail.com>
Message-ID: <CAKzWOxWrCC1igEi59sd9x_cC2_tb=VMTu6Wku4iM0Ybx6CJn2w@mail.gmail.com>

You are much better off with the latest software. If you are in a production
environment where updating is problematic then definitely get a test box and
update it and iron out all the kinks. If you can't get a test box then make
a really, really good backup of your production system and then update &
test. Trying to patch an old version of the software is a maintenance
nightmare...

-MC

On Tue, Sep 13, 2011 at 12:46 PM, Isaac Jurado <ijurado at econcept.es> wrote:

> Hi,
>
> I've been trying to combine mod_loopback and mod_spandsp very similarly
> to the way it is explained here:
>
>    http://jira.freeswitch.org/browse/FS-3494
>
> However, I cannot manage to make any case work, except for the direct
> one not using loopback.
>
> I've checked the issue's configuration files and all, and they don't seem
> to hide any particular variable.
>
> I'm using a git checkout from may 30th of this year.  The issue has been
> fixed in august.  However, I was wondering if between may and august
> this scenario was not supposed to work.
>
> Cheers.
>
> --
> Isaac Jurado
> Internet Busines Solutions eConcept
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From covici at ccs.covici.com  Wed Sep 14 00:15:49 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 13 Sep 2011 16:15:49 -0400
Subject: [Freeswitch-users] problem using mod_managed under linux
In-Reply-To: <CAAtQP54za1xEuT9HzT-sWOjbzzuKcV-FiotHta_ZXEw7y9vFgQ@mail.gmail.com>
References: <rjxwrb84m9olrbg6m7ghyb97.1314822061391@email.android.com>
	<20320.1314825366@ccs.covici.com>
	<CAAtQP55W982E1NvDYYZ5_moKk9aGdddU2j4pDdR-Ncy5JodYTg@mail.gmail.com>
	<30786.1315879273@ccs.covici.com>
	<CAAtQP54za1xEuT9HzT-sWOjbzzuKcV-FiotHta_ZXEw7y9vFgQ@mail.gmail.com>
Message-ID: <28590.1315944949@ccs.covici.com>

OK, it was actually in mono that the error occurred, but last couple of
times, it did work, but I didn't change very much.  I hope it get fixed
soon.  The end of the stack was libc.so which has always been suspect
for me.

Thanks again for your response.



R H <buscom123+fs at gmail.com> wrote:

> Hey Covici,
> 
> I currently have a DLL that uses 3 IAppPlugin interfaces, 2 IApiPlugin
> interfaces, and 2 ILoadNotificationPlugin interfaces. I build the dll on a
> separate machine but using this system I am able to deploy the dll to a
> running FreeSwitch and have it unload/reload the dll on the fly without any
> problems.Its been working great for a week or two. A few things you may want
> to keep in mind:
> 
> 1) DLL's deployed in the managed directory are not exactly going to work
> well if they have dependencies on each-other as you never know which one
> will get loaded first. (Well, if you only drop them in 1 at a time it would
> probably work)
> 
> 2) Any crash in FreeSwitch I have seen is because of the above issue, or
> because of some uncaught exception thrown from my code. If you look at
> your FreeSwitch logs or start FreeSwitch by hand and watch the cli you
> should see the exception that killed it.
> 
> When I run FreeSwitch as a service even my script in /etc/init.d/ has to cd
> into the mod directory and then start the server. Hope this helps!
> 
> Ryan
> 
> 
> On Mon, Sep 12, 2011 at 8:01 PM, <covici at ccs.covici.com> wrote:
> 
> > Well, I spoke too soon.  It works, but if I make a change and try to
> > reload the .dll, it kills fs altogether -- makes it not too usable.
> >
> > R H <buscom123+fs at gmail.com> wrote:
> >
> > > Hey Guys,
> > >
> > > I just noticed this thread in the email queue so this may be coming too
> > > late. I have the same problem and sent an email about it to this group
> > with
> > > the subject line: *Mod_Managed Mono is not following the system LD Path.*
> > > *
> > > *
> > > That said,
> > > I have mod_managed working but it is a bit of a hack to do so. I simply
> > CD
> > > into the mod directory and then start the server from there.
> > >
> > > EG.
> > >
> > > cd /usr/local/freeswitch/mod
> > > ../bin/freeswitch
> > >
> > > This makes that particular problem go away. I have tried changing the LD
> > > Path, configured mono per the instructions on the wiki, etc? None of that
> > > has worked for me but cd'ing into the mod directory and then starting the
> > > server at least gets me running.
> > >
> > > This might be useful to everyone while we wait for this to be resolved. I
> > > have yet to add this as a bug in Jira. I was attempting to get some
> > response
> > > here before doing so.
> > >
> > > Ryan
> > >
> > > On Wed, Aug 31, 2011 at 3:16 PM, <covici at ccs.covici.com> wrote:
> > >
> > > > 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.
> > > >
> > >
> > > ----------------------------------------------------
> > > Alternatives:
> > >
> > > ----------------------------------------------------
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From covici at ccs.covici.com  Wed Sep 14 00:18:24 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 13 Sep 2011 16:18:24 -0400
Subject: [Freeswitch-users] problem using mod_managed under linux
In-Reply-To: <CAAtQP56+7hAThRmb6RqzA45i6ok83TXS-vu6-k2uHt9DE1OTpg@mail.gmail.com>
References: <rjxwrb84m9olrbg6m7ghyb97.1314822061391@email.android.com>
	<20320.1314825366@ccs.covici.com>
	<CAAtQP55W982E1NvDYYZ5_moKk9aGdddU2j4pDdR-Ncy5JodYTg@mail.gmail.com>
	<30786.1315879273@ccs.covici.com>
	<CAAtQP56+7hAThRmb6RqzA45i6ok83TXS-vu6-k2uHt9DE1OTpg@mail.gmail.com>
Message-ID: <28951.1315945104@ccs.covici.com>

I am using gentoo, so the mono version  is 2.10.4.  Also, the glibc
library is 2.13.4 which may make some difference.

R H <buscom123+fs at gmail.com> wrote:

> I guess I should also mention that I am running the following:
> 
> - 2 FreeSwitch Servers (1 Production, 1 Development).
> - Both run Suse 11.4 64bit, Mono 2.8.2, FreeSwitch at HEAD.
> - I also have to keep a softlink at /lib64/libc.so (although these boxes are
> dedicated FreeSwitch only)
> - I had to run the 2.8patch on the mod_managed library before build to make
> all this work
> 
> Aside from this issue mod_managed seems to run fairly well, but I have not
> tested it under load. All-in-all it has done well for me so far.
> 
> Ryan
> 
> On Mon, Sep 12, 2011 at 8:01 PM, <covici at ccs.covici.com> wrote:
> 
> > Well, I spoke too soon.  It works, but if I make a change and try to
> > reload the .dll, it kills fs altogether -- makes it not too usable.
> >
> > R H <buscom123+fs at gmail.com> wrote:
> >
> > > Hey Guys,
> > >
> > > I just noticed this thread in the email queue so this may be coming too
> > > late. I have the same problem and sent an email about it to this group
> > with
> > > the subject line: *Mod_Managed Mono is not following the system LD Path.*
> > > *
> > > *
> > > That said,
> > > I have mod_managed working but it is a bit of a hack to do so. I simply
> > CD
> > > into the mod directory and then start the server from there.
> > >
> > > EG.
> > >
> > > cd /usr/local/freeswitch/mod
> > > ../bin/freeswitch
> > >
> > > This makes that particular problem go away. I have tried changing the LD
> > > Path, configured mono per the instructions on the wiki, etc? None of that
> > > has worked for me but cd'ing into the mod directory and then starting the
> > > server at least gets me running.
> > >
> > > This might be useful to everyone while we wait for this to be resolved. I
> > > have yet to add this as a bug in Jira. I was attempting to get some
> > response
> > > here before doing so.
> > >
> > > Ryan
> > >
> > > On Wed, Aug 31, 2011 at 3:16 PM, <covici at ccs.covici.com> wrote:
> > >
> > > > 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.
> > > >
> > >
> > > ----------------------------------------------------
> > > Alternatives:
> > >
> > > ----------------------------------------------------
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From gcd at i.ph  Wed Sep 14 00:49:27 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Wed, 14 Sep 2011 04:49:27 +0800
Subject: [Freeswitch-users] call forwarding how to test
In-Reply-To: <1315940995184-6788875.post@n2.nabble.com>
References: <1315940995184-6788875.post@n2.nabble.com>
Message-ID: <CAHKqMb6ZFpj5GH5Wj8XQkNQjZUu3zXeOjNbbCHcJVU3yUB0u3Q@mail.gmail.com>

you need 3 extension phones - A, B and C. if u want B to forward calls to C,
dial *72 on B. you will hear a dialtone. dial the extension number of C
followed by #.

from A, dial B. C should ring.

to disable call forwarding, dial *73 on B then hangup.

from A, dial B. B, this time, should ring.

-nandy

On Wed, Sep 14, 2011 at 3:09 AM, Aakash <aakashviswam at gmail.com> wrote:

> Hi,
>
> I am trying to call forward from one extension to another extension.I
> configured default.xml and features.xml by following below link
>
> http://wiki.freeswitch.org/wiki/Call_Forward_Example
>
> Can anyone help me how do i test or activate the call forwarding in the
> extension.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/call-forwarding-how-to-test-tp6788875p6788875.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From chrisbware at interfree.it  Wed Sep 14 00:59:18 2011
From: chrisbware at interfree.it (Chrisbware)
Date: Tue, 13 Sep 2011 22:59:18 +0200
Subject: [Freeswitch-users] Re4: Lua "two steps" bridging
In-Reply-To: <CAKzWOxW2Dwva26ZBdJ4gy9ChT6coGNsq8BJBvPFvWSG2kpK+BQ@mail.gmail.com>
References: <20110913122138.15531.qmail@community17.interfree.it>
	<CAKzWOxW2Dwva26ZBdJ4gy9ChT6coGNsq8BJBvPFvWSG2kpK+BQ@mail.gmail.com>
Message-ID: <4E6FC426.9050805@interfree.it>

Hi Michael,

a PHP script connect to Event socket and launch "luarun  call.lua <legA
number> <legB number> <sip account number>

Script call.lua, upload sip account password from a DB and then call A
number and B number using the same SIP account, bridging them.
What I'd like to do is:

- place call to legB number only after legA has answered the call
- detect hangupCause on both legs to check if something goes wrong
- return SUCCESS or hangup cause to the PHP script

Is it science fiction? With Asterisk isn't so hard to do and, since I'm
in love with FS, please don't disappoint me :-)

Chris B.
 

Il 13/09/2011 18:44, Michael Collins ha scritto:
> How are you calling this Lua script to begin with? What is the big
> picture, that is, what is the problem you're solving? Some context
> might help us give you a better answer.
>
> -MC
>
> On Tue, Sep 13, 2011 at 5:21 AM, <chrisbware at interfree.it
> <mailto:chrisbware at interfree.it>> wrote:
>
>     Hi,
>      
>     thank you for your help, but doesn't work.
>     legA:ready() is true when the first session start and not when the
>     call is answered.
>     So legB is called at the same time.
>      
>     Any other suggestion?
>      
>      
>      
>      
>     -----Messaggio originale-----
>     Da: Sergey Scheglov <sid at eltc.ru <mailto:sid at eltc.ru>>
>     Inviato il: 13 Set 2011 - 17:15
>     A: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>
>
>
>     ? 13 Sep 2011 09:39:33 -0000
>     chrisbware at interfree.it <mailto:chrisbware at interfree.it> wrote:
>
>     >
>     > Hi guys,
>     >
>     > I've read anything on this argument on Wiki but I can't find a
>     > complete answer.
>     >
>     > I need to bridge two calls in Lua, waiting the first to be answered.
>     > My basic script is:
>     >
>     > local GwParams =
>     > "origination_caller_id_number="..gateway..",sip_auth_username="..gateway..",sip_auth_password="..sip_passwd
>     > local slegA =
>     > "[ignore_early_media=false,"..GwParams.."]sofia/external/"..callee.."@"..sip_domain
>     > local slegB =
>     > "[ignore_early_media=true,"..GwParams.."]sofia/external/"..called.."@"..sip_domain
>     >
>     > legA = freeswitch.Session(slegA);
>     > legB = freeswitch.Session(slegB);
>     > freeswitch.bridge(legA, legB);
>     >
>     >
>     > It works but legA and legB are called at the same time. Using:
>     >
>     > while (legA:answered()== false) do end;
>     >
>     > between the two session do the job but called parties can't hear
>     > audio.
>     >
>     > I promise to add a script on wiki if you help me ! :)
>     >
>     > Thanks in advance.
>     >
>     >
>     > -------------------------------------------------------------------------------
>     > Valore legale alle tue mail
>     > InterfreePEC - la tua Posta Elettronica Certificata
>     > http://pec.interfree.it
>     > -------------------------------------------------------------------------------
>     >
>     >
>     >
>     > 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
>     >
>
>     Try:
>
>     legA = freeswitch.Session(slegA);
>     if (legA:ready()) then
>     legB = freeswitch.Session(slegB, legA);
>     freeswitch.bridge(legA, legB);
>     end
>
>     ----
>
>     Scheglov Sergey
>
>
>     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
>
>     -------------------------------------------------------------------------------
>     Valore legale alle tue mail
>     InterfreePEC - la tua Posta Elettronica Certificata
>     http://pec.interfree.it
>     -------------------------------------------------------------------------------
>
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From aakashviswam at gmail.com  Wed Sep 14 01:04:05 2011
From: aakashviswam at gmail.com (Aakash)
Date: Tue, 13 Sep 2011 14:04:05 -0700 (PDT)
Subject: [Freeswitch-users] call forwarding how to test
In-Reply-To: <CAHKqMb6ZFpj5GH5Wj8XQkNQjZUu3zXeOjNbbCHcJVU3yUB0u3Q@mail.gmail.com>
References: <1315940995184-6788875.post@n2.nabble.com>
	<CAHKqMb6ZFpj5GH5Wj8XQkNQjZUu3zXeOjNbbCHcJVU3yUB0u3Q@mail.gmail.com>
Message-ID: <1315947845640-6789287.post@n2.nabble.com>

Hi Nandy,

 When i try to dialed the extension number of C followed by #. It shoe san
error message in temporvary unavailable.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/call-forwarding-how-to-test-tp6788875p6789287.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From wayne at hamilton.net  Wed Sep 14 01:16:05 2011
From: wayne at hamilton.net (Wayne)
Date: Tue, 13 Sep 2011 16:16:05 -0500
Subject: [Freeswitch-users] Sending DTMF on B-leg
In-Reply-To: <CAKzWOxX8tJbPubs=eFaUmajirOKQBYv9WtBp3HNa2vHPKRWvug@mail.gmail.com>
References: <2934141FC0D9453B9150F7BD307120F3@ccs.local><CALNZuwkMZoRx-HbRBkHg5FSutwV2xW2GWiQQEz2dsmAh2WoPhA@mail.gmail.com><9B18FF40055B495BBAE3EDEEDB7388B0@ccs.local><CAKzWOxXVFFhB25Za9xdmkcTWm-76nFfX7hjn5JFU6gfX+Jxyvg@mail.gmail.com><F9A3AD8DC6C3490CAE3F9365B98D1359@ccs.local>
	<CAKzWOxX8tJbPubs=eFaUmajirOKQBYv9WtBp3HNa2vHPKRWvug@mail.gmail.com>
Message-ID: <C57A15E034154A9297EE8DF5EF752971@ccs.local>

Sorry Michael,
 
When I started out I was sending the calls to a SIP provider. I couldn't
here the DTMF tones. For trouble shooting I moved it to a local PRI. I was
thinking that maybe this was an issue with the sip trunk and sending inband
or out of band but that doesn't seem to be the case.
 
Wayne

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, September 13, 2011 2:41 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Sending DTMF on B-leg




On Tue, Sep 13, 2011 at 11:34 AM, Wayne <wayne at hamilton.net> wrote:



That's true I don't hear any tones when I answer the phone. I don't see any
messages in my log from queue_dtmf. Any help would be great.
 
Wayne


Okay, in your original post you said you are making an OB call on a SIP
trunk, yet in your example there is no SIP trunk. You've got an internal
user and a FreeTDM FXO port. Can you please clarify what you actually have?
Who's the caller and who's the callee?
-MC

 


 
 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, September 08, 2011 6:25 PM 

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Sending DTMF on B-leg


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
<tel:%289025556747> )" >
       <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 <tel:1%2F9025556747> 
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 <tel:1%2F9025556747> 
 
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
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From msc at freeswitch.org  Wed Sep 14 01:27:09 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Sep 2011 14:27:09 -0700
Subject: [Freeswitch-users] Re4: Lua "two steps" bridging
In-Reply-To: <4E6FC426.9050805@interfree.it>
References: <20110913122138.15531.qmail@community17.interfree.it>
	<CAKzWOxW2Dwva26ZBdJ4gy9ChT6coGNsq8BJBvPFvWSG2kpK+BQ@mail.gmail.com>
	<4E6FC426.9050805@interfree.it>
Message-ID: <CAKzWOxWNW=eL8=_fQC8J-whYi5cMrur2=Odxd7cnaZD_6Pz4+w@mail.gmail.com>

Chris,

We're glad you like FS - we like it as well.

What you're trying to do is completely possible, but the way you're trying
to do it is probably not the best choice. Instead of a local Lua script to
control both calls, you are better off using either one of two options:
event socket entirely or event socket + dialplan. It depends on the level of
control that you need. If you want ultimate control then just use ESL right
from PHP (or whichever scripting language you are comfortable with) and
create your call legs, watching for the requisite CHANNEL_XXX events. (See
http://wiki.freeswitch.org/wiki/Event_List for a nice list.)

I suppose you could do it all in Lua but you'd have to keep polling legA
channel for the endpoint_disposition variable in a while loop and then break
out on various conditions. I'm not at all a fan of that method. With the
event socket you listen for the events that are of interest to you and then
act on them. Very elegant.

-MC

On Tue, Sep 13, 2011 at 1:59 PM, Chrisbware <chrisbware at interfree.it> wrote:

> **
> Hi Michael,
>
> a PHP script connect to Event socket and launch "luarun  call.lua <legA
> number> <legB number> <sip account number>
>
> Script call.lua, upload sip account password from a DB and then call A
> number and B number using the same SIP account, bridging them.
> What I'd like to do is:
>
> - place call to legB number only after legA has answered the call
> - detect hangupCause on both legs to check if something goes wrong
> - return SUCCESS or hangup cause to the PHP script
>
> Is it science fiction? With Asterisk isn't so hard to do and, since I'm in
> love with FS, please don't disappoint me :-)
>
> Chris B.
>
>
> Il 13/09/2011 18:44, Michael Collins ha scritto:
>
> How are you calling this Lua script to begin with? What is the big picture,
> that is, what is the problem you're solving? Some context might help us give
> you a better answer.
>
>  -MC
>
>  On Tue, Sep 13, 2011 at 5:21 AM, <chrisbware at interfree.it> wrote:
>
>> Hi,
>>
>> thank you for your help, but doesn't work.
>> legA:ready() is true when the first session start and not when the call is
>> answered.
>> So legB is called at the same time.
>>
>> Any other suggestion?
>>
>>
>>
>>
>> -----Messaggio originale-----
>> Da: Sergey Scheglov <sid at eltc.ru>
>> Inviato il: 13 Set 2011 - 17:15
>> A: freeswitch-users at lists.freeswitch.org
>>
>>
>>
>> ? 13 Sep 2011 09:39:33 -0000
>> chrisbware at interfree.it wrote:
>>
>> >
>> > Hi guys,
>> >
>> > I've read anything on this argument on Wiki but I can't find a
>> > complete answer.
>> >
>> > I need to bridge two calls in Lua, waiting the first to be answered.
>> > My basic script is:
>> >
>> > local GwParams =
>> >
>> "origination_caller_id_number="..gateway..",sip_auth_username="..gateway..",sip_auth_password="..sip_passwd
>> > local slegA =
>> >
>> "[ignore_early_media=false,"..GwParams.."]sofia/external/"..callee.."@"..sip_domain
>> > local slegB =
>> >
>> "[ignore_early_media=true,"..GwParams.."]sofia/external/"..called.."@"..sip_domain
>> >
>> > legA = freeswitch.Session(slegA);
>> > legB = freeswitch.Session(slegB);
>> > freeswitch.bridge(legA, legB);
>> >
>> >
>> > It works but legA and legB are called at the same time. Using:
>> >
>> > while (legA:answered()== false) do end;
>> >
>> > between the two session do the job but called parties can't hear
>> > audio.
>> >
>> > I promise to add a script on wiki if you help me ! :)
>> >
>> > Thanks in advance.
>> >
>> >
>> >
>> -------------------------------------------------------------------------------
>> > Valore legale alle tue mail
>> > InterfreePEC - la tua Posta Elettronica Certificata
>> > http://pec.interfree.it
>> >
>> -------------------------------------------------------------------------------
>> >
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> Try:
>>
>> legA = freeswitch.Session(slegA);
>> if (legA:ready()) then
>> legB = freeswitch.Session(slegB, legA);
>> freeswitch.bridge(legA, legB);
>> end
>>
>> ----
>>
>> Scheglov Sergey
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> -------------------------------------------------------------------------------
>> Valore legale alle tue mail
>> InterfreePEC - la tua Posta Elettronica Certificata
>> http://pec.interfree.it
>>
>> -------------------------------------------------------------------------------
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Sep 14 01:28:56 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Sep 2011 14:28:56 -0700
Subject: [Freeswitch-users] Sending DTMF on B-leg
In-Reply-To: <C57A15E034154A9297EE8DF5EF752971@ccs.local>
References: <2934141FC0D9453B9150F7BD307120F3@ccs.local>
	<CALNZuwkMZoRx-HbRBkHg5FSutwV2xW2GWiQQEz2dsmAh2WoPhA@mail.gmail.com>
	<9B18FF40055B495BBAE3EDEEDB7388B0@ccs.local>
	<CAKzWOxXVFFhB25Za9xdmkcTWm-76nFfX7hjn5JFU6gfX+Jxyvg@mail.gmail.com>
	<F9A3AD8DC6C3490CAE3F9365B98D1359@ccs.local>
	<CAKzWOxX8tJbPubs=eFaUmajirOKQBYv9WtBp3HNa2vHPKRWvug@mail.gmail.com>
	<C57A15E034154A9297EE8DF5EF752971@ccs.local>
Message-ID: <CAKzWOxXJ7J+Pi2iart1nHC8-OmwHkZ0vjH3iXsKvAKGdNOt7=Q@mail.gmail.com>

Okay, fair enough. Not sure what queue_dtmf is not working. You could always
try "execute_on_answer" and execute an extension that sends the dtmfs.

-MC

On Tue, Sep 13, 2011 at 2:16 PM, Wayne <wayne at hamilton.net> wrote:

> **
> Sorry Michael,
>
> When I started out I was sending the calls to a SIP provider. I couldn't
> here the DTMF tones. For trouble shooting I moved it to a local PRI. I was
> thinking that maybe this was an issue with the sip trunk and sending inband
> or out of band but that doesn't seem to be the case.
>
> Wayne
>
>  ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Tuesday, September 13, 2011 2:41 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Sending DTMF on B-leg
>
>
>
> On Tue, Sep 13, 2011 at 11:34 AM, Wayne <wayne at hamilton.net> wrote:
>
>> **
>> That's true I don't hear any tones when I answer the phone. I don't see
>> any messages in my log from queue_dtmf. Any help would be great.
>>
>> Wayne
>>
> Okay, in your original post you said you are making an OB call on a SIP
> trunk, yet in your example there is no SIP trunk. You've got an internal
> user and a FreeTDM FXO port. Can you please clarify what you actually have?
> Who's the caller and who's the callee?
> -MC
>
>
>
>>
>>
>>
>>  ------------------------------
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
>> Collins
>> *Sent:* Thursday, September 08, 2011 6:25 PM
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] Sending DTMF on B-leg
>>
>>   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
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chrisbware at interfree.it  Wed Sep 14 01:52:29 2011
From: chrisbware at interfree.it (Chrisbware)
Date: Tue, 13 Sep 2011 23:52:29 +0200
Subject: [Freeswitch-users] Re4: Lua "two steps" bridging
In-Reply-To: <CAKzWOxWNW=eL8=_fQC8J-whYi5cMrur2=Odxd7cnaZD_6Pz4+w@mail.gmail.com>
References: <20110913122138.15531.qmail@community17.interfree.it>	<CAKzWOxW2Dwva26ZBdJ4gy9ChT6coGNsq8BJBvPFvWSG2kpK+BQ@mail.gmail.com>	<4E6FC426.9050805@interfree.it>
	<CAKzWOxWNW=eL8=_fQC8J-whYi5cMrur2=Odxd7cnaZD_6Pz4+w@mail.gmail.com>
Message-ID: <4E6FD09D.2070103@interfree.it>

Thank you for your answer: I'll follow your advice.

Anyway, IMHO, Lua must be able to manage situations like this. Languages
strong integration is a plus
of FS that helps programmers to invent new services. Telling them they
still have to play with events
sounds a bit "old school". Again: it's only my opinon. Keep up the good
work!

Il 13/09/2011 23:27, Michael Collins ha scritto:
> Chris,
>
> We're glad you like FS - we like it as well.
>
> What you're trying to do is completely possible, but the way you're
> trying to do it is probably not the best choice. Instead of a local
> Lua script to control both calls, you are better off using either one
> of two options: event socket entirely or event socket + dialplan. It
> depends on the level of control that you need. If you want ultimate
> control then just use ESL right from PHP (or whichever scripting
> language you are comfortable with) and create your call legs, watching
> for the requisite CHANNEL_XXX events.
> (See http://wiki.freeswitch.org/wiki/Event_List for a nice list.)
>
> I suppose you could do it all in Lua but you'd have to keep polling
> legA channel for the endpoint_disposition variable in a while loop and
> then break out on various conditions. I'm not at all a fan of that
> method. With the event socket you listen for the events that are of
> interest to you and then act on them. Very elegant.
>
> -MC
>
> On Tue, Sep 13, 2011 at 1:59 PM, Chrisbware <chrisbware at interfree.it
> <mailto:chrisbware at interfree.it>> wrote:
>
>     Hi Michael,
>
>     a PHP script connect to Event socket and launch "luarun  call.lua
>     <legA number> <legB number> <sip account number>
>
>     Script call.lua, upload sip account password from a DB and then
>     call A number and B number using the same SIP account, bridging them.
>     What I'd like to do is:
>
>     - place call to legB number only after legA has answered the call
>     - detect hangupCause on both legs to check if something goes wrong
>     - return SUCCESS or hangup cause to the PHP script
>
>     Is it science fiction? With Asterisk isn't so hard to do and,
>     since I'm in love with FS, please don't disappoint me :-)
>
>     Chris B.
>      
>
>     Il 13/09/2011 18:44, Michael Collins ha scritto:
>>     How are you calling this Lua script to begin with? What is the
>>     big picture, that is, what is the problem you're solving? Some
>>     context might help us give you a better answer.
>>
>>     -MC
>>
>>     On Tue, Sep 13, 2011 at 5:21 AM, <chrisbware at interfree.it
>>     <mailto:chrisbware at interfree.it>> wrote:
>>
>>         Hi,
>>          
>>         thank you for your help, but doesn't work.
>>         legA:ready() is true when the first session start and not
>>         when the call is answered.
>>         So legB is called at the same time.
>>          
>>         Any other suggestion?
>>          
>>          
>>          
>>          
>>         -----Messaggio originale-----
>>         Da: Sergey Scheglov <sid at eltc.ru <mailto:sid at eltc.ru>>
>>         Inviato il: 13 Set 2011 - 17:15
>>         A: freeswitch-users at lists.freeswitch.org
>>         <mailto:freeswitch-users at lists.freeswitch.org>
>>
>>
>>
>>         ? 13 Sep 2011 09:39:33 -0000
>>         chrisbware at interfree.it <mailto:chrisbware at interfree.it> wrote:
>>
>>         >
>>         > Hi guys,
>>         >
>>         > I've read anything on this argument on Wiki but I can't find a
>>         > complete answer.
>>         >
>>         > I need to bridge two calls in Lua, waiting the first to be
>>         answered.
>>         > My basic script is:
>>         >
>>         > local GwParams =
>>         > "origination_caller_id_number="..gateway..",sip_auth_username="..gateway..",sip_auth_password="..sip_passwd
>>         > local slegA =
>>         > "[ignore_early_media=false,"..GwParams.."]sofia/external/"..callee.."@"..sip_domain
>>         > local slegB =
>>         > "[ignore_early_media=true,"..GwParams.."]sofia/external/"..called.."@"..sip_domain
>>         >
>>         > legA = freeswitch.Session(slegA);
>>         > legB = freeswitch.Session(slegB);
>>         > freeswitch.bridge(legA, legB);
>>         >
>>         >
>>         > It works but legA and legB are called at the same time. Using:
>>         >
>>         > while (legA:answered()== false) do end;
>>         >
>>         > between the two session do the job but called parties can't hear
>>         > audio.
>>         >
>>         > I promise to add a script on wiki if you help me ! :)
>>         >
>>         > Thanks in advance.
>>         >
>>         >
>>         > -------------------------------------------------------------------------------
>>         > Valore legale alle tue mail
>>         > InterfreePEC - la tua Posta Elettronica Certificata
>>         > http://pec.interfree.it
>>         > -------------------------------------------------------------------------------
>>         >
>>         >
>>         >
>>         > 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
>>         >
>>
>>         Try:
>>
>>         legA = freeswitch.Session(slegA);
>>         if (legA:ready()) then
>>         legB = freeswitch.Session(slegB, legA);
>>         freeswitch.bridge(legA, legB);
>>         end
>>
>>         ----
>>
>>         Scheglov Sergey
>>
>>
>>         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
>>
>>         -------------------------------------------------------------------------------
>>         Valore legale alle tue mail
>>         InterfreePEC - la tua Posta Elettronica Certificata
>>         http://pec.interfree.it
>>         -------------------------------------------------------------------------------
>>
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From peter.schrock at gmail.com  Wed Sep 14 01:59:48 2011
From: peter.schrock at gmail.com (Peter Schrock)
Date: Tue, 13 Sep 2011 14:59:48 -0700
Subject: [Freeswitch-users] No Rule to make target 'tport/libtport.la'
In-Reply-To: <CA+PZoDS4iJhrS=6Z91y5NCeeQ4a1q1+H+hQBNK-DC97X+BZCgQ@mail.gmail.com>
References: <CA+PZoDS4iJhrS=6Z91y5NCeeQ4a1q1+H+hQBNK-DC97X+BZCgQ@mail.gmail.com>
Message-ID: <CA+PZoDRxh5Z=Dx_zMzRu6BGBUOYtGDjJn+cjGMes2OodGF13Cw@mail.gmail.com>

Okay, so I think I know the root of the problem but am not aware of how to
solve the problem. This is the error message:

tport_logging.c: In function 'tport_open_log':
tport_logging.c:209: error: 'AI_NUMERICSERV' undeclared (first use in this
function)
tport_logging.c:209: error: (Each undeclared identifier is reported only
once
tport_logging.c:209: error: for each function it appears in.)
make[9]: *** [tport_logging.lo] Error 1
make[8]: *** [all] Error 2

Anyone know anything that can help?

Peter

On Sun, Sep 11, 2011 at 11:49 PM, Peter Schrock <peter.schrock at gmail.com>wrote:

> So, I figured out how to fix the 'swab' problem using a powerpc. Now, I am
> getting this error message:
>
> LINK libnua.la
> make[8]: *** No rule to make target 'tport/libtport.la', needed by '
> libsofia-sip-us.la'. Stop.
>
> Does any one have any ideas?
>
> Peter
>
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From msc at freeswitch.org  Wed Sep 14 02:02:40 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Sep 2011 15:02:40 -0700
Subject: [Freeswitch-users] Lua XML parser
In-Reply-To: <CD5A1499C8DEF34FAA3509722B00393E74029E2ECA@m-mail03.office.premiertech>
References: <CD5A1499C8DEF34FAA3509722B00393E74029E2ECA@m-mail03.office.premiertech>
Message-ID: <CAKzWOxXjzWA6pH42OzBV9t0Y6Tz4OsreQrEsskh9skb6tE5=dQ@mail.gmail.com>

I discuss basic pattern matching in Lua in chapter 7 of our book. I'm
thinking you'll need to do something like this. Assume that variable
"nuance_res" contains the blurb that you posted below.

freeswitch.consoleLog("INFO","Result is:\n" .. nuance_res .. "\n")





confidence = string.gsub(nuance_res,'.-confidence="(0\.%d%d).+',"%1")


freeswitch.consoleLog("NOTICE","Confidence: " .. confidence .. "\n")





input = string.gsub(nuance_res,'.-<input mode=".-">(.-)<.+',"%1")


freeswitch.consoleLog("NOTICE","Input: " .. input .. "\n")





instance = string.gsub(nuance_res,'.-<instance>(.-)<.+',"%1")


freeswitch.consoleLog("NOTICE","Instance: " .. instance .. "\n")

-MC

On Sun, Sep 11, 2011 at 11:26 PM, Glen Ganderton <
Glen.Ganderton at premier.com.au> wrote:

> Hey Guys,****
>
> ** **
>
> I?m currently trying to create a basic IVR with voice recognition (just a
> simple YES or NO response). I have successfully configured the freeswitch
> MRCP client to connect to my nuance MRCP server and I successfully get a
> responses from the server. Now what I need to do is process the response in
> my freeswitch lua script. The nuance box returns the following:****
>
> ** **
>
>
> ----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
> ****
>
> Completion-Cause: 000 success****
>
> Content-Type: application/nlsml+xml****
>
> Content-Length: 178****
>
> ** **
>
> <?xml version='1.0'?><result><interpretation
> grammar="session:nuance5-mrcp2" confidence="0.99"><input
> mode="speech">yes</input><instance>true</instance></interpretation></result>
> ****
>
> ** **
>
> Now what I want to be able to do is parse the XML in my lua script so that
> I can simply have the XML stored in some variables:****
>
> ** **
>
> Eg.****
>
> ** **
>
> Confidence = 0.99****
>
> Input = yes****
>
> Instance =  true****
>
> ?.etc.****
>
> ** **
>
> I am just starting to learn lua and am unable to do this myself. I have
> seen a few scripts online to parse XML for lua but none seem to work
> correctly (maybe they are meant for different format..not sure).****
>
> ** **
>
> If anybody could point me in the right direction or write some sample code
> for what I need to do that would be a great help.****
>
> ** **
>
> Thanks 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
>
>
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From gcd at i.ph  Wed Sep 14 03:11:56 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Wed, 14 Sep 2011 07:11:56 +0800
Subject: [Freeswitch-users] call forwarding how to test
In-Reply-To: <1315947845640-6789287.post@n2.nabble.com>
References: <1315940995184-6788875.post@n2.nabble.com>
	<CAHKqMb6ZFpj5GH5Wj8XQkNQjZUu3zXeOjNbbCHcJVU3yUB0u3Q@mail.gmail.com>
	<1315947845640-6789287.post@n2.nabble.com>
Message-ID: <CAHKqMb7YqYnMz8pv=Up+7OEoiPqLDDwKnYSHYmx5KsfZQYM3dw@mail.gmail.com>

did u get this message when activating call-forwarding *72? or when u dial C
from A (or B)?  your extension numbers should be at least 3 digits long.

On Wed, Sep 14, 2011 at 5:04 AM, Aakash <aakashviswam at gmail.com> wrote:

> Hi Nandy,
>
>  When i try to dialed the extension number of C followed by #. It shoe san
> error message in temporvary unavailable.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/call-forwarding-how-to-test-tp6788875p6789287.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Sep 14 03:35:18 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Sep 2011 16:35:18 -0700
Subject: [Freeswitch-users] Re4: Lua "two steps" bridging
In-Reply-To: <4E6FD09D.2070103@interfree.it>
References: <20110913122138.15531.qmail@community17.interfree.it>
	<CAKzWOxW2Dwva26ZBdJ4gy9ChT6coGNsq8BJBvPFvWSG2kpK+BQ@mail.gmail.com>
	<4E6FC426.9050805@interfree.it>
	<CAKzWOxWNW=eL8=_fQC8J-whYi5cMrur2=Odxd7cnaZD_6Pz4+w@mail.gmail.com>
	<4E6FD09D.2070103@interfree.it>
Message-ID: <CAKzWOxV-gJA=N1_EmwsXiFBf9G=S7J_GPArfp9YAwVjc+3THdg@mail.gmail.com>

Well, I didn't say you *couldn't* do it, just that it isn't the best option.
For posterity's sake here is a sample of a Lua script that does basically
what you want.

--


-- cc.lua


--


-- call control lua script - can it be done?


--





dialA = "sofia/gateway/fs1/9903"


dialB = "user/1001"





legA = freeswitch.Session(dialA)


dispoA = "None"





while(legA:ready() and dispoA ~= "ANSWER") do


  dispoA = legA:getVariable("endpoint_disposition")


  freeswitch.consoleLog("INFO","Leg A disposition is '" .. dispoA .. "'\n")


  os.execute("sleep 1")


end -- legA while





if( not legA:ready() ) then


  -- oops, lost leg A handle this case


  freeswitch.consoleLog("NOTICE","It appears that " .. dialA .. "
disconnected...\n")

else


  legB = freeswitch.Session(dialB)


  dispoB = "None"


  while(legA:ready() and legB:ready() and dispoB ~= "ANSWER") do


    if ( not legA:ready() ) then


      -- oops, leg A hung up or got disconnected, handle this case


      freeswitch.consoleLog("NOTICE","It appears that " .. dialA .. "
disconnected...\n")

    else


      os.execute("sleep 1")


      dispoB = legB:getVariable("endpoint_disposition")


      freeswitch.consoleLog("NOTICE","Leg B disposition is '" .. dispoB ..
"'\n")

    end -- inner if legA ready


  end -- legB while





  if ( legA:ready() and legB:ready() ) then


    freeswitch.bridge(legA,legB)


  else


    -- oops, one of the above legs hung up, handle this case


   freeswitch.consoleLog("NOTICE","It appears that " .. dialA .. " or " ..
dialB .. " disconnected...\n")

  end





end -- outter if legA ready




-MC

On Tue, Sep 13, 2011 at 2:52 PM, Chrisbware <chrisbware at interfree.it> wrote:

> **
> Thank you for your answer: I'll follow your advice.
>
> Anyway, IMHO, Lua must be able to manage situations like this. Languages
> strong integration is a plus
> of FS that helps programmers to invent new services. Telling them they
> still have to play with events
> sounds a bit "old school". Again: it's only my opinon. Keep up the good
> work!
>
> Il 13/09/2011 23:27, Michael Collins ha scritto:
>
> Chris,
>
>  We're glad you like FS - we like it as well.
>
>  What you're trying to do is completely possible, but the way you're trying
> to do it is probably not the best choice. Instead of a local Lua script to
> control both calls, you are better off using either one of two options:
> event socket entirely or event socket + dialplan. It depends on the level of
> control that you need. If you want ultimate control then just use ESL right
> from PHP (or whichever scripting language you are comfortable with) and
> create your call legs, watching for the requisite CHANNEL_XXX events. (See
> http://wiki.freeswitch.org/wiki/Event_List for a nice list.)
>
>  I suppose you could do it all in Lua but you'd have to keep polling legA
> channel for the endpoint_disposition variable in a while loop and then break
> out on various conditions. I'm not at all a fan of that method. With the
> event socket you listen for the events that are of interest to you and then
> act on them. Very elegant.
>
>  -MC
>
> On Tue, Sep 13, 2011 at 1:59 PM, Chrisbware <chrisbware at interfree.it>wrote:
>
>>  Hi Michael,
>>
>> a PHP script connect to Event socket and launch "luarun  call.lua <legA
>> number> <legB number> <sip account number>
>>
>> Script call.lua, upload sip account password from a DB and then call A
>> number and B number using the same SIP account, bridging them.
>> What I'd like to do is:
>>
>> - place call to legB number only after legA has answered the call
>> - detect hangupCause on both legs to check if something goes wrong
>> - return SUCCESS or hangup cause to the PHP script
>>
>> Is it science fiction? With Asterisk isn't so hard to do and, since I'm in
>> love with FS, please don't disappoint me :-)
>>
>> Chris B.
>>
>>
>> Il 13/09/2011 18:44, Michael Collins ha scritto:
>>
>> How are you calling this Lua script to begin with? What is the big
>> picture, that is, what is the problem you're solving? Some context might
>> help us give you a better answer.
>>
>>  -MC
>>
>>  On Tue, Sep 13, 2011 at 5:21 AM, <chrisbware at interfree.it> wrote:
>>
>>> Hi,
>>>
>>> thank you for your help, but doesn't work.
>>> legA:ready() is true when the first session start and not when the call
>>> is answered.
>>> So legB is called at the same time.
>>>
>>> Any other suggestion?
>>>
>>>
>>>
>>>
>>> -----Messaggio originale-----
>>> Da: Sergey Scheglov <sid at eltc.ru>
>>> Inviato il: 13 Set 2011 - 17:15
>>> A: freeswitch-users at lists.freeswitch.org
>>>
>>>
>>>
>>> ? 13 Sep 2011 09:39:33 -0000
>>> chrisbware at interfree.it wrote:
>>>
>>> >
>>> > Hi guys,
>>> >
>>> > I've read anything on this argument on Wiki but I can't find a
>>> > complete answer.
>>> >
>>> > I need to bridge two calls in Lua, waiting the first to be answered.
>>> > My basic script is:
>>> >
>>> > local GwParams =
>>> >
>>> "origination_caller_id_number="..gateway..",sip_auth_username="..gateway..",sip_auth_password="..sip_passwd
>>> > local slegA =
>>> >
>>> "[ignore_early_media=false,"..GwParams.."]sofia/external/"..callee.."@"..sip_domain
>>> > local slegB =
>>> >
>>> "[ignore_early_media=true,"..GwParams.."]sofia/external/"..called.."@"..sip_domain
>>> >
>>> > legA = freeswitch.Session(slegA);
>>> > legB = freeswitch.Session(slegB);
>>> > freeswitch.bridge(legA, legB);
>>> >
>>> >
>>> > It works but legA and legB are called at the same time. Using:
>>> >
>>> > while (legA:answered()== false) do end;
>>> >
>>> > between the two session do the job but called parties can't hear
>>> > audio.
>>> >
>>> > I promise to add a script on wiki if you help me ! :)
>>> >
>>> > Thanks in advance.
>>> >
>>> >
>>> >
>>> -------------------------------------------------------------------------------
>>> > Valore legale alle tue mail
>>> > InterfreePEC - la tua Posta Elettronica Certificata
>>> > http://pec.interfree.it
>>> >
>>> -------------------------------------------------------------------------------
>>> >
>>> >
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>> Try:
>>>
>>> legA = freeswitch.Session(slegA);
>>> if (legA:ready()) then
>>> legB = freeswitch.Session(slegB, legA);
>>> freeswitch.bridge(legA, legB);
>>> end
>>>
>>> ----
>>>
>>> Scheglov Sergey
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> -------------------------------------------------------------------------------
>>> Valore legale alle tue mail
>>> InterfreePEC - la tua Posta Elettronica Certificata
>>> http://pec.interfree.it
>>>
>>> -------------------------------------------------------------------------------
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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>
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From msc at freeswitch.org  Wed Sep 14 03:40:39 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Sep 2011 16:40:39 -0700
Subject: [Freeswitch-users] No Rule to make target 'tport/libtport.la'
In-Reply-To: <CA+PZoDRxh5Z=Dx_zMzRu6BGBUOYtGDjJn+cjGMes2OodGF13Cw@mail.gmail.com>
References: <CA+PZoDS4iJhrS=6Z91y5NCeeQ4a1q1+H+hQBNK-DC97X+BZCgQ@mail.gmail.com>
	<CA+PZoDRxh5Z=Dx_zMzRu6BGBUOYtGDjJn+cjGMes2OodGF13Cw@mail.gmail.com>
Message-ID: <CAKzWOxUVq4MNSFHUw0_CA-y0tBixEPqZC2PyyS=PoOb3Ye0M0g@mail.gmail.com>

Can you git pull again and try rebuilding? Just want to make sure there
wasn't an issue with the download.

-MC

On Tue, Sep 13, 2011 at 2:59 PM, Peter Schrock <peter.schrock at gmail.com>wrote:

> Okay, so I think I know the root of the problem but am not aware of how to
> solve the problem. This is the error message:
>
> tport_logging.c: In function 'tport_open_log':
> tport_logging.c:209: error: 'AI_NUMERICSERV' undeclared (first use in this
> function)
> tport_logging.c:209: error: (Each undeclared identifier is reported only
> once
> tport_logging.c:209: error: for each function it appears in.)
> make[9]: *** [tport_logging.lo] Error 1
> make[8]: *** [all] Error 2
>
> Anyone know anything that can help?
>
> Peter
>
> On Sun, Sep 11, 2011 at 11:49 PM, Peter Schrock <peter.schrock at gmail.com>wrote:
>
>> So, I figured out how to fix the 'swab' problem using a powerpc. Now, I am
>> getting this error message:
>>
>> LINK libnua.la
>> make[8]: *** No rule to make target 'tport/libtport.la', needed by '
>> libsofia-sip-us.la'. Stop.
>>
>> Does any one have any ideas?
>>
>> Peter
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Wed Sep 14 03:53:42 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 14 Sep 2011 02:53:42 +0300
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION - did FS not offer
	a codec?
In-Reply-To: <CAKzWOxUpFV=8UxNhKByiERD3mTO-uxfi0mwfqswfGTzkpFvQWg@mail.gmail.com>
References: <CALNZuwkvEtVcYXjie4RsBhaKcJiSJsrdBJ_otXijcDHm74EBsA@mail.gmail.com>
	<CAKzWOxUpFV=8UxNhKByiERD3mTO-uxfi0mwfqswfGTzkpFvQWg@mail.gmail.com>
Message-ID: <CALNZuwnfHn6ppS15R97R4kRc1Djw9_eYtMGM=vyk=E8kxTrmjw@mail.gmail.com>

OK then!
Looking at my call logs, it seems to mostly happen with one particular
carrier. Can I set verbose_sdp=true in the bridge string? e.g.
"[verbose_sdp=true]sofia/gateway/... "?

Thanks,
-Avi


On Tue, Sep 13, 2011 at 6:28 PM, Michael Collins <msc at freeswitch.org> wrote:

> For the record, line #107 contains the codecs (and other stuff) that FS
> offered:
>
> m=audio 16480 RTP/AVP 8 0 9 101 13
>
> The "8 0 9 101 13" represent the codecs (and other stuff). What do those
> numbers mean? This wiki page has the list:
>
> http://wiki.freeswitch.org/wiki/RTP_payload_list
>
> The SDP spec says that for non-dynamic payload numbers you are NOT required
> to have the "a=rtpmap" lines. This makes sense because those payload numbers
> don't change meaning until you get up to 96 and higher. The issue is that
> some devices are broken and require the "a=rtpmap" lines for the payload
> types below number 96. The verbose_sdp chan var mentioned by KK tells FS to
> add all those a=rtpmap lines for the non-dynamic payload types. You should
> only set this var when absolutely necessary because those verbose SDP's can
> cause the packets to be larger than the allowable MTU size.
>
> Dontcha just *love* SIP/SDP interop? :)
>
> -MC
>
> On Tue, Sep 13, 2011 at 8:01 AM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> I see a failed bridge to my second carrier. The  sip_local_sdp_str doesn't
>> show PCMU in it - did FS not offer any codecs?
>>  FS log: http://pastebin.freeswitch.org/17316
>> PCAP: http://ge.tt/95tFji7
>>
>> Can someone tell me what's going on here?
>>
>> -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
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Sep 14 04:15:00 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Sep 2011 17:15:00 -0700
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION - did FS not offer
	a codec?
In-Reply-To: <CALNZuwnfHn6ppS15R97R4kRc1Djw9_eYtMGM=vyk=E8kxTrmjw@mail.gmail.com>
References: <CALNZuwkvEtVcYXjie4RsBhaKcJiSJsrdBJ_otXijcDHm74EBsA@mail.gmail.com>
	<CAKzWOxUpFV=8UxNhKByiERD3mTO-uxfi0mwfqswfGTzkpFvQWg@mail.gmail.com>
	<CALNZuwnfHn6ppS15R97R4kRc1Djw9_eYtMGM=vyk=E8kxTrmjw@mail.gmail.com>
Message-ID: <CAKzWOxWwX+aGgk+rS5_7omrQJ3ipZge=TM89yKtsoQemahweXQ@mail.gmail.com>

I believe that is exactly the way to use it.
-MC

On Tue, Sep 13, 2011 at 4:53 PM, Avi Marcus <avi at avimarcus.net> wrote:

> OK then!
> Looking at my call logs, it seems to mostly happen with one particular
> carrier. Can I set verbose_sdp=true in the bridge string? e.g.
> "[verbose_sdp=true]sofia/gateway/... "?
>
> Thanks,
> -Avi
>
>
> On Tue, Sep 13, 2011 at 6:28 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> For the record, line #107 contains the codecs (and other stuff) that FS
>> offered:
>>
>> m=audio 16480 RTP/AVP 8 0 9 101 13
>>
>> The "8 0 9 101 13" represent the codecs (and other stuff). What do those
>> numbers mean? This wiki page has the list:
>>
>> http://wiki.freeswitch.org/wiki/RTP_payload_list
>>
>> The SDP spec says that for non-dynamic payload numbers you are NOT
>> required to have the "a=rtpmap" lines. This makes sense because those
>> payload numbers don't change meaning until you get up to 96 and higher. The
>> issue is that some devices are broken and require the "a=rtpmap" lines for
>> the payload types below number 96. The verbose_sdp chan var mentioned by KK
>> tells FS to add all those a=rtpmap lines for the non-dynamic payload types.
>> You should only set this var when absolutely necessary because those verbose
>> SDP's can cause the packets to be larger than the allowable MTU size.
>>
>> Dontcha just *love* SIP/SDP interop? :)
>>
>> -MC
>>
>> On Tue, Sep 13, 2011 at 8:01 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> I see a failed bridge to my second carrier. The  sip_local_sdp_str
>>> doesn't show PCMU in it - did FS not offer any codecs?
>>>  FS log: http://pastebin.freeswitch.org/17316
>>> PCAP: http://ge.tt/95tFji7
>>>
>>> Can someone tell me what's going on here?
>>>
>>> -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
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Glen.Ganderton at premier.com.au  Wed Sep 14 05:39:58 2011
From: Glen.Ganderton at premier.com.au (Glen Ganderton)
Date: Wed, 14 Sep 2011 11:39:58 +1000
Subject: [Freeswitch-users] Speech Input Timeout issue
In-Reply-To: <CD5A1499C8DEF34FAA3509722B00393E74029E309E@m-mail03.office.premiertech>
References: <CD5A1499C8DEF34FAA3509722B00393E74029E3090@m-mail03.office.premiertech>
	<CD5A1499C8DEF34FAA3509722B00393E74029E309E@m-mail03.office.premiertech>
Message-ID: <CD5A1499C8DEF34FAA3509722B00393E74029E3179@m-mail03.office.premiertech>

Also if there is any paid support I can get for this issue who would be best to see?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Glen Ganderton
Sent: Tuesday, 13 September 2011 3:49 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Speech Input Timeout issue

Just to add one more thing, when I was simply using the XML dialplan I didn't seem to have this issue. It may be an issue with my lua ivr script. Im just learning to program in lua and have hacked something together. So if you want to have a look maybe you will see an issue with it.

http://pastebin.com/Bagn8ESS



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Glen Ganderton
Sent: Tuesday, 13 September 2011 3:30 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Speech Input Timeout issue

Hi Guys,

I am in the process of configuring an speech recognition IVR using freeswitch and nuance, with the module unimrcp to link them.  This seems to be working fine, I have create a basic IVR script that works well however  around 50% - 60% of calls fail with some sort of input timeout issue.

Here is a few relevant lines of the log file for a successful call:

2011-09-11 09:07:18.017882 [DEBUG] mrcp_stream.c:382 Parsed MRCP Message [70]
2011-09-11 09:07:18.017882 [DEBUG] mrcp_client.c:1104 Signal Connection Task Message
2011-09-11 09:07:18.017882 [DEBUG] mrcp_client.c:974 Receive Connection Task Message [3]
2011-09-11 09:07:18.017882 [INFO] mrcp_client_session.c:504 Raise App MRCP Response 0xb7252778 <160>
2011-09-11 09:07:18.017882 [DEBUG] mod_unimrcp.c:3094 (ASR-16) RECOGNIZE IN PROGRESS
2011-09-11 09:07:18.017882 [DEBUG] mod_unimrcp.c:1488 (ASR-16) READY ==> PROCESSING
2011-09-11 09:07:18.017882 [DEBUG] apt_consumer_task.c:90 Wait for Task Messages [MRCP Client]
2011-09-11 09:07:18.017882 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/1000 at 10.3.5.80<mailto:sofia/internal/1000 at 10.3.5.80>
2011-09-11 09:07:18.690084 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed 0xb7252d18 [1000]
2011-09-11 09:07:18.690084 [DEBUG] mpf_timer_manager.c:111 Set Timer 0xb7252d18 [2000]
2011-09-11 09:07:19.446976 [INFO] mrcp_client_connection.c:525 Receive MRCPv2 Stream 10.3.5.80:49844 <-> 10.3.15.180:6075 [136 bytes]
MRCP/2.0 136 START-OF-INPUT 2 IN-PROGRESS
Channel-Identifier: 160 at speechrecog
Proxy-Sync-Id: 0-160 at speechrecog
Input-Type: speech


Here are a few lines for an unsuccessful call:
2011-09-11 09:07:44.605658 [DEBUG] mrcp_stream.c:382 Parsed MRCP Message [70]
2011-09-11 09:07:44.605658 [DEBUG] mrcp_client.c:1104 Signal Connection Task Message
2011-09-11 09:07:44.606901 [DEBUG] mrcp_client.c:974 Receive Connection Task Message [3]
2011-09-11 09:07:44.606901 [INFO] mrcp_client_session.c:504 Raise App MRCP Response 0xb72d8228 <161>
2011-09-11 09:07:44.606901 [DEBUG] mod_unimrcp.c:3094 (ASR-17) RECOGNIZE IN PROGRESS
2011-09-11 09:07:44.606901 [DEBUG] mod_unimrcp.c:1488 (ASR-17) READY ==> PROCESSING
2011-09-11 09:07:44.606901 [DEBUG] apt_consumer_task.c:90 Wait for Task Messages [MRCP Client]
2011-09-11 09:07:44.606901 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/1000 at 10.3.5.80<mailto:sofia/internal/1000 at 10.3.5.80>
2011-09-11 09:07:44.814845 [INFO] mrcp_client_connection.c:525 Receive MRCPv2 Stream 10.3.5.80:49845 <-> 10.3.15.180:6075 [141 bytes]
MRCP/2.0 141 RECOGNITION-COMPLETE 2 COMPLETE
Channel-Identifier: 161 at speechrecog
Waveform-URI:
Completion-Cause: 002 no-input-timeout

It seems to me the main difference is that in the successful call there is a timer that is set. Has anybody seen this issue or have any idea's on how to resolve it?


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From sid at eltc.ru  Wed Sep 14 06:12:09 2011
From: sid at eltc.ru (Sergey Scheglov)
Date: Wed, 14 Sep 2011 09:12:09 +0700
Subject: [Freeswitch-users] Re2:  Lua "two steps" bridging
In-Reply-To: <20110913122138.15531.qmail@community17.interfree.it>
References: <20110913122138.15531.qmail@community17.interfree.it>
Message-ID: <20110914091209.17f67fe1@shadow>

? 13 Sep 2011 12:21:38 -0000
chrisbware at interfree.it wrote:

> Hi,
>  
> thank you for your help, but doesn't work.
> legA:ready() is true when the first session start and not when the
> call is answered. So legB is called at the same time.
>  
> Any other suggestion?
>  
>  
>  
>  
> -----Messaggio originale-----
> Da: Sergey Scheglov <sid at eltc.ru>
> Inviato il: 13 Set 2011 - 17:15
> A: freeswitch-users at lists.freeswitch.org
> 
> 
> ? 13 Sep 2011 09:39:33 -0000
> chrisbware at interfree.it wrote:
> 
> >
> > Hi guys,
> >
> > I've read anything on this argument on Wiki but I can't find a
> > complete answer.
> >
> > I need to bridge two calls in Lua, waiting the first to be answered.
> > My basic script is:
> >
> > local GwParams =
> > "origination_caller_id_number="..gateway..",sip_auth_username="..gateway..",sip_auth_password="..sip_passwd
> > local slegA =
> > "[ignore_early_media=false,"..GwParams.."]sofia/external/"..callee.."@"..sip_domain
> > local slegB =
> > "[ignore_early_media=true,"..GwParams.."]sofia/external/"..called.."@"..sip_domain
> >
> > legA = freeswitch.Session(slegA);
> > legB = freeswitch.Session(slegB);
> > freeswitch.bridge(legA, legB);
> >
> >
> > It works but legA and legB are called at the same time. Using:
> >
> > while (legA:answered()== false) do end;
> >
> > between the two session do the job but called parties can't hear
> > audio.
> >
> > I promise to add a script on wiki if you help me ! :)
> >
> > Thanks in advance.
> >
> >
> > -------------------------------------------------------------------------------
> > Valore legale alle tue mail
> > InterfreePEC - la tua Posta Elettronica Certificata
> > http://pec.interfree.it
> > -------------------------------------------------------------------------------
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> Try:
> 
> legA = freeswitch.Session(slegA);
> if (legA:ready()) then
> legB = freeswitch.Session(slegB, legA);
> freeswitch.bridge(legA, legB);
> end
> 
> ----
> 
> Scheglov Sergey
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> -------------------------------------------------------------------------------
> Valore legale alle tue mail
> InterfreePEC - la tua Posta Elettronica Certificata
> http://pec.interfree.it
> -------------------------------------------------------------------------------

Just checked this script from cli:
luarun test.lua
--
local slegA="user/1000"
local slegB="user/1001"

legA = freeswitch.Session(slegA);
if (legA:ready()) then
legB = freeswitch.Session(slegB, legA);
freeswitch.bridge(legA, legB);
end

Leg B is created after the leg A answered.

Also, look this example
http://wiki.freeswitch.org/wiki/Bridging_two_calls_with_retry
-- 
Scheglov Sergey


From Glen.Ganderton at premier.com.au  Wed Sep 14 09:47:34 2011
From: Glen.Ganderton at premier.com.au (Glen Ganderton)
Date: Wed, 14 Sep 2011 15:47:34 +1000
Subject: [Freeswitch-users] Speech Input Timeout issue
In-Reply-To: <CD5A1499C8DEF34FAA3509722B00393E74029E3179@m-mail03.office.premiertech>
References: <CD5A1499C8DEF34FAA3509722B00393E74029E3090@m-mail03.office.premiertech>
	<CD5A1499C8DEF34FAA3509722B00393E74029E309E@m-mail03.office.premiertech>
	<CD5A1499C8DEF34FAA3509722B00393E74029E3179@m-mail03.office.premiertech>
Message-ID: <CD5A1499C8DEF34FAA3509722B00393E74029E3231@m-mail03.office.premiertech>

Looks like I have this one resolved. Was an issue with the Nuance MRCP server.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Glen Ganderton
Sent: Wednesday, 14 September 2011 11:40 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Speech Input Timeout issue

Also if there is any paid support I can get for this issue who would be best to see?

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Glen Ganderton
Sent: Tuesday, 13 September 2011 3:49 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Speech Input Timeout issue

Just to add one more thing, when I was simply using the XML dialplan I didn't seem to have this issue. It may be an issue with my lua ivr script. Im just learning to program in lua and have hacked something together. So if you want to have a look maybe you will see an issue with it.

http://pastebin.com/Bagn8ESS



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Glen Ganderton
Sent: Tuesday, 13 September 2011 3:30 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Speech Input Timeout issue

Hi Guys,

I am in the process of configuring an speech recognition IVR using freeswitch and nuance, with the module unimrcp to link them.  This seems to be working fine, I have create a basic IVR script that works well however  around 50% - 60% of calls fail with some sort of input timeout issue.

Here is a few relevant lines of the log file for a successful call:

2011-09-11 09:07:18.017882 [DEBUG] mrcp_stream.c:382 Parsed MRCP Message [70]
2011-09-11 09:07:18.017882 [DEBUG] mrcp_client.c:1104 Signal Connection Task Message
2011-09-11 09:07:18.017882 [DEBUG] mrcp_client.c:974 Receive Connection Task Message [3]
2011-09-11 09:07:18.017882 [INFO] mrcp_client_session.c:504 Raise App MRCP Response 0xb7252778 <160>
2011-09-11 09:07:18.017882 [DEBUG] mod_unimrcp.c:3094 (ASR-16) RECOGNIZE IN PROGRESS
2011-09-11 09:07:18.017882 [DEBUG] mod_unimrcp.c:1488 (ASR-16) READY ==> PROCESSING
2011-09-11 09:07:18.017882 [DEBUG] apt_consumer_task.c:90 Wait for Task Messages [MRCP Client]
2011-09-11 09:07:18.017882 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/1000 at 10.3.5.80<mailto:sofia/internal/1000 at 10.3.5.80>
2011-09-11 09:07:18.690084 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed 0xb7252d18 [1000]
2011-09-11 09:07:18.690084 [DEBUG] mpf_timer_manager.c:111 Set Timer 0xb7252d18 [2000]
2011-09-11 09:07:19.446976 [INFO] mrcp_client_connection.c:525 Receive MRCPv2 Stream 10.3.5.80:49844 <-> 10.3.15.180:6075 [136 bytes]
MRCP/2.0 136 START-OF-INPUT 2 IN-PROGRESS
Channel-Identifier: 160 at speechrecog
Proxy-Sync-Id: 0-160 at speechrecog
Input-Type: speech


Here are a few lines for an unsuccessful call:
2011-09-11 09:07:44.605658 [DEBUG] mrcp_stream.c:382 Parsed MRCP Message [70]
2011-09-11 09:07:44.605658 [DEBUG] mrcp_client.c:1104 Signal Connection Task Message
2011-09-11 09:07:44.606901 [DEBUG] mrcp_client.c:974 Receive Connection Task Message [3]
2011-09-11 09:07:44.606901 [INFO] mrcp_client_session.c:504 Raise App MRCP Response 0xb72d8228 <161>
2011-09-11 09:07:44.606901 [DEBUG] mod_unimrcp.c:3094 (ASR-17) RECOGNIZE IN PROGRESS
2011-09-11 09:07:44.606901 [DEBUG] mod_unimrcp.c:1488 (ASR-17) READY ==> PROCESSING
2011-09-11 09:07:44.606901 [DEBUG] apt_consumer_task.c:90 Wait for Task Messages [MRCP Client]
2011-09-11 09:07:44.606901 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/1000 at 10.3.5.80<mailto:sofia/internal/1000 at 10.3.5.80>
2011-09-11 09:07:44.814845 [INFO] mrcp_client_connection.c:525 Receive MRCPv2 Stream 10.3.5.80:49845 <-> 10.3.15.180:6075 [141 bytes]
MRCP/2.0 141 RECOGNITION-COMPLETE 2 COMPLETE
Channel-Identifier: 161 at speechrecog
Waveform-URI:
Completion-Cause: 002 no-input-timeout

It seems to me the main difference is that in the successful call there is a timer that is set. Has anybody seen this issue or have any idea's on how to resolve it?


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From Glen.Ganderton at premier.com.au  Wed Sep 14 10:04:10 2011
From: Glen.Ganderton at premier.com.au (Glen Ganderton)
Date: Wed, 14 Sep 2011 16:04:10 +1000
Subject: [Freeswitch-users] Where to set parameter for input timeout
Message-ID: <CD5A1499C8DEF34FAA3509722B00393E74029E3248@m-mail03.office.premiertech>

Hey Guys,

Im trying to find where I would set the parameter for input timeout. Im using FreeSWITCH as an MRCP client to my Nuance server and what is happening is if FreeSWITCH doesn't detect any speech in around 8-10 seconds I get an input-timeout, any idea's?

MRCP/2.0 158 RECOGNIZE 2
Channel-Identifier: 40 at speechrecog
Cancel-If-Queue: false
Content-Type: text/uri-list
Content-Length: 21

session:nuance5-mrcp2
2011-09-12 10:30:12.728662 [INFO] mpf_rtp_stream.c:1092 Generate RTCP SR [ssrc:2706393383 s:12 o:1920 ts:2160]
2011-09-12 10:30:12.728662 [INFO] mpf_rtp_stream.c:1279 Send Compound RTCP Packet [BYE] [72 bytes] 10.3.5.80:4017 -> 10.3.15.180:7971
2011-09-12 10:30:12.752156 [INFO] mrcp_client_connection.c:525 Receive MRCPv2 Stream 10.3.5.80:40061 <-> 10.3.15.180:6075 [69 bytes]
MRCP/2.0 69 2 200 IN-PROGRESS
Channel-Identifier: 40 at speechrecog


2011-09-12 10:30:12.752156 [DEBUG] mrcp_stream.c:382 Parsed MRCP Message [69]
2011-09-12 10:30:12.752156 [DEBUG] mrcp_client.c:1104 Signal Connection Task Message
2011-09-12 10:30:12.752156 [DEBUG] mrcp_client.c:974 Receive Connection Task Message [3]
2011-09-12 10:30:12.752156 [INFO] mrcp_client_session.c:504 Raise App MRCP Response 0xb7191d88 <40>
2011-09-12 10:30:12.752156 [DEBUG] mod_unimrcp.c:3094 (ASR-8) RECOGNIZE IN PROGRESS
2011-09-12 10:30:12.752156 [DEBUG] mod_unimrcp.c:1488 (ASR-8) READY ==> PROCESSING
2011-09-12 10:30:12.752156 [DEBUG] apt_consumer_task.c:90 Wait for Task Messages [MRCP Client]
2011-09-12 10:30:12.752156 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/1000 at 10.3.5.80
2011-09-12 10:30:12.756369 [DEBUG] switch_ivr_play_say.c:1236 Codec Activated L16 at 8000hz 1 channels 20ms
2011-09-12 10:30:13.388227 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed 0xb7192328 [1000]
2011-09-12 10:30:13.388227 [DEBUG] mpf_timer_manager.c:111 Set Timer 0xb7192328 [2000]
2011-09-12 10:30:14.388378 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed 0xb7192328 [2000]
2011-09-12 10:30:14.388378 [DEBUG] mpf_timer_manager.c:111 Set Timer 0xb7192328 [3000]
2011-09-12 10:30:14.976914 [DEBUG] switch_ivr_play_say.c:1573 done playing file
2011-09-12 10:30:15.387961 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed 0xb7192328 [3000]
2011-09-12 10:30:15.387961 [DEBUG] mpf_timer_manager.c:111 Set Timer 0xb7192328 [4000]
2011-09-12 10:30:16.388597 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed 0xb7192328 [4000]
2011-09-12 10:30:16.388597 [DEBUG] mpf_timer_manager.c:111 Set Timer 0xb7192328 [5000]
2011-09-12 10:30:17.387667 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed 0xb7192310 [5000]
2011-09-12 10:30:17.387667 [INFO] mpf_rtp_stream.c:1092 Generate RTCP SR [ssrc:2706393383 s:244 o:39040 ts:39440]
2011-09-12 10:30:17.387667 [INFO] mpf_rtp_stream.c:1230 Send Compound RTCP Packet [48 bytes] 10.3.5.80:4017 -> 10.3.15.180:7971
2011-09-12 10:30:17.387667 [DEBUG] mpf_timer_manager.c:111 Set Timer 0xb7192310 [10000]
2011-09-12 10:30:17.387667 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed 0xb7192328 [5000]
2011-09-12 10:30:17.387667 [DEBUG] mpf_timer_manager.c:111 Set Timer 0xb7192328 [6000]
2011-09-12 10:30:18.388818 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed 0xb7192328 [6000]
2011-09-12 10:30:18.388818 [DEBUG] mpf_timer_manager.c:111 Set Timer 0xb7192328 [7000]
2011-09-12 10:30:19.387895 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed 0xb7192328 [7000]
2011-09-12 10:30:19.387895 [DEBUG] mpf_timer_manager.c:111 Set Timer 0xb7192328 [8000]
2011-09-12 10:30:19.906918 [INFO] mrcp_client_connection.c:525 Receive MRCPv2 Stream 10.3.5.80:40061 <-> 10.3.15.180:6075 [140 bytes]
MRCP/2.0 140 RECOGNITION-COMPLETE 2 COMPLETE
Channel-Identifier: 40 at speechrecog
Waveform-URI:
Completion-Cause: 002 no-input-timeout
--------------------------------------------------------------------------------------------------------------------------------
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From steveayre at gmail.com  Wed Sep 14 11:47:02 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 14 Sep 2011 08:47:02 +0100
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION - did FS not offer
	a codec?
In-Reply-To: <CALNZuwnfHn6ppS15R97R4kRc1Djw9_eYtMGM=vyk=E8kxTrmjw@mail.gmail.com>
References: <CALNZuwkvEtVcYXjie4RsBhaKcJiSJsrdBJ_otXijcDHm74EBsA@mail.gmail.com>
	<CAKzWOxUpFV=8UxNhKByiERD3mTO-uxfi0mwfqswfGTzkpFvQWg@mail.gmail.com>
	<CALNZuwnfHn6ppS15R97R4kRc1Djw9_eYtMGM=vyk=E8kxTrmjw@mail.gmail.com>
Message-ID: <CAFiqYum740N2tncbpifbbH6a9Mw9JLp84zCkhZaZdJbumVY2xQ@mail.gmail.com>

You should also be able to set it in the gateway configuration, I believe:
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Variables

If that works then that'll probably make it far easier to manage as you
won't need to put in any logic to alter the dialstring.

-Steve



On 14 September 2011 00:53, Avi Marcus <avi at avimarcus.net> wrote:

> OK then!
> Looking at my call logs, it seems to mostly happen with one particular
> carrier. Can I set verbose_sdp=true in the bridge string? e.g.
> "[verbose_sdp=true]sofia/gateway/... "?
>
> Thanks,
> -Avi
>
>
> On Tue, Sep 13, 2011 at 6:28 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> For the record, line #107 contains the codecs (and other stuff) that FS
>> offered:
>>
>> m=audio 16480 RTP/AVP 8 0 9 101 13
>>
>> The "8 0 9 101 13" represent the codecs (and other stuff). What do those
>> numbers mean? This wiki page has the list:
>>
>> http://wiki.freeswitch.org/wiki/RTP_payload_list
>>
>> The SDP spec says that for non-dynamic payload numbers you are NOT
>> required to have the "a=rtpmap" lines. This makes sense because those
>> payload numbers don't change meaning until you get up to 96 and higher. The
>> issue is that some devices are broken and require the "a=rtpmap" lines for
>> the payload types below number 96. The verbose_sdp chan var mentioned by KK
>> tells FS to add all those a=rtpmap lines for the non-dynamic payload types.
>> You should only set this var when absolutely necessary because those verbose
>> SDP's can cause the packets to be larger than the allowable MTU size.
>>
>> Dontcha just *love* SIP/SDP interop? :)
>>
>> -MC
>>
>> On Tue, Sep 13, 2011 at 8:01 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> I see a failed bridge to my second carrier. The  sip_local_sdp_str
>>> doesn't show PCMU in it - did FS not offer any codecs?
>>>  FS log: http://pastebin.freeswitch.org/17316
>>> PCAP: http://ge.tt/95tFji7
>>>
>>> Can someone tell me what's going on here?
>>>
>>> -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
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chrisbware at interfree.it  Wed Sep 14 11:48:25 2011
From: chrisbware at interfree.it (chrisbware at interfree.it)
Date: 14 Sep 2011 07:48:25 -0000
Subject: [Freeswitch-users] Re5:  Lua "two steps" bridging
Message-ID: <20110914074825.30898.qmail@community24.interfree.it>

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From brtantunes at gmail.com  Wed Sep 14 13:13:10 2011
From: brtantunes at gmail.com (Brtantunes)
Date: Wed, 14 Sep 2011 02:13:10 -0700 (PDT)
Subject: [Freeswitch-users] Timeout Message or something similar
In-Reply-To: <CAKzWOxUszOu8xGn70+UtaCOtqFC=WvEObORE+vWpdFmXKXSLgA@mail.gmail.com>
References: <1315837848697-6783735.post@n2.nabble.com>
	<CAMwZ-tK6GT7EdtCFdZqNTahfGp1BHt45jBUXt68eGfbKv7TBig@mail.gmail.com>
	<1315907331256-6786830.post@n2.nabble.com>
	<CAKzWOxUszOu8xGn70+UtaCOtqFC=WvEObORE+vWpdFmXKXSLgA@mail.gmail.com>
Message-ID: <1315991590857-6791788.post@n2.nabble.com>

Hello,

Thank you for your reply,

I tried the first command:
/originate sofia/DOMAIN/1001
bridge:{fail_on_single_reject=true,ignore_early_media=true,originate_continue_on_timeout=true}sofia/DOMAIN/1002,playback:C:/teste2.wav
inline/
but it dosen't work :(.

Then i tried the second approach and it works. The problem is i want to
configure all parameters in runtime, so i need the first approah. Is there a
way to see the command generated using XML default approach? Or can you guys
tell me which command to use?

Tks for your help,
Greetings

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Timeout-Message-or-something-similar-tp6783735p6791788.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From garmt.noname at gmail.com  Wed Sep 14 14:04:43 2011
From: garmt.noname at gmail.com (grmt)
Date: Wed, 14 Sep 2011 12:04:43 +0200
Subject: [Freeswitch-users] Where to set parameter for input timeout
In-Reply-To: <CD5A1499C8DEF34FAA3509722B00393E74029E3248@m-mail03.office.premiertech>
References: <CD5A1499C8DEF34FAA3509722B00393E74029E3248@m-mail03.office.premiertech>
Message-ID: <02b201cc72c5$ba90fe50$2fb2faf0$@gmail.com>

Hi, 

 

Not that I have any experience with UniMRCP in combination with Nuance ASR,
however it seems to me that the MRCP server (i.e. Nuance) tells you that it
did not receive any input. I believe having read somewhere that the default
no-input-timeout on Nuance is 8 (s). 

If you want to manipulate the no-input-timeout on a MRCP request basis, I
think you will have to change mod_unimrcp.

You may also manipulate the global request-timout which by default is 10
(s).

I bet that if you set this lower than 8(s), it is FS that will timeout .

 

Now why Nuance is not receiving any input (rtp) I don't know.

 

If I understand correctly from the log below you play a wav file and you
want that to be recognized?

Maybe you need a little bit more time between triggering the recognizer and
playing the wav file?

You may want to use wireshark to capture the RTP stream between FS and
NUANCE.

 

Garmt

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Glen
Ganderton
Sent: Wednesday, September 14, 2011 08:04
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Where to set parameter for input timeout

 

Hey Guys,

 

Im trying to find where I would set the parameter for input timeout. Im
using FreeSWITCH as an MRCP client to my Nuance server and what is happening
is if FreeSWITCH doesn't detect any speech in around 8-10 seconds I get an
input-timeout, any idea's?

 

MRCP/2.0 158 RECOGNIZE 2

Channel-Identifier: 40 at speechrecog

Cancel-If-Queue: false

Content-Type: text/uri-list

Content-Length: 21

 

session:nuance5-mrcp2

2011-09-12 10:30:12.728662 [INFO] mpf_rtp_stream.c:1092 Generate RTCP SR
[ssrc:2706393383 s:12 o:1920 ts:2160]

2011-09-12 10:30:12.728662 [INFO] mpf_rtp_stream.c:1279 Send Compound RTCP
Packet [BYE] [72 bytes] 10.3.5.80:4017 -> 10.3.15.180:7971

2011-09-12 10:30:12.752156 [INFO] mrcp_client_connection.c:525 Receive
MRCPv2 Stream 10.3.5.80:40061 <-> 10.3.15.180:6075 [69 bytes]

MRCP/2.0 69 2 200 IN-PROGRESS

Channel-Identifier: 40 at speechrecog

 

 

2011-09-12 10:30:12.752156 [DEBUG] mrcp_stream.c:382 Parsed MRCP Message
[69]

2011-09-12 10:30:12.752156 [DEBUG] mrcp_client.c:1104 Signal Connection Task
Message

2011-09-12 10:30:12.752156 [DEBUG] mrcp_client.c:974 Receive Connection Task
Message [3]

2011-09-12 10:30:12.752156 [INFO] mrcp_client_session.c:504 Raise App MRCP
Response 0xb7191d88 <40>

2011-09-12 10:30:12.752156 [DEBUG] mod_unimrcp.c:3094 (ASR-8) RECOGNIZE IN
PROGRESS

2011-09-12 10:30:12.752156 [DEBUG] mod_unimrcp.c:1488 (ASR-8) READY ==>
PROCESSING

2011-09-12 10:30:12.752156 [DEBUG] apt_consumer_task.c:90 Wait for Task
Messages [MRCP Client]

2011-09-12 10:30:12.752156 [DEBUG] switch_core_media_bug.c:360 Attaching BUG
to sofia/internal/1000 at 10.3.5.80

2011-09-12 10:30:12.756369 [DEBUG] switch_ivr_play_say.c:1236 Codec
Activated L16 at 8000hz 1 channels 20ms

2011-09-12 10:30:13.388227 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed
0xb7192328 [1000]

2011-09-12 10:30:13.388227 [DEBUG] mpf_timer_manager.c:111 Set Timer
0xb7192328 [2000]

2011-09-12 10:30:14.388378 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed
0xb7192328 [2000]

2011-09-12 10:30:14.388378 [DEBUG] mpf_timer_manager.c:111 Set Timer
0xb7192328 [3000]

2011-09-12 10:30:14.976914 [DEBUG] switch_ivr_play_say.c:1573 done playing
file

2011-09-12 10:30:15.387961 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed
0xb7192328 [3000]

2011-09-12 10:30:15.387961 [DEBUG] mpf_timer_manager.c:111 Set Timer
0xb7192328 [4000]

2011-09-12 10:30:16.388597 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed
0xb7192328 [4000]

2011-09-12 10:30:16.388597 [DEBUG] mpf_timer_manager.c:111 Set Timer
0xb7192328 [5000]

2011-09-12 10:30:17.387667 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed
0xb7192310 [5000]

2011-09-12 10:30:17.387667 [INFO] mpf_rtp_stream.c:1092 Generate RTCP SR
[ssrc:2706393383 s:244 o:39040 ts:39440]

2011-09-12 10:30:17.387667 [INFO] mpf_rtp_stream.c:1230 Send Compound RTCP
Packet [48 bytes] 10.3.5.80:4017 -> 10.3.15.180:7971

2011-09-12 10:30:17.387667 [DEBUG] mpf_timer_manager.c:111 Set Timer
0xb7192310 [10000]

2011-09-12 10:30:17.387667 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed
0xb7192328 [5000]

2011-09-12 10:30:17.387667 [DEBUG] mpf_timer_manager.c:111 Set Timer
0xb7192328 [6000]

2011-09-12 10:30:18.388818 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed
0xb7192328 [6000]

2011-09-12 10:30:18.388818 [DEBUG] mpf_timer_manager.c:111 Set Timer
0xb7192328 [7000]

2011-09-12 10:30:19.387895 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed
0xb7192328 [7000]

2011-09-12 10:30:19.387895 [DEBUG] mpf_timer_manager.c:111 Set Timer
0xb7192328 [8000]

2011-09-12 10:30:19.906918 [INFO] mrcp_client_connection.c:525 Receive
MRCPv2 Stream 10.3.5.80:40061 <-> 10.3.15.180:6075 [140 bytes]

MRCP/2.0 140 RECOGNITION-COMPLETE 2 COMPLETE

Channel-Identifier: 40 at speechrecog

Waveform-URI:

Completion-Cause: 002 no-input-timeout

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

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From eagle.antonio at gmail.com  Wed Sep 14 14:18:17 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Wed, 14 Sep 2011 10:18:17 +0000
Subject: [Freeswitch-users] DTMF Info
Message-ID: <CAF+3j-syTYQi2uCz4exSXKmqs++HhoHkv0UA=V-Yu_RPC4On2Q@mail.gmail.com>

Good Morning.

I have a soft phone that uses SIP INFO to deliver the DTMF keys to FS.
Now the problem is that FS always reported INFO is not expected on this
channels so off to the wiki i go and i set in the profile dtmf-type = info.
sofia.c:6439 IGNORE INFO DTMF(2) (This channel was not configured to use
INFO DTMF!)

So anyone can help me out with this ?

Best Regards
Antonio Teixeira
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From avi at avimarcus.net  Wed Sep 14 14:25:31 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 14 Sep 2011 13:25:31 +0300
Subject: [Freeswitch-users] DTMF Info
In-Reply-To: <CAF+3j-syTYQi2uCz4exSXKmqs++HhoHkv0UA=V-Yu_RPC4On2Q@mail.gmail.com>
References: <CAF+3j-syTYQi2uCz4exSXKmqs++HhoHkv0UA=V-Yu_RPC4On2Q@mail.gmail.com>
Message-ID: <CALNZuwnzTW0itYkOSPjuN7QHb1tRJi6o2sqqaXBvqYdzg5Ow1Q@mail.gmail.com>

If you set it in the profile, once you reload the profile, it should work.
Or, in the sip profile you can set
liberal-dtmf<http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#liberal-dtmf>
and
then it will accept rfc2833 or INFO.

-Avi


On Wed, Sep 14, 2011 at 1:18 PM, Antonio Teixeira
<eagle.antonio at gmail.com>wrote:

> Good Morning.
>
> I have a soft phone that uses SIP INFO to deliver the DTMF keys to FS.
> Now the problem is that FS always reported INFO is not expected on this
> channels so off to the wiki i go and i set in the profile dtmf-type = info.
> sofia.c:6439 IGNORE INFO DTMF(2) (This channel was not configured to use
> INFO DTMF!)
>
> So anyone can help me out with this ?
>
> Best Regards
> Antonio Teixeira
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.schrock at gmail.com  Wed Sep 14 14:38:15 2011
From: peter.schrock at gmail.com (Peter Schrock)
Date: Wed, 14 Sep 2011 03:38:15 -0700
Subject: [Freeswitch-users] No Rule to make target 'tport/libtport.la'
In-Reply-To: <CAKzWOxUVq4MNSFHUw0_CA-y0tBixEPqZC2PyyS=PoOb3Ye0M0g@mail.gmail.com>
References: <CA+PZoDS4iJhrS=6Z91y5NCeeQ4a1q1+H+hQBNK-DC97X+BZCgQ@mail.gmail.com>
	<CA+PZoDRxh5Z=Dx_zMzRu6BGBUOYtGDjJn+cjGMes2OodGF13Cw@mail.gmail.com>
	<CAKzWOxUVq4MNSFHUw0_CA-y0tBixEPqZC2PyyS=PoOb3Ye0M0g@mail.gmail.com>
Message-ID: <-3231902503382953710@unknownmsgid>

So, I tried the fit pull and to no avail, I got the same tport error
message. Any suggestions?

Peter

On Sep 13, 2011, at 4:42 PM, Michael Collins <msc at freeswitch.org> wrote:

Can you git pull again and try rebuilding? Just want to make sure there
wasn't an issue with the download.

-MC

On Tue, Sep 13, 2011 at 2:59 PM, Peter Schrock <peter.schrock at gmail.com>wrote:

> Okay, so I think I know the root of the problem but am not aware of how to
> solve the problem. This is the error message:
>
> tport_logging.c: In function 'tport_open_log':
> tport_logging.c:209: error: 'AI_NUMERICSERV' undeclared (first use in this
> function)
> tport_logging.c:209: error: (Each undeclared identifier is reported only
> once
> tport_logging.c:209: error: for each function it appears in.)
> make[9]: *** [tport_logging.lo] Error 1
> make[8]: *** [all] Error 2
>
> Anyone know anything that can help?
>
> Peter
>
> On Sun, Sep 11, 2011 at 11:49 PM, Peter Schrock <peter.schrock at gmail.com>wrote:
>
>> So, I figured out how to fix the 'swab' problem using a powerpc. Now, I am
>> getting this error message:
>>
>> LINK libnua.la
>> make[8]: *** No rule to make target 'tport/libtport.la', needed by '
>> libsofia-sip-us.la'. Stop.
>>
>> Does any one have any ideas?
>>
>> Peter
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From eagle.antonio at gmail.com  Wed Sep 14 15:14:43 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Wed, 14 Sep 2011 11:14:43 +0000
Subject: [Freeswitch-users] DTMF Info
In-Reply-To: <CALNZuwnzTW0itYkOSPjuN7QHb1tRJi6o2sqqaXBvqYdzg5Ow1Q@mail.gmail.com>
References: <CAF+3j-syTYQi2uCz4exSXKmqs++HhoHkv0UA=V-Yu_RPC4On2Q@mail.gmail.com>
	<CALNZuwnzTW0itYkOSPjuN7QHb1tRJi6o2sqqaXBvqYdzg5Ow1Q@mail.gmail.com>
Message-ID: <CAF+3j-tA-8=YAMtKAW8abNYWXhUPU2tx7ZWGvFHG2CoipL-E_w@mail.gmail.com>

Hello Avi , thanks for your answer.
I tried setting dtmf_type at the sip profile and user profile still not luck
, liberal-dtmf is not available on my version.

Hope someone can help me , since upgrading will be more complicated
Regards
A/T

2011/9/14 Avi Marcus <avi at avimarcus.net>

> If you set it in the profile, once you reload the profile, it should work.
> Or, in the sip profile you can set liberal-dtmf<http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#liberal-dtmf> and
> then it will accept rfc2833 or INFO.
>
> -Avi
>
>
> On Wed, Sep 14, 2011 at 1:18 PM, Antonio Teixeira <eagle.antonio at gmail.com
> > wrote:
>
>> Good Morning.
>>
>> I have a soft phone that uses SIP INFO to deliver the DTMF keys to FS.
>> Now the problem is that FS always reported INFO is not expected on this
>> channels so off to the wiki i go and i set in the profile dtmf-type = info.
>> sofia.c:6439 IGNORE INFO DTMF(2) (This channel was not configured to use
>> INFO DTMF!)
>>
>> So anyone can help me out with this ?
>>
>> Best Regards
>> Antonio Teixeira
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From cmrienzo at gmail.com  Wed Sep 14 15:38:56 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 14 Sep 2011 07:38:56 -0400
Subject: [Freeswitch-users] Where to set parameter for input timeout
In-Reply-To: <02b201cc72c5$ba90fe50$2fb2faf0$@gmail.com>
References: <CD5A1499C8DEF34FAA3509722B00393E74029E3248@m-mail03.office.premiertech>
	<02b201cc72c5$ba90fe50$2fb2faf0$@gmail.com>
Message-ID: <CAD-qRkjohBZGU7ZQQCUxbRLPXt_UxjtRR3YrY+aBztn3UC=qbA@mail.gmail.com>

You can set MRCP parameters either globally in the MRCP profile with the
<recogparams> tag or on a specific request by prefixing {param="val"} to the
grammar.

Read the MRCP RFCs and the Nuance documentation for all available parameters
and their default values.

For MRCPv1 (RFC 4463):

8.4.5.  No Input Timeout

   When recognition is started and there is no speech detected for a
   certain period of time, the recognizer can send a RECOGNITION-
   COMPLETE event to the client and terminate the recognition operation.
   The no-input-timeout header field can set this timeout value.  The
   value is in milliseconds.  This header field MAY occur in RECOGNIZE,


Shanmugham, et al.           Informational                     [Page 45]

  <http://tools.ietf.org/html/rfc4463#page-46>RFC 4463
<http://tools.ietf.org/html/rfc4463>         MRCP by Cisco, Nuance,
and Speechworks       April 2006


   SET-PARAMS, or GET-PARAMS.  The value for this field ranges from 0 to
   MAXTIMEOUT, where MAXTIMEOUT is platform specific.  The default value
   for this field is platform specific.

     no-input-timeout    =    "No-Input-Timeout" ":" 1*DIGIT CRLF
8.4.6.  Recognition Timeout

   When recognition is started and there is no match for a certain
   period of time, the recognizer can send a RECOGNITION-COMPLETE event
   to the client and terminate the recognition operation.  The
   recognition-timeout parameter field sets this timeout value.  The
   value is in milliseconds.  The value for this field ranges from 0 to
   MAXTIMEOUT, where MAXTIMEOUT is platform specific.  The default value
   is 10 seconds.  This header field MAY occur in RECOGNIZE, SET-PARAMS
   or GET-PARAMS.

     recognition-timeout =    "Recognition-Timeout" ":" 1*DIGIT CRLF

8.4.10.  Recognition Start Timers

   This parameter MAY BE sent as part of the RECOGNIZE request.  A value
   of false tells the recognizer to start recognition, but not to start
   the no-input timer yet.  The recognizer should not start the timers
   until the client sends a RECOGNITION-START-TIMERS request to the
   recognizer.  This is useful in the scenario when the recognizer and
   synthesizer engines are not part of the same session.  Here, when a
   kill-on-barge-in prompt is being played, you want the RECOGNIZE
   request to be simultaneously active so that it can detect and
   implement kill-on-barge-in.  But at the same time, you don't want the
   recognizer to start the no-input timers until the prompt is finished.
   The default value is "true".

     recognizer-start-timers  =    "Recognizer-Start-Timers" ":"
                                   boolean-value CRLF


8.4.12.  Speech Complete Timeout

   This header field specifies the length of silence required following
   user speech before the speech recognizer finalizes a result (either
   accepting it or throwing a nomatch event).  The speech-complete-
   timeout value is used when the recognizer currently has a complete
   match of an active grammar, and specifies how long it should wait for
   more input before declaring a match.  By contrast, the incomplete
   timeout is used when the speech is an incomplete match to an active
   grammar.  The value is in milliseconds.

     speech-complete-timeout = "Speech-Complete-Timeout" ":"
                               1*DIGIT CRLF




Shanmugham, et al.           Informational                     [Page 48]

  <http://tools.ietf.org/html/rfc4463#page-49>RFC 4463
<http://tools.ietf.org/html/rfc4463>         MRCP by Cisco, Nuance,
and Speechworks       April 2006


   A long speech-complete-timeout value delays the result completion
   and, therefore, makes the computer's response slow.  A short speech-
   complete-timeout may lead to an utterance being broken up
   inappropriately.  Reasonable complete timeout values are typically in
   the range of 0.3 seconds to 1.0 seconds.  The value for this field
   ranges from 0 to MAXTIMEOUT, where MAXTIMEOUT is platform specific.
   The default value for this field is platform specific.  This header
   field MAY occur in RECOGNIZE, SET-PARAMS, or GET-PARAMS.
8.4.13.  Speech Incomplete Timeout

   This header field specifies the required length of silence following
   user speech, after which a recognizer finalizes a result.  The
   incomplete timeout applies when the speech prior to the silence is an
   incomplete match of all active grammars.  In this case, once the
   timeout is triggered, the partial result is rejected (with a nomatch
   event).  The value is in milliseconds.  The value for this field
   ranges from 0 to MAXTIMEOUT, where MAXTIMEOUT is platform specific.
   The default value for this field is platform specific.

     speech-incomplete-timeout = "Speech-Incomplete-Timeout" ":"
                                 1*DIGIT CRLF

   The speech-incomplete-timeout also applies when the speech prior to
   the silence is a complete match of an active grammar, but where it is
   possible to speak further and still match the grammar.  By contrast,
   the complete timeout is used when the speech is a complete match to
   an active grammar and no further words can be spoken.

   A long speech-incomplete-timeout value delays the result completion
   and, therefore, makes the computer's response slow.  A short speech-
   incomplete-timeout may lead to an utterance being broken up
   inappropriately.

   The speech-incomplete-timeout is usually longer than the speech-
   complete-timeout to allow users to pause mid-utterance (for example,
   to breathe).  This header field MAY occur in RECOGNIZE, SET-PARAMS,
   or GET-PARAMS.





For MRCPv2:

9.4.6.  No Input Timeout

   When recognition is started and there is no speech detected for a
   certain period of time, the recognizer can send a RECOGNITION-
   COMPLETE event to the client with a Completion-Cause of "no-input-
   timeout" and terminate the recognition operation.  The client can use
   the no-input-timeout header field to set this timeout.  The value is
   in milliseconds and may range from 0 to an implementation specific
   maximum value.  This header field MAY occur in RECOGNIZE,
   "SET-PARAMS" or "GET-PARAMS".  The default value is implementation
   specific.

   no-input-timeout         =  "No-Input-Timeout" ":" 1*19DIGIT CRLF
9.4.7.  Recognition Timeout

   When recognition is started and there is no match for a certain
   period of time, the recognizer can send a RECOGNITION-COMPLETE event
   to the client and terminate the recognition operation.  The
   Recognition-Timeout header field allows the client to set this
   timeout value.  The value is in milliseconds.  The value for this
   header field ranges from 0 to an implementation specific maximum
   value.  The default value is 10 seconds.  This header field MAY occur
   in RECOGNIZE, SET-PARAMS or GET-PARAMS.

   recognition-timeout      =  "Recognition-Timeout" ":" 1*19DIGIT CRLF


9.4.14.  Start Input Timers

   This header field MAY be sent as part of the RECOGNIZE request.  A
   value of false tells the recognizer to start recognition, but not to
   start the no-input timer yet.  The recognizer MUST NOT start the
   timers until the client sends a START-INPUT-TIMERS request to the
   recognizer.  This is useful in the scenario when the recognizer and
   synthesizer engines are not part of the same session.  In such
   configurations, when a kill-on-barge-in prompt is being played (see
   Section 8.4.2
<http://tools.ietf.org/html/draft-ietf-speechsc-mrcpv2-25#section-8.4.2>),
the client wants the RECOGNIZE request to be
   simultaneously active so that it can detect and implement kill-on-
   barge-in.  However, the recognizer SHOULD NOT start the no-input
   timers until the prompt is finished.  The default value is "true".

   start-input-timers  =  "Start-Input-Timers" ":" BOOLEAN CRLF


Burnett & Shanmugham    Expires January 12, 2012               [Page 86]

  <http://tools.ietf.org/html/draft-ietf-speechsc-mrcpv2-25#page-87>Internet-Draft
                  MRCPv2                        July 2011

9.4.15.  Speech Complete Timeout

   This header field specifies the length of silence required following
   user speech before the speech recognizer finalizes a result (either
   accepting it or generating a nomatch event).  The speech-complete-
   timeout value applies when the recognizer currently has a complete
   match against an active grammar, and specifies how long the
   recognizer MUST wait for more input before declaring a match.  By
   contrast, the incomplete timeout is used when the speech is an
   incomplete match to an active grammar.  The value is in milliseconds.

  speech-complete-timeout = "Speech-Complete-Timeout" ":" 1*19DIGIT CRLF

   A long speech-complete-timeout value delays the result to the client
   and therefore makes the application's response to a user slow.  A
   short speech-complete-timeout may lead to an utterance being broken
   up inappropriately.  Reasonable speech complete timeout values are
   typically in the range of 0.3 seconds to 1.0 seconds.  The value for
   this header field ranges from 0 to an implementation specific maximum
   value.  The default value for this header field is implementation
   specific.  This header field MAY occur in RECOGNIZE, "SET-PARAMS" or
   "GET-PARAMS".
9.4.16.  Speech Incomplete Timeout

   This header field specifies the required length of silence following
   user speech after which a recognizer finalizes a result.  The
   incomplete timeout applies when the speech prior to the silence is an
   incomplete match of all active grammars.  In this case, once the
   timeout is triggered, the partial result is rejected (with a
   Completion-Cause of "partial-match").  The value is in milliseconds.
   The value for this header field ranges from 0 to an implementation
   specific maximum value.  The default value for this header field is
   implementation specific.

   speech-incomplete-timeout = "Speech-Incomplete-Timeout" ":" 1*19DIGIT
                                CRLF

   The speech-incomplete-timeout also applies when the speech prior to
   the silence is a complete match of an active grammar, but where it is
   possible to speak further and still match the grammar.  By contrast,
   the complete timeout is used when the speech is a complete match to
   an active grammar and no further spoken words can continue to
   represent a match.

   A long speech-incomplete-timeout value delays the result to the
   client and therefore makes the application's response to a user slow.
   A short speech-incomplete-timeout may lead to an utterance being


Burnett & Shanmugham    Expires January 12, 2012               [Page 87]

  <http://tools.ietf.org/html/draft-ietf-speechsc-mrcpv2-25#page-88>Internet-Draft
                  MRCPv2                        July 2011


   broken up inappropriately.

   The speech-incomplete-timeout is usually longer than the speech-
   complete-timeout to allow users to pause mid-utterance (for example,
   to breathe).  This header field MAY occur in RECOGNIZE, "SET-PARAMS"
   or "GET-PARAMS".




On Wed, Sep 14, 2011 at 6:04 AM, grmt <garmt.noname at gmail.com> wrote:

> Hi, ****
>
> ** **
>
> Not that I have any experience with UniMRCP in combination with Nuance ASR,
> however it seems to me that the MRCP server (i.e. Nuance) tells you that it
> did not receive any input. I believe having read somewhere that the default
> no-input-timeout on Nuance is 8 (s). ****
>
> If you want to manipulate the no-input-timeout on a MRCP request basis, I
> think you will have to change mod_unimrcp.****
>
> You may also manipulate the global request-timout which by default is 10
> (s).****
>
> I bet that if you set this lower than 8(s), it is FS that will timeout ?**
> **
>
> ** **
>
> Now why Nuance is not receiving any input (rtp) I don?t know.****
>
> ** **
>
> If I understand correctly from the log below you play a wav file and you
> want that to be recognized?****
>
> Maybe you need a little bit more time between triggering the recognizer and
> playing the wav file?****
>
> You may want to use wireshark to capture the RTP stream between FS and
> NUANCE.****
>
> ** **
>
> Garmt****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Glen
> Ganderton
> *Sent:* Wednesday, September 14, 2011 08:04
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Where to set parameter for input timeout****
>
> ** **
>
> Hey Guys,****
>
> ** **
>
> Im trying to find where I would set the parameter for input timeout. Im
> using FreeSWITCH as an MRCP client to my Nuance server and what is happening
> is if FreeSWITCH doesn?t detect any speech in around 8-10 seconds I get an
> input-timeout, any idea?s?****
>
> ** **
>
> MRCP/2.0 158 RECOGNIZE 2****
>
> Channel-Identifier: 40 at speechrecog****
>
> Cancel-If-Queue: false****
>
> Content-Type: text/uri-list****
>
> Content-Length: 21****
>
> ** **
>
> session:nuance5-mrcp2****
>
> 2011-09-12 10:30:12.728662 [INFO] mpf_rtp_stream.c:1092 Generate RTCP SR
> [ssrc:2706393383 s:12 o:1920 ts:2160]****
>
> 2011-09-12 10:30:12.728662 [INFO] mpf_rtp_stream.c:1279 Send Compound RTCP
> Packet [BYE] [72 bytes] 10.3.5.80:4017 -> 10.3.15.180:7971****
>
> 2011-09-12 10:30:12.752156 [INFO] mrcp_client_connection.c:525 Receive
> MRCPv2 Stream 10.3.5.80:40061 <-> 10.3.15.180:6075 [69 bytes]****
>
> MRCP/2.0 69 2 200 IN-PROGRESS****
>
> Channel-Identifier: 40 at speechrecog****
>
> ** **
>
> ** **
>
> 2011-09-12 10:30:12.752156 [DEBUG] mrcp_stream.c:382 Parsed MRCP Message
> [69]****
>
> 2011-09-12 10:30:12.752156 [DEBUG] mrcp_client.c:1104 Signal Connection
> Task Message****
>
> 2011-09-12 10:30:12.752156 [DEBUG] mrcp_client.c:974 Receive Connection
> Task Message [3]****
>
> 2011-09-12 10:30:12.752156 [INFO] mrcp_client_session.c:504 Raise App MRCP
> Response 0xb7191d88 <40>****
>
> 2011-09-12 10:30:12.752156 [DEBUG] mod_unimrcp.c:3094 (ASR-8) RECOGNIZE IN
> PROGRESS****
>
> 2011-09-12 10:30:12.752156 [DEBUG] mod_unimrcp.c:1488 (ASR-8) READY ==>
> PROCESSING****
>
> 2011-09-12 10:30:12.752156 [DEBUG] apt_consumer_task.c:90 Wait for Task
> Messages [MRCP Client]****
>
> 2011-09-12 10:30:12.752156 [DEBUG] switch_core_media_bug.c:360 Attaching
> BUG to sofia/internal/1000 at 10.3.5.80****
>
> 2011-09-12 10:30:12.756369 [DEBUG] switch_ivr_play_say.c:1236 Codec
> Activated L16 at 8000hz 1 channels 20ms****
>
> 2011-09-12 10:30:13.388227 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed
> 0xb7192328 [1000]****
>
> 2011-09-12 10:30:13.388227 [DEBUG] mpf_timer_manager.c:111 Set Timer
> 0xb7192328 [2000]****
>
> 2011-09-12 10:30:14.388378 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed
> 0xb7192328 [2000]****
>
> 2011-09-12 10:30:14.388378 [DEBUG] mpf_timer_manager.c:111 Set Timer
> 0xb7192328 [3000]****
>
> 2011-09-12 10:30:14.976914 [DEBUG] switch_ivr_play_say.c:1573 done playing
> file****
>
> 2011-09-12 10:30:15.387961 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed
> 0xb7192328 [3000]****
>
> 2011-09-12 10:30:15.387961 [DEBUG] mpf_timer_manager.c:111 Set Timer
> 0xb7192328 [4000]****
>
> 2011-09-12 10:30:16.388597 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed
> 0xb7192328 [4000]****
>
> 2011-09-12 10:30:16.388597 [DEBUG] mpf_timer_manager.c:111 Set Timer
> 0xb7192328 [5000]****
>
> 2011-09-12 10:30:17.387667 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed
> 0xb7192310 [5000]****
>
> 2011-09-12 10:30:17.387667 [INFO] mpf_rtp_stream.c:1092 Generate RTCP SR
> [ssrc:2706393383 s:244 o:39040 ts:39440]****
>
> 2011-09-12 10:30:17.387667 [INFO] mpf_rtp_stream.c:1230 Send Compound RTCP
> Packet [48 bytes] 10.3.5.80:4017 -> 10.3.15.180:7971****
>
> 2011-09-12 10:30:17.387667 [DEBUG] mpf_timer_manager.c:111 Set Timer
> 0xb7192310 [10000]****
>
> 2011-09-12 10:30:17.387667 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed
> 0xb7192328 [5000]****
>
> 2011-09-12 10:30:17.387667 [DEBUG] mpf_timer_manager.c:111 Set Timer
> 0xb7192328 [6000]****
>
> 2011-09-12 10:30:18.388818 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed
> 0xb7192328 [6000]****
>
> 2011-09-12 10:30:18.388818 [DEBUG] mpf_timer_manager.c:111 Set Timer
> 0xb7192328 [7000]****
>
> 2011-09-12 10:30:19.387895 [DEBUG] mpf_timer_manager.c:180 Timer Elapsed
> 0xb7192328 [7000]****
>
> 2011-09-12 10:30:19.387895 [DEBUG] mpf_timer_manager.c:111 Set Timer
> 0xb7192328 [8000]****
>
> 2011-09-12 10:30:19.906918 [INFO] mrcp_client_connection.c:525 Receive
> MRCPv2 Stream 10.3.5.80:40061 <-> 10.3.15.180:6075 [140 bytes]****
>
> MRCP/2.0 140 RECOGNITION-COMPLETE 2 COMPLETE****
>
> Channel-Identifier: 40 at speechrecog****
>
> Waveform-URI:****
>
> Completion-Cause: 002 no-input-timeout****
>
>
> --------------------------------------------------------------------------------------------------------------------------------
> ****
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From aakashviswam at gmail.com  Wed Sep 14 17:54:56 2011
From: aakashviswam at gmail.com (Aakash)
Date: Wed, 14 Sep 2011 06:54:56 -0700 (PDT)
Subject: [Freeswitch-users] call forwarding how to test
In-Reply-To: <CAHKqMb7YqYnMz8pv=Up+7OEoiPqLDDwKnYSHYmx5KsfZQYM3dw@mail.gmail.com>
References: <1315940995184-6788875.post@n2.nabble.com>
	<CAHKqMb6ZFpj5GH5Wj8XQkNQjZUu3zXeOjNbbCHcJVU3yUB0u3Q@mail.gmail.com>
	<1315947845640-6789287.post@n2.nabble.com>
	<CAHKqMb7YqYnMz8pv=Up+7OEoiPqLDDwKnYSHYmx5KsfZQYM3dw@mail.gmail.com>
Message-ID: <1316008496762-6792618.post@n2.nabble.com>

When i try to dial *72 and entering the extension number eg 1020 and pressing
# .I am getting an error Temporarily unavailable *72

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/call-forwarding-how-to-test-tp6788875p6792618.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From covici at ccs.covici.com  Wed Sep 14 18:57:21 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 14 Sep 2011 10:57:21 -0400
Subject: [Freeswitch-users] How can I get setDTMFCallback in mod_managed
Message-ID: <16805.1316012241@ccs.covici.com>

Hi.  I have a program where I need to use setDTMFCallback because I want
to actually interrupt a menu prompt if a dtmf is received and the other
receive function does not stop the prompt.  I tried returning "stop" but
that did not work -- it works on a file, but not a tts prompt whereas I
know setDTMFCallback does work -- at least in another language.

So how can I either enable setDTMFCallback or otherwise work around this
limitation?

Any assistance would be appreciated.

-- 
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 gcd at i.ph  Wed Sep 14 19:11:42 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Wed, 14 Sep 2011 23:11:42 +0800
Subject: [Freeswitch-users] call forwarding how to test
In-Reply-To: <1316008496762-6792618.post@n2.nabble.com>
References: <1315940995184-6788875.post@n2.nabble.com>
	<CAHKqMb6ZFpj5GH5Wj8XQkNQjZUu3zXeOjNbbCHcJVU3yUB0u3Q@mail.gmail.com>
	<1315947845640-6789287.post@n2.nabble.com>
	<CAHKqMb7YqYnMz8pv=Up+7OEoiPqLDDwKnYSHYmx5KsfZQYM3dw@mail.gmail.com>
	<1316008496762-6792618.post@n2.nabble.com>
Message-ID: <CAHKqMb7Dh4YqMzczijY0tOWi1kWy12HC0sDz3CFa1DZkOFz6zw@mail.gmail.com>

is extension 1020 working? cuz the default configuration doesn't hv
extension 1020.

On Wed, Sep 14, 2011 at 9:54 PM, Aakash <aakashviswam at gmail.com> wrote:

> When i try to dial *72 and entering the extension number eg 1020 and
> pressing
> # .I am getting an error Temporarily unavailable *72
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/call-forwarding-how-to-test-tp6788875p6792618.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rendyfrx at gmail.com  Wed Sep 14 13:02:40 2011
From: rendyfrx at gmail.com (rendyfrx)
Date: Wed, 14 Sep 2011 02:02:40 -0700 (PDT)
Subject: [Freeswitch-users] [mod_xml_curl] Only accept if Return Password is
	blank
Message-ID: <1315990960205-6791769.post@n2.nabble.com>

Hi, 
I'm trying to setup freeswitch to authenticate SIP client login (SIP
register) to my backend database via php, example: username=65972932222,
pwd=1234 
As I couldn't really find sample return that suit my need, so I tried with
below xml return. Problem is if the password is blank, then it will
accepted, but if I change to simple password 1234 as xml return below, I
will get  SIP/2.0 403 Forbidden. 

xml 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>

I tried add in groups, but it doesn't have any impact.
The bindings' auth-scheme value is "basic", mod_xml_curl will call another
server (LAN) and it will validate and return the xml above.

Anybody can advise what I might be missing here? 
Thanks in advance


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-xml-curl-Only-accept-if-Return-Password-is-blank-tp6791769p6791769.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble_01394 at slickdeals.endjunk.com  Wed Sep 14 15:57:14 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Wed, 14 Sep 2011 04:57:14 -0700 (PDT)
Subject: [Freeswitch-users] No Rule to make target 'tport/libtport.la'
In-Reply-To: <CA+PZoDS4iJhrS=6Z91y5NCeeQ4a1q1+H+hQBNK-DC97X+BZCgQ@mail.gmail.com>
References: <CA+PZoDS4iJhrS=6Z91y5NCeeQ4a1q1+H+hQBNK-DC97X+BZCgQ@mail.gmail.com>
Message-ID: <1316001434399-6792206.post@n2.nabble.com>

I have been experiencing and/or seeing this issue for quite sometimes when
trying to cross compile FS.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/No-Rule-to-make-target-tport-libtport-la-tp6782434p6792206.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Wed Sep 14 19:12:09 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Sep 2011 08:12:09 -0700
Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today
Message-ID: <CAKzWOxXPkB2JymJ+EO2c-RanQd0Bued5F_KEf+J29Gku2-sA9g@mail.gmail.com>

Hello all!

Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_09_14

We have Dave Kompel (resident Windows hacking guru) to talk to us about
NuGet. He's also found a possible new solution for our build/test
challenges.

-MC
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From msc at freeswitch.org  Wed Sep 14 19:16:57 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Sep 2011 08:16:57 -0700
Subject: [Freeswitch-users] Timeout Message or something similar
In-Reply-To: <1315991590857-6791788.post@n2.nabble.com>
References: <1315837848697-6783735.post@n2.nabble.com>
	<CAMwZ-tK6GT7EdtCFdZqNTahfGp1BHt45jBUXt68eGfbKv7TBig@mail.gmail.com>
	<1315907331256-6786830.post@n2.nabble.com>
	<CAKzWOxUszOu8xGn70+UtaCOtqFC=WvEObORE+vWpdFmXKXSLgA@mail.gmail.com>
	<1315991590857-6791788.post@n2.nabble.com>
Message-ID: <CAKzWOxVmEjdPAKxbzyk2KUZ7DyC7xzga54g1e0s-Ksd3RR1ozg@mail.gmail.com>

Actually, you can use the second method and still be "dynamic". What are all
of the parameters that must be set at run time?
-MC

On Wed, Sep 14, 2011 at 2:13 AM, Brtantunes <brtantunes at gmail.com> wrote:

> Hello,
>
> Thank you for your reply,
>
> I tried the first command:
> /originate sofia/DOMAIN/1001
>
> bridge:{fail_on_single_reject=true,ignore_early_media=true,originate_continue_on_timeout=true}sofia/DOMAIN/1002,playback:C:/teste2.wav
> inline/
> but it dosen't work :(.
>
> Then i tried the second approach and it works. The problem is i want to
> configure all parameters in runtime, so i need the first approah. Is there
> a
> way to see the command generated using XML default approach? Or can you
> guys
> tell me which command to use?
>
> Tks for your help,
> Greetings
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Timeout-Message-or-something-similar-tp6783735p6791788.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From aakashviswam at gmail.com  Wed Sep 14 19:23:43 2011
From: aakashviswam at gmail.com (Aakash)
Date: Wed, 14 Sep 2011 08:23:43 -0700 (PDT)
Subject: [Freeswitch-users] call forwarding how to test
In-Reply-To: <CAHKqMb7Dh4YqMzczijY0tOWi1kWy12HC0sDz3CFa1DZkOFz6zw@mail.gmail.com>
References: <1315940995184-6788875.post@n2.nabble.com>
	<CAHKqMb6ZFpj5GH5Wj8XQkNQjZUu3zXeOjNbbCHcJVU3yUB0u3Q@mail.gmail.com>
	<1315947845640-6789287.post@n2.nabble.com>
	<CAHKqMb7YqYnMz8pv=Up+7OEoiPqLDDwKnYSHYmx5KsfZQYM3dw@mail.gmail.com>
	<1316008496762-6792618.post@n2.nabble.com>
	<CAHKqMb7Dh4YqMzczijY0tOWi1kWy12HC0sDz3CFa1DZkOFz6zw@mail.gmail.com>
Message-ID: <1316013823287-6793069.post@n2.nabble.com>

Hi Nandy,

Its a working extension.I do no where i am wrong.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/call-forwarding-how-to-test-tp6788875p6793069.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Wed Sep 14 19:26:45 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Sep 2011 08:26:45 -0700
Subject: [Freeswitch-users] Detecting silence in recorded file
In-Reply-To: <4E6F5CCA.3060801@newpace.ca>
References: <4E6F5CCA.3060801@newpace.ca>
Message-ID: <CAKzWOxWFnez-SmsKrbGJ8DR13wXOueLd7OZPgbrnD2qdz8aqFg@mail.gmail.com>

Do you need to do something after the call is over, like in a
post-processing cron job or hangup hook? If so you can probably apply the
principles in Kielhofner's Recqual project. I wrote up some info on it here:
http://www.freeswitch.org/node/297

That story has links plus a recap of what Recqual is all about. The trick is
using ecasound's "chains" and "gates" to allow/deny sections of audio to
pass through from the source to the destination. If you have a gate that
allows only "non silence" to pass through to the target file then you can
compare the size of the source to the size of the destination, or you can
just check the absolute size of the destination file. In any case, if you
don't mind a clever-though-less-than-perfect solution then check it out.

-MC

On Tue, Sep 13, 2011 at 6:38 AM, Adam Kelloway <adam.kelloway at newpace.ca>wrote:

> Hi there,
>
> Is there way to do any the following when using the record dial plan tool?
>
> - prevent the record tool from writing any samples to the destination
> file if the audio is not above the given silence threshold
> - detect whether a recorded file contains any audio above the given
> silence threshold
>
> I would like to treat recordings that are purely "silence" differently
> from those that contain audio above the silence threshold (such as
> speech). I would be interested in knowing if anyone has ever done this,
> and what your approach was.
>
> Thanks,
> Adam
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Hector.Geraldino at ip-soft.net  Wed Sep 14 19:48:23 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Wed, 14 Sep 2011 11:48:23 -0400
Subject: [Freeswitch-users] How can I get setDTMFCallback in mod_managed
In-Reply-To: <16805.1316012241@ccs.covici.com>
References: <16805.1316012241@ccs.covici.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD02206EC6F9@NY1-EXMB-01.ip-soft.net>

I've a java application that controls FS using ESL (mod_event_socket), and I can stop any prompt (playing audio file or performing text-to-speech) by sending a BREAK command.

See: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_break


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
Sent: Wednesday, September 14, 2011 10:57 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] How can I get setDTMFCallback in mod_managed

Hi.  I have a program where I need to use setDTMFCallback because I want
to actually interrupt a menu prompt if a dtmf is received and the other
receive function does not stop the prompt.  I tried returning "stop" but
that did not work -- it works on a file, but not a tts prompt whereas I
know setDTMFCallback does work -- at least in another language.

So how can I either enable setDTMFCallback or otherwise work around this
limitation?

Any assistance would be appreciated.

-- 
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 jack at livecall.com  Wed Sep 14 19:50:55 2011
From: jack at livecall.com (Jack)
Date: Wed, 14 Sep 2011 08:50:55 -0700
Subject: [Freeswitch-users] [mod_xml_curl] Only accept if Return
 Password is blank
In-Reply-To: <1315990960205-6791769.post@n2.nabble.com>
References: <1315990960205-6791769.post@n2.nabble.com>
Message-ID: <4E70CD5F.1010206@livecall.com>

When I check my database and I have no matching entry I return this to 
FS from curl

"<?xml version=\"1.0\" encoding=\"UTF-8\" standalone=\"no\"?>\r\n" +
                             "<document type=\"freeswitch/xml\">\r\n" +
                             " <section name=\"directory\">\r\n" +
                             " <result status=\"not found\"/>\r\n" +
                             " </section>\r\n" +
                             "</document>\r\n"
if it is found I return all information that would be in the static 
directory xml for that user.
You should also be returning the password, FreeSwitch checks it against 
what is sent from your client to authenticate.
This is probably why it works when use a blank password.

this scenario works like a charm for me.
Jack

On 9/14/2011 2:02 AM, rendyfrx wrote:
> Hi,
> I'm trying to setup freeswitch to authenticate SIP client login (SIP
> register) to my backend database via php, example: username=65972932222,
> pwd=1234
> As I couldn't really find sample return that suit my need, so I tried with
> below xml return. Problem is if the password is blank, then it will
> accepted, but if I change to simple password 1234 as xml return below, I
> will get  SIP/2.0 403 Forbidden.
>
> xml 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>
>        *<param name=\"password\" value="1234"/>*
>     </params>
>   </user>
> </domain>
> </section>
>
> I tried add in groups, but it doesn't have any impact.
> The bindings' auth-scheme value is "basic", mod_xml_curl will call another
> server (LAN) and it will validate and return the xml above.
>
> Anybody can advise what I might be missing here?
> Thanks in advance
>
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-xml-curl-Only-accept-if-Return-Password-is-blank-tp6791769p6791769.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jack at livecall.com  Wed Sep 14 19:57:53 2011
From: jack at livecall.com (Jack)
Date: Wed, 14 Sep 2011 08:57:53 -0700
Subject: [Freeswitch-users] [mod_xml_curl] Only accept if Return
 Password is blank
In-Reply-To: <4E70CD5F.1010206@livecall.com>
References: <1315990960205-6791769.post@n2.nabble.com>
	<4E70CD5F.1010206@livecall.com>
Message-ID: <4E70CF01.9080406@livecall.com>

Sorry, the param should not include the backslashes.
When it reaches Freeswitch it shoud look like this:

*<param name="password" value="1234"/>*



On 9/14/2011 8:50 AM, Jack wrote:
> When I check my database and I have no matching entry I return this to 
> FS from curl
>
> "<?xml version=\"1.0\" encoding=\"UTF-8\" standalone=\"no\"?>\r\n" +
>                             "<document type=\"freeswitch/xml\">\r\n" +
>                             " <section name=\"directory\">\r\n" +
>                             " <result status=\"not found\"/>\r\n" +
>                             " </section>\r\n" +
>                             "</document>\r\n"
> if it is found I return all information that would be in the static 
> directory xml for that user.
> You should also be returning the password, FreeSwitch checks it 
> against what is sent from your client to authenticate.
> This is probably why it works when use a blank password.
>
> this scenario works like a charm for me.
> Jack
>
> On 9/14/2011 2:02 AM, rendyfrx wrote:
>> Hi,
>> I'm trying to setup freeswitch to authenticate SIP client login (SIP
>> register) to my backend database via php, example: username=65972932222,
>> pwd=1234
>> As I couldn't really find sample return that suit my need, so I tried with
>> below xml return. Problem is if the password is blank, then it will
>> accepted, but if I change to simple password 1234 as xml return below, I
>> will get  SIP/2.0 403 Forbidden.
>>
>> xml 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>
>>        *<param name=\"password\" value="1234"/>*
>>     </params>
>>   </user>
>> </domain>
>> </section>
>>
>> I tried add in groups, but it doesn't have any impact.
>> The bindings' auth-scheme value is "basic", mod_xml_curl will call another
>> server (LAN) and it will validate and return the xml above.
>>
>> Anybody can advise what I might be missing here?
>> Thanks in advance
>>
>>
>> --
>> View this message in context:http://freeswitch-users.2379917.n2.nabble.com/mod-xml-curl-Only-accept-if-Return-Password-is-blank-tp6791769p6791769.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From gcd at i.ph  Wed Sep 14 20:26:34 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Thu, 15 Sep 2011 00:26:34 +0800
Subject: [Freeswitch-users] call forwarding how to test
In-Reply-To: <1316013823287-6793069.post@n2.nabble.com>
References: <1315940995184-6788875.post@n2.nabble.com>
	<CAHKqMb6ZFpj5GH5Wj8XQkNQjZUu3zXeOjNbbCHcJVU3yUB0u3Q@mail.gmail.com>
	<1315947845640-6789287.post@n2.nabble.com>
	<CAHKqMb7YqYnMz8pv=Up+7OEoiPqLDDwKnYSHYmx5KsfZQYM3dw@mail.gmail.com>
	<1316008496762-6792618.post@n2.nabble.com>
	<CAHKqMb7Dh4YqMzczijY0tOWi1kWy12HC0sDz3CFa1DZkOFz6zw@mail.gmail.com>
	<1316013823287-6793069.post@n2.nabble.com>
Message-ID: <CAHKqMb6m09OnaXhJxiEg_mMpnzOrOWSFjeY5wyF952uRMkm01A@mail.gmail.com>

what is the error message in the log?


On Wed, Sep 14, 2011 at 11:23 PM, Aakash <aakashviswam at gmail.com> wrote:

> Hi Nandy,
>
> Its a working extension.I do no where i am wrong.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/call-forwarding-how-to-test-tp6788875p6793069.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Sep 14 20:33:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Sep 2011 09:33:49 -0700
Subject: [Freeswitch-users] New Chinese sound files for review
Message-ID: <CAKzWOxVnHsP8Q8c-ivTCxAFVV2y8K7bGh0MsCL9EgjNn0Fdbag@mail.gmail.com>

To all Chinese speakers:

Seven Du has graciously put together new zh_CN sound files:

ftp://ftp.x-y-t.com/pub/freeswitch-sounds-zh-cn-link-8000-1.0.0.tar.gz
ftp://ftp.x-y-t.com/pub/freeswitch-sounds-zh-cn-link-8000-1.0.0.tar.gz.md5

Here's a jira for the phrase xml:

http://jira.freeswitch.org/browse/FS-3568

If you understand Chinese then please consider helping Seven Du out by
downloading and listening to the sounds.

Thanks!
-Michael
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From cvogel at lyonl.com  Wed Sep 14 19:44:23 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Wed, 14 Sep 2011 15:44:23 +0000
Subject: [Freeswitch-users] large block of DID's
Message-ID: <D25CC3E5-C0D2-41D7-87C3-843C24A0CA8C@lyonl.com>

I was wondering if anyone has any thoughts what is the best way to manage a large blocks of DID? (we have about 1600 DID's that we are moving off of our asterisk servers) 

Chad


From rendyfrx at gmail.com  Wed Sep 14 20:37:52 2011
From: rendyfrx at gmail.com (rendyfrx)
Date: Wed, 14 Sep 2011 09:37:52 -0700 (PDT)
Subject: [Freeswitch-users] [mod_xml_curl] Only accept if Return
	Password is blank
In-Reply-To: <4E70CF01.9080406@livecall.com>
References: <1315990960205-6791769.post@n2.nabble.com>
	<4E70CD5F.1010206@livecall.com> <4E70CF01.9080406@livecall.com>
Message-ID: <1316018272976-6793355.post@n2.nabble.com>

Hi Jack,
Thanks for your reply.
I think the param did not include the backslashes, probably due to this
forum's text display only, so it should not have backslashes when reaches
FS.

Would you mind sharing with me one sample static directory xml return for
any particular user? I can try it out the format.
Also is there any special config that I might missed?

Your help is much appreciated.
Thanks.



--
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From avi at avimarcus.net  Wed Sep 14 20:43:04 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 14 Sep 2011 19:43:04 +0300
Subject: [Freeswitch-users] large block of DID's
In-Reply-To: <D25CC3E5-C0D2-41D7-87C3-843C24A0CA8C@lyonl.com>
References: <AQHMcvUsnvtPAGNZgUiSUe0ZdZz8gg==>
	<D25CC3E5-C0D2-41D7-87C3-843C24A0CA8C@lyonl.com>
Message-ID: <CALNZuwkUXV5V+7FFS6yJGL-CZcxJb913Nzpe1R62xCZxw9cM8Q@mail.gmail.com>

For routing?
SQL table to keep track.. then to use, call mod_odbc_query or a simple lua
script to get the info.

-Avi


On Wed, Sep 14, 2011 at 6:44 PM, Chad Vogel <cvogel at lyonl.com> wrote:

> I was wondering if anyone has any thoughts what is the best way to manage a
> large blocks of DID? (we have about 1600 DID's that we are moving off of our
> asterisk servers)
>
> Chad
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From aakashviswam at gmail.com  Wed Sep 14 20:50:38 2011
From: aakashviswam at gmail.com (Aakash)
Date: Wed, 14 Sep 2011 09:50:38 -0700 (PDT)
Subject: [Freeswitch-users] call forwarding how to test
In-Reply-To: <CAHKqMb6m09OnaXhJxiEg_mMpnzOrOWSFjeY5wyF952uRMkm01A@mail.gmail.com>
References: <1315940995184-6788875.post@n2.nabble.com>
	<CAHKqMb6ZFpj5GH5Wj8XQkNQjZUu3zXeOjNbbCHcJVU3yUB0u3Q@mail.gmail.com>
	<1315947845640-6789287.post@n2.nabble.com>
	<CAHKqMb7YqYnMz8pv=Up+7OEoiPqLDDwKnYSHYmx5KsfZQYM3dw@mail.gmail.com>
	<1316008496762-6792618.post@n2.nabble.com>
	<CAHKqMb7Dh4YqMzczijY0tOWi1kWy12HC0sDz3CFa1DZkOFz6zw@mail.gmail.com>
	<1316013823287-6793069.post@n2.nabble.com>
	<CAHKqMb6m09OnaXhJxiEg_mMpnzOrOWSFjeY5wyF952uRMkm01A@mail.gmail.com>
Message-ID: <1316019038624-6793424.post@n2.nabble.com>



Please have a look on my logs

http://pastebin.freeswitch.org/17330

--
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From aakashviswam at gmail.com  Wed Sep 14 20:51:00 2011
From: aakashviswam at gmail.com (Aakash)
Date: Wed, 14 Sep 2011 09:51:00 -0700 (PDT)
Subject: [Freeswitch-users] call forwarding how to test
In-Reply-To: <CAHKqMb6m09OnaXhJxiEg_mMpnzOrOWSFjeY5wyF952uRMkm01A@mail.gmail.com>
References: <1315940995184-6788875.post@n2.nabble.com>
	<CAHKqMb6ZFpj5GH5Wj8XQkNQjZUu3zXeOjNbbCHcJVU3yUB0u3Q@mail.gmail.com>
	<1315947845640-6789287.post@n2.nabble.com>
	<CAHKqMb7YqYnMz8pv=Up+7OEoiPqLDDwKnYSHYmx5KsfZQYM3dw@mail.gmail.com>
	<1316008496762-6792618.post@n2.nabble.com>
	<CAHKqMb7Dh4YqMzczijY0tOWi1kWy12HC0sDz3CFa1DZkOFz6zw@mail.gmail.com>
	<1316013823287-6793069.post@n2.nabble.com>
	<CAHKqMb6m09OnaXhJxiEg_mMpnzOrOWSFjeY5wyF952uRMkm01A@mail.gmail.com>
Message-ID: <1316019060771-6793428.post@n2.nabble.com>



Please have a look on my logs

http://pastebin.freeswitch.org/17330

--
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From jack at livecall.com  Wed Sep 14 20:56:10 2011
From: jack at livecall.com (Jack)
Date: Wed, 14 Sep 2011 09:56:10 -0700
Subject: [Freeswitch-users] [mod_xml_curl] Only accept if Return
 Password is blank
In-Reply-To: <1316018272976-6793355.post@n2.nabble.com>
References: <1315990960205-6791769.post@n2.nabble.com>
	<4E70CD5F.1010206@livecall.com> <4E70CF01.9080406@livecall.com>
	<1316018272976-6793355.post@n2.nabble.com>
Message-ID: <4E70DCAA.1040400@livecall.com>

This is returned for a found user.  The querystring[] variables are 
filled from the POST FreeSwitch made in the fetch to curl.
PassWordFromDatabase would be a variable filled from your database lookup.

If it is posting to your curl http server you are probably set up 
correctly.
If you turn your http server off and call into Freeswitch you should see 
the POST attempt on the FS console


"<?xml version=\"1.0\" encoding=\"UTF-8\" standalone=\"no\"?>\r\n" +
                             "<document type=\"freeswitch/xml\">\r\n" +
                             " <section name=\"directory\" 
description=\"Directory For FreeSwitch\">\r\n" +
                             " <domain name=\"" + queryString["domain"] 
+ "\">\r\n" +
                             " <user id=\"" + queryString["user"] + 
"\">\r\n" +
                             " <params>\r\n" +
                             " <param name=\"password\" value=\"" + 
PasswordFromDatabase + "\"/>\r\n" +
                             " </params>\r\n" +
                             " <variables>\r\n" +
                             " <variable name=\"user_context\" 
value=\"default\"/>\r\n" +
                             " </variables>\r\n" +
                             " </user>\r\n" +
                             " </domain>\r\n" +
                             " </section>\r\n" +
                             "</document>\r\n"

On 9/14/2011 9:37 AM, rendyfrx wrote:
> Hi Jack,
> Thanks for your reply.
> I think the param did not include the backslashes, probably due to this
> forum's text display only, so it should not have backslashes when reaches
> FS.
>
> Would you mind sharing with me one sample static directory xml return for
> any particular user? I can try it out the format.
> Also is there any special config that I might missed?
>
> Your help is much appreciated.
> Thanks.
>
>
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-xml-curl-Only-accept-if-Return-Password-is-blank-tp6791769p6793355.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 msc at freeswitch.org  Wed Sep 14 21:07:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Sep 2011 10:07:44 -0700
Subject: [Freeswitch-users] call forwarding how to test
In-Reply-To: <1316019038624-6793424.post@n2.nabble.com>
References: <1315940995184-6788875.post@n2.nabble.com>
	<CAHKqMb6ZFpj5GH5Wj8XQkNQjZUu3zXeOjNbbCHcJVU3yUB0u3Q@mail.gmail.com>
	<1315947845640-6789287.post@n2.nabble.com>
	<CAHKqMb7YqYnMz8pv=Up+7OEoiPqLDDwKnYSHYmx5KsfZQYM3dw@mail.gmail.com>
	<1316008496762-6792618.post@n2.nabble.com>
	<CAHKqMb7Dh4YqMzczijY0tOWi1kWy12HC0sDz3CFa1DZkOFz6zw@mail.gmail.com>
	<1316013823287-6793069.post@n2.nabble.com>
	<CAHKqMb6m09OnaXhJxiEg_mMpnzOrOWSFjeY5wyF952uRMkm01A@mail.gmail.com>
	<1316019038624-6793424.post@n2.nabble.com>
Message-ID: <CAKzWOxULsj-wzGKzF0X1qj2iQoeZ5VRo__mvj8F8QiSGk6bcoA@mail.gmail.com>

That log looks perfect. Note line #108:
EXECUTE sofia/internal/1210 at sip.agshealth.com hash(insert/10.10.10.20
-call_forward/1210/1010)

That's inserting the CF info into the database. This is working. Please
pastebin the debug log of whatever isn't working. (I'm guessing that when
someone calls "1210" it isn't CFing to 1010...)

-MC

On Wed, Sep 14, 2011 at 9:50 AM, Aakash <aakashviswam at gmail.com> wrote:

>
>
> Please have a look on my logs
>
> http://pastebin.freeswitch.org/17330
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/call-forwarding-how-to-test-tp6788875p6793424.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From aakashviswam at gmail.com  Wed Sep 14 21:56:24 2011
From: aakashviswam at gmail.com (Aakash)
Date: Wed, 14 Sep 2011 10:56:24 -0700 (PDT)
Subject: [Freeswitch-users] call forwarding how to test
In-Reply-To: <CAKzWOxULsj-wzGKzF0X1qj2iQoeZ5VRo__mvj8F8QiSGk6bcoA@mail.gmail.com>
References: <1315940995184-6788875.post@n2.nabble.com>
	<CAHKqMb6ZFpj5GH5Wj8XQkNQjZUu3zXeOjNbbCHcJVU3yUB0u3Q@mail.gmail.com>
	<1315947845640-6789287.post@n2.nabble.com>
	<CAHKqMb7YqYnMz8pv=Up+7OEoiPqLDDwKnYSHYmx5KsfZQYM3dw@mail.gmail.com>
	<1316008496762-6792618.post@n2.nabble.com>
	<CAHKqMb7Dh4YqMzczijY0tOWi1kWy12HC0sDz3CFa1DZkOFz6zw@mail.gmail.com>
	<1316013823287-6793069.post@n2.nabble.com>
	<CAHKqMb6m09OnaXhJxiEg_mMpnzOrOWSFjeY5wyF952uRMkm01A@mail.gmail.com>
	<1316019038624-6793424.post@n2.nabble.com>
	<CAKzWOxULsj-wzGKzF0X1qj2iQoeZ5VRo__mvj8F8QiSGk6bcoA@mail.gmail.com>
Message-ID: <1316022984994-6793773.post@n2.nabble.com>

I am not getting any error in my logs.When i try to call 1210 from extension
1234 ,the call is not forwarding to extension 1010.

logs for your reference http://pastebin.freeswitch.org/17333

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From javieraristizabal at gmail.com  Wed Sep 14 22:11:27 2011
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Wed, 14 Sep 2011 13:11:27 -0500
Subject: [Freeswitch-users] large block of DID's
In-Reply-To: <CALNZuwkUXV5V+7FFS6yJGL-CZcxJb913Nzpe1R62xCZxw9cM8Q@mail.gmail.com>
References: <D25CC3E5-C0D2-41D7-87C3-843C24A0CA8C@lyonl.com>
	<CALNZuwkUXV5V+7FFS6yJGL-CZcxJb913Nzpe1R62xCZxw9cM8Q@mail.gmail.com>
Message-ID: <CAF4RhHNpqR99uxPA38=O0Nge333tUb-mu+PvMcC2pOcWzmz0iA@mail.gmail.com>

Look at this: http://wiki.freeswitch.org/wiki/Mod_easyroute



On Wed, Sep 14, 2011 at 11:43 AM, Avi Marcus <avi at avimarcus.net> wrote:

> For routing?
> SQL table to keep track.. then to use, call mod_odbc_query or a simple lua
> script to get the info.
>
> -Avi
>
>
> On Wed, Sep 14, 2011 at 6:44 PM, Chad Vogel <cvogel at lyonl.com> wrote:
>
>> I was wondering if anyone has any thoughts what is the best way to manage
>> a large blocks of DID? (we have about 1600 DID's that we are moving off of
>> our asterisk servers)
>>
>> Chad
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Javier Aristiz?bal
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From msc at freeswitch.org  Wed Sep 14 22:20:03 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Sep 2011 11:20:03 -0700
Subject: [Freeswitch-users] call forwarding how to test
In-Reply-To: <1316022984994-6793773.post@n2.nabble.com>
References: <1315940995184-6788875.post@n2.nabble.com>
	<CAHKqMb6ZFpj5GH5Wj8XQkNQjZUu3zXeOjNbbCHcJVU3yUB0u3Q@mail.gmail.com>
	<1315947845640-6789287.post@n2.nabble.com>
	<CAHKqMb7YqYnMz8pv=Up+7OEoiPqLDDwKnYSHYmx5KsfZQYM3dw@mail.gmail.com>
	<1316008496762-6792618.post@n2.nabble.com>
	<CAHKqMb7Dh4YqMzczijY0tOWi1kWy12HC0sDz3CFa1DZkOFz6zw@mail.gmail.com>
	<1316013823287-6793069.post@n2.nabble.com>
	<CAHKqMb6m09OnaXhJxiEg_mMpnzOrOWSFjeY5wyF952uRMkm01A@mail.gmail.com>
	<1316019038624-6793424.post@n2.nabble.com>
	<CAKzWOxULsj-wzGKzF0X1qj2iQoeZ5VRo__mvj8F8QiSGk6bcoA@mail.gmail.com>
	<1316022984994-6793773.post@n2.nabble.com>
Message-ID: <CAKzWOxV917_bWqPO9_kGbV5Tmu2zu-J0wV4CEWULMsgrrwOyqQ@mail.gmail.com>

Aakash,

The trace you posted looks to me to be from 1210 to *72:
2011-09-14 19:01:00.932980 [INFO] mod_dialplan_xml.c:336 Processing VSI Tech
<1210>->*72 in context default

I don't see any indication of a call from 1234 to 1210. Perhaps you did not
get the whole trace?

-MC

On Wed, Sep 14, 2011 at 10:56 AM, Aakash <aakashviswam at gmail.com> wrote:

> I am not getting any error in my logs.When i try to call 1210 from
> extension
> 1234 ,the call is not forwarding to extension 1010.
>
> logs for your reference http://pastebin.freeswitch.org/17333
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/call-forwarding-how-to-test-tp6788875p6793773.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gerry at pstn2.net  Wed Sep 14 22:37:39 2011
From: gerry at pstn2.net (Gerry Hull)
Date: Wed, 14 Sep 2011 14:37:39 -0400
Subject: [Freeswitch-users] What Provider do you guys use?
In-Reply-To: <CALNZuwm-RsUL56Y=gqmr3=aMysrq0RsyshUjm9yEGznvNukNXg@mail.gmail.com>
References: <CADrcvV86dR63-G6_-_KkpLSVh7UVEosT9gwNOJR-vK-Y37ibNA@mail.gmail.com>
	<CALNZuwm-RsUL56Y=gqmr3=aMysrq0RsyshUjm9yEGznvNukNXg@mail.gmail.com>
Message-ID: <CABWdUKAj8BX=0=YcyrsBg38J_pvBpUAqGGbHSDt00K4bz5kSKw@mail.gmail.com>

I've used voip.ms' value route to western europe for over a year; it's been
about 1000 minutes of traffic, with no call quality issues.   Half a cent a
minute is pretty reasonable, IMHO.


On Sat, Sep 10, 2011 at 2:38 PM, Avi Marcus <avi at avimarcus.net> wrote:

> 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
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From aakashviswam at gmail.com  Wed Sep 14 22:46:24 2011
From: aakashviswam at gmail.com (Aakash)
Date: Wed, 14 Sep 2011 11:46:24 -0700 (PDT)
Subject: [Freeswitch-users] call forwarding how to test
In-Reply-To: <CAKzWOxV917_bWqPO9_kGbV5Tmu2zu-J0wV4CEWULMsgrrwOyqQ@mail.gmail.com>
References: <1315947845640-6789287.post@n2.nabble.com>
	<CAHKqMb7YqYnMz8pv=Up+7OEoiPqLDDwKnYSHYmx5KsfZQYM3dw@mail.gmail.com>
	<1316008496762-6792618.post@n2.nabble.com>
	<CAHKqMb7Dh4YqMzczijY0tOWi1kWy12HC0sDz3CFa1DZkOFz6zw@mail.gmail.com>
	<1316013823287-6793069.post@n2.nabble.com>
	<CAHKqMb6m09OnaXhJxiEg_mMpnzOrOWSFjeY5wyF952uRMkm01A@mail.gmail.com>
	<1316019038624-6793424.post@n2.nabble.com>
	<CAKzWOxULsj-wzGKzF0X1qj2iQoeZ5VRo__mvj8F8QiSGk6bcoA@mail.gmail.com>
	<1316022984994-6793773.post@n2.nabble.com>
	<CAKzWOxV917_bWqPO9_kGbV5Tmu2zu-J0wV4CEWULMsgrrwOyqQ@mail.gmail.com>
Message-ID: <1316025984046-6794016.post@n2.nabble.com>

Please have a look --  http://pastebin.freeswitch.org/17333

You can see the indications like 
2011-09-14 19:01:16.454118 [INFO] mod_dialplan_xml.c:336 Processing Aakash
<1234>->1210 in context default


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From avi at avimarcus.net  Wed Sep 14 23:59:51 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 14 Sep 2011 22:59:51 +0300
Subject: [Freeswitch-users] What Provider do you guys use?
In-Reply-To: <CABWdUKAj8BX=0=YcyrsBg38J_pvBpUAqGGbHSDt00K4bz5kSKw@mail.gmail.com>
References: <CADrcvV86dR63-G6_-_KkpLSVh7UVEosT9gwNOJR-vK-Y37ibNA@mail.gmail.com>
	<CALNZuwm-RsUL56Y=gqmr3=aMysrq0RsyshUjm9yEGznvNukNXg@mail.gmail.com>
	<CABWdUKAj8BX=0=YcyrsBg38J_pvBpUAqGGbHSDt00K4bz5kSKw@mail.gmail.com>
Message-ID: <CALNZuwk3DrTCVMEhnBYSHGBHKkOxS+Q=+wL9aGf6D-LEii70Mw@mail.gmail.com>

I see they have UK for .79cents, but where else is 1/2 cent in Europe..?
-Avi


On Wed, Sep 14, 2011 at 9:37 PM, Gerry Hull <gerry at pstn2.net> wrote:

> I've used voip.ms' value route to western europe for over a year; it's
> been about 1000 minutes of traffic, with no call quality issues.   Half a
> cent a minute is pretty reasonable, IMHO.
>
>
> On Sat, Sep 10, 2011 at 2:38 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> 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
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gerry at pstn2.net  Thu Sep 15 00:38:02 2011
From: gerry at pstn2.net (Gerry Hull)
Date: Wed, 14 Sep 2011 16:38:02 -0400
Subject: [Freeswitch-users] What Provider do you guys use?
In-Reply-To: <CALNZuwk3DrTCVMEhnBYSHGBHKkOxS+Q=+wL9aGf6D-LEii70Mw@mail.gmail.com>
References: <CADrcvV86dR63-G6_-_KkpLSVh7UVEosT9gwNOJR-vK-Y37ibNA@mail.gmail.com>
	<CALNZuwm-RsUL56Y=gqmr3=aMysrq0RsyshUjm9yEGznvNukNXg@mail.gmail.com>
	<CABWdUKAj8BX=0=YcyrsBg38J_pvBpUAqGGbHSDt00K4bz5kSKw@mail.gmail.com>
	<CALNZuwk3DrTCVMEhnBYSHGBHKkOxS+Q=+wL9aGf6D-LEii70Mw@mail.gmail.com>
Message-ID: <CABWdUKAjTrV2X7w6S8m7F9seDnwamfUfHqifKZB6H1_bTAEoLg@mail.gmail.com>

I stand corrected.   I'm routing some calls from eastern Canada to London
and back.  London is 0.0079, and Canada is 0.0052.



On Wed, Sep 14, 2011 at 3:59 PM, Avi Marcus <avi at avimarcus.net> wrote:

> I see they have UK for .79cents, but where else is 1/2 cent in Europe..?
> -Avi
>
>
> On Wed, Sep 14, 2011 at 9:37 PM, Gerry Hull <gerry at pstn2.net> wrote:
>
>> I've used voip.ms' value route to western europe for over a year; it's
>> been about 1000 minutes of traffic, with no call quality issues.   Half a
>> cent a minute is pretty reasonable, IMHO.
>>
>>
>> On Sat, Sep 10, 2011 at 2:38 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> 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
>>>>
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From buscom123+fs at gmail.com  Thu Sep 15 00:41:05 2011
From: buscom123+fs at gmail.com (R H)
Date: Wed, 14 Sep 2011 14:41:05 -0600
Subject: [Freeswitch-users] mod_managed switch_xml_open_cfg and SIGSEGV
Message-ID: <CAAtQP54ec9oTygRGeNdZOvXK7t-U0_79WM_hV9Bc791XDNp5Tg@mail.gmail.com>

Has anyone had success working with xml configuration files using
mod_managed? I have several modules that DO NOT override the xml
configuration but, rather, I actually want to configure them using the
freeswitch conf directory structure. I have not even tried to parse the xml
because I cant even load the xml. Here is a very basic example of what I
tried and the result I got:

The Code:
------------------------------------------------

using System;
using FreeSWITCH;
using FreeSWITCH.Native;

namespace PsiCallCenterAddon
{
public class FreeSwitchConfigLoader
{
public static void loadConfigTest()
{
try
{
switch_xml xml = freeswitch.switch_xml_open_cfg("freeswitch.conf", null,
null);
freeswitch.switch_xml_free(xml);
 Log.WriteLine(LogLevel.Alert, "Config Loader Success!");
}
catch
{
Log.WriteLine(LogLevel.Alert, "Config Loader Fail!");
}
}
}
}

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

The Result:
------------------------------------------------

Stacktrace:

  at (wrapper managed-to-native)
FreeSWITCH.Native.freeswitchPINVOKE.switch_xml_open_cfg
(string,System.Runtime.InteropServices.HandleRef,System.Runtime.InteropServices.HandleRef)
<0x000a3>
  at (wrapper managed-to-native)
FreeSWITCH.Native.freeswitchPINVOKE.switch_xml_open_cfg
(string,System.Runtime.InteropServices.HandleRef,System.Runtime.InteropServices.HandleRef)
<0x000a3>
  at FreeSWITCH.Native.freeswitch.switch_xml_open_cfg
(string,FreeSWITCH.Native.SWIGTYPE_p_p_switch_xml,FreeSWITCH.Native.switch_event)
<0x0004f>
  at PsiCallCenterAddon.FreeSwitchConfigLoader.loadConfigTest () <0x0001b>
  at PsiCallCenterAddon.QueueMonitorLoader.Load () <0x0000b>
  at FreeSWITCH.PluginManager.RunLoadNotify (System.Type[]) <0x00190>
  at FreeSWITCH.AsmPluginManager.LoadInternal (string) <0x001eb>
  at FreeSWITCH.PluginManager.Load (string) <0x000a7>
  at (wrapper remoting-invoke-with-check) FreeSWITCH.PluginManager.Load
(string) <0x00067>
  at (wrapper xdomain-dispatch) FreeSWITCH.PluginManager.Load
(object,byte[]&,byte[]&,string) <0x0017b>
  at (wrapper xdomain-invoke) FreeSWITCH.PluginManager.Load (string)
<0x00126>
  at (wrapper remoting-invoke-with-check) FreeSWITCH.PluginManager.Load
(string) <0x00047>
  at FreeSWITCH.Loader.loadFile (string) <0x004a3>
  at FreeSWITCH.Loader.Load () <0x0027b>
  at (wrapper runtime-invoke) <Module>.runtime_invoke_bool
(object,intptr,intptr,intptr) <0x00046>

Native stacktrace:

        /usr/lib64/libmono-2.0.so.1(+0xacba0) [0x7fc7ea58dba0]
        /usr/lib64/libmono-2.0.so.1(+0xfcedf) [0x7fc7ea5ddedf]
        /lib64/libpthread.so.0(+0xf2d0) [0x7fc7f1ceb2d0]

/usr/local/freeswitch/lib/libfreeswitch.so.1(switch_xml_open_cfg+0x1a)
[0x7fc7f260b0ea]
        [0x40155804]

Debug info from gdb:
=================================================================
Got a SIGSEGV while executing native code. This usually indicates
a fatal error in the mono runtime or one of the native libraries
used by your application.
=================================================================
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From rhuddleston at gmail.com  Thu Sep 15 00:44:59 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Wed, 14 Sep 2011 16:44:59 -0400
Subject: [Freeswitch-users] What Provider do you guys use?
In-Reply-To: <CABWdUKAjTrV2X7w6S8m7F9seDnwamfUfHqifKZB6H1_bTAEoLg@mail.gmail.com>
References: <CADrcvV86dR63-G6_-_KkpLSVh7UVEosT9gwNOJR-vK-Y37ibNA@mail.gmail.com>
	<CALNZuwm-RsUL56Y=gqmr3=aMysrq0RsyshUjm9yEGznvNukNXg@mail.gmail.com>
	<CABWdUKAj8BX=0=YcyrsBg38J_pvBpUAqGGbHSDt00K4bz5kSKw@mail.gmail.com>
	<CALNZuwk3DrTCVMEhnBYSHGBHKkOxS+Q=+wL9aGf6D-LEii70Mw@mail.gmail.com>
	<CABWdUKAjTrV2X7w6S8m7F9seDnwamfUfHqifKZB6H1_bTAEoLg@mail.gmail.com>
Message-ID: <027801cc731f$2c1b2a60$84517f20$@com>

Voip.ms and callwithus are my primary two

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gerry
Hull
Sent: Wednesday, September 14, 2011 4:38 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] What Provider do you guys use?

 


I stand corrected.   I'm routing some calls from eastern Canada to London
and back.  London is 0.0079, and Canada is
<http://p1.wisestamp.com/pixel.png?p=mozilla&v=2.7.4&t=1316032454527&u=ec446
5620c4ca423> 0.0052.  

 

On Wed, Sep 14, 2011 at 3:59 PM, Avi Marcus <avi at avimarcus.net> wrote:

I see they have UK for .79cents, but where else is 1/2 cent in Europe..?


-Avi

 

On Wed, Sep 14, 2011 at 9:37 PM, Gerry Hull <gerry at pstn2.net> wrote:

I've used voip.ms' value route to western europe for over a year; it's been
about 1000 minutes of traffic, with no call quality issues.   Half a cent a
minute is pretty reasonable, IMHO.

Error! Filename not specified.

 

On Sat, Sep 10, 2011 at 2:38 PM, Avi Marcus <avi at avimarcus.net> wrote:

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

 



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




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

 



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

 

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From msc at freeswitch.org  Thu Sep 15 02:50:15 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Sep 2011 15:50:15 -0700
Subject: [Freeswitch-users] call forwarding how to test
In-Reply-To: <1316025984046-6794016.post@n2.nabble.com>
References: <1315947845640-6789287.post@n2.nabble.com>
	<CAHKqMb7YqYnMz8pv=Up+7OEoiPqLDDwKnYSHYmx5KsfZQYM3dw@mail.gmail.com>
	<1316008496762-6792618.post@n2.nabble.com>
	<CAHKqMb7Dh4YqMzczijY0tOWi1kWy12HC0sDz3CFa1DZkOFz6zw@mail.gmail.com>
	<1316013823287-6793069.post@n2.nabble.com>
	<CAHKqMb6m09OnaXhJxiEg_mMpnzOrOWSFjeY5wyF952uRMkm01A@mail.gmail.com>
	<1316019038624-6793424.post@n2.nabble.com>
	<CAKzWOxULsj-wzGKzF0X1qj2iQoeZ5VRo__mvj8F8QiSGk6bcoA@mail.gmail.com>
	<1316022984994-6793773.post@n2.nabble.com>
	<CAKzWOxV917_bWqPO9_kGbV5Tmu2zu-J0wV4CEWULMsgrrwOyqQ@mail.gmail.com>
	<1316025984046-6794016.post@n2.nabble.com>
Message-ID: <CAKzWOxXMk_Za-xkM_H-WZQparVyRCk+G8H7K4pO+0obqaah6ZQ@mail.gmail.com>

Apologies, I didn't look down far enough.

Okay, what's happening is that the call from 1234 to 1210 is never hitting a
dialplan extension that checks the value of the CF information that you
entered with the *72 process. My guess is that you didn't do the part
mentioned on this page:

http://wiki.freeswitch.org/wiki/Call_Forward_Example

...under "Also, default.xml:"

So, the questions are:
Did you add the stuff to features.xml that the author shows?
Did you add the "call_forwarding_number" extension to default.xml?
If so, did you put it near the top of the file? (Don't forget that the
dialplan is parsed in the order listed in the file, so if you put that
"call_forwarding_number" too low in the file it won't be processed. I
recommend putting as the first extension in the file, just to be safe.)
Did you add the "is_forward" check inside the Local_Extension? (By the look
of your log I'd say not...)

If you want an alternative way of doing this then check out the
call-forwarding example that I posted on the wiki. There's a link to "Simple
CF with IVR" at the bottom of the wiki page that you are looking at. Perhaps
the information I present will give you some more clarity on how these kinds
of things work. I give a lot of explanatory text in that example, so check
it out.

Don't be discouraged - if you're new to all of this then the pains you're
experiencing are part of the learning process. My advice is to keep hacking
away at it. Also, buy our book! Darren and I wrote some good stuff in
chapters 8 and 5 (respectively) on the dialplan and how it works. The book
will help you build a good foundation. :)

-MC

On Wed, Sep 14, 2011 at 11:46 AM, Aakash <aakashviswam at gmail.com> wrote:

> Please have a look --  http://pastebin.freeswitch.org/17333
>
> You can see the indications like
> 2011-09-14 19:01:16.454118 [INFO] mod_dialplan_xml.c:336 Processing Aakash
> <1234>->1210 in context default
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/call-forwarding-how-to-test-tp6788875p6794016.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Sep 15 03:25:35 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Sep 2011 16:25:35 -0700
Subject: [Freeswitch-users] call forwarding how to test
In-Reply-To: <CAKzWOxXMk_Za-xkM_H-WZQparVyRCk+G8H7K4pO+0obqaah6ZQ@mail.gmail.com>
References: <1315947845640-6789287.post@n2.nabble.com>
	<CAHKqMb7YqYnMz8pv=Up+7OEoiPqLDDwKnYSHYmx5KsfZQYM3dw@mail.gmail.com>
	<1316008496762-6792618.post@n2.nabble.com>
	<CAHKqMb7Dh4YqMzczijY0tOWi1kWy12HC0sDz3CFa1DZkOFz6zw@mail.gmail.com>
	<1316013823287-6793069.post@n2.nabble.com>
	<CAHKqMb6m09OnaXhJxiEg_mMpnzOrOWSFjeY5wyF952uRMkm01A@mail.gmail.com>
	<1316019038624-6793424.post@n2.nabble.com>
	<CAKzWOxULsj-wzGKzF0X1qj2iQoeZ5VRo__mvj8F8QiSGk6bcoA@mail.gmail.com>
	<1316022984994-6793773.post@n2.nabble.com>
	<CAKzWOxV917_bWqPO9_kGbV5Tmu2zu-J0wV4CEWULMsgrrwOyqQ@mail.gmail.com>
	<1316025984046-6794016.post@n2.nabble.com>
	<CAKzWOxXMk_Za-xkM_H-WZQparVyRCk+G8H7K4pO+0obqaah6ZQ@mail.gmail.com>
Message-ID: <CAKzWOxVJ1_dfJ3OsgVZnsM00nmJ=cN0Ps6WH6t=zSOZLEpSR_A@mail.gmail.com>

Okay, I couldn't stand it! I added *72 and *73 to my example:

http://wiki.freeswitch.org/wiki/Simple_CF_with_IVR

If you follow the instructions on my page you'll have magic *72/*73 for
setting and canceling. Also, feel free to omit the 909x extensions that I
show unless you like the idea of being able to set and cancel from a remote
location... :)

-MC

On Wed, Sep 14, 2011 at 3:50 PM, Michael Collins <msc at freeswitch.org> wrote:

> Apologies, I didn't look down far enough.
>
> Okay, what's happening is that the call from 1234 to 1210 is never hitting
> a dialplan extension that checks the value of the CF information that you
> entered with the *72 process. My guess is that you didn't do the part
> mentioned on this page:
>
> http://wiki.freeswitch.org/wiki/Call_Forward_Example
>
> ...under "Also, default.xml:"
>
> So, the questions are:
> Did you add the stuff to features.xml that the author shows?
> Did you add the "call_forwarding_number" extension to default.xml?
> If so, did you put it near the top of the file? (Don't forget that the
> dialplan is parsed in the order listed in the file, so if you put that
> "call_forwarding_number" too low in the file it won't be processed. I
> recommend putting as the first extension in the file, just to be safe.)
> Did you add the "is_forward" check inside the Local_Extension? (By the look
> of your log I'd say not...)
>
> If you want an alternative way of doing this then check out the
> call-forwarding example that I posted on the wiki. There's a link to "Simple
> CF with IVR" at the bottom of the wiki page that you are looking at. Perhaps
> the information I present will give you some more clarity on how these kinds
> of things work. I give a lot of explanatory text in that example, so check
> it out.
>
> Don't be discouraged - if you're new to all of this then the pains you're
> experiencing are part of the learning process. My advice is to keep hacking
> away at it. Also, buy our book! Darren and I wrote some good stuff in
> chapters 8 and 5 (respectively) on the dialplan and how it works. The book
> will help you build a good foundation. :)
>
> -MC
>
> On Wed, Sep 14, 2011 at 11:46 AM, Aakash <aakashviswam at gmail.com> wrote:
>
>> Please have a look --  http://pastebin.freeswitch.org/17333
>>
>> You can see the indications like
>> 2011-09-14 19:01:16.454118 [INFO] mod_dialplan_xml.c:336 Processing Aakash
>> <1234>->1210 in context default
>>
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/call-forwarding-how-to-test-tp6788875p6794016.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From djbinter at gmail.com  Thu Sep 15 03:31:27 2011
From: djbinter at gmail.com (DJB International)
Date: Wed, 14 Sep 2011 16:31:27 -0700
Subject: [Freeswitch-users] call forwarding how to test
In-Reply-To: <CAKzWOxVJ1_dfJ3OsgVZnsM00nmJ=cN0Ps6WH6t=zSOZLEpSR_A@mail.gmail.com>
References: <1315947845640-6789287.post@n2.nabble.com>
	<CAHKqMb7YqYnMz8pv=Up+7OEoiPqLDDwKnYSHYmx5KsfZQYM3dw@mail.gmail.com>
	<1316008496762-6792618.post@n2.nabble.com>
	<CAHKqMb7Dh4YqMzczijY0tOWi1kWy12HC0sDz3CFa1DZkOFz6zw@mail.gmail.com>
	<1316013823287-6793069.post@n2.nabble.com>
	<CAHKqMb6m09OnaXhJxiEg_mMpnzOrOWSFjeY5wyF952uRMkm01A@mail.gmail.com>
	<1316019038624-6793424.post@n2.nabble.com>
	<CAKzWOxULsj-wzGKzF0X1qj2iQoeZ5VRo__mvj8F8QiSGk6bcoA@mail.gmail.com>
	<1316022984994-6793773.post@n2.nabble.com>
	<CAKzWOxV917_bWqPO9_kGbV5Tmu2zu-J0wV4CEWULMsgrrwOyqQ@mail.gmail.com>
	<1316025984046-6794016.post@n2.nabble.com>
	<CAKzWOxXMk_Za-xkM_H-WZQparVyRCk+G8H7K4pO+0obqaah6ZQ@mail.gmail.com>
	<CAKzWOxVJ1_dfJ3OsgVZnsM00nmJ=cN0Ps6WH6t=zSOZLEpSR_A@mail.gmail.com>
Message-ID: <CAAB3jWsnySAe_ZKdDaxPZqkzuWHrxfHcV_HbJcvrPOqvOB+x0Q@mail.gmail.com>

Aakash,

You can also try this:

     Add this in your dialplan for Local_Extension extension:
        <action application="execute_extension" data="is_forward XML
features"/>
      before this line:
       <action application="bridge" data="user/${dialed_extension}@
${domain_name}"/>
 And, make sure you have "is_forward" in features.xml so that when the
system has it when you call.

-djbinter

On Wed, Sep 14, 2011 at 4:25 PM, Michael Collins <msc at freeswitch.org> wrote:

> Okay, I couldn't stand it! I added *72 and *73 to my example:
>
> http://wiki.freeswitch.org/wiki/Simple_CF_with_IVR
>
> If you follow the instructions on my page you'll have magic *72/*73 for
> setting and canceling. Also, feel free to omit the 909x extensions that I
> show unless you like the idea of being able to set and cancel from a remote
> location... :)
>
> -MC
>
>
> On Wed, Sep 14, 2011 at 3:50 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Apologies, I didn't look down far enough.
>>
>> Okay, what's happening is that the call from 1234 to 1210 is never hitting
>> a dialplan extension that checks the value of the CF information that you
>> entered with the *72 process. My guess is that you didn't do the part
>> mentioned on this page:
>>
>> http://wiki.freeswitch.org/wiki/Call_Forward_Example
>>
>> ...under "Also, default.xml:"
>>
>> So, the questions are:
>> Did you add the stuff to features.xml that the author shows?
>> Did you add the "call_forwarding_number" extension to default.xml?
>> If so, did you put it near the top of the file? (Don't forget that the
>> dialplan is parsed in the order listed in the file, so if you put that
>> "call_forwarding_number" too low in the file it won't be processed. I
>> recommend putting as the first extension in the file, just to be safe.)
>> Did you add the "is_forward" check inside the Local_Extension? (By the
>> look of your log I'd say not...)
>>
>> If you want an alternative way of doing this then check out the
>> call-forwarding example that I posted on the wiki. There's a link to "Simple
>> CF with IVR" at the bottom of the wiki page that you are looking at. Perhaps
>> the information I present will give you some more clarity on how these kinds
>> of things work. I give a lot of explanatory text in that example, so check
>> it out.
>>
>> Don't be discouraged - if you're new to all of this then the pains you're
>> experiencing are part of the learning process. My advice is to keep hacking
>> away at it. Also, buy our book! Darren and I wrote some good stuff in
>> chapters 8 and 5 (respectively) on the dialplan and how it works. The book
>> will help you build a good foundation. :)
>>
>> -MC
>>
>> On Wed, Sep 14, 2011 at 11:46 AM, Aakash <aakashviswam at gmail.com> wrote:
>>
>>> Please have a look --  http://pastebin.freeswitch.org/17333
>>>
>>> You can see the indications like
>>> 2011-09-14 19:01:16.454118 [INFO] mod_dialplan_xml.c:336 Processing
>>> Aakash
>>> <1234>->1210 in context default
>>>
>>>
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/call-forwarding-how-to-test-tp6788875p6794016.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ljjimenez at gmail.com  Thu Sep 15 07:39:07 2011
From: ljjimenez at gmail.com (ljjimenez at gmail.com)
Date: Thu, 15 Sep 2011 03:39:07 +0000
Subject: [Freeswitch-users] large block of DID's
In-Reply-To: <D25CC3E5-C0D2-41D7-87C3-843C24A0CA8C@lyonl.com>
References: <D25CC3E5-C0D2-41D7-87C3-843C24A0CA8C@lyonl.com>
Message-ID: <801470822-1316057954-cardhu_decombobulator_blackberry.rim.net-146923873-@b5.c27.bise6.blackberry>

We do this with about 10k DIDs, we use MySQL and a Lua script via ODBC

Luis Jimenez

-----Original Message-----
From: Chad Vogel <cvogel at lyonl.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Wed, 14 Sep 2011 15:44:23 
To: FreeSWITCH-users at lists.freeswitch.org<FreeSWITCH-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] large block of DID's

I was wondering if anyone has any thoughts what is the best way to manage a large blocks of DID? (we have about 1600 DID's that we are moving off of our asterisk servers) 

Chad


FreeSWITCH-users mailing list
FreeSWITCH-users at 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 curriegrad2004 at gmail.com  Thu Sep 15 10:20:15 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 14 Sep 2011 23:20:15 -0700
Subject: [Freeswitch-users] New Chinese sound files for review
In-Reply-To: <CAKzWOxVnHsP8Q8c-ivTCxAFVV2y8K7bGh0MsCL9EgjNn0Fdbag@mail.gmail.com>
References: <CAKzWOxVnHsP8Q8c-ivTCxAFVV2y8K7bGh0MsCL9EgjNn0Fdbag@mail.gmail.com>
Message-ID: <CAMWiMy-VoB2UXBpXWjZrAj90sg9B=8oueBpYUnTf2cMnQNDGKA@mail.gmail.com>

I'd say the mandarin sounds can be placed on the mainline git repository.

However, dollar.wav and dollars.wav can be symlinked together as the
Mandarin language doesn't have the concept of plural tenses. The same
goes for cent.wav and cents.wav.

On Wed, Sep 14, 2011 at 9:33 AM, Michael Collins <msc at freeswitch.org> wrote:
> To all Chinese speakers:
> Seven Du has graciously put together new zh_CN sound files:
> ftp://ftp.x-y-t.com/pub/freeswitch-sounds-zh-cn-link-8000-1.0.0.tar.gz
> ftp://ftp.x-y-t.com/pub/freeswitch-sounds-zh-cn-link-8000-1.0.0.tar.gz.md5
> Here's a jira for the phrase xml:
> http://jira.freeswitch.org/browse/FS-3568
> If you understand Chinese then please consider helping Seven Du out by
> downloading and listening to the sounds.
> Thanks!
> -Michael
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 chrisg.lists at gmail.com  Thu Sep 15 13:14:41 2011
From: chrisg.lists at gmail.com (Chris Graham)
Date: Thu, 15 Sep 2011 11:14:41 +0200
Subject: [Freeswitch-users] Dialplan delimiter transform
Message-ID: <CAMKRqpjhPgxpgPomNn=tadpH6NSDmZJ3=TsyfwJFj3eoB9-+bg@mail.gmail.com>

Hi All,

I have a simple issue I seem to not be making progress on. Due to topology
hiding in one of the headquaters I send a invite as such:

INVITE sip:0114712500-10.3.12.153 at x.x.x.x SIP/2.0.

You can see the first part of the invite is the dialstring then a "-" used
as a delimiter to bill against the hidden IP. FS duly passed this dialstring
on. The carrier then rejects the call.

I want to modify the dialstring in the context below not to dial
0114712500-10.3.12.153 but rather just 0114712500. I hope I am clear, and
thanks in advance.

<document type="freeswitch/xml">
  <section name="dialplan" description="RE FS DP">
    <context name="lcr_trunks">
      <extension name="Catch all send to carriers">
        <condition field="destination_number" expression="^(.*)$">
          <action application="set" data="nibble_account=10.3.12.153"/>
          <action application="lcr" data="$1 peak"/>
          <action application="bridge" data="${lcr_auto_route}"/>
        </condition>
      </extension>
    </context>
  </section>
</document>

Chris G
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From steveayre at gmail.com  Thu Sep 15 14:19:03 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 15 Sep 2011 11:19:03 +0100
Subject: [Freeswitch-users] Dialplan delimiter transform
In-Reply-To: <CAMKRqpjhPgxpgPomNn=tadpH6NSDmZJ3=TsyfwJFj3eoB9-+bg@mail.gmail.com>
References: <CAMKRqpjhPgxpgPomNn=tadpH6NSDmZJ3=TsyfwJFj3eoB9-+bg@mail.gmail.com>
Message-ID: <CAFiqYu=5XeFoN-oq_O7T+Mc=5nzXy59-rFDvFb3v5nkLeVChMQ@mail.gmail.com>

Change the regex to only pull out the first part into $1. Something like
this:

<condition field="destination_number"
expression="^(\d+)\-(\d+\.\d+\.\d+\.\d+)$">

$1 will be 0114712500
$2 will be 10.3.12.153

-Steve


On 15 September 2011 10:14, Chris Graham <chrisg.lists at gmail.com> wrote:

> Hi All,
>
> I have a simple issue I seem to not be making progress on. Due to topology
> hiding in one of the headquaters I send a invite as such:
>
> INVITE sip:0114712500-10.3.12.153 at x.x.x.x SIP/2.0.
>
> You can see the first part of the invite is the dialstring then a "-" used
> as a delimiter to bill against the hidden IP. FS duly passed this dialstring
> on. The carrier then rejects the call.
>
> I want to modify the dialstring in the context below not to dial
> 0114712500-10.3.12.153 but rather just 0114712500. I hope I am clear, and
> thanks in advance.
>
> <document type="freeswitch/xml">
>   <section name="dialplan" description="RE FS DP">
>     <context name="lcr_trunks">
>       <extension name="Catch all send to carriers">
>         <condition field="destination_number" expression="^(.*)$">
>           <action application="set" data="nibble_account=10.3.12.153"/>
>           <action application="lcr" data="$1 peak"/>
>           <action application="bridge" data="${lcr_auto_route}"/>
>         </condition>
>       </extension>
>     </context>
>   </section>
> </document>
>
> Chris G
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveu at coppice.org  Thu Sep 15 15:53:11 2011
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 15 Sep 2011 19:53:11 +0800
Subject: [Freeswitch-users] New Chinese sound files for review
In-Reply-To: <CAMWiMy-VoB2UXBpXWjZrAj90sg9B=8oueBpYUnTf2cMnQNDGKA@mail.gmail.com>
References: <CAKzWOxVnHsP8Q8c-ivTCxAFVV2y8K7bGh0MsCL9EgjNn0Fdbag@mail.gmail.com>
	<CAMWiMy-VoB2UXBpXWjZrAj90sg9B=8oueBpYUnTf2cMnQNDGKA@mail.gmail.com>
Message-ID: <4E71E727.40302@coppice.org>

It looks like these prompts were produced without looking at what the 
Chinese say module currently uses.

On 09/15/2011 02:20 PM, curriegrad2004 wrote:
> I'd say the mandarin sounds can be placed on the mainline git repository.
>
> However, dollar.wav and dollars.wav can be symlinked together as the
> Mandarin language doesn't have the concept of plural tenses. The same
> goes for cent.wav and cents.wav.
They shouldn't be symlinked. They should be deleted. The Chinese say 
module doesn't use them. Also, the say module use the two Chinese senses 
of "2". There should be a file called time/2s.wav with ? in it.
> On Wed, Sep 14, 2011 at 9:33 AM, Michael Collins<msc at freeswitch.org>  wrote:
>> To all Chinese speakers:
>> Seven Du has graciously put together new zh_CN sound files:
>> ftp://ftp.x-y-t.com/pub/freeswitch-sounds-zh-cn-link-8000-1.0.0.tar.gz
>> ftp://ftp.x-y-t.com/pub/freeswitch-sounds-zh-cn-link-8000-1.0.0.tar.gz.md5
>> Here's a jira for the phrase xml:
>> http://jira.freeswitch.org/browse/FS-3568
>> If you understand Chinese then please consider helping Seven Du out by
>> downloading and listening to the sounds.
>> Thanks!
>> -Michael
Steve



From gavin.henry at gmail.com  Thu Sep 15 16:07:25 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Thu, 15 Sep 2011 13:07:25 +0100
Subject: [Freeswitch-users] One Way Audio - Auto Change RTP port?
In-Reply-To: <CAA8_NKC1Fgtw9vBEKVffYG9NHO9toOMEkcA0qto03Ttn1UJ7JA@mail.gmail.com>
References: <CAAuJ8MBnFKYzEjPnCNeNmiU-T=jRu3M1vcrR49-p2LPyRU=+uQ@mail.gmail.com>
	<CAKbxfG8vU1pe-TU5_PHT8BcabYPceLem9QiyysCDxnf+gKSBYQ@mail.gmail.com>
	<CAAuJ8MDdz-EoaLRzR2es+aH0tURL9kN0K+6bQcKEpP_RJ0BEGw@mail.gmail.com>
	<CAA8_NKC1Fgtw9vBEKVffYG9NHO9toOMEkcA0qto03Ttn1UJ7JA@mail.gmail.com>
Message-ID: <CAA8_NKB9=MSVYfaaQKyHzBddJhqL2LpvdC1A55Y0Qc6xuzaDfw@mail.gmail.com>

We've found firmware version 3.3.3 fixed all issues on the Draytek 2820 routers.

On 12 September 2011 21:36, Gavin Henry <gavin.henry at gmail.com> wrote:
> Hi Dan,
>
> We're currently debugging this also with Snom 3XX/8XX on Draytek 2820 routers:
>
> http://jira.freeswitch.org/browse/FS-3552
>
> On 9 September 2011 00:17, Dan Lan <danlanweb at gmail.com> wrote:
>> 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
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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.suretecsystems.com/services/openldap/
> http://www.surevoip.co.uk
>



-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk


From gavin.henry at gmail.com  Thu Sep 15 16:09:17 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Thu, 15 Sep 2011 13:09:17 +0100
Subject: [Freeswitch-users] DrayTek 2820n routers and FreeSWITCH
Message-ID: <CAA8_NKDK6-pDxsC8ciwbamkcXXn1yv3JDJDAriC1YgmXr8tWcA@mail.gmail.com>

Just a quick note to all to say if you have any one-way audio issues
or SIP stability issues with these routers and FreeSWITCH then
downgrade to firmware 3.3.3 and all issues disappear.

This is what DrayTek recommend.

Thanks,

Gavin.

-- 
http://www.suretecgroup.com
http://www.surevoip.co.uk


From eagle.antonio at gmail.com  Thu Sep 15 16:22:38 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Thu, 15 Sep 2011 12:22:38 +0000
Subject: [Freeswitch-users] DTMF Info
In-Reply-To: <CAF+3j-tA-8=YAMtKAW8abNYWXhUPU2tx7ZWGvFHG2CoipL-E_w@mail.gmail.com>
References: <CAF+3j-syTYQi2uCz4exSXKmqs++HhoHkv0UA=V-Yu_RPC4On2Q@mail.gmail.com>
	<CALNZuwnzTW0itYkOSPjuN7QHb1tRJi6o2sqqaXBvqYdzg5Ow1Q@mail.gmail.com>
	<CAF+3j-tA-8=YAMtKAW8abNYWXhUPU2tx7ZWGvFHG2CoipL-E_w@mail.gmail.com>
Message-ID: <CAF+3j-u8y1943hbksTLTC6OgQwpMwveXgOUZ=RKWRkrnMD2UHQ@mail.gmail.com>

So is there any other way to capture DTMF-type INFO :\ without an upgrade ?

Regards
A/T

2011/9/14 Antonio Teixeira <eagle.antonio at gmail.com>

> Hello Avi , thanks for your answer.
> I tried setting dtmf_type at the sip profile and user profile still not
> luck , liberal-dtmf is not available on my version.
>
> Hope someone can help me , since upgrading will be more complicated
> Regards
> A/T
>
>
> 2011/9/14 Avi Marcus <avi at avimarcus.net>
>
>> If you set it in the profile, once you reload the profile, it should work.
>> Or, in the sip profile you can set liberal-dtmf<http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#liberal-dtmf> and
>> then it will accept rfc2833 or INFO.
>>
>> -Avi
>>
>>
>> On Wed, Sep 14, 2011 at 1:18 PM, Antonio Teixeira <
>> eagle.antonio at gmail.com> wrote:
>>
>>> Good Morning.
>>>
>>> I have a soft phone that uses SIP INFO to deliver the DTMF keys to FS.
>>> Now the problem is that FS always reported INFO is not expected on this
>>> channels so off to the wiki i go and i set in the profile dtmf-type = info.
>>> sofia.c:6439 IGNORE INFO DTMF(2) (This channel was not configured to use
>>> INFO DTMF!)
>>>
>>> So anyone can help me out with this ?
>>>
>>> Best Regards
>>> Antonio Teixeira
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From avi at avimarcus.net  Thu Sep 15 17:13:15 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 15 Sep 2011 16:13:15 +0300
Subject: [Freeswitch-users] DTMF Info
In-Reply-To: <CAF+3j-u8y1943hbksTLTC6OgQwpMwveXgOUZ=RKWRkrnMD2UHQ@mail.gmail.com>
References: <CAF+3j-syTYQi2uCz4exSXKmqs++HhoHkv0UA=V-Yu_RPC4On2Q@mail.gmail.com>
	<CALNZuwnzTW0itYkOSPjuN7QHb1tRJi6o2sqqaXBvqYdzg5Ow1Q@mail.gmail.com>
	<CAF+3j-tA-8=YAMtKAW8abNYWXhUPU2tx7ZWGvFHG2CoipL-E_w@mail.gmail.com>
	<CAF+3j-u8y1943hbksTLTC6OgQwpMwveXgOUZ=RKWRkrnMD2UHQ@mail.gmail.com>
Message-ID: <CALNZuwmM=Q=UKyBsBLJLdGhdyxu8ZWTBDV-=KVdZSHJznocZ+A@mail.gmail.com>

a) change the softphone to send rfc2833
b) set the variable in the profile or in the gateway - it's too late to set
it in the channel
http://wiki.freeswitch.org/wiki/Sofia.conf.xml#DTMF
in the profile - internal? - <param name="dtmf-type" value="info"/> and
reload the profile

Or in the user/gateway profile:


<variables>
  <variable direction="inbound|outbound|both" name="dtmf_type" value="info">
</variables>

and reloadxml.. kill/start the gateway and probably need to wait for the
user to re-reg.
-Avi



On Thu, Sep 15, 2011 at 3:22 PM, Antonio Teixeira
<eagle.antonio at gmail.com>wrote:

> So is there any other way to capture DTMF-type INFO :\ without an upgrade ?
>
> Regards
> A/T
>
>
> 2011/9/14 Antonio Teixeira <eagle.antonio at gmail.com>
>
>> Hello Avi , thanks for your answer.
>> I tried setting dtmf_type at the sip profile and user profile still not
>> luck , liberal-dtmf is not available on my version.
>>
>> Hope someone can help me , since upgrading will be more complicated
>> Regards
>> A/T
>>
>>
>> 2011/9/14 Avi Marcus <avi at avimarcus.net>
>>
>>> If you set it in the profile, once you reload the profile, it should
>>> work.
>>> Or, in the sip profile you can set liberal-dtmf<http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#liberal-dtmf> and
>>> then it will accept rfc2833 or INFO.
>>>
>>> -Avi
>>>
>>>
>>> On Wed, Sep 14, 2011 at 1:18 PM, Antonio Teixeira <
>>> eagle.antonio at gmail.com> wrote:
>>>
>>>> Good Morning.
>>>>
>>>> I have a soft phone that uses SIP INFO to deliver the DTMF keys to FS.
>>>> Now the problem is that FS always reported INFO is not expected on this
>>>> channels so off to the wiki i go and i set in the profile dtmf-type = info.
>>>> sofia.c:6439 IGNORE INFO DTMF(2) (This channel was not configured to use
>>>> INFO DTMF!)
>>>>
>>>> So anyone can help me out with this ?
>>>>
>>>> Best Regards
>>>> Antonio Teixeira
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From michel.daggelinckx at gmail.com  Thu Sep 15 18:37:02 2011
From: michel.daggelinckx at gmail.com (Michel Daggelinckx)
Date: Thu, 15 Sep 2011 16:37:02 +0200
Subject: [Freeswitch-users] Lua API for CouchDB
In-Reply-To: <C48166F7-EB91-4466-BEDC-B79A42073D1F@apartmentlines.com>
References: <C48166F7-EB91-4466-BEDC-B79A42073D1F@apartmentlines.com>
Message-ID: <CAFYZzbuY9w1BZM4rXMD9OZ1UEXPS_L9rS+LBMVta6OexUxbXRw@mail.gmail.com>

2600hz.org has done allot of work on freeswitch and couchdb in their
whistle project.
might be of interest to take a look into it


On 9/13/11, Chad Phillips -- Apartment Lines <chad at apartmentlines.com> wrote:
> For those interested in connecting to CouchDB (http://couchdb.apache.org)
> from FreeSWITCH, I've written a Lua API (http://www.lua.org) for it,
> available here:
>
> https://github.com/thehunmonkgroup/luchia
>
> With Lua being the recommended scripting language for FreeSWITCH, this seems
> an excellent way to work with Couch.
>
> For those that have the LuaRocks package manager installed, it's a simple
> 'luarocks install luchia' to install.
>
> I'm interested in exploring CouchDB as the datastore for a distributed,
> fault tolerant voicemail system -- I think it's attachment feature and
> bi-directional synchronization are well suited for the task.
>
> Chad
>
>


From eagle.antonio at gmail.com  Thu Sep 15 18:57:37 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Thu, 15 Sep 2011 14:57:37 +0000
Subject: [Freeswitch-users] DTMF Info
In-Reply-To: <CALNZuwmM=Q=UKyBsBLJLdGhdyxu8ZWTBDV-=KVdZSHJznocZ+A@mail.gmail.com>
References: <CAF+3j-syTYQi2uCz4exSXKmqs++HhoHkv0UA=V-Yu_RPC4On2Q@mail.gmail.com>
	<CALNZuwnzTW0itYkOSPjuN7QHb1tRJi6o2sqqaXBvqYdzg5Ow1Q@mail.gmail.com>
	<CAF+3j-tA-8=YAMtKAW8abNYWXhUPU2tx7ZWGvFHG2CoipL-E_w@mail.gmail.com>
	<CAF+3j-u8y1943hbksTLTC6OgQwpMwveXgOUZ=RKWRkrnMD2UHQ@mail.gmail.com>
	<CALNZuwmM=Q=UKyBsBLJLdGhdyxu8ZWTBDV-=KVdZSHJznocZ+A@mail.gmail.com>
Message-ID: <CAF+3j-vUwTTuBSinaFpvEhXBTGWmYYOOKeDJ6cFoyG+GpfasHQ@mail.gmail.com>

Hello Avi.

I tried all those steps , unfortunately the softphone is a legacy system
that is incapable of rfc2833 so option 1 is out.
I tried B and C ensuring by wireshark to verify that the data is actually in
transit and FS still raises and error :(.

Regards
A/T

2011/9/15 Avi Marcus <avi at avimarcus.net>

> a) change the softphone to send rfc2833
> b) set the variable in the profile or in the gateway - it's too late to set
> it in the channel
> http://wiki.freeswitch.org/wiki/Sofia.conf.xml#DTMF
> in the profile - internal? - <param name="dtmf-type" value="info"/> and
> reload the profile
>
> Or in the user/gateway profile:
>
> <variables>
>   <variable direction="inbound|outbound|both" name="dtmf_type" value="info">
> </variables>
>
> and reloadxml.. kill/start the gateway and probably need to wait for the
> user to re-reg.
> -Avi
>
>
>
> On Thu, Sep 15, 2011 at 3:22 PM, Antonio Teixeira <eagle.antonio at gmail.com
> > wrote:
>
>> So is there any other way to capture DTMF-type INFO :\ without an upgrade
>> ?
>>
>> Regards
>> A/T
>>
>>
>> 2011/9/14 Antonio Teixeira <eagle.antonio at gmail.com>
>>
>>> Hello Avi , thanks for your answer.
>>> I tried setting dtmf_type at the sip profile and user profile still not
>>> luck , liberal-dtmf is not available on my version.
>>>
>>> Hope someone can help me , since upgrading will be more complicated
>>> Regards
>>> A/T
>>>
>>>
>>> 2011/9/14 Avi Marcus <avi at avimarcus.net>
>>>
>>>> If you set it in the profile, once you reload the profile, it should
>>>> work.
>>>> Or, in the sip profile you can set liberal-dtmf<http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#liberal-dtmf> and
>>>> then it will accept rfc2833 or INFO.
>>>>
>>>> -Avi
>>>>
>>>>
>>>> On Wed, Sep 14, 2011 at 1:18 PM, Antonio Teixeira <
>>>> eagle.antonio at gmail.com> wrote:
>>>>
>>>>> Good Morning.
>>>>>
>>>>> I have a soft phone that uses SIP INFO to deliver the DTMF keys to FS.
>>>>> Now the problem is that FS always reported INFO is not expected on this
>>>>> channels so off to the wiki i go and i set in the profile dtmf-type = info.
>>>>> sofia.c:6439 IGNORE INFO DTMF(2) (This channel was not configured to
>>>>> use INFO DTMF!)
>>>>>
>>>>> So anyone can help me out with this ?
>>>>>
>>>>> Best Regards
>>>>> Antonio Teixeira
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Thu Sep 15 19:10:37 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 15 Sep 2011 18:10:37 +0300
Subject: [Freeswitch-users] DTMF Info
In-Reply-To: <CAF+3j-vUwTTuBSinaFpvEhXBTGWmYYOOKeDJ6cFoyG+GpfasHQ@mail.gmail.com>
References: <CAF+3j-syTYQi2uCz4exSXKmqs++HhoHkv0UA=V-Yu_RPC4On2Q@mail.gmail.com>
	<CALNZuwnzTW0itYkOSPjuN7QHb1tRJi6o2sqqaXBvqYdzg5Ow1Q@mail.gmail.com>
	<CAF+3j-tA-8=YAMtKAW8abNYWXhUPU2tx7ZWGvFHG2CoipL-E_w@mail.gmail.com>
	<CAF+3j-u8y1943hbksTLTC6OgQwpMwveXgOUZ=RKWRkrnMD2UHQ@mail.gmail.com>
	<CALNZuwmM=Q=UKyBsBLJLdGhdyxu8ZWTBDV-=KVdZSHJznocZ+A@mail.gmail.com>
	<CAF+3j-vUwTTuBSinaFpvEhXBTGWmYYOOKeDJ6cFoyG+GpfasHQ@mail.gmail.com>
Message-ID: <CALNZuwkd=j1AZv4FFhLaYXmF-Zwu_D=CdP3E-+9auGPW2S053g@mail.gmail.com>

That doesn't sound possible. Are you sure you reloaded the profile?
Regardless, I suppose it's time to upgrade!

-Avi


On Thu, Sep 15, 2011 at 5:57 PM, Antonio Teixeira
<eagle.antonio at gmail.com>wrote:

> Hello Avi.
>
> I tried all those steps , unfortunately the softphone is a legacy system
> that is incapable of rfc2833 so option 1 is out.
> I tried B and C ensuring by wireshark to verify that the data is actually
> in transit and FS still raises and error :(.
>
> Regards
> A/T
>
>
> 2011/9/15 Avi Marcus <avi at avimarcus.net>
>
>> a) change the softphone to send rfc2833
>> b) set the variable in the profile or in the gateway - it's too late to
>> set it in the channel
>> http://wiki.freeswitch.org/wiki/Sofia.conf.xml#DTMF
>> in the profile - internal? - <param name="dtmf-type" value="info"/> and
>> reload the profile
>>
>> Or in the user/gateway profile:
>>
>>
>> <variables>
>>   <variable direction="inbound|outbound|both" name="dtmf_type" value="info">
>> </variables>
>>
>> and reloadxml.. kill/start the gateway and probably need to wait for the
>> user to re-reg.
>> -Avi
>>
>>
>>
>> On Thu, Sep 15, 2011 at 3:22 PM, Antonio Teixeira <
>> eagle.antonio at gmail.com> wrote:
>>
>>> So is there any other way to capture DTMF-type INFO :\ without an upgrade
>>> ?
>>>
>>> Regards
>>> A/T
>>>
>>>
>>> 2011/9/14 Antonio Teixeira <eagle.antonio at gmail.com>
>>>
>>>> Hello Avi , thanks for your answer.
>>>> I tried setting dtmf_type at the sip profile and user profile still not
>>>> luck , liberal-dtmf is not available on my version.
>>>>
>>>> Hope someone can help me , since upgrading will be more complicated
>>>> Regards
>>>> A/T
>>>>
>>>>
>>>> 2011/9/14 Avi Marcus <avi at avimarcus.net>
>>>>
>>>>> If you set it in the profile, once you reload the profile, it should
>>>>> work.
>>>>> Or, in the sip profile you can set liberal-dtmf<http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#liberal-dtmf> and
>>>>> then it will accept rfc2833 or INFO.
>>>>>
>>>>> -Avi
>>>>>
>>>>>
>>>>> On Wed, Sep 14, 2011 at 1:18 PM, Antonio Teixeira <
>>>>> eagle.antonio at gmail.com> wrote:
>>>>>
>>>>>> Good Morning.
>>>>>>
>>>>>> I have a soft phone that uses SIP INFO to deliver the DTMF keys to FS.
>>>>>> Now the problem is that FS always reported INFO is not expected on
>>>>>> this channels so off to the wiki i go and i set in the profile dtmf-type =
>>>>>> info.
>>>>>> sofia.c:6439 IGNORE INFO DTMF(2) (This channel was not configured to
>>>>>> use INFO DTMF!)
>>>>>>
>>>>>> So anyone can help me out with this ?
>>>>>>
>>>>>> Best Regards
>>>>>> Antonio Teixeira
>>>>>>
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From eagle.antonio at gmail.com  Thu Sep 15 19:22:57 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Thu, 15 Sep 2011 15:22:57 +0000
Subject: [Freeswitch-users] DTMF Info
In-Reply-To: <CALNZuwkd=j1AZv4FFhLaYXmF-Zwu_D=CdP3E-+9auGPW2S053g@mail.gmail.com>
References: <CAF+3j-syTYQi2uCz4exSXKmqs++HhoHkv0UA=V-Yu_RPC4On2Q@mail.gmail.com>
	<CALNZuwnzTW0itYkOSPjuN7QHb1tRJi6o2sqqaXBvqYdzg5Ow1Q@mail.gmail.com>
	<CAF+3j-tA-8=YAMtKAW8abNYWXhUPU2tx7ZWGvFHG2CoipL-E_w@mail.gmail.com>
	<CAF+3j-u8y1943hbksTLTC6OgQwpMwveXgOUZ=RKWRkrnMD2UHQ@mail.gmail.com>
	<CALNZuwmM=Q=UKyBsBLJLdGhdyxu8ZWTBDV-=KVdZSHJznocZ+A@mail.gmail.com>
	<CAF+3j-vUwTTuBSinaFpvEhXBTGWmYYOOKeDJ6cFoyG+GpfasHQ@mail.gmail.com>
	<CALNZuwkd=j1AZv4FFhLaYXmF-Zwu_D=CdP3E-+9auGPW2S053g@mail.gmail.com>
Message-ID: <CAF+3j-uRvgE+3=Oy+-e3ReLJtbNKSS71JS2+w-PTpH10m9rsqA@mail.gmail.com>

Both phones are in the internal profile.
So i did sofia profile internal rescan reloadxml

http://pastebin.freeswitch.org/17338
for more info

Rgds
A/T


2011/9/15 Avi Marcus <avi at avimarcus.net>

> That doesn't sound possible. Are you sure you reloaded the profile?
> Regardless, I suppose it's time to upgrade!
>
> -Avi
>
>
> On Thu, Sep 15, 2011 at 5:57 PM, Antonio Teixeira <eagle.antonio at gmail.com
> > wrote:
>
>> Hello Avi.
>>
>> I tried all those steps , unfortunately the softphone is a legacy system
>> that is incapable of rfc2833 so option 1 is out.
>> I tried B and C ensuring by wireshark to verify that the data is actually
>> in transit and FS still raises and error :(.
>>
>> Regards
>> A/T
>>
>>
>> 2011/9/15 Avi Marcus <avi at avimarcus.net>
>>
>>> a) change the softphone to send rfc2833
>>> b) set the variable in the profile or in the gateway - it's too late to
>>> set it in the channel
>>> http://wiki.freeswitch.org/wiki/Sofia.conf.xml#DTMF
>>> in the profile - internal? - <param name="dtmf-type" value="info"/> and
>>> reload the profile
>>>
>>> Or in the user/gateway profile:
>>>
>>> <variables>
>>>   <variable direction="inbound|outbound|both" name="dtmf_type" value="info">
>>> </variables>
>>>
>>> and reloadxml.. kill/start the gateway and probably need to wait for the
>>> user to re-reg.
>>> -Avi
>>>
>>>
>>>
>>> On Thu, Sep 15, 2011 at 3:22 PM, Antonio Teixeira <
>>> eagle.antonio at gmail.com> wrote:
>>>
>>>> So is there any other way to capture DTMF-type INFO :\ without an
>>>> upgrade ?
>>>>
>>>> Regards
>>>> A/T
>>>>
>>>>
>>>> 2011/9/14 Antonio Teixeira <eagle.antonio at gmail.com>
>>>>
>>>>> Hello Avi , thanks for your answer.
>>>>> I tried setting dtmf_type at the sip profile and user profile still not
>>>>> luck , liberal-dtmf is not available on my version.
>>>>>
>>>>> Hope someone can help me , since upgrading will be more complicated
>>>>> Regards
>>>>> A/T
>>>>>
>>>>>
>>>>> 2011/9/14 Avi Marcus <avi at avimarcus.net>
>>>>>
>>>>>> If you set it in the profile, once you reload the profile, it should
>>>>>> work.
>>>>>> Or, in the sip profile you can set liberal-dtmf<http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#liberal-dtmf> and
>>>>>> then it will accept rfc2833 or INFO.
>>>>>>
>>>>>> -Avi
>>>>>>
>>>>>>
>>>>>> On Wed, Sep 14, 2011 at 1:18 PM, Antonio Teixeira <
>>>>>> eagle.antonio at gmail.com> wrote:
>>>>>>
>>>>>>> Good Morning.
>>>>>>>
>>>>>>> I have a soft phone that uses SIP INFO to deliver the DTMF keys to
>>>>>>> FS.
>>>>>>> Now the problem is that FS always reported INFO is not expected on
>>>>>>> this channels so off to the wiki i go and i set in the profile dtmf-type =
>>>>>>> info.
>>>>>>> sofia.c:6439 IGNORE INFO DTMF(2) (This channel was not configured to
>>>>>>> use INFO DTMF!)
>>>>>>>
>>>>>>> So anyone can help me out with this ?
>>>>>>>
>>>>>>> Best Regards
>>>>>>> Antonio Teixeira
>>>>>>>
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nish.chn at gmail.com  Thu Sep 15 16:34:40 2011
From: nish.chn at gmail.com (mg123)
Date: Thu, 15 Sep 2011 05:34:40 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH never hangup at nobal_amount
Message-ID: <1316090079670-6796618.post@n2.nabble.com>

freeswitch is not supporting billing during the call and never hangup at
nobal_amount.

nibblebil_conf.xml configurations

    


During the call fs_cli display below lines for each heartbeat

     2011-09-15 17:30:42.507425 [DEBUG] mod_nibblebill.c:573 Received
request via SESSION_HEARTBEAT!
     2011-09-15 17:30:42.507425 [DEBUG] mod_nibblebill.c:434 Attempting to
bill at $1.00 per minute to account 1005
     2011-09-15 17:31:42.544169 [DEBUG] mod_nibblebill.c:573 Received
request via SESSION_HEARTBEAT!
     2011-09-15 17:31:42.544169 [DEBUG] mod_nibblebill.c:434 Attempting to
bill at $1.00 per minute to account 1005

Please advice.
 
Thanks.


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-never-hangup-at-nobal-amount-tp6796618p6796618.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Thu Sep 15 19:26:17 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 15 Sep 2011 10:26:17 -0500
Subject: [Freeswitch-users] mod_managed switch_xml_open_cfg and SIGSEGV
In-Reply-To: <CAAtQP54ec9oTygRGeNdZOvXK7t-U0_79WM_hV9Bc791XDNp5Tg@mail.gmail.com>
References: <CAAtQP54ec9oTygRGeNdZOvXK7t-U0_79WM_hV9Bc791XDNp5Tg@mail.gmail.com>
Message-ID: <CAKbxfG-vq2XPF6cVFi2v8XZpvJrr_=GOFYejHPh2nC0BKwWffg@mail.gmail.com>

you have to supply the 2nd arg too.

the returned obj is the whole xml root, the 2nd param is a pointer to
pointer of the section you searched for inside that root.


On Wed, Sep 14, 2011 at 3:41 PM, R H <buscom123+fs at gmail.com> wrote:
> Has anyone had success working with xml configuration files using
> mod_managed? I have several modules that DO NOT override the xml
> configuration but, rather, I actually want to configure them using the
> freeswitch conf directory structure.?I have not even tried to?parse the xml
> because I cant even load the xml.?Here is a very basic example of what I
> tried and the result I got:
> The Code:
> ------------------------------------------------
> using System;
> using FreeSWITCH;
> using FreeSWITCH.Native;
> namespace PsiCallCenterAddon
> {
> public class FreeSwitchConfigLoader
> {
> public static void loadConfigTest()
> {
> try
> {
> switch_xml xml = freeswitch.switch_xml_open_cfg("freeswitch.conf", null,
> null);
> freeswitch.switch_xml_free(xml);
> Log.WriteLine(LogLevel.Alert, "Config Loader Success!");
> }
> catch
> {
> Log.WriteLine(LogLevel.Alert, "Config Loader Fail!");
> }
> }
> }
> }
> ------------------------------------------------
> The Result:
> ------------------------------------------------
> Stacktrace:
> ? at (wrapper managed-to-native)
> FreeSWITCH.Native.freeswitchPINVOKE.switch_xml_open_cfg
> (string,System.Runtime.InteropServices.HandleRef,System.Runtime.InteropServices.HandleRef)
> <0x000a3>
> ? at (wrapper managed-to-native)
> FreeSWITCH.Native.freeswitchPINVOKE.switch_xml_open_cfg
> (string,System.Runtime.InteropServices.HandleRef,System.Runtime.InteropServices.HandleRef)
> <0x000a3>
> ? at FreeSWITCH.Native.freeswitch.switch_xml_open_cfg
> (string,FreeSWITCH.Native.SWIGTYPE_p_p_switch_xml,FreeSWITCH.Native.switch_event)
> <0x0004f>
> ? at PsiCallCenterAddon.FreeSwitchConfigLoader.loadConfigTest () <0x0001b>
> ? at PsiCallCenterAddon.QueueMonitorLoader.Load () <0x0000b>
> ? at FreeSWITCH.PluginManager.RunLoadNotify (System.Type[]) <0x00190>
> ? at FreeSWITCH.AsmPluginManager.LoadInternal (string) <0x001eb>
> ? at FreeSWITCH.PluginManager.Load (string) <0x000a7>
> ? at (wrapper remoting-invoke-with-check) FreeSWITCH.PluginManager.Load
> (string) <0x00067>
> ? at (wrapper xdomain-dispatch) FreeSWITCH.PluginManager.Load
> (object,byte[]&,byte[]&,string) <0x0017b>
> ? at (wrapper xdomain-invoke) FreeSWITCH.PluginManager.Load (string)
> <0x00126>
> ? at (wrapper remoting-invoke-with-check) FreeSWITCH.PluginManager.Load
> (string) <0x00047>
> ? at FreeSWITCH.Loader.loadFile (string) <0x004a3>
> ? at FreeSWITCH.Loader.Load () <0x0027b>
> ? at (wrapper runtime-invoke) <Module>.runtime_invoke_bool
> (object,intptr,intptr,intptr) <0x00046>
> Native stacktrace:
> ? ? ? ? /usr/lib64/libmono-2.0.so.1(+0xacba0) [0x7fc7ea58dba0]
> ? ? ? ? /usr/lib64/libmono-2.0.so.1(+0xfcedf) [0x7fc7ea5ddedf]
> ? ? ? ? /lib64/libpthread.so.0(+0xf2d0) [0x7fc7f1ceb2d0]
>
> /usr/local/freeswitch/lib/libfreeswitch.so.1(switch_xml_open_cfg+0x1a)
> [0x7fc7f260b0ea]
> ? ? ? ? [0x40155804]
> Debug info from gdb:
> =================================================================
> Got a SIGSEGV while executing native code. This usually indicates
> a fatal error in the mono runtime or one of the native libraries
> used by your application.
> =================================================================
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 15 19:28:57 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 15 Sep 2011 08:28:57 -0700
Subject: [Freeswitch-users] Dialplan delimiter transform
In-Reply-To: <CAFiqYu=5XeFoN-oq_O7T+Mc=5nzXy59-rFDvFb3v5nkLeVChMQ@mail.gmail.com>
References: <CAMKRqpjhPgxpgPomNn=tadpH6NSDmZJ3=TsyfwJFj3eoB9-+bg@mail.gmail.com>
	<CAFiqYu=5XeFoN-oq_O7T+Mc=5nzXy59-rFDvFb3v5nkLeVChMQ@mail.gmail.com>
Message-ID: <CAKzWOxW-rgHgoYZ-2fh0Nb7+TT8ipU86jn10vaucEJth70=PGw@mail.gmail.com>

On Thu, Sep 15, 2011 at 3:19 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Change the regex to only pull out the first part into $1. Something like
> this:
>
> <condition field="destination_number"
> expression="^(\d+)\-(\d+\.\d+\.\d+\.\d+)$">
>
FYI, a bare - in a regex does not need to be escaped.
-MC


>
> $1 will be 0114712500
> $2 will be 10.3.12.153
>
> -Steve
>
>
> On 15 September 2011 10:14, Chris Graham <chrisg.lists at gmail.com> wrote:
>
>> Hi All,
>>
>> I have a simple issue I seem to not be making progress on. Due to topology
>> hiding in one of the headquaters I send a invite as such:
>>
>> INVITE sip:0114712500-10.3.12.153 at x.x.x.x SIP/2.0.
>>
>> You can see the first part of the invite is the dialstring then a "-" used
>> as a delimiter to bill against the hidden IP. FS duly passed this dialstring
>> on. The carrier then rejects the call.
>>
>> I want to modify the dialstring in the context below not to dial
>> 0114712500-10.3.12.153 but rather just 0114712500. I hope I am clear, and
>> thanks in advance.
>>
>> <document type="freeswitch/xml">
>>   <section name="dialplan" description="RE FS DP">
>>     <context name="lcr_trunks">
>>       <extension name="Catch all send to carriers">
>>         <condition field="destination_number" expression="^(.*)$">
>>           <action application="set" data="nibble_account=10.3.12.153"/>
>>           <action application="lcr" data="$1 peak"/>
>>           <action application="bridge" data="${lcr_auto_route}"/>
>>         </condition>
>>       </extension>
>>     </context>
>>   </section>
>> </document>
>>
>> Chris G
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wayne at hamilton.net  Thu Sep 15 20:55:23 2011
From: wayne at hamilton.net (Wayne)
Date: Thu, 15 Sep 2011 11:55:23 -0500
Subject: [Freeswitch-users] Sending DTMF on B-leg
In-Reply-To: <CAKzWOxXJ7J+Pi2iart1nHC8-OmwHkZ0vjH3iXsKvAKGdNOt7=Q@mail.gmail.com>
References: <2934141FC0D9453B9150F7BD307120F3@ccs.local><CALNZuwkMZoRx-HbRBkHg5FSutwV2xW2GWiQQEz2dsmAh2WoPhA@mail.gmail.com><9B18FF40055B495BBAE3EDEEDB7388B0@ccs.local><CAKzWOxXVFFhB25Za9xdmkcTWm-76nFfX7hjn5JFU6gfX+Jxyvg@mail.gmail.com><F9A3AD8DC6C3490CAE3F9365B98D1359@ccs.local><CAKzWOxX8tJbPubs=eFaUmajirOKQBYv9WtBp3HNa2vHPKRWvug@mail.gmail.com><C57A15E034154A9297EE8DF5EF752971@ccs.local>
	<CAKzWOxXJ7J+Pi2iart1nHC8-OmwHkZ0vjH3iXsKvAKGdNOt7=Q@mail.gmail.com>
Message-ID: <B5489AF9DCC040E2AAA414038FFBB657@ccs.local>

Okay,
 
I have tried send_dtmf and queue_dtmf and I can not get any DTMF to pass.
Would anyone be so kind and send me an extension that has it working on the
B-leg. I am getting nowhere.
 
Thanks Wayne 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, September 13, 2011 4:29 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Sending DTMF on B-leg


Okay, fair enough. Not sure what queue_dtmf is not working. You could always
try "execute_on_answer" and execute an extension that sends the dtmfs.  

-MC


On Tue, Sep 13, 2011 at 2:16 PM, Wayne <wayne at hamilton.net> wrote:



Sorry Michael,
 
When I started out I was sending the calls to a SIP provider. I couldn't
here the DTMF tones. For trouble shooting I moved it to a local PRI. I was
thinking that maybe this was an issue with the sip trunk and sending inband
or out of band but that doesn't seem to be the case.
 
Wayne

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, September 13, 2011 2:41 PM 

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Sending DTMF on B-leg




On Tue, Sep 13, 2011 at 11:34 AM, Wayne <wayne at hamilton.net> wrote:



That's true I don't hear any tones when I answer the phone. I don't see any
messages in my log from queue_dtmf. Any help would be great.
 
Wayne


Okay, in your original post you said you are making an OB call on a SIP
trunk, yet in your example there is no SIP trunk. You've got an internal
user and a FreeTDM FXO port. Can you please clarify what you actually have?
Who's the caller and who's the callee?
-MC

 


 
 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, September 08, 2011 6:25 PM 

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Sending DTMF on B-leg


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
<tel:%289025556747> )" >
       <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 <tel:1%2F9025556747> 
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 <tel:1%2F9025556747> 
 
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






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






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




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From bill.evergreen at gmail.com  Thu Sep 15 20:59:34 2011
From: bill.evergreen at gmail.com (bill evergreen)
Date: Thu, 15 Sep 2011 18:59:34 +0200
Subject: [Freeswitch-users] calling Skype-users and receiving calls from
	Skype-users
Message-ID: <CAAmYSev5CFAvTXzu3iYvsp1S8hN+vFCWA5gkbdX=XkhhB2aRTQ@mail.gmail.com>

Hello,

i would like to know, if it's possible from FreeSWITCH to call Skype-users
and to receive calls from
Skype-users.

As I understand, with "SipTheeSkype" that is to some extent possible. But I
am not sure whether or not
on the Skype client, some addtional software is required (what is not the
thing i'd like)

What approach would you recommend to go?


Thank's a lot for any help!

Bill
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From gmaruzz at gmail.com  Thu Sep 15 21:22:11 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 15 Sep 2011 19:22:11 +0200
Subject: [Freeswitch-users] calling Skype-users and receiving calls from
	Skype-users
In-Reply-To: <CAAmYSev5CFAvTXzu3iYvsp1S8hN+vFCWA5gkbdX=XkhhB2aRTQ@mail.gmail.com>
References: <CAAmYSev5CFAvTXzu3iYvsp1S8hN+vFCWA5gkbdX=XkhhB2aRTQ@mail.gmail.com>
Message-ID: <CALXCt0oVAaP=VEGBm8W50H3F2CusPa+q-aPKVofGZVC3kj7hew@mail.gmail.com>

Please search for "skypopen" in the wiki htttp://wiki.freeswitch.org .
Mod-skypopen connects FreeSWITCH to the skype network.
-giovanni


On 9/15/11, bill evergreen <bill.evergreen at gmail.com> wrote:
> Hello,
>
> i would like to know, if it's possible from FreeSWITCH to call Skype-users
> and to receive calls from
> Skype-users.
>
> As I understand, with "SipTheeSkype" that is to some extent possible. But I
> am not sure whether or not
> on the Skype client, some addtional software is required (what is not the
> thing i'd like)
>
> What approach would you recommend to go?
>
>
> Thank's a lot for any help!
>
> Bill
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From rhuddleston at gmail.com  Thu Sep 15 21:24:13 2011
From: rhuddleston at gmail.com (Robert-iPhone)
Date: Thu, 15 Sep 2011 13:24:13 -0400
Subject: [Freeswitch-users] calling Skype-users and receiving calls from
	Skype-users
In-Reply-To: <CAAmYSev5CFAvTXzu3iYvsp1S8hN+vFCWA5gkbdX=XkhhB2aRTQ@mail.gmail.com>
References: <CAAmYSev5CFAvTXzu3iYvsp1S8hN+vFCWA5gkbdX=XkhhB2aRTQ@mail.gmail.com>
Message-ID: <8407970B-7080-4300-BA54-EAE0494A4ABF@gmail.com>

Have you reviewed wiki?

Sent from my iPhone

On Sep 15, 2011, at 12:59 PM, bill evergreen <bill.evergreen at gmail.com> wrote:

> Hello,
> 
> i would like to know, if it's possible from FreeSWITCH to call Skype-users and to receive calls from 
> Skype-users.
> 
> As I understand, with "SipTheeSkype" that is to some extent possible. But I am not sure whether or not
> on the Skype client, some addtional software is required (what is not the thing i'd like)
> 
> What approach would you recommend to go?
> 
> 
> Thank's a lot for any help!
> 
> Bill
> 
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mi.ke at null.net  Thu Sep 15 23:39:10 2011
From: mi.ke at null.net (Mi Ke)
Date: Thu, 15 Sep 2011 15:39:10 -0400
Subject: [Freeswitch-users] changing default FS for G729
Message-ID: <20110915193911.129460@gmx.com>

Hi ALL !

 I'm trying to change default frame size from 30 to 40 bytes for G729 (commercial version) but setting correspondent codec prefs to G729 at 40i (vars.conf) gives no effect. Codec still starts with 30 bytes frames:

 2011-09-15 19:00:37.538174 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1 channels 30ms

 Is it possible to do that with mod_com_g729 ?

 Thanks,

 Mike
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From sunwood360 at gmail.com  Fri Sep 16 00:56:26 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Thu, 15 Sep 2011 13:56:26 -0700
Subject: [Freeswitch-users] FreeSWITCH with external registrar
In-Reply-To: <CAMwZ-t+EMF3C=Wwoc=Jhhy4SMg47Jqn4-jYLJDsv-3BnBCEWmw@mail.gmail.com>
References: <CAKq5mS7qfy_u2Q6cG6aF1rZPctvXpHBkyzCUHk8feP=NMSv93g@mail.gmail.com>
	<CAKq5mS7vSz4B2kxzgUn8vOJ9zPWp2Kwfw=HZEzW_Avd9UR2jHw@mail.gmail.com>
	<CAMwZ-t+EMF3C=Wwoc=Jhhy4SMg47Jqn4-jYLJDsv-3BnBCEWmw@mail.gmail.com>
Message-ID: <CAKq5mS4nA6s6k7hriCpY_hsU6saVYWi-K+M=h7Yi=4nXWEqRDA@mail.gmail.com>

Most of them are cases where FS access external as gateways.  What I am
looking for is usage case that extensions are registered to 3rd party
registrar, and FS is integrated with the registrar.
On Sep 10, 2011 5:28 PM, "Gabriel Gunderson" <gabe at gundy.org> wrote:
> 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
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jayesh.voip at gmail.com  Fri Sep 16 02:08:27 2011
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Fri, 16 Sep 2011 03:38:27 +0530
Subject: [Freeswitch-users] ESL: uuid events with filtering
Message-ID: <CALvF6vAfd+2AqgQrJehR0BvJdxiqgiAUkwbXe_Gs9_p7GkYg7Q@mail.gmail.com>

Hi,
Is it possible to filter specific events for a particular uuid through ESL.
I am using it from a PHP application as follows:

$sock->SendRecv("events plain ALL");
$sock->SendRecv("filter unique-id 12345");
$sock->sendRecv("filter Event-Name CHANNEL_ANSWER");
$sock->sendRecv("filter Event-Name CHANNEL_HANGUP");
$sock->sendRecv("filter Event-Name CHANNEL_HANGUP_COMPLETE");

In the case above, I start seeing the events coming from CHANNEL_ANSWER and
thereafter all events for that uuid is received.
Even if I write:

$sock2->SendRecv("events plain ALL");
$sock2->SendRecv("filter unique-id 12345");
$sock2->sendRecv("filter Event-Name CHANNEL_HANGUP_COMPLETE");

I get all the events belonging to uuid 12345 starting from CHANNEL_ANSWER
event name. I also tried doing "myevents 12345 plain" and then applying
filters but it still doesnt work as expected.
I tried googling a bit but could'nt find much help. Anybody pointing me to a
proper direction in this regards is highly appreciated.

Thanks,

--- Jayesh
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From Nabble_01394 at slickdeals.endjunk.com  Fri Sep 16 03:13:10 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Thu, 15 Sep 2011 16:13:10 -0700 (PDT)
Subject: [Freeswitch-users] Dialplan delimiter transform
In-Reply-To: <CAKzWOxW-rgHgoYZ-2fh0Nb7+TT8ipU86jn10vaucEJth70=PGw@mail.gmail.com>
References: <CAMKRqpjhPgxpgPomNn=tadpH6NSDmZJ3=TsyfwJFj3eoB9-+bg@mail.gmail.com>
	<CAFiqYu=5XeFoN-oq_O7T+Mc=5nzXy59-rFDvFb3v5nkLeVChMQ@mail.gmail.com>
	<CAKzWOxW-rgHgoYZ-2fh0Nb7+TT8ipU86jn10vaucEJth70=PGw@mail.gmail.com>
Message-ID: <1316128390835-6798950.post@n2.nabble.com>

So do the dots. BTW, using *^(\d+)-* as ReGex will suffice too to pass digits
before the -.

mercutioviz wrote:
> 
> On Thu, Sep 15, 2011 at 3:19 AM, Steven Ayre &lt;steveayre at gmail.com&gt;
> wrote:
> 
>> Change the regex to only pull out the first part into $1. Something like
>> this:
>>
>> <condition field="destination_number"
>> expression="^(\d+)\-(\d+\.\d+\.\d+\.\d+)$">
>>
> FYI, a bare - in a regex does not need to be escaped.
> -MC
> 
> 
>>
>> $1 will be 0114712500
>> $2 will be 10.3.12.153
>>
>> -Steve
>>
>>
>> On 15 September 2011 10:14, Chris Graham &lt;chrisg.lists at gmail.com&gt;
>> wrote:
>>
>>> Hi All,
>>>
>>> I have a simple issue I seem to not be making progress on. Due to
>>> topology
>>> hiding in one of the headquaters I send a invite as such:
>>>
>>> INVITE sip:0114712500-10.3.12.153 at x.x.x.x SIP/2.0.
>>>
>>> You can see the first part of the invite is the dialstring then a "-"
>>> used
>>> as a delimiter to bill against the hidden IP. FS duly passed this
>>> dialstring
>>> on. The carrier then rejects the call.
>>>
>>> I want to modify the dialstring in the context below not to dial
>>> 0114712500-10.3.12.153 but rather just 0114712500. I hope I am clear,
>>> and
>>> thanks in advance.
>>>
>>> <document type="freeswitch/xml">
>>>   <section name="dialplan" description="RE FS DP">
>>>     <context name="lcr_trunks">
>>>       <extension name="Catch all send to carriers">
>>>         <condition field="destination_number" expression="^(.*)$">
>>>           <action application="set" data="nibble_account=10.3.12.153"/>
>>>           <action application="lcr" data="$1 peak"/>
>>>           <action application="bridge" data="${lcr_auto_route}"/>
>>>         </condition>
>>>       </extension>
>>>     </context>
>>>   </section>
>>> </document>
>>>
>>> Chris G
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dialplan-delimiter-transform-tp6796094p6798950.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From nabble at mralston.com  Thu Sep 15 22:45:58 2011
From: nabble at mralston.com (mralston)
Date: Thu, 15 Sep 2011 11:45:58 -0700 (PDT)
Subject: [Freeswitch-users] Answer issue on inbound call
In-Reply-To: <BANLkTin+f=uDMiOouyLWAorPvNTd1Z6xaA@mail.gmail.com>
References: <BANLkTin+f=uDMiOouyLWAorPvNTd1Z6xaA@mail.gmail.com>
Message-ID: <1316112358210-6798070.post@n2.nabble.com>

It's this line:

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

Comment it out and FreeSWITCH stops automatically answering the call.

I have no idea what side effects this my have. I've just been having the
same issue myself, which is how I found this post.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Answer-issue-on-inbound-call-tp6504591p6798070.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mi.ke at null.net  Fri Sep 16 05:22:35 2011
From: mi.ke at null.net (Mi Ke)
Date: Thu, 15 Sep 2011 21:22:35 -0400
Subject: [Freeswitch-users] changing default FS for G729
Message-ID: <20110916012235.29220@gmx.com>

Please disregard my below request - the problem was a buggy endpoint. FS works as expected.

 Thanks / Mike

----- Original Message -----
From: Mi Ke
Sent: 09/15/11 10:39 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] changing default FS for G729

Hi ALL !

 I'm trying to change default frame size from 30 to 40 bytes for G729 (commercial version) but setting correspondent codec prefs to G729 at 40i (vars.conf) gives no effect. Codec still starts with 30 bytes frames:

 2011-09-15 19:00:37.538174 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1 channels 30ms

 Is it possible to do that with mod_com_g729 ?

 Thanks,

 Mike
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From lakindia89 at gmail.com  Fri Sep 16 08:03:41 2011
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 16 Sep 2011 09:33:41 +0530
Subject: [Freeswitch-users] ESL: uuid events with filtering
In-Reply-To: <CALvF6vAfd+2AqgQrJehR0BvJdxiqgiAUkwbXe_Gs9_p7GkYg7Q@mail.gmail.com>
References: <CALvF6vAfd+2AqgQrJehR0BvJdxiqgiAUkwbXe_Gs9_p7GkYg7Q@mail.gmail.com>
Message-ID: <CAMx6C7fgq9z9M_9=KUM88yKq=ovHtkiVH+OeJTFPWjrOvHo4JQ@mail.gmail.com>

I've not tried applying filter for Event-Name. But what I do is to register
only for the specific events that I need.

If you want to get only specific events, you can do as follows

$sock->SendRecv("events plain CHANNEL_ANSWER") # You will see only the
CHANNEL_ANSWER event....

Before getting into the code, you can actually experiment this using nc (
Outbound Socket ) or telnet ( Inbound Socket ), so that you will have a
better idea on how to in-corporate it in code.



On Fri, Sep 16, 2011 at 3:38 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:

> Hi,
> Is it possible to filter specific events for a particular uuid through ESL.
> I am using it from a PHP application as follows:
>
> $sock->SendRecv("events plain ALL");
> $sock->SendRecv("filter unique-id 12345");
> $sock->sendRecv("filter Event-Name CHANNEL_ANSWER");
> $sock->sendRecv("filter Event-Name CHANNEL_HANGUP");
> $sock->sendRecv("filter Event-Name CHANNEL_HANGUP_COMPLETE");
>
> In the case above, I start seeing the events coming from CHANNEL_ANSWER and
> thereafter all events for that uuid is received.
> Even if I write:
>
> $sock2->SendRecv("events plain ALL");
> $sock2->SendRecv("filter unique-id 12345");
> $sock2->sendRecv("filter Event-Name CHANNEL_HANGUP_COMPLETE");
>
> I get all the events belonging to uuid 12345 starting from CHANNEL_ANSWER
> event name. I also tried doing "myevents 12345 plain" and then applying
> filters but it still doesnt work as expected.
> I tried googling a bit but could'nt find much help. Anybody pointing me to
> a proper direction in this regards is highly appreciated.
>
> Thanks,
>
> --- Jayesh
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ocset at the800group.com  Fri Sep 16 09:14:13 2011
From: ocset at the800group.com (ocset)
Date: Fri, 16 Sep 2011 13:14:13 +0800
Subject: [Freeswitch-users] GXW4104 gateway setup for outgoing calls
In-Reply-To: <CAHKqMb7BV1oVa5NqvZP+mgS1QSjuaUu-JPpHRFpqmNBs8G8mHQ@mail.gmail.com>
References: <4E6EFCFA.6000803@the800group.com>
	<CAHKqMb7vF4ULKjWsHqP20xVRy0oOrDo9A3fpwCBr2NXAjsJ6iw@mail.gmail.com>
	<4E6F2C07.80800@the800group.com>
	<CAHKqMb7BV1oVa5NqvZP+mgS1QSjuaUu-JPpHRFpqmNBs8G8mHQ@mail.gmail.com>
Message-ID: <4E72DB25.7010706@the800group.com>

Hi Nandy

Thanks for your help so far - unfortunately this is still not working.

Based on your examples, I have created the following  two files

1.
../sip_profile/internal/01_custom.xml

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

<gateway name="gxw4104-fxo">
<param name="username" value="1019"/>
<param name="password" value="1234"/>
<param name="realm" value="192.168.0.160"/>
<param name="sip-port" value="5060"/>
<param name="rtp_ip" value="192.168.0.160"/>
<param name="dtmf-type" value="rfc2833"/>
<param name="expire-seconds" value="600"/>
<param name="register" value="false"/>
<param name="caller-id-in-from" value="false"/>
</gateway>
</include>

2.
../dialplan/default/01_custom.xml

<include>
<extension name="gxw4104-fxo-local">
<condition field="${toll_allow}" expression="local"/>
<condition field="destination_number" expression="^(\d{10})$">
<action application="set" data="effective_caller_id_number=5555555555"/>
<action application="set" data="effective_caller_id_name=ThisIsMyCompany"/>
<action application="set" data="ignore_early_media=ring_ready"/>
<action application="set" data="ringback=${us-ring}"/>
<action application="bridge" 
data="sofia/gateway/gxw4104-fxo/$1 at 192.168.0.160:5060"/>
</condition>
</extension>
</include>



Command "sofia status"  shows the followiing

internal                               profile                
sip:mod_sofia at 192.168.0.23:5060         RUNNING (0)
internal::gxw4104-fxo        gateway             
sip:1019 at 192.168.0.160                        NOREG
external                              profile                
sip:mod_sofia at 192.168.0.23:5080         RUNNING (0)
external::example.com      gateway             
sip:joeuser at example.com                      NOREG
internal-ipv6                       profile                
sip:mod_sofia@[::1]:5060                        RUNNING (0)
192.168.0.23                      alias                   
internal                                                    ALIASED


Here is the call log when I try do dial out:

2011-09-16 12:57:30.903616 [NOTICE] switch_channel.c:669 New Channel 
sofia/internal/1014 at 192.168.0.23 [0a83e803-b3fc-4a46-bc74-9d6786dac8e2]
2011-09-16 12:57:30.933863 [INFO] mod_dialplan_xml.c:418 Processing User 
1014->0412345678 in context default
2011-09-16 12:57:30.950359 [NOTICE] switch_channel.c:669 New Channel 
sofia/internal/0412345678 at 192.168.0.160:5060 
[24111b8d-25f0-4433-a49f-88973730ebfb]
2011-09-16 12:57:40.957391 [NOTICE] sofia.c:4789 Hangup 
sofia/internal/0412345678 at 192.168.0.160:5060 [CS_CONSUME_MEDIA] 
[NORMAL_TEMPORARY_FAILURE]
2011-09-16 12:57:40.958554 [NOTICE] switch_core_session.c:1182 Session 7 
(sofia/internal/0412345678 at 192.168.0.160:5060) Ended
2011-09-16 12:57:40.958554 [NOTICE] switch_core_session.c:1184 Close 
Channel sofia/internal/0412345678 at 192.168.0.160:5060 [CS_DESTROY]
2011-09-16 12:57:40.958554 [INFO] mod_dptools.c:2355 Originate Failed.  
Cause: NORMAL_TEMPORARY_FAILURE
2011-09-16 12:57:40.958554 [NOTICE] mod_dptools.c:2418 Hangup 
sofia/internal/1014 at 192.168.0.23 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
2011-09-16 12:57:40.995191 [NOTICE] switch_core_session.c:1182 Session 6 
(sofia/internal/1014 at 192.168.0.23) Ended
2011-09-16 12:57:40.995191 [NOTICE] switch_core_session.c:1184 Close 
Channel sofia/internal/1014 at 192.168.0.23 [CS_DESTROY]



I have the following questions

1. What is the significance of the user 1019? I have a default install 
of FS so that user does exist but I am not logged in as that user on my 
sip phone. I am logged in as user 1014.

2. The resultant log does not show my gateway being used but instead 
shows "/sofia/internal/0412345678 at 192.168.0.160:5060". Is that expected 
behaviour?

3. I assumed that the IP address 192.168.0.9 in your example is the 
address of your HT503 and not FS. I have thus replaced it with the IP 
address from my GXW4104 (192.168.0.160). Is that correct?






On 09/13/2011 11:42 PM, Nandy Dagondon wrote:
> i inserted my answers to your questions below. for point #3), here's 
> an example how i configured my FXO port of ht503.
>
> included in  sip_profile/internal:
> <include>
> <gateway name="ht503-fxo">
> <param name="username" value="1019"/> <-- it's registered to receive 
> incoming calls
> <param name="realm" value="192.168.0.9"/>
> <param name="sip-port" value="5062"/> <-- port 5060 is set to the FXS port
> <param name="password" value="1234"/>
> <param name="rtp_ip" value="192.168.0.9"/>
> <param name="dtmf-type" value="rfc2833"/>
> <param name="expire-seconds" value="600"/>
> <param name="register" value="false"/>
> <param name="caller-id-in-from" value="false"/>
> </gateway>
> </include>
>
> included  in dialplan/default
>
> <include>
> <extension name="ht503-fxo-local">
> <condition field="${toll_allow}" expression="local"/>
> <condition field="destination_number" expression="^9([2-9]\d{6})$">
> <action application="set" data="effective_caller_id_number=0321234567"/>
> <action application="set" data="effective_caller_id_name=ThisIsMy 
> Company"/>
> <action application="set" data="ignore_early_media=ring_ready"/>
> <action application="set" data="ringback=${us-ring}"/>
> <action application="bridge" 
> data="sofia/gateway/ht503-fxo/$1 at 192.168.0.9:5062 
> <http://1 at 192.168.0.9:5062>"/>
> </condition>
> </extension>
> </include>
>
> it looks you can create 4 internal gateways for the every port, fxo-1 
> to fxo-4, w/ the same realm/rtp_ip values but setting different 
> sip-port values. then  your bridge app would be:
>
> <action application="bridge" 
> data="sofia/gateway/fxo-1/$1|sofia/gateway/fxo-2/$1|sofia/gateway/fxo-3/$1sofia/gateway/fxo-4/$1"/>
>
> if u want to dialout any free port.
>
> i haven't tested the above. just try it. i hope it works.
>
> -nandy
>
>
> On Tue, Sep 13, 2011 at 6:10 PM, ocset <ocset at the800group.com 
> <mailto:ocset at the800group.com>> wrote:
>
>     Hi Nandy
>
>     Thanks for your reply. I assume 192.168.0.9 in your example is the
>     IP address of the GXW4104?
>
> yes.
>
>
>     Some more questions
>
>     1. When you say port number, is this something I should be setting
>     up on the GXW4104 so that it is listening on those 4 port numbers?
>     If yes, what would be the setting I am looking for?
>
> not for every port. the gateway has a base port number e.g. 5060 for 
> port#1. add 2 to the subsequent ports e.g. 5062 for port#2 and so on. 
> this is pointed out by sergey.
>
>
>     2. Does that mean I don't define a new gateway in FreeSWITCH?
>
> it's an option. but defining a gateway is cleaner.
>
>
>     3. In your example, you said the bridge data would be
>     7654321 at 192.168.0.9:5063 <mailto:7654321 at 192.168.0.9:5063>. What
>     would the whole line look like in the dialplan?*
>
>     <action application="bridge"
>     data="sofia/gateway/7654321 at 192.168.0.9:5063
>     <mailto:sofia/gateway/7654321 at 192.168.0.9:5063>**"/>*
>
>     Still very confused :-)
>
>     Thanks
>
>
>     On 09/13/2011 03:45 PM, Nandy Dagondon wrote:
>>     hi,
>>
>>     if GWX4104 is in your local network, use the internal profile for
>>     the gateway. register your FXO accounts to receive incoming calls
>>     (i think you did this already).
>>
>>     to dialout the ports, specify the port number 5060~5063 assuming
>>     Port1 starts at 5060. to dialout via port4, the bridge data
>>     should look like:
>>
>>     7654321 at 192.168.0.9:5063 <http://7654321 at 192.168.0.9:5063>
>>
>>     hope it helps.
>>
>>     -nandy
>>
>>
>>     On Tue, Sep 13, 2011 at 2:49 PM, ocset <ocset at the800group.com
>>     <mailto:ocset at the800group.com>> wrote:
>>
>>         Hi
>>
>>         I have recently bought a Grandstream GXW4104 (4 FXO ports)
>>         and need some help setting up a gateway to call out using the
>>         GXW4104. I am really out of my depth here and may be looking
>>         at this the wrong way so please bear with me.
>>
>>         I followed the advice on this website
>>         "http://www.timhunt.net/wiki/FreeSwitch:GXW4104"
>>         <http://www.timhunt.net/wiki/FreeSwitch:GXW4104> and incoming
>>         calls from a PSTN line are working great. Now I need to setup
>>         a dialplan so that outgoing calls are routed through the same
>>         PSTN line on the GXW4104. I will eventually have 4 PSTN lines
>>         with a dialplan to use the first available line (if that is
>>         possible).
>>
>>         According to the FreeSWITCH 1.0.6 book (and many online
>>         posts) I need to create a gateway and a dialplan but all the
>>         gateway examples are for SIP accounts.
>>
>>         So, the gateway definition seems to need a username and
>>         password but the GXW4104 does not have that capability. I
>>         found this gateway definition in the  freeswitch.xml.fsxml
>>         file but am not sure how many of these variables are required.
>>
>>         <gateways>
>>         <gateway name="example.com <http://example.com>">
>>         <param name="username" value="joeuser"/>
>>         <param name="password" value="password"/>
>>         <param name="from-user" value="joeuser"/>
>>         <param name="from-domain" value="example.com
>>         <http://example.com>"/>
>>         <param name="expire-seconds" value="600"/>
>>         <param name="register" value="false"/>
>>         <param name="retry-seconds" value="30"/>
>>         <param name="extension" value="5000"/>
>>         <param name="context" value="public"/>
>>         </gateway>
>>         </gateways>
>>
>>
>>         If I define a gateway called "gxw4104", then this is what I
>>         think a simple dialplan should look like but I'm not sure of
>>         the gateway details in the "bridge" section of the definition.
>>
>>         <extension name="gxw4104_out">
>>         <condition field="destination_number" expression="^(\d{10})$">
>>         *<action application="bridge"
>>         data="sofia/gateway/gxw4104/........"/>        (what should
>>         this be???)*
>>         </condition>
>>         </extension>
>>
>>         Am I moving in the right direction and can someone fill in
>>         the blanks for me please
>>
>>         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
>>
>>
>>
>>
>>     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
>
>
>
>
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From govoiper at gmail.com  Fri Sep 16 09:56:42 2011
From: govoiper at gmail.com (Sam Govind)
Date: Fri, 16 Sep 2011 10:56:42 +0500
Subject: [Freeswitch-users] FreeSWITCH with external registrar
In-Reply-To: <CAKq5mS4nA6s6k7hriCpY_hsU6saVYWi-K+M=h7Yi=4nXWEqRDA@mail.gmail.com>
References: <CAKq5mS7qfy_u2Q6cG6aF1rZPctvXpHBkyzCUHk8feP=NMSv93g@mail.gmail.com>
	<CAKq5mS7vSz4B2kxzgUn8vOJ9zPWp2Kwfw=HZEzW_Avd9UR2jHw@mail.gmail.com>
	<CAMwZ-t+EMF3C=Wwoc=Jhhy4SMg47Jqn4-jYLJDsv-3BnBCEWmw@mail.gmail.com>
	<CAKq5mS4nA6s6k7hriCpY_hsU6saVYWi-K+M=h7Yi=4nXWEqRDA@mail.gmail.com>
Message-ID: <CAJUJwthZfWHN1xn0XW9PJKOrNruvUyRmGHNeBsD37yBY=HoH1A@mail.gmail.com>

I guess you are looking for a SIP Proxy acting as SBC. This could be
OpenSIPS forwarding REGISTER attempt to REGISTRAR server while INVITEs are
handled by FS !
I don't think FreeSwitch can't handle forwarding of Registrations (is not a
SIP proxy)

On Fri, Sep 16, 2011 at 1:56 AM, envelopes envelopes
<sunwood360 at gmail.com>wrote:

> Most of them are cases where FS access external as gateways.  What I am
> looking for is usage case that extensions are registered to 3rd party
> registrar, and FS is integrated with the registrar.
> On Sep 10, 2011 5:28 PM, "Gabriel Gunderson" <gabe at gundy.org> wrote:
> > 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
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sunwood360 at gmail.com  Fri Sep 16 11:12:55 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Fri, 16 Sep 2011 00:12:55 -0700
Subject: [Freeswitch-users] FreeSWITCH with external registrar
In-Reply-To: <CAJUJwthZfWHN1xn0XW9PJKOrNruvUyRmGHNeBsD37yBY=HoH1A@mail.gmail.com>
References: <CAKq5mS7qfy_u2Q6cG6aF1rZPctvXpHBkyzCUHk8feP=NMSv93g@mail.gmail.com>
	<CAKq5mS7vSz4B2kxzgUn8vOJ9zPWp2Kwfw=HZEzW_Avd9UR2jHw@mail.gmail.com>
	<CAMwZ-t+EMF3C=Wwoc=Jhhy4SMg47Jqn4-jYLJDsv-3BnBCEWmw@mail.gmail.com>
	<CAKq5mS4nA6s6k7hriCpY_hsU6saVYWi-K+M=h7Yi=4nXWEqRDA@mail.gmail.com>
	<CAJUJwthZfWHN1xn0XW9PJKOrNruvUyRmGHNeBsD37yBY=HoH1A@mail.gmail.com>
Message-ID: <CAKq5mS7=q4oAPFEug=9yap4Pk2bc0jT3+LpbTuFfuw-zD6-w0A@mail.gmail.com>

Sip clients are registered to a 3rd party registrar via OpenSIPS and
FreeSwitch has no knowledge of this. How does FreeSWITCH get the contact
address of sip client and determinate the client is currently active
registered If FreeSwitch receives INVITE?  I am wondering what is the proper
way to handle this in FS?

option 1 : build a module to query REGISTRAR's location service that
provides special API.
option 2 : forward INVITE to REGISTRAR , which rewrites SIP headers and
routes request back to FS.

please advice!

On Thu, Sep 15, 2011 at 10:56 PM, Sam Govind <govoiper at gmail.com> wrote:

> I guess you are looking for a SIP Proxy acting as SBC. This could be
> OpenSIPS forwarding REGISTER attempt to REGISTRAR server while INVITEs are
> handled by FS !
> I don't think FreeSwitch can't handle forwarding of Registrations (is not a
> SIP proxy)
>
>
> On Fri, Sep 16, 2011 at 1:56 AM, envelopes envelopes <sunwood360 at gmail.com
> > wrote:
>
>> Most of them are cases where FS access external as gateways.  What I am
>> looking for is usage case that extensions are registered to 3rd party
>> registrar, and FS is integrated with the registrar.
>> On Sep 10, 2011 5:28 PM, "Gabriel Gunderson" <gabe at gundy.org> wrote:
>> > 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
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lfurrea at gmail.com  Fri Sep 16 11:30:24 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Fri, 16 Sep 2011 01:30:24 -0600
Subject: [Freeswitch-users] non-blocking session:streamFile on LUA script
Message-ID: <CAB2KLydoKywnEBkYhH-yLzNRVukutGLsNZpxCW87eULD5HeswA@mail.gmail.com>

Hello all,

I am trying to write a script in which I need to simultaneously:

1. Play a file
2. Read digits
3. Ring a phone

I can play the file an read digits using setInputCallback

But the playing of file blocks for the origination of a b leg using
freeswitch.Session()

Am I SOL?

TIA
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From gcd at i.ph  Fri Sep 16 11:35:05 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Fri, 16 Sep 2011 15:35:05 +0800
Subject: [Freeswitch-users] GXW4104 gateway setup for outgoing calls
In-Reply-To: <4E72DB25.7010706@the800group.com>
References: <4E6EFCFA.6000803@the800group.com>
	<CAHKqMb7vF4ULKjWsHqP20xVRy0oOrDo9A3fpwCBr2NXAjsJ6iw@mail.gmail.com>
	<4E6F2C07.80800@the800group.com>
	<CAHKqMb7BV1oVa5NqvZP+mgS1QSjuaUu-JPpHRFpqmNBs8G8mHQ@mail.gmail.com>
	<4E72DB25.7010706@the800group.com>
Message-ID: <CAHKqMb69+OqObSnJdGEOVwreahbaw0SEinJgyixYgzG9FG3UdQ@mail.gmail.com>

pls see my comments below. you're welcome.
-nandy

On Fri, Sep 16, 2011 at 1:14 PM, ocset <ocset at the800group.com> wrote:

>  Hi Nandy
>
> Thanks for your help so far - unfortunately this is still not working.
>
> Based on your examples, I have created the following  two files
>
> 1.
> ../sip_profile/internal/01_custom.xml
>
> <include>
>   <param name="multiple-registrations" value="true"/>
>   <param name="accept-blind-reg" value="true"/>
>   <param name="accept-blind-auth" value="true"/>
>
>   <gateway name="gxw4104-fxo">
>
>      <param name="username" value="1019"/>
>      <param name="password" value="1234"/>
>      <param name="realm" value="192.168.0.160"/>
>      <param name="sip-port" value="5060"/>
>      <param name="rtp_ip" value="192.168.0.160"/>
>
>      <param name="dtmf-type" value="rfc2833"/>
>      <param name="expire-seconds" value="600"/>
>      <param name="register" value="false"/>
>      <param name="caller-id-in-from" value="false"/>
>   </gateway>
> </include>
>
> 2.
> ../dialplan/default/01_custom.xml
>
> <include>
>   <extension name="gxw4104-fxo-local">
>
>     <condition field="${toll_allow}" expression="local"/>
>     <condition field="destination_number" expression="^(\d{10})$">
>     <action application="set"
> data="effective_caller_id_number=5555555555"/>
>
>     <action application="set"
> data="effective_caller_id_name=ThisIsMyCompany"/>
>     <action application="set" data="ignore_early_media=ring_ready"/>
>     <action application="set" data="ringback=${us-ring}"/>
>       <action application="bridge" data=
> "sofia/gateway/gxw4104-fxo/$1 at 192.168.0.160:5060"<sofia/gateway/gxw4104-fxo/$1 at 192.168.0.160:5060>
> />
>     </condition>
>   </extension>
> </include>
>
>
>
> Command "sofia status"  shows the followiing
>
> internal                               profile
> sip:mod_sofia at 192.168.0.23:5060         RUNNING (0)
> internal::gxw4104-fxo        gateway             sip:1019 at 192.168.0.160
> NOREG
> external                              profile
> sip:mod_sofia at 192.168.0.23:5080         RUNNING (0)
> external::example.com      gateway             sip:joeuser at example.com
> NOREG
> internal-ipv6                       profile                sip:mod_sofia@[::1]:5060
> RUNNING (0)
> 192.168.0.23                      alias
> internal                                                    ALIASED
>
>
> Here is the call log when I try do dial out:
>
> 2011-09-16 12:57:30.903616 [NOTICE] switch_channel.c:669 New Channel
> sofia/internal/1014 at 192.168.0.23 [0a83e803-b3fc-4a46-bc74-9d6786dac8e2]
> 2011-09-16 12:57:30.933863 [INFO] mod_dialplan_xml.c:418 Processing User
> 1014->0412345678 in context default
> 2011-09-16 12:57:30.950359 [NOTICE] switch_channel.c:669 New Channel
> sofia/internal/0412345678 at 192.168.0.160:5060[24111b8d-25f0-4433-a49f-88973730ebfb]
> 2011-09-16 12:57:40.957391 [NOTICE] sofia.c:4789 Hangup
> sofia/internal/0412345678 at 192.168.0.160:5060 [CS_CONSUME_MEDIA]
> [NORMAL_TEMPORARY_FAILURE]
> 2011-09-16 12:57:40.958554 [NOTICE] switch_core_session.c:1182 Session 7 (
> sofia/internal/0412345678 at 192.168.0.160:5060) Ended
> 2011-09-16 12:57:40.958554 [NOTICE] switch_core_session.c:1184 Close
> Channel sofia/internal/0412345678 at 192.168.0.160:5060 [CS_DESTROY]
> 2011-09-16 12:57:40.958554 [INFO] mod_dptools.c:2355 Originate Failed.
> Cause: NORMAL_TEMPORARY_FAILURE
> 2011-09-16 12:57:40.958554 [NOTICE] mod_dptools.c:2418 Hangup
> sofia/internal/1014 at 192.168.0.23 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
> 2011-09-16 12:57:40.995191 [NOTICE] switch_core_session.c:1182 Session 6 (
> sofia/internal/1014 at 192.168.0.23) Ended
> 2011-09-16 12:57:40.995191 [NOTICE] switch_core_session.c:1184 Close
> Channel sofia/internal/1014 at 192.168.0.23 [CS_DESTROY]
>
>
>
> I have the following questions
>
> 1. What is the significance of the user 1019? I have a default install of
> FS so that user does exist but I am not logged in as that user on my sip
> phone. I am logged in as user 1014.
>
> 1019 is the user account for the FXO port to handle incoming calls. so, the
login details must be set on directory/default/1019.xml.

actually,  the user and password entries can be deleted because FS is not
registering to the FXO gateway.


> 2. The resultant log does not show my gateway being used but instead shows
> "/sofia/internal/0412345678 at 192.168.0.160:5060"</sofia/internal/0412345678 at 192.168.0.160:5060>.
> Is that expected behaviour?
>
> it's not hitting the dialplan, i guess. i suggest you create prefix 9 for
PSTN calls e.g.
    <condition field="destination_number" expression="^9(\d+)$">

and make the file ../dialplan/default/01_custom.xml is on top of
dialplan/default directory so it will be scanned early.


> 3. I assumed that the IP address 192.168.0.9 in your example is the address
> of your HT503 and not FS. I have thus replaced it with the IP address from
> my GXW4104 (192.168.0.160). Is that correct?
>
> that is correct.

>
>
>
>
>
>
> On 09/13/2011 11:42 PM, Nandy Dagondon wrote:
>
> i inserted my answers to your questions below. for point #3), here's an
> example how i configured my FXO port of ht503.
>
> included in  sip_profile/internal:
> <include>
>   <gateway name="ht503-fxo">
>   <param name="username" value="1019"/>   <-- it's registered to receive
> incoming calls
>   <param name="realm" value="192.168.0.9"/>
>   <param name="sip-port" value="5062"/>   <-- port 5060 is set to the FXS
> port
>   <param name="password" value="1234"/>
>   <param name="rtp_ip" value="192.168.0.9"/>
>   <param name="dtmf-type" value="rfc2833"/>
>   <param name="expire-seconds" value="600"/>
>   <param name="register" value="false"/>
>   <param name="caller-id-in-from" value="false"/>
>   </gateway>
> </include>
>
> included  in dialplan/default
>
> <include>
>   <extension name="ht503-fxo-local">
>     <condition field="${toll_allow}" expression="local"/>
>     <condition field="destination_number" expression="^9([2-9]\d{6})$">
>     <action application="set" data="effective_caller_id_number=0321234567
> "/>
>     <action application="set" data="effective_caller_id_name=ThisIsMy
> Company"/>
>     <action application="set" data="ignore_early_media=ring_ready"/>
>     <action application="set" data="ringback=${us-ring}"/>
>       <action application="bridge" data="sofia/gateway/ht503-fxo/$
> 1 at 192.168.0.9:5062"/>
>     </condition>
>   </extension>
> </include>
>
> it looks you can create 4 internal gateways for the every port, fxo-1 to
> fxo-4, w/ the same realm/rtp_ip values but setting different sip-port
> values. then  your bridge app would be:
>
>       <action application="bridge"
> data="sofia/gateway/fxo-1/$1|sofia/gateway/fxo-2/$1|sofia/gateway/fxo-3/$1sofia/gateway/fxo-4/$1"/>
>
> if u want to dialout any free port.
>
> i haven't tested the above. just try it. i hope it works.
>
> -nandy
>
>
> On Tue, Sep 13, 2011 at 6:10 PM, ocset <ocset at the800group.com> wrote:
>
>>  Hi Nandy
>>
>> Thanks for your reply. I assume 192.168.0.9 in your example is the IP
>> address of the GXW4104?
>>
> yes.
>
>>
>>
>  Some more questions
>>
>> 1. When you say port number, is this something I should be setting up on
>> the GXW4104 so that it is listening on those 4 port numbers? If yes, what
>> would be the setting I am looking for?
>>
> not for every port. the gateway has a base port number e.g. 5060 for
> port#1. add 2 to the subsequent ports e.g. 5062 for port#2 and so on. this
> is pointed out by sergey.
>
>>
>> 2. Does that mean I don't define a new gateway in FreeSWITCH?
>>
> it's an option. but defining a gateway is cleaner.
>
>
>>
>>
>  3. In your example, you said the bridge data would be
>> 7654321 at 192.168.0.9:5063. What would the whole line look like in the
>> dialplan?*
>>
>> <action application="bridge" data="sofia/gateway/7654321 at 192.168.0.9:5063
>> **"/>*
>>
>> Still very confused :-)
>>
>> Thanks
>>
>>
>> On 09/13/2011 03:45 PM, Nandy Dagondon wrote:
>>
>> hi,
>>
>> if GWX4104 is in your local network, use the internal profile for the
>> gateway. register your FXO accounts to receive incoming calls (i think you
>> did this already).
>>
>> to dialout the ports, specify the port number 5060~5063 assuming Port1
>> starts at 5060. to dialout via port4, the bridge data should look like:
>>
>> 7654321 at 192.168.0.9:5063
>>
>> hope it helps.
>>
>> -nandy
>>
>>
>> On Tue, Sep 13, 2011 at 2:49 PM, ocset <ocset at the800group.com> wrote:
>>
>>>  Hi
>>>
>>> I have recently bought a Grandstream GXW4104 (4 FXO ports) and need some
>>> help setting up a gateway to call out using the GXW4104. I am really out of
>>> my depth here and may be looking at this the wrong way so please bear with
>>> me.
>>>
>>> I followed the advice on this website
>>> "http://www.timhunt.net/wiki/FreeSwitch:GXW4104"<http://www.timhunt.net/wiki/FreeSwitch:GXW4104>and incoming calls from a PSTN line are working great. Now I need to setup a
>>> dialplan so that outgoing calls are routed through the same PSTN line on the
>>> GXW4104. I will eventually have 4 PSTN lines with a dialplan to use the
>>> first available line (if that is possible).
>>>
>>> According to the FreeSWITCH 1.0.6 book (and many online posts) I need to
>>> create a gateway and a dialplan but all the gateway examples are for SIP
>>> accounts.
>>>
>>> So, the gateway definition seems to need a username and password but the
>>> GXW4104 does not have that capability. I found this gateway definition in
>>> the  freeswitch.xml.fsxml file but am not sure how many of these variables
>>> are required.
>>>
>>> <gateways>
>>>    <gateway name="example.com">
>>>       <param name="username" value="joeuser"/>
>>>       <param name="password" value="password"/>
>>>       <param name="from-user" value="joeuser"/>
>>>       <param name="from-domain" value="example.com"/>
>>>       <param name="expire-seconds" value="600"/>
>>>       <param name="register" value="false"/>
>>>       <param name="retry-seconds" value="30"/>
>>>       <param name="extension" value="5000"/>
>>>       <param name="context" value="public"/>
>>>    </gateway>
>>> </gateways>
>>>
>>>
>>> If I define a gateway called "gxw4104", then this is what I think a
>>> simple dialplan should look like but I'm not sure of the gateway details in
>>> the "bridge" section of the definition.
>>>
>>> <extension name="gxw4104_out">
>>>    <condition field="destination_number" expression="^(\d{10})$">
>>>       *<action application="bridge"
>>> data="sofia/gateway/gxw4104/........"/>        (what should this be???)*
>>>    </condition>
>>> </extension>
>>>
>>> Am I moving in the right direction and can someone fill in the blanks for
>>> me please
>>>
>>> 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
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From govoiper at gmail.com  Fri Sep 16 12:49:09 2011
From: govoiper at gmail.com (Sam Govind)
Date: Fri, 16 Sep 2011 13:49:09 +0500
Subject: [Freeswitch-users] FreeSWITCH with external registrar
In-Reply-To: <CAKq5mS7=q4oAPFEug=9yap4Pk2bc0jT3+LpbTuFfuw-zD6-w0A@mail.gmail.com>
References: <CAKq5mS7qfy_u2Q6cG6aF1rZPctvXpHBkyzCUHk8feP=NMSv93g@mail.gmail.com>
	<CAKq5mS7vSz4B2kxzgUn8vOJ9zPWp2Kwfw=HZEzW_Avd9UR2jHw@mail.gmail.com>
	<CAMwZ-t+EMF3C=Wwoc=Jhhy4SMg47Jqn4-jYLJDsv-3BnBCEWmw@mail.gmail.com>
	<CAKq5mS4nA6s6k7hriCpY_hsU6saVYWi-K+M=h7Yi=4nXWEqRDA@mail.gmail.com>
	<CAJUJwthZfWHN1xn0XW9PJKOrNruvUyRmGHNeBsD37yBY=HoH1A@mail.gmail.com>
	<CAKq5mS7=q4oAPFEug=9yap4Pk2bc0jT3+LpbTuFfuw-zD6-w0A@mail.gmail.com>
Message-ID: <CAJUJwtg2isQ85UjyY3zxu66RVg+vbNBku77Gxug4HULFTqHUXw@mail.gmail.com>

I'm not very very good at FreeSwitch ATM. I hope for option 1 some other
expert reply.

As far as second option is concerned, this seems good but I think it'll
complicate the code on first layer SBC/proxy.

What I'm thinking is using a single DB for REGISTRAR and Front layer proxy,
whenever an INVITE comes in just check the DB and if usrloc module returns
info corresponding to the callee then use the headers and forward call to FS
to bridge.
REGISTER requests still go to REGISTRAR, I don't think sharing the same DB
would overload front Proxy.


On Fri, Sep 16, 2011 at 12:12 PM, envelopes envelopes
<sunwood360 at gmail.com>wrote:

> Sip clients are registered to a 3rd party registrar via OpenSIPS and
> FreeSwitch has no knowledge of this. How does FreeSWITCH get the contact
> address of sip client and determinate the client is currently active
> registered If FreeSwitch receives INVITE?  I am wondering what is the proper
> way to handle this in FS?
>
> option 1 : build a module to query REGISTRAR's location service that
> provides special API.
> option 2 : forward INVITE to REGISTRAR , which rewrites SIP headers and
> routes request back to FS.
>
> please advice!
>
>
> On Thu, Sep 15, 2011 at 10:56 PM, Sam Govind <govoiper at gmail.com> wrote:
>
>> I guess you are looking for a SIP Proxy acting as SBC. This could be
>> OpenSIPS forwarding REGISTER attempt to REGISTRAR server while INVITEs are
>> handled by FS !
>> I don't think FreeSwitch can't handle forwarding of Registrations (is not
>> a SIP proxy)
>>
>>
>> On Fri, Sep 16, 2011 at 1:56 AM, envelopes envelopes <
>> sunwood360 at gmail.com> wrote:
>>
>>> Most of them are cases where FS access external as gateways.  What I am
>>> looking for is usage case that extensions are registered to 3rd party
>>> registrar, and FS is integrated with the registrar.
>>> On Sep 10, 2011 5:28 PM, "Gabriel Gunderson" <gabe at gundy.org> wrote:
>>> > 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
>>> >
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sascha.daniels at amooma.de  Fri Sep 16 14:49:50 2011
From: sascha.daniels at amooma.de (Sascha Daniels)
Date: Fri, 16 Sep 2011 12:49:50 +0200
Subject: [Freeswitch-users] mod_cdr_sqlite schema.sql
Message-ID: <4E7329CE.1050800@amooma.de>

Hi together,

I would like to write cdr data into the sqlite file of my rails application.

According to http://wiki.freeswitch.org/wiki/Mod_cdr I can find the
schema.sql in the source directory.

Unfortunately it is not there.

All needed fields are in the MySQL example, but I need to know what the
default table name is.

Can someone tell me the table name, so I can generate my migration?

Kind regards

                        Sascha

-- 
AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied  -->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister Montabaur B14998

B?cher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de


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From michal.bielicki at seventhsignal.de  Fri Sep 16 16:41:23 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Fri, 16 Sep 2011 14:41:23 +0200
Subject: [Freeswitch-users] mod_cdr_sqlite schema.sql
In-Reply-To: <4E7329CE.1050800@amooma.de>
References: <4E7329CE.1050800@amooma.de>
Message-ID: <ABFDC784-B031-41E5-B8D6-0D616BE9C251@seventhsignal.de>

It is in the sourcecode since the files with the tables get created if they are not there.
#
Am 16.09.2011 um 12:49 schrieb Sascha Daniels:

> Hi together,
> 
> I would like to write cdr data into the sqlite file of my rails application.
> 
> According to http://wiki.freeswitch.org/wiki/Mod_cdr I can find the
> schema.sql in the source directory.
> 
> Unfortunately it is not there.
> 
> All needed fields are in the MySQL example, but I need to know what the
> default table name is.
> 
> Can someone tell me the table name, so I can generate my migration?
> 
> Kind regards
> 
>                        Sascha
> 
> -- 
> AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied  -->  http://www.amooma.de
> Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister Montabaur B14998
> 
> B?cher: http://das-asterisk-buch.de - http://ruby-auf-schienen.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

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 chrisg.lists at gmail.com  Fri Sep 16 17:13:36 2011
From: chrisg.lists at gmail.com (Chris Graham)
Date: Fri, 16 Sep 2011 15:13:36 +0200
Subject: [Freeswitch-users] Dialplan delimiter transform
In-Reply-To: <1316128390835-6798950.post@n2.nabble.com>
References: <CAMKRqpjhPgxpgPomNn=tadpH6NSDmZJ3=TsyfwJFj3eoB9-+bg@mail.gmail.com>
	<CAFiqYu=5XeFoN-oq_O7T+Mc=5nzXy59-rFDvFb3v5nkLeVChMQ@mail.gmail.com>
	<CAKzWOxW-rgHgoYZ-2fh0Nb7+TT8ipU86jn10vaucEJth70=PGw@mail.gmail.com>
	<1316128390835-6798950.post@n2.nabble.com>
Message-ID: <CAMKRqphpNXBRXLT-R_STRS2rST2Ch2vR+sNsUNAJsSbb4X+6oA@mail.gmail.com>

Hi All,

Thanks for the prompt response. I ended up with:

<condition field="destination_number" expression="^.*-(\d+)$">

By changing the invite from:

INVITE sip:0114712500-10.3.12.153 at x.x.x.x SIP/2.0.

to:

INVITE sip:10.3.12.153-0114712500 at x.x.x.x SIP/2.0.

Really loving the regex's now am using them in the correct manner!

Thanks,
Chris G


On Fri, Sep 16, 2011 at 1:13 AM, mazilo <Nabble_01394 at slickdeals.endjunk.com
> wrote:

> So do the dots. BTW, using *^(\d+)-* as ReGex will suffice too to pass
> digits
> before the -.
>
> mercutioviz wrote:
> >
> > On Thu, Sep 15, 2011 at 3:19 AM, Steven Ayre &lt;steveayre at gmail.com&gt;
> > wrote:
> >
> >> Change the regex to only pull out the first part into $1. Something like
> >> this:
> >>
> >> <condition field="destination_number"
> >> expression="^(\d+)\-(\d+\.\d+\.\d+\.\d+)$">
> >>
> > FYI, a bare - in a regex does not need to be escaped.
> > -MC
> >
> >
> >>
> >> $1 will be 0114712500
> >> $2 will be 10.3.12.153
> >>
> >> -Steve
> >>
> >>
> >> On 15 September 2011 10:14, Chris Graham &lt;chrisg.lists at gmail.com&gt;
> >> wrote:
> >>
> >>> Hi All,
> >>>
> >>> I have a simple issue I seem to not be making progress on. Due to
> >>> topology
> >>> hiding in one of the headquaters I send a invite as such:
> >>>
> >>> INVITE sip:0114712500-10.3.12.153 at x.x.x.x SIP/2.0.
> >>>
> >>> You can see the first part of the invite is the dialstring then a "-"
> >>> used
> >>> as a delimiter to bill against the hidden IP. FS duly passed this
> >>> dialstring
> >>> on. The carrier then rejects the call.
> >>>
> >>> I want to modify the dialstring in the context below not to dial
> >>> 0114712500-10.3.12.153 but rather just 0114712500. I hope I am clear,
> >>> and
> >>> thanks in advance.
> >>>
> >>> <document type="freeswitch/xml">
> >>>   <section name="dialplan" description="RE FS DP">
> >>>     <context name="lcr_trunks">
> >>>       <extension name="Catch all send to carriers">
> >>>         <condition field="destination_number" expression="^(.*)$">
> >>>           <action application="set" data="nibble_account=10.3.12.153"/>
> >>>           <action application="lcr" data="$1 peak"/>
> >>>           <action application="bridge" data="${lcr_auto_route}"/>
> >>>         </condition>
> >>>       </extension>
> >>>     </context>
> >>>   </section>
> >>> </document>
> >>>
> >>> Chris G
> >>>
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3
> Watts of electricity.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Dialplan-delimiter-transform-tp6796094p6798950.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brtantunes at gmail.com  Fri Sep 16 18:04:07 2011
From: brtantunes at gmail.com (Brtantunes)
Date: Fri, 16 Sep 2011 07:04:07 -0700 (PDT)
Subject: [Freeswitch-users] Timeout Message or something similar
In-Reply-To: <CAKzWOxVmEjdPAKxbzyk2KUZ7DyC7xzga54g1e0s-Ksd3RR1ozg@mail.gmail.com>
References: <1315837848697-6783735.post@n2.nabble.com>
	<CAMwZ-tK6GT7EdtCFdZqNTahfGp1BHt45jBUXt68eGfbKv7TBig@mail.gmail.com>
	<1315907331256-6786830.post@n2.nabble.com>
	<CAKzWOxUszOu8xGn70+UtaCOtqFC=WvEObORE+vWpdFmXKXSLgA@mail.gmail.com>
	<1315991590857-6791788.post@n2.nabble.com>
	<CAKzWOxVmEjdPAKxbzyk2KUZ7DyC7xzga54g1e0s-Ksd3RR1ozg@mail.gmail.com>
Message-ID: <1316181847759-6801030.post@n2.nabble.com>

Hello,

I figured it out. I'm using variables, Right now the command is:
/originate
{ringback=c:/teste.wav}[timeoutMsg=C:/teste2.wav,domain=DOMAIN,maxCallTime=30,ringTimeout=30]sofia/DOMAIN/1001
TRY_1002 XML default/

And in the dialplan i have:
/<context name="myDialPlan">
	<extension name="try to dial an extension">
	  <condition field="destination_number" expression="^TRY_(\d+)$">
		<action application="sched_hangup" data="+${maxCallTime}
allotted_timeout"/>
		<action application="bridge"
data="{fail_on_single_reject=true,ignore_early_media=true,originate_continue_on_timeout=true,ignore_early_media=true,originate_timeout=${ringTimeout}}sofia/${domain}/$1"/>
		<action application="playback" data="${timeoutMsg}"/>
		<action application="hangup"/>
	  </condition>
	</extension>
</context>/

It is working fine.
Now i just want to include the originate in the dialplan and just pass all
parameters like variables, but i'm having problems. Any sugestions? 

Tks for your help,
Greetings

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Timeout-Message-or-something-similar-tp6783735p6801030.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From lfurrea at gmail.com  Fri Sep 16 19:49:35 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Fri, 16 Sep 2011 09:49:35 -0600
Subject: [Freeswitch-users] non-blocking session:streamFile on LUA script
In-Reply-To: <CAB2KLydoKywnEBkYhH-yLzNRVukutGLsNZpxCW87eULD5HeswA@mail.gmail.com>
References: <CAB2KLydoKywnEBkYhH-yLzNRVukutGLsNZpxCW87eULD5HeswA@mail.gmail.com>
Message-ID: <CAB2KLye4PpdsBod=oSTRNSJUyG0ANs7WG7ytVRLYkxWC2_NiOQ@mail.gmail.com>

This is just for an IVR

So that I can start ringing phones immediately and whatever comes first user
DTMF or phones being answered determines where to bridge the call?

Can it be done on a script?

On Fri, Sep 16, 2011 at 1:30 AM, Luis F Urrea <lfurrea at gmail.com> wrote:

> Hello all,
>
> I am trying to write a script in which I need to simultaneously:
>
> 1. Play a file
> 2. Read digits
> 3. Ring a phone
>
> I can play the file an read digits using setInputCallback
>
> But the playing of file blocks for the origination of a b leg using
> freeswitch.Session()
>
> Am I SOL?
>
> TIA
>
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From curriegrad2004 at gmail.com  Fri Sep 16 19:57:33 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 16 Sep 2011 08:57:33 -0700
Subject: [Freeswitch-users] mod_cdr_sqlite schema.sql
In-Reply-To: <ABFDC784-B031-41E5-B8D6-0D616BE9C251@seventhsignal.de>
References: <4E7329CE.1050800@amooma.de>
	<ABFDC784-B031-41E5-B8D6-0D616BE9C251@seventhsignal.de>
Message-ID: <CAMWiMy_eKN+n1biMRP6i2PH-eCpgiCCrY8nnd5pudiUpvm=g8w@mail.gmail.com>

You can always load the sqlite database file and extract the schema from there

On Fri, Sep 16, 2011 at 5:41 AM, Michal Bielicki
<michal.bielicki at seventhsignal.de> wrote:
> It is in the sourcecode since the files with the tables get created if they are not there.
> #
> Am 16.09.2011 um 12:49 schrieb Sascha Daniels:
>
>> Hi together,
>>
>> I would like to write cdr data into the sqlite file of my rails application.
>>
>> According to http://wiki.freeswitch.org/wiki/Mod_cdr I can find the
>> schema.sql in the source directory.
>>
>> Unfortunately it is not there.
>>
>> All needed fields are in the MySQL example, but I need to know what the
>> default table name is.
>>
>> Can someone tell me the table name, so I can generate my migration?
>>
>> Kind regards
>>
>> ? ? ? ? ? ? ? ? ? ? ? ?Sascha
>>
>> --
>> AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied ?--> ?http://www.amooma.de
>> Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister Montabaur B14998
>>
>> B?cher: http://das-asterisk-buch.de - http://ruby-auf-schienen.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
>
> 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
>


From msc at freeswitch.org  Fri Sep 16 20:25:31 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 16 Sep 2011 09:25:31 -0700
Subject: [Freeswitch-users] Timeout Message or something similar
In-Reply-To: <1316181847759-6801030.post@n2.nabble.com>
References: <1315837848697-6783735.post@n2.nabble.com>
	<CAMwZ-tK6GT7EdtCFdZqNTahfGp1BHt45jBUXt68eGfbKv7TBig@mail.gmail.com>
	<1315907331256-6786830.post@n2.nabble.com>
	<CAKzWOxUszOu8xGn70+UtaCOtqFC=WvEObORE+vWpdFmXKXSLgA@mail.gmail.com>
	<1315991590857-6791788.post@n2.nabble.com>
	<CAKzWOxVmEjdPAKxbzyk2KUZ7DyC7xzga54g1e0s-Ksd3RR1ozg@mail.gmail.com>
	<1316181847759-6801030.post@n2.nabble.com>
Message-ID: <CAKzWOxVdS41bgNpTeBUDOTVVvbmzHyypPa8oMYy2cAHKreNwhg@mail.gmail.com>

What do you mean by "include the originate in the dialplan"? The originate
command is what creates the two call legs. Anything in the dialplan requires
a call leg already to exist.

-MC

On Fri, Sep 16, 2011 at 7:04 AM, Brtantunes <brtantunes at gmail.com> wrote:

> Hello,
>
> I figured it out. I'm using variables, Right now the command is:
> /originate
>
> {ringback=c:/teste.wav}[timeoutMsg=C:/teste2.wav,domain=DOMAIN,maxCallTime=30,ringTimeout=30]sofia/DOMAIN/1001
> TRY_1002 XML default/
>
> And in the dialplan i have:
> /<context name="myDialPlan">
>         <extension name="try to dial an extension">
>          <condition field="destination_number" expression="^TRY_(\d+)$">
>                 <action application="sched_hangup" data="+${maxCallTime}
> allotted_timeout"/>
>                <action application="bridge"
>
> data="{fail_on_single_reject=true,ignore_early_media=true,originate_continue_on_timeout=true,ignore_early_media=true,originate_timeout=${ringTimeout}}sofia/${domain}/$1"/>
>                <action application="playback" data="${timeoutMsg}"/>
>                 <action application="hangup"/>
>          </condition>
>        </extension>
> </context>/
>
> It is working fine.
> Now i just want to include the originate in the dialplan and just pass all
> parameters like variables, but i'm having problems. Any sugestions?
>
> Tks for your help,
> Greetings
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Timeout-Message-or-something-similar-tp6783735p6801030.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fs-list at communicatefreely.net  Fri Sep 16 21:10:42 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 16 Sep 2011 13:10:42 -0400
Subject: [Freeswitch-users] Can bind-meta-app hangup b-leg?
Message-ID: <4E738312.3010408@communicatefreely.net>

Hello,

I'm building a little DISA application in lua, and one of the things I 
want to do is give the user the option to hang up on their current call 
and come back into the app so they can make another call without 
re-authenticating.  I was trying to use bind-meta-app to execute hangup 
on the other leg, but it doesn't seem to work.

Here's the important part of my lua script:

function place_call()
        destination = session:playAndGetDigits(3, 14, 1, 3000, 
'#','phrase:disa_enter_number','','\\d+')
        session:execute("bind_meta_app","9 a oi transfer::test-tone XML 
targets")
        freeswitch.consoleLog("info","User ".. user.." calling 
"..destination.." from DISA tool\n")
        session:execute("execute_extension",destination.." XML internal")
end

If I change the flag to be the same leg, *9 hangs up on me, but I want 
it to hang up on the other guy.

I am using execute_extension to run some dialplan that takes care of 
routing, caller ID setup, etc.  It isn't really practical to call a 
bridge from the script, since there is a lot of pre-processing to do, 
and I have it already sorted out in the dialplan.  The function is set 
to loop as long as session:ready() is true, so killing the b-leg should 
bring us back to the top of this function.

Why doesn't this work, or is there a better way to achieve the same result?

Thanks!

-Tim


From fs-list at communicatefreely.net  Fri Sep 16 21:17:22 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 16 Sep 2011 13:17:22 -0400
Subject: [Freeswitch-users] Can bind-meta-app hangup b-leg?
In-Reply-To: <4E738312.3010408@communicatefreely.net>
References: <4E738312.3010408@communicatefreely.net>
Message-ID: <4E7384A2.4050709@communicatefreely.net>

Sorry, looks like I sent the wrong code.  That was just for testing.


function place_call()
        destination = session:playAndGetDigits(3, 14, 1, 3000, 
'#','phrase:disa_enter_number','','\\d+')
        session:execute("bind_meta_app","9 a oi hangup::")
        freeswitch.consoleLog("info","User ".. user.." calling 
"..destination.." from DISA tool\n")
        session:execute("execute_extension",destination.." XML internal")
end


This is what I think should let me hangup on the other leg.


Tim St. Pierre wrote:
> Hello,
>
> I'm building a little DISA application in lua, and one of the things I 
> want to do is give the user the option to hang up on their current call 
> and come back into the app so they can make another call without 
> re-authenticating.  I was trying to use bind-meta-app to execute hangup 
> on the other leg, but it doesn't seem to work.
>
> Here's the important part of my lua script:
>
> function place_call()
>         destination = session:playAndGetDigits(3, 14, 1, 3000, 
> '#','phrase:disa_enter_number','','\\d+')
>         session:execute("bind_meta_app","9 a oi transfer::test-tone XML 
> targets")
>         freeswitch.consoleLog("info","User ".. user.." calling 
> "..destination.." from DISA tool\n")
>         session:execute("execute_extension",destination.." XML internal")
> end
>
> If I change the flag to be the same leg, *9 hangs up on me, but I want 
> it to hang up on the other guy.
>
> I am using execute_extension to run some dialplan that takes care of 
> routing, caller ID setup, etc.  It isn't really practical to call a 
> bridge from the script, since there is a lot of pre-processing to do, 
> and I have it already sorted out in the dialplan.  The function is set 
> to loop as long as session:ready() is true, so killing the b-leg should 
> bring us back to the top of this function.
>
> Why doesn't this work, or is there a better way to achieve the same result?
>
> Thanks!
>
> -Tim
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From shaon.khan at gmail.com  Fri Sep 16 21:22:43 2011
From: shaon.khan at gmail.com (Arafath-uz-zaman khan)
Date: Fri, 16 Sep 2011 23:22:43 +0600
Subject: [Freeswitch-users] No variable name specified
Message-ID: <CACcfM9YQT_9vouA2vqJVbqZpbhyqowQ8otC7CY=Ue0mWAs-BXw@mail.gmail.com>

hello

i tried to use nibblebill, here is my configuration:

*User Directory:*
<include>
  <user id="3001">
    <params>
      <param name="password" value="1234"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="user_context" value="default"/>
      <variable name="nibble_account" value="3001"/>
    </variables>
  </user>
</include>

*Dialplan:*
<include>
  <extension name="Test">
   <condition field="destination_number" expression="880">
    <action application="set" data="nibble_rate=0.035"/>
    <action application="set" data="$${nibble_account}"/>
    <action application="bridge"
data="sofia/gateway/asterisk/${destination_number}"/>
   </condition>
  </extension>

I got following error

2011-09-16 17:18:36.866801 [DEBUG] mod_dptools.c:1167 sofia/external/
3001 at 192.168.7.21 SET [nibble_rate]=[0.035]
EXECUTE sofia/external/3001 at 192.168.7.21 set()
2011-09-16 17:18:36.866801 [ERR] mod_dptools.c:1145 No variable name
specified.

can anyone help me to find the issue, i follwed the *Mod_nibblebill wiki
Single Rate for all users section*.

Regards
shaon
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From larclap at yahoo.com  Fri Sep 16 21:34:50 2011
From: larclap at yahoo.com (Lars Zeb)
Date: Fri, 16 Sep 2011 10:34:50 -0700
Subject: [Freeswitch-users] FollowMe timing on wiki?
Message-ID: <008901cc7496$f0440dd0$d0cc2970$@yahoo.com>

On the Dialplan FollowMe wiki, the two following examples are given:

<action application="bridge"
data="user/1000 at mydomain.com,[leg_delay_start=15,leg_timeout=25]sofia/gatewa
y/flowroute/12345678901" />

   <!-- ring my desk extension for 10 seconds. -->
   <action application="set" data="call_timeout=10"/>
   <action application="bridge" data="sofia/$${domain}/1001"/>
   <!-- Now try my cell phone, hangup before cellphone voicemail picks up.
-->
   <!-- Dial out through my voicepulse gateway -->
   <action application="set" data="call_timeout=13"/>
   <action application="bridge" data="sofia/gateway/voicepulse/16501234567"
/>

Is the call_timeout=13 variable equivalent functionally to the
leg_timeout=25 in the first example?

Thanks, Lars



From rgelfand2 at gmail.com  Fri Sep 16 22:59:45 2011
From: rgelfand2 at gmail.com (Roman Gelfand)
Date: Fri, 16 Sep 2011 14:59:45 -0400
Subject: [Freeswitch-users] ESL Memory Leak
Message-ID: <CAJbW+rmb7zpztb_6fTkN5sKeh+9huJHvxkhDv4_Hz8LVG_N9Qw@mail.gmail.com>

There seem to be reports of ESL memory leak.  If so, has this been fixed?

Thanks in advance
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From yungwei at resolvity.com  Fri Sep 16 23:17:02 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Fri, 16 Sep 2011 15:17:02 -0400
Subject: [Freeswitch-users] sending a call to a queue on a remote FS
Message-ID: <33095823FD21DF429B481B5163264B795118922016@VMBX102.ihostexchange.net>

Hi,

I want to send a call from one FS to a queue on another FS. 
So I'm wondering if that is supported. If so, how can I do that (FS config-wise)? If not, what is the recommended way of doing this? 
Thanks.


From peter.olsson at visionutveckling.se  Sat Sep 17 00:22:53 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 16 Sep 2011 22:22:53 +0200
Subject: [Freeswitch-users] ESL Memory Leak
In-Reply-To: <CAJbW+rmb7zpztb_6fTkN5sKeh+9huJHvxkhDv4_Hz8LVG_N9Qw@mail.gmail.com>
References: <CAJbW+rmb7zpztb_6fTkN5sKeh+9huJHvxkhDv4_Hz8LVG_N9Qw@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF126@cooper>

Any references? I don't know of any memory leaks in ESL for the moment.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Roman Gelfand [rgelfand2 at gmail.com]
Skickat: den 16 september 2011 20:59
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] ESL Memory Leak

There seem to be reports of ESL memory leak.  If so, has this been fixed?

Thanks in advance
!DSPAM:4e739ceb32761772330274!


From buscom123+fs at gmail.com  Sat Sep 17 01:52:58 2011
From: buscom123+fs at gmail.com (R H)
Date: Fri, 16 Sep 2011 15:52:58 -0600
Subject: [Freeswitch-users] mod_managed switch_xml_open_cfg and SIGSEGV
In-Reply-To: <CAKbxfG-vq2XPF6cVFi2v8XZpvJrr_=GOFYejHPh2nC0BKwWffg@mail.gmail.com>
References: <CAAtQP54ec9oTygRGeNdZOvXK7t-U0_79WM_hV9Bc791XDNp5Tg@mail.gmail.com>
	<CAKbxfG-vq2XPF6cVFi2v8XZpvJrr_=GOFYejHPh2nC0BKwWffg@mail.gmail.com>
Message-ID: <CAAtQP54j7Sm_3rGG2vwjRj_ZgDA=e+-e55csdedg9=CZSdoMig@mail.gmail.com>

Thanks for your response Anthony,

I should of mentioned this in my previous email but The method signature
for switch_xml_open_cfg looks like the following:

*public static switch_xml switch_xml_open_cfg(string file_path,
SWIGTYPE_p_p_switch_xml node, switch_event arg2)*

and i have been unable to figure out how to create an instance of *
SWIGTYPE_p_p_switch_xml* to pass into that. As far as I can tell, is not
instantiable unless you write an interface for swig to use before you build.
Perhaps you can give me a hint on how to use that object or is there
something that has to be changed on the mod_managed code itself? Sorry, I am
pretty new to freeswitch and swig so I am not terribly sure why that object
is not available for me to instantiate.

Thank you so much for your help.

Ryan


On Thu, Sep 15, 2011 at 9:26 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you have to supply the 2nd arg too.
>
> the returned obj is the whole xml root, the 2nd param is a pointer to
> pointer of the section you searched for inside that root.
>
>
> On Wed, Sep 14, 2011 at 3:41 PM, R H <buscom123+fs at gmail.com> wrote:
> > Has anyone had success working with xml configuration files using
> > mod_managed? I have several modules that DO NOT override the xml
> > configuration but, rather, I actually want to configure them using the
> > freeswitch conf directory structure. I have not even tried to parse the
> xml
> > because I cant even load the xml. Here is a very basic example of what I
> > tried and the result I got:
> > The Code:
> > ------------------------------------------------
> > using System;
> > using FreeSWITCH;
> > using FreeSWITCH.Native;
> > namespace PsiCallCenterAddon
> > {
> > public class FreeSwitchConfigLoader
> > {
> > public static void loadConfigTest()
> > {
> > try
> > {
> > switch_xml xml = freeswitch.switch_xml_open_cfg("freeswitch.conf", null,
> > null);
> > freeswitch.switch_xml_free(xml);
> > Log.WriteLine(LogLevel.Alert, "Config Loader Success!");
> > }
> > catch
> > {
> > Log.WriteLine(LogLevel.Alert, "Config Loader Fail!");
> > }
> > }
> > }
> > }
> > ------------------------------------------------
> > The Result:
> > ------------------------------------------------
> > Stacktrace:
> >   at (wrapper managed-to-native)
> > FreeSWITCH.Native.freeswitchPINVOKE.switch_xml_open_cfg
> >
> (string,System.Runtime.InteropServices.HandleRef,System.Runtime.InteropServices.HandleRef)
> > <0x000a3>
> >   at (wrapper managed-to-native)
> > FreeSWITCH.Native.freeswitchPINVOKE.switch_xml_open_cfg
> >
> (string,System.Runtime.InteropServices.HandleRef,System.Runtime.InteropServices.HandleRef)
> > <0x000a3>
> >   at FreeSWITCH.Native.freeswitch.switch_xml_open_cfg
> >
> (string,FreeSWITCH.Native.SWIGTYPE_p_p_switch_xml,FreeSWITCH.Native.switch_event)
> > <0x0004f>
> >   at PsiCallCenterAddon.FreeSwitchConfigLoader.loadConfigTest ()
> <0x0001b>
> >   at PsiCallCenterAddon.QueueMonitorLoader.Load () <0x0000b>
> >   at FreeSWITCH.PluginManager.RunLoadNotify (System.Type[]) <0x00190>
> >   at FreeSWITCH.AsmPluginManager.LoadInternal (string) <0x001eb>
> >   at FreeSWITCH.PluginManager.Load (string) <0x000a7>
> >   at (wrapper remoting-invoke-with-check) FreeSWITCH.PluginManager.Load
> > (string) <0x00067>
> >   at (wrapper xdomain-dispatch) FreeSWITCH.PluginManager.Load
> > (object,byte[]&,byte[]&,string) <0x0017b>
> >   at (wrapper xdomain-invoke) FreeSWITCH.PluginManager.Load (string)
> > <0x00126>
> >   at (wrapper remoting-invoke-with-check) FreeSWITCH.PluginManager.Load
> > (string) <0x00047>
> >   at FreeSWITCH.Loader.loadFile (string) <0x004a3>
> >   at FreeSWITCH.Loader.Load () <0x0027b>
> >   at (wrapper runtime-invoke) <Module>.runtime_invoke_bool
> > (object,intptr,intptr,intptr) <0x00046>
> > Native stacktrace:
> >         /usr/lib64/libmono-2.0.so.1(+0xacba0) [0x7fc7ea58dba0]
> >         /usr/lib64/libmono-2.0.so.1(+0xfcedf) [0x7fc7ea5ddedf]
> >         /lib64/libpthread.so.0(+0xf2d0) [0x7fc7f1ceb2d0]
> >
> > /usr/local/freeswitch/lib/libfreeswitch.so.1(switch_xml_open_cfg+0x1a)
> > [0x7fc7f260b0ea]
> >         [0x40155804]
> > Debug info from gdb:
> > =================================================================
> > Got a SIGSEGV while executing native code. This usually indicates
> > a fatal error in the mono runtime or one of the native libraries
> > used by your application.
> > =================================================================
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Sat Sep 17 02:04:51 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 16 Sep 2011 17:04:51 -0500
Subject: [Freeswitch-users] mod_managed switch_xml_open_cfg and SIGSEGV
In-Reply-To: <CAAtQP54j7Sm_3rGG2vwjRj_ZgDA=e+-e55csdedg9=CZSdoMig@mail.gmail.com>
References: <CAAtQP54ec9oTygRGeNdZOvXK7t-U0_79WM_hV9Bc791XDNp5Tg@mail.gmail.com>
	<CAKbxfG-vq2XPF6cVFi2v8XZpvJrr_=GOFYejHPh2nC0BKwWffg@mail.gmail.com>
	<CAAtQP54j7Sm_3rGG2vwjRj_ZgDA=e+-e55csdedg9=CZSdoMig@mail.gmail.com>
Message-ID: <CAKbxfG939qUeOD97QLmp3jKcra1-qNrubCpUg2f0xzjE3Mt5Vg@mail.gmail.com>

I dont use managed stuff but in c you would have

switch_xml_t xml, cfg;

 xml = switch_xml_open_cfg("freeswitch.conf", &cfg, NULL);

cfg is the same type of xml a node inside the xml pointed in "xml"

it always returns the whole root and then the area you are searching for.

When you are done you release the xml pointer and the cfg will also be released.



On Fri, Sep 16, 2011 at 4:52 PM, R H <buscom123+fs at gmail.com> wrote:
> Thanks for your response Anthony,
> I should of mentioned this in my previous email but?The method signature
> for?switch_xml_open_cfg looks like the following:
>
> public static switch_xml switch_xml_open_cfg(string file_path,
> SWIGTYPE_p_p_switch_xml node, switch_event arg2)
>
> and i have been unable to figure out how to create an instance
> of?SWIGTYPE_p_p_switch_xml to pass into that.?As far as I can tell, is not
> instantiable unless you write an interface for swig to use before you build.
> Perhaps you can give me a hint on how to use that object or is there
> something that has to be changed on the mod_managed code itself? Sorry, I am
> pretty new to freeswitch and swig so I am not terribly sure why that object
> is not available for me to instantiate.
> Thank you so much for your help.
> Ryan
>
> On Thu, Sep 15, 2011 at 9:26 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> you have to supply the 2nd arg too.
>>
>> the returned obj is the whole xml root, the 2nd param is a pointer to
>> pointer of the section you searched for inside that root.
>>
>>
>> On Wed, Sep 14, 2011 at 3:41 PM, R H <buscom123+fs at gmail.com> wrote:
>> > Has anyone had success working with xml configuration files using
>> > mod_managed? I have several modules that DO NOT override the xml
>> > configuration but, rather, I actually want to configure them using the
>> > freeswitch conf directory structure.?I have not even tried to?parse the
>> > xml
>> > because I cant even load the xml.?Here is a very basic example of what I
>> > tried and the result I got:
>> > The Code:
>> > ------------------------------------------------
>> > using System;
>> > using FreeSWITCH;
>> > using FreeSWITCH.Native;
>> > namespace PsiCallCenterAddon
>> > {
>> > public class FreeSwitchConfigLoader
>> > {
>> > public static void loadConfigTest()
>> > {
>> > try
>> > {
>> > switch_xml xml = freeswitch.switch_xml_open_cfg("freeswitch.conf", null,
>> > null);
>> > freeswitch.switch_xml_free(xml);
>> > Log.WriteLine(LogLevel.Alert, "Config Loader Success!");
>> > }
>> > catch
>> > {
>> > Log.WriteLine(LogLevel.Alert, "Config Loader Fail!");
>> > }
>> > }
>> > }
>> > }
>> > ------------------------------------------------
>> > The Result:
>> > ------------------------------------------------
>> > Stacktrace:
>> > ? at (wrapper managed-to-native)
>> > FreeSWITCH.Native.freeswitchPINVOKE.switch_xml_open_cfg
>> >
>> > (string,System.Runtime.InteropServices.HandleRef,System.Runtime.InteropServices.HandleRef)
>> > <0x000a3>
>> > ? at (wrapper managed-to-native)
>> > FreeSWITCH.Native.freeswitchPINVOKE.switch_xml_open_cfg
>> >
>> > (string,System.Runtime.InteropServices.HandleRef,System.Runtime.InteropServices.HandleRef)
>> > <0x000a3>
>> > ? at FreeSWITCH.Native.freeswitch.switch_xml_open_cfg
>> >
>> > (string,FreeSWITCH.Native.SWIGTYPE_p_p_switch_xml,FreeSWITCH.Native.switch_event)
>> > <0x0004f>
>> > ? at PsiCallCenterAddon.FreeSwitchConfigLoader.loadConfigTest ()
>> > <0x0001b>
>> > ? at PsiCallCenterAddon.QueueMonitorLoader.Load () <0x0000b>
>> > ? at FreeSWITCH.PluginManager.RunLoadNotify (System.Type[]) <0x00190>
>> > ? at FreeSWITCH.AsmPluginManager.LoadInternal (string) <0x001eb>
>> > ? at FreeSWITCH.PluginManager.Load (string) <0x000a7>
>> > ? at (wrapper remoting-invoke-with-check) FreeSWITCH.PluginManager.Load
>> > (string) <0x00067>
>> > ? at (wrapper xdomain-dispatch) FreeSWITCH.PluginManager.Load
>> > (object,byte[]&,byte[]&,string) <0x0017b>
>> > ? at (wrapper xdomain-invoke) FreeSWITCH.PluginManager.Load (string)
>> > <0x00126>
>> > ? at (wrapper remoting-invoke-with-check) FreeSWITCH.PluginManager.Load
>> > (string) <0x00047>
>> > ? at FreeSWITCH.Loader.loadFile (string) <0x004a3>
>> > ? at FreeSWITCH.Loader.Load () <0x0027b>
>> > ? at (wrapper runtime-invoke) <Module>.runtime_invoke_bool
>> > (object,intptr,intptr,intptr) <0x00046>
>> > Native stacktrace:
>> > ? ? ? ? /usr/lib64/libmono-2.0.so.1(+0xacba0) [0x7fc7ea58dba0]
>> > ? ? ? ? /usr/lib64/libmono-2.0.so.1(+0xfcedf) [0x7fc7ea5ddedf]
>> > ? ? ? ? /lib64/libpthread.so.0(+0xf2d0) [0x7fc7f1ceb2d0]
>> >
>> > /usr/local/freeswitch/lib/libfreeswitch.so.1(switch_xml_open_cfg+0x1a)
>> > [0x7fc7f260b0ea]
>> > ? ? ? ? [0x40155804]
>> > Debug info from gdb:
>> > =================================================================
>> > Got a SIGSEGV while executing native code. This usually indicates
>> > a fatal error in the mono runtime or one of the native libraries
>> > used by your application.
>> > =================================================================
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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 cchirag at gmail.com  Sat Sep 17 01:24:06 2011
From: cchirag at gmail.com (cchhat01)
Date: Fri, 16 Sep 2011 14:24:06 -0700 (PDT)
Subject: [Freeswitch-users] FreeSwitch with Googlevoice (dingaling) on
	OpenWrt based Router
Message-ID: <1316208246493-6802276.post@n2.nabble.com>

Hi All,

I am in the process of getting the freeswitch installation up and running on
my router (Netgear WNDR3700v2) which runs trunk Openwrt trunk (attitude
adjustment). 
This is freeswitch 1.0.7.
0. dingaling.conf.xml was not being copied for my installation type and is
now addressed (thanks to mazilo for committing this checkin)
1. once started, fs_cli shows "sofia status" with the IP address of my WAN
IP Address (207.a.b.c) and not that of the router (192.168.1.1). Digging
through the config, i found that $${local_ip_v4} results in this value, even
though it should really be 192.168.1.1. Is there an option which disables
running FS on the public IP? As a result of this problem, I can't even get
my SIP ATA device to register. Once I manually change the configuration to
use the domain="192.168.1.1" instead of "${local_ip_v4}", my ATA clients can
register.

2. Once i copied over the dingaling.conf.xml file, i was successfully
connected to gtalk account, however all my inbound calls did not route to
the sofia directory user 1000 (as specified in my
jingle_profiles/client.xml).
This file is a replica of the info found on
http://wiki.freeswitch.org/wiki/Google_Voice#Setup_FreeSWITCH_-_Dingaling_to_work_with_your_Gmail_account
with my gtalk login details.

I need these two aspects to start communicating with each other.
Please advise if anyone has any ideas as to what needs to be done.
Thanks,

Chirag

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-with-Googlevoice-dingaling-on-OpenWrt-based-Router-tp6802276p6802276.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Sat Sep 17 02:49:02 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 16 Sep 2011 17:49:02 -0500
Subject: [Freeswitch-users] FreeSwitch with Googlevoice (dingaling) on
 OpenWrt based Router
In-Reply-To: <1316208246493-6802276.post@n2.nabble.com>
References: <1316208246493-6802276.post@n2.nabble.com>
Message-ID: <CAKbxfG8fqtG1rxuPQYuTwz-yYAiA1ycK+QCjDgS3MqOkhx2qnA@mail.gmail.com>

set local_ip_v4 manually in vars.xml at the top to override what the
default tries to guess (it will always go for the first interface that
can reach the internet unless you override it)


On Fri, Sep 16, 2011 at 4:24 PM, cchhat01 <cchirag at gmail.com> wrote:
> Hi All,
>
> I am in the process of getting the freeswitch installation up and running on
> my router (Netgear WNDR3700v2) which runs trunk Openwrt trunk (attitude
> adjustment).
> This is freeswitch 1.0.7.
> 0. dingaling.conf.xml was not being copied for my installation type and is
> now addressed (thanks to mazilo for committing this checkin)
> 1. once started, fs_cli shows "sofia status" with the IP address of my WAN
> IP Address (207.a.b.c) and not that of the router (192.168.1.1). Digging
> through the config, i found that $${local_ip_v4} results in this value, even
> though it should really be 192.168.1.1. Is there an option which disables
> running FS on the public IP? As a result of this problem, I can't even get
> my SIP ATA device to register. Once I manually change the configuration to
> use the domain="192.168.1.1" instead of "${local_ip_v4}", my ATA clients can
> register.
>
> 2. Once i copied over the dingaling.conf.xml file, i was successfully
> connected to gtalk account, however all my inbound calls did not route to
> the sofia directory user 1000 (as specified in my
> jingle_profiles/client.xml).
> This file is a replica of the info found on
> http://wiki.freeswitch.org/wiki/Google_Voice#Setup_FreeSWITCH_-_Dingaling_to_work_with_your_Gmail_account
> with my gtalk login details.
>
> I need these two aspects to start communicating with each other.
> Please advise if anyone has any ideas as to what needs to be done.
> Thanks,
>
> Chirag
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-with-Googlevoice-dingaling-on-OpenWrt-based-Router-tp6802276p6802276.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 admin at blindi.net  Sat Sep 17 06:11:18 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Sat, 17 Sep 2011 04:11:18 +0200 (CEST)
Subject: [Freeswitch-users] Problem send_dtmf send a aleg
In-Reply-To: <C2CAC53C1A8B4CB3ADC3B5EF97F56DDD@e1705>
References: <EBED8C31D06F4A319D978371EE1CF24E@e1705>
	<AANLkTi=AVwJ4c80z9F_06+8EyUHrokLzEtRcPGGpa4jG@mail.gmail.com>
	<C2CAC53C1A8B4CB3ADC3B5EF97F56DDD@e1705>
Message-ID: <Pine.LNX.4.64.1109170400420.9960@tomback.antagus.de>

Hi all,
I like to create a extension to make a dtmf login to my
voicemailbox on my cellphone.

Fs sends only dtmf tones to myself, not to the  phonenumber.
Here is my extension:
<extension name="remote voicemaillogin to my cellphone" continue="true">
<condition field="destination_number" expression="^701$">
<action application="send_dtmf" data="*312666 at 100"/>
<action application="bridge" data="loopback/0174559909990/disa"/>

</condition>
</extension>
Can you help me please?
thanks.


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From jeff at jefflenk.com  Sat Sep 17 10:31:49 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 16 Sep 2011 23:31:49 -0700 (PDT)
Subject: [Freeswitch-users] mod_managed switch_xml_open_cfg and SIGSEGV
In-Reply-To: <CAAtQP54j7Sm_3rGG2vwjRj_ZgDA=e+-e55csdedg9=CZSdoMig@mail.gmail.com>
References: <CAAtQP54ec9oTygRGeNdZOvXK7t-U0_79WM_hV9Bc791XDNp5Tg@mail.gmail.com>
	<CAKbxfG-vq2XPF6cVFi2v8XZpvJrr_=GOFYejHPh2nC0BKwWffg@mail.gmail.com>
	<CAAtQP54j7Sm_3rGG2vwjRj_ZgDA=e+-e55csdedg9=CZSdoMig@mail.gmail.com>
Message-ID: <1316241109104-6803332.post@n2.nabble.com>

These SWIG references can be quite difficult and cryptic but you can get used
to them if you spend enough time. Look in eventbinding.cs in mod_managed for
several examples.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-managed-switch-xml-open-cfg-and-SIGSEGV-tp6794420p6803332.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From joe.jflemmings at gmail.com  Sat Sep 17 11:42:49 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Sat, 17 Sep 2011 00:42:49 -0700
Subject: [Freeswitch-users] Say Euros instead of Dollars
Message-ID: <CALH5ZrZ6giwx1BaLjrRwAFKN2KZ=QZ-9CB+1EbcfrjQAb81XRg@mail.gmail.com>

How can i make freeswitch Say applications to  say Euros instead of dollars

eg

<action application="say" data="en CURRENCY PRONOUNCED 1.96"/>

will by default say dollars

Joe
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From joe.jflemmings at gmail.com  Sat Sep 17 11:44:11 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Sat, 17 Sep 2011 00:44:11 -0700
Subject: [Freeswitch-users] Sophia authentication without using editing
	acl.conf.xml
In-Reply-To: <CA+GmxbYRhaLOZm7ami4zEa9nkMYESE24baK-13vbXdzcH-3PrA@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>
	<CA+GmxbYRhaLOZm7ami4zEa9nkMYESE24baK-13vbXdzcH-3PrA@mail.gmail.com>
Message-ID: <CALH5ZrYpQrnGd3kB_zY4+j_ct0QbmJHPKR3FEUP5qLeyA+RWkQ@mail.gmail.com>

Is their a solution without reloading acl?

On Fri, Sep 9, 2011 at 4:16 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> 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
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Nabble_01394 at slickdeals.endjunk.com  Sat Sep 17 17:52:48 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Sat, 17 Sep 2011 06:52:48 -0700 (PDT)
Subject: [Freeswitch-users] FreeSwitch with Googlevoice (dingaling) on
	OpenWrt based Router
In-Reply-To: <1316235342445-6803177.post@n2.nabble.com>
References: <1316208246493-6802276.post@n2.nabble.com>
	<CAKbxfG8fqtG1rxuPQYuTwz-yYAiA1ycK+QCjDgS3MqOkhx2qnA@mail.gmail.com>
	<1316235342445-6803177.post@n2.nabble.com>
Message-ID: <1316267568824-6803939.post@n2.nabble.com>


cchhat01 wrote:
> i then checked the cli to see whats happening with the inbound and
> outbound calls, turned out it was an issue with the:
> 
> external_rtp_ip=stun:stun.freeswitch.org
> external_sip_ip=stun:stun.freeswitch.org
> 
> changed this to 
> 
> external_rtp_ip=stun:stunserver.org
> external_sip_ip=stun:stunserver.org
> 
> SUCCESS!
AFAICR, stun.freeswitch.org has been defunct for a few weeks now.


1. When I dial a number, I don't hear the ringing as I did with my asterisk
setup. Do i need to make some changes to hear the "ringing" sound. All i
hear is silence until the other other party picks up the phone.
Take a closer look at the dialplan shown in the FS Wiki on 
http://wiki.freeswitch.org/wiki/Google_Voice Make Free Outgoing Calls to
USA/Canada using Google Voice , it uses /ring_ready/ to generate ring tones
during a GV outbound call.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-with-Googlevoice-dingaling-on-OpenWrt-based-Router-tp6802276p6803939.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From avi at avimarcus.net  Sun Sep 18 09:37:31 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 18 Sep 2011 08:37:31 +0300
Subject: [Freeswitch-users] 302 From phone / redirect not working
Message-ID: <CALNZuwnR2GckNNdN9un7pJbbhZw+HfwfH7nbNgxLmJ4utT1hCA@mail.gmail.com>

I set up "forwarding" from a Yealink sip-t22 and I know it worked before.
I'm not sure if something changed on the phone or on FS or in my dialplan,
but it's attempts to froward don't work anymore.

Pcap of the yealink's response with 302: http://ge.tt/9bnNXs7
FS Calllog: http://pastebin.freeswitch.org/17359

Callflow: ext 1000->pstn -> incoming via 410 number -> ext 1105s1106 ->
rings to user/1105 :_: user/1106 -> 1105 sends the 302 -> FS ends up with: "

   1. 67230312-de1d-11e0-aecd-83184b5ae345 2011-09-13 18:31:09.460091 [DEBUG
   ] sofia.c:6093 Process REFER to [0548481732 at 178.79.147.47]
   2. 67230312-de1d-11e0-aecd-83184b5ae345 2011-09-13 18:31:09.460091 [ERR]
    sofia.c:6537 Cannot Blind Transfer 1 Legged calls"

The 302 is SUPPOSED to be hitting the dialplan (under their accountcode) - I
know it used to. Did I fudge something with the redirect settings?

Thanks!!

-Avi
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From dujinfang at gmail.com  Sun Sep 18 12:21:25 2011
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 18 Sep 2011 16:21:25 +0800
Subject: [Freeswitch-users] New Chinese sound files for review
In-Reply-To: <4E71E727.40302@coppice.org>
References: <CAKzWOxVnHsP8Q8c-ivTCxAFVV2y8K7bGh0MsCL9EgjNn0Fdbag@mail.gmail.com>
	<CAMWiMy-VoB2UXBpXWjZrAj90sg9B=8oueBpYUnTf2cMnQNDGKA@mail.gmail.com>
	<4E71E727.40302@coppice.org>
Message-ID: <3FA6356B04EC42C386FF033BE568E6EA@gmail.com>

Thanks all.


On Thursday, September 15, 2011 at 7:53 PM, Steve Underwood wrote:

> It looks like these prompts were produced without looking at what the  
> Chinese say module currently uses.
>  
I simply translated the phrase_en.xml, so there's some miss match.

> On 09/15/2011 02:20 PM, curriegrad2004 wrote:
> > I'd say the mandarin sounds can be placed on the mainline git repository.
> >  
> > However, dollar.wav and dollars.wav can be symlinked together as the
> > Mandarin language doesn't have the concept of plural tenses. The same
> > goes for cent.wav and cents.wav.
> They shouldn't be symlinked. They should be deleted.  
I think leave them alone should be fine for now(in case we may use?). It should not be symlinked as windows don't have symlink.

> The Chinese say  
> module doesn't use them. Also, the say module use the two Chinese senses  
> of "2". There should be a file called time/2s.wav with ? in it.
will look into this.  
> > On Wed, Sep 14, 2011 at 9:33 AM, Michael Collins<msc at freeswitch.org (mailto:msc at freeswitch.org)> wrote:
> > > To all Chinese speakers:
> > > Seven Du has graciously put together new zh_CN sound files:
> > > ftp://ftp.x-y-t.com/pub/freeswitch-sounds-zh-cn-link-8000-1.0.0.tar.gz
> > > ftp://ftp.x-y-t.com/pub/freeswitch-sounds-zh-cn-link-8000-1.0.0.tar.gz.md5
> > > Here's a jira for the phrase xml:
> > > http://jira.freeswitch.org/browse/FS-3568
> > > If you understand Chinese then please consider helping Seven Du out by
> > > downloading and listening to the sounds.
> > > Thanks!
> > > -Michael
> Steve
>  
>  
>  
> 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

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From curriegrad2004 at gmail.com  Sun Sep 18 19:13:13 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 18 Sep 2011 08:13:13 -0700
Subject: [Freeswitch-users] Say Euros instead of Dollars
In-Reply-To: <CALH5ZrZ6giwx1BaLjrRwAFKN2KZ=QZ-9CB+1EbcfrjQAb81XRg@mail.gmail.com>
References: <CALH5ZrZ6giwx1BaLjrRwAFKN2KZ=QZ-9CB+1EbcfrjQAb81XRg@mail.gmail.com>
Message-ID: <CAMWiMy8v-o8ruJEDe784tTxhZn=fQVqtTifxC8yv13eyRqY6bw@mail.gmail.com>

You can replace the dollars.wav's conents with the phrase euros

On Sat, Sep 17, 2011 at 12:42 AM, Joe Flemmings
<joe.jflemmings at gmail.com> wrote:
> How can i make freeswitch Say applications to? say Euros instead of dollars
>
> eg
>
> <action application="say" data="en CURRENCY PRONOUNCED 1.96"/>
>
> will by default say dollars
>
> 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
>
>


From brian at freeswitch.org  Sun Sep 18 21:58:24 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 18 Sep 2011 12:58:24 -0500
Subject: [Freeswitch-users] 302 From phone / redirect not working
In-Reply-To: <CALNZuwnR2GckNNdN9un7pJbbhZw+HfwfH7nbNgxLmJ4utT1hCA@mail.gmail.com>
References: <CALNZuwnR2GckNNdN9un7pJbbhZw+HfwfH7nbNgxLmJ4utT1hCA@mail.gmail.com>
Message-ID: <0C174237-49E0-42C0-9FAF-B2A4C551EE15@freeswitch.org>

the context on redirected calls will be "redirected" if you're really being redirected manually.

/b

On Sep 18, 2011, at 12:37 AM, Avi Marcus wrote:

> I set up "forwarding" from a Yealink sip-t22 and I know it worked before.
> I'm not sure if something changed on the phone or on FS or in my dialplan,
> but it's attempts to froward don't work anymore.
> 
> Pcap of the yealink's response with 302: http://ge.tt/9bnNXs7
> FS Calllog: http://pastebin.freeswitch.org/17359
> 
> Callflow: ext 1000->pstn -> incoming via 410 number -> ext 1105s1106 ->
> rings to user/1105 :_: user/1106 -> 1105 sends the 302 -> FS ends up with: "
> 
>   1. 67230312-de1d-11e0-aecd-83184b5ae345 2011-09-13 18:31:09.460091 [DEBUG
>   ] sofia.c:6093 Process REFER to [0548481732 at 178.79.147.47]
>   2. 67230312-de1d-11e0-aecd-83184b5ae345 2011-09-13 18:31:09.460091 [ERR]
>    sofia.c:6537 Cannot Blind Transfer 1 Legged calls"
> 
> The 302 is SUPPOSED to be hitting the dialplan (under their accountcode) - I
> know it used to. Did I fudge something with the redirect settings?
> 
> Thanks!!
> 
> -Avi

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From chrisbware at interfree.it  Sun Sep 18 22:44:40 2011
From: chrisbware at interfree.it (Chrisbware)
Date: Sun, 18 Sep 2011 20:44:40 +0200
Subject: [Freeswitch-users] Lua Call Control ring back
Message-ID: <4E763C18.9000208@interfree.it>

Hi guys,

testing Lua script made by MSC calle Call Control (you can find it on
wiki, http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control) ,
everything works perfectly.
Just one issue is still open on my tests: when LegA answers, there's no
ringback.

I'd like to use ringback coming from my provider in early media. How can
I do it?

I've tried setting ignore_early_media=true/false on both legs without
success.
In my case both Leg A and B are calls through my sip provider.

I've tried simplify the script in this way:

legA = freeswitch.Session(slegA);
while (Adisp~="ANSWER" and legA:ready()) do
   Adisp  = legA:getVariable("endpoint_disposition");
   os.execute("sleep 1")
end -- waiting for A to answer

if (Adisp=="ANSWER") then
-- Call LegB
     legB = freeswitch.Session(slegB);
     freeswitch.consoleLog("NOTICE","Fine while B\n");
else
     freeswitch.consoleLog("NOTICE","LegA down!\n");
end

freeswitch.bridge(legA,legB)


and I can hear the ringback! Is there someting on while-do cycle
which block ringback?

Than ks in advance for you help.

Chris








From brad at tech21.com  Sun Sep 18 23:37:57 2011
From: brad at tech21.com (Brad Mina)
Date: Sun, 18 Sep 2011 12:37:57 -0700
Subject: [Freeswitch-users] sending a call to a queue on a remote FS
In-Reply-To: <33095823FD21DF429B481B5163264B795118922016@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B795118922016@VMBX102.ihostexchange.net>
Message-ID: <CAGRr059JtMhW0DDqpmp+GKiVbfhJd+wUVZmMDs_ef=f_8=u2cw@mail.gmail.com>

bridge sofia/external/5900 at remote.ip.add.ress ?

On Fri, Sep 16, 2011 at 12:17 PM, Yungwei Chen <yungwei at resolvity.com>wrote:

> Hi,
>
> I want to send a call from one FS to a queue on another FS.
> So I'm wondering if that is supported. If so, how can I do that (FS
> config-wise)? If not, what is the recommended way of doing this?
> Thanks.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ljjimenez at gmail.com  Sun Sep 18 23:48:28 2011
From: ljjimenez at gmail.com (Luis Jimenez)
Date: Sun, 18 Sep 2011 15:48:28 -0400
Subject: [Freeswitch-users] uuid_broadcast not working
Message-ID: <CAKUOwoH0B1830c-QJj0v+JjZUC9E_d=yJ+BWhvOfN3b9zd2bGQ@mail.gmail.com>

Hello, list

I have this lua script, i try to put music to the aleg while do another
task, but it is not working (not audio at all), any advice on what am doing
wrong?

THanks in advance, here my script:

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

require 'socket'
local host = "127.0.0.1"
local port = 8021

function set_hold(uuid)
    local conn = socket.connect(host, port)

    if (conn) then
        local cmd = "bgapi uuid_broadcast " .. uuid .. "
/opt/freeswitch/sounds/es/dm/dm-promo-recharge-hold.G729 aleg";
        freeswitch.consoleLog("info", "CMD: " .. cmd .. "\n")
        conn:send(string.format("auth
ClueCon\r\n\r\n%s\r\n\r\nexit\r\n\r\n", cmd))
        line, err = conn:receive()
        while (not err) do
            if (line ~= nil) then freeswitch.consoleLog("info", "SCKT: " ..
line .. "\n") end
            line, err = conn:receive()
        end
        conn:close()
    else
        debug.notice("Error posting to server: " .. host .. ":" .. port)
    end
end

function sleep(n)
   os.execute("sleep " .. tonumber(n))
end

session:answer()
session:sleep(500);
uuid = session:get_uuid();
freeswitch.consoleLog("info", "UUID: " .. uuid .. "\n");

api = freeswitch.API();

set_hold(uuid);
freeswitch.consoleLog("info", " - - - RETURNED - - -\n");


for i=1,50 do
    if not session:ready() then break end
    sleep(1);
end

if session:ready() then api:executeString("uuid_break" .. uuid); end
freeswitch.consoleLog("info", " - - - FINNISH COUNTING - - -\n");

session:hangup();

----------------------------------------------------------------------------------
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From Nabble_01394 at slickdeals.endjunk.com  Mon Sep 19 04:30:15 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Sun, 18 Sep 2011 17:30:15 -0700 (PDT)
Subject: [Freeswitch-users] FreeSwitch with Googlevoice (dingaling) on
	OpenWrt based Router
In-Reply-To: <1316292307191-6804611.post@n2.nabble.com>
References: <1316208246493-6802276.post@n2.nabble.com>
	<CAKbxfG8fqtG1rxuPQYuTwz-yYAiA1ycK+QCjDgS3MqOkhx2qnA@mail.gmail.com>
	<1316235342445-6803177.post@n2.nabble.com>
	<1316267568824-6803939.post@n2.nabble.com>
	<1316292307191-6804611.post@n2.nabble.com>
Message-ID: <1316392215467-6806906.post@n2.nabble.com>


cchhat01 wrote:
> Now I have a working freeswitch install on my router..
> Sweeeeeet.
I am glad to hear that you finally made it to work.


If there is any way to set my gtalk status as "away" or "invisible" while on
freeswitch, I would like to look into enabling that.
You are way ahead of me on this. Hopefully, FS experts here may be able to
chime in.


I wonder if anyone else has FreeSwitch installed and working on the same
model router as mine (Netgear WNDR3700v2).
I lurked around on this mailing list and notice posts related to FS hosted
on a Linux embedded device are almost non-existence. AFAICT, most FS users
here are using a multi-core X86 platform to host their FS system. Since my
FS system is for personal and/or SOHO uses only, it certainly will be more
appropriate to have my FS system hosted on a Linux embedded system that
consumes no more than 3Watts of electricity.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-with-Googlevoice-dingaling-on-OpenWrt-based-Router-tp6802276p6806906.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From joe.jflemmings at gmail.com  Mon Sep 19 08:18:15 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Sun, 18 Sep 2011 21:18:15 -0700
Subject: [Freeswitch-users] Say Euros instead of Dollars
In-Reply-To: <CAMWiMy8v-o8ruJEDe784tTxhZn=fQVqtTifxC8yv13eyRqY6bw@mail.gmail.com>
References: <CALH5ZrZ6giwx1BaLjrRwAFKN2KZ=QZ-9CB+1EbcfrjQAb81XRg@mail.gmail.com>
	<CAMWiMy8v-o8ruJEDe784tTxhZn=fQVqtTifxC8yv13eyRqY6bw@mail.gmail.com>
Message-ID: <CALH5ZrabOuAzm67NxXBHoDLB9jtg6D5=VEtdZ1G7XLQPiE8_=A@mail.gmail.com>

I want the ability to use both. That is not a good solution buddy.

On Sun, Sep 18, 2011 at 8:13 AM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> You can replace the dollars.wav's conents with the phrase euros
>
> On Sat, Sep 17, 2011 at 12:42 AM, Joe Flemmings
> <joe.jflemmings at gmail.com> wrote:
> > How can i make freeswitch Say applications to  say Euros instead of
> dollars
> >
> > eg
> >
> > <action application="say" data="en CURRENCY PRONOUNCED 1.96"/>
> >
> > will by default say dollars
> >
> > 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
> >
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From curriegrad2004 at gmail.com  Mon Sep 19 08:29:34 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 18 Sep 2011 21:29:34 -0700
Subject: [Freeswitch-users] Say Euros instead of Dollars
In-Reply-To: <CALH5ZrabOuAzm67NxXBHoDLB9jtg6D5=VEtdZ1G7XLQPiE8_=A@mail.gmail.com>
References: <CALH5ZrZ6giwx1BaLjrRwAFKN2KZ=QZ-9CB+1EbcfrjQAb81XRg@mail.gmail.com>
	<CAMWiMy8v-o8ruJEDe784tTxhZn=fQVqtTifxC8yv13eyRqY6bw@mail.gmail.com>
	<CALH5ZrabOuAzm67NxXBHoDLB9jtg6D5=VEtdZ1G7XLQPiE8_=A@mail.gmail.com>
Message-ID: <CAMWiMy9E7-UF=Db3rQP4C_+=VcXMi6gt1o3G1dEOooeqNYuiRQ@mail.gmail.com>

The other solution is to edit the phrase file that does this. It's
under the lang/en/dir directory in the default dialplan.

On Sun, Sep 18, 2011 at 9:18 PM, Joe Flemmings <joe.jflemmings at gmail.com> wrote:
> I want the ability to use both. That is not a good solution buddy.
>
> On Sun, Sep 18, 2011 at 8:13 AM, curriegrad2004 <curriegrad2004 at gmail.com>
> wrote:
>>
>> You can replace the dollars.wav's conents with the phrase euros
>>
>> On Sat, Sep 17, 2011 at 12:42 AM, Joe Flemmings
>> <joe.jflemmings at gmail.com> wrote:
>> > How can i make freeswitch Say applications to? say Euros instead of
>> > dollars
>> >
>> > eg
>> >
>> > <action application="say" data="en CURRENCY PRONOUNCED 1.96"/>
>> >
>> > will by default say dollars
>> >
>> > 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
>> >
>> >
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 joe.jflemmings at gmail.com  Mon Sep 19 08:40:46 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Sun, 18 Sep 2011 21:40:46 -0700
Subject: [Freeswitch-users] Say Euros instead of Dollars
In-Reply-To: <CAMWiMy9E7-UF=Db3rQP4C_+=VcXMi6gt1o3G1dEOooeqNYuiRQ@mail.gmail.com>
References: <CALH5ZrZ6giwx1BaLjrRwAFKN2KZ=QZ-9CB+1EbcfrjQAb81XRg@mail.gmail.com>
	<CAMWiMy8v-o8ruJEDe784tTxhZn=fQVqtTifxC8yv13eyRqY6bw@mail.gmail.com>
	<CALH5ZrabOuAzm67NxXBHoDLB9jtg6D5=VEtdZ1G7XLQPiE8_=A@mail.gmail.com>
	<CAMWiMy9E7-UF=Db3rQP4C_+=VcXMi6gt1o3G1dEOooeqNYuiRQ@mail.gmail.com>
Message-ID: <CALH5ZrZXG_NAzkzBgc9W6f=M-tV9OwzPp2wAFwWLSPnkXx1h9A@mail.gmail.com>

I dont think the say application bu default uses the phrase xml.

i'm actually using phrase xml and by default when currency is used it says
dollars at the end.



On Sun, Sep 18, 2011 at 9:29 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> The other solution is to edit the phrase file that does this. It's
> under the lang/en/dir directory in the default dialplan.
>
> On Sun, Sep 18, 2011 at 9:18 PM, Joe Flemmings <joe.jflemmings at gmail.com>
> wrote:
> > I want the ability to use both. That is not a good solution buddy.
> >
> > On Sun, Sep 18, 2011 at 8:13 AM, curriegrad2004 <
> curriegrad2004 at gmail.com>
> > wrote:
> >>
> >> You can replace the dollars.wav's conents with the phrase euros
> >>
> >> On Sat, Sep 17, 2011 at 12:42 AM, Joe Flemmings
> >> <joe.jflemmings at gmail.com> wrote:
> >> > How can i make freeswitch Say applications to  say Euros instead of
> >> > dollars
> >> >
> >> > eg
> >> >
> >> > <action application="say" data="en CURRENCY PRONOUNCED 1.96"/>
> >> >
> >> > will by default say dollars
> >> >
> >> > 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
> >> >
> >> >
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From cvogel at lyonl.com  Mon Sep 19 08:42:21 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Mon, 19 Sep 2011 04:42:21 +0000
Subject: [Freeswitch-users] Dial plan help
Message-ID: <8D04992C-91A6-4B8F-9424-6CFC901B8B37@lyonl.com>

I'm trying to build a dialplan however it doesn't seem to work. on the first number I would like to forward and all other numbers i would like to play an an audio file.

  <include>
    <extension name="14142211800">
      <condition field="destination_number" expression=""^+14142211800$">
        <action application="bridge" data="sofia/gateway/level3/+15618911806"/>
      </condition>
    </extension>
    <extension name="13127567001">
      <condition field="destination_number" expression=""^+(1{0,1}\d{10})$">
        <action application="answer"/>
        <action application="set" data="playback_terminators=#"/>
        <action application="playback" data="C:\Program Files\FreeSWITCH\sounds\music\48000\danza-espanola-op-37-h-142-xii-arabesca.wav"/>
      </condition>
    </extension>
  </include>

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From gcd at i.ph  Mon Sep 19 09:05:25 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Mon, 19 Sep 2011 13:05:25 +0800
Subject: [Freeswitch-users] Dial plan help
In-Reply-To: <8D04992C-91A6-4B8F-9424-6CFC901B8B37@lyonl.com>
References: <8D04992C-91A6-4B8F-9424-6CFC901B8B37@lyonl.com>
Message-ID: <CAHKqMb58=VjvOjo2qm1rjMNS67QAvtVYHoJ_9=410zGSD-zZNA@mail.gmail.com>

hi chad,

nothing could match "+" in the destination_number.

-nandy

On Mon, Sep 19, 2011 at 12:42 PM, Chad Vogel <cvogel at lyonl.com> wrote:

>  I'm trying to build a dialplan however it doesn't seem to work. on the
> first number I would like to forward and all other numbers i would like to
> play an an audio file.
>
>    <include>
>      <extension name="14142211800">
>        <condition field="destination_number" expression=""^+14142211800$">
>          <action application="bridge" data="sofia/gateway/level3/
> +15618911806"/>
>        </condition>
>      </extension>
>      <extension name="13127567001">
>        <condition field="destination_number" expression=""^+(1{0,1}\d{10
> })$">
>          <action application="answer"/>
>          <action application="set" data="playback_terminators=#"/>
>          <action application="playback" data="C:\Program
> Files\FreeSWITCH\sounds\music\48000\danza-espanola-op-37-h-142-xii-arabesca.wav
> "/>
>        </condition>
>      </extension>
>    </include>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rendyfrx at gmail.com  Mon Sep 19 09:23:04 2011
From: rendyfrx at gmail.com (rendyfrx)
Date: Sun, 18 Sep 2011 22:23:04 -0700 (PDT)
Subject: [Freeswitch-users] [mod_xml_curl] Only accept if Return
	Password is blank
In-Reply-To: <4E70DCAA.1040400@livecall.com>
References: <1315990960205-6791769.post@n2.nabble.com>
	<4E70CD5F.1010206@livecall.com> <4E70CF01.9080406@livecall.com>
	<1316018272976-6793355.post@n2.nabble.com>
	<4E70DCAA.1040400@livecall.com>
Message-ID: <1316409784934-6807266.post@n2.nabble.com>

Hi Jack,
I have tried out your suggestion but still no luck. I have cross check with
the post attempt (FS console) and the domain/user info is correct.
The return in FS console:

2011-09-19 13:21:57.001319 [CONSOLE] mod_xml_curl.c:313 XML response is in
/tmp/c0a9e205-c738-45bf-aaa2-50a6ac0168f4.tmp.xml
2011-09-19 13:21:57.001319 [WARNING] sofia_reg.c:1199 SIP auth failure
(REGISTER) on sofia profile 'external' for [+6597293279 at 10.1.1.47] from ip
111.65.28.60
send 554 bytes to udp/[111.65.28.60]:33913 at 05:21:57.005661:
   ------------------------------------------------------------------------
   SIP/2.0 403 Forbidden
   Via: SIP/2.0/UDP
10.223.215.249:5060;rport=33913;branch=z9hG4bKPj0LZUqL1S3xjOHQ5mFNQDWrf5TSZZUWER;received=111.65.28.60
   .....

my XML return is like in image
http://freeswitch-users.2379917.n2.nabble.com/file/n6807266/mod_xml_curl_content.jpg 

Do you have any other idea of what might be the problem?

Thank you.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-xml-curl-Only-accept-if-Return-Password-is-blank-tp6791769p6807266.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From cvogel at lyonl.com  Mon Sep 19 09:35:38 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Mon, 19 Sep 2011 05:35:38 +0000
Subject: [Freeswitch-users] Dial plan help
In-Reply-To: <CAHKqMb58=VjvOjo2qm1rjMNS67QAvtVYHoJ_9=410zGSD-zZNA@mail.gmail.com>
References: <8D04992C-91A6-4B8F-9424-6CFC901B8B37@lyonl.com>
	<CAHKqMb58=VjvOjo2qm1rjMNS67QAvtVYHoJ_9=410zGSD-zZNA@mail.gmail.com>
Message-ID: <DF04C9DB-FC65-45A5-810A-AD458CA19025@lyonl.com>

level 3 is sending us + on every incoming DID, how would i write the correct expression?

On Sep 19, 2011, at 12:05 AM, Nandy Dagondon wrote:

hi chad,

nothing could match "+" in the destination_number.

-nandy

On Mon, Sep 19, 2011 at 12:42 PM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:
I'm trying to build a dialplan however it doesn't seem to work. on the first number I would like to forward and all other numbers i would like to play an an audio file.

  <include>
    <extension name="14142211800">
      <condition field="destination_number" expression=""^+14142211800$">
        <action application="bridge" data="sofia/gateway/level3/+15618911806<tel:%2B15618911806>"/>
      </condition>
    </extension>
    <extension name="13127567001">
      <condition field="destination_number" expression=""^+(1{0,1}\d{10})$">
        <action application="answer"/>
        <action application="set" data="playback_terminators=#"/>
        <action application="playback" data="C:\Program Files\FreeSWITCH\sounds\music\48000\danza-espanola-op-37-h-142-xii-arabesca.wav"/>
      </condition>
    </extension>
  </include>



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From avi at avimarcus.net  Mon Sep 19 09:46:59 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 19 Sep 2011 08:46:59 +0300
Subject: [Freeswitch-users] Dial plan help
In-Reply-To: <DF04C9DB-FC65-45A5-810A-AD458CA19025@lyonl.com>
References: <8D04992C-91A6-4B8F-9424-6CFC901B8B37@lyonl.com>
	<CAHKqMb58=VjvOjo2qm1rjMNS67QAvtVYHoJ_9=410zGSD-zZNA@mail.gmail.com>
	<DF04C9DB-FC65-45A5-810A-AD458CA19025@lyonl.com>
Message-ID: <CALNZuwkd8gEaBNQdKJVLyUmB9suw0JtGa7bEFn=r7RqmYL0TzA@mail.gmail.com>

+ is a regex operator (see a basic intro at
http://wiki.freeswitch.org/wiki/Regular_Expression)
If you want to use a literal +, you need to escape it, e.g \+
-Avi


On Mon, Sep 19, 2011 at 8:35 AM, Chad Vogel <cvogel at lyonl.com> wrote:

>  level 3 is sending us + on every incoming DID, how would i write the
> correct expression?
>
>   On Sep 19, 2011, at 12:05 AM, Nandy Dagondon wrote:
>
> hi chad,
>
> nothing could match "+" in the destination_number.
>
> -nandy
>
> On Mon, Sep 19, 2011 at 12:42 PM, Chad Vogel <cvogel at lyonl.com> wrote:
>
>>  I'm trying to build a dialplan however it doesn't seem to work. on the
>> first number I would like to forward and all other numbers i would like to
>> play an an audio file.
>>
>>    <include>
>>     <extension name="14142211800">
>>       <condition field="destination_number" expression=""^+14142211800$">
>>         <action application="bridge" data="sofia/gateway/level3/
>> +15618911806"/>
>>       </condition>
>>     </extension>
>>     <extension name="13127567001">
>>       <condition field="destination_number" expression=""^+(1{0,1}\d{10
>> })$">
>>         <action application="answer"/>
>>         <action application="set" data="playback_terminators=#"/>
>>         <action application="playback" data="C:\Program
>> Files\FreeSWITCH\sounds\music\48000\danza-espanola-op-37-h-142-xii-arabesca.wav
>> "/>
>>       </condition>
>>     </extension>
>>   </include>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cvogel at lyonl.com  Mon Sep 19 10:40:33 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Mon, 19 Sep 2011 06:40:33 +0000
Subject: [Freeswitch-users] Dial plan help
In-Reply-To: <CALNZuwkd8gEaBNQdKJVLyUmB9suw0JtGa7bEFn=r7RqmYL0TzA@mail.gmail.com>
References: <8D04992C-91A6-4B8F-9424-6CFC901B8B37@lyonl.com>
	<CAHKqMb58=VjvOjo2qm1rjMNS67QAvtVYHoJ_9=410zGSD-zZNA@mail.gmail.com>
	<DF04C9DB-FC65-45A5-810A-AD458CA19025@lyonl.com>
	<CALNZuwkd8gEaBNQdKJVLyUmB9suw0JtGa7bEFn=r7RqmYL0TzA@mail.gmail.com>
Message-ID: <8C96D7E4-CBE0-4C76-A4C2-FCA3F3A6B71B@lyonl.com>

it seems to almost work, but call forwarding doesnt work, I get this error: "[INFO] mod_dptools.c:2801 Originate Failed.  Cause: ORIGINATOR_CANCEL"

  <include>
    <extension name="4142211800">
      <condition field="destination_number" expression="^(\+?1)?(4142211800)$">
        <action application="bridge" data="sofia/gateway/level3/+15618911806"/>
      </condition>
    </extension>
    <extension name="13127567001">
      <condition field="destination_number" expression="^(\+?1)?(\d{10})$">
        <action application="answer"/>
        <action application="set" data="playback_terminators=#"/>
        <action application="playback" data="C:\Program Files\FreeSWITCH\sounds\music\48000\danza-espanola-op-37-h-142-xii-arabesca.wav"/>
      </condition>
    </extension>
  </include>

Our gateway config (i did replace the username's and passwords):

  <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="false"/>
    <param name="dtmf-type" value="rfc2833"/>
  </gateway>>
</include>

On Sep 19, 2011, at 12:46 AM, Avi Marcus wrote:

+ is a regex operator (see a basic intro at http://wiki.freeswitch.org/wiki/Regular_Expression)
If you want to use a literal +, you need to escape it, e.g \+
-Avi


On Mon, Sep 19, 2011 at 8:35 AM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:
level 3 is sending us + on every incoming DID, how would i write the correct expression?

On Sep 19, 2011, at 12:05 AM, Nandy Dagondon wrote:

hi chad,

nothing could match "+" in the destination_number.

-nandy

On Mon, Sep 19, 2011 at 12:42 PM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:
I'm trying to build a dialplan however it doesn't seem to work. on the first number I would like to forward and all other numbers i would like to play an an audio file.

  <include>
    <extension name="14142211800<tel:14142211800>">
      <condition field="destination_number" expression=""^+14142211800<tel:14142211800>$">
        <action application="bridge" data="sofia/gateway/level3/+15618911806<tel:%2B15618911806>"/>
      </condition>
    </extension>
    <extension name="13127567001<tel:13127567001>">
      <condition field="destination_number" expression=""^+(1{0,1}\d{10})$">
        <action application="answer"/>
        <action application="set" data="playback_terminators=#"/>
        <action application="playback" data="C:\Program Files\FreeSWITCH\sounds\music\48000\danza-espanola-op-37-h-142-xii-arabesca.wav"/>
      </condition>
    </extension>
  </include>



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From cchirag at gmail.com  Sat Sep 17 08:02:49 2011
From: cchirag at gmail.com (Chirag Chhatriwala)
Date: Sat, 17 Sep 2011 00:02:49 -0400
Subject: [Freeswitch-users] FreeSwitch with Googlevoice (dingaling) on
 OpenWrt based Router
In-Reply-To: <CAKbxfG8fqtG1rxuPQYuTwz-yYAiA1ycK+QCjDgS3MqOkhx2qnA@mail.gmail.com>
References: <1316208246493-6802276.post@n2.nabble.com>
	<CAKbxfG8fqtG1rxuPQYuTwz-yYAiA1ycK+QCjDgS3MqOkhx2qnA@mail.gmail.com>
Message-ID: <CAO0n5wq3Wi8ZZqh2ddyvJ1KVKvVONen_LJs-jXFZCmiVyix+yg@mail.gmail.com>

Thank you Anthony. That did help.
After doing so, I got stuck at another step.
external_rtp_ip
external_sip_ip

Apparently, these get set to stun.freeswitch.org, which was just dumping all
my calls.
I changed these settings to stunserver.org and my installation started
working like magic.

Now, I need to understand the things below:
1. How to change the incoming caller id to the actual numeric value and not
show "Google Voice" for caller name field
2. How to call a SIP URI
3. How to call a googleid.. ie.. calling joe at gmail.com

Chirag


On Fri, Sep 16, 2011 at 6:49 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> set local_ip_v4 manually in vars.xml at the top to override what the
> default tries to guess (it will always go for the first interface that
> can reach the internet unless you override it)
>
>
> On Fri, Sep 16, 2011 at 4:24 PM, cchhat01 <cchirag at gmail.com> wrote:
> > Hi All,
> >
> > I am in the process of getting the freeswitch installation up and running
> on
> > my router (Netgear WNDR3700v2) which runs trunk Openwrt trunk (attitude
> > adjustment).
> > This is freeswitch 1.0.7.
> > 0. dingaling.conf.xml was not being copied for my installation type and
> is
> > now addressed (thanks to mazilo for committing this checkin)
> > 1. once started, fs_cli shows "sofia status" with the IP address of my
> WAN
> > IP Address (207.a.b.c) and not that of the router (192.168.1.1). Digging
> > through the config, i found that $${local_ip_v4} results in this value,
> even
> > though it should really be 192.168.1.1. Is there an option which disables
> > running FS on the public IP? As a result of this problem, I can't even
> get
> > my SIP ATA device to register. Once I manually change the configuration
> to
> > use the domain="192.168.1.1" instead of "${local_ip_v4}", my ATA clients
> can
> > register.
> >
> > 2. Once i copied over the dingaling.conf.xml file, i was successfully
> > connected to gtalk account, however all my inbound calls did not route to
> > the sofia directory user 1000 (as specified in my
> > jingle_profiles/client.xml).
> > This file is a replica of the info found on
> >
> http://wiki.freeswitch.org/wiki/Google_Voice#Setup_FreeSWITCH_-_Dingaling_to_work_with_your_Gmail_account
> > with my gtalk login details.
> >
> > I need these two aspects to start communicating with each other.
> > Please advise if anyone has any ideas as to what needs to be done.
> > Thanks,
> >
> > Chirag
> >
> > --
> > View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-with-Googlevoice-dingaling-on-OpenWrt-based-Router-tp6802276p6802276.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
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From cchirag at gmail.com  Sat Sep 17 08:04:32 2011
From: cchirag at gmail.com (Chirag Chhatriwala)
Date: Sat, 17 Sep 2011 00:04:32 -0400
Subject: [Freeswitch-users] FreeSwitch with Googlevoice (dingaling) on
 OpenWrt based Router
In-Reply-To: <CAO0n5wq3Wi8ZZqh2ddyvJ1KVKvVONen_LJs-jXFZCmiVyix+yg@mail.gmail.com>
References: <1316208246493-6802276.post@n2.nabble.com>
	<CAKbxfG8fqtG1rxuPQYuTwz-yYAiA1ycK+QCjDgS3MqOkhx2qnA@mail.gmail.com>
	<CAO0n5wq3Wi8ZZqh2ddyvJ1KVKvVONen_LJs-jXFZCmiVyix+yg@mail.gmail.com>
Message-ID: <CAO0n5wqbT8om2BMXLmQGOL+Stya+yn3kKc6ZJFBmChk5x3wsdQ@mail.gmail.com>

Also,

When making an outgoing call, I'm not hearing the "ringing phone" sound,
instead all i hear is silence until the call gets connected. Did I screw
something up?

Chirag


On Sat, Sep 17, 2011 at 12:02 AM, Chirag Chhatriwala <cchirag at gmail.com>wrote:

> Thank you Anthony. That did help.
> After doing so, I got stuck at another step.
> external_rtp_ip
> external_sip_ip
>
> Apparently, these get set to stun.freeswitch.org, which was just dumping
> all my calls.
> I changed these settings to stunserver.org and my installation started
> working like magic.
>
> Now, I need to understand the things below:
> 1. How to change the incoming caller id to the actual numeric value and not
> show "Google Voice" for caller name field
> 2. How to call a SIP URI
> 3. How to call a googleid.. ie.. calling joe at gmail.com
>
> Chirag
>
>
>
> On Fri, Sep 16, 2011 at 6:49 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> set local_ip_v4 manually in vars.xml at the top to override what the
>> default tries to guess (it will always go for the first interface that
>> can reach the internet unless you override it)
>>
>>
>> On Fri, Sep 16, 2011 at 4:24 PM, cchhat01 <cchirag at gmail.com> wrote:
>> > Hi All,
>> >
>> > I am in the process of getting the freeswitch installation up and
>> running on
>> > my router (Netgear WNDR3700v2) which runs trunk Openwrt trunk (attitude
>> > adjustment).
>> > This is freeswitch 1.0.7.
>> > 0. dingaling.conf.xml was not being copied for my installation type and
>> is
>> > now addressed (thanks to mazilo for committing this checkin)
>> > 1. once started, fs_cli shows "sofia status" with the IP address of my
>> WAN
>> > IP Address (207.a.b.c) and not that of the router (192.168.1.1). Digging
>> > through the config, i found that $${local_ip_v4} results in this value,
>> even
>> > though it should really be 192.168.1.1. Is there an option which
>> disables
>> > running FS on the public IP? As a result of this problem, I can't even
>> get
>> > my SIP ATA device to register. Once I manually change the configuration
>> to
>> > use the domain="192.168.1.1" instead of "${local_ip_v4}", my ATA clients
>> can
>> > register.
>> >
>> > 2. Once i copied over the dingaling.conf.xml file, i was successfully
>> > connected to gtalk account, however all my inbound calls did not route
>> to
>> > the sofia directory user 1000 (as specified in my
>> > jingle_profiles/client.xml).
>> > This file is a replica of the info found on
>> >
>> http://wiki.freeswitch.org/wiki/Google_Voice#Setup_FreeSWITCH_-_Dingaling_to_work_with_your_Gmail_account
>> > with my gtalk login details.
>> >
>> > I need these two aspects to start communicating with each other.
>> > Please advise if anyone has any ideas as to what needs to be done.
>> > Thanks,
>> >
>> > Chirag
>> >
>> > --
>> > View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-with-Googlevoice-dingaling-on-OpenWrt-based-Router-tp6802276p6802276.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
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From cchirag at gmail.com  Sat Sep 17 08:55:42 2011
From: cchirag at gmail.com (cchhat01)
Date: Fri, 16 Sep 2011 21:55:42 -0700 (PDT)
Subject: [Freeswitch-users] FreeSwitch with Googlevoice (dingaling) on
	OpenWrt based Router
In-Reply-To: <CAKbxfG8fqtG1rxuPQYuTwz-yYAiA1ycK+QCjDgS3MqOkhx2qnA@mail.gmail.com>
References: <1316208246493-6802276.post@n2.nabble.com>
	<CAKbxfG8fqtG1rxuPQYuTwz-yYAiA1ycK+QCjDgS3MqOkhx2qnA@mail.gmail.com>
Message-ID: <1316235342445-6803177.post@n2.nabble.com>

Thanks Anthony.

That did help infact. 
i then checked the cli to see whats happening with the inbound and outbound
calls, turned out it was an issue with the:

external_rtp_ip=stun:stun.freeswitch.org
external_sip_ip=stun:stun.freeswitch.org

changed this to 

external_rtp_ip=stun:stunserver.org
external_sip_ip=stun:stunserver.org

SUCCESS!

Now i need to know 2 additional things:
1. When I dial a number, I don't hear the ringing as I did with my asterisk
setup. Do i need to make some changes to hear the "ringing" sound. All i
hear is silence until the other other party picks up the phone.

2. in the jingle_profile/client.xml which has my gtalk credentials, is it
possible to pass a "priority" value as a param to the gtalk connection? The
reason I ask is because I always want my freeswitch connection to ring ALL
THE TIME when someone calls my google voice number of initiates a "computer
call" via gtalk.

Please help.
Thank you.


Anthony Minessale wrote:
> 
> set local_ip_v4 manually in vars.xml at the top to override what the
> default tries to guess (it will always go for the first interface that
> can reach the internet unless you override it)
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 

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From cchirag at gmail.com  Sat Sep 17 09:51:29 2011
From: cchirag at gmail.com (cchhat01)
Date: Fri, 16 Sep 2011 22:51:29 -0700 (PDT)
Subject: [Freeswitch-users] FreeSwitch with Googlevoice (dingaling) on
	OpenWrt based Router
In-Reply-To: <1316235342445-6803177.post@n2.nabble.com>
References: <1316208246493-6802276.post@n2.nabble.com>
	<CAKbxfG8fqtG1rxuPQYuTwz-yYAiA1ycK+QCjDgS3MqOkhx2qnA@mail.gmail.com>
	<1316235342445-6803177.post@n2.nabble.com>
Message-ID: <1316238689400-6803279.post@n2.nabble.com>

Well, twice over. I have been told that my voice becomes extremely choppy
while being on call.
I can hear the other party just well.

I recorded the call (as it was incoming for me) on the google voice server
by pressing the "4" key.
Indeed my voice was choppy.

Not sure if this is because of CPU utilization or something else.

I'll try back tomorrow to see if there is any help on this matter.

Thanks.

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From cchirag at gmail.com  Sun Sep 18 00:45:07 2011
From: cchirag at gmail.com (cchhat01)
Date: Sat, 17 Sep 2011 13:45:07 -0700 (PDT)
Subject: [Freeswitch-users] FreeSwitch with Googlevoice (dingaling) on
	OpenWrt based Router
In-Reply-To: <1316267568824-6803939.post@n2.nabble.com>
References: <1316208246493-6802276.post@n2.nabble.com>
	<CAKbxfG8fqtG1rxuPQYuTwz-yYAiA1ycK+QCjDgS3MqOkhx2qnA@mail.gmail.com>
	<1316235342445-6803177.post@n2.nabble.com>
	<1316267568824-6803939.post@n2.nabble.com>
Message-ID: <1316292307191-6804611.post@n2.nabble.com>

Mazilo,

Thank you sir.
I completely missed that line when creating the dialplan.

Because i was getting choppy audio, I looked into the "vad" param in my
jingle_profile and set it to "none" instead of "both." Problem solved.

Also, as I read carefully through the vars.xml file, it should have
defaulted to $${bind_server_ip} so I manually changed 

external_rtp_ip=stun:stunserver.org 
external_sip_ip=stun:stunserver.org

to

external_rtp_ip=$${bind_server_ip}
external_sip_ip=$${bind_server_ip}

Now I have a working freeswitch install on my router..
Sweeeeeet.

If there is any way to set my gtalk status as "away" or "invisible" while on
freeswitch, I would like to look into enabling that. I liked asterisk for
that and I'm sure it's probably not so hard. Maybe I can look at how
asterisk did it, but asterisk relies heavily on iksemel, which will require
days and days of effort. 
Maybe a project in a little bit of time.

For now, I'm just glad to have a working FreeSwitch Install.

I wonder if anyone else has FreeSwitch installed and working on the same
model router as mine (Netgear WNDR3700v2).

thank you,
cchhat01

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From avi at avimarcus.net  Mon Sep 19 10:58:49 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 19 Sep 2011 09:58:49 +0300
Subject: [Freeswitch-users] Dial plan help
In-Reply-To: <8C96D7E4-CBE0-4C76-A4C2-FCA3F3A6B71B@lyonl.com>
References: <8D04992C-91A6-4B8F-9424-6CFC901B8B37@lyonl.com>
	<CAHKqMb58=VjvOjo2qm1rjMNS67QAvtVYHoJ_9=410zGSD-zZNA@mail.gmail.com>
	<DF04C9DB-FC65-45A5-810A-AD458CA19025@lyonl.com>
	<CALNZuwkd8gEaBNQdKJVLyUmB9suw0JtGa7bEFn=r7RqmYL0TzA@mail.gmail.com>
	<8C96D7E4-CBE0-4C76-A4C2-FCA3F3A6B71B@lyonl.com>
Message-ID: <CALNZuwkXHH=MNBLd3TOyh1KPDUcVTvOWF4a+JexOJcJvvec_yA@mail.gmail.com>

That means the person calling hung up. If you don't think that's the case,
can you pastebin a full call log on /log 7 from the fs_cli?

-Avi

On Mon, Sep 19, 2011 at 9:40 AM, Chad Vogel <cvogel at lyonl.com> wrote:

>  it seems to almost work, but call forwarding doesnt work, I get this
> error: "[INFO] mod_dptools.c:2801 Originate Failed.  Cause:
> ORIGINATOR_CANCEL"
>
>     <include>
>      <extension name="4142211800">
>        <condition field="destination_number" expression="^(\+?1)?
> (4142211800)$">
>          <action application="bridge" data="sofia/gateway/level3/
> +15618911806"/>
>        </condition>
>      </extension>
>      <extension name="13127567001">
>        <condition field="destination_number" expression="^(\+?1)?(\d{10})$
> ">
>          <action application="answer"/>
>          <action application="set" data="playback_terminators=#"/>
>          <action application="playback" data="C:\Program
> Files\FreeSWITCH\sounds\music\48000\danza-espanola-op-37-h-142-xii-arabesca.wav
> "/>
>        </condition>
>      </extension>
>    </include>
>
>  Our gateway config (i did replace the username's and passwords):
>
>     <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="false"/>
>      <param name="dtmf-type" value="rfc2833"/>
>    </gateway>>
>  </include>
>
>   On Sep 19, 2011, at 12:46 AM, Avi Marcus wrote:
>
>  + is a regex operator (see a basic intro at
> http://wiki.freeswitch.org/wiki/Regular_Expression)
> If you want to use a literal +, you need to escape it, e.g \+
> -Avi
>
>
> On Mon, Sep 19, 2011 at 8:35 AM, Chad Vogel <cvogel at lyonl.com> wrote:
>
>> level 3 is sending us + on every incoming DID, how would i write the
>> correct expression?
>>
>>   On Sep 19, 2011, at 12:05 AM, Nandy Dagondon wrote:
>>
>> hi chad,
>>
>> nothing could match "+" in the destination_number.
>>
>> -nandy
>>
>> On Mon, Sep 19, 2011 at 12:42 PM, Chad Vogel <cvogel at lyonl.com> wrote:
>>
>>>  I'm trying to build a dialplan however it doesn't seem to work. on the
>>> first number I would like to forward and all other numbers i would like to
>>> play an an audio file.
>>>
>>>    <include>
>>>     <extension name="14142211800">
>>>       <condition field="destination_number" expression=""^+14142211800$"
>>> >
>>>         <action application="bridge" data="sofia/gateway/level3/
>>> +15618911806"/>
>>>       </condition>
>>>     </extension>
>>>     <extension name="13127567001">
>>>       <condition field="destination_number" expression=""^+(1{0,1}\d{10
>>> })$">
>>>         <action application="answer"/>
>>>         <action application="set" data="playback_terminators=#"/>
>>>         <action application="playback" data="C:\Program
>>> Files\FreeSWITCH\sounds\music\48000\danza-espanola-op-37-h-142-xii-arabesca.wav
>>> "/>
>>>       </condition>
>>>     </extension>
>>>   </include>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anton.vazir at gmail.com  Mon Sep 19 12:19:04 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Mon, 19 Sep 2011 13:19:04 +0500
Subject: [Freeswitch-users] ESL Inbound - what would be the best known
 way to wait in XML dialplan for any command execution via ESL?
In-Reply-To: <CAMwZ-t+L0iwpYpcWi3vkGMM2yk-=V9NJGLx76Dk_RxqC41QYzw@mail.gmail.com>
References: <BANLkTikt2R7Qgtm6LL2jr8CP5X4_0_=aeA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEF56437@cooper>
	<BANLkTi=CXOqpYtNDQYwHpz1Ds9nPSAP_ww@mail.gmail.com>
	<CAMwZ-t+L0iwpYpcWi3vkGMM2yk-=V9NJGLx76Dk_RxqC41QYzw@mail.gmail.com>
Message-ID: <CAGX+XzstBe-8Vnfg_Koj=nR9_mRhFGoSXgb95URc4d5EU5gd5A@mail.gmail.com>

Gabe,
Try implementing yourself outbound stuff in python, and you will
quickly find out, after trying feeding several hundred cps to it,
while keeping several hundred sessions online.

2011/8/25 Gabriel Gunderson <gabe at gundy.org>:
> On Sat, Jun 11, 2011 at 10:07 AM, Anton VG <anton.vazir at gmail.com> wrote:
>> PS. Outbound ESL showed itself as a terribly bad scaling solution if
>> implemented in python...
>
> What?
>
> I don't mean to troll here, but what does the language (in this case,
> Python) have to do with scaling Oubound ESL? Any long-running process
> is going to be able to handle the setup and teardown of TCP
> connections without much trouble.
>
> One could foul up ESL (in or out) in any language, and maybe you've
> see it done poorly in Python, but I'd have to hear/see more before
> accepting that it's a particularly bad choice.
>
> I'd love to understand better where you are coming from, if you'd care to share.
>
> Best,
> Gabe
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 admin at blindi.net  Mon Sep 19 17:44:23 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Mon, 19 Sep 2011 15:44:23 +0200 (CEST)
Subject: [Freeswitch-users] Problem send_dtmf send to aleg
Message-ID: <Pine.LNX.4.64.1109191542060.3270@tomback.antagus.de>

Hi all,
I like to create a extension to make a dtmf login to my
voicemailbox on my cellphone.

Fs sends only dtmf tones to myself.
Here is my extension:
<extension name="remote voicemaillogin to my cellphone" continue="true">
<condition field="destination_number" expression="^701$">
<action application="send_dtmf" data="*312666 at 100"/>
<action application="bridge" data="loopback/0174559909990/disa"/>

</condition>
</extension>
Can you help me please?
thanks.


---------------
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From cvogel at lyonl.com  Mon Sep 19 19:12:06 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Mon, 19 Sep 2011 15:12:06 +0000
Subject: [Freeswitch-users] Dial plan help
In-Reply-To: <CALNZuwkXHH=MNBLd3TOyh1KPDUcVTvOWF4a+JexOJcJvvec_yA@mail.gmail.com>
References: <8D04992C-91A6-4B8F-9424-6CFC901B8B37@lyonl.com>
	<CAHKqMb58=VjvOjo2qm1rjMNS67QAvtVYHoJ_9=410zGSD-zZNA@mail.gmail.com>
	<DF04C9DB-FC65-45A5-810A-AD458CA19025@lyonl.com>
	<CALNZuwkd8gEaBNQdKJVLyUmB9suw0JtGa7bEFn=r7RqmYL0TzA@mail.gmail.com>
	<8C96D7E4-CBE0-4C76-A4C2-FCA3F3A6B71B@lyonl.com>
	<CALNZuwkXHH=MNBLd3TOyh1KPDUcVTvOWF4a+JexOJcJvvec_yA@mail.gmail.com>
Message-ID: <58C870B4-CFF9-42DF-9B5A-2E20FA7D91E2@lyonl.com>

here is a copy of our logs, i'm seeing "Regex (FAIL)" do i need an except char in the string "sofia/gateway/level3/+15618911806<tel:%2B15618911806>", also how can i tell if fs is sending the remote gateway our authentication info because i can see we are receiving 487 error from level3 and my understanding this happens when we dont pass our username and password.

2011-09-19 14:54:20.601492 [NOTICE] switch_channel.c:908 Ne
w Channel sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> [4e94849a-35b6-4e8f-8fd4-84ac03
e3da7c]
2011-09-19 14:54:20.601492 [DEBUG] sofia.c:5156 Channel sofia/external/+15618911
806 at 4.55.35.60<mailto:806 at 4.55.35.60> entering state [received][100]
2011-09-19 14:54:20.601492 [DEBUG] sofia.c:5167 Remote SDP:
v=0
o=BroadWorks 4208062 1 IN IP4 4.55.35.60
s=-
c=IN IP4 4.55.35.60
t=0 0
m=audio 49972 RTP/AVP 0 8 18 101
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=maxptime:20

2011-09-19 14:54:20.601492 [DEBUG] sofia_glue.c:4739 Audio Codec Compare [PCMU:0
:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-09-19 14:54:20.601492 [DEBUG] sofia_glue.c:2857 Set Codec sofia/external/+1
5618911806 at 4.55.35.60<mailto:5618911806 at 4.55.35.60> PCMU/8000 20 ms 160 samples 64000 bits
2011-09-19 14:54:20.601492 [DEBUG] sofia_glue.c:4859 Set 2833 dtmf send/recv pay
load to 101
2011-09-19 14:54:20.601492 [DEBUG] sofia.c:5357 (sofia/external/+15618911806 at 4.5
5.35.60) State Change CS_NEW -> CS_INIT
2011-09-19 14:54:20.601492 [DEBUG] switch_core_session.c:1175 Send signal sofia/
external/+15618911806 at 4.55.35.60<mailto:external/+15618911806 at 4.55.35.60> [BREAK]
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:336 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) Running State Change CS_INIT
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:375 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) State INIT
2011-09-19 14:54:20.601492 [DEBUG] mod_sofia.c:85 sofia/external/+15618911806 at 4.
55.35.60 SOFIA INIT
2011-09-19 14:54:20.601492 [DEBUG] mod_sofia.c:125 (sofia/external/+15618911806@
4.55.35.60) State Change CS_INIT -> CS_ROUTING
2011-09-19 14:54:20.601492 [DEBUG] switch_core_session.c:1175 Send signal sofia/
external/+15618911806 at 4.55.35.60<mailto:external/+15618911806 at 4.55.35.60> [BREAK]
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:375 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) State INIT going to sleep
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:336 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) Running State Change CS_ROUTING
2011-09-19 14:54:20.601492 [DEBUG] switch_channel.c:1837 (sofia/external/+156189
11806 at 4.55.35.60<mailto:11806 at 4.55.35.60>) Callstate Change DOWN -> RINGING
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:384 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) State ROUTING
2011-09-19 14:54:20.601492 [DEBUG] mod_sofia.c:148 sofia/external/+15618911806 at 4
.55.35.60 SOFIA ROUTING
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:78 sofia/external
/+15618911806 at 4.55.35.60 Standard ROUTING
2011-09-19 14:54:20.601492 [INFO] mod_dialplan_xml.c:336 Processing +15618911806
 <+15618911806>->+14142211800 in context public
Dialplan: sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> parsing [public->unloop] contin
ue=false
Dialplan: sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> Regex (PASS) [unloop] ${unroll_
loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> Regex (FAIL) [unloop] ${sip_loo
ped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> parsing [public->outside_call]
continue=true
Dialplan: sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> Absolute Condition [outside_cal
l]
Dialplan: sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> Action set(outside_call=true)
Dialplan: sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> Action set(RFC2822_DATE=${strft
ime(%a, %d %b %Y %T %z)})
Dialplan: sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> parsing [public->call_debug] co
ntinue=true
Dialplan: sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> Regex (FAIL) [call_debug] ${cal
l_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> parsing [public->public_extensi
ons] continue=false
Dialplan: sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> Regex (FAIL) [public_extensions
] destination_number(+14142211800) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> parsing [public->4142211800] co
ntinue=false
Dialplan: sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> Regex (PASS) [4142211800] desti
nation_number(+14142211800) =~ /^(\+?1)?(4142211800)$/ break=on-false
Dialplan: sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> Action bridge(sofia/gateway/lev
el3/+15618911806)
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:128 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) State Change CS_ROUTING -> CS_EXECUTE
2011-09-19 14:54:20.601492 [DEBUG] switch_core_session.c:1175 Send signal sofia/
external/+15618911806 at 4.55.35.60<mailto:external/+15618911806 at 4.55.35.60> [BREAK]
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:384 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) State ROUTING going to sleep
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:336 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) Running State Change CS_EXECUTE
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:391 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) State EXECUTE
2011-09-19 14:54:20.601492 [DEBUG] mod_sofia.c:241 sofia/external/+15618911806 at 4
.55.35.60 SOFIA EXECUTE
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:166 sofia/externa
l/+15618911806 at 4.55.35.60<mailto:l/+15618911806 at 4.55.35.60> Standard EXECUTE
EXECUTE sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> set(outside_call=true)
2011-09-19 14:54:20.601492 [DEBUG] mod_dptools.c:1167 sofia/external/+1561891180
6 at 4.55.35.60<mailto:6 at 4.55.35.60> SET [outside_call]=[true]
EXECUTE sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> set(RFC2822_DATE=Mon, 19 Sep 2011
 14:54:20 Coordinated Universal Time)
2011-09-19 14:54:20.601492 [DEBUG] mod_dptools.c:1167 sofia/external/+1561891180
6 at 4.55.35.60<mailto:6 at 4.55.35.60> SET [RFC2822_DATE]=[Mon, 19 Sep 2011 14:54:20 Coordinated Universal
 Time]
EXECUTE sofia/external/+15618911806 at 4.55.35.60<mailto:sofia/external/+15618911806 at 4.55.35.60> bridge(sofia/gateway/level3/+1561
8911806)
2011-09-19 14:54:20.601492 [DEBUG] switch_ivr_originate.c:1873 Parsing global va
riables
2011-09-19 14:54:20.601492 [NOTICE] switch_channel.c:908 New Channel sofia/exter
nal/+15618911806 [1278470d-f578-4200-9892-be93025bdd62]
2011-09-19 14:54:20.601492 [DEBUG] mod_sofia.c:4491 (sofia/external/+15618911806
) State Change CS_NEW -> CS_INIT
2011-09-19 14:54:20.601492 [DEBUG] switch_core_session.c:1175 Send signal sofia/
external/+15618911806 [BREAK]
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:336 (sofia/extern
al/+15618911806) Running State Change CS_INIT
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:375 (sofia/extern
al/+15618911806) State INIT
2011-09-19 14:54:20.601492 [DEBUG] mod_sofia.c:85 sofia/external/+15618911806 SO
FIA INIT
2011-09-19 14:54:20.601492 [DEBUG] switch_core_session.c:870 Send signal sofia/e
xternal/+15618911806 [BREAK]
2011-09-19 14:54:20.601492 [DEBUG] mod_sofia.c:125 (sofia/external/+15618911806)
 State Change CS_INIT -> CS_ROUTING
2011-09-19 14:54:20.601492 [DEBUG] switch_core_session.c:1175 Send signal sofia/
external/+15618911806 [BREAK]
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:375 (sofia/extern
al/+15618911806) State INIT going to sleep
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:336 (sofia/extern
al/+15618911806) Running State Change CS_ROUTING
2011-09-19 14:54:20.601492 [DEBUG] switch_channel.c:1837 (sofia/external/+156189
11806) Callstate Change DOWN -> RINGING
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:384 (sofia/extern
al/+15618911806) State ROUTING
2011-09-19 14:54:20.601492 [DEBUG] mod_sofia.c:148 sofia/external/+15618911806 S
OFIA ROUTING
2011-09-19 14:54:20.601492 [DEBUG] switch_ivr_originate.c:66 (sofia/external/+15
618911806) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-09-19 14:54:20.601492 [DEBUG] switch_core_session.c:1175 Send signal sofia/
external/+15618911806 [BREAK]
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:384 (sofia/extern
al/+15618911806) State ROUTING going to sleep
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:336 (sofia/extern
al/+15618911806) Running State Change CS_CONSUME_MEDIA
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:403 (sofia/extern
al/+15618911806) State CONSUME_MEDIA
2011-09-19 14:54:20.601492 [DEBUG] switch_core_state_machine.c:403 (sofia/extern
al/+15618911806) State CONSUME_MEDIA going to sleep
2011-09-19 14:54:20.601492 [DEBUG] sofia.c:5156 Channel sofia/external/+15618911
806 entering state [calling][0]
2011-09-19 14:54:23.187429 [DEBUG] switch_core_session.c:870 Send signal sofia/e
xternal/+15618911806 at 4.55.35.60<mailto:xternal/+15618911806 at 4.55.35.60> [BREAK]
2011-09-19 14:54:23.187429 [DEBUG] switch_core_session.c:870 Send signal sofia/e
xternal/+15618911806 at 4.55.35.60<mailto:xternal/+15618911806 at 4.55.35.60> [BREAK]
2011-09-19 14:54:23.187429 [DEBUG] switch_core_session.c:870 Send signal sofia/e
xternal/+15618911806 at 4.55.35.60<mailto:xternal/+15618911806 at 4.55.35.60> [BREAK]
2011-09-19 14:54:23.207937 [DEBUG] sofia.c:5156 Channel sofia/external/+15618911
806 at 4.55.35.60<mailto:806 at 4.55.35.60> entering state [terminated][487]
2011-09-19 14:54:23.207937 [DEBUG] switch_channel.c:2797 (sofia/external/+156189
11806 at 4.55.35.60<mailto:11806 at 4.55.35.60>) Callstate Change RINGING -> HANGUP
2011-09-19 14:54:23.207937 [NOTICE] sofia.c:5881 Hangup sofia/external/+15618911
806 at 4.55.35.60<mailto:806 at 4.55.35.60> [CS_EXECUTE] [ORIGINATOR_CANCEL]
2011-09-19 14:54:23.207937 [DEBUG] switch_channel.c:2813 Send signal sofia/exter
nal/+15618911806 at 4.55.35.60<mailto:nal/+15618911806 at 4.55.35.60> [KILL]
2011-09-19 14:54:23.207937 [DEBUG] switch_core_session.c:1175 Send signal sofia/
external/+15618911806 at 4.55.35.60<mailto:external/+15618911806 at 4.55.35.60> [BREAK]
2011-09-19 14:54:23.207937 [DEBUG] switch_channel.c:2797 (sofia/external/+156189
11806) Callstate Change RINGING -> HANGUP
2011-09-19 14:54:23.207937 [NOTICE] switch_ivr_originate.c:3156 Hangup sofia/ext
ernal/+15618911806 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2011-09-19 14:54:23.207937 [DEBUG] switch_channel.c:2813 Send signal sofia/exter
nal/+15618911806 [KILL]
2011-09-19 14:54:23.207937 [DEBUG] switch_core_session.c:1175 Send signal sofia/
external/+15618911806 [BREAK]
2011-09-19 14:54:23.207937 [DEBUG] switch_ivr_originate.c:3331 Originate Cancell
ed by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2011-09-19 14:54:23.207937 [INFO] mod_dptools.c:2801 Originate Failed.  Cause: O
RIGINATOR_CANCEL
2011-09-19 14:54:23.207937 [DEBUG] switch_core_session.c:2209 sofia/external/+15
618911806 at 4.55.35.60<mailto:618911806 at 4.55.35.60> skip receive message [APPLICATION_EXEC_COMPLETE] (channel i
s hungup already)
2011-09-19 14:54:23.207937 [DEBUG] switch_core_state_machine.c:391 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) State EXECUTE going to sleep
2011-09-19 14:54:23.207937 [DEBUG] switch_core_state_machine.c:336 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) Running State Change CS_HANGUP
2011-09-19 14:54:23.207937 [DEBUG] switch_core_state_machine.c:576 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) State HANGUP
2011-09-19 14:54:23.207937 [DEBUG] mod_sofia.c:452 sofia/external/+15618911806 at 4
.55.35.60 Overriding SIP cause 487 with 487 from the other leg
2011-09-19 14:54:23.207937 [DEBUG] mod_sofia.c:458 Channel sofia/external/+15618
911806 at 4.55.35.60<mailto:911806 at 4.55.35.60> hanging up, cause: ORIGINATOR_CANCEL
2011-09-19 14:54:23.207937 [DEBUG] switch_core_state_machine.c:46 sofia/external
/+15618911806 at 4.55.35.60 Standard HANGUP, cause: ORIGINATOR_CANCEL
2011-09-19 14:54:23.207937 [DEBUG] switch_core_state_machine.c:576 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) State HANGUP going to sleep
2011-09-19 14:54:23.207937 [DEBUG] switch_core_state_machine.c:367 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) State Change CS_HANGUP -> CS_REPORTING
2011-09-19 14:54:23.207937 [DEBUG] switch_core_session.c:1175 Send signal sofia/
external/+15618911806 at 4.55.35.60<mailto:external/+15618911806 at 4.55.35.60> [BREAK]
2011-09-19 14:54:23.207937 [DEBUG] switch_core_state_machine.c:336 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) Running State Change CS_REPORTING
2011-09-19 14:54:23.207937 [DEBUG] switch_core_state_machine.c:636 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) State REPORTING
2011-09-19 14:54:23.207937 [DEBUG] switch_core_state_machine.c:53 sofia/external
/+15618911806 at 4.55.35.60 Standard REPORTING, cause: ORIGINATOR_CANCEL
2011-09-19 14:54:23.207937 [DEBUG] switch_core_state_machine.c:636 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) State REPORTING going to sleep
2011-09-19 14:54:23.207937 [DEBUG] switch_core_state_machine.c:361 (sofia/extern
al/+15618911806 at 4.55.35.60<mailto:al/+15618911806 at 4.55.35.60>) State Change CS_REPORTING -> CS_DESTROY
2011-09-19 14:54:23.207937 [DEBUG] switch_core_session.c:1175 Send signal sofia/
external/+15618911806 at 4.55.35.60<mailto:external/+15618911806 at 4.55.35.60> [BREAK]
2011-09-19 14:54:23.207937 [DEBUG] switch_core_session.c:1349 Session 12 (sofia/
external/+15618911806 at 4.55.35.60<mailto:external/+15618911806 at 4.55.35.60>) Locked, Waiting on external entities
2011-09-19 14:54:23.207937 [NOTICE] switch_core_session.c:1367 Session 12 (sofia
/external/+15618911806 at 4.55.35.60) Ended
2011-09-19 14:54:23.207937 [NOTICE] switch_core_session.c:1369 Close Channel sof
ia/external/+15618911806 at 4.55.35.60<mailto:ia/external/+15618911806 at 4.55.35.60> [CS_DESTROY]

On Sep 19, 2011, at 1:58 AM, Avi Marcus wrote:

That means the person calling hung up. If you don't think that's the case, can you pastebin a full call log on /log 7 from the fs_cli?

-Avi

On Mon, Sep 19, 2011 at 9:40 AM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:
it seems to almost work, but call forwarding doesnt work, I get this error: "[INFO] mod_dptools.c:2801 Originate Failed.  Cause: ORIGINATOR_CANCEL"

  <include>
    <extension name="4142211800<tel:4142211800>">
      <condition field="destination_number" expression="^(\+?1)?(4142211800<tel:%284142211800>)$">
        <action application="bridge" data="sofia/gateway/level3/+15618911806<tel:%2B15618911806>"/>
      </condition>
    </extension>
    <extension name="13127567001<tel:13127567001>">
      <condition field="destination_number" expression="^(\+?1)?(\d{10})$">
        <action application="answer"/>
        <action application="set" data="playback_terminators=#"/>
        <action application="playback" data="C:\Program Files\FreeSWITCH\sounds\music\48000\danza-espanola-op-37-h-142-xii-arabesca.wav"/>
      </condition>
    </extension>
  </include>

Our gateway config (i did replace the username's and passwords):

  <gateway name="level3">
    <param name="username" value="username"/>
    <param name="password" value="password"/>
    <param name="proxy" value="4.55.35.60:5070<http://4.55.35.60:5070/>"/>
    <param name="expire-seconds" value="3600"/>
    <param name="register-transport" value="udp"/>
    <param name="register" value="false"/>
    <param name="dtmf-type" value="rfc2833"/>
  </gateway>>
</include>

On Sep 19, 2011, at 12:46 AM, Avi Marcus wrote:

+ is a regex operator (see a basic intro at http://wiki.freeswitch.org/wiki/Regular_Expression)
If you want to use a literal +, you need to escape it, e.g \+
-Avi


On Mon, Sep 19, 2011 at 8:35 AM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:
level 3 is sending us + on every incoming DID, how would i write the correct expression?

On Sep 19, 2011, at 12:05 AM, Nandy Dagondon wrote:

hi chad,

nothing could match "+" in the destination_number.

-nandy

On Mon, Sep 19, 2011 at 12:42 PM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:
I'm trying to build a dialplan however it doesn't seem to work. on the first number I would like to forward and all other numbers i would like to play an an audio file.

  <include>
    <extension name="14142211800<tel:14142211800>">
      <condition field="destination_number" expression=""^+14142211800<tel:14142211800>$">
        <action application="bridge" data="sofia/gateway/level3/+15618911806<tel:%2B15618911806>"/>
      </condition>
    </extension>
    <extension name="13127567001<tel:13127567001>">
      <condition field="destination_number" expression=""^+(1{0,1}\d{10})$">
        <action application="answer"/>
        <action application="set" data="playback_terminators=#"/>
        <action application="playback" data="C:\Program Files\FreeSWITCH\sounds\music\48000\danza-espanola-op-37-h-142-xii-arabesca.wav"/>
      </condition>
    </extension>
  </include>



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-------------- next part --------------
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From kris at kriskinc.com  Mon Sep 19 19:13:06 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 19 Sep 2011 11:13:06 -0400
Subject: [Freeswitch-users] Sophia authentication without using editing
	acl.conf.xml
In-Reply-To: <CALH5ZrYpQrnGd3kB_zY4+j_ct0QbmJHPKR3FEUP5qLeyA+RWkQ@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>
	<CA+GmxbYRhaLOZm7ami4zEa9nkMYESE24baK-13vbXdzcH-3PrA@mail.gmail.com>
	<CALH5ZrYpQrnGd3kB_zY4+j_ct0QbmJHPKR3FEUP5qLeyA+RWkQ@mail.gmail.com>
Message-ID: <CAKDfjgcXbFZa=iZJ=0DOu+kLoXcvoZXrVo3K-KPSU_Q6bREU_Q@mail.gmail.com>

Disable the ACL on the SIP profile and:

1)  Do IP checking using conditions in the dialplan and use respond to
require auth/reject calls when you need it.  This will still require a
reloadxml when you make changes.
2)  Use xml_curl like msc said and look at sip_from_host and use
respond (and xml_curl) to reject/auth calls when you need it.  This
will not require any reloading when you auth calls.

On Sat, Sep 17, 2011 at 3:44 AM, Joe Flemmings <joe.jflemmings at gmail.com> wrote:
> Is their a solution without reloading acl?
>
> On Fri, Sep 9, 2011 at 4:16 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>>
>> 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
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From anthony.minessale at gmail.com  Mon Sep 19 19:24:32 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Sep 2011 10:24:32 -0500
Subject: [Freeswitch-users] uuid_broadcast not working
In-Reply-To: <CAKUOwoH0B1830c-QJj0v+JjZUC9E_d=yJ+BWhvOfN3b9zd2bGQ@mail.gmail.com>
References: <CAKUOwoH0B1830c-QJj0v+JjZUC9E_d=yJ+BWhvOfN3b9zd2bGQ@mail.gmail.com>
Message-ID: <CAKbxfG8-MJAEyaYYzWKsCQ3_UT6eXyVcQfJi6GDaOj44q3AC5g@mail.gmail.com>

are you running a lua script on the session and then trying to connect
to event socket on the same server just to send it a command you can
already do with a freeswitch.API object ?


On Sun, Sep 18, 2011 at 2:48 PM, Luis Jimenez <ljjimenez at gmail.com> wrote:
> Hello, list
>
> I have this lua script, i try to put music to the aleg while do another
> task, but it is not working (not audio at all), any advice on what am doing
> wrong?
>
> THanks in advance, here my script:
>
> ----------------------------------------------------------------------------------------------
>
> require 'socket'
> local host = "127.0.0.1"
> local port = 8021
>
> function set_hold(uuid)
> ??? local conn = socket.connect(host, port)
>
> ??? if (conn) then
> ??? ??? local cmd = "bgapi uuid_broadcast " .. uuid .. "
> /opt/freeswitch/sounds/es/dm/dm-promo-recharge-hold.G729 aleg";
> ??? ??? freeswitch.consoleLog("info", "CMD: " .. cmd .. "\n")
> ??? ??? conn:send(string.format("auth
> ClueCon\r\n\r\n%s\r\n\r\nexit\r\n\r\n", cmd))
> ??? ??? line, err = conn:receive()
> ??? ??? while (not err) do
> ??? ??? ??? if (line ~= nil) then freeswitch.consoleLog("info", "SCKT: " ..
> line .. "\n") end
> ??? ??? ??? line, err = conn:receive()
> ??? ??? end
> ??? ??? conn:close()
> ??? else
> ??? ??? debug.notice("Error posting to server: " .. host .. ":" .. port)
> ??? end
> end
>
> function sleep(n)
> ?? os.execute("sleep " .. tonumber(n))
> end
>
> session:answer()
> session:sleep(500);
> uuid = session:get_uuid();
> freeswitch.consoleLog("info", "UUID: " .. uuid .. "\n");
>
> api = freeswitch.API();
>
> set_hold(uuid);
> freeswitch.consoleLog("info", " - - - RETURNED - - -\n");
>
>
> for i=1,50 do
> ??? if not session:ready() then break end
> ??? sleep(1);
> end
>
> if session:ready() then api:executeString("uuid_break" .. uuid); end
> freeswitch.consoleLog("info", " - - - FINNISH COUNTING - - -\n");
>
> session:hangup();
>
> ----------------------------------------------------------------------------------
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 wayne at hamilton.net  Mon Sep 19 21:03:01 2011
From: wayne at hamilton.net (Wayne)
Date: Mon, 19 Sep 2011 12:03:01 -0500
Subject: [Freeswitch-users] Problem send_dtmf send to aleg
In-Reply-To: <Pine.LNX.4.64.1109191542060.3270@tomback.antagus.de>
References: <Pine.LNX.4.64.1109191542060.3270@tomback.antagus.de>
Message-ID: <D0074D20C5754B59BAAD6DAC363B284F@ccs.local>

Have you tried.
<action application="export" data="nolocal:execute_on_answer=queue_dtmf
312666" /> 
This seems to send tones on answer. I have been working on the same type of
thing but have an issue with the tones being sent multiple times without any
real pattern. Give it a try and check the debug out put. 

Wayne

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Thomas
Hoellriegel
Sent: Monday, September 19, 2011 8:44 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Problem send_dtmf send to aleg

Hi all,
I like to create a extension to make a dtmf login to my voicemailbox on my
cellphone.

Fs sends only dtmf tones to myself.
Here is my extension:
<extension name="remote voicemaillogin to my cellphone" continue="true">
<condition field="destination_number" expression="^701$"> <action
application="send_dtmf" data="*312666 at 100"/> <action application="bridge"
data="loopback/0174559909990/disa"/>

</condition>
</extension>
Can you help me please?
thanks.


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc



From cvogel at lyonl.com  Mon Sep 19 21:59:08 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Mon, 19 Sep 2011 17:59:08 +0000
Subject: [Freeswitch-users] sip headers
Message-ID: <1F7AF579-D88E-4617-B210-BA5AC9C60E20@lyonl.com>

hello,

I was wondering how can modify the sip from and contact headers? I need to pass the phone number in the username not "1-F2la9<sip:1-F2la9 at 4.55.35.60:5070;transport=udp>" to level 3. also i need to do the same with the contact header the username needs to the phone number not  "gw+level3<sip:gw+level3 at 10.126.200.6:5060;transport=udp;gw=level3>". also why is fs switch send our internal ip address and not our external ip address in the contact field, do i not have something setup correctly?

Request URI: sip:+15618911806 at 4.55.35.60:5070
From: "+15745844329" <sip:1-F2la9 at 4.55.35.60:5070;transport=udp>;tag=69eUt3Fm7pyNg
To: <sip:+15618911806 at 4.55.35.60:5070>
Call-ID: a3db0f98-5d82-122f-c3a5-9595615a4f01
CSeq: 17889518 INVITE
Contact: <sip:gw+level3 at 10.126.200.6:5060;transport=udp;gw=level3>


Chad
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From djbinter at gmail.com  Mon Sep 19 22:22:00 2011
From: djbinter at gmail.com (DJB International)
Date: Mon, 19 Sep 2011 11:22:00 -0700
Subject: [Freeswitch-users] sip headers
In-Reply-To: <1F7AF579-D88E-4617-B210-BA5AC9C60E20@lyonl.com>
References: <1F7AF579-D88E-4617-B210-BA5AC9C60E20@lyonl.com>
Message-ID: <CAAB3jWtb-fc5c+FbHx1OhJxzSqHVY3r=DkNUQbrq_ft5guLXQw@mail.gmail.com>

Check the wiki page
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files on

<param name="extension-in-contact" value="true"/>

-djbinter

On Mon, Sep 19, 2011 at 10:59 AM, Chad Vogel <cvogel at lyonl.com> wrote:

>  hello,
>
>  I was wondering how can modify the sip from and contact headers? I need
> to pass the phone number in the username not "1-F2la9" to level 3. also i
> need to do the same with the contact header the username needs to the phone
> number not  "gw+level3". also why is fs switch send our internal ip
> address and not our external ip address in the contact field, do i not have
> something setup correctly?
>
>  Request URI: sip:+15618911806 at 4.55.35.60:5070
> From: "+15745844329" <sip:1-F2la9 at 4.55.35.60:5070;transport=udp
> >;tag=69eUt3Fm7pyNg
> To: <sip:+15618911806 at 4.55.35.60:5070>
> Call-ID: a3db0f98-5d82-122f-c3a5-9595615a4f01
> CSeq: 17889518 INVITE
> Contact: <sip:gw+level3 at 10.126.200.6:5060;transport=udp;gw=level3>
>
>
>  Chad
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ljjimenez at gmail.com  Mon Sep 19 22:25:28 2011
From: ljjimenez at gmail.com (ljjimenez at gmail.com)
Date: Mon, 19 Sep 2011 18:25:28 +0000
Subject: [Freeswitch-users] uuid_broadcast not working
In-Reply-To: <CAKbxfG8-MJAEyaYYzWKsCQ3_UT6eXyVcQfJi6GDaOj44q3AC5g@mail.gmail.com>
References: <CAKUOwoH0B1830c-QJj0v+JjZUC9E_d=yJ+BWhvOfN3b9zd2bGQ@mail.gmail.com><CAKbxfG8-MJAEyaYYzWKsCQ3_UT6eXyVcQfJi6GDaOj44q3AC5g@mail.gmail.com>
Message-ID: <839669831-1316456736-cardhu_decombobulator_blackberry.rim.net-1498515184-@b5.c27.bise6.blackberry>

I did it that way because I tried thru the api, but got the same results

Luis Jimenez

-----Original Message-----
From: Anthony Minessale <anthony.minessale at gmail.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Mon, 19 Sep 2011 10:24:32 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] uuid_broadcast not working

are you running a lua script on the session and then trying to connect
to event socket on the same server just to send it a command you can
already do with a freeswitch.API object ?


On Sun, Sep 18, 2011 at 2:48 PM, Luis Jimenez <ljjimenez at gmail.com> wrote:
> Hello, list
>
> I have this lua script, i try to put music to the aleg while do another
> task, but it is not working (not audio at all), any advice on what am doing
> wrong?
>
> THanks in advance, here my script:
>
> ----------------------------------------------------------------------------------------------
>
> require 'socket'
> local host = "127.0.0.1"
> local port = 8021
>
> function set_hold(uuid)
> ??? local conn = socket.connect(host, port)
>
> ??? if (conn) then
> ??? ??? local cmd = "bgapi uuid_broadcast " .. uuid .. "
> /opt/freeswitch/sounds/es/dm/dm-promo-recharge-hold.G729 aleg";
> ??? ??? freeswitch.consoleLog("info", "CMD: " .. cmd .. "\n")
> ??? ??? conn:send(string.format("auth
> ClueCon\r\n\r\n%s\r\n\r\nexit\r\n\r\n", cmd))
> ??? ??? line, err = conn:receive()
> ??? ??? while (not err) do
> ??? ??? ??? if (line ~= nil) then freeswitch.consoleLog("info", "SCKT: " ..
> line .. "\n") end
> ??? ??? ??? line, err = conn:receive()
> ??? ??? end
> ??? ??? conn:close()
> ??? else
> ??? ??? debug.notice("Error posting to server: " .. host .. ":" .. port)
> ??? end
> end
>
> function sleep(n)
> ?? os.execute("sleep " .. tonumber(n))
> end
>
> session:answer()
> session:sleep(500);
> uuid = session:get_uuid();
> freeswitch.consoleLog("info", "UUID: " .. uuid .. "\n");
>
> api = freeswitch.API();
>
> set_hold(uuid);
> freeswitch.consoleLog("info", " - - - RETURNED - - -\n");
>
>
> for i=1,50 do
> ??? if not session:ready() then break end
> ??? sleep(1);
> end
>
> if session:ready() then api:executeString("uuid_break" .. uuid); end
> freeswitch.consoleLog("info", " - - - FINNISH COUNTING - - -\n");
>
> session:hangup();
>
> ----------------------------------------------------------------------------------
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jack at livecall.com  Mon Sep 19 23:13:31 2011
From: jack at livecall.com (Jack)
Date: Mon, 19 Sep 2011 12:13:31 -0700
Subject: [Freeswitch-users] [mod_xml_curl] Only accept if Return
 Password is blank
In-Reply-To: <1316409784934-6807266.post@n2.nabble.com>
References: <1315990960205-6791769.post@n2.nabble.com>
	<4E70CD5F.1010206@livecall.com> <4E70CF01.9080406@livecall.com>
	<1316018272976-6793355.post@n2.nabble.com>
	<4E70DCAA.1040400@livecall.com>
	<1316409784934-6807266.post@n2.nabble.com>
Message-ID: <4E77945B.9070107@livecall.com>

What are you using as an http server to process the fetch that 
Freeswitch makes?
I use minihttpd and wrapped it in a C# program to access my database and 
make the returns.

You might check modules.config.xml  and make sure <load 
module="*mod_xml_curl*"/>   processes prior to <!--<load 
module="mod_directory"/>--> (which I commented out so it doesn't load)



On 9/18/2011 10:23 PM, rendyfrx wrote:
> Hi Jack,
> I have tried out your suggestion but still no luck. I have cross check with
> the post attempt (FS console) and the domain/user info is correct.
> The return in FS console:
>
> 2011-09-19 13:21:57.001319 [CONSOLE] mod_xml_curl.c:313 XML response is in
> /tmp/c0a9e205-c738-45bf-aaa2-50a6ac0168f4.tmp.xml
> 2011-09-19 13:21:57.001319 [WARNING] sofia_reg.c:1199 SIP auth failure
> (REGISTER) on sofia profile 'external' for [+6597293279 at 10.1.1.47] from ip
> 111.65.28.60
> send 554 bytes to udp/[111.65.28.60]:33913 at 05:21:57.005661:
>     ------------------------------------------------------------------------
>     SIP/2.0 403 Forbidden
>     Via: SIP/2.0/UDP
> 10.223.215.249:5060;rport=33913;branch=z9hG4bKPj0LZUqL1S3xjOHQ5mFNQDWrf5TSZZUWER;received=111.65.28.60
>     .....
>
> my XML return is like in image
> http://freeswitch-users.2379917.n2.nabble.com/file/n6807266/mod_xml_curl_content.jpg
>
> Do you have any other idea of what might be the problem?
>
> Thank you.
>
>
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-xml-curl-Only-accept-if-Return-Password-is-blank-tp6791769p6807266.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cvogel at lyonl.com  Mon Sep 19 23:44:39 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Mon, 19 Sep 2011 19:44:39 +0000
Subject: [Freeswitch-users] sip headers
In-Reply-To: <CAAB3jWtb-fc5c+FbHx1OhJxzSqHVY3r=DkNUQbrq_ft5guLXQw@mail.gmail.com>
References: <1F7AF579-D88E-4617-B210-BA5AC9C60E20@lyonl.com>
	<CAAB3jWtb-fc5c+FbHx1OhJxzSqHVY3r=DkNUQbrq_ft5guLXQw@mail.gmail.com>
Message-ID: <B44849A6-CA46-4D8C-BAD1-B91A17B3C00C@lyonl.com>

after enabling this, now fs send our gateway username but we need to send a phone number. how can we override Contact: <sip:1-F2la9 at 10.126.200.6:5060;transport=udp;gw=level3> and make it to: Contact: <sip:+14142211800 at 207.67.30.226:5060;transport=udp;gw=level3<sip:207.67.30.226 at 10.126.200.6:5060;transport=udp;gw=level3>>

gateway config is:

<include>
  <gateway name="level3">
    <param name="username" value="1-F2la9"/>
    <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="false"/>
    <param name="dtmf-type" value="rfc2833"/>
    <param name="extension-in-contact" value="true"/>
  </gateway>
</include>



On Sep 19, 2011, at 1:22 PM, DJB International wrote:

Check the wiki page http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files on -djbinter

On Mon, Sep 19, 2011 at 10:59 AM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:
hello,

I was wondering how can modify the sip from and contact headers? I need to pass the phone number in the username not "1-F2la9" to level 3. also i need to do the same with the contact header the username needs to the phone number not  "gw+level3". also why is fs switch send our internal ip address and not our external ip address in the contact field, do i not have something setup correctly?

Request URI: sip:+15618911806 at 4.55.35.60:5070
From: "+15745844329<tel:%2B15745844329>" <sip:1-F2la9 at 4.55.35.60:5070;transport=udp>;tag=69eUt3Fm7pyNg
To: <sip:+15618911806 at 4.55.35.60:5070>
Call-ID: a3db0f98-5d82-122f-c3a5-9595615a4f01
CSeq: 17889518 INVITE
Contact: <sip:gw+level3 at 10.126.200.6:5060;transport=udp;gw=level3>


Chad


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
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From joe.jflemmings at gmail.com  Mon Sep 19 23:45:28 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Mon, 19 Sep 2011 12:45:28 -0700
Subject: [Freeswitch-users] Sophia authentication without using editing
	acl.conf.xml
In-Reply-To: <CAKDfjgcXbFZa=iZJ=0DOu+kLoXcvoZXrVo3K-KPSU_Q6bREU_Q@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>
	<CA+GmxbYRhaLOZm7ami4zEa9nkMYESE24baK-13vbXdzcH-3PrA@mail.gmail.com>
	<CALH5ZrYpQrnGd3kB_zY4+j_ct0QbmJHPKR3FEUP5qLeyA+RWkQ@mail.gmail.com>
	<CAKDfjgcXbFZa=iZJ=0DOu+kLoXcvoZXrVo3K-KPSU_Q6bREU_Q@mail.gmail.com>
Message-ID: <CALH5Zra=NZyXeOPEP3HTDhRV7Eg+8SauEdB2oVB9NbGhADuawA@mail.gmail.com>

Thank You,

Did not know you can disable ACL, will try this.

On Mon, Sep 19, 2011 at 8:13 AM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Disable the ACL on the SIP profile and:
>
> 1)  Do IP checking using conditions in the dialplan and use respond to
> require auth/reject calls when you need it.  This will still require a
> reloadxml when you make changes.
> 2)  Use xml_curl like msc said and look at sip_from_host and use
> respond (and xml_curl) to reject/auth calls when you need it.  This
> will not require any reloading when you auth calls.
>
> On Sat, Sep 17, 2011 at 3:44 AM, Joe Flemmings <joe.jflemmings at gmail.com>
> wrote:
> > Is their a solution without reloading acl?
> >
> > On Fri, Sep 9, 2011 at 4:16 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> >>
> >> 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
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Kristian Kielhofner
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Tue Sep 20 00:38:22 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Sep 2011 15:38:22 -0500
Subject: [Freeswitch-users] uuid_broadcast not working
In-Reply-To: <839669831-1316456736-cardhu_decombobulator_blackberry.rim.net-1498515184-@b5.c27.bise6.blackberry>
References: <CAKUOwoH0B1830c-QJj0v+JjZUC9E_d=yJ+BWhvOfN3b9zd2bGQ@mail.gmail.com>
	<CAKbxfG8-MJAEyaYYzWKsCQ3_UT6eXyVcQfJi6GDaOj44q3AC5g@mail.gmail.com>
	<839669831-1316456736-cardhu_decombobulator_blackberry.rim.net-1498515184-@b5.c27.bise6.blackberry>
Message-ID: <CAKbxfG_Q=sZzjbzA8S==ZqJM3PqKH6ZZi43VGD_ATWCpdg1JwQ@mail.gmail.com>

When playing a native file you should not specify the extension.
Apart from that, I have too little information based on such a vague account.

You would probably want to check your console logs at debug level.


On Mon, Sep 19, 2011 at 1:25 PM,  <ljjimenez at gmail.com> wrote:
> I did it that way because I tried thru the api, but got the same results
>
> Luis Jimenez
>
> -----Original Message-----
> From: Anthony Minessale <anthony.minessale at gmail.com>
> Sender: freeswitch-users-bounces at lists.freeswitch.org
> Date: Mon, 19 Sep 2011 10:24:32
> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] uuid_broadcast not working
>
> are you running a lua script on the session and then trying to connect
> to event socket on the same server just to send it a command you can
> already do with a freeswitch.API object ?
>
>
> On Sun, Sep 18, 2011 at 2:48 PM, Luis Jimenez <ljjimenez at gmail.com> wrote:
>> Hello, list
>>
>> I have this lua script, i try to put music to the aleg while do another
>> task, but it is not working (not audio at all), any advice on what am doing
>> wrong?
>>
>> THanks in advance, here my script:
>>
>> ----------------------------------------------------------------------------------------------
>>
>> require 'socket'
>> local host = "127.0.0.1"
>> local port = 8021
>>
>> function set_hold(uuid)
>> ??? local conn = socket.connect(host, port)
>>
>> ??? if (conn) then
>> ??? ??? local cmd = "bgapi uuid_broadcast " .. uuid .. "
>> /opt/freeswitch/sounds/es/dm/dm-promo-recharge-hold.G729 aleg";
>> ??? ??? freeswitch.consoleLog("info", "CMD: " .. cmd .. "\n")
>> ??? ??? conn:send(string.format("auth
>> ClueCon\r\n\r\n%s\r\n\r\nexit\r\n\r\n", cmd))
>> ??? ??? line, err = conn:receive()
>> ??? ??? while (not err) do
>> ??? ??? ??? if (line ~= nil) then freeswitch.consoleLog("info", "SCKT: " ..
>> line .. "\n") end
>> ??? ??? ??? line, err = conn:receive()
>> ??? ??? end
>> ??? ??? conn:close()
>> ??? else
>> ??? ??? debug.notice("Error posting to server: " .. host .. ":" .. port)
>> ??? end
>> end
>>
>> function sleep(n)
>> ?? os.execute("sleep " .. tonumber(n))
>> end
>>
>> session:answer()
>> session:sleep(500);
>> uuid = session:get_uuid();
>> freeswitch.consoleLog("info", "UUID: " .. uuid .. "\n");
>>
>> api = freeswitch.API();
>>
>> set_hold(uuid);
>> freeswitch.consoleLog("info", " - - - RETURNED - - -\n");
>>
>>
>> for i=1,50 do
>> ??? if not session:ready() then break end
>> ??? sleep(1);
>> end
>>
>> if session:ready() then api:executeString("uuid_break" .. uuid); end
>> freeswitch.consoleLog("info", " - - - FINNISH COUNTING - - -\n");
>>
>> session:hangup();
>>
>> ----------------------------------------------------------------------------------
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 ljjimenez at gmail.com  Tue Sep 20 02:27:23 2011
From: ljjimenez at gmail.com (Luis Jimenez)
Date: Mon, 19 Sep 2011 18:27:23 -0400
Subject: [Freeswitch-users] uuid_broadcast not working
In-Reply-To: <CAKbxfG_Q=sZzjbzA8S==ZqJM3PqKH6ZZi43VGD_ATWCpdg1JwQ@mail.gmail.com>
References: <CAKUOwoH0B1830c-QJj0v+JjZUC9E_d=yJ+BWhvOfN3b9zd2bGQ@mail.gmail.com>
	<CAKbxfG8-MJAEyaYYzWKsCQ3_UT6eXyVcQfJi6GDaOj44q3AC5g@mail.gmail.com>
	<839669831-1316456736-cardhu_decombobulator_blackberry.rim.net-1498515184-@b5.c27.bise6.blackberry>
	<CAKbxfG_Q=sZzjbzA8S==ZqJM3PqKH6ZZi43VGD_ATWCpdg1JwQ@mail.gmail.com>
Message-ID: <CAKUOwoGmmmCYB+qBhDuBBUc0Qe5vwJ8sTrpJsAYc1KYKLn2xeg@mail.gmail.com>

Hello Anthony, i am trying to put the caller to listen background music,
while the script continue sending a SMS using curl, the duration of this
process (send SMS) is about 4 secs.

Can you advice me on how to resolve this?

Thanks in advance.

Luis Jimenez



On Mon, Sep 19, 2011 at 4:38 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> When playing a native file you should not specify the extension.
> Apart from that, I have too little information based on such a vague
> account.
>
> You would probably want to check your console logs at debug level.
>
>
> On Mon, Sep 19, 2011 at 1:25 PM,  <ljjimenez at gmail.com> wrote:
> > I did it that way because I tried thru the api, but got the same results
> >
> > Luis Jimenez
> >
> > -----Original Message-----
> > From: Anthony Minessale <anthony.minessale at gmail.com>
> > Sender: freeswitch-users-bounces at lists.freeswitch.org
> > Date: Mon, 19 Sep 2011 10:24:32
> > To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
> > Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> > Subject: Re: [Freeswitch-users] uuid_broadcast not working
> >
> > are you running a lua script on the session and then trying to connect
> > to event socket on the same server just to send it a command you can
> > already do with a freeswitch.API object ?
> >
> >
> > On Sun, Sep 18, 2011 at 2:48 PM, Luis Jimenez <ljjimenez at gmail.com>
> wrote:
> >> Hello, list
> >>
> >> I have this lua script, i try to put music to the aleg while do another
> >> task, but it is not working (not audio at all), any advice on what am
> doing
> >> wrong?
> >>
> >> THanks in advance, here my script:
> >>
> >>
> ----------------------------------------------------------------------------------------------
> >>
> >> require 'socket'
> >> local host = "127.0.0.1"
> >> local port = 8021
> >>
> >> function set_hold(uuid)
> >>     local conn = socket.connect(host, port)
> >>
> >>     if (conn) then
> >>         local cmd = "bgapi uuid_broadcast " .. uuid .. "
> >> /opt/freeswitch/sounds/es/dm/dm-promo-recharge-hold.G729 aleg";
> >>         freeswitch.consoleLog("info", "CMD: " .. cmd .. "\n")
> >>         conn:send(string.format("auth
> >> ClueCon\r\n\r\n%s\r\n\r\nexit\r\n\r\n", cmd))
> >>         line, err = conn:receive()
> >>         while (not err) do
> >>             if (line ~= nil) then freeswitch.consoleLog("info", "SCKT: "
> ..
> >> line .. "\n") end
> >>             line, err = conn:receive()
> >>         end
> >>         conn:close()
> >>     else
> >>         debug.notice("Error posting to server: " .. host .. ":" .. port)
> >>     end
> >> end
> >>
> >> function sleep(n)
> >>    os.execute("sleep " .. tonumber(n))
> >> end
> >>
> >> session:answer()
> >> session:sleep(500);
> >> uuid = session:get_uuid();
> >> freeswitch.consoleLog("info", "UUID: " .. uuid .. "\n");
> >>
> >> api = freeswitch.API();
> >>
> >> set_hold(uuid);
> >> freeswitch.consoleLog("info", " - - - RETURNED - - -\n");
> >>
> >>
> >> for i=1,50 do
> >>     if not session:ready() then break end
> >>     sleep(1);
> >> end
> >>
> >> if session:ready() then api:executeString("uuid_break" .. uuid); end
> >> freeswitch.consoleLog("info", " - - - FINNISH COUNTING - - -\n");
> >>
> >> session:hangup();
> >>
> >>
> ----------------------------------------------------------------------------------
> >>
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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
>
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From yungwei at resolvity.com  Tue Sep 20 02:54:55 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Mon, 19 Sep 2011 18:54:55 -0400
Subject: [Freeswitch-users] Making changes to fifo.conf.xml requires
	restarting FS?
Message-ID: <33095823FD21DF429B481B5163264B795118922232@VMBX102.ihostexchange.net>

Hi,

I noticed that making changes to fifo.conf.xml requires FS to be restarted, and reloadxml command doesn't seem to reload fifo.conf.xml.
Am I missing something? Thanks.


From joe.jflemmings at gmail.com  Tue Sep 20 08:15:29 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Mon, 19 Sep 2011 21:15:29 -0700
Subject: [Freeswitch-users] Sophia authentication without using editing
	acl.conf.xml
In-Reply-To: <CALH5Zra=NZyXeOPEP3HTDhRV7Eg+8SauEdB2oVB9NbGhADuawA@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>
	<CA+GmxbYRhaLOZm7ami4zEa9nkMYESE24baK-13vbXdzcH-3PrA@mail.gmail.com>
	<CALH5ZrYpQrnGd3kB_zY4+j_ct0QbmJHPKR3FEUP5qLeyA+RWkQ@mail.gmail.com>
	<CAKDfjgcXbFZa=iZJ=0DOu+kLoXcvoZXrVo3K-KPSU_Q6bREU_Q@mail.gmail.com>
	<CALH5Zra=NZyXeOPEP3HTDhRV7Eg+8SauEdB2oVB9NbGhADuawA@mail.gmail.com>
Message-ID: <CALH5ZrYStTeu-ttpvYDqOh1CRy-ZuZ-zaUYYjso9WeEUAaA1iQ@mail.gmail.com>

How do you disable ACL in SIP Profile?

On Mon, Sep 19, 2011 at 12:45 PM, Joe Flemmings <joe.jflemmings at gmail.com>wrote:

> Thank You,
>
> Did not know you can disable ACL, will try this.
>
>
> On Mon, Sep 19, 2011 at 8:13 AM, Kristian Kielhofner <kris at kriskinc.com>wrote:
>
>> Disable the ACL on the SIP profile and:
>>
>> 1)  Do IP checking using conditions in the dialplan and use respond to
>> require auth/reject calls when you need it.  This will still require a
>> reloadxml when you make changes.
>> 2)  Use xml_curl like msc said and look at sip_from_host and use
>> respond (and xml_curl) to reject/auth calls when you need it.  This
>> will not require any reloading when you auth calls.
>>
>> On Sat, Sep 17, 2011 at 3:44 AM, Joe Flemmings <joe.jflemmings at gmail.com>
>> wrote:
>> > Is their a solution without reloading acl?
>> >
>> > On Fri, Sep 9, 2011 at 4:16 AM, Dmitry Sytchev <kbdfck at gmail.com>
>> wrote:
>> >>
>> >> 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
>> >>
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Kristian Kielhofner
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From rendyfrx at gmail.com  Tue Sep 20 10:18:30 2011
From: rendyfrx at gmail.com (rendyfrx)
Date: Mon, 19 Sep 2011 23:18:30 -0700 (PDT)
Subject: [Freeswitch-users] [mod_xml_curl] Only accept if Return
	Password is blank
In-Reply-To: <4E77945B.9070107@livecall.com>
References: <1315990960205-6791769.post@n2.nabble.com>
	<4E70CD5F.1010206@livecall.com> <4E70CF01.9080406@livecall.com>
	<1316018272976-6793355.post@n2.nabble.com>
	<4E70DCAA.1040400@livecall.com>
	<1316409784934-6807266.post@n2.nabble.com>
	<4E77945B.9070107@livecall.com>
Message-ID: <1316499510378-6811033.post@n2.nabble.com>

Hi Jack,
I think we finally managed to resolved the issue "accidentally" :)
We set enable-compact-headers=true in internal sip_profiles and somehow it
works, probably due to network packet issue too big.

Anyway, thanks for your kind help.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-xml-curl-Only-accept-if-Return-Password-is-blank-tp6791769p6811033.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fabio.bigliardi at gmail.com  Tue Sep 20 13:56:13 2011
From: fabio.bigliardi at gmail.com (Fabio Bigliardi)
Date: Tue, 20 Sep 2011 11:56:13 +0200
Subject: [Freeswitch-users] Codec byte order
Message-ID: <CABtP9Qj62ihNHYgZoGb-_1Q7JjbxSbQSmCh=P976GPytHHoBQg@mail.gmail.com>

Hi all,
I have an audio decoder registered to freeswitch server.
It worked with a codec L16 at 24000h little endian.
After software update to FreeSWITCH Version 1.0.head (git-25c725c 2011-06-30
18-30-24 -0500), the audio decoder ceased to work with this codec. It
produces just noise.
With the big endian version of the codec, it works again.
The hardware on which freeswitch is running, is the same.
How could you explain this behaviour?

Thank you very much.

Fabio Bigliardi
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From vaad.fabi at gmail.com  Tue Sep 20 14:19:35 2011
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Tue, 20 Sep 2011 13:19:35 +0300
Subject: [Freeswitch-users] srtp  RTP/SAVP
In-Reply-To: <CALH5Zra=NZyXeOPEP3HTDhRV7Eg+8SauEdB2oVB9NbGhADuawA@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>	<CA+GmxbYRhaLOZm7ami4zEa9nkMYESE24baK-13vbXdzcH-3PrA@mail.gmail.com>	<CALH5ZrYpQrnGd3kB_zY4+j_ct0QbmJHPKR3FEUP5qLeyA+RWkQ@mail.gmail.com>	<CAKDfjgcXbFZa=iZJ=0DOu+kLoXcvoZXrVo3K-KPSU_Q6bREU_Q@mail.gmail.com>
	<CALH5Zra=NZyXeOPEP3HTDhRV7Eg+8SauEdB2oVB9NbGhADuawA@mail.gmail.com>
Message-ID: <4E7868B7.9010407@gmail.com>

  Hi all,

Got problem with FS\srtp settings.
New CounterPath Bria3 works only with softswitch which offer RTP/SAVP in 
the SDP for srtp, otherwise
"No media type supported". Is it possible set some FS srtp 
configurations\settings  to offer RTP/SAVP in the SDP ?


-- 
Best Regards,
Vadim F.



From chris at ghosttelecom.com  Tue Sep 20 15:48:08 2011
From: chris at ghosttelecom.com (Chris Martineau)
Date: Tue, 20 Sep 2011 12:48:08 +0100
Subject: [Freeswitch-users] Manipulate sip error codes
Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF359A38@SVR01.ghosttelecom.local>

 

Hi,

 

Is it possible to translate B leg sip error codes so that the A leg
always gives a fixed error code?

 

i.e make call and get 408 for instance from the far end but would like
486 to be sent on the a leg.

 

Need this as one of our suppliers retransmits a call multiple times
unless they recieve a 403 or 486. No major issue but just clogs the
error cdrs with loads of duplicate records otherwise.

 

Any ideas would be greatly appreciated.

 

Many thanks

 

Chris 

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From ocset at the800group.com  Tue Sep 20 16:06:21 2011
From: ocset at the800group.com (ocset)
Date: Tue, 20 Sep 2011 20:06:21 +0800
Subject: [Freeswitch-users] GXW4104 gateway setup for outgoing calls
In-Reply-To: <CAHKqMb69+OqObSnJdGEOVwreahbaw0SEinJgyixYgzG9FG3UdQ@mail.gmail.com>
References: <4E6EFCFA.6000803@the800group.com>
	<CAHKqMb7vF4ULKjWsHqP20xVRy0oOrDo9A3fpwCBr2NXAjsJ6iw@mail.gmail.com>
	<4E6F2C07.80800@the800group.com>
	<CAHKqMb7BV1oVa5NqvZP+mgS1QSjuaUu-JPpHRFpqmNBs8G8mHQ@mail.gmail.com>
	<4E72DB25.7010706@the800group.com>
	<CAHKqMb69+OqObSnJdGEOVwreahbaw0SEinJgyixYgzG9FG3UdQ@mail.gmail.com>
Message-ID: <4E7881BD.70306@the800group.com>

I fixed the problem.

The guide I followed for setting up the incoming calls suggested this 
setting: "Wait for Dial-Tone(Y/N): ch1-4:Y;"

I changed this to "N" and now it works. Not sure how that affects the 
use of the system but at least incoming call are now working.

Thanks for all your help.


On 09/16/2011 03:35 PM, Nandy Dagondon wrote:
> pls see my comments below. you're welcome.
> -nandy
>
> On Fri, Sep 16, 2011 at 1:14 PM, ocset <ocset at the800group.com 
> <mailto:ocset at the800group.com>> wrote:
>
>     Hi Nandy
>
>     Thanks for your help so far - unfortunately this is still not working.
>
>     Based on your examples, I have created the following  two files
>
>     1.
>     ../sip_profile/internal/01_custom.xml
>
>     <include>
>     <param name="multiple-registrations" value="true"/>
>     <param name="accept-blind-reg" value="true"/>
>     <param name="accept-blind-auth" value="true"/>
>
>     <gateway name="gxw4104-fxo">
>
>     <param name="username" value="1019"/>
>     <param name="password" value="1234"/>
>     <param name="realm" value="192.168.0.160"/>
>     <param name="sip-port" value="5060"/>
>     <param name="rtp_ip" value="192.168.0.160"/>
>
>     <param name="dtmf-type" value="rfc2833"/>
>     <param name="expire-seconds" value="600"/>
>     <param name="register" value="false"/>
>     <param name="caller-id-in-from" value="false"/>
>     </gateway>
>     </include>
>
>     2.
>     ../dialplan/default/01_custom.xml
>
>     <include>
>     <extension name="gxw4104-fxo-local">
>
>     <condition field="${toll_allow}" expression="local"/>
>     <condition field="destination_number" expression="^(\d{10})$">
>     <action application="set"
>     data="effective_caller_id_number=5555555555"/>
>
>     <action application="set"
>     data="effective_caller_id_name=ThisIsMyCompany"/>
>     <action application="set" data="ignore_early_media=ring_ready"/>
>     <action application="set" data="ringback=${us-ring}"/>
>     <action application="bridge"
>     data="sofia/gateway/gxw4104-fxo/$1 at 192.168.0.160:5060"
>     <mailto:sofia/gateway/gxw4104-fxo/$1 at 192.168.0.160:5060>/>
>     </condition>
>     </extension>
>     </include>
>
>
>
>     Command "sofia status"  shows the followiing
>
>     internal                               profile
>     sip:mod_sofia at 192.168.0.23:5060
>     <mailto:sip:mod_sofia at 192.168.0.23:5060>         RUNNING (0)
>     internal::gxw4104-fxo        gateway sip:1019 at 192.168.0.160
>     <mailto:sip:1019 at 192.168.0.160>                        NOREG
>     external                              profile
>     sip:mod_sofia at 192.168.0.23:5080
>     <mailto:sip:mod_sofia at 192.168.0.23:5080>         RUNNING (0)
>     external::example.com <http://example.com>      gateway
>     sip:joeuser at example.com
>     <mailto:sip:joeuser at example.com>                      NOREG
>     internal-ipv6                       profile               
>     sip:mod_sofia@[::1]:5060                        RUNNING (0)
>     192.168.0.23                      alias                  
>     internal                                                    ALIASED
>
>
>     Here is the call log when I try do dial out:
>
>     2011-09-16 <tel:2011-09-16> 12:57:30.903616 [NOTICE]
>     switch_channel.c:669 New Channel sofia/internal/1014 at 192.168.0.23
>     <mailto:sofia/internal/1014 at 192.168.0.23>
>     [0a83e803-b3fc-4a46-bc74-9d6786dac8e2]
>     2011-09-16 <tel:2011-09-16> 12:57:30.933863 [INFO]
>     mod_dialplan_xml.c:418 Processing User 1014->0412345678 in context
>     default
>     2011-09-16 <tel:2011-09-16> 12:57:30.950359 [NOTICE]
>     switch_channel.c:669 New Channel
>     sofia/internal/0412345678 at 192.168.0.160:5060
>     <mailto:sofia/internal/0412345678 at 192.168.0.160:5060>
>     [24111b8d-25f0-4433-a49f-88973730ebfb]
>     2011-09-16 <tel:2011-09-16> 12:57:40.957391 [NOTICE] sofia.c:4789
>     Hangup sofia/internal/0412345678 at 192.168.0.160:5060
>     <mailto:sofia/internal/0412345678 at 192.168.0.160:5060>
>     [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
>     2011-09-16 <tel:2011-09-16> 12:57:40.958554 [NOTICE]
>     switch_core_session.c:1182 Session 7
>     (sofia/internal/0412345678 at 192.168.0.160:5060
>     <mailto:sofia/internal/0412345678 at 192.168.0.160:5060>) Ended
>     2011-09-16 <tel:2011-09-16> 12:57:40.958554 [NOTICE]
>     switch_core_session.c:1184 Close Channel
>     sofia/internal/0412345678 at 192.168.0.160:5060
>     <mailto:sofia/internal/0412345678 at 192.168.0.160:5060> [CS_DESTROY]
>     2011-09-16 <tel:2011-09-16> 12:57:40.958554 [INFO]
>     mod_dptools.c:2355 Originate Failed.  Cause: NORMAL_TEMPORARY_FAILURE
>     2011-09-16 <tel:2011-09-16> 12:57:40.958554 [NOTICE]
>     mod_dptools.c:2418 Hangup sofia/internal/1014 at 192.168.0.23
>     <mailto:sofia/internal/1014 at 192.168.0.23> [CS_EXECUTE]
>     [NORMAL_TEMPORARY_FAILURE]
>     2011-09-16 <tel:2011-09-16> 12:57:40.995191 [NOTICE]
>     switch_core_session.c:1182 Session 6
>     (sofia/internal/1014 at 192.168.0.23
>     <mailto:sofia/internal/1014 at 192.168.0.23>) Ended
>     2011-09-16 <tel:2011-09-16> 12:57:40.995191 [NOTICE]
>     switch_core_session.c:1184 Close Channel
>     sofia/internal/1014 at 192.168.0.23
>     <mailto:sofia/internal/1014 at 192.168.0.23> [CS_DESTROY]
>
>
>
>     I have the following questions
>
>     1. What is the significance of the user 1019? I have a default
>     install of FS so that user does exist but I am not logged in as
>     that user on my sip phone. I am logged in as user 1014.
>
> 1019 is the user account for the FXO port to handle incoming calls. 
> so, the login details must be set on directory/default/1019.xml.
>
> actually,  the user and password entries can be deleted because FS is 
> not registering to the FXO gateway.
>
>     2. The resultant log does not show my gateway being used but
>     instead shows "/sofia/internal/0412345678 at 192.168.0.160:5060"
>     <mailto:/sofia/internal/0412345678 at 192.168.0.160:5060>. Is that
>     expected behaviour?
>
> it's not hitting the dialplan, i guess. i suggest you create prefix 9 
> for PSTN calls e.g.
> <condition field="destination_number" expression="^9(\d+)$">
>
> and make the file ../dialplan/default/01_custom.xml is on top of 
> dialplan/default directory so it will be scanned early.
>
>     3. I assumed that the IP address 192.168.0.9 in your example is
>     the address of your HT503 and not FS. I have thus replaced it with
>     the IP address from my GXW4104 (192.168.0.160). Is that correct?
>
> that is correct.
>
>
>
>
>
>
>
>     On 09/13/2011 11:42 PM, Nandy Dagondon wrote:
>>     i inserted my answers to your questions below. for point #3),
>>     here's an example how i configured my FXO port of ht503.
>>
>>     included in  sip_profile/internal:
>>     <include>
>>     <gateway name="ht503-fxo">
>>     <param name="username" value="1019"/> <-- it's registered to
>>     receive incoming calls
>>     <param name="realm" value="192.168.0.9"/>
>>     <param name="sip-port" value="5062"/> <-- port 5060 is set to the
>>     FXS port
>>     <param name="password" value="1234"/>
>>     <param name="rtp_ip" value="192.168.0.9"/>
>>     <param name="dtmf-type" value="rfc2833"/>
>>     <param name="expire-seconds" value="600"/>
>>     <param name="register" value="false"/>
>>     <param name="caller-id-in-from" value="false"/>
>>     </gateway>
>>     </include>
>>
>>     included  in dialplan/default
>>
>>     <include>
>>     <extension name="ht503-fxo-local">
>>     <condition field="${toll_allow}" expression="local"/>
>>     <condition field="destination_number" expression="^9([2-9]\d{6})$">
>>     <action application="set"
>>     data="effective_caller_id_number=0321234567 <tel:0321234567>"/>
>>     <action application="set" data="effective_caller_id_name=ThisIsMy
>>     Company"/>
>>     <action application="set" data="ignore_early_media=ring_ready"/>
>>     <action application="set" data="ringback=${us-ring}"/>
>>     <action application="bridge"
>>     data="sofia/gateway/ht503-fxo/$1 at 192.168.0.9:5062
>>     <http://1 at 192.168.0.9:5062>"/>
>>     </condition>
>>     </extension>
>>     </include>
>>
>>     it looks you can create 4 internal gateways for the every port,
>>     fxo-1 to fxo-4, w/ the same realm/rtp_ip values but setting
>>     different sip-port values. then  your bridge app would be:
>>
>>     <action application="bridge"
>>     data="sofia/gateway/fxo-1/$1|sofia/gateway/fxo-2/$1|sofia/gateway/fxo-3/$1sofia/gateway/fxo-4/$1"/>
>>
>>     if u want to dialout any free port.
>>
>>     i haven't tested the above. just try it. i hope it works.
>>
>>     -nandy
>>
>>
>>     On Tue, Sep 13, 2011 at 6:10 PM, ocset <ocset at the800group.com
>>     <mailto:ocset at the800group.com>> wrote:
>>
>>         Hi Nandy
>>
>>         Thanks for your reply. I assume 192.168.0.9 in your example
>>         is the IP address of the GXW4104?
>>
>>     yes.
>>
>>
>>         Some more questions
>>
>>         1. When you say port number, is this something I should be
>>         setting up on the GXW4104 so that it is listening on those 4
>>         port numbers? If yes, what would be the setting I am looking for?
>>
>>     not for every port. the gateway has a base port number e.g. 5060
>>     for port#1. add 2 to the subsequent ports e.g. 5062 for port#2
>>     and so on. this is pointed out by sergey.
>>
>>
>>         2. Does that mean I don't define a new gateway in FreeSWITCH?
>>
>>     it's an option. but defining a gateway is cleaner.
>>
>>
>>         3. In your example, you said the bridge data would be
>>         7654321 at 192.168.0.9:5063 <mailto:7654321 at 192.168.0.9:5063>.
>>         What would the whole line look like in the dialplan?*
>>
>>         <action application="bridge"
>>         data="sofia/gateway/7654321 at 192.168.0.9:5063
>>         <mailto:sofia/gateway/7654321 at 192.168.0.9:5063>**"/>*
>>
>>         Still very confused :-)
>>
>>         Thanks
>>
>>
>>         On 09/13/2011 03:45 PM, Nandy Dagondon wrote:
>>>         hi,
>>>
>>>         if GWX4104 is in your local network, use the internal
>>>         profile for the gateway. register your FXO accounts to
>>>         receive incoming calls (i think you did this already).
>>>
>>>         to dialout the ports, specify the port number 5060~5063
>>>         assuming Port1 starts at 5060. to dialout via port4, the
>>>         bridge data should look like:
>>>
>>>         7654321 at 192.168.0.9:5063 <http://7654321 at 192.168.0.9:5063>
>>>
>>>         hope it helps.
>>>
>>>         -nandy
>>>
>>>
>>>         On Tue, Sep 13, 2011 at 2:49 PM, ocset
>>>         <ocset at the800group.com <mailto:ocset at the800group.com>> wrote:
>>>
>>>             Hi
>>>
>>>             I have recently bought a Grandstream GXW4104 (4 FXO
>>>             ports) and need some help setting up a gateway to call
>>>             out using the GXW4104. I am really out of my depth here
>>>             and may be looking at this the wrong way so please bear
>>>             with me.
>>>
>>>             I followed the advice on this website
>>>             "http://www.timhunt.net/wiki/FreeSwitch:GXW4104"
>>>             <http://www.timhunt.net/wiki/FreeSwitch:GXW4104> and
>>>             incoming calls from a PSTN line are working great. Now I
>>>             need to setup a dialplan so that outgoing calls are
>>>             routed through the same PSTN line on the GXW4104. I will
>>>             eventually have 4 PSTN lines with a dialplan to use the
>>>             first available line (if that is possible).
>>>
>>>             According to the FreeSWITCH 1.0.6 book (and many online
>>>             posts) I need to create a gateway and a dialplan but all
>>>             the gateway examples are for SIP accounts.
>>>
>>>             So, the gateway definition seems to need a username and
>>>             password but the GXW4104 does not have that capability.
>>>             I found this gateway definition in the 
>>>             freeswitch.xml.fsxml file but am not sure how many of
>>>             these variables are required.
>>>
>>>             <gateways>
>>>             <gateway name="example.com <http://example.com>">
>>>             <param name="username" value="joeuser"/>
>>>             <param name="password" value="password"/>
>>>             <param name="from-user" value="joeuser"/>
>>>             <param name="from-domain" value="example.com
>>>             <http://example.com>"/>
>>>             <param name="expire-seconds" value="600"/>
>>>             <param name="register" value="false"/>
>>>             <param name="retry-seconds" value="30"/>
>>>             <param name="extension" value="5000"/>
>>>             <param name="context" value="public"/>
>>>             </gateway>
>>>             </gateways>
>>>
>>>
>>>             If I define a gateway called "gxw4104", then this is
>>>             what I think a simple dialplan should look like but I'm
>>>             not sure of the gateway details in the "bridge" section
>>>             of the definition.
>>>
>>>             <extension name="gxw4104_out">
>>>             <condition field="destination_number"
>>>             expression="^(\d{10})$">
>>>             *<action application="bridge"
>>>             data="sofia/gateway/gxw4104/........"/>        (what
>>>             should this be???)*
>>>             </condition>
>>>             </extension>
>>>
>>>             Am I moving in the right direction and can someone fill
>>>             in the blanks for me please
>>>
>>>             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
>>>
>>>
>>>
>>>
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
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From ira at connectmevoice.com  Tue Sep 20 17:03:09 2011
From: ira at connectmevoice.com (Ira Tessler)
Date: Tue, 20 Sep 2011 09:03:09 -0400
Subject: [Freeswitch-users] External IVR and MWI
Message-ID: <2201b1154e174794473a5792a0649a60@mail.gmail.com>

I am planning to use an external IVR with a Freeswitch system. The phones
will register with Freeswitch and forward to the external IVR for
voicemails. When the external IVR receives a voicemail, I need to send a
message to Freeswitch to turn on the message waiting indicator on the phone.
I will be using a Multi Tenant Freeswitch system. The code will need to be
programmed in C#.



What is the best way to go about this? Using the ESL API? Does anyone have
any examples?



Thanks,



Ira Tessler
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From gcd at i.ph  Tue Sep 20 17:42:44 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 20 Sep 2011 21:42:44 +0800
Subject: [Freeswitch-users] GXW4104 gateway setup for outgoing calls
In-Reply-To: <4E7881BD.70306@the800group.com>
References: <4E6EFCFA.6000803@the800group.com>
	<CAHKqMb7vF4ULKjWsHqP20xVRy0oOrDo9A3fpwCBr2NXAjsJ6iw@mail.gmail.com>
	<4E6F2C07.80800@the800group.com>
	<CAHKqMb7BV1oVa5NqvZP+mgS1QSjuaUu-JPpHRFpqmNBs8G8mHQ@mail.gmail.com>
	<4E72DB25.7010706@the800group.com>
	<CAHKqMb69+OqObSnJdGEOVwreahbaw0SEinJgyixYgzG9FG3UdQ@mail.gmail.com>
	<4E7881BD.70306@the800group.com>
Message-ID: <CAHKqMb5Fb1f_gmBOhiKhZ9gG3Qbjogy5kWeDwF-CXz7y+Jq1dg@mail.gmail.com>

perhaps, your dialtone parameters are wrong causing the problem. try to
enable it w/ the right parameters.
-nandy


On Tue, Sep 20, 2011 at 8:06 PM, ocset <ocset at the800group.com> wrote:

>  I fixed the problem.
>
> The guide I followed for setting up the incoming calls suggested this
> setting:  "Wait for Dial-Tone(Y/N): ch1-4:Y;"
>
> I changed this to "N" and now it works. Not sure how that affects the use
> of the system but at least incoming call are now working.
>
> Thanks for all your help.
>
>
>
> On 09/16/2011 03:35 PM, Nandy Dagondon wrote:
>
> pls see my comments below. you're welcome.
> -nandy
>
> On Fri, Sep 16, 2011 at 1:14 PM, ocset <ocset at the800group.com> wrote:
>
>>  Hi Nandy
>>
>> Thanks for your help so far - unfortunately this is still not working.
>>
>> Based on your examples, I have created the following  two files
>>
>> 1.
>> ../sip_profile/internal/01_custom.xml
>>
>> <include>
>>   <param name="multiple-registrations" value="true"/>
>>   <param name="accept-blind-reg" value="true"/>
>>   <param name="accept-blind-auth" value="true"/>
>>
>>   <gateway name="gxw4104-fxo">
>>
>>      <param name="username" value="1019"/>
>>       <param name="password" value="1234"/>
>>       <param name="realm" value="192.168.0.160"/>
>>      <param name="sip-port" value="5060"/>
>>      <param name="rtp_ip" value="192.168.0.160"/>
>>
>>      <param name="dtmf-type" value="rfc2833"/>
>>      <param name="expire-seconds" value="600"/>
>>      <param name="register" value="false"/>
>>      <param name="caller-id-in-from" value="false"/>
>>   </gateway>
>> </include>
>>
>>  2.
>> ../dialplan/default/01_custom.xml
>>
>> <include>
>>   <extension name="gxw4104-fxo-local">
>>
>>     <condition field="${toll_allow}" expression="local"/>
>>      <condition field="destination_number" expression="^(\d{10})$">
>>     <action application="set"
>> data="effective_caller_id_number=5555555555"/>
>>
>>     <action application="set"
>> data="effective_caller_id_name=ThisIsMyCompany"/>
>>     <action application="set" data="ignore_early_media=ring_ready"/>
>>     <action application="set" data="ringback=${us-ring}"/>
>>        <action application="bridge" data=
>> "sofia/gateway/gxw4104-fxo/$1 at 192.168.0.160:5060"<sofia/gateway/gxw4104-fxo/$1 at 192.168.0.160:5060>
>> />
>>     </condition>
>>   </extension>
>> </include>
>>
>>
>>
>> Command "sofia status"  shows the followiing
>>
>> internal                               profile
>> sip:mod_sofia at 192.168.0.23:5060         RUNNING (0)
>> internal::gxw4104-fxo        gateway             sip:1019 at 192.168.0.160
>> NOREG
>> external                              profile
>> sip:mod_sofia at 192.168.0.23:5080         RUNNING (0)
>> external::example.com      gateway             sip:joeuser at example.com
>> NOREG
>> internal-ipv6                       profile                sip:mod_sofia@[::1]:5060
>> RUNNING (0)
>> 192.168.0.23                      alias
>> internal                                                    ALIASED
>>
>>
>> Here is the call log when I try do dial out:
>>
>> 2011-09-16 12:57:30.903616 [NOTICE] switch_channel.c:669 New Channel
>> sofia/internal/1014 at 192.168.0.23 [0a83e803-b3fc-4a46-bc74-9d6786dac8e2]
>> 2011-09-16 12:57:30.933863 [INFO] mod_dialplan_xml.c:418 Processing User
>> 1014->0412345678 in context default
>> 2011-09-16 12:57:30.950359 [NOTICE] switch_channel.c:669 New Channel
>> sofia/internal/0412345678 at 192.168.0.160:5060[24111b8d-25f0-4433-a49f-88973730ebfb]
>> 2011-09-16 12:57:40.957391 [NOTICE] sofia.c:4789 Hangup
>> sofia/internal/0412345678 at 192.168.0.160:5060 [CS_CONSUME_MEDIA]
>> [NORMAL_TEMPORARY_FAILURE]
>> 2011-09-16 12:57:40.958554 [NOTICE] switch_core_session.c:1182 Session 7
>> (sofia/internal/0412345678 at 192.168.0.160:5060) Ended
>> 2011-09-16 12:57:40.958554 [NOTICE] switch_core_session.c:1184 Close
>> Channel sofia/internal/0412345678 at 192.168.0.160:5060 [CS_DESTROY]
>> 2011-09-16 12:57:40.958554 [INFO] mod_dptools.c:2355 Originate Failed.
>> Cause: NORMAL_TEMPORARY_FAILURE
>> 2011-09-16 12:57:40.958554 [NOTICE] mod_dptools.c:2418 Hangup
>> sofia/internal/1014 at 192.168.0.23 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
>> 2011-09-16 12:57:40.995191 [NOTICE] switch_core_session.c:1182 Session 6
>> (sofia/internal/1014 at 192.168.0.23) Ended
>> 2011-09-16 12:57:40.995191 [NOTICE] switch_core_session.c:1184 Close
>> Channel sofia/internal/1014 at 192.168.0.23 [CS_DESTROY]
>>
>>
>>
>> I have the following questions
>>
>> 1. What is the significance of the user 1019? I have a default install of
>> FS so that user does exist but I am not logged in as that user on my sip
>> phone. I am logged in as user 1014.
>>
>>  1019 is the user account for the FXO port to handle incoming calls. so,
> the login details must be set on directory/default/1019.xml.
>
> actually,  the user and password entries can be deleted because FS is not
> registering to the FXO gateway.
>
>
>>  2. The resultant log does not show my gateway being used but instead
>> shows "/sofia/internal/0412345678 at 192.168.0.160:5060"</sofia/internal/0412345678 at 192.168.0.160:5060>.
>> Is that expected behaviour?
>>
>>  it's not hitting the dialplan, i guess. i suggest you create prefix 9
> for PSTN calls e.g.
>     <condition field="destination_number" expression="^9(\d+)$">
>
> and make the file ../dialplan/default/01_custom.xml is on top of
> dialplan/default directory so it will be scanned early.
>
>
>>  3. I assumed that the IP address 192.168.0.9 in your example is the
>> address of your HT503 and not FS. I have thus replaced it with the IP
>> address from my GXW4104 (192.168.0.160). Is that correct?
>>
>>   that is correct.
>
>>
>>
>>
>>
>>
>>
>> On 09/13/2011 11:42 PM, Nandy Dagondon wrote:
>>
>> i inserted my answers to your questions below. for point #3), here's an
>> example how i configured my FXO port of ht503.
>>
>> included in  sip_profile/internal:
>> <include>
>>   <gateway name="ht503-fxo">
>>   <param name="username" value="1019"/>   <-- it's registered to receive
>> incoming calls
>>   <param name="realm" value="192.168.0.9"/>
>>   <param name="sip-port" value="5062"/>   <-- port 5060 is set to the FXS
>> port
>>   <param name="password" value="1234"/>
>>   <param name="rtp_ip" value="192.168.0.9"/>
>>   <param name="dtmf-type" value="rfc2833"/>
>>   <param name="expire-seconds" value="600"/>
>>   <param name="register" value="false"/>
>>   <param name="caller-id-in-from" value="false"/>
>>   </gateway>
>> </include>
>>
>> included  in dialplan/default
>>
>> <include>
>>   <extension name="ht503-fxo-local">
>>     <condition field="${toll_allow}" expression="local"/>
>>     <condition field="destination_number" expression="^9([2-9]\d{6})$">
>>     <action application="set" data="effective_caller_id_number=0321234567
>> "/>
>>     <action application="set" data="effective_caller_id_name=ThisIsMy
>> Company"/>
>>     <action application="set" data="ignore_early_media=ring_ready"/>
>>     <action application="set" data="ringback=${us-ring}"/>
>>       <action application="bridge" data="sofia/gateway/ht503-fxo/$
>> 1 at 192.168.0.9:5062"/>
>>     </condition>
>>   </extension>
>> </include>
>>
>> it looks you can create 4 internal gateways for the every port, fxo-1 to
>> fxo-4, w/ the same realm/rtp_ip values but setting different sip-port
>> values. then  your bridge app would be:
>>
>>       <action application="bridge"
>> data="sofia/gateway/fxo-1/$1|sofia/gateway/fxo-2/$1|sofia/gateway/fxo-3/$1sofia/gateway/fxo-4/$1"/>
>>
>> if u want to dialout any free port.
>>
>> i haven't tested the above. just try it. i hope it works.
>>
>> -nandy
>>
>>
>> On Tue, Sep 13, 2011 at 6:10 PM, ocset <ocset at the800group.com> wrote:
>>
>>>  Hi Nandy
>>>
>>> Thanks for your reply. I assume 192.168.0.9 in your example is the IP
>>> address of the GXW4104?
>>>
>> yes.
>>
>>>
>>>
>>  Some more questions
>>>
>>> 1. When you say port number, is this something I should be setting up on
>>> the GXW4104 so that it is listening on those 4 port numbers? If yes, what
>>> would be the setting I am looking for?
>>>
>> not for every port. the gateway has a base port number e.g. 5060 for
>> port#1. add 2 to the subsequent ports e.g. 5062 for port#2 and so on. this
>> is pointed out by sergey.
>>
>>>
>>> 2. Does that mean I don't define a new gateway in FreeSWITCH?
>>>
>> it's an option. but defining a gateway is cleaner.
>>
>>
>>>
>>>
>>  3. In your example, you said the bridge data would be
>>> 7654321 at 192.168.0.9:5063. What would the whole line look like in the
>>> dialplan?*
>>>
>>> <action application="bridge" data="
>>> sofia/gateway/7654321 at 192.168.0.9:5063**"/>*
>>>
>>> Still very confused :-)
>>>
>>> Thanks
>>>
>>>
>>> On 09/13/2011 03:45 PM, Nandy Dagondon wrote:
>>>
>>> hi,
>>>
>>> if GWX4104 is in your local network, use the internal profile for the
>>> gateway. register your FXO accounts to receive incoming calls (i think you
>>> did this already).
>>>
>>> to dialout the ports, specify the port number 5060~5063 assuming Port1
>>> starts at 5060. to dialout via port4, the bridge data should look like:
>>>
>>> 7654321 at 192.168.0.9:5063
>>>
>>> hope it helps.
>>>
>>> -nandy
>>>
>>>
>>> On Tue, Sep 13, 2011 at 2:49 PM, ocset <ocset at the800group.com> wrote:
>>>
>>>>  Hi
>>>>
>>>> I have recently bought a Grandstream GXW4104 (4 FXO ports) and need some
>>>> help setting up a gateway to call out using the GXW4104. I am really out of
>>>> my depth here and may be looking at this the wrong way so please bear with
>>>> me.
>>>>
>>>> I followed the advice on this website
>>>> "http://www.timhunt.net/wiki/FreeSwitch:GXW4104"<http://www.timhunt.net/wiki/FreeSwitch:GXW4104>and incoming calls from a PSTN line are working great. Now I need to setup a
>>>> dialplan so that outgoing calls are routed through the same PSTN line on the
>>>> GXW4104. I will eventually have 4 PSTN lines with a dialplan to use the
>>>> first available line (if that is possible).
>>>>
>>>> According to the FreeSWITCH 1.0.6 book (and many online posts) I need to
>>>> create a gateway and a dialplan but all the gateway examples are for SIP
>>>> accounts.
>>>>
>>>> So, the gateway definition seems to need a username and password but the
>>>> GXW4104 does not have that capability. I found this gateway definition in
>>>> the  freeswitch.xml.fsxml file but am not sure how many of these variables
>>>> are required.
>>>>
>>>> <gateways>
>>>>    <gateway name="example.com">
>>>>       <param name="username" value="joeuser"/>
>>>>       <param name="password" value="password"/>
>>>>       <param name="from-user" value="joeuser"/>
>>>>       <param name="from-domain" value="example.com"/>
>>>>       <param name="expire-seconds" value="600"/>
>>>>       <param name="register" value="false"/>
>>>>       <param name="retry-seconds" value="30"/>
>>>>       <param name="extension" value="5000"/>
>>>>       <param name="context" value="public"/>
>>>>    </gateway>
>>>> </gateways>
>>>>
>>>>
>>>> If I define a gateway called "gxw4104", then this is what I think a
>>>> simple dialplan should look like but I'm not sure of the gateway details in
>>>> the "bridge" section of the definition.
>>>>
>>>> <extension name="gxw4104_out">
>>>>    <condition field="destination_number" expression="^(\d{10})$">
>>>>
>>>>       *<action application="bridge"
>>>> data="sofia/gateway/gxw4104/........"/>        (what should this be???)
>>>> *
>>>>    </condition>
>>>> </extension>
>>>>
>>>> Am I moving in the right direction and can someone fill in the blanks
>>>> for me please
>>>>
>>>> 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
>>>>
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From kris at kriskinc.com  Tue Sep 20 17:55:39 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 20 Sep 2011 09:55:39 -0400
Subject: [Freeswitch-users] Manipulate sip error codes
In-Reply-To: <1D10AB188D6CCA46BB4369E3268E36EF359A38@SVR01.ghosttelecom.local>
References: <Acx3iyrBjVQzk4+pSSurt4lpUfwy2A==>
	<1D10AB188D6CCA46BB4369E3268E36EF359A38@SVR01.ghosttelecom.local>
Message-ID: <CAKDfjgdZwoBC9SxNY3Zov3REa229ZYxH2CKmxUtbkQx-tjd52Q@mail.gmail.com>

Chris,

  <action application="set" data="continue_on_fail=true"/>
  <action application="bridge" data="bridge/string/to/somewhere"/>
  <action application="respond" data="486"/>

  This will cause any failed bridge to respond 486 on the A leg.  This
syntax can be made conditional using some dialplan tricks...

On Tue, Sep 20, 2011 at 7:48 AM, Chris Martineau <chris at ghosttelecom.com> wrote:
>
>
> Hi,
>
>
>
> Is it possible to translate B leg sip error codes so that the A leg always
> gives a fixed error code?
>
>
>
> i.e make call and get 408 for instance from the far end but would like 486
> to be sent on the a leg.
>
>
>
> Need this as one of our suppliers retransmits a call multiple times unless
> they recieve a 403 or 486. No major issue but just clogs the error cdrs with
> loads of duplicate records otherwise.
>
>
>
> Any ideas would be greatly appreciated.
>
>
>
> Many thanks
>
>
>
> Chris
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From kris at kriskinc.com  Tue Sep 20 17:57:17 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 20 Sep 2011 09:57:17 -0400
Subject: [Freeswitch-users] Sophia authentication without using editing
	acl.conf.xml
In-Reply-To: <CALH5ZrYStTeu-ttpvYDqOh1CRy-ZuZ-zaUYYjso9WeEUAaA1iQ@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>
	<CA+GmxbYRhaLOZm7ami4zEa9nkMYESE24baK-13vbXdzcH-3PrA@mail.gmail.com>
	<CALH5ZrYpQrnGd3kB_zY4+j_ct0QbmJHPKR3FEUP5qLeyA+RWkQ@mail.gmail.com>
	<CAKDfjgcXbFZa=iZJ=0DOu+kLoXcvoZXrVo3K-KPSU_Q6bREU_Q@mail.gmail.com>
	<CALH5Zra=NZyXeOPEP3HTDhRV7Eg+8SauEdB2oVB9NbGhADuawA@mail.gmail.com>
	<CALH5ZrYStTeu-ttpvYDqOh1CRy-ZuZ-zaUYYjso9WeEUAaA1iQ@mail.gmail.com>
Message-ID: <CAKDfjgf+wseuJpc8Fxpsxq_YUv87JWQ5aBNVQ8QGL3HU3QhxqA@mail.gmail.com>

Comment inbound ACL or apply and ACL that defaults to accept.

On Tue, Sep 20, 2011 at 12:15 AM, Joe Flemmings
<joe.jflemmings at gmail.com> wrote:
> How do you disable ACL in SIP Profile?
>
> On Mon, Sep 19, 2011 at 12:45 PM, Joe Flemmings <joe.jflemmings at gmail.com>
> wrote:
>>
>> Thank You,
>>
>> Did not know you can disable ACL, will try this.
>>
>> On Mon, Sep 19, 2011 at 8:13 AM, Kristian Kielhofner <kris at kriskinc.com>
>> wrote:
>>>
>>> Disable the ACL on the SIP profile and:
>>>
>>> 1) ?Do IP checking using conditions in the dialplan and use respond to
>>> require auth/reject calls when you need it. ?This will still require a
>>> reloadxml when you make changes.
>>> 2) ?Use xml_curl like msc said and look at sip_from_host and use
>>> respond (and xml_curl) to reject/auth calls when you need it. ?This
>>> will not require any reloading when you auth calls.
>>>
>>> On Sat, Sep 17, 2011 at 3:44 AM, Joe Flemmings <joe.jflemmings at gmail.com>
>>> wrote:
>>> > Is their a solution without reloading acl?
>>> >
>>> > On Fri, Sep 9, 2011 at 4:16 AM, Dmitry Sytchev <kbdfck at gmail.com>
>>> > wrote:
>>> >>
>>> >> 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
>>> >>
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Kristian Kielhofner
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From nagalenoj at gmail.com  Tue Sep 20 18:19:45 2011
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Tue, 20 Sep 2011 19:49:45 +0530
Subject: [Freeswitch-users] DTMF issue when using execute_extension with
	play_and_get_digits
In-Reply-To: <CAA1Qw2sLHtu8ZHQn7GJFOs=f2k9H2pLa+mgu6Kt+FEtGcwWBOw@mail.gmail.com>
References: <CAA1Qw2sLHtu8ZHQn7GJFOs=f2k9H2pLa+mgu6Kt+FEtGcwWBOw@mail.gmail.com>
Message-ID: <CAA1Qw2usLq9rYHparEm_BdR6Ers8vo8C8xUej6bYhXzHqWJkKw@mail.gmail.com>

Hi,
   I've updated the source to latest Git and checked the same. It is not
working. Extension 4567 is not receiving the DTMFs given for play_and_get_
digits application.


On Tue, Aug 9, 2011 at 8:37 PM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Hi Friends,
>      Facing an issue when using bind_meta_app and execute_extension(with
> play_and_get_digits) combined.
>
> Here is my dialplan,
> <condition field="destination_number" expression="^4567$" break="on-true">
> <action application="play_and_get_digits" data="1 5 3 7000 #
> '/voice_files/A.wav' '/voice_files/failed.wav' dtmf_received \d+"/>
> </condition>
> <condition field="destination_number" expression="^.*$" break="on-true">
> <action application="answer"/>
> <action application="bind_meta_app" data="5 b o execute_extension::4567 XML
> default"/>
> <action application="bridge" data="freetdm/1/a/9952248266"/>
> <action application="park"/>
> </condition>
>
> So, when callee enters *5,  I want the caller to enter a number. I get the
> extension executed as expected. The caller is able to hear the voice file
> played and when he enters the digits, it is not received. Digits are not
> even present in FS log.
>
> In the normal cases, there is no issues in getting DTMFs. I don't know,
> what am I doing wrong here. Kindly, help me to resolve this.
>
> --
> Regards,
> Nagalenoj H.
>



-- 
Regards,
Nagalenoj H.
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From chris at ghosttelecom.com  Tue Sep 20 19:52:31 2011
From: chris at ghosttelecom.com (Chris Martineau)
Date: Tue, 20 Sep 2011 16:52:31 +0100
Subject: [Freeswitch-users] Manipulate sip error codes
In-Reply-To: <CAKDfjgdZwoBC9SxNY3Zov3REa229ZYxH2CKmxUtbkQx-tjd52Q@mail.gmail.com>
References: <Acx3iyrBjVQzk4+pSSurt4lpUfwy2A==><1D10AB188D6CCA46BB4369E3268E36EF359A38@SVR01.ghosttelecom.local>
	<CAKDfjgdZwoBC9SxNY3Zov3REa229ZYxH2CKmxUtbkQx-tjd52Q@mail.gmail.com>
Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF359A74@SVR01.ghosttelecom.local>

Thanks for that,

Could you help with the following?

Trying to play a welcome announcement to the a leg on receiving a 183
before the main early media from the far end. Sort of have it working
but it tends to destroy the call or kills all media.

Have tried a straight playback just before the bridge but this kills the
call when it completes.
Have tried the execute on media but that kills all media so not sure if
the syntax is correct.

Do you have any examples how this can be achieved?

Ideally it would be useful to wait for incoming media from the a leg to
confirm a media path through a nat but not sure if this is possible.

Many thanks for any help you can offer.

Regards


Chris

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Kristian Kielhofner
Sent: 20 September 2011 14:56
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Manipulate sip error codes

Chris,

  <action application="set" data="continue_on_fail=true"/>
  <action application="bridge" data="bridge/string/to/somewhere"/>
  <action application="respond" data="486"/>

  This will cause any failed bridge to respond 486 on the A leg.  This
syntax can be made conditional using some dialplan tricks...

On Tue, Sep 20, 2011 at 7:48 AM, Chris Martineau
<chris at ghosttelecom.com> wrote:
>
>
> Hi,
>
>
>
> Is it possible to translate B leg sip error codes so that the A leg
always
> gives a fixed error code?
>
>
>
> i.e make call and get 408 for instance from the far end but would like
486
> to be sent on the a leg.
>
>
>
> Need this as one of our suppliers retransmits a call multiple times
unless
> they recieve a 403 or 486. No major issue but just clogs the error
cdrs with
> loads of duplicate records otherwise.
>
>
>
> Any ideas would be greatly appreciated.
>
>
>
> Many thanks
>
>
>
> Chris
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


FreeSWITCH-users mailing list
FreeSWITCH-users at 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 robert.hadley at teotech.com  Tue Sep 20 20:06:42 2011
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 20 Sep 2011 09:06:42 -0700
Subject: [Freeswitch-users] Making changes to fifo.conf.xml
	requires	restarting FS?
In-Reply-To: <33095823FD21DF429B481B5163264B795118922232@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B795118922232@VMBX102.ihostexchange.net>
Message-ID: <E826C94A3C6BE44198FAF4D66B5AB4D81BDED675F0@VA3DIAXVS351.RED001.local>


After making changes, from FS CLI, try
reload mod_fifo


-----Original Message-----
From: Yungwei Chen [mailto:yungwei at resolvity.com] 
Sent: Monday, September 19, 2011 3:55 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Making changes to fifo.conf.xml requires restarting FS?

Hi,

I noticed that making changes to fifo.conf.xml requires FS to be restarted, and reloadxml command doesn't seem to reload fifo.conf.xml.
Am I missing something? Thanks.



From cvogel at lyonl.com  Tue Sep 20 20:34:27 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Tue, 20 Sep 2011 16:34:27 +0000
Subject: [Freeswitch-users] Gateway Authentication
Message-ID: <D2DD0DB0-1560-459E-9194-9ADB754B534B@lyonl.com>

Hello, I'm trying to make the move from Asterix, but I'm running into some difficulties. I'm try to bridge a call using our gateway however it doesn't work. In wireshark I can see I'm getting an SIP 401 Unauthorized error with a WWW-Authenticate header, after FS send the INVITE message to the gateway. However FS doesnt seem to respond to the request for Authentication.  Asterix responds correctly however I cant seem to make FS to do the same. Any help would be appreciated


INVITE sip:+15618911806 at 4.55.35.60:5070 SIP/2.0
Via: SIP/2.0/UDP 207.67.30.226;rport;branch=z9hG4bKB49SZQHrgaaKc
Max-Forwards: 8
From: "LyonL" <sip:+14142211800 at 207.67.30.226:5060>;tag=eZe8gcQgXXv5c
To: <sip:+15618911806 at 4.55.35.60:5070>
Call-ID: 4f8edae0-5e45-122f-6399-07d4dbeff43f
CSeq: 17931324 INVITE
Contact: <sip:1-F2la9 at 207.67.30.226:5060;transport=udp;gw=level3>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 227
X-FS-Support: update_display
Remote-Party-ID: "LyonL" <sip:+14142211800 at 207.67.30.226:5060>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1316516832 1316516833 IN IP4 10.126.200.6
s=FreeSWITCH
c=IN IP4 10.126.200.6
t=0 0
m=audio 17944 RTP/AVP 0 8 18 101 13
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20


SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 207.67.30.226;received=207.67.30.226;branch=z9hG4bKB49SZQHrgaaKc;rport=42534
From: "LyonL" <sip:+14142211800 at 207.67.30.226:5060>;tag=eZe8gcQgXXv5c
To: <sip:+15618911806 at 4.55.35.60:5070>;tag=SD6soqf99-1367649635-1316534779161
Call-ID: 4f8edae0-5e45-122f-6399-07d4dbeff43f
CSeq: 17931324 INVITE
WWW-Authenticate: DIGEST qop="auth",nonce="BroadWorksXgst2td09Tbihi2qBW",algorithm=MD5,realm="BroadWorks"
Content-Length: 0

  <include>
    <extension name="4142211800">
      <condition field="destination_number" expression="^(\+?1)?(4142211800)$">
        <action application="set" data="effective_caller_id_name=LyonL"/>
        <action application="set" data="effective_caller_id_number=+14142211800"/>
        <action application="bridge" data="sofia/gateway/level3/+15618911806"/>
      </condition>
    </extension>
  </include>

<include>
  <gateway name="level3">
    <param name="apply-inbound-acl" value="level3"/>
    <param name="username" value="1-F2la9"/>
    <param name="password" value="password"/>>
    <param name="realm" value="BroadWorks"/>
    <param name="proxy" value="4.55.35.60:5070"/>
    <param name="from-domain" value="207.67.30.226:5060"/>
    <param name="dtmf-type" value="rfc2833"/>
    <param name="extension-in-contact" value="true"/>
    <param name="caller-id-in-from" value="true"/>
    <param name="register" value="false"/>
  </gateway>
</include>

<profile name="external">
  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
  <!-- This profile is only for outbound registrations to providers -->
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>

  <aliases>
    <!--
    <alias name="outbound"/>
    <alias name="nat"/>
    -->
  </aliases>

  <domains>
    <domain name="all" alias="false" parse="true"/>
  </domains>

  <settings>
    <param name="debug" value="0"/>
<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
<!-- <param name="shutdown-on-fail" value="true"/> -->
    <param name="sip-trace" value="no"/>
    <param name="sip-capture" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="$${external_sip_port}"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="rtp-timer-name" value="soft"/>
    <!--<param name="enable-100rel" value="true"/>-->
    <!--<param name="disable-srv503" value="true"/>-->
    <!-- This could be set to "passive" -->
    <param name="local-network-acl" value="localnet.auto"/>
    <param name="manage-presence" value="false"/>

    <!-- used to share presence info across sofia profiles
manage-presence needs to be set to passive on this profile
if you want it to behave as if it were the internal profile
for presence.
    -->
    <!-- Name of the db to use for this profile -->
    <!--<param name="dbname" value="share_presence"/>-->
    <!--<param name="presence-hosts" value="$${domain}"/>-->
    <!--<param name="force-register-domain" value="$${domain}"/>-->
    <!--all inbound reg will stored in the db using this domain -->
    <!--<param name="force-register-db-domain" value="$${domain}"/>-->
    <!-- ************************************************* -->

    <!--<param name="aggressive-nat-detection" value="true"/>-->
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="false"/>
    <!--<param name="challenge-realm" value="auto_from"/>-->
    <param name="user-agent-string" value="FreeSWITCH"/>
    <!--
DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
    -->
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
    <!--<param name="enable-3pcc" value="true"/>-->

    <!-- TLS: disabled by default, set to "true" to enable -->
    <param name="tls" value="$${external_ssl_enable}"/>
    <!-- additional bind parameters for TLS -->
    <param name="tls-bind-params" value="transport=tls"/>
    <!-- Port to listen on for TLS requests. (5081 will be used if unspecified) -->
    <param name="tls-sip-port" value="$${external_tls_port}"/>
    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
    <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
    <param name="tls-version" value="$${sip_tls_version}"/>

  </settings>
</profile>

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From dimskraft at gmail.com  Tue Sep 20 20:58:54 2011
From: dimskraft at gmail.com (Dmitry Kravchenko)
Date: Tue, 20 Sep 2011 20:58:54 +0400
Subject: [Freeswitch-users] Authenticate any user free and by demand?
Message-ID: <CADDV4gyy=4VYhSUufVKPPnshuKMCMuRN7F-9UNjcSuqB1G+2kw@mail.gmail.com>

Is it possible to configure freeswitch to authenticate any SIP user
connected to it, regardless it's password and presense in freeswitch
configs, receiving it's name from SIP phone and allowing calls to that
names?
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From fizban79 at gmail.com  Tue Sep 20 18:36:37 2011
From: fizban79 at gmail.com (Igor Borisov)
Date: Tue, 20 Sep 2011 21:36:37 +0700
Subject: [Freeswitch-users] Stuck registration.
Message-ID: <4E78A4F5.804@gmail.com>

Hello!

Trying to resolve problem with text messages passing between SIP users 
I've noticed strange registration:

freeswitch at internal> sofia status profile internal

Call-ID:        ArOr7DiiGiLEUoA0..mgUbX30vJ7p3V2
User:           9001 at static.XX.XX.XX.X.clients.your-server.de
Contact:        "Alexander" <sip:9001 at YY.YY.YY.YY:52121;transport=tls>
Agent:          Bria iPhone 1.3.1
Status:         Registered(TLS)(unknown) EXP(2011-08-22 18:07:11) 
EXPSECS(-2506676)
Host:           static.34.160.4.46.clients.your-server.de
IP:             YY.YY.YY.YY
Port:           52121
Auth-User:      9001
Auth-Realm:     static.XX.XX.XX.X.clients.your-server.de
MWI-Account:    9001 at static.XX.XX.XX.X.clients.your-server.de

This registration is expired long time ago but still exists in the list. 
Registration from correct user with login 9001 exists too. I suppose it 
can be cause of messaging problems.
Profile reload or FreeSwitch restart doesn't help - registration remains 
in the list.
How can I get rid of it and avoid its appearance in the future?

Regards,
Igor A. Borisov

-- 
? ?????????,
??????? ?????

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From Nabble_01394 at slickdeals.endjunk.com  Wed Sep 21 04:34:12 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Tue, 20 Sep 2011 17:34:12 -0700 (PDT)
Subject: [Freeswitch-users] Missing mod_blacklist.conf file?
Message-ID: <1316565252256-6814430.post@n2.nabble.com>

I took a look at and studied the applications/mod_blacklist.c file which was
recently committed to FS git source. I found a reference to access
mod_blacklist.conf file on line #116 of the applications/mod_blacklist.c
file; however, I searched through my local FS source directories and could
not find the mod_blacklist.conf file. So, anyone here who has used
mod_blacklist can point to me where I can get a copy of mod_blacklist.conf
file?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Missing-mod-blacklist-conf-file-tp6814430p6814430.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From danlanweb at gmail.com  Wed Sep 21 04:53:53 2011
From: danlanweb at gmail.com (Dan Lan)
Date: Tue, 20 Sep 2011 17:53:53 -0700
Subject: [Freeswitch-users] Goto Voicemail When Network Down
Message-ID: <CAAuJ8MBBuSo4WhoGRqWy-8fm9KNV4pWfF1qzTrwRpjwCNZECUA@mail.gmail.com>

Hi,
I install the Freeswitch without much modification.
I have an extension, and with incoming DID transfer to that extension.
Some Test Scenarios
1. When the extension is registered, the incoming call will go to VM after
30 secs, if no one answer the call (This is normal)
2. When the extension is not registered, the incoming call will go directly
to VM
3. However, if the endpoint is registered, but for some reason, the
endpoint's network is down. The incoming call will just go to blank
scilience. after about 60 secs then I can hear the busy signal.

>From the packet capture, I can see FS try to reach the end point by sending
INVITE during the 60 secs, but because the end point is not responding, so
the FS keep sending INVITE.

I am just wondering, are there anyway to break the loop? see, if the FS send
INVITE without response for some certain time, then go to Voice Mail without
the caller waiting and dont know what to do?

Dan
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From curriegrad2004 at gmail.com  Wed Sep 21 05:21:48 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 20 Sep 2011 18:21:48 -0700
Subject: [Freeswitch-users] Goto Voicemail When Network Down
In-Reply-To: <CAAuJ8MBBuSo4WhoGRqWy-8fm9KNV4pWfF1qzTrwRpjwCNZECUA@mail.gmail.com>
References: <CAAuJ8MBBuSo4WhoGRqWy-8fm9KNV4pWfF1qzTrwRpjwCNZECUA@mail.gmail.com>
Message-ID: <CAMWiMy-bP+eA=vzGoNGs4ATPN2HTY+VUQNJ6BWnwM4RbFwrLRQ@mail.gmail.com>

You can insert this in the dialplan:
<application set="call_timeout=[x seconds here]" />
<application set="continue_on_fail=true" />
<application set="instant_ringback=true" />

and FreeSWITCH will wait 60 seconds before failing the call and move
on with the next item on the dialplan, give the user a ringback tone
(depending on what you choose earlier that is) regardless if the
endpoint is ringing or not. Hope this solves your problem

On Tue, Sep 20, 2011 at 5:53 PM, Dan Lan <danlanweb at gmail.com> wrote:
> Hi,
> I install the Freeswitch without much modification.
> I have an extension, and with incoming DID transfer to that extension.
> Some Test Scenarios
> 1. When the extension is registered, the incoming call will go to VM after
> 30 secs, if no one answer the call (This is normal)
> 2. When the extension is not registered, the incoming call will go directly
> to VM
> 3. However, if the endpoint is registered, but for some reason, the
> endpoint's network is down. The incoming call will just go to blank
> scilience. after about 60 secs then I can hear the busy signal.
>
> From the packet capture, I can see FS try to reach the end point by sending
> INVITE during the 60 secs, but because the end point is not responding, so
> the FS keep sending INVITE.
>
> I am just wondering, are there anyway to break the loop? see, if the FS send
> INVITE without response for some certain time, then go to Voice Mail without
> the caller waiting and dont know what to do?
>
> Dan
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 lfurrea at gmail.com  Wed Sep 21 06:53:52 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Tue, 20 Sep 2011 20:53:52 -0600
Subject: [Freeswitch-users] FreeTDM kickstart help ERRORS inline
Message-ID: <CAB2KLyeJVd3mP+7LUSMTRq5sMgcoz5FoKFwTZukyk_dnyFMFQw@mail.gmail.com>

This dAHDI stuff is like voodoo!

I know this has to be configuration issues but sh****t after a few years in
the telecom industry it is really hard to find a more awful syntax for
configuring such a stupid device as an FXO port.

I am trying to get a Xorcom Astribank going with FreeTDM. So far after a
bunch of firmware and driver BS (come on there is probably nothing more
buggy in a freking kernel than the USB layer, how can a telecom device be
using a USB driver!!!!!!!).

I get the following output from

#lsdahdi
### Span  1: XBUS-00/XPD-00 "Xorcom XPD #00/00: FXO" (MASTER)
  1 FXO                     RED
  2 FXO                     RED
  3 FXO                     RED
  4 FXO                     RED
  5 FXO                     RED
  6 FXO                     RED
  7 FXO                     RED
  8 FXO                     RED
### Span  2: XBUS-00/XPD-10 "Xorcom XPD #00/10: FXO"
  9 FXO                     RED
 10 FXO                     RED
 11 FXO                     RED
 12 FXO                     RED
 13 FXO                     RED
 14 FXO                     RED
 15 FXO                     RED
 16 FXO                    (battery)
### Span  3: XBUS-00/XPD-20 "Xorcom XPD #00/20: FXO"
 17 FXO                     RED
 18 FXO                    (battery)
 19 FXO                    (battery)
 20 FXO                    (battery)
 21 FXO                    (battery)
 22 FXO                    (battery)
 23 FXO                    (battery)
 24 FXO                    (battery)

But when loading mod_freetdm I am getting:

2011-09-19 15:44:03.149105 [NOTICE] ftdm_io.c:5723 Modules configured: 1
2011-09-19 15:44:03.149105 [DEBUG] ftdm_config.c:80 Configuration file is
/usr/local/freeswitch/conf/freetdm.conf.
2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4633 Reading FreeTDM
configuration file
2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4649 found config for span
2011-09-19 15:44:03.149105 [NOTICE] ftmod_zt.c:1323 Using DAHDI control
device
2011-09-19 15:44:03.149105 [INFO] ftdm_io.c:4963 Loading IO from
/usr/local/freeswitch/mod/ftmod_zt.so [zt]
2011-09-19 15:44:03.149105 [DEBUG] ftdm_config.c:80 Configuration file is
/usr/local/freeswitch/conf/zt.conf.
2011-09-19 15:44:03.149105 [INFO] ftmod_zt.c:585 Setting rxgain val to
0.000000
2011-09-19 15:44:03.149105 [INFO] ftmod_zt.c:593 Setting txgain val to
0.000000
2011-09-19 15:44:03.149105 [INFO] ftdm_io.c:787 Auto-loaded I/O module 'zt'
2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4670 created span 1 (SPAN1) of
type zt
2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4688 span 1
[fxo-channel]=[17-24]
2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4718 setting trunk type to
'FXO' start(KEWL)
2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device
/dev/dahdi/channel chan 17 fd 48 (No such device or address)
2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device
/dev/dahdi/channel chan 18 fd 48 (No such device or address)
2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device
/dev/dahdi/channel chan 19 fd 48 (No such device or address)
2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device
/dev/dahdi/channel chan 20 fd 48 (No such device or address)
2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device
/dev/dahdi/channel chan 21 fd 48 (No such device or address)
2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device
/dev/dahdi/channel chan 22 fd 48 (No such device or address)
2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device
/dev/dahdi/channel chan 23 fd 48 (No such device or address)
2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device
/dev/dahdi/channel chan 24 fd 48 (No such device or address)
2011-09-19 15:44:03.149105 [ERR] ftdm_io.c:4579 1:Failed to configure span
2011-09-19 15:44:03.149105 [INFO] ftdm_io.c:4885 Configured 0 channel(s)
2011-09-19 15:44:03.149105 [ERR] ftdm_io.c:5733 FreeTDM global configuration
failed!
2011-09-19 15:44:03.149105 [ERR] mod_freetdm.c:4805 Error configuring
FreeTDM
2011-09-19 15:44:03.149105 [INFO] switch_time.c:1028 Timezone reloaded 530
definitions
2011-09-19 15:44:03.149105 [CRIT] switch_loadable_module.c:1019 Error
Loading module /usr/local/freeswitch/mod/mod_freetdm.so
**Module load routine returned an error**
2011-09-19 15:44:03.149105 [NOTICE] ftdm_sched.c:147 Main scheduling thread
going out ...


Thanks in advance and sorry for the rant.
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From bobc at devassert.com  Wed Sep 21 07:46:59 2011
From: bobc at devassert.com (Bob Coleman)
Date: Wed, 21 Sep 2011 15:46:59 +1200
Subject: [Freeswitch-users] External IVR and MWI
In-Reply-To: <2201b1154e174794473a5792a0649a60@mail.gmail.com>
References: <2201b1154e174794473a5792a0649a60@mail.gmail.com>
Message-ID: <CAN=3f1-fA9esoZM+2CyNdq-ibzU-z+btCckrOuVj+7ONW7=QDQ@mail.gmail.com>

Yes, use the ESL API.

Havent dealt with voicemail message waiting indicator before.

But it is pretty simple to hook up C# to freeswitch via esl api.

The ESL API just wraps up the socket communication from Freeswitch,
which is fairly easy to do yourself if you really have too as well.

Bob

On Wed, Sep 21, 2011 at 1:03 AM, Ira Tessler <ira at connectmevoice.com> wrote:
> I am planning to use an external IVR with a Freeswitch system. The phones
> will register with Freeswitch and forward to the external IVR for
> voicemails. When the external IVR receives a voicemail, I need to send a
> message to Freeswitch to turn on the message waiting indicator on the phone.
> I will be using a Multi Tenant Freeswitch system. The code will need to be
> programmed in C#.
>
> What is the best way to go about this? Using the ESL API? Does anyone have
> any examples?


From chris at ghosttelecom.com  Wed Sep 21 13:24:46 2011
From: chris at ghosttelecom.com (Chris Martineau)
Date: Wed, 21 Sep 2011 10:24:46 +0100
Subject: [Freeswitch-users] Early media playback
Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF359A8F@SVR01.ghosttelecom.local>

Hi

 

Trying to play a custom announcement/welcome to the a leg on receiving a
183 from the far end. 

Have tried a straight playback just before the bridge which confirms
that it finds the playback file as the recording is heard but as this is
a bit nasty FS complains with a codec error and closes the call.

Tried execute on media with the same playback details as follows...

<action application="set" data="execute_on_media=playback
/path/mediafile"

Which according to the logs seems to work but I get just silence until I
presume the file stops playing which is strange as the file plays fine
when using a forced playback?

Logs in both cases show the same file being found and used
mediafile.PCMA. 

 

Also I would like to terminate the playback on answer but cannot find
any obvious process to do this. Obviously you could do an
execute_on_answer but can't see what I would execute to achieve the
desired effect?

 

Many thanks for any help you can offer.

 

Regards

 

 

Chris

 

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From Stefan.Weigel at allianz-warranty.com  Wed Sep 21 17:12:46 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Wed, 21 Sep 2011 15:12:46 +0200
Subject: [Freeswitch-users] change sip_to_user in dialplan
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36DF1@AZWSMS03.azwarranty.int>

Hello all,



I'm trying to change some incoming calls 'sip_to_uri' (it's a rerouting of our ITSP) so it matches our main number:



0891234567 -> 08998765



I'm checking the sip_to_uri and then trying to export/set/set_profile_var to the new number. But in debug trace it's still comparing against the original sip_to_uri.



Is there a way to change as soon it's hitting the dialplan ?





Thanks and best regards



Stefan

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From avi at avimarcus.net  Wed Sep 21 17:20:03 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 21 Sep 2011 16:20:03 +0300
Subject: [Freeswitch-users] change sip_to_user in dialplan
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C04D3B36DF1@AZWSMS03.azwarranty.int>
References: <5003D7D3E06F514E8C682F18D223265C04D3B36DF1@AZWSMS03.azwarranty.int>
Message-ID: <CALNZuwnGjmSHTGrEfGzTAzUFu6EJ3qQPvYbWHedA_b3fm2Zd+w@mail.gmail.com>

Why bother? regex match on the destination_number and then bridge/transfer
accordingly. What are you trying to gain by messing with the actual
sip_to_user?
And perhaps set on the inbound gateway where it gets sent to..

-Avi

On Wed, Sep 21, 2011 at 4:12 PM, Weigel, Stefan <
Stefan.Weigel at allianz-warranty.com> wrote:

>  Hello all,****
>
> ** **
>
> I?m trying to change some incoming calls ?sip_to_uri? (it?s a rerouting of
> our ITSP) so it matches our main number:****
>
> ** **
>
> 0891234567 -> 08998765****
>
> ** **
>
> I?m checking the sip_to_uri and then trying to export/set/set_profile_var
> to the new number. But in debug trace it?s still comparing against the
> original sip_to_uri.****
>
> ** **
>
> Is there a way to change as soon it?s hitting the dialplan ?****
>
> ** **
>
> ** **
>
> Thanks and best regards****
>
> ** **
>
> Stefan ****
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fernando.berretta at gmail.com  Wed Sep 21 17:38:01 2011
From: fernando.berretta at gmail.com (Fernando Berretta)
Date: Wed, 21 Sep 2011 10:38:01 -0300
Subject: [Freeswitch-users] G.729
Message-ID: <4E79E8B9.1040402@gmail.com>

Hi,

Is there some way to install g729 codec in freepbx for free, in order to 
"try" it like is provided for asterisk ?

Best Regards,
Fernando


From kris at kriskinc.com  Wed Sep 21 17:54:41 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 21 Sep 2011 09:54:41 -0400
Subject: [Freeswitch-users] G.729
In-Reply-To: <4E79E8B9.1040402@gmail.com>
References: <4E79E8B9.1040402@gmail.com>
Message-ID: <CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>

This is a mailing list for FreeSWITCH discussion, not FreePBX.

On Wed, Sep 21, 2011 at 9:38 AM, Fernando Berretta
<fernando.berretta at gmail.com> wrote:
> Hi,
>
> Is there some way to install g729 codec in freepbx for free, in order to
> "try" it like is provided for asterisk ?
>
> Best Regards,
> Fernando
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


From kris at kriskinc.com  Wed Sep 21 18:03:50 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 21 Sep 2011 10:03:50 -0400
Subject: [Freeswitch-users] Early media playback
In-Reply-To: <1D10AB188D6CCA46BB4369E3268E36EF359A8F@SVR01.ghosttelecom.local>
References: <1D10AB188D6CCA46BB4369E3268E36EF359A8F@SVR01.ghosttelecom.local>
Message-ID: <CAKDfjgdYDXpN6oLR3dy9jG2syCcdyXxNLhKmJ5--mQm73cj-rA@mail.gmail.com>

While I haven't done this before you probably need to ignore the 183
from the remote end and set execute_on_preanswer or possibly
execute_on_media:

<action application="export"
data="nolocal:execute_on_preanswer=playback somefile"/>
<action application="export" data="nolocal:ignore_early_media=true"/>
<action application="bridge" data="bridge/string"/>

  However if you're getting a codec error now you probably have some
other issues to deal with.  It looks like you're trying to use
mod_native_file.  Can you try a normal wav (using mod_sndfile) and try
it again with the above code snippet?  Send the FS console logs to
http://pastebin.freeswitch.org.

On Wed, Sep 21, 2011 at 5:24 AM, Chris Martineau <chris at ghosttelecom.com> wrote:
> Hi
>
>
>
> Trying to play a custom announcement/welcome to the a leg on receiving a 183
> from the far end.
>
> Have tried a straight playback just before the bridge which confirms that it
> finds the playback file as the recording is heard but as this is a bit nasty
> FS complains with a codec error and closes the call.
>
> Tried execute on media with the same playback details as follows...
>
> <action application=?set? data=?execute_on_media=playback /path/mediafile?
>
> Which according to the logs seems to work but I get just silence until I
> presume the file stops playing which is strange as the file plays fine when
> using a forced playback?
>
> Logs in both cases show the same file being found and used mediafile.PCMA.
>
>
>
> Also I would like to terminate the playback on answer but cannot find any
> obvious process to do this. Obviously you could do an execute_on_answer but
> can?t see what I would execute to achieve the desired effect?
>
>
>
> Many thanks for any help you can offer.
>
>
>
> Regards
>
>
>
>
>
> Chris
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From chris at ghosttelecom.com  Wed Sep 21 18:30:13 2011
From: chris at ghosttelecom.com (Chris Martineau)
Date: Wed, 21 Sep 2011 15:30:13 +0100
Subject: [Freeswitch-users] Early media playback
In-Reply-To: <CAKDfjgdYDXpN6oLR3dy9jG2syCcdyXxNLhKmJ5--mQm73cj-rA@mail.gmail.com>
References: <1D10AB188D6CCA46BB4369E3268E36EF359A8F@SVR01.ghosttelecom.local>
	<CAKDfjgdYDXpN6oLR3dy9jG2syCcdyXxNLhKmJ5--mQm73cj-rA@mail.gmail.com>
Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF359AE2@SVR01.ghosttelecom.local>

Hi,

Managed to fix this by trying every possible combination starting from a
very basic scenario.

Works as follows...

Execute on media <playback etc> inline
Pre-answer inline (must be inline)
<whatever else you want to do>
Bridge

My big mistake was having proxy media enabled. Must be full fs
processing for early media playback to work.
You can use proxy media but only works when set as part of the bridge
action.

I don't ignore far end 183 as I want this to kick in after my playback
and works as I want.

Only a couple of things I am still stuck on.

1. I would like to terminate the playback on answer but cannot find any
obvious process to do this. Obviously you could do an execute_on_answer
but can't see what I would execute to achieve the desired effect?
Currently if the far end answers quickly they get silence until the
playback has finished and then they get through.

2. How do you execute multiple applications on answer? I need to execute
a sched_hangup and a sched_broadcast on answer so that the timers start
from answer. However cannot find any combination which allows me to put
multiple apps in the same execute on answer action. If I just create
multiple action entries it only seems to action the last one entered?
So have...
application='set' data='execute_on_answer=sched_hangup +XX
allotted_timeout'
if I add...
application='set' data='execute_on_answer=sched_broadcast +XX
/path/file'
only the broadcast triggers?

Many thanks again

Chris

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Kristian Kielhofner
Sent: 21 September 2011 15:04
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Early media playback

While I haven't done this before you probably need to ignore the 183
from the remote end and set execute_on_preanswer or possibly
execute_on_media:

<action application="export"
data="nolocal:execute_on_preanswer=playback somefile"/>
<action application="export" data="nolocal:ignore_early_media=true"/>
<action application="bridge" data="bridge/string"/>

  However if you're getting a codec error now you probably have some
other issues to deal with.  It looks like you're trying to use
mod_native_file.  Can you try a normal wav (using mod_sndfile) and try
it again with the above code snippet?  Send the FS console logs to
http://pastebin.freeswitch.org.

On Wed, Sep 21, 2011 at 5:24 AM, Chris Martineau
<chris at ghosttelecom.com> wrote:
> Hi
>
>
>
> Trying to play a custom announcement/welcome to the a leg on receiving
a 183
> from the far end.
>
> Have tried a straight playback just before the bridge which confirms
that it
> finds the playback file as the recording is heard but as this is a bit
nasty
> FS complains with a codec error and closes the call.
>
> Tried execute on media with the same playback details as follows...
>
> <action application="set" data="execute_on_media=playback
/path/mediafile"
>
> Which according to the logs seems to work but I get just silence until
I
> presume the file stops playing which is strange as the file plays fine
when
> using a forced playback?
>
> Logs in both cases show the same file being found and used
mediafile.PCMA.
>
>
>
> Also I would like to terminate the playback on answer but cannot find
any
> obvious process to do this. Obviously you could do an
execute_on_answer but
> can't see what I would execute to achieve the desired effect?
>
>
>
> Many thanks for any help you can offer.
>
>
>
> Regards
>
>
>
>
>
> Chris
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


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


From kris at kriskinc.com  Wed Sep 21 19:00:44 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 21 Sep 2011 11:00:44 -0400
Subject: [Freeswitch-users] Early media playback
In-Reply-To: <1D10AB188D6CCA46BB4369E3268E36EF359AE2@SVR01.ghosttelecom.local>
References: <1D10AB188D6CCA46BB4369E3268E36EF359A8F@SVR01.ghosttelecom.local>
	<CAKDfjgdYDXpN6oLR3dy9jG2syCcdyXxNLhKmJ5--mQm73cj-rA@mail.gmail.com>
	<1D10AB188D6CCA46BB4369E3268E36EF359AE2@SVR01.ghosttelecom.local>
Message-ID: <CAKDfjgdBmwtHKvziW=ruFT0r_hT9p-cAYFgKLnEe7u48i68mow@mail.gmail.com>

Please try to be as specific as possible when describing issues.  As
far as proxy_media, well, this comes up all of the time:

http://wiki.freeswitch.org/wiki/Proxy_Media#Common_misconceptions_.28READ_THIS.29

I have no idea why you'd need to use inline but whatever works, I guess...

On Wed, Sep 21, 2011 at 10:30 AM, Chris Martineau
<chris at ghosttelecom.com> wrote:
> Hi,
>
> Managed to fix this by trying every possible combination starting from a
> very basic scenario.
>
> Works as follows...
>
> Execute on media <playback etc> inline
> Pre-answer inline (must be inline)
> <whatever else you want to do>
> Bridge
>
> My big mistake was having proxy media enabled. Must be full fs
> processing for early media playback to work.
> You can use proxy media but only works when set as part of the bridge
> action.
>
> I don't ignore far end 183 as I want this to kick in after my playback
> and works as I want.
>
> Only a couple of things I am still stuck on.
>
> 1. I would like to terminate the playback on answer but cannot find any
> obvious process to do this. Obviously you could do an execute_on_answer
> but can't see what I would execute to achieve the desired effect?
> Currently if the far end answers quickly they get silence until the
> playback has finished and then they get through.
>
> 2. How do you execute multiple applications on answer? I need to execute
> a sched_hangup and a sched_broadcast on answer so that the timers start
> from answer. However cannot find any combination which allows me to put
> multiple apps in the same execute on answer action. If I just create
> multiple action entries it only seems to action the last one entered?
> So have...
> application='set' data='execute_on_answer=sched_hangup +XX
> allotted_timeout'
> if I add...
> application='set' data='execute_on_answer=sched_broadcast +XX
> /path/file'
> only the broadcast triggers?
>
> Many thanks again
>
> Chris
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Kristian Kielhofner
> Sent: 21 September 2011 15:04
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Early media playback
>
> While I haven't done this before you probably need to ignore the 183
> from the remote end and set execute_on_preanswer or possibly
> execute_on_media:
>
> <action application="export"
> data="nolocal:execute_on_preanswer=playback somefile"/>
> <action application="export" data="nolocal:ignore_early_media=true"/>
> <action application="bridge" data="bridge/string"/>
>
> ?However if you're getting a codec error now you probably have some
> other issues to deal with. ?It looks like you're trying to use
> mod_native_file. ?Can you try a normal wav (using mod_sndfile) and try
> it again with the above code snippet? ?Send the FS console logs to
> http://pastebin.freeswitch.org.
>
> On Wed, Sep 21, 2011 at 5:24 AM, Chris Martineau
> <chris at ghosttelecom.com> wrote:
>> Hi
>>
>>
>>
>> Trying to play a custom announcement/welcome to the a leg on receiving
> a 183
>> from the far end.
>>
>> Have tried a straight playback just before the bridge which confirms
> that it
>> finds the playback file as the recording is heard but as this is a bit
> nasty
>> FS complains with a codec error and closes the call.
>>
>> Tried execute on media with the same playback details as follows...
>>
>> <action application="set" data="execute_on_media=playback
> /path/mediafile"
>>
>> Which according to the logs seems to work but I get just silence until
> I
>> presume the file stops playing which is strange as the file plays fine
> when
>> using a forced playback?
>>
>> Logs in both cases show the same file being found and used
> mediafile.PCMA.
>>
>>
>>
>> Also I would like to terminate the playback on answer but cannot find
> any
>> obvious process to do this. Obviously you could do an
> execute_on_answer but
>> can't see what I would execute to achieve the desired effect?
>>
>>
>>
>> Many thanks for any help you can offer.
>>
>>
>>
>> Regards
>>
>>
>>
>>
>>
>> Chris
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Kristian Kielhofner
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


From adam.kelloway at newpace.ca  Wed Sep 21 19:08:58 2011
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Wed, 21 Sep 2011 12:08:58 -0300
Subject: [Freeswitch-users] Inaccurate variable_record_ms
Message-ID: <4E79FE0A.1000802@newpace.ca>

Hi there,

I've noticed that the variable_record_ms and variable_record_seconds 
values are often inaccurate. Is there any explanation for this?

The example below shows that while the RECORD_START and RECORD_STOP 
events were about 4 seconds apart, the variables claim that the file is 
7 seconds long. See the soxi output at the end, which shows that the 
length of the file is 00:00:03.50, thus more in line with what the event 
timestamps suggest.

RECV EVENT
Event-Name: RECORD_START
Core-UUID: 235bab4c-36c6-48af-87d7-0f30a1140d52
*Event-Date-GMT: Wed, 21 Sep 2011 14:52:24 GMT*
Event-Date-Timestamp: 1316616744483384
Event-Calling-File: switch_ivr_play_say.c
Event-Calling-Function: switch_ivr_record_file
Event-Calling-Line-Number: 615
Unique-ID: fa39d249-c9da-4072-aebe-4b606e661e5f
Caller-Profile-Created-Time: 1316616744483384
Caller-Channel-Created-Time: 1316616740683382
Caller-Channel-Answered-Time: 1316616740683382
*variable_current_application_data: 
/tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav 120 200 2*
variable_current_application: record
Record-File-Path: /tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav

RECV EVENT
Event-Name: RECORD_STOP
Core-UUID: 235bab4c-36c6-48af-87d7-0f30a1140d52
*Event-Date-GMT: Wed, 21 Sep 2011 14:52:28 GMT*
Event-Date-Timestamp: 1316616748083413
Event-Calling-File: switch_ivr_play_say.c
Event-Calling-Function: switch_ivr_record_file
Event-Calling-Line-Number: 790
Unique-ID: fa39d249-c9da-4072-aebe-4b606e661e5f
Caller-Profile-Created-Time: 1316616744483384
Caller-Channel-Created-Time: 1316616740683382
Caller-Channel-Answered-Time: 1316616740683382
*variable_current_application_data: 
/tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav 120 200 2*
variable_current_application: record
*variable_record_seconds: 7*
*variable_record_ms: 7000*
variable_record_samples: 56000
Record-File-Path: /tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav

root at dev:/usr/local/freeswitch# soxi 
/tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav

Input File     : '/tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav'
Channels       : 1
Sample Rate    : 8000
Precision      : 16-bit
Duration       : *00:00:03.50* = 28000 samples ~ 262.5 CDDA sectors
File Size      : 56.0k
Bit Rate       : 128k
Sample Encoding: 16-bit Signed Integer PCM
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From Nabble_01394 at slickdeals.endjunk.com  Wed Sep 21 19:09:03 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Wed, 21 Sep 2011 08:09:03 -0700 (PDT)
Subject: [Freeswitch-users] Missing mod_blacklist.conf file?
In-Reply-To: <1316565252256-6814430.post@n2.nabble.com>
References: <1316565252256-6814430.post@n2.nabble.com>
Message-ID: <1316617743617-6816434.post@n2.nabble.com>

I just chatted with Stefan Knoblich on IRC and he committed the
blacklist.conf.xml to FS git repo.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Missing-mod-blacklist-conf-file-tp6814430p6816434.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Wed Sep 21 20:33:28 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Sep 2011 11:33:28 -0500
Subject: [Freeswitch-users] Inaccurate variable_record_ms
In-Reply-To: <4E79FE0A.1000802@newpace.ca>
References: <4E79FE0A.1000802@newpace.ca>
Message-ID: <CAKbxfG8VqY+B=-38ZW6JJ1NA0Vorm3G8uXTCmZv0LEtFd9hoUA@mail.gmail.com>

It was a bug. next time file it in JIRA, its only by chance that I
caught this email floating in a sea of 1000 other ones.

commit 5fe3a22d83867d07f30ef7974329b2fa966747ea
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Wed Sep 21 11:05:33 2011 -0500

    fix inaccurate sample count in file handle, buffered samples were
being double tallied



On Wed, Sep 21, 2011 at 10:08 AM, Adam Kelloway
<adam.kelloway at newpace.ca> wrote:
> Hi there,
>
> I've noticed that the variable_record_ms and variable_record_seconds values
> are often inaccurate. Is there any explanation for this?
>
> The example below shows that while the RECORD_START and RECORD_STOP events
> were about 4 seconds apart, the variables claim that the file is 7 seconds
> long. See the soxi output at the end, which shows that the length of the
> file is 00:00:03.50, thus more in line with what the event timestamps
> suggest.
>
> RECV EVENT
> Event-Name: RECORD_START
> Core-UUID: 235bab4c-36c6-48af-87d7-0f30a1140d52
> Event-Date-GMT: Wed, 21 Sep 2011 14:52:24 GMT
> Event-Date-Timestamp: 1316616744483384
> Event-Calling-File: switch_ivr_play_say.c
> Event-Calling-Function: switch_ivr_record_file
> Event-Calling-Line-Number: 615
> Unique-ID: fa39d249-c9da-4072-aebe-4b606e661e5f
> Caller-Profile-Created-Time: 1316616744483384
> Caller-Channel-Created-Time: 1316616740683382
> Caller-Channel-Answered-Time: 1316616740683382
> variable_current_application_data:
> /tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav 120 200 2
> variable_current_application: record
> Record-File-Path: /tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav
>
> RECV EVENT
> Event-Name: RECORD_STOP
> Core-UUID: 235bab4c-36c6-48af-87d7-0f30a1140d52
> Event-Date-GMT: Wed, 21 Sep 2011 14:52:28 GMT
> Event-Date-Timestamp: 1316616748083413
> Event-Calling-File: switch_ivr_play_say.c
> Event-Calling-Function: switch_ivr_record_file
> Event-Calling-Line-Number: 790
> Unique-ID: fa39d249-c9da-4072-aebe-4b606e661e5f
> Caller-Profile-Created-Time: 1316616744483384
> Caller-Channel-Created-Time: 1316616740683382
> Caller-Channel-Answered-Time: 1316616740683382
> variable_current_application_data:
> /tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav 120 200 2
> variable_current_application: record
> variable_record_seconds: 7
> variable_record_ms: 7000
> variable_record_samples: 56000
> Record-File-Path: /tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav
>
> root at dev:/usr/local/freeswitch# soxi
> /tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav
>
> Input File???? : '/tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav'
> Channels?????? : 1
> Sample Rate??? : 8000
> Precision????? : 16-bit
> Duration?????? : 00:00:03.50 = 28000 samples ~ 262.5 CDDA sectors
> File Size????? : 56.0k
> Bit Rate?????? : 128k
> Sample Encoding: 16-bit Signed Integer PCM
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 krice at freeswitch.org  Wed Sep 21 20:35:54 2011
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 21 Sep 2011 11:35:54 -0500
Subject: [Freeswitch-users] Weekly FreeSwitch conference Call
Message-ID: <CA9F7C9A.1A39B%krice@freeswitch.org>

Hey Guys,

Michael and Brian have asked me to help out with the conference today since
Michael is otherwise occupied today...

Open Forum for the most part today, however, we will have some discussion of
hotels for Cluecon 2012, and some more embedded talk. Plus more....

Join us at 1PM Eastern thats 10AM Pacific or 5PM GMT Today!

Just dial *9888 from your FreeSwitch installs that have the default dialplan
or join as via the web at http://conference.freeswitch.org/

K
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From adam.kelloway at newpace.ca  Wed Sep 21 20:40:44 2011
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Wed, 21 Sep 2011 13:40:44 -0300
Subject: [Freeswitch-users] Inaccurate variable_record_ms
In-Reply-To: <CAKbxfG8VqY+B=-38ZW6JJ1NA0Vorm3G8uXTCmZv0LEtFd9hoUA@mail.gmail.com>
References: <4E79FE0A.1000802@newpace.ca>
	<CAKbxfG8VqY+B=-38ZW6JJ1NA0Vorm3G8uXTCmZv0LEtFd9hoUA@mail.gmail.com>
Message-ID: <4E7A138C.8040707@newpace.ca>

Will do, thanks very much.

Adam

On 3:59 PM, Anthony Minessale wrote:
> It was a bug. next time file it in JIRA, its only by chance that I
> caught this email floating in a sea of 1000 other ones.
>
> commit 5fe3a22d83867d07f30ef7974329b2fa966747ea
> Author: Anthony Minessale<anthm at freeswitch.org>
> Date:   Wed Sep 21 11:05:33 2011 -0500
>
>      fix inaccurate sample count in file handle, buffered samples were
> being double tallied
>
>
>
> On Wed, Sep 21, 2011 at 10:08 AM, Adam Kelloway
> <adam.kelloway at newpace.ca>  wrote:
>> Hi there,
>>
>> I've noticed that the variable_record_ms and variable_record_seconds values
>> are often inaccurate. Is there any explanation for this?
>>
>> The example below shows that while the RECORD_START and RECORD_STOP events
>> were about 4 seconds apart, the variables claim that the file is 7 seconds
>> long. See the soxi output at the end, which shows that the length of the
>> file is 00:00:03.50, thus more in line with what the event timestamps
>> suggest.
>>
>> RECV EVENT
>> Event-Name: RECORD_START
>> Core-UUID: 235bab4c-36c6-48af-87d7-0f30a1140d52
>> Event-Date-GMT: Wed, 21 Sep 2011 14:52:24 GMT
>> Event-Date-Timestamp: 1316616744483384
>> Event-Calling-File: switch_ivr_play_say.c
>> Event-Calling-Function: switch_ivr_record_file
>> Event-Calling-Line-Number: 615
>> Unique-ID: fa39d249-c9da-4072-aebe-4b606e661e5f
>> Caller-Profile-Created-Time: 1316616744483384
>> Caller-Channel-Created-Time: 1316616740683382
>> Caller-Channel-Answered-Time: 1316616740683382
>> variable_current_application_data:
>> /tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav 120 200 2
>> variable_current_application: record
>> Record-File-Path: /tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav
>>
>> RECV EVENT
>> Event-Name: RECORD_STOP
>> Core-UUID: 235bab4c-36c6-48af-87d7-0f30a1140d52
>> Event-Date-GMT: Wed, 21 Sep 2011 14:52:28 GMT
>> Event-Date-Timestamp: 1316616748083413
>> Event-Calling-File: switch_ivr_play_say.c
>> Event-Calling-Function: switch_ivr_record_file
>> Event-Calling-Line-Number: 790
>> Unique-ID: fa39d249-c9da-4072-aebe-4b606e661e5f
>> Caller-Profile-Created-Time: 1316616744483384
>> Caller-Channel-Created-Time: 1316616740683382
>> Caller-Channel-Answered-Time: 1316616740683382
>> variable_current_application_data:
>> /tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav 120 200 2
>> variable_current_application: record
>> variable_record_seconds: 7
>> variable_record_ms: 7000
>> variable_record_samples: 56000
>> Record-File-Path: /tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav
>>
>> root at dev:/usr/local/freeswitch# soxi
>> /tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav
>>
>> Input File     : '/tmp/fa39d249-c9da-4072-aebe-4b606e661e5f.wav'
>> Channels       : 1
>> Sample Rate    : 8000
>> Precision      : 16-bit
>> Duration       : 00:00:03.50 =8000 samples ~ 262.5 CDDA sectors
>> File Size      : 56.0k
>> Bit Rate       : 128k
>> Sample Encoding: 16-bit Signed Integer PCM
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>

-- 
Adam
-- 
NewPace Logo 	
	
	
	Adam Kelloway

	Software Engineer, NewPace
phone 	+1 (902) 406--8375  x1031
email 	Adam.Kelloway at NewPace.com <mailto:Adam.Kelloway at newpace.com>
aim <aim:GoIm?screenname=Adam.Kelloway at newpace.com>/msn 
<msnim:chat?contact=Adam.Kelloway at newpace.com> 	Adam.Kelloway 
<aim:GoIm?screenname=Adam.Kelloway at newpace.ca>@NewPace.ca

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From Stefan.Weigel at allianz-warranty.com  Wed Sep 21 22:54:13 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Wed, 21 Sep 2011 20:54:13 +0200
Subject: [Freeswitch-users] change sip_to_user in dialplan
In-Reply-To: <CALNZuwnGjmSHTGrEfGzTAzUFu6EJ3qQPvYbWHedA_b3fm2Zd+w@mail.gmail.com>
References: <5003D7D3E06F514E8C682F18D223265C04D3B36DF1@AZWSMS03.azwarranty.int>
	<CALNZuwnGjmSHTGrEfGzTAzUFu6EJ3qQPvYbWHedA_b3fm2Zd+w@mail.gmail.com>
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36DF6@AZWSMS03.azwarranty.int>

Hi Avi,



our provider doesn't send the whole number in the INVITE, we need to parse the 'sip_to_uri'.

I just found a way to transfer through the dialplan and putting my information together.



Thanks and best regards



Stefan







Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Avi Marcus
Gesendet: Mittwoch, 21. September 2011 15:20
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] change sip_to_user in dialplan



Why bother? regex match on the destination_number and then bridge/transfer accordingly. What are you trying to gain by messing with the actual sip_to_user?

And perhaps set on the inbound gateway where it gets sent to..




-Avi



On Wed, Sep 21, 2011 at 4:12 PM, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>> wrote:

Hello all,



I'm trying to change some incoming calls 'sip_to_uri' (it's a rerouting of our ITSP) so it matches our main number:



0891234567 -> 08998765



I'm checking the sip_to_uri and then trying to export/set/set_profile_var to the new number. But in debug trace it's still comparing against the original sip_to_uri.



Is there a way to change as soon it's hitting the dialplan ?





Thanks and best regards



Stefan



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



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From zhulizhong at live.com  Wed Sep 21 11:27:42 2011
From: zhulizhong at live.com (James zhu)
Date: Wed, 21 Sep 2011 07:27:42 +0000
Subject: [Freeswitch-users] please help for Failed Registration,
	setting retry to 1020 seconds
Message-ID: <BLU121-W2467447ED24E90F6E665D7B50D0@phx.gbl>


hi:
I am using GSM gateway to regiester to freeswitch, the incoming from gsm gateway is ok, but i can not regiester
into freeswitch due to the gateway connection.
===the debug info from fs_CLI
   ------------------------------------------------------------------------
2011-09-21 23:14:50.645350 [ERR] sofia_reg.c:1496 gsm Registration Failed with status Request Timeout [408]. failure #33
2011-09-21 23:14:53.318802 [WARNING] sofia_reg.c:386 gsm Failed Registration, setting retry to 1020 seconds.
=====sip trace ===============

   ------------------------------------------------------------------------
send 590 bytes to udp/[172.16.33.11]:5060 at 15:14:50.147741:
   ------------------------------------------------------------------------
   REGISTER sip:172.16.33.11 SIP/2.0
   Via: SIP/2.0/UDP 172.16.33.100:5080;rport;branch=z9hG4bKB09cFa6Uvyc1H
   Max-Forwards: 70
   From: <sip:1000 at 172.16.33.11;transport=udp>;tag=B1ytyyXQS3NKr
   To: <sip:1000 at 172.16.33.11;transport=udp>
   Call-ID: 4809dacc-aeee-4a33-a245-884c6788a4a6
   CSeq: 17972965 REGISTER
   Contact: <sip:gw+gsm at 172.16.33.100:5080;transport=udp;gw=gsm>
   Expires: 30
   User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Length: 0
   ------------------------------------------------------------------------
send 590 bytes to udp/[172.16.33.11]:5060 at 15:14:46.146854:
   ------------------------------------------------------------------------
   REGISTER sip:172.16.33.11 SIP/2.0
   Via: SIP/2.0/UDP 172.16.33.100:5080;rport;branch=z9hG4bKB09cFa6Uvyc1H
   Max-Forwards: 70
   From: <sip:1000 at 172.16.33.11;transport=udp>;tag=B1ytyyXQS3NKr
   To: <sip:1000 at 172.16.33.11;transport=udp>
   Call-ID: 4809dacc-aeee-4a33-a245-884c6788a4a6
   CSeq: 17972965 REGISTER
   Contact: <sip:gw+gsm at 172.16.33.100:5080;transport=udp;gw=gsm>
   Expires: 30
   User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Length: 0

=======gateway confi: ======
<include>
  <gateway name="gsm">
  <!--/// account username *required* ///-->
  <!-- Extension 1000 in my case -->
  <param name="username" value="1000"/>
  <!--/// auth realm: *optional* same as gateway name, if blank ///-->
  <!-- It must aim to the  GateWay IP 192.168.2.2 in my case -->
  <param name="realm" value="172.16.33.11"/>
  <!--1234 in my case-->
  <param name="password" value="1234"/>
  <!--/// expire in seconds: *optional* 3600, if blank ///-->
  <param name="expire-seconds" value=""/>
  <!--/// do not register ///-->
  <param name="register" value="false"/>
  <!-- which transport to use for register -->
  <param name="sip-trace" value="yes"/>

 </gateway>
 </include>
freeswitch at internal> sofia status
                     Name          Type                                       Data      State
=================================================================================================
                 external       profile           sip:mod_sofia at 172.16.33.100:5080      RUNNING (0)
    external::example.com       gateway                    sip:joeuser at example.com      NOREG
            external::gsm       gateway                      sip:1000 at 172.16.33.11      FAIL_WAIT
            internal-ipv6       profile                   sip:mod_sofia@[::1]:5060      RUNNING (0)
                 internal       profile           sip:mod_sofia at 172.16.33.100:5060      RUNNING (0)
            172.16.33.100         alias                                   internal      ALIASED
=================================================================================================
somehow, the gateway never sends anything.  anyone can give a hit?

Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri<->SIP).
website: www.voipviews.com 
 		 	   		  
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From kashif at kashifbukhari.com  Wed Sep 21 16:02:22 2011
From: kashif at kashifbukhari.com (Kashif Ali)
Date: Wed, 21 Sep 2011 17:02:22 +0500
Subject: [Freeswitch-users] vBilling Beta Program!!
In-Reply-To: <CAHLYDCu5cGU3KFguy28CzuM3RxQ+RYA7nYZnzLmZF4UTOJNH5A@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>
	<CAHLYDCu5cGU3KFguy28CzuM3RxQ+RYA7nYZnzLmZF4UTOJNH5A@mail.gmail.com>
Message-ID: <CANwDMSWcfk9ex39TwQU4mZNGj2RedWTtN4dhZXcYQD0ooCqXjQ@mail.gmail.com>

Dear Naseer
when installation will be online again?


2011/9/8 Juan Antonio Iba?ez Santorum <juanito1982 at 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
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ga at steadfasttelecom.com  Wed Sep 21 17:41:39 2011
From: ga at steadfasttelecom.com (Gilad Abada)
Date: Wed, 21 Sep 2011 09:41:39 -0400
Subject: [Freeswitch-users] FreeTDM kickstart help ERRORS inline
In-Reply-To: <CAB2KLyeJVd3mP+7LUSMTRq5sMgcoz5FoKFwTZukyk_dnyFMFQw@mail.gmail.com>
References: <CAB2KLyeJVd3mP+7LUSMTRq5sMgcoz5FoKFwTZukyk_dnyFMFQw@mail.gmail.com>
Message-ID: <1069397243424704580@unknownmsgid>

Return the astribanks!! Stay away from xorcom. Once you do get them up
you will have MWI issues. Best thing to do is to go with grandstream
gateways. They are 1/3 the price and easy to set up. Also if one of
your astribanks fails (I've had 3 fail) all your astribanks are down
until you get them to over night you a new one (at your cost) then you
need to configure it again!!

Sent from my mobile device.

On Sep 20, 2011, at 10:59 PM, Luis F Urrea <lfurrea at gmail.com> wrote:

> This dAHDI stuff is like voodoo!
>
> I know this has to be configuration issues but sh****t after a few years in the telecom industry it is really hard to find a more awful syntax for configuring such a stupid device as an FXO port.
>
> I am trying to get a Xorcom Astribank going with FreeTDM. So far after a bunch of firmware and driver BS (come on there is probably nothing more buggy in a freking kernel than the USB layer, how can a telecom device be using a USB driver!!!!!!!).
>
> I get the following output from
>
> #lsdahdi
> ### Span  1: XBUS-00/XPD-00 "Xorcom XPD #00/00: FXO" (MASTER)
>   1 FXO                     RED
>   2 FXO                     RED
>   3 FXO                     RED
>   4 FXO                     RED
>   5 FXO                     RED
>   6 FXO                     RED
>   7 FXO                     RED
>   8 FXO                     RED
> ### Span  2: XBUS-00/XPD-10 "Xorcom XPD #00/10: FXO"
>   9 FXO                     RED
>  10 FXO                     RED
>  11 FXO                     RED
>  12 FXO                     RED
>  13 FXO                     RED
>  14 FXO                     RED
>  15 FXO                     RED
>  16 FXO                    (battery)
> ### Span  3: XBUS-00/XPD-20 "Xorcom XPD #00/20: FXO"
>  17 FXO                     RED
>  18 FXO                    (battery)
>  19 FXO                    (battery)
>  20 FXO                    (battery)
>  21 FXO                    (battery)
>  22 FXO                    (battery)
>  23 FXO                    (battery)
>  24 FXO                    (battery)
>
> But when loading mod_freetdm I am getting:
>
> 2011-09-19 15:44:03.149105 [NOTICE] ftdm_io.c:5723 Modules configured: 1
> 2011-09-19 15:44:03.149105 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch/conf/freetdm.conf.
> 2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4633 Reading FreeTDM configuration file
> 2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4649 found config for span
> 2011-09-19 15:44:03.149105 [NOTICE] ftmod_zt.c:1323 Using DAHDI control device
> 2011-09-19 15:44:03.149105 [INFO] ftdm_io.c:4963 Loading IO from /usr/local/freeswitch/mod/ftmod_zt.so [zt]
> 2011-09-19 15:44:03.149105 [DEBUG] ftdm_config.c:80 Configuration file is /usr/local/freeswitch/conf/zt.conf.
> 2011-09-19 15:44:03.149105 [INFO] ftmod_zt.c:585 Setting rxgain val to 0.000000
> 2011-09-19 15:44:03.149105 [INFO] ftmod_zt.c:593 Setting txgain val to 0.000000
> 2011-09-19 15:44:03.149105 [INFO] ftdm_io.c:787 Auto-loaded I/O module 'zt'
> 2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4670 created span 1 (SPAN1) of type zt
> 2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4688 span 1 [fxo-channel]=[17-24]
> 2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4718 setting trunk type to 'FXO' start(KEWL)
> 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device /dev/dahdi/channel chan 17 fd 48 (No such device or address)
> 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device /dev/dahdi/channel chan 18 fd 48 (No such device or address)
> 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device /dev/dahdi/channel chan 19 fd 48 (No such device or address)
> 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device /dev/dahdi/channel chan 20 fd 48 (No such device or address)
> 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device /dev/dahdi/channel chan 21 fd 48 (No such device or address)
> 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device /dev/dahdi/channel chan 22 fd 48 (No such device or address)
> 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device /dev/dahdi/channel chan 23 fd 48 (No such device or address)
> 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring device /dev/dahdi/channel chan 24 fd 48 (No such device or address)
> 2011-09-19 15:44:03.149105 [ERR] ftdm_io.c:4579 1:Failed to configure span
> 2011-09-19 15:44:03.149105 [INFO] ftdm_io.c:4885 Configured 0 channel(s)
> 2011-09-19 15:44:03.149105 [ERR] ftdm_io.c:5733 FreeTDM global configuration failed!
> 2011-09-19 15:44:03.149105 [ERR] mod_freetdm.c:4805 Error configuring FreeTDM
> 2011-09-19 15:44:03.149105 [INFO] switch_time.c:1028 Timezone reloaded 530 definitions
> 2011-09-19 15:44:03.149105 [CRIT] switch_loadable_module.c:1019 Error Loading module /usr/local/freeswitch/mod/mod_freetdm.so
> **Module load routine returned an error**
> 2011-09-19 15:44:03.149105 [NOTICE] ftdm_sched.c:147 Main scheduling thread going out ...
>
>
> Thanks in advance and sorry for the rant.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 f.erfurth at googlemail.com  Wed Sep 21 20:09:46 2011
From: f.erfurth at googlemail.com (f.erfurth)
Date: Wed, 21 Sep 2011 18:09:46 +0200
Subject: [Freeswitch-users] IAX
Message-ID: <CANFMcCj9jqiWyM32caaR5qLAFizDhOrLbbpWS=_fFX30RJKUMQ@mail.gmail.com>

Hi,
I installed FreeSWITCH for BigBlueButton (Conference-System). I want call by
using IAX. Is IAX available by default or do I have some configurations for
that?

I have a dialplan which is responsible for connecting to Conference of
BigBlueButton. I should use the context bbb-voip.
cu Floh
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From fssupport at highergear.com  Wed Sep 21 21:20:31 2011
From: fssupport at highergear.com (fssupport at highergear.com)
Date: Wed, 21 Sep 2011 12:20:31 -0500 (CDT)
Subject: [Freeswitch-users] Share PRI line?
Message-ID: <56122.67.95.76.98.1316625631.squirrel@webmail.highergear.com>

We have a client with a Mitel PBX with a PRI connected to it.  We are
deploying a freeswitch box that will do some autodialing tasks for them. 
Is there a way to share the PRI between the two systems?  If nothing else,
I'm pretty sure I can put a two line interface card in the Freeswitch
server, plug the pri into it, then connected the other interface to the
Mitel box, but that would make our box a point of failure and I'd like to
avoid that if possible.



From nbhatti at gmail.com  Wed Sep 21 23:48:13 2011
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Wed, 21 Sep 2011 22:48:13 +0300
Subject: [Freeswitch-users] vBilling Beta Program!!
In-Reply-To: <CANwDMSWcfk9ex39TwQU4mZNGj2RedWTtN4dhZXcYQD0ooCqXjQ@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>
	<CAHLYDCu5cGU3KFguy28CzuM3RxQ+RYA7nYZnzLmZF4UTOJNH5A@mail.gmail.com>
	<CANwDMSWcfk9ex39TwQU4mZNGj2RedWTtN4dhZXcYQD0ooCqXjQ@mail.gmail.com>
Message-ID: <CACak_yyTVb4OuO_vendQnFVUV6GK5iyvQ53kKDH+AE-N4U2VUA@mail.gmail.com>

Dear Kashif,
It is not online at the moment, going to take a few more days. But we can
install it for you without any charges :) Contact me off-list if you are
interested.

-B

On Wed, Sep 21, 2011 at 3:02 PM, Kashif Ali <kashif at kashifbukhari.com>wrote:

> Dear Naseer
> when installation will be online again?
>
>
> 2011/9/8 Juan Antonio Iba?ez Santorum <juanito1982 at 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
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lfurrea at gmail.com  Wed Sep 21 23:53:32 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Wed, 21 Sep 2011 13:53:32 -0600
Subject: [Freeswitch-users] FreeTDM kickstart help ERRORS inline
In-Reply-To: <1069397243424704580@unknownmsgid>
References: <CAB2KLyeJVd3mP+7LUSMTRq5sMgcoz5FoKFwTZukyk_dnyFMFQw@mail.gmail.com>
	<1069397243424704580@unknownmsgid>
Message-ID: <CAB2KLydQxsJnQ+np2-KtJ24hrAb0LzRKmkvWUQ4OX1F3oTs8wg@mail.gmail.com>

Hahahaha

I wish o could convince my customer to do so.

Grandstream, Audiocodes, Patton, you name it, but good old SIP communication
over the network!

But it's a requirement for the project.

Any input is welcome I don't want to be spoon fed but it looks like all this
dahdi configuration is most of the times generated by scripts and not even
documented.

On Wed, Sep 21, 2011 at 7:41 AM, Gilad Abada <ga at steadfasttelecom.com>wrote:

> Return the astribanks!! Stay away from xorcom. Once you do get them up
> you will have MWI issues. Best thing to do is to go with grandstream
> gateways. They are 1/3 the price and easy to set up. Also if one of
> your astribanks fails (I've had 3 fail) all your astribanks are down
> until you get them to over night you a new one (at your cost) then you
> need to configure it again!!
>
> Sent from my mobile device.
>
> On Sep 20, 2011, at 10:59 PM, Luis F Urrea <lfurrea at gmail.com> wrote:
>
> > This dAHDI stuff is like voodoo!
> >
> > I know this has to be configuration issues but sh****t after a few years
> in the telecom industry it is really hard to find a more awful syntax for
> configuring such a stupid device as an FXO port.
> >
> > I am trying to get a Xorcom Astribank going with FreeTDM. So far after a
> bunch of firmware and driver BS (come on there is probably nothing more
> buggy in a freking kernel than the USB layer, how can a telecom device be
> using a USB driver!!!!!!!).
> >
> > I get the following output from
> >
> > #lsdahdi
> > ### Span  1: XBUS-00/XPD-00 "Xorcom XPD #00/00: FXO" (MASTER)
> >   1 FXO                     RED
> >   2 FXO                     RED
> >   3 FXO                     RED
> >   4 FXO                     RED
> >   5 FXO                     RED
> >   6 FXO                     RED
> >   7 FXO                     RED
> >   8 FXO                     RED
> > ### Span  2: XBUS-00/XPD-10 "Xorcom XPD #00/10: FXO"
> >   9 FXO                     RED
> >  10 FXO                     RED
> >  11 FXO                     RED
> >  12 FXO                     RED
> >  13 FXO                     RED
> >  14 FXO                     RED
> >  15 FXO                     RED
> >  16 FXO                    (battery)
> > ### Span  3: XBUS-00/XPD-20 "Xorcom XPD #00/20: FXO"
> >  17 FXO                     RED
> >  18 FXO                    (battery)
> >  19 FXO                    (battery)
> >  20 FXO                    (battery)
> >  21 FXO                    (battery)
> >  22 FXO                    (battery)
> >  23 FXO                    (battery)
> >  24 FXO                    (battery)
> >
> > But when loading mod_freetdm I am getting:
> >
> > 2011-09-19 15:44:03.149105 [NOTICE] ftdm_io.c:5723 Modules configured: 1
> > 2011-09-19 15:44:03.149105 [DEBUG] ftdm_config.c:80 Configuration file is
> /usr/local/freeswitch/conf/freetdm.conf.
> > 2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4633 Reading FreeTDM
> configuration file
> > 2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4649 found config for span
> > 2011-09-19 15:44:03.149105 [NOTICE] ftmod_zt.c:1323 Using DAHDI control
> device
> > 2011-09-19 15:44:03.149105 [INFO] ftdm_io.c:4963 Loading IO from
> /usr/local/freeswitch/mod/ftmod_zt.so [zt]
> > 2011-09-19 15:44:03.149105 [DEBUG] ftdm_config.c:80 Configuration file is
> /usr/local/freeswitch/conf/zt.conf.
> > 2011-09-19 15:44:03.149105 [INFO] ftmod_zt.c:585 Setting rxgain val to
> 0.000000
> > 2011-09-19 15:44:03.149105 [INFO] ftmod_zt.c:593 Setting txgain val to
> 0.000000
> > 2011-09-19 15:44:03.149105 [INFO] ftdm_io.c:787 Auto-loaded I/O module
> 'zt'
> > 2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4670 created span 1 (SPAN1)
> of type zt
> > 2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4688 span 1
> [fxo-channel]=[17-24]
> > 2011-09-19 15:44:03.149105 [DEBUG] ftdm_io.c:4718 setting trunk type to
> 'FXO' start(KEWL)
> > 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring
> device /dev/dahdi/channel chan 17 fd 48 (No such device or address)
> > 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring
> device /dev/dahdi/channel chan 18 fd 48 (No such device or address)
> > 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring
> device /dev/dahdi/channel chan 19 fd 48 (No such device or address)
> > 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring
> device /dev/dahdi/channel chan 20 fd 48 (No such device or address)
> > 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring
> device /dev/dahdi/channel chan 21 fd 48 (No such device or address)
> > 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring
> device /dev/dahdi/channel chan 22 fd 48 (No such device or address)
> > 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring
> device /dev/dahdi/channel chan 23 fd 48 (No such device or address)
> > 2011-09-19 15:44:03.149105 [ERR] ftmod_zt.c:288 failure configuring
> device /dev/dahdi/channel chan 24 fd 48 (No such device or address)
> > 2011-09-19 15:44:03.149105 [ERR] ftdm_io.c:4579 1:Failed to configure
> span
> > 2011-09-19 15:44:03.149105 [INFO] ftdm_io.c:4885 Configured 0 channel(s)
> > 2011-09-19 15:44:03.149105 [ERR] ftdm_io.c:5733 FreeTDM global
> configuration failed!
> > 2011-09-19 15:44:03.149105 [ERR] mod_freetdm.c:4805 Error configuring
> FreeTDM
> > 2011-09-19 15:44:03.149105 [INFO] switch_time.c:1028 Timezone reloaded
> 530 definitions
> > 2011-09-19 15:44:03.149105 [CRIT] switch_loadable_module.c:1019 Error
> Loading module /usr/local/freeswitch/mod/mod_freetdm.so
> > **Module load routine returned an error**
> > 2011-09-19 15:44:03.149105 [NOTICE] ftdm_sched.c:147 Main scheduling
> thread going out ...
> >
> >
> > Thanks in advance and sorry for the rant.
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sdame at 207me.com  Thu Sep 22 00:28:18 2011
From: sdame at 207me.com (Stephen Dame)
Date: Wed, 21 Sep 2011 16:28:18 -0400
Subject: [Freeswitch-users] Weekly FreeSwitch conference Call
In-Reply-To: <CA9F7C9A.1A39B%krice@freeswitch.org>
References: <CA9F7C9A.1A39B%krice@freeswitch.org>
Message-ID: <018101cc789d$004ac340$00e049c0$@com>

Ken,

 

Is there a link with the recorded archives of the weekly meetings, in case
we miss one.

 

Thanks

stephen

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Wednesday, September 21, 2011 12:36 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Weekly FreeSwitch conference Call

 

Hey Guys,

Michael and Brian have asked me to help out with the conference today since
Michael is otherwise occupied today... 

Open Forum for the most part today, however, we will have some discussion of
hotels for Cluecon 2012, and some more embedded talk. Plus more....

Join us at 1PM Eastern thats 10AM Pacific or 5PM GMT Today!

Just dial *9888 from your FreeSwitch installs that have the default dialplan
or join as via the web at http://conference.freeswitch.org/

K 

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From shouldbeq931 at gmail.com  Thu Sep 22 02:56:36 2011
From: shouldbeq931 at gmail.com (shouldbe q931)
Date: Wed, 21 Sep 2011 23:56:36 +0100
Subject: [Freeswitch-users] Share PRI line?
In-Reply-To: <56122.67.95.76.98.1316625631.squirrel@webmail.highergear.com>
References: <56122.67.95.76.98.1316625631.squirrel@webmail.highergear.com>
Message-ID: <CAP1_MOC5QOY2XQw2GgktyWH8qHDPtj5w57JQJ0=zRZftazKo0g@mail.gmail.com>

On Wed, Sep 21, 2011 at 6:20 PM,  <fssupport at highergear.com> wrote:
> We have a client with a Mitel PBX with a PRI connected to it. ?We are
> deploying a freeswitch box that will do some autodialing tasks for them.
> Is there a way to share the PRI between the two systems? ?If nothing else,
> I'm pretty sure I can put a two line interface card in the Freeswitch
> server, plug the pri into it, then connected the other interface to the
> Mitel box, but that would make our box a point of failure and I'd like to
> avoid that if possible.
>

It is not possible to "split" or "share" a PRI interface with passive
components, your three options are

a/ put a second PRI interface onto "your" system, and configure it to
route inbound calls to the Mitel system and outbound calls from the
Mitel system
b/ put a second PRI interface onto the Mitel, and configure it as per
a/ but with your system instead
c/ purchase a device with 3 PRI interfaces that can route inbound
calls and outbound calls appropriately as per a/ and b/


From curriegrad2004 at gmail.com  Thu Sep 22 03:47:18 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 21 Sep 2011 16:47:18 -0700
Subject: [Freeswitch-users] IAX
In-Reply-To: <CANFMcCj9jqiWyM32caaR5qLAFizDhOrLbbpWS=_fFX30RJKUMQ@mail.gmail.com>
References: <CANFMcCj9jqiWyM32caaR5qLAFizDhOrLbbpWS=_fFX30RJKUMQ@mail.gmail.com>
Message-ID: <CAMWiMy_b4239c9CehXfe-qMEMam4tsCffWBjkokTAzExtD88DA@mail.gmail.com>

FreeSWITCH doesn't support IAX... officially that is. There's always
mod_opal, but that is considered beta quality.

http://wiki.freeswitch.org/wiki/Mod_opal

...wiki gives you the necessary instructions to get you up and running
on the IAX front ;)

On Wed, Sep 21, 2011 at 9:09 AM, f.erfurth <f.erfurth at googlemail.com> wrote:
> Hi,
> I installed FreeSWITCH for BigBlueButton (Conference-System). I want call by
> using IAX. Is IAX available by default or do I have some configurations for
> that?
>
> I have a dialplan which is responsible for connecting to Conference of
> BigBlueButton. I should use the context bbb-voip.
> cu Floh
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 robert.hadley at teotech.com  Thu Sep 22 04:09:18 2011
From: robert.hadley at teotech.com (Robert Hadley)
Date: Wed, 21 Sep 2011 17:09:18 -0700
Subject: [Freeswitch-users] Share PRI line?
In-Reply-To: <56122.67.95.76.98.1316625631.squirrel@webmail.highergear.com>
References: <56122.67.95.76.98.1316625631.squirrel@webmail.highergear.com>
Message-ID: <E826C94A3C6BE44198FAF4D66B5AB4D8245D7CAE27@VA3DIAXVS351.RED001.local>


We use an AdTran product to share a single PRI among our test servers.


-----Original Message-----
From: fssupport at highergear.com [mailto:fssupport at highergear.com] 
Sent: Wednesday, September 21, 2011 10:21 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Share PRI line?

We have a client with a Mitel PBX with a PRI connected to it.  We are deploying a freeswitch box that will do some autodialing tasks for them. 
Is there a way to share the PRI between the two systems?  If nothing else, I'm pretty sure I can put a two line interface card in the Freeswitch server, plug the pri into it, then connected the other interface to the Mitel box, but that would make our box a point of failure and I'd like to avoid that if possible.




From gcd at i.ph  Thu Sep 22 07:51:54 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Thu, 22 Sep 2011 11:51:54 +0800
Subject: [Freeswitch-users] IAX
In-Reply-To: <CAMWiMy_b4239c9CehXfe-qMEMam4tsCffWBjkokTAzExtD88DA@mail.gmail.com>
References: <CANFMcCj9jqiWyM32caaR5qLAFizDhOrLbbpWS=_fFX30RJKUMQ@mail.gmail.com>
	<CAMWiMy_b4239c9CehXfe-qMEMam4tsCffWBjkokTAzExtD88DA@mail.gmail.com>
Message-ID: <CAHKqMb4B9cOodoAFnBSeODaJB-T921TuKZX5yD+V4Pk+X_mbZQ@mail.gmail.com>

how to do IAX is sketchy in the Wiki.  we need a more concrete example. tks.
-nandy

On Thu, Sep 22, 2011 at 7:47 AM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> FreeSWITCH doesn't support IAX... officially that is. There's always
> mod_opal, but that is considered beta quality.
>
> http://wiki.freeswitch.org/wiki/Mod_opal
>
> ...wiki gives you the necessary instructions to get you up and running
> on the IAX front ;)
>
> On Wed, Sep 21, 2011 at 9:09 AM, f.erfurth <f.erfurth at googlemail.com>
> wrote:
> > Hi,
> > I installed FreeSWITCH for BigBlueButton (Conference-System). I want call
> by
> > using IAX. Is IAX available by default or do I have some configurations
> for
> > that?
> >
> > I have a dialplan which is responsible for connecting to Conference of
> > BigBlueButton. I should use the context bbb-voip.
> > cu Floh
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gabe at gundy.org  Thu Sep 22 10:28:39 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 22 Sep 2011 00:28:39 -0600
Subject: [Freeswitch-users] how to redirect $${sounds_dir} ?
In-Reply-To: <BANLkTimfH8pR7jJb-pUxLe9b_K9WHXth4A@mail.gmail.com>
References: <BANLkTinv+-0wP0NPyAVTRsG595HPCmUObtTNfy8m0o7f7_Gspw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEE587AA@cooper>
	<BANLkTikiLd0OWyHMz0znc14MNH79HAOzSg@mail.gmail.com>
	<BANLkTimfH8pR7jJb-pUxLe9b_K9WHXth4A@mail.gmail.com>
Message-ID: <CAMwZ-tL6VvccWrhFdqfGBxhd+oM19Qw7i5Whyd1_bn2Cfk5hHg@mail.gmail.com>

On Fri, Jun 10, 2011 at 4:46 PM, Michael Collins <msc at freeswitch.org> wrote:
> On Fri, Jun 10, 2011 at 10:40 AM, jesse <chat2jesse at gmail.com> wrote:
>>
>> vars.xml only sets sound_prefix :
>>
>> ?<X-PRE-PROCESS cmd="set"
>> data="sound_prefix=$${sounds_dir}/en/us/callie"/>
>>
>> $${sounds_dir} is like $${local_ip_v4}, automatically fetched by the
>> application.
>>
>> not sure whether i can set sounds_dir in vars.xml...
>
> the sounds_dir variable you cannot - it is like the base_dir, htdocs_dir,
> etc. variables. What are you doing that requires you to change
> ${sounds_dir}? Perhaps we can help you with an alternate solution.
> -MC


I just had this same thought too... It would be nice it you could
specify a different dir for sounds as a command-line option like you
can with -conf -db -log and -run.

There are two reasons I would favor that approach...

If you have your own sounds and you don't want to put them into the
/opt/freeswitch dir because it might get in the way of updating
FreeSWITCH with debs or rpms.

Also, that kind of media is likely to be provided via some sort of
file sharing (to make sure all FS servers have the same versions of
the media). Those shares are typically mounted at
/mnt/some/random/path.  It would be pretty nice to just start with a
"-sounds /mnt/foo/bar"

Anyway, just another perspective on it.

Best,
Gabe


From chris at ghosttelecom.com  Thu Sep 22 12:30:37 2011
From: chris at ghosttelecom.com (Chris Martineau)
Date: Thu, 22 Sep 2011 09:30:37 +0100
Subject: [Freeswitch-users] Multiple apps with Execute_on_answer
Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF359B0A@SVR01.ghosttelecom.local>

Hi,

 

How do you execute multiple applications on answer?

I need to execute a sched_hangup and a sched_broadcast on answer so that
the timers start from answer. However cannot find any combination which 

allows me to put multiple apps in the same execute on answer action. 

If I just create multiple action entries it only seems to action the
last one entered?

So have...

application='set' data='execute_on_answer=sched_hangup +XX
allotted_timeout'

if I add...

application='set' data='execute_on_answer=sched_broadcast +XX
/path/file'

only the broadcast triggers?

 

Many thanks again

 

Chris

 

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From chris at ghosttelecom.com  Thu Sep 22 13:50:15 2011
From: chris at ghosttelecom.com (Chris Martineau)
Date: Thu, 22 Sep 2011 10:50:15 +0100
Subject: [Freeswitch-users] Terminate early media playback on answer
Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF359B29@SVR01.ghosttelecom.local>

Hi,

 

I am using a playback on early media to provide a welcome message and
would like to terminate the playback on answer.

 

I have tried

 

application='set' data='execute_on_answer=break all'

 

but this doesn't seem to do anything even though the log shows it being
executed on answer.

 

Currently if the far end answers quickly they get silence until the
playback has finished and then they get through.

 

Any ideas on how I can do this?

 

Many thanks again

 

Chris

 

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From chris at ghosttelecom.com  Thu Sep 22 13:55:00 2011
From: chris at ghosttelecom.com (Chris Martineau)
Date: Thu, 22 Sep 2011 10:55:00 +0100
Subject: [Freeswitch-users] Multiple apps with Execute_on_answer
In-Reply-To: <1D10AB188D6CCA46BB4369E3268E36EF359B0A@SVR01.ghosttelecom.local>
References: <1D10AB188D6CCA46BB4369E3268E36EF359B0A@SVR01.ghosttelecom.local>
Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF359B2B@SVR01.ghosttelecom.local>

Sorry, just found this.

 

Thanks

 

Chris

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Chris Martineau
Sent: 22 September 2011 09:31
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Multiple apps with Execute_on_answer

 

Hi,

 

How do you execute multiple applications on answer?

I need to execute a sched_hangup and a sched_broadcast on answer so that
the timers start from answer. However cannot find any combination which 

allows me to put multiple apps in the same execute on answer action. 

If I just create multiple action entries it only seems to action the
last one entered?

So have...

application='set' data='execute_on_answer=sched_hangup +XX
allotted_timeout'

if I add...

application='set' data='execute_on_answer=sched_broadcast +XX
/path/file'

only the broadcast triggers?

 

Many thanks again

 

Chris

 

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From nbhatti at gmail.com  Thu Sep 22 15:31:42 2011
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Thu, 22 Sep 2011 14:31:42 +0300
Subject: [Freeswitch-users] ACL to fallback digest authentication
Message-ID: <CACak_yxqX0WLp7Pa_zgMbTfWE2NnGLEoSuvVg9Yjj9M9bzJzXw@mail.gmail.com>

Hello,
I am trying to implement ACL in FreeSWITCH. What I want to do is, check the
user's IP in the ACL. If that is present, let the user send INVITE. If ACL
fails, fallback to digest authentication and authentication user/pass.
Problem is that when I do auth_calls true, I am not able to INVITE. We have
actually two scenarios, 1 make call without REGISTER, second REGISTER and
make calls. I have set default ACL to deny, but that still not helping.
Either one of them works, with the digest or ACL. I don't want to create
another profile and make 1 for authentication, and the other without auth.
Is there any workaround for this?

-B
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From fernando.berretta at gmail.com  Thu Sep 22 15:39:25 2011
From: fernando.berretta at gmail.com (Fernando Berretta)
Date: Thu, 22 Sep 2011 08:39:25 -0300
Subject: [Freeswitch-users] G.729
In-Reply-To: <CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>
References: <4E79E8B9.1040402@gmail.com>
	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>
Message-ID: <4E7B1E6D.4090705@gmail.com>

Kristian,

It was a typo.. the question would be.. is there some way to install 
g729 codec in Freeswitch for free, in order to "try" .....

Regards,

On 9/21/2011 10:54 AM, Kristian Kielhofner wrote:
> This is a mailing list for FreeSWITCH discussion, not FreePBX.
>
> On Wed, Sep 21, 2011 at 9:38 AM, Fernando Berretta
> <fernando.berretta at gmail.com>  wrote:
>> Hi,
>>
>> Is there some way to install g729 codec in freepbx for free, in order to
>> "try" it like is provided for asterisk ?
>>
>> Best Regards,
>> Fernando
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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  Thu Sep 22 15:51:32 2011
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Thu, 22 Sep 2011 14:51:32 +0300
Subject: [Freeswitch-users] G.729
In-Reply-To: <4E7B1E6D.4090705@gmail.com>
References: <4E79E8B9.1040402@gmail.com>
	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>
	<4E7B1E6D.4090705@gmail.com>
Message-ID: <CACak_yzPA0e4kk0guxZXhrVPYt+jzqU-bT20c8bCo3X3oDVqMQ@mail.gmail.com>

Take a look here, https://github.com/Deepwalker/fs_itu_g729


On Thu, Sep 22, 2011 at 2:39 PM, Fernando Berretta <
fernando.berretta at gmail.com> wrote:

> Kristian,
>
> It was a typo.. the question would be.. is there some way to install
> g729 codec in Freeswitch for free, in order to "try" .....
>
> Regards,
>
> On 9/21/2011 10:54 AM, Kristian Kielhofner wrote:
> > This is a mailing list for FreeSWITCH discussion, not FreePBX.
> >
> > On Wed, Sep 21, 2011 at 9:38 AM, Fernando Berretta
> > <fernando.berretta at gmail.com>  wrote:
> >> Hi,
> >>
> >> Is there some way to install g729 codec in freepbx for free, in order to
> >> "try" it like is provided for asterisk ?
> >>
> >> Best Regards,
> >> Fernando
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From avi at avimarcus.net  Thu Sep 22 15:54:57 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 22 Sep 2011 14:54:57 +0300
Subject: [Freeswitch-users] G.729
In-Reply-To: <4E7B1E6D.4090705@gmail.com>
References: <4E79E8B9.1040402@gmail.com>
	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>
	<4E7B1E6D.4090705@gmail.com>
Message-ID: <CALNZuw=ZGO5WqQ3rYcHMcgnq0+5TQ=2HabHdLnAuMAMtZ6dq5A@mail.gmail.com>

What do you need to test? The g729 works for as many channels as you pay
for...

-Avi


On Thu, Sep 22, 2011 at 2:39 PM, Fernando Berretta <
fernando.berretta at gmail.com> wrote:

> Kristian,
>
> It was a typo.. the question would be.. is there some way to install
> g729 codec in Freeswitch for free, in order to "try" .....
>
> Regards,
>
> On 9/21/2011 10:54 AM, Kristian Kielhofner wrote:
> > This is a mailing list for FreeSWITCH discussion, not FreePBX.
> >
> > On Wed, Sep 21, 2011 at 9:38 AM, Fernando Berretta
> > <fernando.berretta at gmail.com>  wrote:
> >> Hi,
> >>
> >> Is there some way to install g729 codec in freepbx for free, in order to
> >> "try" it like is provided for asterisk ?
> >>
> >> Best Regards,
> >> Fernando
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From avi at avimarcus.net  Thu Sep 22 15:56:29 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 22 Sep 2011 14:56:29 +0300
Subject: [Freeswitch-users] Multiple apps with Execute_on_answer
In-Reply-To: <1D10AB188D6CCA46BB4369E3268E36EF359B2B@SVR01.ghosttelecom.local>
References: <1D10AB188D6CCA46BB4369E3268E36EF359B0A@SVR01.ghosttelecom.local>
	<1D10AB188D6CCA46BB4369E3268E36EF359B2B@SVR01.ghosttelecom.local>
Message-ID: <CALNZuwncsno2cGpwsLEAy-psPVxoG_dxVw_iB+1cc7mfrLbnmQ@mail.gmail.com>

How to do multiple:
http://wiki.freeswitch.org/wiki/Channel_Variables#The_execute_on_family

... in case anyone else wanted to know.

-Avi


On Thu, Sep 22, 2011 at 12:55 PM, Chris Martineau <chris at ghosttelecom.com>wrote:

> Sorry, just found this.****
>
> ** **
>
> Thanks****
>
> ** **
>
> Chris****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris
> Martineau
> *Sent:* 22 September 2011 09:31
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Multiple apps with Execute_on_answer****
>
> ** **
>
> Hi,****
>
> ** **
>
> How do you execute multiple applications on answer?****
>
> I need to execute a sched_hangup and a sched_broadcast on answer so that
> the timers start from answer. However cannot find any combination which **
> **
>
> allows me to put multiple apps in the same execute on answer action. ****
>
> If I just create multiple action entries it only seems to action the last
> one entered?****
>
> So have...****
>
> application='set' data='execute_on_answer=sched_hangup +XX
> allotted_timeout'****
>
> if I add...****
>
> application='set' data='execute_on_answer=sched_broadcast +XX /path/file'*
> ***
>
> only the broadcast triggers?****
>
> ** **
>
> Many thanks again****
>
> ** **
>
> Chris****
>
> ** **
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From alec.taylor6 at gmail.com  Thu Sep 22 18:59:53 2011
From: alec.taylor6 at gmail.com (Alec Taylor)
Date: Fri, 23 Sep 2011 00:59:53 +1000
Subject: [Freeswitch-users] Weekly FreeSwitch conference Call
In-Reply-To: <CA9F7C9A.1A39B%krice@freeswitch.org>
References: <Acx4fIgBDyyomwMLDkyUbnHT/Dww6w==>
	<CA9F7C9A.1A39B%krice@freeswitch.org>
Message-ID: <CAO+9iGc_WcXTwwBOFcwBtfAkCF9=-xTmsV4pxJ1AKLoLL87fbQ@mail.gmail.com>

Hi Ken,

Kind of unrelated, but I've been searching for quite some time to
create a setup just like freeswitch conference has.

How can I setup my own website like yours?

Thanks for all tutorials,

Alec Taylor

On Thu, Sep 22, 2011 at 2:35 AM, Ken Rice <krice at freeswitch.org> wrote:
> Hey Guys,
>
> Michael and Brian have asked me to help out with the conference today since
> Michael is otherwise occupied today...
>
> Open Forum for the most part today, however, we will have some discussion of
> hotels for Cluecon 2012, and some more embedded talk. Plus more....
>
> Join us at 1PM Eastern thats 10AM Pacific or 5PM GMT Today!
>
> Just dial *9888 from your FreeSwitch installs that have the default dialplan
> or join as via the web at http://conference.freeswitch.org/
>
> K
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From krice at freeswitch.org  Thu Sep 22 19:14:01 2011
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 22 Sep 2011 10:14:01 -0500
Subject: [Freeswitch-users] Weekly FreeSwitch conference Call
In-Reply-To: <CAO+9iGc_WcXTwwBOFcwBtfAkCF9=-xTmsV4pxJ1AKLoLL87fbQ@mail.gmail.com>
Message-ID: <CAA0BAE9.1A4CE%krice@freeswitch.org>

The Conference.freeswitch.org website uses mod_rtmp and uses the RTMP client
that's in tree. Most of the code is there to create a similar site.

Also check out some of the conference code in the various contrib trees.

Also if you need some professional help contact me offlist or contact
consulting at freeswitch.org

K


On 9/22/11 9:59 AM, "Alec Taylor" <alec.taylor6 at gmail.com> wrote:

> Hi Ken,
> 
> Kind of unrelated, but I've been searching for quite some time to
> create a setup just like freeswitch conference has.
> 
> How can I setup my own website like yours?
> 
> Thanks for all tutorials,
> 
> Alec Taylor
> 
> On Thu, Sep 22, 2011 at 2:35 AM, Ken Rice <krice at freeswitch.org> wrote:
>> Hey Guys,
>> 
>> Michael and Brian have asked me to help out with the conference today since
>> Michael is otherwise occupied today...
>> 
>> Open Forum for the most part today, however, we will have some discussion of
>> hotels for Cluecon 2012, and some more embedded talk. Plus more....
>> 
>> Join us at 1PM Eastern thats 10AM Pacific or 5PM GMT Today!
>> 
>> Just dial *9888 from your FreeSwitch installs that have the default dialplan
>> or join as via the web at http://conference.freeswitch.org/
>> 
>> 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
>> 
>> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 vipkilla at gmail.com  Thu Sep 22 19:45:55 2011
From: vipkilla at gmail.com (vip killa)
Date: Thu, 22 Sep 2011 11:45:55 -0400
Subject: [Freeswitch-users] Fwd: [asterisk-users] VoIP Abuse to Twitter
	(real time VoIP Abuse)
In-Reply-To: <033301cc7939$d5cf2140$816d63c0$@com>
References: <4E7B483F.2030901@tormenting.net>
	<CAC-LwPN4MrMzatm64izoos8V+ZhagaPRxNuN8M9z+CnDWOxocw@mail.gmail.com>
	<033301cc7939$d5cf2140$816d63c0$@com>
Message-ID: <CAC-LwPNsMs6QFE5G2B0==yh5ZmRZKx4WZHOxRSCgCgE1ERzVyw@mail.gmail.com>

---------- Forwarded message ----------
From: Robert Huddleston <rhuddleston at gmail.com>
Date: Thu, Sep 22, 2011 at 11:10 AM
Subject: Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP Abuse)
To: Asterisk Users Mailing List - Non-Commercial Discussion <
asterisk-users at lists.digium.com>


Sounds like a great idea.. Hopefully the page/account never gets hacked and
bad IP?s published.. I could see a great hack of ****

127.0.0.1  ****

192.168.0.0/16 ****

10.0.0.0/8 ****

getting up there somehow and next thing you know ? BAM!****

** **

But I haven?t RTFM ? I?m guessing there is probably a white list that
supersedes the naughty list.****

** **

** **

*From:* asterisk-users-bounces at lists.digium.com [mailto:
asterisk-users-bounces at lists.digium.com] *On Behalf Of *vip killa
*Sent:* Thursday, September 22, 2011 11:06 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP Abuse)
****

** **

very cool!****

On Thu, Sep 22, 2011 at 10:37 AM, J. Oquendo <asterisk at tormenting.net>
wrote:****


Apologies for cross posting but some of us aren't on the other list
(vice/versa) and thought both groups would benefit.

For those familiar with the VoIP Abuse Project, no need to explain the
gist of this. I got tired of parsing through the alerts (lists) I
receive via email daily. They're long and sometimes I don't have the
time to post them all. So for now, posting VoIP Abuse addresses straight
to Twitter.

So, anyone trying to compromise a pbx, is now autoposted on an hourly
basis to Twitter. Still working on pulling, have about 4 machines linked
up now, will mop em up during the week.

http://twitter.com/#!/voipabuse

Now, you can concoct a quick script off of it, e.g.:

links -dump "http://twitter.com/voipabuse"|awk '/attacker/{print
"iptables -A INPUT -s "$2" -j DROP"| "sort -u"}'

Will get a quickie soon from my Acme's, nCites, etc. when I have time.

For those NOT familiar with it, please Google it as I don't feel like
typing anymore ;) (sorry)



--

=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT, RWSP, GREM

"It takes 20 years to build a reputation and five minutes to
ruin it. If you think about that, you'll do things
differently." - Warren Buffett

42B0 5A53 6505 6638 44BB  3943 2BF7 D83F 210A 95AF
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x2BF7D83F210A95AF


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users****

** **

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
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From curriegrad2004 at gmail.com  Thu Sep 22 20:42:19 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 22 Sep 2011 09:42:19 -0700
Subject: [Freeswitch-users] G.729
In-Reply-To: <CALNZuw=ZGO5WqQ3rYcHMcgnq0+5TQ=2HabHdLnAuMAMtZ6dq5A@mail.gmail.com>
References: <4E79E8B9.1040402@gmail.com>
	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>
	<4E7B1E6D.4090705@gmail.com>
	<CALNZuw=ZGO5WqQ3rYcHMcgnq0+5TQ=2HabHdLnAuMAMtZ6dq5A@mail.gmail.com>
Message-ID: <CAMWiMy_WfpAiiw8-tEKe9F5mcq3Q34sqStp5FdHjdr1uAoS6VA@mail.gmail.com>

There is no G729 trial codecs in FreeSWITCH. You need to pay royalties
to SIPPRO regardless. There's always the option of waiting until
somewhere 2014 for the patents to expire ;)

On Thu, Sep 22, 2011 at 4:54 AM, Avi Marcus <avi at avimarcus.net> wrote:
> What do you need to test? The g729 works for as many channels as you pay
> for...
> -Avi
>
> On Thu, Sep 22, 2011 at 2:39 PM, Fernando Berretta
> <fernando.berretta at gmail.com> wrote:
>>
>> Kristian,
>>
>> It was a typo.. the question would be.. is there some way to install
>> g729 codec in Freeswitch for free, in order to "try" .....
>>
>> Regards,
>>
>> On 9/21/2011 10:54 AM, Kristian Kielhofner wrote:
>> > This is a mailing list for FreeSWITCH discussion, not FreePBX.
>> >
>> > On Wed, Sep 21, 2011 at 9:38 AM, Fernando Berretta
>> > <fernando.berretta at gmail.com>  wrote:
>> >> Hi,
>> >>
>> >> Is there some way to install g729 codec in freepbx for free, in order
>> >> to
>> >> "try" it like is provided for asterisk ?
>> >>
>> >> Best Regards,
>> >> Fernando
>> >>
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 curriegrad2004 at gmail.com  Thu Sep 22 20:43:01 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 22 Sep 2011 09:43:01 -0700
Subject: [Freeswitch-users] Weekly FreeSwitch conference Call
In-Reply-To: <CAA0BAE9.1A4CE%krice@freeswitch.org>
References: <CAO+9iGc_WcXTwwBOFcwBtfAkCF9=-xTmsV4pxJ1AKLoLL87fbQ@mail.gmail.com>
	<CAA0BAE9.1A4CE%krice@freeswitch.org>
Message-ID: <CAMWiMy_-Hvtx7Aty9ug1Hi69yOq+TzJSv_s5Aw_3+4cUsGhQ7g@mail.gmail.com>

Yeah, where's last weeks and this week's conf call?

On Thu, Sep 22, 2011 at 8:14 AM, Ken Rice <krice at freeswitch.org> wrote:
> The Conference.freeswitch.org website uses mod_rtmp and uses the RTMP client
> that's in tree. Most of the code is there to create a similar site.
>
> Also check out some of the conference code in the various contrib trees.
>
> Also if you need some professional help contact me offlist or contact
> consulting at freeswitch.org
>
> K
>
>
> On 9/22/11 9:59 AM, "Alec Taylor" <alec.taylor6 at gmail.com> wrote:
>
>> Hi Ken,
>>
>> Kind of unrelated, but I've been searching for quite some time to
>> create a setup just like freeswitch conference has.
>>
>> How can I setup my own website like yours?
>>
>> Thanks for all tutorials,
>>
>> Alec Taylor
>>
>> On Thu, Sep 22, 2011 at 2:35 AM, Ken Rice <krice at freeswitch.org> wrote:
>>> Hey Guys,
>>>
>>> Michael and Brian have asked me to help out with the conference today since
>>> Michael is otherwise occupied today...
>>>
>>> Open Forum for the most part today, however, we will have some discussion of
>>> hotels for Cluecon 2012, and some more embedded talk. Plus more....
>>>
>>> Join us at 1PM Eastern thats 10AM Pacific or 5PM GMT Today!
>>>
>>> Just dial *9888 from your FreeSwitch installs that have the default dialplan
>>> or join as via the web at http://conference.freeswitch.org/
>>>
>>> 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
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 alec.taylor6 at gmail.com  Thu Sep 22 20:49:20 2011
From: alec.taylor6 at gmail.com (Alec Taylor)
Date: Fri, 23 Sep 2011 02:49:20 +1000
Subject: [Freeswitch-users] Weekly FreeSwitch conference Call
In-Reply-To: <CAA0BAE9.1A4CE%krice@freeswitch.org>
References: <CAO+9iGc_WcXTwwBOFcwBtfAkCF9=-xTmsV4pxJ1AKLoLL87fbQ@mail.gmail.com>
	<CAA0BAE9.1A4CE%krice@freeswitch.org>
Message-ID: <CAO+9iGeM0JpDvper6d+dQmzjh+QNy4VbQNiRHmFf=4CKR1ad0Q@mail.gmail.com>

Thanks, I'll see what I can build from it.

What are the various licenses? - AGPL?

On Fri, Sep 23, 2011 at 1:14 AM, Ken Rice <krice at freeswitch.org> wrote:
> The Conference.freeswitch.org website uses mod_rtmp and uses the RTMP client
> that's in tree. Most of the code is there to create a similar site.
>
> Also check out some of the conference code in the various contrib trees.
>
> Also if you need some professional help contact me offlist or contact
> consulting at freeswitch.org
>
> K
>
>
> On 9/22/11 9:59 AM, "Alec Taylor" <alec.taylor6 at gmail.com> wrote:
>
>> Hi Ken,
>>
>> Kind of unrelated, but I've been searching for quite some time to
>> create a setup just like freeswitch conference has.
>>
>> How can I setup my own website like yours?
>>
>> Thanks for all tutorials,
>>
>> Alec Taylor
>>
>> On Thu, Sep 22, 2011 at 2:35 AM, Ken Rice <krice at freeswitch.org> wrote:
>>> Hey Guys,
>>>
>>> Michael and Brian have asked me to help out with the conference today since
>>> Michael is otherwise occupied today...
>>>
>>> Open Forum for the most part today, however, we will have some discussion of
>>> hotels for Cluecon 2012, and some more embedded talk. Plus more....
>>>
>>> Join us at 1PM Eastern thats 10AM Pacific or 5PM GMT Today!
>>>
>>> Just dial *9888 from your FreeSwitch installs that have the default dialplan
>>> or join as via the web at http://conference.freeswitch.org/
>>>
>>> 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
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 krice at freeswitch.org  Thu Sep 22 20:53:04 2011
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 22 Sep 2011 11:53:04 -0500
Subject: [Freeswitch-users] Weekly FreeSwitch conference Call
In-Reply-To: <CAO+9iGeM0JpDvper6d+dQmzjh+QNy4VbQNiRHmFf=4CKR1ad0Q@mail.gmail.com>
Message-ID: <CAA0D220.1A4ED%krice@freeswitch.org>

Most everything in FreeSwitch is MPL...

I'll fight the urge to comment on the AGPL tho...

K


On 9/22/11 11:49 AM, "Alec Taylor" <alec.taylor6 at gmail.com> wrote:

> Thanks, I'll see what I can build from it.
> 
> What are the various licenses? - AGPL?
> 
> On Fri, Sep 23, 2011 at 1:14 AM, Ken Rice <krice at freeswitch.org> wrote:
>> The Conference.freeswitch.org website uses mod_rtmp and uses the RTMP client
>> that's in tree. Most of the code is there to create a similar site.
>> 
>> Also check out some of the conference code in the various contrib trees.
>> 
>> Also if you need some professional help contact me offlist or contact
>> consulting at freeswitch.org
>> 
>> K
>> 
>> 
>> On 9/22/11 9:59 AM, "Alec Taylor" <alec.taylor6 at gmail.com> wrote:
>> 
>>> Hi Ken,
>>> 
>>> Kind of unrelated, but I've been searching for quite some time to
>>> create a setup just like freeswitch conference has.
>>> 
>>> How can I setup my own website like yours?
>>> 
>>> Thanks for all tutorials,
>>> 
>>> Alec Taylor
>>> 
>>> On Thu, Sep 22, 2011 at 2:35 AM, Ken Rice <krice at freeswitch.org> wrote:
>>>> Hey Guys,
>>>> 
>>>> Michael and Brian have asked me to help out with the conference today since
>>>> Michael is otherwise occupied today...
>>>> 
>>>> Open Forum for the most part today, however, we will have some discussion
>>>> of
>>>> hotels for Cluecon 2012, and some more embedded talk. Plus more....
>>>> 
>>>> Join us at 1PM Eastern thats 10AM Pacific or 5PM GMT Today!
>>>> 
>>>> Just dial *9888 from your FreeSwitch installs that have the default
>>>> dialplan
>>>> or join as via the web at http://conference.freeswitch.org/
>>>> 
>>>> 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
>>>> 
>>>> 
>>> 
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 alec.taylor6 at gmail.com  Thu Sep 22 20:54:56 2011
From: alec.taylor6 at gmail.com (Alec Taylor)
Date: Fri, 23 Sep 2011 02:54:56 +1000
Subject: [Freeswitch-users] Weekly FreeSwitch conference Call
In-Reply-To: <CAA0D220.1A4ED%krice@freeswitch.org>
References: <CAO+9iGeM0JpDvper6d+dQmzjh+QNy4VbQNiRHmFf=4CKR1ad0Q@mail.gmail.com>
	<CAA0D220.1A4ED%krice@freeswitch.org>
Message-ID: <CAO+9iGfd_jUFXodDE0TEpTftYnkX5LuAEpDwuq8QwpWN-kv8OA@mail.gmail.com>

Any chance of releasing the website under MPL?

On Fri, Sep 23, 2011 at 2:53 AM, Ken Rice <krice at freeswitch.org> wrote:
> Most everything in FreeSwitch is MPL...
>
> I'll fight the urge to comment on the AGPL tho...
>
> K
>
>
> On 9/22/11 11:49 AM, "Alec Taylor" <alec.taylor6 at gmail.com> wrote:
>
>> Thanks, I'll see what I can build from it.
>>
>> What are the various licenses? - AGPL?
>>
>> On Fri, Sep 23, 2011 at 1:14 AM, Ken Rice <krice at freeswitch.org> wrote:
>>> The Conference.freeswitch.org website uses mod_rtmp and uses the RTMP client
>>> that's in tree. Most of the code is there to create a similar site.
>>>
>>> Also check out some of the conference code in the various contrib trees.
>>>
>>> Also if you need some professional help contact me offlist or contact
>>> consulting at freeswitch.org
>>>
>>> K
>>>
>>>
>>> On 9/22/11 9:59 AM, "Alec Taylor" <alec.taylor6 at gmail.com> wrote:
>>>
>>>> Hi Ken,
>>>>
>>>> Kind of unrelated, but I've been searching for quite some time to
>>>> create a setup just like freeswitch conference has.
>>>>
>>>> How can I setup my own website like yours?
>>>>
>>>> Thanks for all tutorials,
>>>>
>>>> Alec Taylor
>>>>
>>>> On Thu, Sep 22, 2011 at 2:35 AM, Ken Rice <krice at freeswitch.org> wrote:
>>>>> Hey Guys,
>>>>>
>>>>> Michael and Brian have asked me to help out with the conference today since
>>>>> Michael is otherwise occupied today...
>>>>>
>>>>> Open Forum for the most part today, however, we will have some discussion
>>>>> of
>>>>> hotels for Cluecon 2012, and some more embedded talk. Plus more....
>>>>>
>>>>> Join us at 1PM Eastern thats 10AM Pacific or 5PM GMT Today!
>>>>>
>>>>> Just dial *9888 from your FreeSwitch installs that have the default
>>>>> dialplan
>>>>> or join as via the web at http://conference.freeswitch.org/
>>>>>
>>>>> 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
>>>>>
>>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Sep 22 23:00:42 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Thu, 22 Sep 2011 19:00:42 +0000
Subject: [Freeswitch-users] large block of DID's
In-Reply-To: <801470822-1316057954-cardhu_decombobulator_blackberry.rim.net-146923873-@b5.c27.bise6.blackberry>
References: <D25CC3E5-C0D2-41D7-87C3-843C24A0CA8C@lyonl.com>
	<801470822-1316057954-cardhu_decombobulator_blackberry.rim.net-146923873-@b5.c27.bise6.blackberry>
Message-ID: <2C4ADC11-8559-45ED-94E9-435F1D60CA78@lyonl.com>

Are you also using a lua script via odbc to manage users?


On Sep 14, 2011, at 10:39 PM, <ljjimenez at gmail.com>
 <ljjimenez at gmail.com> wrote:

> We do this with about 10k DIDs, we use MySQL and a Lua script via ODBC
> 
> Luis Jimenez
> 
> -----Original Message-----
> From: Chad Vogel <cvogel at lyonl.com>
> Sender: freeswitch-users-bounces at lists.freeswitch.org
> Date: Wed, 14 Sep 2011 15:44:23 
> To: FreeSWITCH-users at lists.freeswitch.org<FreeSWITCH-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] large block of DID's
> 
> I was wondering if anyone has any thoughts what is the best way to manage a large blocks of DID? (we have about 1600 DID's that we are moving off of our asterisk servers) 
> 
> Chad
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From infos at madovsky.org  Thu Sep 22 23:15:47 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 22 Sep 2011 15:15:47 -0400
Subject: [Freeswitch-users] last git and postgresql 9.1
Message-ID: <BC90B94B20914D3DA9C82856F6FFABAF@e1705>

Hi Folks,

only to say that I just upgraded to PG 9.1 and everything is working well

Cheers

Franck
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From moises.silva at gmail.com  Thu Sep 22 23:46:12 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 22 Sep 2011 15:46:12 -0400
Subject: [Freeswitch-users] Codec byte order
In-Reply-To: <CABtP9Qj62ihNHYgZoGb-_1Q7JjbxSbQSmCh=P976GPytHHoBQg@mail.gmail.com>
References: <CABtP9Qj62ihNHYgZoGb-_1Q7JjbxSbQSmCh=P976GPytHHoBQg@mail.gmail.com>
Message-ID: <CAA4nhyDjWPdGra7CCeUwjigm3DsTB8wqJjN0StJB7ijbYxjCKA@mail.gmail.com>

On Tue, Sep 20, 2011 at 5:56 AM, Fabio Bigliardi
<fabio.bigliardi at gmail.com> wrote:
> Hi all,
> I have an audio decoder registered to freeswitch server.
> It worked with a codec L16 at 24000h little endian.
> After software update to FreeSWITCH Version 1.0.head (git-25c725c 2011-06-30
> 18-30-24 -0500), the audio decoder ceased to work with this codec. It
> produces just noise.
> With the big endian version of the codec, it works again.
> The hardware on which freeswitch is running, is the same.
> How could you explain this behaviour?

This explains it:

commit e657e32fcac6f29fabfea8e38ce7a7dcf5beb8af
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Mon Mar 21 14:31:10 2011 -0500

You can use rtp_disable_byteswap variable to disable the byte
swapping. In the RTP stream, the byte order is supposed to be big
endian.

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 avi at avimarcus.net  Thu Sep 22 23:54:10 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 22 Sep 2011 22:54:10 +0300
Subject: [Freeswitch-users] large block of DID's
In-Reply-To: <2C4ADC11-8559-45ED-94E9-435F1D60CA78@lyonl.com>
References: <D25CC3E5-C0D2-41D7-87C3-843C24A0CA8C@lyonl.com>
	<801470822-1316057954-cardhu_decombobulator_blackberry.rim.net-146923873-@b5.c27.bise6.blackberry>
	<2C4ADC11-8559-45ED-94E9-435F1D60CA78@lyonl.com>
Message-ID: <CALNZuwnhT26xOEDAR9iGtcSicOtx6L8F1O6vzEBBCAwAX4ziig@mail.gmail.com>

I'd be interested in seeing the Lua script that does the users/directory. I
don't think I've seen one yet, I'm not quite sure how that would work. I've
only used xml_curl for a dialplan with a web server..
-Avi



On Thu, Sep 22, 2011 at 10:00 PM, Chad Vogel <cvogel at lyonl.com> wrote:

> Are you also using a lua script via odbc to manage users?
>
>
> On Sep 14, 2011, at 10:39 PM, <ljjimenez at gmail.com>
>  <ljjimenez at gmail.com> wrote:
>
> > We do this with about 10k DIDs, we use MySQL and a Lua script via ODBC
> >
> > Luis Jimenez
> >
> > -----Original Message-----
> > From: Chad Vogel <cvogel at lyonl.com>
> > Sender: freeswitch-users-bounces at lists.freeswitch.org
> > Date: Wed, 14 Sep 2011 15:44:23
> > To: FreeSWITCH-users at lists.freeswitch.org<
> FreeSWITCH-users at lists.freeswitch.org>
> > Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> > Subject: [Freeswitch-users] large block of DID's
> >
> > I was wondering if anyone has any thoughts what is the best way to manage
> a large blocks of DID? (we have about 1600 DID's that we are moving off of
> our asterisk servers)
> >
> > Chad
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Fri Sep 23 00:04:56 2011
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 22 Sep 2011 15:04:56 -0500
Subject: [Freeswitch-users] Weekly FreeSwitch conference Call
In-Reply-To: <CAO+9iGfd_jUFXodDE0TEpTftYnkX5LuAEpDwuq8QwpWN-kv8OA@mail.gmail.com>
Message-ID: <CAA0FF18.1A508%krice@freeswitch.org>

I didn't do that website... But maybe one of the guys that did can speak to
that if they choose... Keep in mind that you can build this yourself to your
specs... 

The FreeSwitch Developers already give you most of the tools to do these
things. Tony, Brian, Mike and Crew work very hard so that we can do the
things we want to do, however, they all have families and bills they need to
take care of...

Ken


On 9/22/11 11:54 AM, "Alec Taylor" <alec.taylor6 at gmail.com> wrote:

> Any chance of releasing the website under MPL?
> 
> On Fri, Sep 23, 2011 at 2:53 AM, Ken Rice <krice at freeswitch.org> wrote:
>> Most everything in FreeSwitch is MPL...
>> 
>> I'll fight the urge to comment on the AGPL tho...
>> 
>> K
>> 
>> 
>> On 9/22/11 11:49 AM, "Alec Taylor" <alec.taylor6 at gmail.com> wrote:
>> 
>>> Thanks, I'll see what I can build from it.
>>> 
>>> What are the various licenses? - AGPL?
>>> 
>>> On Fri, Sep 23, 2011 at 1:14 AM, Ken Rice <krice at freeswitch.org> wrote:
>>>> The Conference.freeswitch.org website uses mod_rtmp and uses the RTMP
>>>> client
>>>> that's in tree. Most of the code is there to create a similar site.
>>>> 
>>>> Also check out some of the conference code in the various contrib trees.
>>>> 
>>>> Also if you need some professional help contact me offlist or contact
>>>> consulting at freeswitch.org
>>>> 
>>>> K
>>>> 
>>>> 
>>>> On 9/22/11 9:59 AM, "Alec Taylor" <alec.taylor6 at gmail.com> wrote:
>>>> 
>>>>> Hi Ken,
>>>>> 
>>>>> Kind of unrelated, but I've been searching for quite some time to
>>>>> create a setup just like freeswitch conference has.
>>>>> 
>>>>> How can I setup my own website like yours?
>>>>> 
>>>>> Thanks for all tutorials,
>>>>> 
>>>>> Alec Taylor
>>>>> 
>>>>> On Thu, Sep 22, 2011 at 2:35 AM, Ken Rice <krice at freeswitch.org> wrote:
>>>>>> Hey Guys,
>>>>>> 
>>>>>> Michael and Brian have asked me to help out with the conference today
>>>>>> since
>>>>>> Michael is otherwise occupied today...
>>>>>> 
>>>>>> Open Forum for the most part today, however, we will have some discussion
>>>>>> of
>>>>>> hotels for Cluecon 2012, and some more embedded talk. Plus more....
>>>>>> 
>>>>>> Join us at 1PM Eastern thats 10AM Pacific or 5PM GMT Today!
>>>>>> 
>>>>>> Just dial *9888 from your FreeSwitch installs that have the default
>>>>>> dialplan
>>>>>> or join as via the web at http://conference.freeswitch.org/
>>>>>> 
>>>>>> 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
>>>>>> 
>>>>>> 
>>>>> 
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From eagle.antonio at gmail.com  Fri Sep 23 00:54:18 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Thu, 22 Sep 2011 21:54:18 +0100
Subject: [Freeswitch-users] Freeswitch Does Not Build
Message-ID: <CAF+3j-us-Oi5rDYNMvez++ASFt5BP__oaErR0bJhBnYvXkdsYA@mail.gmail.com>

At the time Date:   Thu Sep 22 21:11:43 2011 +0200

Freeswitch fails with
make[5]: Entering directory
`/usr/local/src/freeswitch/src/mod/asr_tts/mod_unimrcp'
make[6]: Entering directory `/usr/local/src/freeswitch/libs/unimrcp'
make[6]: *** No rule to make target `build/acmacros/swift.m4', needed by
`Makefile.in'.  Stop.
make[6]: Leaving directory `/usr/local/src/freeswitch/libs/unimrcp'
make[5]: ***
[/usr/local/src/freeswitch/libs/unimrcp/platforms/libunimrcp-client/
libunimrcpclient.la] Error 2
make[5]: Leaving directory
`/usr/local/src/freeswitch/src/mod/asr_tts/mod_unimrcp'
make[4]: *** [mod_unimrcp-all] Error 1
make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod'
make[3]: *** [all-recursive] Error 1
make[3]: Leaving directory `/usr/local/src/freeswitch/src'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/usr/local/src/freeswitch'
make[1]: *** [all] Error 2
make[1]: Leaving directory `/usr/local/src/freeswitch'
make: *** [current] Error 2


Or is just me ?

Regards
A/T
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From fssupport at highergear.com  Thu Sep 22 17:23:28 2011
From: fssupport at highergear.com (fssupport at highergear.com)
Date: Thu, 22 Sep 2011 08:23:28 -0500 (CDT)
Subject: [Freeswitch-users] Share PRI line?
In-Reply-To: <CAP1_MOC5QOY2XQw2GgktyWH8qHDPtj5w57JQJ0=zRZftazKo0g@mail.gmail.com>
References: <56122.67.95.76.98.1316625631.squirrel@webmail.highergear.com>
	<CAP1_MOC5QOY2XQw2GgktyWH8qHDPtj5w57JQJ0=zRZftazKo0g@mail.gmail.com>
Message-ID: <60205.67.95.76.98.1316697808.squirrel@webmail.highergear.com>

> On Wed, Sep 21, 2011 at 6:20 PM,  <fssupport at highergear.com> wrote:
>> We have a client with a Mitel PBX with a PRI connected to it. ?We are
>> deploying a freeswitch box that will do some autodialing tasks for them.
>> Is there a way to share the PRI between the two systems? ?If nothing
>> else,
>> I'm pretty sure I can put a two line interface card in the Freeswitch
>> server, plug the pri into it, then connected the other interface to the
>> Mitel box, but that would make our box a point of failure and I'd like
>> to
>> avoid that if possible.
>>
>
> It is not possible to "split" or "share" a PRI interface with passive
> components, your three options are
>
> a/ put a second PRI interface onto "your" system, and configure it to
> route inbound calls to the Mitel system and outbound calls from the
> Mitel system
> b/ put a second PRI interface onto the Mitel, and configure it as per
> a/ but with your system instead
> c/ purchase a device with 3 PRI interfaces that can route inbound
> calls and outbound calls appropriately as per a/ and b/
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
Thanks for your response on this.  With regards to option c, I found this:
http://www.patapsco.co.uk/Products/Liberator/ISDN_Splitter_ISDN-Converter_Liberator_s.html
I think this might be what I need?



From shouldbeq931 at gmail.com  Fri Sep 23 01:23:08 2011
From: shouldbeq931 at gmail.com (shouldbe q931)
Date: Thu, 22 Sep 2011 22:23:08 +0100
Subject: [Freeswitch-users] Share PRI line?
In-Reply-To: <60205.67.95.76.98.1316697808.squirrel@webmail.highergear.com>
References: <56122.67.95.76.98.1316625631.squirrel@webmail.highergear.com>
	<CAP1_MOC5QOY2XQw2GgktyWH8qHDPtj5w57JQJ0=zRZftazKo0g@mail.gmail.com>
	<60205.67.95.76.98.1316697808.squirrel@webmail.highergear.com>
Message-ID: <CAP1_MOALK=VABcNqYj=wHA8vVfzFr9oWfswh5pHtXVAgoxxmLQ@mail.gmail.com>

On Thu, Sep 22, 2011 at 2:23 PM,  <fssupport at highergear.com> wrote:
>> On Wed, Sep 21, 2011 at 6:20 PM, ?<fssupport at highergear.com> wrote:
>>> We have a client with a Mitel PBX with a PRI connected to it. ?We are
>>> deploying a freeswitch box that will do some autodialing tasks for them.
>>> Is there a way to share the PRI between the two systems? ?If nothing
>>> else,
>>> I'm pretty sure I can put a two line interface card in the Freeswitch
>>> server, plug the pri into it, then connected the other interface to the
>>> Mitel box, but that would make our box a point of failure and I'd like
>>> to
>>> avoid that if possible.
>>>
>>
>> It is not possible to "split" or "share" a PRI interface with passive
>> components, your three options are
>>
>> a/ put a second PRI interface onto "your" system, and configure it to
>> route inbound calls to the Mitel system and outbound calls from the
>> Mitel system
>> b/ put a second PRI interface onto the Mitel, and configure it as per
>> a/ but with your system instead
>> c/ purchase a device with 3 PRI interfaces that can route inbound
>> calls and outbound calls appropriately as per a/ and b/
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> Thanks for your response on this. ?With regards to option c, I found this:
> http://www.patapsco.co.uk/Products/Liberator/ISDN_Splitter_ISDN-Converter_Liberator_s.html
> I think this might be what I need?
>
>
>

this might be more appropriate
http://www.patapsco.co.uk/Products/Liberator/Liberator_Q_4-Port_PRI_ISDN_Switch.html

however a 2nd PRI card in the Mitel would probably be more cost effective


From cmrienzo at gmail.com  Fri Sep 23 01:27:37 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Thu, 22 Sep 2011 17:27:37 -0400
Subject: [Freeswitch-users] Freeswitch Does Not Build
In-Reply-To: <CAF+3j-us-Oi5rDYNMvez++ASFt5BP__oaErR0bJhBnYvXkdsYA@mail.gmail.com>
References: <CAF+3j-us-Oi5rDYNMvez++ASFt5BP__oaErR0bJhBnYvXkdsYA@mail.gmail.com>
Message-ID: <CAD-qRkgPKw=4YHwUpp-1H2kcZ976Ksum81MUVYHRkANtjWP98A@mail.gmail.com>

It builds for me on CentOS 5.4 x86_64.


On Thu, Sep 22, 2011 at 4:54 PM, Antonio Teixeira
<eagle.antonio at gmail.com>wrote:

> At the time Date:   Thu Sep 22 21:11:43 2011 +0200
>
> Freeswitch fails with
> make[5]: Entering directory
> `/usr/local/src/freeswitch/src/mod/asr_tts/mod_unimrcp'
> make[6]: Entering directory `/usr/local/src/freeswitch/libs/unimrcp'
> make[6]: *** No rule to make target `build/acmacros/swift.m4', needed by
> `Makefile.in'.  Stop.
> make[6]: Leaving directory `/usr/local/src/freeswitch/libs/unimrcp'
> make[5]: ***
> [/usr/local/src/freeswitch/libs/unimrcp/platforms/libunimrcp-client/
> libunimrcpclient.la] Error 2
> make[5]: Leaving directory
> `/usr/local/src/freeswitch/src/mod/asr_tts/mod_unimrcp'
> make[4]: *** [mod_unimrcp-all] Error 1
> make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod'
> make[3]: *** [all-recursive] Error 1
> make[3]: Leaving directory `/usr/local/src/freeswitch/src'
> make[2]: *** [all-recursive] Error 1
> make[2]: Leaving directory `/usr/local/src/freeswitch'
> make[1]: *** [all] Error 2
> make[1]: Leaving directory `/usr/local/src/freeswitch'
> make: *** [current] Error 2
>
>
> Or is just me ?
>
> Regards
> A/T
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fernando.berretta at gmail.com  Fri Sep 23 01:44:54 2011
From: fernando.berretta at gmail.com (Fernando Berretta)
Date: Thu, 22 Sep 2011 18:44:54 -0300
Subject: [Freeswitch-users] G.729
In-Reply-To: <CACak_yzPA0e4kk0guxZXhrVPYt+jzqU-bT20c8bCo3X3oDVqMQ@mail.gmail.com>
References: <4E79E8B9.1040402@gmail.com>
	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>
	<4E7B1E6D.4090705@gmail.com>
	<CACak_yzPA0e4kk0guxZXhrVPYt+jzqU-bT20c8bCo3X3oDVqMQ@mail.gmail.com>
Message-ID: <4E7BAC56.1000906@gmail.com>

Muhammad,

Thanks for the link, I'm gonna analyze it.

Best Regards,
Fernando

On 9/22/2011 8:51 AM, Muhammad Naseer Bhatti wrote:
>
> Take a look here, https://github.com/Deepwalker/fs_itu_g729
>
>
> On Thu, Sep 22, 2011 at 2:39 PM, Fernando Berretta 
> <fernando.berretta at gmail.com <mailto:fernando.berretta at gmail.com>> wrote:
>
>     Kristian,
>
>     It was a typo.. the question would be.. is there some way to install
>     g729 codec in Freeswitch for free, in order to "try" .....
>
>     Regards,
>
>     On 9/21/2011 10:54 AM, Kristian Kielhofner wrote:
>     > This is a mailing list for FreeSWITCH discussion, not FreePBX.
>     >
>     > On Wed, Sep 21, 2011 at 9:38 AM, Fernando Berretta
>     > <fernando.berretta at gmail.com
>     <mailto:fernando.berretta at gmail.com>>  wrote:
>     >> Hi,
>     >>
>     >> Is there some way to install g729 codec in freepbx for free, in
>     order to
>     >> "try" it like is provided for asterisk ?
>     >>
>     >> Best Regards,
>     >> Fernando
>     >>
>     >>
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >>
>     >
>     >
>
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From fernando.berretta at gmail.com  Fri Sep 23 01:51:04 2011
From: fernando.berretta at gmail.com (Fernando Berretta)
Date: Thu, 22 Sep 2011 18:51:04 -0300
Subject: [Freeswitch-users] G.729
In-Reply-To: <CALNZuw=ZGO5WqQ3rYcHMcgnq0+5TQ=2HabHdLnAuMAMtZ6dq5A@mail.gmail.com>
References: <4E79E8B9.1040402@gmail.com>
	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>
	<4E7B1E6D.4090705@gmail.com>
	<CALNZuw=ZGO5WqQ3rYcHMcgnq0+5TQ=2HabHdLnAuMAMtZ6dq5A@mail.gmail.com>
Message-ID: <4E7BADC8.60309@gmail.com>

In order to use in a laboratory to test would be a good example

On 9/22/2011 8:54 AM, Avi Marcus wrote:
> What do you need to test? The g729 works for as many channels as you 
> pay for...
>
> -Avi
>
>
> On Thu, Sep 22, 2011 at 2:39 PM, Fernando Berretta 
> <fernando.berretta at gmail.com <mailto:fernando.berretta at gmail.com>> wrote:
>
>     Kristian,
>
>     It was a typo.. the question would be.. is there some way to install
>     g729 codec in Freeswitch for free, in order to "try" .....
>
>     Regards,
>
>     On 9/21/2011 10:54 AM, Kristian Kielhofner wrote:
>     > This is a mailing list for FreeSWITCH discussion, not FreePBX.
>     >
>     > On Wed, Sep 21, 2011 at 9:38 AM, Fernando Berretta
>     > <fernando.berretta at gmail.com
>     <mailto:fernando.berretta at gmail.com>>  wrote:
>     >> Hi,
>     >>
>     >> Is there some way to install g729 codec in freepbx for free, in
>     order to
>     >> "try" it like is provided for asterisk ?
>     >>
>     >> Best Regards,
>     >> Fernando
>     >>
>     >>
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >>
>     >
>     >
>
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From cvogel at lyonl.com  Fri Sep 23 02:16:05 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Thu, 22 Sep 2011 22:16:05 +0000
Subject: [Freeswitch-users] via header
Message-ID: <C8C19B9C-6564-478F-B4B0-A0A1D24DC28A@lyonl.com>

Hello, 

does anyone know why fs isn't setting rport in the sip via header? is there a way to set this?

Thanks

Chad


From cvogel at lyonl.com  Fri Sep 23 02:51:06 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Thu, 22 Sep 2011 22:51:06 +0000
Subject: [Freeswitch-users] via header
In-Reply-To: <C8C19B9C-6564-478F-B4B0-A0A1D24DC28A@lyonl.com>
References: <C8C19B9C-6564-478F-B4B0-A0A1D24DC28A@lyonl.com>
Message-ID: <0A42770C-89D5-4234-8FB2-2801179B03C5@lyonl.com>

or add the port to address in the via header? which is in return making our upstream provider respond on random ports

here is what fs is setting the via header to: SIP/2.0/UDP 207.67.30.226;rport;branch=z9hG4bKc25D3me5N2a3r


 
On Sep 22, 2011, at 5:16 PM, Chad Vogel wrote:

> Hello, 
> 
> does anyone know why fs isn't setting rport in the sip via header? is there a way to set this?
> 
> Thanks
> 
> Chad
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 yudha2008 at gmail.com  Fri Sep 23 03:33:29 2011
From: yudha2008 at gmail.com (baskar)
Date: Thu, 22 Sep 2011 16:33:29 -0700 (PDT)
Subject: [Freeswitch-users] Call forward
Message-ID: <1316734409939-6822324.post@n2.nabble.com>

Hi All,

I try to make call forward form once extension to another extension.
Example:
I have three extensions 4100, 4107, 4102
Call forwarded from 4107 to 4102 but it directly go to voicemail. If i
cancel call forward it directly reaches the extension.
Can any one help to resolve this issue.
I have paste my log in http://pastebin.freeswitch.org/17386.

Thanks in advance.

N.Baskar






--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Call-forward-tp6822324p6822324.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From bmoore at statirasystems.com  Fri Sep 23 04:33:40 2011
From: bmoore at statirasystems.com (William Moore)
Date: Thu, 22 Sep 2011 20:33:40 -0400
Subject: [Freeswitch-users] Error loading mod_shout
Message-ID: <4E7BD3E4.7090200@statirasystems.com>

I get the following error when I load mod_shout through the cli.

freeswitch at internal> load mod_shout
+OK Reloading XML
-ERR [module load file routine returned an error]

2011-09-23 00:31:10.654023 [INFO] switch_time.c:1028 Timezone reloaded 
530 definitions
freeswitch at internal> 2011-09-23 00:31:10.654023 [CRIT] 
switch_loadable_module.c:1271 Error Loading module 
/usr/local/freeswitch/mod/mod_shout.so
**/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: 
curl_easy_setopt**

Any ideas?

-- 
William J. Moore
Statira Systems
611 Caroline St
Fredericksburg, VA
http://www.statirasystems.com



From anthony.minessale at gmail.com  Fri Sep 23 05:04:44 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 22 Sep 2011 20:04:44 -0500
Subject: [Freeswitch-users] via header
In-Reply-To: <0A42770C-89D5-4234-8FB2-2801179B03C5@lyonl.com>
References: <C8C19B9C-6564-478F-B4B0-A0A1D24DC28A@lyonl.com>
	<0A42770C-89D5-4234-8FB2-2801179B03C5@lyonl.com>
Message-ID: <CAKbxfG87=ZR6NDka-RiSfO_75tiss3RSH60Q18UtnWRXAWMtcQ@mail.gmail.com>

As noted in your jira, the uac side sends an empty rport param to the
uas who will fill it in with the received port in the response.


On Thu, Sep 22, 2011 at 5:51 PM, Chad Vogel <cvogel at lyonl.com> wrote:
> or add the port to address in the via header? which is in return making our upstream provider respond on random ports
>
> here is what fs is setting the via header to: SIP/2.0/UDP 207.67.30.226;rport;branch=z9hG4bKc25D3me5N2a3r
>
>
>
> On Sep 22, 2011, at 5:16 PM, Chad Vogel wrote:
>
>> Hello,
>>
>> does anyone know why fs isn't setting rport in the sip via header? is there a way to set this?
>>
>> Thanks
>>
>> Chad
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ahmed.ajmal at gmail.com  Fri Sep 23 09:02:32 2011
From: ahmed.ajmal at gmail.com (Ahmed Bhaila)
Date: Fri, 23 Sep 2011 10:02:32 +0500
Subject: [Freeswitch-users] Nibble Bill Issue
Message-ID: <CAArTj4PYyvTR1Uwe+aJfbcpieQAc5waqBpjZE89r6_JF-M5Kww@mail.gmail.com>

Hi



I am trying to run nibble bill in our test environment and unable to see
calls hanging up when the balance depletes to 0. Following is the xml from
dialplan:



<context name="public">

<extension name="Internal-XXX_Mobile">

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

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

  <action application="set" data="continue_on_fail=true" inline="true"/>

 <action application="bridge"
data="[nibble_account=1,nibble_rate=1.458]sofia/gateway/NEXTONE/1291892xxxxxxxxx"/>

</condition>

</extension>



<extension name="hangup">

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

    <action application="log" data="NOTICE Hanging up"/>

    <action application="hangup"/>

  </condition>

</extension>

</context>



Now the balance for this account is already below zero and interestingly the
A leg hangs up while B leg stays connected. This is what I am seeing on my
console



Transfer sofia/external/1291892xxxxxxxxx to XML[hangup at public]



So the call does get transferred to the hangup extension but only manages to
hangup the A leg.





Thanks

Ahmed
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From viraptor at gmail.com  Fri Sep 23 01:59:53 2011
From: viraptor at gmail.com (=?ISO-8859-2?Q?Stanis=B3aw_Pitucha?=)
Date: Thu, 22 Sep 2011 22:59:53 +0100
Subject: [Freeswitch-users] callerid name on a gateway
Message-ID: <CAFrYPTOztL0+6dbStGu4FcahrjQH3RzRoq_8CVwiEtfqSGSn_g@mail.gmail.com>

Hi all,
I've got a gateway which I contact with:
caller-id-in-from = false

I've got the from-user set to the number (remote username), let's say 1234567.

Unfortunately when I send the call, the invite has header:
From: "" <1234567 at ....>

The gateway doesn't like it very much (rejects the call, complaining
about the from header) for some reason and I'd like to either fill in
the from name with the username, or remove the from name completely.
Is there some way to do this?

-- 
KTHXBYE,

Stanis?aw Pitucha


From chris at ghosttelecom.com  Fri Sep 23 11:39:30 2011
From: chris at ghosttelecom.com (Chris Martineau)
Date: Fri, 23 Sep 2011 08:39:30 +0100
Subject: [Freeswitch-users] Early media playback
In-Reply-To: <CAKDfjgdBmwtHKvziW=ruFT0r_hT9p-cAYFgKLnEe7u48i68mow@mail.gmail.com>
References: <1D10AB188D6CCA46BB4369E3268E36EF359A8F@SVR01.ghosttelecom.local><CAKDfjgdYDXpN6oLR3dy9jG2syCcdyXxNLhKmJ5--mQm73cj-rA@mail.gmail.com><1D10AB188D6CCA46BB4369E3268E36EF359AE2@SVR01.ghosttelecom.local>
	<CAKDfjgdBmwtHKvziW=ruFT0r_hT9p-cAYFgKLnEe7u48i68mow@mail.gmail.com>
Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF359B76@SVR01.ghosttelecom.local>

Hi,

Testing this a bit more and it isn't quite doing what I expect.

You were right about the inline, wasn't need and it worked mainly because putting inline on the preanswer actually prevents it from being executed as it isn't allowed.

Basically once proxy media was removed all worked as scripted.

application='set' data='execute_on_media=playback /path/file' triggers the playback on the a leg on receiving a 183 on the b leg but this was the way I did it on opensips which was a proxy whereas we are a b2b environment and this takes no account of whether there is a valid rtp path back to the a leg source.

By chance it works as there is no nat on my source but it wouldn't in a nat environment.

I tried your export nolocal version which I assume then looks for media back on the a leg from the b leg but this never seems to trigger anything.

Now I am not 100% on what on_media or on_preanswer are looking for, I assume just a 183 and 180 respectively because it doesn't seem to do anything on receiving actual rtp. If this is the case will execute on media actually trigger b leg to a leg seeing as the a leg doesn't receive a 183 (only sends it out). From looking at the log it would seem this is the case?

Here is a list of what seems to happen with different setups for the purposes of the test i use execute on media as this is looking for 183s.
1.
application='set' data='execute_on_media=playback /path/file'
application='preanswer'
Result->Plays immediately without waiting for preanswer or b leg 183?
2.
application='preanswer'
application='set' data='execute_on_media=playback /path/file'
Result->Preanswer sent to a leg but no subsequent playback heard?
3.
application='set' data='execute_on_media=playback /path/file'
Result->Plays okay on b leg 183 however does not deliver 183 to a leg until playback finished?
4.
application=' playback' data=/path/file'
application='preanswer'
Result->Plays immediately?
5.
application='preanswer'
application=' playback' data=/path/file'
Result->Plays after a leg preanswer sent however does not initiate b leg bridge until playback finished?

The one that would be perfect is number 2 with preanswer followed by execute on media which would allow time for the a leg rtp to set up whilst the call was still progressing but for some reason this just will not playback even though the log says it is?

I have got it sort of working by setting it as a sched_broadcast after 1 second which seems to allow the preanswer to be sent and the bridge to progress before playing.

Any ideas on how to make this a bit cleaner would be greatly appreciated.

Regards


Chris



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner
Sent: 21 September 2011 16:01
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Early media playback

Please try to be as specific as possible when describing issues.  As
far as proxy_media, well, this comes up all of the time:

http://wiki.freeswitch.org/wiki/Proxy_Media#Common_misconceptions_.28READ_THIS.29

I have no idea why you'd need to use inline but whatever works, I guess...

On Wed, Sep 21, 2011 at 10:30 AM, Chris Martineau
<chris at ghosttelecom.com> wrote:
> Hi,
>
> Managed to fix this by trying every possible combination starting from a
> very basic scenario.
>
> Works as follows...
>
> Execute on media <playback etc> inline
> Pre-answer inline (must be inline)
> <whatever else you want to do>
> Bridge
>
> My big mistake was having proxy media enabled. Must be full fs
> processing for early media playback to work.
> You can use proxy media but only works when set as part of the bridge
> action.
>
> I don't ignore far end 183 as I want this to kick in after my playback
> and works as I want.
>
> Only a couple of things I am still stuck on.
>
> 1. I would like to terminate the playback on answer but cannot find any
> obvious process to do this. Obviously you could do an execute_on_answer
> but can't see what I would execute to achieve the desired effect?
> Currently if the far end answers quickly they get silence until the
> playback has finished and then they get through.
>
> 2. How do you execute multiple applications on answer? I need to execute
> a sched_hangup and a sched_broadcast on answer so that the timers start
> from answer. However cannot find any combination which allows me to put
> multiple apps in the same execute on answer action. If I just create
> multiple action entries it only seems to action the last one entered?
> So have...
> application='set' data='execute_on_answer=sched_hangup +XX
> allotted_timeout'
> if I add...
> application='set' data='execute_on_answer=sched_broadcast +XX
> /path/file'
> only the broadcast triggers?
>
> Many thanks again
>
> Chris
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Kristian Kielhofner
> Sent: 21 September 2011 15:04
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Early media playback
>
> While I haven't done this before you probably need to ignore the 183
> from the remote end and set execute_on_preanswer or possibly
> execute_on_media:
>
> <action application="export"
> data="nolocal:execute_on_preanswer=playback somefile"/>
> <action application="export" data="nolocal:ignore_early_media=true"/>
> <action application="bridge" data="bridge/string"/>
>
> ?However if you're getting a codec error now you probably have some
> other issues to deal with. ?It looks like you're trying to use
> mod_native_file. ?Can you try a normal wav (using mod_sndfile) and try
> it again with the above code snippet? ?Send the FS console logs to
> http://pastebin.freeswitch.org.
>
> On Wed, Sep 21, 2011 at 5:24 AM, Chris Martineau
> <chris at ghosttelecom.com> wrote:
>> Hi
>>
>>
>>
>> Trying to play a custom announcement/welcome to the a leg on receiving
> a 183
>> from the far end.
>>
>> Have tried a straight playback just before the bridge which confirms
> that it
>> finds the playback file as the recording is heard but as this is a bit
> nasty
>> FS complains with a codec error and closes the call.
>>
>> Tried execute on media with the same playback details as follows...
>>
>> <action application="set" data="execute_on_media=playback
> /path/mediafile"
>>
>> Which according to the logs seems to work but I get just silence until
> I
>> presume the file stops playing which is strange as the file plays fine
> when
>> using a forced playback?
>>
>> Logs in both cases show the same file being found and used
> mediafile.PCMA.
>>
>>
>>
>> Also I would like to terminate the playback on answer but cannot find
> any
>> obvious process to do this. Obviously you could do an
> execute_on_answer but
>> can't see what I would execute to achieve the desired effect?
>>
>>
>>
>> Many thanks for any help you can offer.
>>
>>
>>
>> Regards
>>
>>
>>
>>
>>
>> Chris
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Kristian Kielhofner
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


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


From eagle.antonio at gmail.com  Fri Sep 23 15:52:58 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Fri, 23 Sep 2011 11:52:58 +0000
Subject: [Freeswitch-users] Freeswitch Does Not Build
In-Reply-To: <CAD-qRkgPKw=4YHwUpp-1H2kcZ976Ksum81MUVYHRkANtjWP98A@mail.gmail.com>
References: <CAF+3j-us-Oi5rDYNMvez++ASFt5BP__oaErR0bJhBnYvXkdsYA@mail.gmail.com>
	<CAD-qRkgPKw=4YHwUpp-1H2kcZ976Ksum81MUVYHRkANtjWP98A@mail.gmail.com>
Message-ID: <CAF+3j-shhJ_Tau+3VSkdPR+AfHMhC90JQ7GbWnX_VAd_rHOpVQ@mail.gmail.com>

Had to ./bootstrap && ./configure FS again :\
Strange it never happened to me , sorry to brother guys :D



2011/9/22 Christopher Rienzo <cmrienzo at gmail.com>

> It builds for me on CentOS 5.4 x86_64.
>
>
> On Thu, Sep 22, 2011 at 4:54 PM, Antonio Teixeira <eagle.antonio at gmail.com
> > wrote:
>
>> At the time Date:   Thu Sep 22 21:11:43 2011 +0200
>>
>> Freeswitch fails with
>> make[5]: Entering directory
>> `/usr/local/src/freeswitch/src/mod/asr_tts/mod_unimrcp'
>> make[6]: Entering directory `/usr/local/src/freeswitch/libs/unimrcp'
>> make[6]: *** No rule to make target `build/acmacros/swift.m4', needed by
>> `Makefile.in'.  Stop.
>> make[6]: Leaving directory `/usr/local/src/freeswitch/libs/unimrcp'
>> make[5]: ***
>> [/usr/local/src/freeswitch/libs/unimrcp/platforms/libunimrcp-client/
>> libunimrcpclient.la] Error 2
>> make[5]: Leaving directory
>> `/usr/local/src/freeswitch/src/mod/asr_tts/mod_unimrcp'
>> make[4]: *** [mod_unimrcp-all] Error 1
>> make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod'
>> make[3]: *** [all-recursive] Error 1
>> make[3]: Leaving directory `/usr/local/src/freeswitch/src'
>> make[2]: *** [all-recursive] Error 1
>> make[2]: Leaving directory `/usr/local/src/freeswitch'
>> make[1]: *** [all] Error 2
>> make[1]: Leaving directory `/usr/local/src/freeswitch'
>> make: *** [current] Error 2
>>
>>
>> Or is just me ?
>>
>> Regards
>> A/T
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From eagle.antonio at gmail.com  Fri Sep 23 15:56:59 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Fri, 23 Sep 2011 11:56:59 +0000
Subject: [Freeswitch-users] Freeswitch Does Not Build
In-Reply-To: <CAF+3j-shhJ_Tau+3VSkdPR+AfHMhC90JQ7GbWnX_VAd_rHOpVQ@mail.gmail.com>
References: <CAF+3j-us-Oi5rDYNMvez++ASFt5BP__oaErR0bJhBnYvXkdsYA@mail.gmail.com>
	<CAD-qRkgPKw=4YHwUpp-1H2kcZ976Ksum81MUVYHRkANtjWP98A@mail.gmail.com>
	<CAF+3j-shhJ_Tau+3VSkdPR+AfHMhC90JQ7GbWnX_VAd_rHOpVQ@mail.gmail.com>
Message-ID: <CAF+3j-tC7XqKQUBDATejSyqdzE7er47qYyoGMdwfOcV-q5THgA@mail.gmail.com>

Oopsy shameless kick in the dictionary.
Bother not brother ... :\

2011/9/23 Antonio Teixeira <eagle.antonio at gmail.com>

> Had to ./bootstrap && ./configure FS again :\
> Strange it never happened to me , sorry to brother guys :D
>
>
>
> 2011/9/22 Christopher Rienzo <cmrienzo at gmail.com>
>
>> It builds for me on CentOS 5.4 x86_64.
>>
>>
>> On Thu, Sep 22, 2011 at 4:54 PM, Antonio Teixeira <
>> eagle.antonio at gmail.com> wrote:
>>
>>> At the time Date:   Thu Sep 22 21:11:43 2011 +0200
>>>
>>> Freeswitch fails with
>>> make[5]: Entering directory
>>> `/usr/local/src/freeswitch/src/mod/asr_tts/mod_unimrcp'
>>> make[6]: Entering directory `/usr/local/src/freeswitch/libs/unimrcp'
>>> make[6]: *** No rule to make target `build/acmacros/swift.m4', needed by
>>> `Makefile.in'.  Stop.
>>> make[6]: Leaving directory `/usr/local/src/freeswitch/libs/unimrcp'
>>> make[5]: ***
>>> [/usr/local/src/freeswitch/libs/unimrcp/platforms/libunimrcp-client/
>>> libunimrcpclient.la] Error 2
>>> make[5]: Leaving directory
>>> `/usr/local/src/freeswitch/src/mod/asr_tts/mod_unimrcp'
>>> make[4]: *** [mod_unimrcp-all] Error 1
>>> make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod'
>>> make[3]: *** [all-recursive] Error 1
>>> make[3]: Leaving directory `/usr/local/src/freeswitch/src'
>>> make[2]: *** [all-recursive] Error 1
>>> make[2]: Leaving directory `/usr/local/src/freeswitch'
>>> make[1]: *** [all] Error 2
>>> make[1]: Leaving directory `/usr/local/src/freeswitch'
>>> make: *** [current] Error 2
>>>
>>>
>>> Or is just me ?
>>>
>>> Regards
>>> A/T
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From jcasale at activenetwerx.com  Fri Sep 23 17:15:08 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 23 Sep 2011 13:15:08 +0000
Subject: [Freeswitch-users] Web based reporting tools
Message-ID: <assp.02472e9d79.92097A6A775D5147B1078E3F15430B929E0AA7@prato.activenetwerx.local>

What are you guys using that can produce good reports for current and
accumulated statistics on an FS box?

Thanks!
jlc


From jmesquita at freeswitch.org  Fri Sep 23 19:40:58 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 23 Sep 2011 12:40:58 -0300
Subject: [Freeswitch-users] Web based reporting tools
In-Reply-To: <assp.02472e9d79.92097A6A775D5147B1078E3F15430B929E0AA7@prato.activenetwerx.local>
References: <assp.02472e9d79.92097A6A775D5147B1078E3F15430B929E0AA7@prato.activenetwerx.local>
Message-ID: <CABH8QX0jhuf5vSY=6qrs7u5O-taUAndKkVhvy2wib77b5iQ6LA@mail.gmail.com>

I personally use Munin with a couple of plugins I wrote for it to graph # of
channels, calls and some other things.

Regards,
Jo?o Mesquita



On Fri, Sep 23, 2011 at 10:15 AM, Joseph L. Casale <
jcasale at activenetwerx.com> wrote:

> What are you guys using that can produce good reports for current and
> accumulated statistics on an FS box?
>
> Thanks!
> jlc
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From robert.hadley at teotech.com  Fri Sep 23 20:26:21 2011
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 23 Sep 2011 09:26:21 -0700
Subject: [Freeswitch-users] large block of DID's
In-Reply-To: <CALNZuwnhT26xOEDAR9iGtcSicOtx6L8F1O6vzEBBCAwAX4ziig@mail.gmail.com>
References: <D25CC3E5-C0D2-41D7-87C3-843C24A0CA8C@lyonl.com>
	<801470822-1316057954-cardhu_decombobulator_blackberry.rim.net-146923873-@b5.c27.bise6.blackberry>
	<2C4ADC11-8559-45ED-94E9-435F1D60CA78@lyonl.com>
	<CALNZuwnhT26xOEDAR9iGtcSicOtx6L8F1O6vzEBBCAwAX4ziig@mail.gmail.com>
Message-ID: <E826C94A3C6BE44198FAF4D66B5AB4D8245DAAA0A6@VA3DIAXVS351.RED001.local>


I created a C didmap module I can share that uses ODBC to lookup a variable length DID number and return the user's extension, plus flags for fax server and mobile DIDs.  If you pass in the caller_id_number it also checks for blocked callers.

Robert


From: Avi Marcus [mailto:avi at avimarcus.net]
Sent: Thursday, September 22, 2011 12:54 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] large block of DID's

I'd be interested in seeing the Lua script that does the users/directory. I don't think I've seen one yet, I'm not quite sure how that would work. I've only used xml_curl for a dialplan with a web server..
-Avi


On Thu, Sep 22, 2011 at 10:00 PM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:
Are you also using a lua script via odbc to manage users?


On Sep 14, 2011, at 10:39 PM, <ljjimenez at gmail.com<mailto:ljjimenez at gmail.com>>
 <ljjimenez at gmail.com<mailto:ljjimenez at gmail.com>> wrote:

> We do this with about 10k DIDs, we use MySQL and a Lua script via ODBC
>
> Luis Jimenez
>
> -----Original Message-----
> From: Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>>
> Sender: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>
> Date: Wed, 14 Sep 2011 15:44:23
> To: FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org><FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: [Freeswitch-users] large block of DID's
>
> I was wondering if anyone has any thoughts what is the best way to manage a large blocks of DID? (we have about 1600 DID's that we are moving off of our asterisk servers)
>
> Chad
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From cvogel at lyonl.com  Fri Sep 23 23:30:30 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Fri, 23 Sep 2011 19:30:30 +0000
Subject: [Freeswitch-users] large block of DID's
In-Reply-To: <E826C94A3C6BE44198FAF4D66B5AB4D8245DAAA0A6@VA3DIAXVS351.RED001.local>
References: <D25CC3E5-C0D2-41D7-87C3-843C24A0CA8C@lyonl.com>
	<801470822-1316057954-cardhu_decombobulator_blackberry.rim.net-146923873-@b5.c27.bise6.blackberry>
	<2C4ADC11-8559-45ED-94E9-435F1D60CA78@lyonl.com>
	<CALNZuwnhT26xOEDAR9iGtcSicOtx6L8F1O6vzEBBCAwAX4ziig@mail.gmail.com>
	<E826C94A3C6BE44198FAF4D66B5AB4D8245DAAA0A6@VA3DIAXVS351.RED001.local>
Message-ID: <465713D3-D7A4-4CD5-99B8-72F583836410@lyonl.com>

I would love to see your module because I'm are thinking about writing one c or c# to do almost the same thing. are you then routing the DID to an application server to manage the users?


On Sep 23, 2011, at 11:26 AM, Robert Hadley wrote:


I created a C didmap module I can share that uses ODBC to lookup a variable length DID number and return the user's extension, plus flags for fax server and mobile DIDs.  If you pass in the caller_id_number it also checks for blocked callers.

Robert


From: Avi Marcus [mailto:avi at avimarcus.net]
Sent: Thursday, September 22, 2011 12:54 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] large block of DID's

I'd be interested in seeing the Lua script that does the users/directory. I don't think I've seen one yet, I'm not quite sure how that would work. I've only used xml_curl for a dialplan with a web server..
-Avi


On Thu, Sep 22, 2011 at 10:00 PM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:
Are you also using a lua script via odbc to manage users?


On Sep 14, 2011, at 10:39 PM, <ljjimenez at gmail.com<mailto:ljjimenez at gmail.com>>
 <ljjimenez at gmail.com<mailto:ljjimenez at gmail.com>> wrote:

> We do this with about 10k DIDs, we use MySQL and a Lua script via ODBC
>
> Luis Jimenez
>
> -----Original Message-----
> From: Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>>
> Sender: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>
> Date: Wed, 14 Sep 2011 15:44:23
> To: FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org><FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: [Freeswitch-users] large block of DID's
>
> I was wondering if anyone has any thoughts what is the best way to manage a large blocks of DID? (we have about 1600 DID's that we are moving off of our asterisk servers)
>
> Chad
>
>
> 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
>




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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
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From s.lam at keensystems.eu  Fri Sep 23 12:23:19 2011
From: s.lam at keensystems.eu (Steven Lam, KeenSystems B.V.)
Date: Fri, 23 Sep 2011 08:23:19 +0000
Subject: [Freeswitch-users] Problem compiling
Message-ID: <038DA521FD819D47845F6EA1B563BEB224981375@dc2.keensystems.eu>

Hi,

I'm having problems compiling FreeSWITCH on a Gentoo x86_64 machine.
I did a git checkout and still have the same problem, the error i see whem compiling is:

make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
Making all in .
cc1: warnings being treated as errors
libs/esl/fs_cli.c: In function 'clear_cli':
libs/esl/fs_cli.c:109: error: ignoring return value of 'write', declared with attribute warn_unused_result
libs/esl/fs_cli.c: In function 'clear_line':
libs/esl/fs_cli.c:578: error: ignoring return value of 'write', declared with attribute warn_unused_result
libs/esl/fs_cli.c:580: error: ignoring return value of 'write', declared with attribute warn_unused_result
libs/esl/fs_cli.c:581: error: ignoring return value of 'write', declared with attribute warn_unused_result
libs/esl/fs_cli.c: In function 'redisplay':
libs/esl/fs_cli.c:588: error: ignoring return value of 'write', declared with attribute warn_unused_result
libs/esl/fs_cli.c:589: error: ignoring return value of 'write', declared with attribute warn_unused_result
make[2]: *** [fs_cli-fs_cli.o] Error 1

Is there someone who can help me on this?

Thanks!

Steven

From robert.hadley at teotech.com  Sat Sep 24 00:13:59 2011
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 23 Sep 2011 13:13:59 -0700
Subject: [Freeswitch-users] large block of DID's
In-Reply-To: <465713D3-D7A4-4CD5-99B8-72F583836410@lyonl.com>
References: <D25CC3E5-C0D2-41D7-87C3-843C24A0CA8C@lyonl.com>
	<801470822-1316057954-cardhu_decombobulator_blackberry.rim.net-146923873-@b5.c27.bise6.blackberry>
	<2C4ADC11-8559-45ED-94E9-435F1D60CA78@lyonl.com>
	<CALNZuwnhT26xOEDAR9iGtcSicOtx6L8F1O6vzEBBCAwAX4ziig@mail.gmail.com>
	<E826C94A3C6BE44198FAF4D66B5AB4D8245DAAA0A6@VA3DIAXVS351.RED001.local>
	<465713D3-D7A4-4CD5-99B8-72F583836410@lyonl.com>
Message-ID: <E826C94A3C6BE44198FAF4D66B5AB4D8245DAAA191@VA3DIAXVS351.RED001.local>


No, the custom didmap module doesn't manage users, another app does.  I call the didmap app from public.xml and transfer to whatever destination it returns in defaults.xml, e.g. a local_extension, faxserver, or caller_blocked.

    <extension name="didmap_lookup_transfer" continue="true">
      <condition field="destination_number" expression="^(\d+)$">
        <action application="set" data="did_number=$1"/>
        <action application="didmap" data="${did_number} ${caller_id_number}"/>
        <action application="transfer" data="${didmaplookup_dest} XML default"/>
      </condition>
    </extension>

DID map DB table:
CREATE TABLE inbounddidnumbermap
(
  id bigserial NOT NULL,
  phonenumber character varying(255),
  fax boolean NOT NULL DEFAULT false,
  mobile boolean NOT NULL DEFAULT false,
  extension_id bigint,
  CONSTRAINT inbounddidnumbermap_pkey PRIMARY KEY (id),
)


From: Chad Vogel [mailto:cvogel at lyonl.com]
Sent: Friday, September 23, 2011 12:31 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] large block of DID's

I would love to see your module because I'm are thinking about writing one c or c# to do almost the same thing. are you then routing the DID to an application server to manage the users?


On Sep 23, 2011, at 11:26 AM, Robert Hadley wrote:



I created a C didmap module I can share that uses ODBC to lookup a variable length DID number and return the user's extension, plus flags for fax server and mobile DIDs.  If you pass in the caller_id_number it also checks for blocked callers.

Robert


From: Avi Marcus [mailto:avi at avimarcus.net]<mailto:[mailto:avi at avimarcus.net]>
Sent: Thursday, September 22, 2011 12:54 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] large block of DID's

I'd be interested in seeing the Lua script that does the users/directory. I don't think I've seen one yet, I'm not quite sure how that would work. I've only used xml_curl for a dialplan with a web server..
-Avi



On Thu, Sep 22, 2011 at 10:00 PM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:
Are you also using a lua script via odbc to manage users?


On Sep 14, 2011, at 10:39 PM, <ljjimenez at gmail.com<mailto:ljjimenez at gmail.com>>
 <ljjimenez at gmail.com<mailto:ljjimenez at gmail.com>> wrote:

> We do this with about 10k DIDs, we use MySQL and a Lua script via ODBC
>
> Luis Jimenez
>
> -----Original Message-----
> From: Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>>
> Sender: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>
> Date: Wed, 14 Sep 2011 15:44:23
> To: FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org><FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: [Freeswitch-users] large block of DID's
>
> I was wondering if anyone has any thoughts what is the best way to manage a large blocks of DID? (we have about 1600 DID's that we are moving off of our asterisk servers)
>
> Chad
>
>
> 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
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> http://www.freeswitch.org
>




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From covici at ccs.covici.com  Sat Sep 24 00:50:11 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 23 Sep 2011 16:50:11 -0400
Subject: [Freeswitch-users] Problem compiling
In-Reply-To: <038DA521FD819D47845F6EA1B563BEB224981375@dc2.keensystems.eu>
References: <038DA521FD819D47845F6EA1B563BEB224981375@dc2.keensystems.eu>
Message-ID: <32340.1316811011@ccs.covici.com>

I am not seeing this here -- I am running gentoo unstable amd64.  You
might try running bootstrap -- save your modules.conf somewhere and put
it back and try again.

Steven Lam, KeenSystems B.V. <s.lam at keensystems.eu> wrote:

> Hi,
> 
> I'm having problems compiling FreeSWITCH on a Gentoo x86_64 machine.
> I did a git checkout and still have the same problem, the error i see whem compiling is:
> 
> make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /source/fre!
 es!
>  witch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
> Making all in .
> cc1: warnings being treated as errors
> libs/esl/fs_cli.c: In function 'clear_cli':
> libs/esl/fs_cli.c:109: error: ignoring return value of 'write', declared with attribute warn_unused_result
> libs/esl/fs_cli.c: In function 'clear_line':
> libs/esl/fs_cli.c:578: error: ignoring return value of 'write', declared with attribute warn_unused_result
> libs/esl/fs_cli.c:580: error: ignoring return value of 'write', declared with attribute warn_unused_result
> libs/esl/fs_cli.c:581: error: ignoring return value of 'write', declared with attribute warn_unused_result
> libs/esl/fs_cli.c: In function 'redisplay':
> libs/esl/fs_cli.c:588: error: ignoring return value of 'write', declared with attribute warn_unused_result
> libs/esl/fs_cli.c:589: error: ignoring return value of 'write', declared with attribute warn_unused_result
> make[2]: *** [fs_cli-fs_cli.o] Error 1
> 
> Is there someone who can help me on this?
> 
> Thanks!
> 
> Steven
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.villasmil.work at gmail.com  Sat Sep 24 04:03:36 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 24 Sep 2011 02:03:36 +0200
Subject: [Freeswitch-users] Problem with Freeswitch/Sangoma D100.
In-Reply-To: <aa54cbcc6870ad57562eae38969f10a9@mail.gmail.com>
References: <AcxIgTD+GPg2tEvuR8+l9k98C+DfAg==>
	<aa54cbcc6870ad57562eae38969f10a9@mail.gmail.com>
Message-ID: <CAFGRPVp5y=_AOSSEuzy3Cv6Wd9SqGdsUDSoy2GTKT+3EKsY-sA@mail.gmail.com>

Hello Ricardo,

Did you figure out what your problem was?

Thanks

David

On Fri, Jul 22, 2011 at 5:56 PM, Ricardo Martinez <rmartinez at redvoiss.net>wrote:

> Hello list.
>
> I have the next problem with my Freeswitch.  I?m using a Sangoma D100
> transcoding card, but ?im having problems with G729.  This is my scenario.
>
>
>
> GatewayA ---------> FreeSwitch+D100 -----------> GatewayB
>
> (10.0.0.220)                    (10.0.0.148)
>       (10.0.0.222)
>
>
>
> Gateway A is calling through Freeswitch to Gateway B.  The problem is with
> the ?200 - OK? message that goes from FreeSwitch to Gateway A.  The ?200 -
> OK?  coming from Gateway B has the parameter ?a=fmtp:18 annexb=no? in the
> SDP, but Freeswitch is not attaching this parameter to the ?Leg A? and this
> is causing a one-way audio in the call.  (please see the attached
> sdp-problem.jpg file).
>
> The weird thing is when i unload the ?mod_sangoma_codec? and load the
> ?mod_g729?, this time the ?200 OK? message from the Freeswitch is using the
> ?a=fmtp:18 annexb=no? to the Leg A, and the call is successfully established
> without one way audio problem.
>
> I really don?t know what could be happening, I ask the Sangoma support but
> thay said that this is a Freeswitch bug.
>
> Can someone help me here?
>
>
>
> These are part of my configuration files.
>
>
>
> ?default.xml?
>
>
>
>     <extension name="to_prueba2">
>
>       <condition field="network_addr" expression="^10\.0\.0\.220$">
>
>                   <action application="set" data="call_timeout=50"/>
>
>                   <action application="set"
> data="hangup_after_bridge=true"/>
>
>                   <action application="export"
> data="nolocal:absolute_codec_string=G729,G723"/>
>
>                   <action application="set"
> data="sip_invite_domain=10.0.0.222"/>
>
>                   <action application="export"
> data="sip_append_audio_sdp=a=rtpmap:18 G729/8000,a=fmtp:18 annexb=no"/>
>
>                   <action application="bridge" data="
> sofia/$${domain}/${destination_number}@10.0.0.222"/>
>
>                   <action application="answer"/>
>
>       </condition>
>
>     </extension>
>
>
>
>     <extension name="to_prueba1">
>
>       <condition field="network_addr" expression="^10\.0\.0\.222$">
>
>                   <action application="set" data="call_timeout=50"/>
>
>                   <action application="set"
> data="hangup_after_bridge=true"/>
>
>                   <action application="set"
> data="sip_invite_domain=10.0.0.220"/>
>
>                   <action application="bridge" data="
> sofia/$${domain}/${destination_number}@10.0.0.220"/>
>
>                   <action application="answer"/>
>
>       </condition>
>
>     </extension>
>
>
>
>
>
> Part of the  ?interior.xml? file:
>
>     <!--Uncomment to let calls hit the dialplan *before* you decide if the
> codec is ok-->
>
>     <param name="inbound-late-negotiation" value="true"/>
>
>     <!--set to 'greedy' if you want your codec list to take precedence -->
>
>     <param name="inbound-codec-negotiation" value="generous"/>
>
>
>
> Y finalmente en los codec preferentes tengo (vars.xml)
>
>   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G723,PCMU,PCMA"/>
>
>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,G723"/>
>
>
>
>
>
>
>
> Thanks in advance.
>
> Regards,
>
> Ricardo.-
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cvogel at lyonl.com  Sat Sep 24 06:09:34 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Sat, 24 Sep 2011 02:09:34 +0000
Subject: [Freeswitch-users] Normalization Rules
Message-ID: <1999F1B2-416E-4F05-84E8-943DD9A3A818@lyonl.com>

What is the best way to do number normalization in a dial plan to reformat call to e164 before processing? Is this the best way of doing it or is there a better way where I wouldn't need to loopback?

    <extension name="CleanUp">
      <condition field="destination_number" expression="^(44|380)?(\d)$">
        <action application="bridge" data="loopback/+$1"/>
      </condition>
    </extension>

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From bnaylor at sirran.com  Sat Sep 24 16:14:49 2011
From: bnaylor at sirran.com (Ben Naylor)
Date: Sat, 24 Sep 2011 13:14:49 +0100
Subject: [Freeswitch-users] TLS issue
Message-ID: <001e01cc7ab3$8f6f4330$ae4dc990$@sirran.com>

Hello Freeswitch community!

 

I was wondering if someone could help me with an issue I am experiencing
when using TLS to secure SIP traffic?

Our current test environment consists of a Freeswitch test server, a
wireless access point, and 2 x iPhones using SIP client, all connected to
the same closed off VLAN.

 

The Freeswitch server is set to bypass media=true, so that the RTP traffic
flows directly between the iPhones.  When testing a call using standard
SIP/RTP, the call is successful, and terminates correctly when one of the
devices hangs up the call.

 

The problem occurs when using TLS, and trying to terminate the call.  The
call establishes successfully, but when either of the handsets hangs up, the
other remains connected to the call.  This happens when hanging up the call
from both the calling and the called handset.

 

One thing I noticed when looking at the console output, is that two messages
appear along the lines of 'unknown Sip packet' (apologies for the vagueness,
I am not currently in front of the test server).  These appear when the call
is terminated on one of the hand sets.

 

Has anyone had a similar experience when using TLS in this type of scenario?


 

Thanks for the help!

 

Regards

 

Ben

 

 

 

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From avi at avimarcus.net  Sat Sep 24 20:49:53 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 24 Sep 2011 19:49:53 +0300
Subject: [Freeswitch-users] Normalization Rules
In-Reply-To: <1999F1B2-416E-4F05-84E8-943DD9A3A818@lyonl.com>
References: <1999F1B2-416E-4F05-84E8-943DD9A3A818@lyonl.com>
Message-ID: <CALNZuwnjQoQ+tdEntXnJ-TZ=j_K-FW3i3VDG1RZ19VFqUXhyRA@mail.gmail.com>

You can transfer to the new number.
That means it starts the dialplan again, so put this cleanup at the top of
your dialplan to minimize wasted routing.
-Avi



On Sat, Sep 24, 2011 at 5:09 AM, Chad Vogel <cvogel at lyonl.com> wrote:

>  What is the best way to do number normalization in a dial plan to reformat
> call to e164 before processing? Is this the best way of doing it or is there
> a better way where I wouldn't need to loopback?
>
>       <extension name="CleanUp">
>        <condition field="destination_number" expression="^(44|380)?(\d)$">
>          <action application="bridge" data="loopback/+$1"/>
>        </condition>
>      </extension>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cvogel at lyonl.com  Sat Sep 24 22:12:24 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Sat, 24 Sep 2011 18:12:24 +0000
Subject: [Freeswitch-users] Normalization Rules
In-Reply-To: <CALNZuwnjQoQ+tdEntXnJ-TZ=j_K-FW3i3VDG1RZ19VFqUXhyRA@mail.gmail.com>
References: <1999F1B2-416E-4F05-84E8-943DD9A3A818@lyonl.com>
	<CALNZuwnjQoQ+tdEntXnJ-TZ=j_K-FW3i3VDG1RZ19VFqUXhyRA@mail.gmail.com>
Message-ID: <9C777FEB-B9EA-414D-A936-2FC761EF3279@lyonl.com>

so how would rewrite the destination_number?

On Sep 24, 2011, at 11:49 AM, Avi Marcus wrote:

You can transfer to the new number.
That means it starts the dialplan again, so put this cleanup at the top of your dialplan to minimize wasted routing.
-Avi



On Sat, Sep 24, 2011 at 5:09 AM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:
What is the best way to do number normalization in a dial plan to reformat call to e164 before processing? Is this the best way of doing it or is there a better way where I wouldn't need to loopback?

    <extension name="CleanUp">
      <condition field="destination_number" expression="^(44|380)?(\d)$">
        <action application="bridge" data="loopback/+$1"/>
      </condition>
    </extension>



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
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From avi at avimarcus.net  Sat Sep 24 22:17:06 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 24 Sep 2011 21:17:06 +0300
Subject: [Freeswitch-users] Normalization Rules
In-Reply-To: <9C777FEB-B9EA-414D-A936-2FC761EF3279@lyonl.com>
References: <1999F1B2-416E-4F05-84E8-943DD9A3A818@lyonl.com>
	<CALNZuwnjQoQ+tdEntXnJ-TZ=j_K-FW3i3VDG1RZ19VFqUXhyRA@mail.gmail.com>
	<9C777FEB-B9EA-414D-A936-2FC761EF3279@lyonl.com>
Message-ID: <CALNZuwnPNUQ+DYvWP3u1RKBdXYomBY_Ndery4EURX_5jiBcpqA@mail.gmail.com>

Transfer to it, rather than bridge to loopback.
-Avi



On Sat, Sep 24, 2011 at 9:12 PM, Chad Vogel <cvogel at lyonl.com> wrote:

>  so how would rewrite the destination_number?
>
>  On Sep 24, 2011, at 11:49 AM, Avi Marcus wrote:
>
>  You can transfer to the new number.
> That means it starts the dialplan again, so put this cleanup at the top of
> your dialplan to minimize wasted routing.
> -Avi
>
>
>
> On Sat, Sep 24, 2011 at 5:09 AM, Chad Vogel <cvogel at lyonl.com> wrote:
>
>> What is the best way to do number normalization in a dial plan to reformat
>> call to e164 before processing? Is this the best way of doing it or is there
>> a better way where I wouldn't need to loopback?
>>
>>       <extension name="CleanUp">
>>        <condition field="destination_number" expression="^(44|380)?(\d)$"
>> >
>>          <action application="bridge" data="loopback/+$1"/>
>>        </condition>
>>      </extension>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Sat Sep 24 22:55:22 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 24 Sep 2011 13:55:22 -0500
Subject: [Freeswitch-users] Early media playback
In-Reply-To: <1D10AB188D6CCA46BB4369E3268E36EF359B76@SVR01.ghosttelecom.local>
References: <1D10AB188D6CCA46BB4369E3268E36EF359A8F@SVR01.ghosttelecom.local>
	<CAKDfjgdYDXpN6oLR3dy9jG2syCcdyXxNLhKmJ5--mQm73cj-rA@mail.gmail.com>
	<1D10AB188D6CCA46BB4369E3268E36EF359AE2@SVR01.ghosttelecom.local>
	<CAKDfjgdBmwtHKvziW=ruFT0r_hT9p-cAYFgKLnEe7u48i68mow@mail.gmail.com>
	<1D10AB188D6CCA46BB4369E3268E36EF359B76@SVR01.ghosttelecom.local>
Message-ID: <CAKbxfG-rwCoyfQoKL74Rq18U0qmEDGh_kKXPrjK+fNHYMgK2oQ@mail.gmail.com>

here are your 2 options.

(beware that the file will loop if it finishes)

<action application="set" data="ringback=/path/to/file.wav"/>
<action application="bridge"
data="{ignore_early_media=true}sofia/internal/foo at bar.com"/>


<action application="set"
data="execute_on_pre_answer=playback::/path/to/file.wav"/>
<action application="bridge" data="sofia/internal/foo at bar.com"/>

You can do neither in conjunction with proxy_media mode because by
design, FreeSWITCH stays out of the media negotiation and has no idea
how to stream audio to the endpoints when you choose this setting.

There is no benefit at all to proxy_media mode unless you are trying
to use codecs that FreeSWITCH does not understand.



On Fri, Sep 23, 2011 at 2:39 AM, Chris Martineau <chris at ghosttelecom.com> wrote:
> Hi,
>
> Testing this a bit more and it isn't quite doing what I expect.
>
> You were right about the inline, wasn't need and it worked mainly because putting inline on the preanswer actually prevents it from being executed as it isn't allowed.
>
> Basically once proxy media was removed all worked as scripted.
>
> application='set' data='execute_on_media=playback /path/file' triggers the playback on the a leg on receiving a 183 on the b leg but this was the way I did it on opensips which was a proxy whereas we are a b2b environment and this takes no account of whether there is a valid rtp path back to the a leg source.
>
> By chance it works as there is no nat on my source but it wouldn't in a nat environment.
>
> I tried your export nolocal version which I assume then looks for media back on the a leg from the b leg but this never seems to trigger anything.
>
> Now I am not 100% on what on_media or on_preanswer are looking for, I assume just a 183 and 180 respectively because it doesn't seem to do anything on receiving actual rtp. If this is the case will execute on media actually trigger b leg to a leg seeing as the a leg doesn't receive a 183 (only sends it out). From looking at the log it would seem this is the case?
>
> Here is a list of what seems to happen with different setups for the purposes of the test i use execute on media as this is looking for 183s.
> 1.
> application='set' data='execute_on_media=playback /path/file'
> application='preanswer'
> Result->Plays immediately without waiting for preanswer or b leg 183?
> 2.
> application='preanswer'
> application='set' data='execute_on_media=playback /path/file'
> Result->Preanswer sent to a leg but no subsequent playback heard?
> 3.
> application='set' data='execute_on_media=playback /path/file'
> Result->Plays okay on b leg 183 however does not deliver 183 to a leg until playback finished?
> 4.
> application=' playback' data=/path/file'
> application='preanswer'
> Result->Plays immediately?
> 5.
> application='preanswer'
> application=' playback' data=/path/file'
> Result->Plays after a leg preanswer sent however does not initiate b leg bridge until playback finished?
>
> The one that would be perfect is number 2 with preanswer followed by execute on media which would allow time for the a leg rtp to set up whilst the call was still progressing but for some reason this just will not playback even though the log says it is?
>
> I have got it sort of working by setting it as a sched_broadcast after 1 second which seems to allow the preanswer to be sent and the bridge to progress before playing.
>
> Any ideas on how to make this a bit cleaner would be greatly appreciated.
>
> Regards
>
>
> Chris
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner
> Sent: 21 September 2011 16:01
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Early media playback
>
> Please try to be as specific as possible when describing issues. ?As
> far as proxy_media, well, this comes up all of the time:
>
> http://wiki.freeswitch.org/wiki/Proxy_Media#Common_misconceptions_.28READ_THIS.29
>
> I have no idea why you'd need to use inline but whatever works, I guess...
>
> On Wed, Sep 21, 2011 at 10:30 AM, Chris Martineau
> <chris at ghosttelecom.com> wrote:
>> Hi,
>>
>> Managed to fix this by trying every possible combination starting from a
>> very basic scenario.
>>
>> Works as follows...
>>
>> Execute on media <playback etc> inline
>> Pre-answer inline (must be inline)
>> <whatever else you want to do>
>> Bridge
>>
>> My big mistake was having proxy media enabled. Must be full fs
>> processing for early media playback to work.
>> You can use proxy media but only works when set as part of the bridge
>> action.
>>
>> I don't ignore far end 183 as I want this to kick in after my playback
>> and works as I want.
>>
>> Only a couple of things I am still stuck on.
>>
>> 1. I would like to terminate the playback on answer but cannot find any
>> obvious process to do this. Obviously you could do an execute_on_answer
>> but can't see what I would execute to achieve the desired effect?
>> Currently if the far end answers quickly they get silence until the
>> playback has finished and then they get through.
>>
>> 2. How do you execute multiple applications on answer? I need to execute
>> a sched_hangup and a sched_broadcast on answer so that the timers start
>> from answer. However cannot find any combination which allows me to put
>> multiple apps in the same execute on answer action. If I just create
>> multiple action entries it only seems to action the last one entered?
>> So have...
>> application='set' data='execute_on_answer=sched_hangup +XX
>> allotted_timeout'
>> if I add...
>> application='set' data='execute_on_answer=sched_broadcast +XX
>> /path/file'
>> only the broadcast triggers?
>>
>> Many thanks again
>>
>> Chris
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Kristian Kielhofner
>> Sent: 21 September 2011 15:04
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Early media playback
>>
>> While I haven't done this before you probably need to ignore the 183
>> from the remote end and set execute_on_preanswer or possibly
>> execute_on_media:
>>
>> <action application="export"
>> data="nolocal:execute_on_preanswer=playback somefile"/>
>> <action application="export" data="nolocal:ignore_early_media=true"/>
>> <action application="bridge" data="bridge/string"/>
>>
>> ?However if you're getting a codec error now you probably have some
>> other issues to deal with. ?It looks like you're trying to use
>> mod_native_file. ?Can you try a normal wav (using mod_sndfile) and try
>> it again with the above code snippet? ?Send the FS console logs to
>> http://pastebin.freeswitch.org.
>>
>> On Wed, Sep 21, 2011 at 5:24 AM, Chris Martineau
>> <chris at ghosttelecom.com> wrote:
>>> Hi
>>>
>>>
>>>
>>> Trying to play a custom announcement/welcome to the a leg on receiving
>> a 183
>>> from the far end.
>>>
>>> Have tried a straight playback just before the bridge which confirms
>> that it
>>> finds the playback file as the recording is heard but as this is a bit
>> nasty
>>> FS complains with a codec error and closes the call.
>>>
>>> Tried execute on media with the same playback details as follows...
>>>
>>> <action application="set" data="execute_on_media=playback
>> /path/mediafile"
>>>
>>> Which according to the logs seems to work but I get just silence until
>> I
>>> presume the file stops playing which is strange as the file plays fine
>> when
>>> using a forced playback?
>>>
>>> Logs in both cases show the same file being found and used
>> mediafile.PCMA.
>>>
>>>
>>>
>>> Also I would like to terminate the playback on answer but cannot find
>> any
>>> obvious process to do this. Obviously you could do an
>> execute_on_answer but
>>> can't see what I would execute to achieve the desired effect?
>>>
>>>
>>>
>>> Many thanks for any help you can offer.
>>>
>>>
>>>
>>> Regards
>>>
>>>
>>>
>>>
>>>
>>> Chris
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Kristian Kielhofner
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Kristian Kielhofner
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From anthony.minessale at gmail.com  Sat Sep 24 22:57:33 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 24 Sep 2011 13:57:33 -0500
Subject: [Freeswitch-users] Problem compiling
In-Reply-To: <32340.1316811011@ccs.covici.com>
References: <038DA521FD819D47845F6EA1B563BEB224981375@dc2.keensystems.eu>
	<32340.1316811011@ccs.covici.com>
Message-ID: <CAKbxfG-GKDBTaQ03dLHUjZCpxW29QjDPzwvshp+P1NAMK9my-g@mail.gmail.com>

There were a round of patches to make FS build on newer compilers, are
you running an older checkout ?


On Fri, Sep 23, 2011 at 3:50 PM,  <covici at ccs.covici.com> wrote:
> I am not seeing this here -- I am running gentoo unstable amd64. ?You
> might try running bootstrap -- save your modules.conf somewhere and put
> it back and try again.
>
> Steven Lam, KeenSystems B.V. <s.lam at keensystems.eu> wrote:
>
>> Hi,
>>
>> I'm having problems compiling FreeSWITCH on a Gentoo x86_64 machine.
>> I did a git checkout and still have the same problem, the error i see whem compiling is:
>>
>> make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /source/fre!
> ?es!
>> ?witch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; ?mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
>> Making all in .
>> cc1: warnings being treated as errors
>> libs/esl/fs_cli.c: In function 'clear_cli':
>> libs/esl/fs_cli.c:109: error: ignoring return value of 'write', declared with attribute warn_unused_result
>> libs/esl/fs_cli.c: In function 'clear_line':
>> libs/esl/fs_cli.c:578: error: ignoring return value of 'write', declared with attribute warn_unused_result
>> libs/esl/fs_cli.c:580: error: ignoring return value of 'write', declared with attribute warn_unused_result
>> libs/esl/fs_cli.c:581: error: ignoring return value of 'write', declared with attribute warn_unused_result
>> libs/esl/fs_cli.c: In function 'redisplay':
>> libs/esl/fs_cli.c:588: error: ignoring return value of 'write', declared with attribute warn_unused_result
>> libs/esl/fs_cli.c:589: error: ignoring return value of 'write', declared with attribute warn_unused_result
>> make[2]: *** [fs_cli-fs_cli.o] Error 1
>>
>> Is there someone who can help me on this?
>>
>> Thanks!
>>
>> Steven
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>



-- 
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  Sat Sep 24 22:58:45 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 24 Sep 2011 13:58:45 -0500
Subject: [Freeswitch-users] TLS issue
In-Reply-To: <001e01cc7ab3$8f6f4330$ae4dc990$@sirran.com>
References: <001e01cc7ab3$8f6f4330$ae4dc990$@sirran.com>
Message-ID: <CAKbxfG_mRH=7Or8=bRuixyYhDVHGiAGo37ovmShB4nocVoyrbg@mail.gmail.com>

you could try using the command

sofia global siptrace on

to see what you are getting.

On Sat, Sep 24, 2011 at 7:14 AM, Ben Naylor <bnaylor at sirran.com> wrote:
> Hello Freeswitch community!
>
>
>
> I was wondering if someone could help me with an issue I am experiencing
> when using TLS to secure SIP traffic?
>
> Our current test environment consists of a Freeswitch test server, a
> wireless access point, and 2 x iPhones using SIP client, all connected to
> the same closed off VLAN.
>
>
>
> The Freeswitch server is set to bypass media=true, so that the RTP traffic
> flows directly between the iPhones.? When testing a call using standard
> SIP/RTP, the call is successful, and terminates correctly when one of the
> devices hangs up the call.
>
>
>
> The problem occurs when using TLS, and trying to terminate the call.? The
> call establishes successfully, but when either of the handsets hangs up, the
> other remains connected to the call.? This happens when hanging up the call
> from both the calling and the called handset.
>
>
>
> One thing I noticed when looking at the console output, is that two messages
> appear along the lines of ?unknown Sip packet? (apologies for the vagueness,
> I am not currently in front of the test server).? These appear when the call
> is terminated on one of the hand sets.
>
>
>
> Has anyone had a similar experience when using TLS in this type of
> scenario?
>
>
>
> Thanks for the help!
>
>
>
> Regards
>
>
>
> Ben
>
>
>
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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FreeSWITCH Developer Conference
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From anthony.minessale at gmail.com  Sat Sep 24 23:10:40 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 24 Sep 2011 14:10:40 -0500
Subject: [Freeswitch-users] G.729
In-Reply-To: <4E7BADC8.60309@gmail.com>
References: <4E79E8B9.1040402@gmail.com>
	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>
	<4E7B1E6D.4090705@gmail.com>
	<CALNZuw=ZGO5WqQ3rYcHMcgnq0+5TQ=2HabHdLnAuMAMtZ6dq5A@mail.gmail.com>
	<4E7BADC8.60309@gmail.com>
Message-ID: <CAKbxfG-Rg1MqJ48LfHQWmTbC-5dPU4Vwb3kFwFtbXQP7fmVWwg@mail.gmail.com>

This unsupported code is not the same code you would get if you
purchase the official G729 module.  It's a very slow, poorly written
reference implementation just designed to prove the process that is
under patent.

It will NOT do you any good in evaluating our commercial product, you
know, the one we worked on for years to roll out in an effort to
support our community.

Every time you make a call with that unauthorized codec or suggest it
to someone on our mailing list, a little kitten somewhere falls in a
snake pit..........



On Thu, Sep 22, 2011 at 4:51 PM, Fernando Berretta
<fernando.berretta at gmail.com> wrote:
> In order to use in a laboratory to test would be a good example
>
> On 9/22/2011 8:54 AM, Avi Marcus wrote:
>
> What do you need to test? The g729 works for as many channels as you pay
> for...
> -Avi
>
> On Thu, Sep 22, 2011 at 2:39 PM, Fernando Berretta
> <fernando.berretta at gmail.com> wrote:
>>
>> Kristian,
>>
>> It was a typo.. the question would be.. is there some way to install
>> g729 codec in Freeswitch for free, in order to "try" .....
>>
>> Regards,
>>
>> On 9/21/2011 10:54 AM, Kristian Kielhofner wrote:
>> > This is a mailing list for FreeSWITCH discussion, not FreePBX.
>> >
>> > On Wed, Sep 21, 2011 at 9:38 AM, Fernando Berretta
>> > <fernando.berretta at gmail.com> ?wrote:
>> >> Hi,
>> >>
>> >> Is there some way to install g729 codec in freepbx for free, in order
>> >> to
>> >> "try" it like is provided for asterisk ?
>> >>
>> >> Best Regards,
>> >> Fernando
>> >>
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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FreeSWITCH Developer Conference
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From curriegrad2004 at gmail.com  Sun Sep 25 00:27:13 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 24 Sep 2011 13:27:13 -0700
Subject: [Freeswitch-users] G.729
In-Reply-To: <CAKbxfG-Rg1MqJ48LfHQWmTbC-5dPU4Vwb3kFwFtbXQP7fmVWwg@mail.gmail.com>
References: <4E79E8B9.1040402@gmail.com>
	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>
	<4E7B1E6D.4090705@gmail.com>
	<CALNZuw=ZGO5WqQ3rYcHMcgnq0+5TQ=2HabHdLnAuMAMtZ6dq5A@mail.gmail.com>
	<4E7BADC8.60309@gmail.com>
	<CAKbxfG-Rg1MqJ48LfHQWmTbC-5dPU4Vwb3kFwFtbXQP7fmVWwg@mail.gmail.com>
Message-ID: <CAMWiMy9aS4UVn2VYf0YqJnSSbjrZLZC55nj7yKUHYdVS7BD=fA@mail.gmail.com>

There's always the hardware PCMU-G729 transcoding cards out there, but
they don't come cheap. Those perform much better than the commercial
FS G729, ITU G.729 and any other G729 implementations.

On Sat, Sep 24, 2011 at 12:10 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> This unsupported code is not the same code you would get if you
> purchase the official G729 module. ?It's a very slow, poorly written
> reference implementation just designed to prove the process that is
> under patent.
>
> It will NOT do you any good in evaluating our commercial product, you
> know, the one we worked on for years to roll out in an effort to
> support our community.
>
> Every time you make a call with that unauthorized codec or suggest it
> to someone on our mailing list, a little kitten somewhere falls in a
> snake pit..........
>
>
>
> On Thu, Sep 22, 2011 at 4:51 PM, Fernando Berretta
> <fernando.berretta at gmail.com> wrote:
>> In order to use in a laboratory to test would be a good example
>>
>> On 9/22/2011 8:54 AM, Avi Marcus wrote:
>>
>> What do you need to test? The g729 works for as many channels as you pay
>> for...
>> -Avi
>>
>> On Thu, Sep 22, 2011 at 2:39 PM, Fernando Berretta
>> <fernando.berretta at gmail.com> wrote:
>>>
>>> Kristian,
>>>
>>> It was a typo.. the question would be.. is there some way to install
>>> g729 codec in Freeswitch for free, in order to "try" .....
>>>
>>> Regards,
>>>
>>> On 9/21/2011 10:54 AM, Kristian Kielhofner wrote:
>>> > This is a mailing list for FreeSWITCH discussion, not FreePBX.
>>> >
>>> > On Wed, Sep 21, 2011 at 9:38 AM, Fernando Berretta
>>> > <fernando.berretta at gmail.com> ?wrote:
>>> >> Hi,
>>> >>
>>> >> Is there some way to install g729 codec in freepbx for free, in order
>>> >> to
>>> >> "try" it like is provided for asterisk ?
>>> >>
>>> >> Best Regards,
>>> >> Fernando
>>> >>
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 brad at tech21.com  Sun Sep 25 10:28:16 2011
From: brad at tech21.com (Brad Mina)
Date: Sat, 24 Sep 2011 23:28:16 -0700
Subject: [Freeswitch-users] large block of DID's
In-Reply-To: <E826C94A3C6BE44198FAF4D66B5AB4D8245DAAA191@VA3DIAXVS351.RED001.local>
References: <D25CC3E5-C0D2-41D7-87C3-843C24A0CA8C@lyonl.com>
	<801470822-1316057954-cardhu_decombobulator_blackberry.rim.net-146923873-@b5.c27.bise6.blackberry>
	<2C4ADC11-8559-45ED-94E9-435F1D60CA78@lyonl.com>
	<CALNZuwnhT26xOEDAR9iGtcSicOtx6L8F1O6vzEBBCAwAX4ziig@mail.gmail.com>
	<E826C94A3C6BE44198FAF4D66B5AB4D8245DAAA0A6@VA3DIAXVS351.RED001.local>
	<465713D3-D7A4-4CD5-99B8-72F583836410@lyonl.com>
	<E826C94A3C6BE44198FAF4D66B5AB4D8245DAAA191@VA3DIAXVS351.RED001.local>
Message-ID: <CAGRr05-3GjyrEU+74oQbprayxo+K2eKPzrkpDgqaSFtRZqMp9A@mail.gmail.com>

Robert,

Is your module in the git contrib? Is there a public place we can download
it?

On Fri, Sep 23, 2011 at 1:13 PM, Robert Hadley <robert.hadley at teotech.com>wrote:

> ** **
>
> No, the custom didmap module doesn't manage users, another app does.  I
> call the didmap app from public.xml and transfer to whatever destination it
> returns in defaults.xml, e.g. a local_extension, faxserver, or
> caller_blocked.****
>
> ** **
>
>     <extension name="didmap_lookup_transfer" continue="true">****
>
>       <condition field="destination_number" expression="^(\d+)$">****
>
>         <action application="set" data="did_number=$1"/>****
>
>         <action application="didmap" data="${did_number}
> ${caller_id_number}"/>****
>
>         <action application="transfer" data="${didmaplookup_dest} XML
> default"/>****
>
>       </condition>****
>
>     </extension>****
>
> ** **
>
> DID map DB table:****
>
> CREATE TABLE inbounddidnumbermap****
>
> (****
>
>   id bigserial NOT NULL,****
>
>   phonenumber character varying(255),****
>
>   fax boolean NOT NULL DEFAULT false,****
>
>   mobile boolean NOT NULL DEFAULT false,****
>
>   extension_id bigint,****
>
>   CONSTRAINT inbounddidnumbermap_pkey PRIMARY KEY (id),****
>
> )****
>
> ** **
>
> ** **
>
> *From:* Chad Vogel [mailto:cvogel at lyonl.com]
> *Sent:* Friday, September 23, 2011 12:31 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] large block of DID's****
>
> ** **
>
> I would love to see your module because I'm are thinking about writing one
> c or c# to do almost the same thing. are you then routing the DID to an
> application server to manage the users? ****
>
> ** **
>
> ** **
>
> On Sep 23, 2011, at 11:26 AM, Robert Hadley wrote:****
>
>
>
> ****
>
>  ****
>
> I created a C didmap module I can share that uses ODBC to lookup a variable
> length DID number and return the user's extension, plus flags for fax server
> and mobile DIDs.  If you pass in the caller_id_number it also checks for
> blocked callers.****
>
>  ****
>
> Robert****
>
>  ****
>
>  ****
>
> *From:* Avi Marcus [mailto:avi at avimarcus.net]
> *Sent:* Thursday, September 22, 2011 12:54 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] large block of DID's****
>
>  ****
>
> I'd be interested in seeing the Lua script that does the users/directory. I
> don't think I've seen one yet, I'm not quite sure how that would work. I've
> only used xml_curl for a dialplan with a web server..
> ****
>
> -Avi****
>
>
>
>
> ****
>
> On Thu, Sep 22, 2011 at 10:00 PM, Chad Vogel <cvogel at lyonl.com> wrote:****
>
> Are you also using a lua script via odbc to manage users?
>
>
> On Sep 14, 2011, at 10:39 PM, <ljjimenez at gmail.com>****
>
>  <ljjimenez at gmail.com> wrote:
>
> > We do this with about 10k DIDs, we use MySQL and a Lua script via ODBC
> >
> > Luis Jimenez
> >
> > -----Original Message-----
> > From: Chad Vogel <cvogel at lyonl.com>
> > Sender: freeswitch-users-bounces at lists.freeswitch.org
> > Date: Wed, 14 Sep 2011 15:44:23
> > To: FreeSWITCH-users at lists.freeswitch.org<
> FreeSWITCH-users at lists.freeswitch.org>
> > Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> > Subject: [Freeswitch-users] large block of DID's
> >
> > I was wondering if anyone has any thoughts what is the best way to manage
> a large blocks of DID? (we have about 1600 DID's that we are moving off of
> our asterisk servers)
> >
> > Chad
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>  ****
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From s.lam at keensystems.eu  Sun Sep 25 12:13:41 2011
From: s.lam at keensystems.eu (Steven Lam, KeenSystems B.V.)
Date: Sun, 25 Sep 2011 08:13:41 +0000
Subject: [Freeswitch-users] Problem compiling
In-Reply-To: <CAKbxfG-GKDBTaQ03dLHUjZCpxW29QjDPzwvshp+P1NAMK9my-g@mail.gmail.com>
References: <038DA521FD819D47845F6EA1B563BEB224981375@dc2.keensystems.eu>
	<32340.1316811011@ccs.covici.com>
	<CAKbxfG-GKDBTaQ03dLHUjZCpxW29QjDPzwvshp+P1NAMK9my-g@mail.gmail.com>
Message-ID: <57D97596-98B6-4D6B-9C37-484B23481254@keensystems.eu>

I was using git master.
After some testing i discovered going back to git a78ec2588a solves the problem.

Can it be there is a problem in libs/esl/fs_cli.c atm?

Steven

On 24 sep 2011, at 20:57, Anthony Minessale wrote:

> There were a round of patches to make FS build on newer compilers, are
> you running an older checkout ?
> 
> 
> On Fri, Sep 23, 2011 at 3:50 PM,  <covici at ccs.covici.com> wrote:
>> I am not seeing this here -- I am running gentoo unstable amd64.  You
>> might try running bootstrap -- save your modules.conf somewhere and put
>> it back and try again..
>> 
>> Steven Lam, KeenSystems B.V. <s.lam at keensystems.eu> wrote:
>> 
>>> Hi,
>>> 
>>> I'm having problems compiling FreeSWITCH on a Gentoo x86_64 machine..
>>> I did a git checkout and still have the same problem, the error i see whem compiling is:
>>> 
>>> make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /source/fre!
>>  es!
>>>  witch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
>>> Making all in ..
>>> cc1: warnings being treated as errors
>>> libs/esl/fs_cli.c: In function 'clear_cli':
>>> libs/esl/fs_cli.c:109: error: ignoring return value of 'write', declared with attribute warn_unused_result
>>> libs/esl/fs_cli.c: In function 'clear_line':
>>> libs/esl/fs_cli.c:578: error: ignoring return value of 'write', declared with attribute warn_unused_result
>>> libs/esl/fs_cli.c:580: error: ignoring return value of 'write', declared with attribute warn_unused_result
>>> libs/esl/fs_cli.c:581: error: ignoring return value of 'write', declared with attribute warn_unused_result
>>> libs/esl/fs_cli.c: In function 'redisplay':
>>> libs/esl/fs_cli.c:588: error: ignoring return value of 'write', declared with attribute warn_unused_result
>>> libs/esl/fs_cli.c:589: error: ignoring return value of 'write', declared with attribute warn_unused_result
>>> make[2]: *** [fs_cli-fs_cli.o] Error 1
>>> 
>>> Is there someone who can help me on this?
>>> 
>>> Thanks!
>>> 
>>> Steven
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>> 
> 
> 
> 
> -- 
> 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 steveu at coppice.org  Sun Sep 25 14:24:55 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 25 Sep 2011 18:24:55 +0800
Subject: [Freeswitch-users] G.729
In-Reply-To: <4E7BADC8.60309@gmail.com>
References: <4E79E8B9.1040402@gmail.com>	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>	<4E7B1E6D.4090705@gmail.com>	<CALNZuw=ZGO5WqQ3rYcHMcgnq0+5TQ=2HabHdLnAuMAMtZ6dq5A@mail.gmail.com>
	<4E7BADC8.60309@gmail.com>
Message-ID: <4E7F0177.3010106@coppice.org>

We already tested it in a lab. I think you have other reasons for 
wanting for free. :-)

Neither Freeswitch nor Asterisk offer free trial licences for G.729, but 
the patent licencing does not permit copies of the codec to be handed 
out without paying royalties.

The codec termed "free" are ones were the source code is being 
distributed without any regard for licencing. They are different, and 
generally quite a bit slower than the commercial codecs, so using them 
doesn't tell you a lot in a lab test.

Steve


On 09/23/2011 05:51 AM, Fernando Berretta wrote:
> In order to use in a laboratory to test would be a good example
>
> On 9/22/2011 8:54 AM, Avi Marcus wrote:
>> What do you need to test? The g729 works for as many channels as you 
>> pay for...
>>
>> -Avi
>>
>>
>> On Thu, Sep 22, 2011 at 2:39 PM, Fernando Berretta 
>> <fernando.berretta at gmail.com <mailto:fernando.berretta at gmail.com>> wrote:
>>
>>     Kristian,
>>
>>     It was a typo.. the question would be.. is there some way to install
>>     g729 codec in Freeswitch for free, in order to "try" .....
>>
>>     Regards,
>>
>>     On 9/21/2011 10:54 AM, Kristian Kielhofner wrote:
>>     > This is a mailing list for FreeSWITCH discussion, not FreePBX.
>>     >
>>     > On Wed, Sep 21, 2011 at 9:38 AM, Fernando Berretta
>>     > <fernando.berretta at gmail.com
>>     <mailto:fernando.berretta at gmail.com>>  wrote:
>>     >> Hi,
>>     >>
>>     >> Is there some way to install g729 codec in freepbx for free,
>>     in order to
>>     >> "try" it like is provided for asterisk ?
>>     >>
>>     >> Best Regards,
>>     >> Fernando
>>     >>
>>



From steveu at coppice.org  Sun Sep 25 14:27:13 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 25 Sep 2011 18:27:13 +0800
Subject: [Freeswitch-users] G.729
In-Reply-To: <CAMWiMy_WfpAiiw8-tEKe9F5mcq3Q34sqStp5FdHjdr1uAoS6VA@mail.gmail.com>
References: <4E79E8B9.1040402@gmail.com>	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>	<4E7B1E6D.4090705@gmail.com>	<CALNZuw=ZGO5WqQ3rYcHMcgnq0+5TQ=2HabHdLnAuMAMtZ6dq5A@mail.gmail.com>
	<CAMWiMy_WfpAiiw8-tEKe9F5mcq3Q34sqStp5FdHjdr1uAoS6VA@mail.gmail.com>
Message-ID: <4E7F0201.30804@coppice.org>

Hi,

On 09/23/2011 12:42 AM, curriegrad2004 wrote:
> There is no G729 trial codecs in FreeSWITCH. You need to pay royalties
> to SIPPRO regardless. There's always the option of waiting until
> somewhere 2014 for the patents to expire ;)
>
The last of the G.723.1 patents expires in 2014. There are G.729 patents 
running a bit longer than that, but its hard to find the exact date.

Steve



From jcgpoza at gmail.com  Sun Sep 25 16:47:18 2011
From: jcgpoza at gmail.com (jcgpoza gonzalez)
Date: Sun, 25 Sep 2011 14:47:18 +0200
Subject: [Freeswitch-users] please help for Failed Registration,
 setting retry to 1020 seconds
In-Reply-To: <BLU121-W2467447ED24E90F6E665D7B50D0@phx.gbl>
References: <BLU121-W2467447ED24E90F6E665D7B50D0@phx.gbl>
Message-ID: <CAMDVdeiThDYyKQMA8VWHwsGm88zG1c2iLmr8X58NK4XpxfakUw@mail.gmail.com>

Hello James,

are you using a Dinstar GSM Gateway or this is a another issue? ??

-I used to get the same error with my gateway but only when I forced
registration so your profile doesn't make much sense to me.
-Check your gateway configuration, check the port and IP where your FS is
listening. (5080 is external by default in FS and you'll probably have to
set it in your GW)
-your   <param name="expire-seconds" value=""/> is empty.

Please let me know if you need some help, I have the feeling you've solved
this by now. ;)

Best Regards



2011/9/21 James zhu <zhulizhong at live.com>

>  hi:
> I am using GSM gateway to regiester to freeswitch, the incoming from gsm
> gateway is ok, but i can not regiester
> into freeswitch due to the gateway connection.
> ===the debug info from fs_CLI
>    ------------------------------------------------------------------------
> 2011-09-21 23:14:50.645350 [ERR] sofia_reg.c:1496 gsm Registration Failed
> with status Request Timeout [408]. failure #33
> 2011-09-21 23:14:53.318802 [WARNING] sofia_reg.c:386 gsm Failed
> Registration, setting retry to 1020 seconds.
> =====sip trace ===============
>
>    ------------------------------------------------------------------------
> send 590 bytes to udp/[172.16.33.11]:5060 at 15:14:50.147741:
>    ------------------------------------------------------------------------
>    REGISTER sip:172.16.33.11 SIP/2.0
>    Via: SIP/2.0/UDP 172.16.33.100:5080;rport;branch=z9hG4bKB09cFa6Uvyc1H
>    Max-Forwards: 70
>    From: <sip:1000 at 172.16.33.11;transport=udp>;tag=B1ytyyXQS3NKr
>    To: <sip:1000 at 172.16.33.11;transport=udp>
>    Call-ID: 4809dacc-aeee-4a33-a245-884c6788a4a6
>    CSeq: 17972965 REGISTER
>    Contact: <sip:gw+gsm at 172.16.33.100:5080;transport=udp;gw=gsm>
>    Expires: 30
>    User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Content-Length: 0
>    ------------------------------------------------------------------------
> send 590 bytes to udp/[172.16.33.11]:5060 at 15:14:46.146854:
>    ------------------------------------------------------------------------
>    REGISTER sip:172.16.33.11 SIP/2.0
>    Via: SIP/2.0/UDP 172.16.33.100:5080;rport;branch=z9hG4bKB09cFa6Uvyc1H
>    Max-Forwards: 70
>    From: <sip:1000 at 172.16.33.11;transport=udp>;tag=B1ytyyXQS3NKr
>    To: <sip:1000 at 172.16.33.11;transport=udp>
>    Call-ID: 4809dacc-aeee-4a33-a245-884c6788a4a6
>    CSeq: 17972965 REGISTER
>    Contact: <sip:gw+gsm at 172.16.33.100:5080;transport=udp;gw=gsm>
>    Expires: 30
>    User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Content-Length: 0
>
> =======gateway confi: ======
> <include>
>   <gateway name="gsm">
>   <!--/// account username *required* ///-->
>   <!-- Extension 1000 in my case -->
>   <param name="username" value="1000"/>
>   <!--/// auth realm: *optional* same as gateway name, if blank ///-->
>   <!-- It must aim to the  GateWay IP 192.168.2.2 in my case -->
>  * <param name="realm" value="172.16.33.11"/>*
>   <!--1234 in my case-->
>   <param name="password" value="1234"/>
>   <!--/// expire in seconds: *optional* 3600, if blank ///-->
>   <param name="expire-seconds" value=""/>
>   <!--/// do not register ///-->
>   <param name="register" value="false"/>
>   <!-- which transport to use for register -->
>   <param name="sip-trace" value="yes"/>
>
>  </gateway>
>  </include>
> freeswitch at internal> sofia status
>                      Name
> Type                                       Data      State
>
> =================================================================================================
>                  external       profile
> sip:mod_sofia at 172.16.33.100:5080      RUNNING (0)
>     external::example.com       gateway
> sip:joeuser at example.com      NOREG
>             *external::gsm       gateway
> sip:1000 at 172.16.33.11      FAIL_WAIT*
>             internal-ipv6       profile                   sip:mod_sofia@[::1]:5060
> RUNNING (0)
>                  internal       profile
> sip:mod_sofia at 172.16.33.100:5060      RUNNING (0)
>             172.16.33.100         alias
> internal      ALIASED
>
> =================================================================================================
> somehow, the gateway never sends anything.  anyone can give a hit?
>
> Best regards,
> James.zhu
> Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards,
> gateway(fxs/fxo/pri<->SIP).
> website: www.voipviews.com
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yehavi.bourvine at gmail.com  Sun Sep 25 16:53:46 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 25 Sep 2011 15:53:46 +0300
Subject: [Freeswitch-users] Polycom auto answer with microphone muted?
Message-ID: <CAAtWKeGtedAAL_KYhn+fg+7_DJfQ+wnANT8Uorq_BmSRvfs7qw@mail.gmail.com>

Hello,

  I have Polycom phones with which I want to implement Intercom. The users
would like it to auto answer with the microphone muted at first. The phones
indeed answer but with the mic opened...  Anyone know what should be done?

I have the following in the dial plan:

        <action application="set" data="sip_auto_answer=true" />
       <action application="set"
data="sip_h_Call-Info=<sip:${domain_name}>;answer-after=2" />
          <action application="bridge" data="${RingAt}" />
And I have in the phone's config file:

  <call.autoAnswer call.autoAnswer.micMute="1" />
However, the mic is not muted...

                                Thanks! __Yehavi:
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From curriegrad2004 at gmail.com  Sun Sep 25 18:44:38 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 25 Sep 2011 07:44:38 -0700
Subject: [Freeswitch-users] G.729
In-Reply-To: <4E7F0201.30804@coppice.org>
References: <4E79E8B9.1040402@gmail.com>
	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>
	<4E7B1E6D.4090705@gmail.com>
	<CALNZuw=ZGO5WqQ3rYcHMcgnq0+5TQ=2HabHdLnAuMAMtZ6dq5A@mail.gmail.com>
	<CAMWiMy_WfpAiiw8-tEKe9F5mcq3Q34sqStp5FdHjdr1uAoS6VA@mail.gmail.com>
	<4E7F0201.30804@coppice.org>
Message-ID: <CAMWiMy9Q96AaH36mvFERaSE4fOUWUOpmjvhgvsyh-YBF1m3Eyw@mail.gmail.com>

Steve, do you know around what year when the G.729 patents expire?

On Sun, Sep 25, 2011 at 3:27 AM, Steve Underwood <steveu at coppice.org> wrote:
> Hi,
>
> On 09/23/2011 12:42 AM, curriegrad2004 wrote:
>> There is no G729 trial codecs in FreeSWITCH. You need to pay royalties
>> to SIPPRO regardless. There's always the option of waiting until
>> somewhere 2014 for the patents to expire ;)
>>
> The last of the G.723.1 patents expires in 2014. There are G.729 patents
> running a bit longer than that, but its hard to find the exact date.
>
> Steve
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From steveu at coppice.org  Sun Sep 25 20:34:05 2011
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 26 Sep 2011 00:34:05 +0800
Subject: [Freeswitch-users] G.729
In-Reply-To: <CAMWiMy9Q96AaH36mvFERaSE4fOUWUOpmjvhgvsyh-YBF1m3Eyw@mail.gmail.com>
References: <4E79E8B9.1040402@gmail.com>	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>	<4E7B1E6D.4090705@gmail.com>	<CALNZuw=ZGO5WqQ3rYcHMcgnq0+5TQ=2HabHdLnAuMAMtZ6dq5A@mail.gmail.com>	<CAMWiMy_WfpAiiw8-tEKe9F5mcq3Q34sqStp5FdHjdr1uAoS6VA@mail.gmail.com>	<4E7F0201.30804@coppice.org>
	<CAMWiMy9Q96AaH36mvFERaSE4fOUWUOpmjvhgvsyh-YBF1m3Eyw@mail.gmail.com>
Message-ID: <4E7F57FD.2010201@coppice.org>

That's a good question. Many of the patents listed in the ITU database 
have already expired. I think the last one in the ITU database to expire 
will be some time in 2016, so its pretty much 20 years after the spec 
was first published - i.e. people were patenting techniques used by the 
codec until close to its publication data. One or two of those final 
patents might be things you could design your way around. G.729A was 
published a little later, but I don't remember any patents specific to 
it. Not all the relevant patents are listed in the ITU database, so 
there might be further problematic ones, which won't expire until 2016.

The bottom line seems to be that it will be something like 2015 or 2016 
before the G.729. G.729A and G.729B specs are in the clear.

Steve


On 09/25/2011 10:44 PM, curriegrad2004 wrote:
> Steve, do you know around what year when the G.729 patents expire?
>
> On Sun, Sep 25, 2011 at 3:27 AM, Steve Underwood<steveu at coppice.org>  wrote:
>> Hi,
>>
>> On 09/23/2011 12:42 AM, curriegrad2004 wrote:
>>> There is no G729 trial codecs in FreeSWITCH. You need to pay royalties
>>> to SIPPRO regardless. There's always the option of waiting until
>>> somewhere 2014 for the patents to expire ;)
>>>
>> The last of the G.723.1 patents expires in 2014. There are G.729 patents
>> running a bit longer than that, but its hard to find the exact date.
>>
>> Steve



From curriegrad2004 at gmail.com  Sun Sep 25 20:38:24 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 25 Sep 2011 09:38:24 -0700
Subject: [Freeswitch-users] G.729
In-Reply-To: <4E7F57FD.2010201@coppice.org>
References: <4E79E8B9.1040402@gmail.com>
	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>
	<4E7B1E6D.4090705@gmail.com>
	<CALNZuw=ZGO5WqQ3rYcHMcgnq0+5TQ=2HabHdLnAuMAMtZ6dq5A@mail.gmail.com>
	<CAMWiMy_WfpAiiw8-tEKe9F5mcq3Q34sqStp5FdHjdr1uAoS6VA@mail.gmail.com>
	<4E7F0201.30804@coppice.org>
	<CAMWiMy9Q96AaH36mvFERaSE4fOUWUOpmjvhgvsyh-YBF1m3Eyw@mail.gmail.com>
	<4E7F57FD.2010201@coppice.org>
Message-ID: <CAMWiMy-FztJB=kSNM4qMZO_c2K5EH30GugsXxLchWQZBQ6nydA@mail.gmail.com>

That's what I figured about anyways, +/- 2 years after G723.1 patents
expire. No wonder they're ramping up the enforcement nowadays

On Sun, Sep 25, 2011 at 9:34 AM, Steve Underwood <steveu at coppice.org> wrote:
> That's a good question. Many of the patents listed in the ITU database
> have already expired. I think the last one in the ITU database to expire
> will be some time in 2016, so its pretty much 20 years after the spec
> was first published - i.e. people were patenting techniques used by the
> codec until close to its publication data. One or two of those final
> patents might be things you could design your way around. G.729A was
> published a little later, but I don't remember any patents specific to
> it. Not all the relevant patents are listed in the ITU database, so
> there might be further problematic ones, which won't expire until 2016.
>
> The bottom line seems to be that it will be something like 2015 or 2016
> before the G.729. G.729A and G.729B specs are in the clear.
>
> Steve
>
>
> On 09/25/2011 10:44 PM, curriegrad2004 wrote:
>> Steve, do you know around what year when the G.729 patents expire?
>>
>> On Sun, Sep 25, 2011 at 3:27 AM, Steve Underwood<steveu at coppice.org> ?wrote:
>>> Hi,
>>>
>>> On 09/23/2011 12:42 AM, curriegrad2004 wrote:
>>>> There is no G729 trial codecs in FreeSWITCH. You need to pay royalties
>>>> to SIPPRO regardless. There's always the option of waiting until
>>>> somewhere 2014 for the patents to expire ;)
>>>>
>>> The last of the G.723.1 patents expires in 2014. There are G.729 patents
>>> running a bit longer than that, but its hard to find the exact date.
>>>
>>> Steve
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From cvogel at lyonl.com  Sun Sep 25 21:57:13 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Sun, 25 Sep 2011 17:57:13 +0000
Subject: [Freeswitch-users] Problem compiling
In-Reply-To: <57D97596-98B6-4D6B-9C37-484B23481254@keensystems.eu>
References: <038DA521FD819D47845F6EA1B563BEB224981375@dc2.keensystems.eu>
	<32340.1316811011@ccs.covici.com>
	<CAKbxfG-GKDBTaQ03dLHUjZCpxW29QjDPzwvshp+P1NAMK9my-g@mail.gmail.com>
	<57D97596-98B6-4D6B-9C37-484B23481254@keensystems.eu>
Message-ID: <E55278A0-C9A7-4D7F-B5EA-70B879FF687D@lyonl.com>

i'm having a problem compiling libsofia_sip_ua_static on windows 

On Sep 25, 2011, at 3:13 AM, Steven Lam, KeenSystems B.V. wrote:

> I was using git master.
> After some testing i discovered going back to git a78ec2588a solves the problem.
> 
> Can it be there is a problem in libs/esl/fs_cli.c atm?
> 
> Steven
> 
> On 24 sep 2011, at 20:57, Anthony Minessale wrote:
> 
>> There were a round of patches to make FS build on newer compilers, are
>> you running an older checkout ?
>> 
>> 
>> On Fri, Sep 23, 2011 at 3:50 PM,  <covici at ccs.covici.com> wrote:
>>> I am not seeing this here -- I am running gentoo unstable amd64.  You
>>> might try running bootstrap -- save your modules.conf somewhere and put
>>> it back and try again..
>>> 
>>> Steven Lam, KeenSystems B.V. <s.lam at keensystems.eu> wrote:
>>> 
>>>> Hi,
>>>> 
>>>> I'm having problems compiling FreeSWITCH on a Gentoo x86_64 machine..
>>>> I did a git checkout and still have the same problem, the error i see whem compiling is:
>>>> 
>>>> make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /source/fr
> e!
>>> es!
>>>> witch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
>>>> Making all in ..
>>>> cc1: warnings being treated as errors
>>>> libs/esl/fs_cli.c: In function 'clear_cli':
>>>> libs/esl/fs_cli.c:109: error: ignoring return value of 'write', declared with attribute warn_unused_result
>>>> libs/esl/fs_cli.c: In function 'clear_line':
>>>> libs/esl/fs_cli.c:578: error: ignoring return value of 'write', declared with attribute warn_unused_result
>>>> libs/esl/fs_cli.c:580: error: ignoring return value of 'write', declared with attribute warn_unused_result
>>>> libs/esl/fs_cli.c:581: error: ignoring return value of 'write', declared with attribute warn_unused_result
>>>> libs/esl/fs_cli.c: In function 'redisplay':
>>>> libs/esl/fs_cli.c:588: error: ignoring return value of 'write', declared with attribute warn_unused_result
>>>> libs/esl/fs_cli.c:589: error: ignoring return value of 'write', declared with attribute warn_unused_result
>>>> make[2]: *** [fs_cli-fs_cli.o] Error 1
>>>> 
>>>> Is there someone who can help me on this?
>>>> 
>>>> Thanks!
>>>> 
>>>> Steven
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>> 
>> 
>> 
>> 
>> -- 
>> 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
> 




From marhbere at hotmail.com  Mon Sep 26 00:29:07 2011
From: marhbere at hotmail.com (MarhBere)
Date: Sun, 25 Sep 2011 13:29:07 -0700 (PDT)
Subject: [Freeswitch-users] Auto adjust Change port RTP
Message-ID: <1316982547007-6829940.post@n2.nabble.com>

Hi, I have an issue with extension (eyebeam) behind nat.

Scenario: The call Incoming From ITSP  and is bridged with the a user
registered; The RTP arrive to FS from register user, but don't have the auto
adjust IP/Port. So only had silence in both points. IF the phone Hold the
call and Unhold the call. FS now yet change the RTP to correct private
IP/Port and call is now with bidirectional audio.

I red that is hapened in old release :
http://freeswitch-users.2379917.n2.nabble.com/No-RTP-destination-auto-correction-tt3906730.html#none
But I running the last Head trunk. "FreeSWITCH Version 1.0.head (git-8df1872
2011-09-19 13-51-41 -0400)" SO: CentOs 5. 64x

Someone know what is wrong?
PD: Sorry for my traduction.

Thanks,
Marcelo Bereterbide.


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Auto-adjust-Change-port-RTP-tp6829940p6829940.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble_01394 at slickdeals.endjunk.com  Mon Sep 26 03:16:55 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Sun, 25 Sep 2011 16:16:55 -0700 (PDT)
Subject: [Freeswitch-users] Problem compiling
In-Reply-To: <32340.1316811011@ccs.covici.com>
References: <038DA521FD819D47845F6EA1B563BEB224981375@dc2.keensystems.eu>
	<32340.1316811011@ccs.covici.com>
Message-ID: <1316992615828-6830211.post@n2.nabble.com>


covici wrote:
> 
> I am not seeing this here -- I am running gentoo unstable amd64.
Me too, FS git (23c981df68e486725addc1e2dc77010adb0f4c31) hosted on a
Seagate DockStar under OpenWRT.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-compiling-tp6825542p6830211.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From henrikaagaardsorensen at gmail.com  Mon Sep 26 01:32:10 2011
From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=)
Date: Sun, 25 Sep 2011 23:32:10 +0200
Subject: [Freeswitch-users] Voicemail counts saved messages wrong.
Message-ID: <CAGH8SeY5roEsVXhfqozMuiTS8HOt9t2-Js64YLG8v32Y6tvogA@mail.gmail.com>

I'm running FreeSwitch 1.0.head on Ubuntu 10.04 LTS.

Everything seems to work pretty well, but I'm experience some problems with
the voicemail and how it counts saved messages.

Currently I have 2 saved messages, but when it announce the number of
messages in the voicemail it says "You have 3 saved messages".
Looking in the directory storage/... there's also only 2 messages.

So how come it counts that as 3 messages?
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From reza at lethalnetworks.com  Mon Sep 26 03:59:49 2011
From: reza at lethalnetworks.com (reza a)
Date: Sun, 25 Sep 2011 16:59:49 -0700 (PDT)
Subject: [Freeswitch-users] prepaid solutions
In-Reply-To: <b20533d5-ae5f-4f7a-b1e3-1e48f2cbeab5@zcs>
Message-ID: <2c4a73f4-b37a-4067-9818-d0bbfe26fded@zcs>

Freeswitch users,
Does anyone use freeswitch for prepaid applications? Do you have a walk through on setting up an environment like this? I'd like to play with freeswitch a bit, I know there alot of modules out there but I'm wondering what's the best cohesive set of modules/admin tools to use for a prepaid style applications.
Thanks a lot


From cvogel at lyonl.com  Mon Sep 26 06:17:39 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Mon, 26 Sep 2011 02:17:39 +0000
Subject: [Freeswitch-users] G.729
In-Reply-To: <CAMWiMy-FztJB=kSNM4qMZO_c2K5EH30GugsXxLchWQZBQ6nydA@mail.gmail.com>
References: <4E79E8B9.1040402@gmail.com>
	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>
	<4E7B1E6D.4090705@gmail.com>
	<CALNZuw=ZGO5WqQ3rYcHMcgnq0+5TQ=2HabHdLnAuMAMtZ6dq5A@mail.gmail.com>
	<CAMWiMy_WfpAiiw8-tEKe9F5mcq3Q34sqStp5FdHjdr1uAoS6VA@mail.gmail.com>
	<4E7F0201.30804@coppice.org>
	<CAMWiMy9Q96AaH36mvFERaSE4fOUWUOpmjvhgvsyh-YBF1m3Eyw@mail.gmail.com>
	<4E7F57FD.2010201@coppice.org>
	<CAMWiMy-FztJB=kSNM4qMZO_c2K5EH30GugsXxLchWQZBQ6nydA@mail.gmail.com>
Message-ID: <25515286-0632-4245-820E-8C2E0E55C3E5@lyonl.com>

Check out: http://www.voiceage.com/prodg729.php

On Sep 25, 2011, at 11:38 AM, curriegrad2004 wrote:

> That's what I figured about anyways, +/- 2 years after G723.1 patents
> expire. No wonder they're ramping up the enforcement nowadays
> 
> On Sun, Sep 25, 2011 at 9:34 AM, Steve Underwood <steveu at coppice.org> wrote:
>> That's a good question. Many of the patents listed in the ITU database
>> have already expired. I think the last one in the ITU database to expire
>> will be some time in 2016, so its pretty much 20 years after the spec
>> was first published - i.e. people were patenting techniques used by the
>> codec until close to its publication data. One or two of those final
>> patents might be things you could design your way around. G.729A was
>> published a little later, but I don't remember any patents specific to
>> it. Not all the relevant patents are listed in the ITU database, so
>> there might be further problematic ones, which won't expire until 2016.
>> 
>> The bottom line seems to be that it will be something like 2015 or 2016
>> before the G.729. G.729A and G.729B specs are in the clear.
>> 
>> Steve
>> 
>> 
>> On 09/25/2011 10:44 PM, curriegrad2004 wrote:
>>> Steve, do you know around what year when the G.729 patents expire?
>>> 
>>> On Sun, Sep 25, 2011 at 3:27 AM, Steve Underwood<steveu at coppice.org>  wrote:
>>>> Hi,
>>>> 
>>>> On 09/23/2011 12:42 AM, curriegrad2004 wrote:
>>>>> There is no G729 trial codecs in FreeSWITCH. You need to pay royalties
>>>>> to SIPPRO regardless. There's always the option of waiting until
>>>>> somewhere 2014 for the patents to expire ;)
>>>>> 
>>>> The last of the G.723.1 patents expires in 2014. There are G.729 patents
>>>> running a bit longer than that, but its hard to find the exact date.
>>>> 
>>>> 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
>> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 darklion11 at yahoo.com  Mon Sep 26 07:44:47 2011
From: darklion11 at yahoo.com (Edmar Cruz)
Date: Sun, 25 Sep 2011 20:44:47 -0700 (PDT)
Subject: [Freeswitch-users]  Invalid Application reject
Message-ID: <32503926.post@talk.nabble.com>


Hi Guys,
 
 I always getting error "Invalid application reject" calling in local
extensions and I cannot call locally. But external calls are ok. This are
the results.


 2011-09-26 11:44:14.564848 [NOTICE] switch_channel.c:669 New Channel
sofia/internal/8011105 at 202.6.85.46 [883ec35a-4f52-419a-81ff-fd6d43709868]
2011-09-26 11:44:14.616854 [INFO] mod_dialplan_xml.c:418 Processing
1001->1003 in context default
2011-09-26 11:44:14.868869 [ERR] switch_core_session.c:1731 Invalid
Application reject
2011-09-26 11:44:14.868869 [NOTICE] switch_core_session.c:1732 Hangup
sofia/internal/1001 at xxx [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
2011-09-26 11:44:15.260894 [NOTICE] switch_core_session.c:1182 Session 1
(sofia/internal/8011105 at xxx) Ended
2011-09-26 11:44:15.260894 [NOTICE] switch_core_session.c:1184 Close Channel
sofia/internal/8011105 at xxx [CS_DESTROY]

What maybe the problem? Please help...

Thanks

-- 
View this message in context: http://old.nabble.com/Invalid-Application-reject-tp32503926p32503926.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From msc at freeswitch.org  Mon Sep 26 07:46:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 25 Sep 2011 20:46:36 -0700
Subject: [Freeswitch-users] ACL to fallback digest authentication
In-Reply-To: <CACak_yxqX0WLp7Pa_zgMbTfWE2NnGLEoSuvVg9Yjj9M9bzJzXw@mail.gmail.com>
References: <CACak_yxqX0WLp7Pa_zgMbTfWE2NnGLEoSuvVg9Yjj9M9bzJzXw@mail.gmail.com>
Message-ID: <CAKzWOxVAc3=Sb8e1_43c6NLQ5WcSEGyie6agFn7bqrWQapAPpA@mail.gmail.com>

Have you tried the "cidr=x.x.x.x" in the user's config file?
http://wiki.freeswitch.org/wiki/Acl#Users

-MC

On Thu, Sep 22, 2011 at 4:31 AM, Muhammad Naseer Bhatti
<nbhatti at gmail.com>wrote:

>
> Hello,
> I am trying to implement ACL in FreeSWITCH. What I want to do is, check the
> user's IP in the ACL. If that is present, let the user send INVITE. If ACL
> fails, fallback to digest authentication and authentication user/pass.
> Problem is that when I do auth_calls true, I am not able to INVITE. We have
> actually two scenarios, 1 make call without REGISTER, second REGISTER and
> make calls. I have set default ACL to deny, but that still not helping.
> Either one of them works, with the digest or ACL. I don't want to create
> another profile and make 1 for authentication, and the other without auth.
> Is there any workaround for this?
>
> -B
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Mon Sep 26 12:27:09 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 26 Sep 2011 09:27:09 +0100
Subject: [Freeswitch-users] G.729
In-Reply-To: <4E7F57FD.2010201@coppice.org>
References: <4E79E8B9.1040402@gmail.com>
	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>
	<4E7B1E6D.4090705@gmail.com>
	<CALNZuw=ZGO5WqQ3rYcHMcgnq0+5TQ=2HabHdLnAuMAMtZ6dq5A@mail.gmail.com>
	<CAMWiMy_WfpAiiw8-tEKe9F5mcq3Q34sqStp5FdHjdr1uAoS6VA@mail.gmail.com>
	<4E7F0201.30804@coppice.org>
	<CAMWiMy9Q96AaH36mvFERaSE4fOUWUOpmjvhgvsyh-YBF1m3Eyw@mail.gmail.com>
	<4E7F57FD.2010201@coppice.org>
Message-ID: <CAFiqYunwLBHmYLx6C5wgUxKya9oE5GEfROsXMhNAsz39X9doGw@mail.gmail.com>

>
> Not all the relevant patents are listed in the ITU database, so
> there might be further problematic ones, which won't expire until 2016.
>
> The bottom line seems to be that it will be something like 2015 or 2016
> before the G.729. G.729A and G.729B specs are in the clear.
>

Damn, there I was looking forward to next year... :(

-Steve



On 25 September 2011 17:34, Steve Underwood <steveu at coppice.org> wrote:

> That's a good question. Many of the patents listed in the ITU database
> have already expired. I think the last one in the ITU database to expire
> will be some time in 2016, so its pretty much 20 years after the spec
> was first published - i.e. people were patenting techniques used by the
> codec until close to its publication data. One or two of those final
> patents might be things you could design your way around. G.729A was
> published a little later, but I don't remember any patents specific to
> it. Not all the relevant patents are listed in the ITU database, so
> there might be further problematic ones, which won't expire until 2016.
>
> The bottom line seems to be that it will be something like 2015 or 2016
> before the G.729. G.729A and G.729B specs are in the clear.
>
> Steve
>
>
> On 09/25/2011 10:44 PM, curriegrad2004 wrote:
> > Steve, do you know around what year when the G.729 patents expire?
> >
> > On Sun, Sep 25, 2011 at 3:27 AM, Steve Underwood<steveu at coppice.org>
>  wrote:
> >> Hi,
> >>
> >> On 09/23/2011 12:42 AM, curriegrad2004 wrote:
> >>> There is no G729 trial codecs in FreeSWITCH. You need to pay royalties
> >>> to SIPPRO regardless. There's always the option of waiting until
> >>> somewhere 2014 for the patents to expire ;)
> >>>
> >> The last of the G.723.1 patents expires in 2014. There are G.729 patents
> >> running a bit longer than that, but its hard to find the exact date.
> >>
> >> Steve
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Mon Sep 26 12:26:16 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 26 Sep 2011 09:26:16 +0100
Subject: [Freeswitch-users] G.729
In-Reply-To: <CACak_yzPA0e4kk0guxZXhrVPYt+jzqU-bT20c8bCo3X3oDVqMQ@mail.gmail.com>
References: <4E79E8B9.1040402@gmail.com>
	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>
	<4E7B1E6D.4090705@gmail.com>
	<CACak_yzPA0e4kk0guxZXhrVPYt+jzqU-bT20c8bCo3X3oDVqMQ@mail.gmail.com>
Message-ID: <CAFiqYu=9VfgMzCMgiTKUsehz_DRspa+S89uYJc4wwj3ozXQ7nA@mail.gmail.com>

NO NO NO NO NO

1) This codec is ILLEGAL. You MUST have a G729 license to use G729
encoders/decoders, until the codecs expire (I've head this is in 2012?)

2) You are not "trying" anything. Testing with one codec implementation
doesn't tell you anything about how well a different implementation of the
same codec will work!

3) That codec is less stable and not as optimised as the mod_com_g729 one.

-Steve





On 22 September 2011 12:51, Muhammad Naseer Bhatti <nbhatti at gmail.com>wrote:

>
> Take a look here, https://github.com/Deepwalker/fs_itu_g729
>
>
>
> On Thu, Sep 22, 2011 at 2:39 PM, Fernando Berretta <
> fernando.berretta at gmail.com> wrote:
>
>> Kristian,
>>
>> It was a typo.. the question would be.. is there some way to install
>> g729 codec in Freeswitch for free, in order to "try" .....
>>
>> Regards,
>>
>> On 9/21/2011 10:54 AM, Kristian Kielhofner wrote:
>> > This is a mailing list for FreeSWITCH discussion, not FreePBX.
>> >
>> > On Wed, Sep 21, 2011 at 9:38 AM, Fernando Berretta
>> > <fernando.berretta at gmail.com>  wrote:
>> >> Hi,
>> >>
>> >> Is there some way to install g729 codec in freepbx for free, in order
>> to
>> >> "try" it like is provided for asterisk ?
>> >>
>> >> Best Regards,
>> >> Fernando
>> >>
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Mon Sep 26 12:30:41 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 26 Sep 2011 09:30:41 +0100
Subject: [Freeswitch-users] Invalid Application reject
In-Reply-To: <32503926.post@talk.nabble.com>
References: <32503926.post@talk.nabble.com>
Message-ID: <CAFiqYukvkgSdZHuFDmu7KNrbHO1nrjmhX9PGxDfUMOx8Pq2vhQ@mail.gmail.com>

Somewhere in your dialplan you have this:
<action application="reject" />

There's no such application anywhere in FreeSWITCH (any longer).
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_reject
"Deprecated, has been replaced with respond"

Local calls are probably hitting a different dialplan context from external,
to explain the different behaviour.

-Steve



On 26 September 2011 04:44, Edmar Cruz <darklion11 at yahoo.com> wrote:

>
> Hi Guys,
>
>  I always getting error "Invalid application reject" calling in local
> extensions and I cannot call locally. But external calls are ok. This are
> the results.
>
>
>  2011-09-26 11:44:14.564848 [NOTICE] switch_channel.c:669 New Channel
> sofia/internal/8011105 at 202.6.85.46 [883ec35a-4f52-419a-81ff-fd6d43709868]
> 2011-09-26 11:44:14.616854 [INFO] mod_dialplan_xml.c:418 Processing
> 1001->1003 in context default
> 2011-09-26 11:44:14.868869 [ERR] switch_core_session.c:1731 Invalid
> Application reject
> 2011-09-26 11:44:14.868869 [NOTICE] switch_core_session.c:1732 Hangup
> sofia/internal/1001 at xxx [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
> 2011-09-26 11:44:15.260894 [NOTICE] switch_core_session.c:1182 Session 1
> (sofia/internal/8011105 at xxx) Ended
> 2011-09-26 11:44:15.260894 [NOTICE] switch_core_session.c:1184 Close
> Channel
> sofia/internal/8011105 at xxx [CS_DESTROY]
>
> What maybe the problem? Please help...
>
> Thanks
>
> --
> View this message in context:
> http://old.nabble.com/Invalid-Application-reject-tp32503926p32503926.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From andy at fabulous4.co.uk  Mon Sep 26 13:11:19 2011
From: andy at fabulous4.co.uk (Andy Ayers)
Date: Mon, 26 Sep 2011 10:11:19 +0100
Subject: [Freeswitch-users] Incompelete calls in the log files
Message-ID: <019a01cc7c2c$41ac7c10$c5057430$@fabulous4.co.uk>

Hi,

 

I have been investigating a number of calls coming in to my switch that
don't appear to complete properly. These are incoming calls to a service
which records voicemail messgaes.

 

When I look in the logs to see what has happened to these calls the logging
information seems to simply come to an abrupt halt at a certain point in the
call. In other words there is no information about the call being hung up or
the session ended. They all seem to end at the same point in the call and
the last few lines in the log file are:

 

018afe16-e402-11e0-8e40-a5b8fd8e2a79 2011-09-21 04:30:14.848903 [DEBUG]
switch_ivr_play_say.c:452 Raw Codec Activated, ready to waste resources!

2011-09-21 04:30:14.849956 [WARNING] switch_core_file.c:176 Sample rate
doesn't match

018afe16-e402-11e0-8e40-a5b8fd8e2a79 2011-09-21 04:30:14.849956 [DEBUG]
switch_ivr_play_say.c:560 Raw Codec Activated

018afe16-e402-11e0-8e40-a5b8fd8e2a79 2011-09-21 04:30:14.849956 [DEBUG]
switch_core_codec.c:122 sofia/external/anonymous at localhost Push codec L16:10

018afe16-e402-11e0-8e40-a5b8fd8e2a79 2011-09-21 04:30:29.028452 [DEBUG]
switch_core_io.c:403 Engaging Read Buffer at 320 bytes vs 40

 

It's also then in most cases followed by a new call coming in:

 

39cade04-e402-11e0-8e41-a5b8fd8e2a79 2011-09-21 04:31:43.177540 [NOTICE]
switch_channel.c:669 New Channel sofia/external/anonymous at localhost
[39cade04-e402-11e0-8e41-a5b8fd8e2a79]

39cade04-e402-11e0-8e41-a5b8fd8e2a79 2011-09-21 04:31:43.177540 [DEBUG]
switch_core_state_machine.c:314 (sofia/external/anonymous at localhost) Running
State Change CS_NEW

39cade04-e402-11e0-8e41-a5b8fd8e2a79 2011-09-21 04:31:43.177540 [DEBUG]
switch_core_state_machine.c:320 (sofia/external/anonymous at localhost) State
NEW

39cade04-e402-11e0-8e41-a5b8fd8e2a79 2011-09-21 04:31:43.211238 [DEBUG]
sofia.c:4153 Channel sofia/external/anonymous at localhost entering state
[received][100]

 

I'm guessing the session has simply crashed somehow and freeswitch has been
good enough to recover from it.

 

Anyone had a similar experience or can shed any light on what's actually
happening and how to prevent it?

 

I'm on version 1.06

 

Cheers

Andy

 

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From steveu at coppice.org  Mon Sep 26 15:52:06 2011
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 26 Sep 2011 19:52:06 +0800
Subject: [Freeswitch-users] G.729
In-Reply-To: <25515286-0632-4245-820E-8C2E0E55C3E5@lyonl.com>
References: <4E79E8B9.1040402@gmail.com>	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>	<4E7B1E6D.4090705@gmail.com>	<CALNZuw=ZGO5WqQ3rYcHMcgnq0+5TQ=2HabHdLnAuMAMtZ6dq5A@mail.gmail.com>	<CAMWiMy_WfpAiiw8-tEKe9F5mcq3Q34sqStp5FdHjdr1uAoS6VA@mail.gmail.com>	<4E7F0201.30804@coppice.org>	<CAMWiMy9Q96AaH36mvFERaSE4fOUWUOpmjvhgvsyh-YBF1m3Eyw@mail.gmail.com>	<4E7F57FD.2010201@coppice.org>	<CAMWiMy-FztJB=kSNM4qMZO_c2K5EH30GugsXxLchWQZBQ6nydA@mail.gmail.com>
	<25515286-0632-4245-820E-8C2E0E55C3E5@lyonl.com>
Message-ID: <4E806766.4060205@coppice.org>

On 09/26/2011 10:17 AM, Chad Vogel wrote:
> Check out: http://www.voiceage.com/prodg729.php
Is there supposed to be something interesting amongst the ads on that 
page? If so, it might have been helpful to point it out.

Steve

> On Sep 25, 2011, at 11:38 AM, curriegrad2004 wrote:
>
>> That's what I figured about anyways, +/- 2 years after G723.1 patents
>> expire. No wonder they're ramping up the enforcement nowadays
>>
>> On Sun, Sep 25, 2011 at 9:34 AM, Steve Underwood<steveu at coppice.org>  wrote:
>>> That's a good question. Many of the patents listed in the ITU database
>>> have already expired. I think the last one in the ITU database to expire
>>> will be some time in 2016, so its pretty much 20 years after the spec
>>> was first published - i.e. people were patenting techniques used by the
>>> codec until close to its publication data. One or two of those final
>>> patents might be things you could design your way around. G.729A was
>>> published a little later, but I don't remember any patents specific to
>>> it. Not all the relevant patents are listed in the ITU database, so
>>> there might be further problematic ones, which won't expire until 2016.
>>>
>>> The bottom line seems to be that it will be something like 2015 or 2016
>>> before the G.729. G.729A and G.729B specs are in the clear.
>>>
>>> Steve
>>>
>>>
>>> On 09/25/2011 10:44 PM, curriegrad2004 wrote:
>>>> Steve, do you know around what year when the G.729 patents expire?
>>>>
>>>> On Sun, Sep 25, 2011 at 3:27 AM, Steve Underwood<steveu at coppice.org>   wrote:
>>>>> Hi,
>>>>>
>>>>> On 09/23/2011 12:42 AM, curriegrad2004 wrote:
>>>>>> There is no G729 trial codecs in FreeSWITCH. You need to pay royalties
>>>>>> to SIPPRO regardless. There's always the option of waiting until
>>>>>> somewhere 2014 for the patents to expire ;)
>>>>>>
>>>>> The last of the G.723.1 patents expires in 2014. There are G.729 patents
>>>>> running a bit longer than that, but its hard to find the exact date.
>>>>>
>>>>> Steve
>>>



From curriegrad2004 at gmail.com  Mon Sep 26 18:48:51 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 26 Sep 2011 07:48:51 -0700
Subject: [Freeswitch-users] G.729
In-Reply-To: <4E806766.4060205@coppice.org>
References: <4E79E8B9.1040402@gmail.com>
	<CAKDfjgc+5pvxQW4yw7_ebir7j6b36kC0uF=VLfqj2PKHhJKOPA@mail.gmail.com>
	<4E7B1E6D.4090705@gmail.com>
	<CALNZuw=ZGO5WqQ3rYcHMcgnq0+5TQ=2HabHdLnAuMAMtZ6dq5A@mail.gmail.com>
	<CAMWiMy_WfpAiiw8-tEKe9F5mcq3Q34sqStp5FdHjdr1uAoS6VA@mail.gmail.com>
	<4E7F0201.30804@coppice.org>
	<CAMWiMy9Q96AaH36mvFERaSE4fOUWUOpmjvhgvsyh-YBF1m3Eyw@mail.gmail.com>
	<4E7F57FD.2010201@coppice.org>
	<CAMWiMy-FztJB=kSNM4qMZO_c2K5EH30GugsXxLchWQZBQ6nydA@mail.gmail.com>
	<25515286-0632-4245-820E-8C2E0E55C3E5@lyonl.com>
	<4E806766.4060205@coppice.org>
Message-ID: <CAMWiMy8cy13m1t=C=xfp00PTtNxvVWf+3ORKc9-jvwtCWbSscA@mail.gmail.com>

2014-2015 is still a short wait, really

On Mon, Sep 26, 2011 at 4:52 AM, Steve Underwood <steveu at coppice.org> wrote:
> On 09/26/2011 10:17 AM, Chad Vogel wrote:
>> Check out: http://www.voiceage.com/prodg729.php
> Is there supposed to be something interesting amongst the ads on that
> page? If so, it might have been helpful to point it out.
>
> Steve
>
>> On Sep 25, 2011, at 11:38 AM, curriegrad2004 wrote:
>>
>>> That's what I figured about anyways, +/- 2 years after G723.1 patents
>>> expire. No wonder they're ramping up the enforcement nowadays
>>>
>>> On Sun, Sep 25, 2011 at 9:34 AM, Steve Underwood<steveu at coppice.org>  wrote:
>>>> That's a good question. Many of the patents listed in the ITU database
>>>> have already expired. I think the last one in the ITU database to expire
>>>> will be some time in 2016, so its pretty much 20 years after the spec
>>>> was first published - i.e. people were patenting techniques used by the
>>>> codec until close to its publication data. One or two of those final
>>>> patents might be things you could design your way around. G.729A was
>>>> published a little later, but I don't remember any patents specific to
>>>> it. Not all the relevant patents are listed in the ITU database, so
>>>> there might be further problematic ones, which won't expire until 2016.
>>>>
>>>> The bottom line seems to be that it will be something like 2015 or 2016
>>>> before the G.729. G.729A and G.729B specs are in the clear.
>>>>
>>>> Steve
>>>>
>>>>
>>>> On 09/25/2011 10:44 PM, curriegrad2004 wrote:
>>>>> Steve, do you know around what year when the G.729 patents expire?
>>>>>
>>>>> On Sun, Sep 25, 2011 at 3:27 AM, Steve Underwood<steveu at coppice.org>   wrote:
>>>>>> Hi,
>>>>>>
>>>>>> On 09/23/2011 12:42 AM, curriegrad2004 wrote:
>>>>>>> There is no G729 trial codecs in FreeSWITCH. You need to pay royalties
>>>>>>> to SIPPRO regardless. There's always the option of waiting until
>>>>>>> somewhere 2014 for the patents to expire ;)
>>>>>>>
>>>>>> The last of the G.723.1 patents expires in 2014. There are G.729 patents
>>>>>> running a bit longer than that, but its hard to find the exact date.
>>>>>>
>>>>>> Steve
>>>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Mon Sep 26 19:39:44 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Sep 2011 10:39:44 -0500
Subject: [Freeswitch-users] Incompelete calls in the log files
In-Reply-To: <019a01cc7c2c$41ac7c10$c5057430$@fabulous4.co.uk>
References: <019a01cc7c2c$41ac7c10$c5057430$@fabulous4.co.uk>
Message-ID: <CAKbxfG_F5UtuNcsi81bGVe+RMeTp9QU-q-_t+yR9hnER1T2HBQ@mail.gmail.com>

Your first step is to get the latest GIT version from
http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide
This is always the best first step when troubleshooting because in the
event we need to make any changes to the code we always need to start
with this revision anyway plus 1.0.6 is more than a year old.


On Mon, Sep 26, 2011 at 4:11 AM, Andy Ayers <andy at fabulous4.co.uk> wrote:
> Hi,
>
>
>
> I have been investigating a number of calls coming in to my switch that
> don?t appear to complete properly. These are incoming calls to a service
> which records voicemail messgaes.
>
>
>
> When I look in the logs to see what has happened to these calls the logging
> information seems to simply come to an abrupt halt at a certain point in the
> call. In other words there is no information about the call being hung up or
> the session ended. They all seem to end at the same point in the call and
> the last few lines in the log file are:
>
>
>
> 018afe16-e402-11e0-8e40-a5b8fd8e2a79 2011-09-21 04:30:14.848903 [DEBUG]
> switch_ivr_play_say.c:452 Raw Codec Activated, ready to waste resources!
>
> 2011-09-21 04:30:14.849956 [WARNING] switch_core_file.c:176 Sample rate
> doesn't match
>
> 018afe16-e402-11e0-8e40-a5b8fd8e2a79 2011-09-21 04:30:14.849956 [DEBUG]
> switch_ivr_play_say.c:560 Raw Codec Activated
>
> 018afe16-e402-11e0-8e40-a5b8fd8e2a79 2011-09-21 04:30:14.849956 [DEBUG]
> switch_core_codec.c:122 sofia/external/anonymous at localhost Push codec L16:10
>
> 018afe16-e402-11e0-8e40-a5b8fd8e2a79 2011-09-21 04:30:29.028452 [DEBUG]
> switch_core_io.c:403 Engaging Read Buffer at 320 bytes vs 40
>
>
>
> It?s also then in most cases followed by a new call coming in:
>
>
>
> 39cade04-e402-11e0-8e41-a5b8fd8e2a79 2011-09-21 04:31:43.177540 [NOTICE]
> switch_channel.c:669 New Channel sofia/external/anonymous at localhost
> [39cade04-e402-11e0-8e41-a5b8fd8e2a79]
>
> 39cade04-e402-11e0-8e41-a5b8fd8e2a79 2011-09-21 04:31:43.177540 [DEBUG]
> switch_core_state_machine.c:314 (sofia/external/anonymous at localhost) Running
> State Change CS_NEW
>
> 39cade04-e402-11e0-8e41-a5b8fd8e2a79 2011-09-21 04:31:43.177540 [DEBUG]
> switch_core_state_machine.c:320 (sofia/external/anonymous at localhost) State
> NEW
>
> 39cade04-e402-11e0-8e41-a5b8fd8e2a79 2011-09-21 04:31:43.211238 [DEBUG]
> sofia.c:4153 Channel sofia/external/anonymous at localhost entering state
> [received][100]
>
>
>
> I?m guessing the session has simply crashed somehow and freeswitch has been
> good enough to recover from it.
>
>
>
> Anyone had a similar experience or can shed any light on what?s actually
> happening and how to prevent it?
>
>
>
> I?m on version 1.06
>
>
>
> Cheers
>
> Andy
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 wstephen80 at gmail.com  Mon Sep 26 20:30:36 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 26 Sep 2011 18:30:36 +0200
Subject: [Freeswitch-users] Call Hangup with INCOMPATIBLE_DESTINATION cause
Message-ID: <CAO4LeU0FcQuinzgGqm7ztwH1w2YBjkpO-XFvCabqY3rq6b6Mwg@mail.gmail.com>

Hi all,
I have some calls dropped with INCOMPATIBLE_DESTINATION cause.
I have checked the SDP of the bridged legA and legB and to me appear ok
(both in G729).
What I see in the log is:

http://pastebin.freeswitch.org/17396

Attached to this email an example of captured trace.
Any advice on that?

Stephen
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From steveu at coppice.org  Mon Sep 26 20:32:33 2011
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 27 Sep 2011 00:32:33 +0800
Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re:
 Is	there any license G729?
In-Reply-To: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com>
References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com>
Message-ID: <4E80A921.8070600@coppice.org>

On 06/30/2009 09:26 AM, Craig Askings wrote:
> Are there any hardware phones that support 48 Khz Celt and
> automated/mass deployment?
>
> Craig.
Most of the pure VoIP chipsets aren't capable of running at 48k 
samples/second, so off the shelf phone hardware platforms probably won't 
cut it. The phones with video capabilities might have 48k sample/second 
facilities.
> 2009/6/30 Jason White<jason at jasonjgw.net>:
>> Brian West<brian at freeswitch.org>  wrote:
>>> Everyone has this need for lower bandwidth calls... I tend to march the
>>> other way. 48kHz baby!  (btw you can do 48kHz in the same bandwidth as a
>>> single ulaw call)
>> 48khz Celt (celt at 48000 in your codec preferences) sounds wonderful with
>> FreeSWITCH. To test, run two FreeSWITCH instances, both with mod_portaudio.
>> This also works well in 48khz conferences.
>>
>> I wouldn't use G.729 even if it weren't encumbered by patents - it's G.711,
>> G.722, G.722.1 and (my current favourite) Celt all the way.
Steve



From avi at avimarcus.net  Mon Sep 26 20:38:04 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 26 Sep 2011 19:38:04 +0300
Subject: [Freeswitch-users] Call Hangup with INCOMPATIBLE_DESTINATION
	cause
In-Reply-To: <CAO4LeU0FcQuinzgGqm7ztwH1w2YBjkpO-XFvCabqY3rq6b6Mwg@mail.gmail.com>
References: <CAO4LeU0FcQuinzgGqm7ztwH1w2YBjkpO-XFvCabqY3rq6b6Mwg@mail.gmail.com>
Message-ID: <CALNZuwnoufXdeB12eYE+6__MxijCR418+8o284EgLkF8a=+Twg@mail.gmail.com>

Not g729.. but g729B vs g729.
Supposedly this might work:
http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-bad-iananame

But best is setting the codec "name" on the.. linksys device? to simply
"g729".

-Avi


On Mon, Sep 26, 2011 at 7:30 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> Hi all,
> I have some calls dropped with INCOMPATIBLE_DESTINATION cause.
> I have checked the SDP of the bridged legA and legB and to me appear ok
> (both in G729).
> What I see in the log is:
>
> http://pastebin.freeswitch.org/17396
>
> Attached to this email an example of captured trace.
> Any advice on that?
>
> Stephen
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wstephen80 at gmail.com  Mon Sep 26 20:52:25 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 26 Sep 2011 18:52:25 +0200
Subject: [Freeswitch-users] Call Hangup with INCOMPATIBLE_DESTINATION
	cause
In-Reply-To: <CALNZuwnoufXdeB12eYE+6__MxijCR418+8o284EgLkF8a=+Twg@mail.gmail.com>
References: <CAO4LeU0FcQuinzgGqm7ztwH1w2YBjkpO-XFvCabqY3rq6b6Mwg@mail.gmail.com>
	<CALNZuwnoufXdeB12eYE+6__MxijCR418+8o284EgLkF8a=+Twg@mail.gmail.com>
Message-ID: <CAO4LeU3zf3wmyXgOxQbUN9Jk7auL35k3zbi+hXV34u5BAU+aag@mail.gmail.com>

Than you for the information.
So the problem is the "G729B" name?
I receive this SDP sometimes from one of my service provider so if this SDP
is wrong I prefer to ask him to solve the issue.
There are no "side effect" enabling NDLB-allow-bad-iananame?
Thanks in advance,

Stephen

On Mon, Sep 26, 2011 at 6:38 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Not g729.. but g729B vs g729.
> Supposedly this might work:
> http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-bad-iananame
>
> But best is setting the codec "name" on the.. linksys device? to simply
> "g729".
>
> -Avi
>
>
> On Mon, Sep 26, 2011 at 7:30 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> Hi all,
>> I have some calls dropped with INCOMPATIBLE_DESTINATION cause.
>> I have checked the SDP of the bridged legA and legB and to me appear ok
>> (both in G729).
>> What I see in the log is:
>>
>> http://pastebin.freeswitch.org/17396
>>
>> Attached to this email an example of captured trace.
>> Any advice on that?
>>
>> Stephen
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From michel.daggelinckx at gmail.com  Mon Sep 26 21:45:39 2011
From: michel.daggelinckx at gmail.com (Michel Daggelinckx)
Date: Mon, 26 Sep 2011 19:45:39 +0200
Subject: [Freeswitch-users] prepaid solutions
In-Reply-To: <2c4a73f4-b37a-4067-9818-d0bbfe26fded@zcs>
References: <b20533d5-ae5f-4f7a-b1e3-1e48f2cbeab5@zcs>
	<2c4a73f4-b37a-4067-9818-d0bbfe26fded@zcs>
Message-ID: <CAFYZzbtNCsoWe+z7k2iuM59acaxKXwjz+8Tv4DBx7kOORaw0=g@mail.gmail.com>

mod nibblebill and mod lcr is a good start


On Mon, Sep 26, 2011 at 1:59 AM, reza a <reza at lethalnetworks.com> wrote:

> Freeswitch users,
> Does anyone use freeswitch for prepaid applications? Do you have a walk
> through on setting up an environment like this? I'd like to play with
> freeswitch a bit, I know there alot of modules out there but I'm wondering
> what's the best cohesive set of modules/admin tools to use for a prepaid
> style applications.
> Thanks a lot
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nbhatti at gmail.com  Mon Sep 26 21:50:31 2011
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Mon, 26 Sep 2011 20:50:31 +0300
Subject: [Freeswitch-users] prepaid solutions
In-Reply-To: <CAFYZzbtNCsoWe+z7k2iuM59acaxKXwjz+8Tv4DBx7kOORaw0=g@mail.gmail.com>
References: <b20533d5-ae5f-4f7a-b1e3-1e48f2cbeab5@zcs>
	<2c4a73f4-b37a-4067-9818-d0bbfe26fded@zcs>
	<CAFYZzbtNCsoWe+z7k2iuM59acaxKXwjz+8Tv4DBx7kOORaw0=g@mail.gmail.com>
Message-ID: <CACak_yzYfL=o05sP-=bnEzGRdbYHzVF18atPsWMRD8njxaKQPQ@mail.gmail.com>

Open source solution. http://www.vbilling.org/


On Mon, Sep 26, 2011 at 8:45 PM, Michel Daggelinckx
<michel.daggelinckx at gmail.com> wrote:
> mod nibblebill and mod lcr is a good start
>
> On Mon, Sep 26, 2011 at 1:59 AM, reza a <reza at lethalnetworks.com> wrote:
>>
>> Freeswitch users,
>> Does anyone use freeswitch for prepaid applications? Do you have a walk
>> through on setting up an environment like this? I'd like to play with
>> freeswitch a bit, I know there alot of modules out there but I'm wondering
>> what's the best cohesive set of modules/admin tools to use for a prepaid
>> style applications.
>> Thanks a lot
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 michel.daggelinckx at gmail.com  Mon Sep 26 21:58:14 2011
From: michel.daggelinckx at gmail.com (Michel Daggelinckx)
Date: Mon, 26 Sep 2011 19:58:14 +0200
Subject: [Freeswitch-users] prepaid solutions
In-Reply-To: <CACak_yzYfL=o05sP-=bnEzGRdbYHzVF18atPsWMRD8njxaKQPQ@mail.gmail.com>
References: <b20533d5-ae5f-4f7a-b1e3-1e48f2cbeab5@zcs>
	<2c4a73f4-b37a-4067-9818-d0bbfe26fded@zcs>
	<CAFYZzbtNCsoWe+z7k2iuM59acaxKXwjz+8Tv4DBx7kOORaw0=g@mail.gmail.com>
	<CACak_yzYfL=o05sP-=bnEzGRdbYHzVF18atPsWMRD8njxaKQPQ@mail.gmail.com>
Message-ID: <CAFYZzbt=mmPGKEgAvNQ1cft_1Uez6LBV08LbAQR5y2_s825pRQ@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Billing

other billing solutions

On Mon, Sep 26, 2011 at 7:50 PM, Muhammad Naseer Bhatti
<nbhatti at gmail.com>wrote:

> Open source solution. http://www.vbilling.org/
>
>
> On Mon, Sep 26, 2011 at 8:45 PM, Michel Daggelinckx
> <michel.daggelinckx at gmail.com> wrote:
> > mod nibblebill and mod lcr is a good start
> >
> > On Mon, Sep 26, 2011 at 1:59 AM, reza a <reza at lethalnetworks.com> wrote:
> >>
> >> Freeswitch users,
> >> Does anyone use freeswitch for prepaid applications? Do you have a walk
> >> through on setting up an environment like this? I'd like to play with
> >> freeswitch a bit, I know there alot of modules out there but I'm
> wondering
> >> what's the best cohesive set of modules/admin tools to use for a prepaid
> >> style applications.
> >> Thanks a lot
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Mon Sep 26 22:06:17 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 26 Sep 2011 14:06:17 -0400
Subject: [Freeswitch-users] prepaid solutions
References: <b20533d5-ae5f-4f7a-b1e3-1e48f2cbeab5@zcs><2c4a73f4-b37a-4067-9818-d0bbfe26fded@zcs><CAFYZzbtNCsoWe+z7k2iuM59acaxKXwjz+8Tv4DBx7kOORaw0=g@mail.gmail.com>
	<CACak_yzYfL=o05sP-=bnEzGRdbYHzVF18atPsWMRD8njxaKQPQ@mail.gmail.com>
Message-ID: <18E2EE7F7C0C4E9380372B5100E3C7E1@e1705>

when I click on
http://vbilling.org/about-us/
-> server not found

maybe it would be cool to know who are behind this solution...


----- Original Message ----- 
From: "Muhammad Naseer Bhatti" <nbhatti at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, September 26, 2011 1:50 PM
Subject: Re: [Freeswitch-users] prepaid solutions


> Open source solution. http://www.vbilling.org/
>
>
> On Mon, Sep 26, 2011 at 8:45 PM, Michel Daggelinckx
> <michel.daggelinckx at gmail.com> wrote:
>> mod nibblebill and mod lcr is a good start
>>
>> On Mon, Sep 26, 2011 at 1:59 AM, reza a <reza at lethalnetworks.com> wrote:
>>>
>>> Freeswitch users,
>>> Does anyone use freeswitch for prepaid applications? Do you have a walk
>>> through on setting up an environment like this? I'd like to play with
>>> freeswitch a bit, I know there alot of modules out there but I'm 
>>> wondering
>>> what's the best cohesive set of modules/admin tools to use for a prepaid
>>> style applications.
>>> Thanks a lot
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 26 22:10:53 2011
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Mon, 26 Sep 2011 21:10:53 +0300
Subject: [Freeswitch-users] prepaid solutions
In-Reply-To: <18E2EE7F7C0C4E9380372B5100E3C7E1@e1705>
References: <b20533d5-ae5f-4f7a-b1e3-1e48f2cbeab5@zcs>
	<2c4a73f4-b37a-4067-9818-d0bbfe26fded@zcs>
	<CAFYZzbtNCsoWe+z7k2iuM59acaxKXwjz+8Tv4DBx7kOORaw0=g@mail.gmail.com>
	<CACak_yzYfL=o05sP-=bnEzGRdbYHzVF18atPsWMRD8njxaKQPQ@mail.gmail.com>
	<18E2EE7F7C0C4E9380372B5100E3C7E1@e1705>
Message-ID: <CACak_ywcv40pfLxpVhKeg1B6mzNW74PFptfJPdxVyaw6qGo+Wg@mail.gmail.com>

Check your connectivity. Works fine here. Verified from different locations too.

On Mon, Sep 26, 2011 at 9:06 PM, Madovsky <infos at madovsky.org> wrote:
> when I click on
> http://vbilling.org/about-us/
> -> server not found
>
> maybe it would be cool to know who are behind this solution...
>
>
> ----- Original Message -----
> From: "Muhammad Naseer Bhatti" <nbhatti at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, September 26, 2011 1:50 PM
> Subject: Re: [Freeswitch-users] prepaid solutions
>
>
>> Open source solution. http://www.vbilling.org/
>>
>>
>> On Mon, Sep 26, 2011 at 8:45 PM, Michel Daggelinckx
>> <michel.daggelinckx at gmail.com> wrote:
>>> mod nibblebill and mod lcr is a good start
>>>
>>> On Mon, Sep 26, 2011 at 1:59 AM, reza a <reza at lethalnetworks.com> wrote:
>>>>
>>>> Freeswitch users,
>>>> Does anyone use freeswitch for prepaid applications? Do you have a walk
>>>> through on setting up an environment like this? I'd like to play with
>>>> freeswitch a bit, I know there alot of modules out there but I'm
>>>> wondering
>>>> what's the best cohesive set of modules/admin tools to use for a prepaid
>>>> style applications.
>>>> Thanks a lot
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 cesar.bermudez at gmail.com  Mon Sep 26 22:12:59 2011
From: cesar.bermudez at gmail.com (Cesar Bermudez)
Date: Mon, 26 Sep 2011 12:12:59 -0600
Subject: [Freeswitch-users] prepaid solutions
In-Reply-To: <18E2EE7F7C0C4E9380372B5100E3C7E1@e1705>
References: <b20533d5-ae5f-4f7a-b1e3-1e48f2cbeab5@zcs>
	<2c4a73f4-b37a-4067-9818-d0bbfe26fded@zcs>
	<CAFYZzbtNCsoWe+z7k2iuM59acaxKXwjz+8Tv4DBx7kOORaw0=g@mail.gmail.com>
	<CACak_yzYfL=o05sP-=bnEzGRdbYHzVF18atPsWMRD8njxaKQPQ@mail.gmail.com>
	<18E2EE7F7C0C4E9380372B5100E3C7E1@e1705>
Message-ID: <CADrcvV-0bU-BduThP0-vOZ7LzJ+cmVq6a5RO49X8FUnSj+_h5w@mail.gmail.com>

works for me.

On Mon, Sep 26, 2011 at 12:06 PM, Madovsky <infos at madovsky.org> wrote:

> when I click on
> http://vbilling.org/about-us/
> -> server not found
>
> maybe it would be cool to know who are behind this solution...
>
>
> ----- Original Message -----
> From: "Muhammad Naseer Bhatti" <nbhatti at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, September 26, 2011 1:50 PM
> Subject: Re: [Freeswitch-users] prepaid solutions
>
>
> > Open source solution. http://www.vbilling.org/
> >
> >
> > On Mon, Sep 26, 2011 at 8:45 PM, Michel Daggelinckx
> > <michel.daggelinckx at gmail.com> wrote:
> >> mod nibblebill and mod lcr is a good start
> >>
> >> On Mon, Sep 26, 2011 at 1:59 AM, reza a <reza at lethalnetworks.com>
> wrote:
> >>>
> >>> Freeswitch users,
> >>> Does anyone use freeswitch for prepaid applications? Do you have a walk
> >>> through on setting up an environment like this? I'd like to play with
> >>> freeswitch a bit, I know there alot of modules out there but I'm
> >>> wondering
> >>> what's the best cohesive set of modules/admin tools to use for a
> prepaid
> >>> style applications.
> >>> Thanks a lot
> >>>
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From michel.daggelinckx at gmail.com  Mon Sep 26 22:14:09 2011
From: michel.daggelinckx at gmail.com (Michel Daggelinckx)
Date: Mon, 26 Sep 2011 20:14:09 +0200
Subject: [Freeswitch-users] prepaid solutions
In-Reply-To: <CACak_ywcv40pfLxpVhKeg1B6mzNW74PFptfJPdxVyaw6qGo+Wg@mail.gmail.com>
References: <b20533d5-ae5f-4f7a-b1e3-1e48f2cbeab5@zcs>
	<2c4a73f4-b37a-4067-9818-d0bbfe26fded@zcs>
	<CAFYZzbtNCsoWe+z7k2iuM59acaxKXwjz+8Tv4DBx7kOORaw0=g@mail.gmail.com>
	<CACak_yzYfL=o05sP-=bnEzGRdbYHzVF18atPsWMRD8njxaKQPQ@mail.gmail.com>
	<18E2EE7F7C0C4E9380372B5100E3C7E1@e1705>
	<CACak_ywcv40pfLxpVhKeg1B6mzNW74PFptfJPdxVyaw6qGo+Wg@mail.gmail.com>
Message-ID: <CAFYZzbtbGWLFW9KLVr2=GpBmSZzvROfcffDt6ERtwVSyP-+4kw@mail.gmail.com>

digitallinx is behind this

On Mon, Sep 26, 2011 at 8:10 PM, Muhammad Naseer Bhatti
<nbhatti at gmail.com>wrote:

> Check your connectivity. Works fine here. Verified from different locations
> too.
>
> On Mon, Sep 26, 2011 at 9:06 PM, Madovsky <infos at madovsky.org> wrote:
> > when I click on
> > http://vbilling.org/about-us/
> > -> server not found
> >
> > maybe it would be cool to know who are behind this solution...
> >
> >
> > ----- Original Message -----
> > From: "Muhammad Naseer Bhatti" <nbhatti at gmail.com>
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Sent: Monday, September 26, 2011 1:50 PM
> > Subject: Re: [Freeswitch-users] prepaid solutions
> >
> >
> >> Open source solution. http://www.vbilling.org/
> >>
> >>
> >> On Mon, Sep 26, 2011 at 8:45 PM, Michel Daggelinckx
> >> <michel.daggelinckx at gmail.com> wrote:
> >>> mod nibblebill and mod lcr is a good start
> >>>
> >>> On Mon, Sep 26, 2011 at 1:59 AM, reza a <reza at lethalnetworks.com>
> wrote:
> >>>>
> >>>> Freeswitch users,
> >>>> Does anyone use freeswitch for prepaid applications? Do you have a
> walk
> >>>> through on setting up an environment like this? I'd like to play with
> >>>> freeswitch a bit, I know there alot of modules out there but I'm
> >>>> wondering
> >>>> what's the best cohesive set of modules/admin tools to use for a
> prepaid
> >>>> style applications.
> >>>> Thanks a lot
> >>>>
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
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From infos at madovsky.org  Mon Sep 26 23:08:01 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 26 Sep 2011 15:08:01 -0400
Subject: [Freeswitch-users] prepaid solutions
References: <b20533d5-ae5f-4f7a-b1e3-1e48f2cbeab5@zcs><2c4a73f4-b37a-4067-9818-d0bbfe26fded@zcs><CAFYZzbtNCsoWe+z7k2iuM59acaxKXwjz+8Tv4DBx7kOORaw0=g@mail.gmail.com><CACak_yzYfL=o05sP-=bnEzGRdbYHzVF18atPsWMRD8njxaKQPQ@mail.gmail.com><18E2EE7F7C0C4E9380372B5100E3C7E1@e1705><CACak_ywcv40pfLxpVhKeg1B6mzNW74PFptfJPdxVyaw6qGo+Wg@mail.gmail.com>
	<CAFYZzbtbGWLFW9KLVr2=GpBmSZzvROfcffDt6ERtwVSyP-+4kw@mail.gmail.com>
Message-ID: <0344E3C7596C466F882CCE8292FC796A@e1705>

it works now...
  ----- Original Message ----- 
  From: Michel Daggelinckx 
  To: FreeSWITCH Users Help 
  Sent: Monday, September 26, 2011 2:14 PM
  Subject: Re: [Freeswitch-users] prepaid solutions


  digitallinx is behind this


  On Mon, Sep 26, 2011 at 8:10 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com> wrote:

    Check your connectivity. Works fine here. Verified from different locations too.


    On Mon, Sep 26, 2011 at 9:06 PM, Madovsky <infos at madovsky.org> wrote:
    > when I click on
    > http://vbilling.org/about-us/
    > -> server not found
    >
    > maybe it would be cool to know who are behind this solution...
    >
    >
    > ----- Original Message -----
    > From: "Muhammad Naseer Bhatti" <nbhatti at gmail.com>
    > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
    > Sent: Monday, September 26, 2011 1:50 PM
    > Subject: Re: [Freeswitch-users] prepaid solutions
    >
    >
    >> Open source solution. http://www.vbilling.org/
    >>
    >>
    >> On Mon, Sep 26, 2011 at 8:45 PM, Michel Daggelinckx
    >> <michel.daggelinckx at gmail.com> wrote:
    >>> mod nibblebill and mod lcr is a good start
    >>>
    >>> On Mon, Sep 26, 2011 at 1:59 AM, reza a <reza at lethalnetworks.com> wrote:
    >>>>
    >>>> Freeswitch users,
    >>>> Does anyone use freeswitch for prepaid applications? Do you have a walk
    >>>> through on setting up an environment like this? I'd like to play with
    >>>> freeswitch a bit, I know there alot of modules out there but I'm
    >>>> wondering
    >>>> what's the best cohesive set of modules/admin tools to use for a prepaid
    >>>> style applications.
    >>>> Thanks a lot
    >>>>
    >>>>
    >>>> FreeSWITCH-users mailing list
    >>>> FreeSWITCH-users at lists.freeswitch.org
    >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >>>> http://www.freeswitch.org
    >>>
    >>>
    >>>
    >>> FreeSWITCH-users mailing list
    >>> FreeSWITCH-users at lists.freeswitch.org
    >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >>> http://www.freeswitch.org
    >>>
    >>>
    >>
    >>
    >> FreeSWITCH-users mailing list
    >> FreeSWITCH-users at lists.freeswitch.org
    >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >> http://www.freeswitch.org
    >
    >
    >
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >


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





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



  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From aromberg at gmail.com  Mon Sep 26 10:38:53 2011
From: aromberg at gmail.com (aromberg at gmail.com)
Date: Mon, 26 Sep 2011 01:38:53 -0500
Subject: [Freeswitch-users] Mitel 52xx Broadsoft SLA/SCA Issues
Message-ID: <CAEC58iSKKamqXAq6pG3FxX8g6TvJwFXwbvZfAnh2JWTKM9OpvA@mail.gmail.com>

Here's the quick synopsis:  Freeswitch on CentOS 6 freshly installed 2
days ago. Freeswitch was grabbed via git.  Phones are Mitel 5220 SIP
phones that are in SIP mode.  Mitel has a Broadsoft implementation
called SIP_SCA in the device registration.  There is 1 Cisco 7960 but
its not configured at all for the shared extension.

I can't get it to work.  I *think* I have it narrowed down to the
presence ID being different on a per-phone basis.  The LEDS show the
correct call status (on-hook, ringing, on hold, etc).  However, when
you press the shared line from another phone, it dials the extension
again and goes straight to voicemail/busy instead of seizing the line.
All extensions are configured correctly (calling the shared extension
from the Cisco results in both phones ringing and able to pick up the
call initially)

Logfiles are located at:
http://iis.powertogrow.com/fs

Each one has the same actions:

Mitel phone 1 (192.168.1.106/extention 1002) calls Cisco Phone 2
(192.168.1.103/#1000) on Shared Line 1003.  Cisco answers.  Mitel
Phone 1 puts call on hold.  Mitel phone 2 (192.168.1.100/#1001) tries
to pick up Shared Line 1003 and then hangs up when it hears the
voicemail system.  It normally does this twice.  Then Mitel phone 1
picks up Shared Line 1003 and ends it.

No matter what order the steps are done with whatever phone (well,
except the Cisco doesn't have a shared line), the results are the
same.  Picking up the shared line results in a forward to the VM
system

Thanks,

Adam


From bnaylor at sirran.com  Mon Sep 26 15:07:09 2011
From: bnaylor at sirran.com (Ben Naylor)
Date: Mon, 26 Sep 2011 12:07:09 +0100
Subject: [Freeswitch-users] TLS issue
In-Reply-To: <CAKbxfG_mRH=7Or8=bRuixyYhDVHGiAGo37ovmShB4nocVoyrbg@mail.gmail.com>
References: <001e01cc7ab3$8f6f4330$ae4dc990$@sirran.com>
	<CAKbxfG_mRH=7Or8=bRuixyYhDVHGiAGo37ovmShB4nocVoyrbg@mail.gmail.com>
Message-ID: <005701cc7c3c$7044a290$50cde7b0$@sirran.com>

Hi Anthony

Thanks for the information, we have been using the suggested command today.

It appears that the BYE messages may not be reaching the called device
properly when using TLS.  When using UDP, the BYE messages are relayed to
the called party as normal.

One thing we did notice is that BYE messages from the calling client are
sent in the format '1002 at 192.168.250.18' , whereas the called client sends
BYE messages in the format 'mod_sofia at 192.168.250.18'.  However we assumed
this is normal behaviour.

We have also communicated these issues to the SIP client developer to see if
they can shed any light on this.

Here is a SIP trace for a call between the two clients -


freeswitch at localhost.localdomain> 
freeswitch at localhost.localdomain> recv 1053 bytes from
tls/[192.168.250.20]:49220 at 10:30:51.137867:
   ------------------------------------------------------------------------
   INVITE sip:1003 at 192.168.250.18 SIP/2.0
   Via: SIP/2.0/TLS
192.168.250.20:49220;branch=z9hG4bK3PhRphI56AiqPIBh;rport
   Contact: <sip:1002 at 192.168.250.20:49220;transport=tls>
   Max-Forwards: 70
   From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
   Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
   Supported: replaces, path
   User-Agent: Acrobits Softphone Business/1.8.8
   To: <sip:1003 at 192.168.250.18>
   Content-Type: application/sdp
   Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
   CSeq: 1 INVITE
   Proxy-Authorization: Digest
username="1002",realm="192.168.250.18",algorithm=MD5,uri="sip:1003 at 192.168.2
50.18",nonce="df6eed19-bd20-460a-b5ac-7e44946e729a",qop=auth,cnonce="4042a69
2f4f53c4479be6b2e765c37de",nc=00000005,response="bc47467ac4faac81bac59f37e4d
df7cb"
   Content-Length: 246
   
   v=0
   o=- 19191 16302 IN IP4 192.168.250.20
   s=epkccjl
   c=IN IP4 192.168.250.20
   t=0 0
   m=audio 17570 RTP/AVP 0 8 9 102 3 101
   a=rtpmap:101 telephone-event/8000
   a=rtpmap:102 ILBC/8000
   a=fmtp:102 mode=20
   a=fmtp:101 0-15
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 372 bytes to tls/[192.168.250.20]:49220 at 10:30:51.138777:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/TLS
192.168.250.20:49220;branch=z9hG4bK3PhRphI56AiqPIBh;rport=49220
   From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
   To: <sip:1003 at 192.168.250.18>
   Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-6724d7a 2011-09-23 01-25-43
+0000
   Content-Length: 0
   
   ------------------------------------------------------------------------
2011-09-26 11:30:51.132657 [NOTICE] switch_channel.c:911 New Channel
sofia/internal/1002 at 192.168.250.18 [f8c38b20-5662-4c82-b364-47b316f4fd13]
2011-09-26 11:30:51.172669 [NOTICE] switch_channel.c:911 New Channel
sofia/internal/sip:1003 at 192.168.250.11:49179
[439d5373-b000-4ed0-b299-4b4808f8ddda]
send 1252 bytes to tls/[192.168.250.11]:49179 at 10:30:51.179780:
   ------------------------------------------------------------------------
   INVITE sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls
SIP/2.0
   Via: SIP/2.0/TLS 192.168.250.18;branch=z9hG4bKyHm8ttBjB7Dmj
   Max-Forwards: 69
   From: "Extension 1002" <sip:1002 at 192.168.250.18>;tag=861g6FpZm4KHe
   To: <sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls>
   Call-ID: 729a5dfc-62cd-122f-2d9c-001cc071031a
   CSeq: 18180461 INVITE
   Contact: <sip:mod_sofia at 192.168.250.18:5061;transport=tls>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-6724d7a 2011-09-23 01-25-43
+0000
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 234
   X-FS-Support: update_display
   Remote-Party-ID: "Extension 1002"
<sip:1002 at 192.168.250.18>;party=calling;screen=yes;privacy=off
   
   v=0
   o=- 19191 16302 IN IP4 192.168.250.20
   s=epkccjl
   c=IN IP4 192.168.250.20
   t=0 0
   m=audio 17570 RTP/AVP 0 8 9 102 3 101
   a=rtpmap:102 ILBC/8000
   a=fmtp:102 mode=20
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:20
   ------------------------------------------------------------------------
recv 321 bytes from tls/[192.168.250.11]:49179 at 10:30:51.400065:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/TLS 192.168.250.18;branch=z9hG4bKyHm8ttBjB7Dmj;rport=5061
   From: "Extension 1002" <sip:1002 at 192.168.250.18>;tag=861g6FpZm4KHe
   Call-ID: 729a5dfc-62cd-122f-2d9c-001cc071031a
   CSeq: 18180461 INVITE
   To: <sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls>
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 499 bytes from tls/[192.168.250.11]:49179 at 10:30:51.417353:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/TLS 192.168.250.18;branch=z9hG4bKyHm8ttBjB7Dmj;rport=5061
   Contact: <sip:1003 at 192.168.250.11:49179;transport=tls>
   From: "Extension 1002" <sip:1002 at 192.168.250.18>;tag=861g6FpZm4KHe
   Call-ID: 729a5dfc-62cd-122f-2d9c-001cc071031a
   CSeq: 18180461 INVITE
   To:
<sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls>;tag=5A4771A
2806D8CDF36D39CDFEB4B0E50
   Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
   Supported: replaces, path
   Content-Length: 0
   
   ------------------------------------------------------------------------
2011-09-26 11:30:51.412650 [NOTICE] sofia.c:5235 Ring-Ready
sofia/internal/sip:1003 at 192.168.250.11:49179!
2011-09-26 11:30:51.412650 [NOTICE] mod_sofia.c:2391 Ring-Ready
sofia/internal/1002 at 192.168.250.18!
send 866 bytes to tls/[192.168.250.20]:49220 at 10:30:51.422883:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/TLS
192.168.250.20:49220;branch=z9hG4bK3PhRphI56AiqPIBh;rport=49220
   From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
   To: <sip:1003 at 192.168.250.18>;tag=7X8Q4m5UQUXyj
   Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
   CSeq: 1 INVITE
   Contact: <sip:1003 at 192.168.250.18:5061;transport=tls>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-6724d7a 2011-09-23 01-25-43
+0000
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   Remote-Party-ID: "Outbound Call"
<sip:1003 at 192.168.250.18>;party=calling;privacy=off;screen=no
   
   ------------------------------------------------------------------------
2011-09-26 11:30:51.412650 [NOTICE] switch_ivr_originate.c:481 Ring Ready
sofia/internal/1002 at 192.168.250.18!
recv 719 bytes from tls/[192.168.250.11]:49179 at 10:30:54.585965:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS 192.168.250.18;branch=z9hG4bKyHm8ttBjB7Dmj;rport=5061
   Contact: <sip:1003 at 192.168.250.11:49179;transport=tls>
   From: "Extension 1002" <sip:1002 at 192.168.250.18>;tag=861g6FpZm4KHe
   Call-ID: 729a5dfc-62cd-122f-2d9c-001cc071031a
   CSeq: 18180461 INVITE
   To:
<sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls>;tag=5A4771A
2806D8CDF36D39CDFEB4B0E50
   Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
   Supported: replaces, path
   Content-Type: application/sdp
   Content-Length: 192
   
   v=0
   o=- 41521 15327 IN IP4 192.168.250.11
   s=yqsmhzj
   c=IN IP4 192.168.250.11
   t=0 0
   m=audio 60032 RTP/AVP 0 101
   a=rtpmap:101 TELEPHONE-EVENT/8000
   a=fmtp:101 0-15
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 459 bytes to tls/[192.168.250.11]:49179 at 10:30:54.589056:
   ------------------------------------------------------------------------
   ACK sip:1003 at 192.168.250.11:49179;transport=tls SIP/2.0
   Via: SIP/2.0/TLS 192.168.250.18;branch=z9hG4bKZtD1vNvN8F46D
   Max-Forwards: 70
   From: "Extension 1002" <sip:1002 at 192.168.250.18>;tag=861g6FpZm4KHe
   To:
<sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls>;tag=5A4771A
2806D8CDF36D39CDFEB4B0E50
   Call-ID: 729a5dfc-62cd-122f-2d9c-001cc071031a
   CSeq: 18180461 ACK
   Contact: <sip:mod_sofia at 192.168.250.18:5061;transport=tls>
   Content-Length: 0
   
   ------------------------------------------------------------------------
2011-09-26 11:30:54.572557 [NOTICE] sofia.c:5802 Channel
[sofia/internal/sip:1003 at 192.168.250.11:49179] has been answered
2011-09-26 11:30:54.592567 [NOTICE] switch_ivr_originate.c:481 Ring Ready
sofia/internal/1002 at 192.168.250.18!
send 1079 bytes to tls/[192.168.250.20]:49220 at 10:30:54.596443:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS
192.168.250.20:49220;branch=z9hG4bK3PhRphI56AiqPIBh;rport=49220
   From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
   To: <sip:1003 at 192.168.250.18>;tag=7X8Q4m5UQUXyj
   Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
   CSeq: 1 INVITE
   Contact: <sip:1003 at 192.168.250.18:5061;transport=tls>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-6724d7a 2011-09-23 01-25-43
+0000
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 180
   Remote-Party-ID: "Outbound Call"
<sip:1003 at 192.168.250.18>;party=calling;privacy=off;screen=no
   
   v=02011-09-26 11:30:54.592567 [NOTICE] switch_ivr.c:706 Channel
[sofia/internal/1002 at 192.168.250.18] has been answered

   o=- 41521 15327 IN IP4 192.168.250.11
   s=yqsmhzj
   c=IN IP4 192.168.250.11
   t=0 0
   m=audio 60032 RTP/AVP 0 101
   a=rtpmap:101 TELEPHONE-EVENT/8000
   a=fmtp:101 0-15
   a=ptime:20
   ------------------------------------------------------------------------
recv 618 bytes from tls/[192.168.250.20]:49220 at 10:30:54.756293:
   ------------------------------------------------------------------------
   ACK sip:1003 at 192.168.250.18:5061;transport=tls SIP/2.0
   Via: SIP/2.0/TLS
192.168.250.20:49220;branch=z9hG4bK3PhRphI56AiqPIBh;rport
   Max-Forwards: 70
   To: <sip:1003 at 192.168.250.18>;tag=7X8Q4m5UQUXyj
   From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
   Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
   CSeq: 1 ACK
   Proxy-Authorization: Digest
username="1002",realm="192.168.250.18",algorithm=MD5,uri="sip:1003 at 192.168.2
50.18",nonce="df6eed19-bd20-460a-b5ac-7e44946e729a",qop=auth,cnonce="4042a69
2f4f53c4479be6b2e765c37de",nc=00000005,response="bc47467ac4faac81bac59f37e4d
df7cb"
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 825 bytes from tls/[192.168.250.20]:49221 at 10:31:00.397189:
   ------------------------------------------------------------------------
   BYE sip:1003 at 192.168.250.18:5061;transport=tls SIP/2.0
   Via: SIP/2.0/TCP
192.168.250.20:49221;branch=z9hG4bKLDntiDuXixOceTKx;rport
   Contact: <sips:1002 at 192.168.250.20:49221;transport=tcp>
   Max-Forwards: 70
   From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
   Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
   Supported: replaces, path
   User-Agent: Acrobits Softphone Business/1.8.8
   To: <sip:1003 at 192.168.250.18>;tag=7X8Q4m5UQUXyj
   Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
   CSeq: 2 BYE
   Proxy-Authorization: Digest
username="1002",realm="192.168.250.18",algorithm=MD5,uri="sip:1003 at 192.168.2
50.18:5061;transport=tls",nonce="df6eed19-bd20-460a-b5ac-7e44946e729a",qop=a
uth,cnonce="e2160338300c683de4a96cacce593376",nc=00000006,response="b1546d9b
c75688462874c2617671f1f7"
   Content-Length: 0
   
   ------------------------------------------------------------------------



Kind regards

Ben



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: 24 September 2011 19:59
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] TLS issue

you could try using the command

sofia global siptrace on

to see what you are getting.

On Sat, Sep 24, 2011 at 7:14 AM, Ben Naylor <bnaylor at sirran.com> wrote:
> Hello Freeswitch community!
>
>
>
> I was wondering if someone could help me with an issue I am 
> experiencing when using TLS to secure SIP traffic?
>
> Our current test environment consists of a Freeswitch test server, a 
> wireless access point, and 2 x iPhones using SIP client, all connected 
> to the same closed off VLAN.
>
>
>
> The Freeswitch server is set to bypass media=true, so that the RTP 
> traffic flows directly between the iPhones.? When testing a call using 
> standard SIP/RTP, the call is successful, and terminates correctly 
> when one of the devices hangs up the call.
>
>
>
> The problem occurs when using TLS, and trying to terminate the call.? 
> The call establishes successfully, but when either of the handsets 
> hangs up, the other remains connected to the call.? This happens when 
> hanging up the call from both the calling and the called handset.
>
>
>
> One thing I noticed when looking at the console output, is that two 
> messages appear along the lines of ?unknown Sip packet? (apologies for 
> the vagueness, I am not currently in front of the test server).? These 
> appear when the call is terminated on one of the hand sets.
>
>
>
> Has anyone had a similar experience when using TLS in this type of 
> scenario?
>
>
>
> Thanks for the help!
>
>
>
> Regards
>
>
>
> Ben
>
>
>
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>



--
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 f.erfurth at googlemail.com  Mon Sep 26 19:41:23 2011
From: f.erfurth at googlemail.com (f.erfurth)
Date: Mon, 26 Sep 2011 17:41:23 +0200
Subject: [Freeswitch-users] IAX
In-Reply-To: <CAHKqMb4B9cOodoAFnBSeODaJB-T921TuKZX5yD+V4Pk+X_mbZQ@mail.gmail.com>
References: <CANFMcCj9jqiWyM32caaR5qLAFizDhOrLbbpWS=_fFX30RJKUMQ@mail.gmail.com>
	<CAMWiMy_b4239c9CehXfe-qMEMam4tsCffWBjkokTAzExtD88DA@mail.gmail.com>
	<CAHKqMb4B9cOodoAFnBSeODaJB-T921TuKZX5yD+V4Pk+X_mbZQ@mail.gmail.com>
Message-ID: <CANFMcCjNEtWracv4RWJf78H-3z2bgBdL89SYZt1J=TdGhmDsCA@mail.gmail.com>

Hi

@Curriegrad: Thank you for your information.

@All:
Like Nandy I feel same about documentation of IAX. Could someone of you
provide some more information how to setup IAX for FreeSWITCH. I tried to
compile opal but got following error:

[CC] src/h460/h46018_h225.cxx
/home/f.erfurth/svn/opal/src/h460/h46018_h225.cxx: In member function ?void
H46019UDPSocket::BuildProbe(RTP_ControlFrame&, bool)?:
/home/f.erfurth/svn/opal/src/h460/h46018_h225.cxx:940: error:
?PMessageDigestSHA1? has not been declared
/home/f.erfurth/svn/opal/src/h460/h46018_h225.cxx: In member function
?PBoolean H46019UDPSocket::ReceivedProbePacket(const RTP_ControlFrame&,
bool&, bool&)?:
/home/f.erfurth/svn/opal/src/h460/h46018_h225.cxx:1144: error:
?PMessageDigestSHA1? has not been declared
make[1]: *** [/home/f.erfurth/svn/opal/lib_linux_x86/obj/h46018_h225.o]
Error 1
make[1]: Leaving directory `/home/f.erfurth/svn/opal'
make: *** [opt] Error 2

Well, where do I get PMessageDigestSHA1? Isn't that a perl module?

cu Floh

On Thu, Sep 22, 2011 at 5:51 AM, Nandy Dagondon <gcd at i.ph> wrote:

> how to do IAX is sketchy in the Wiki.  we need a more concrete example.
> tks.
> -nandy
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From marhbere at hotmail.com  Mon Sep 26 23:00:26 2011
From: marhbere at hotmail.com (MarhBere)
Date: Mon, 26 Sep 2011 12:00:26 -0700 (PDT)
Subject: [Freeswitch-users] Auto adjust Change port RTP
In-Reply-To: <1316982547007-6829940.post@n2.nabble.com>
References: <1316982547007-6829940.post@n2.nabble.com>
Message-ID: <1317063626867-6833057.post@n2.nabble.com>

Confirmed: This only happen if used Eyebeam. Not with "User-Agent:
Grandstream GXP285 1.2.5.3"

Regards,
MB

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Auto-adjust-Change-port-RTP-tp6829940p6833057.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Mon Sep 26 23:39:58 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 26 Sep 2011 14:39:58 -0500
Subject: [Freeswitch-users] Mitel 52xx Broadsoft SLA/SCA Issues
In-Reply-To: <CAEC58iSKKamqXAq6pG3FxX8g6TvJwFXwbvZfAnh2JWTKM9OpvA@mail.gmail.com>
References: <CAEC58iSKKamqXAq6pG3FxX8g6TvJwFXwbvZfAnh2JWTKM9OpvA@mail.gmail.com>
Message-ID: <CAKbxfG_zKf40FR6FnH5j7_qjGVAwyKOwE2oBZaYTeU8NGPzvOQ@mail.gmail.com>

you should not separate everything

you should do

sofia global siptrace on
console loglevel debug


Probably the barge in call is missing the replaces info or the correct
appearance-index in the call-info or possibly you are not using the
user/ channel in your dialplan which would mean you have to manually
set presence_id as done in the default.xml from the stock dialplan
template.




On Mon, Sep 26, 2011 at 1:38 AM,  <aromberg at gmail.com> wrote:
> Here's the quick synopsis: ?Freeswitch on CentOS 6 freshly installed 2
> days ago. Freeswitch was grabbed via git. ?Phones are Mitel 5220 SIP
> phones that are in SIP mode. ?Mitel has a Broadsoft implementation
> called SIP_SCA in the device registration. ?There is 1 Cisco 7960 but
> its not configured at all for the shared extension.
>
> I can't get it to work. ?I *think* I have it narrowed down to the
> presence ID being different on a per-phone basis. ?The LEDS show the
> correct call status (on-hook, ringing, on hold, etc). ?However, when
> you press the shared line from another phone, it dials the extension
> again and goes straight to voicemail/busy instead of seizing the line.
> All extensions are configured correctly (calling the shared extension
> from the Cisco results in both phones ringing and able to pick up the
> call initially)
>
> Logfiles are located at:
> http://iis.powertogrow.com/fs
>
> Each one has the same actions:
>
> Mitel phone 1 (192.168.1.106/extention 1002) calls Cisco Phone 2
> (192.168.1.103/#1000) on Shared Line 1003. ?Cisco answers. ?Mitel
> Phone 1 puts call on hold. ?Mitel phone 2 (192.168.1.100/#1001) tries
> to pick up Shared Line 1003 and then hangs up when it hears the
> voicemail system. ?It normally does this twice. ?Then Mitel phone 1
> picks up Shared Line 1003 and ends it.
>
> No matter what order the steps are done with whatever phone (well,
> except the Cisco doesn't have a shared line), the results are the
> same. ?Picking up the shared line results in a forward to the VM
> system
>
> Thanks,
>
> Adam
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From msc at freeswitch.org  Mon Sep 26 23:41:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Sep 2011 12:41:13 -0700
Subject: [Freeswitch-users] Call forward
In-Reply-To: <1316734409939-6822324.post@n2.nabble.com>
References: <1316734409939-6822324.post@n2.nabble.com>
Message-ID: <CAKzWOxXCGhK8OvvLuB2DYN+f2Z+4bKSDuMTBHGORaPiqtZiXgw@mail.gmail.com>

Can you pastebin the dialplan part that handles the callforward? It is not
properly building the bridge string. Look at line #196:
EXECUTE sofia/internal/4100 at 10.10.22.31 bridge(@10.10.22.31)
it's not getting the "4102" into that bridge app's argument. Show us the
dialplan and we can help you figure out what is happening.

-MC

On Thu, Sep 22, 2011 at 4:33 PM, baskar <yudha2008 at gmail.com> wrote:

> Hi All,
>
> I try to make call forward form once extension to another extension.
> Example:
> I have three extensions 4100, 4107, 4102
> Call forwarded from 4107 to 4102 but it directly go to voicemail. If i
> cancel call forward it directly reaches the extension.
> Can any one help to resolve this issue.
> I have paste my log in http://pastebin.freeswitch.org/17386.
>
> Thanks in advance.
>
> N.Baskar
>
>
>
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Call-forward-tp6822324p6822324.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From curriegrad2004 at gmail.com  Mon Sep 26 23:46:46 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 26 Sep 2011 12:46:46 -0700
Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re:
 Is there any license G729?
In-Reply-To: <4E80A921.8070600@coppice.org>
References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com>
	<4E80A921.8070600@coppice.org>
Message-ID: <CAMWiMy9dE72qi8ph8DqXpuFwtLHj4q+C0u0jPPV5HB1RegZQmg@mail.gmail.com>

Unless you want to spend loads of money on prototyping an VoIP IC that
does 48KHz, then be my guest and build one

On Mon, Sep 26, 2011 at 9:32 AM, Steve Underwood <steveu at coppice.org> wrote:
> On 06/30/2009 09:26 AM, Craig Askings wrote:
>> Are there any hardware phones that support 48 Khz Celt and
>> automated/mass deployment?
>>
>> Craig.
> Most of the pure VoIP chipsets aren't capable of running at 48k
> samples/second, so off the shelf phone hardware platforms probably won't
> cut it. The phones with video capabilities might have 48k sample/second
> facilities.
>> 2009/6/30 Jason White<jason at jasonjgw.net>:
>>> Brian West<brian at freeswitch.org> ?wrote:
>>>> Everyone has this need for lower bandwidth calls... I tend to march the
>>>> other way. 48kHz baby! ?(btw you can do 48kHz in the same bandwidth as a
>>>> single ulaw call)
>>> 48khz Celt (celt at 48000 in your codec preferences) sounds wonderful with
>>> FreeSWITCH. To test, run two FreeSWITCH instances, both with mod_portaudio.
>>> This also works well in 48khz conferences.
>>>
>>> I wouldn't use G.729 even if it weren't encumbered by patents - it's G.711,
>>> G.722, G.722.1 and (my current favourite) Celt all the way.
> Steve
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Mon Sep 26 23:52:43 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 26 Sep 2011 12:52:43 -0700
Subject: [Freeswitch-users] Call Hangup with INCOMPATIBLE_DESTINATION
	cause
In-Reply-To: <CAO4LeU3zf3wmyXgOxQbUN9Jk7auL35k3zbi+hXV34u5BAU+aag@mail.gmail.com>
References: <CAO4LeU0FcQuinzgGqm7ztwH1w2YBjkpO-XFvCabqY3rq6b6Mwg@mail.gmail.com>
	<CALNZuwnoufXdeB12eYE+6__MxijCR418+8o284EgLkF8a=+Twg@mail.gmail.com>
	<CAO4LeU3zf3wmyXgOxQbUN9Jk7auL35k3zbi+hXV34u5BAU+aag@mail.gmail.com>
Message-ID: <CAKzWOxVBLC0v6BQRh9disCvzPg+QP_V-FjES7_oNewoccr+yzQ@mail.gmail.com>

No side effects that any have reported. Still, we do recommend that he fix
the issue. I believe there is a separate way to specify annexb for g729.
Google reports this:

a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes

Hope that helps.
-MC

On Mon, Sep 26, 2011 at 9:52 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> Than you for the information.
> So the problem is the "G729B" name?
> I receive this SDP sometimes from one of my service provider so if this SDP
> is wrong I prefer to ask him to solve the issue.
> There are no "side effect" enabling NDLB-allow-bad-iananame?
> Thanks in advance,
>
> Stephen
>
>
> On Mon, Sep 26, 2011 at 6:38 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> Not g729.. but g729B vs g729.
>> Supposedly this might work:
>> http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-bad-iananame
>>
>> But best is setting the codec "name" on the.. linksys device? to simply
>> "g729".
>>
>> -Avi
>>
>>
>> On Mon, Sep 26, 2011 at 7:30 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>
>>> Hi all,
>>> I have some calls dropped with INCOMPATIBLE_DESTINATION cause.
>>> I have checked the SDP of the bridged legA and legB and to me appear ok
>>> (both in G729).
>>> What I see in the log is:
>>>
>>> http://pastebin.freeswitch.org/17396
>>>
>>> Attached to this email an example of captured trace.
>>> Any advice on that?
>>>
>>> Stephen
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wstephen80 at gmail.com  Tue Sep 27 01:01:52 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 26 Sep 2011 23:01:52 +0200
Subject: [Freeswitch-users] Call Hangup with INCOMPATIBLE_DESTINATION
	cause
In-Reply-To: <CAKzWOxVBLC0v6BQRh9disCvzPg+QP_V-FjES7_oNewoccr+yzQ@mail.gmail.com>
References: <CAO4LeU0FcQuinzgGqm7ztwH1w2YBjkpO-XFvCabqY3rq6b6Mwg@mail.gmail.com>
	<CALNZuwnoufXdeB12eYE+6__MxijCR418+8o284EgLkF8a=+Twg@mail.gmail.com>
	<CAO4LeU3zf3wmyXgOxQbUN9Jk7auL35k3zbi+hXV34u5BAU+aag@mail.gmail.com>
	<CAKzWOxVBLC0v6BQRh9disCvzPg+QP_V-FjES7_oNewoccr+yzQ@mail.gmail.com>
Message-ID: <CAO4LeU0+Z9qwiSB+-rv8o1bmU+oe8YhT_1hJRKPzGXsGk6_ydw@mail.gmail.com>

Ok, thank you, as you say is preferable to force to the standard way.

Stephen

On Mon, Sep 26, 2011 at 9:52 PM, Michael Collins <msc at freeswitch.org> wrote:

> No side effects that any have reported. Still, we do recommend that he fix
> the issue. I believe there is a separate way to specify annexb for g729.
> Google reports this:
>
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=yes
>
> Hope that helps.
> -MC
>
> On Mon, Sep 26, 2011 at 9:52 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> Than you for the information.
>> So the problem is the "G729B" name?
>> I receive this SDP sometimes from one of my service provider so if this
>> SDP is wrong I prefer to ask him to solve the issue.
>>  There are no "side effect" enabling NDLB-allow-bad-iananame?
>> Thanks in advance,
>>
>> Stephen
>>
>>
>> On Mon, Sep 26, 2011 at 6:38 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> Not g729.. but g729B vs g729.
>>> Supposedly this might work:
>>> http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-bad-iananame
>>>
>>> But best is setting the codec "name" on the.. linksys device? to simply
>>> "g729".
>>>
>>> -Avi
>>>
>>>
>>> On Mon, Sep 26, 2011 at 7:30 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>
>>>> Hi all,
>>>> I have some calls dropped with INCOMPATIBLE_DESTINATION cause.
>>>> I have checked the SDP of the bridged legA and legB and to me appear ok
>>>> (both in G729).
>>>> What I see in the log is:
>>>>
>>>> http://pastebin.freeswitch.org/17396
>>>>
>>>> Attached to this email an example of captured trace.
>>>> Any advice on that?
>>>>
>>>> Stephen
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brad at tech21.com  Tue Sep 27 01:34:32 2011
From: brad at tech21.com (Brad Mina)
Date: Mon, 26 Sep 2011 14:34:32 -0700
Subject: [Freeswitch-users] FSGUI
Message-ID: <CAGRr05_efk_i211d8eNBWm_2tDdVxRKzCjcHMwJg6ErRX8Uoxg@mail.gmail.com>

I've stumbled across fsgui a few times, (
http://wiki.freeswitch.org/wiki/Fsgui ), and am happy to report the windows
release of this program works flawlessly!

After toying with the windows version myself and a few other community
members wanted to see this application running in linux as well.
I downloaded the fsgui.tar.gz and unzipped it to /tmp then installed the
following packages on Ubuntu 10.04, which also installed related
dependencies:
libqt4-dev
libqt4-core

I tried to chmod+x and execute the application with ./fsgui - however bash
reports the following:

fsgui: cannot execute binary file


Is there something I'm missing here? I'm hoping jmesquita or a fellow
community member can respond with insight.

Thanks in advance.
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From kris at kriskinc.com  Tue Sep 27 06:25:34 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 26 Sep 2011 22:25:34 -0400
Subject: [Freeswitch-users] TLS issue
In-Reply-To: <005701cc7c3c$7044a290$50cde7b0$@sirran.com>
References: <001e01cc7ab3$8f6f4330$ae4dc990$@sirran.com>
	<CAKbxfG_mRH=7Or8=bRuixyYhDVHGiAGo37ovmShB4nocVoyrbg@mail.gmail.com>
	<005701cc7c3c$7044a290$50cde7b0$@sirran.com>
Message-ID: <CAKDfjgct7tp7k_Pmom8mTcz1Hq+k2xsvxOC6BV6izuzNqkVyVQ@mail.gmail.com>

Ben,

  This SIP trace looks pretty clean except for the final BYE (big
surprise).  Especially the Contact: field generated by 192.168.250.20
and the IP:port used by the client:

Contact: <sips:1002 at 192.168.250.20:49221;transport=tcp>

  You'll notice the Contact field in the other packets uses the SIP
URI scheme with a SIP transport=tls URI parameter:

Contact: <sip:1002 at 192.168.250.20:49220;transport=tls>

  Either URI scheme/format is considered "compliant" depending on who
you talk to, which implementation you're using, etc, etc.  It's SIP
;).   I'm almost completely certain that changing the URI scheme mid
call, however, is not compliant.  This seems to be a bug in the
Acrobits client.

  Another concern - the other packets seem to come from
192.168.250.20:49221, not 192.168.250.20:49220.  Now your client is:

1)  Switching the URI scheme and parameters mid-call.
2)  Incrementing the source TCP port number (for some reason).

  Either one (certainly both) of these conditions will most likely
cause the received BYE to not be recognized by FreeSWITCH.

On Mon, Sep 26, 2011 at 7:07 AM, Ben Naylor <bnaylor at sirran.com> wrote:
> Hi Anthony
>
> Thanks for the information, we have been using the suggested command today.
>
> It appears that the BYE messages may not be reaching the called device
> properly when using TLS. ?When using UDP, the BYE messages are relayed to
> the called party as normal.
>
> One thing we did notice is that BYE messages from the calling client are
> sent in the format '1002 at 192.168.250.18' , whereas the called client sends
> BYE messages in the format 'mod_sofia at 192.168.250.18'. ?However we assumed
> this is normal behaviour.
>
> We have also communicated these issues to the SIP client developer to see if
> they can shed any light on this.
>
> Here is a SIP trace for a call between the two clients -
>
>
> freeswitch at localhost.localdomain>
> freeswitch at localhost.localdomain> recv 1053 bytes from
> tls/[192.168.250.20]:49220 at 10:30:51.137867:
> ? ------------------------------------------------------------------------
> ? INVITE sip:1003 at 192.168.250.18 SIP/2.0
> ? Via: SIP/2.0/TLS
> 192.168.250.20:49220;branch=z9hG4bK3PhRphI56AiqPIBh;rport
> ? Contact: <sip:1002 at 192.168.250.20:49220;transport=tls>
> ? Max-Forwards: 70
> ? From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
> ? Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
> ? Supported: replaces, path
> ? User-Agent: Acrobits Softphone Business/1.8.8
> ? To: <sip:1003 at 192.168.250.18>
> ? Content-Type: application/sdp
> ? Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
> ? CSeq: 1 INVITE
> ? Proxy-Authorization: Digest
> username="1002",realm="192.168.250.18",algorithm=MD5,uri="sip:1003 at 192.168.2
> 50.18",nonce="df6eed19-bd20-460a-b5ac-7e44946e729a",qop=auth,cnonce="4042a69
> 2f4f53c4479be6b2e765c37de",nc=00000005,response="bc47467ac4faac81bac59f37e4d
> df7cb"
> ? Content-Length: 246
>
> ? v=0
> ? o=- 19191 16302 IN IP4 192.168.250.20
> ? s=epkccjl
> ? c=IN IP4 192.168.250.20
> ? t=0 0
> ? m=audio 17570 RTP/AVP 0 8 9 102 3 101
> ? a=rtpmap:101 telephone-event/8000
> ? a=rtpmap:102 ILBC/8000
> ? a=fmtp:102 mode=20
> ? a=fmtp:101 0-15
> ? a=ptime:20
> ? a=sendrecv
> ? ------------------------------------------------------------------------
> send 372 bytes to tls/[192.168.250.20]:49220 at 10:30:51.138777:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 100 Trying
> ? Via: SIP/2.0/TLS
> 192.168.250.20:49220;branch=z9hG4bK3PhRphI56AiqPIBh;rport=49220
> ? From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
> ? To: <sip:1003 at 192.168.250.18>
> ? Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
> ? CSeq: 1 INVITE
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-6724d7a 2011-09-23 01-25-43
> +0000
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
> 2011-09-26 11:30:51.132657 [NOTICE] switch_channel.c:911 New Channel
> sofia/internal/1002 at 192.168.250.18 [f8c38b20-5662-4c82-b364-47b316f4fd13]
> 2011-09-26 11:30:51.172669 [NOTICE] switch_channel.c:911 New Channel
> sofia/internal/sip:1003 at 192.168.250.11:49179
> [439d5373-b000-4ed0-b299-4b4808f8ddda]
> send 1252 bytes to tls/[192.168.250.11]:49179 at 10:30:51.179780:
> ? ------------------------------------------------------------------------
> ? INVITE sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls
> SIP/2.0
> ? Via: SIP/2.0/TLS 192.168.250.18;branch=z9hG4bKyHm8ttBjB7Dmj
> ? Max-Forwards: 69
> ? From: "Extension 1002" <sip:1002 at 192.168.250.18>;tag=861g6FpZm4KHe
> ? To: <sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls>
> ? Call-ID: 729a5dfc-62cd-122f-2d9c-001cc071031a
> ? CSeq: 18180461 INVITE
> ? Contact: <sip:mod_sofia at 192.168.250.18:5061;transport=tls>
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-6724d7a 2011-09-23 01-25-43
> +0000
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> ? Supported: timer, precondition, path, replaces
> ? Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> ? Content-Type: application/sdp
> ? Content-Disposition: session
> ? Content-Length: 234
> ? X-FS-Support: update_display
> ? Remote-Party-ID: "Extension 1002"
> <sip:1002 at 192.168.250.18>;party=calling;screen=yes;privacy=off
>
> ? v=0
> ? o=- 19191 16302 IN IP4 192.168.250.20
> ? s=epkccjl
> ? c=IN IP4 192.168.250.20
> ? t=0 0
> ? m=audio 17570 RTP/AVP 0 8 9 102 3 101
> ? a=rtpmap:102 ILBC/8000
> ? a=fmtp:102 mode=20
> ? a=rtpmap:101 telephone-event/8000
> ? a=fmtp:101 0-15
> ? a=ptime:20
> ? ------------------------------------------------------------------------
> recv 321 bytes from tls/[192.168.250.11]:49179 at 10:30:51.400065:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 100 Trying
> ? Via: SIP/2.0/TLS 192.168.250.18;branch=z9hG4bKyHm8ttBjB7Dmj;rport=5061
> ? From: "Extension 1002" <sip:1002 at 192.168.250.18>;tag=861g6FpZm4KHe
> ? Call-ID: 729a5dfc-62cd-122f-2d9c-001cc071031a
> ? CSeq: 18180461 INVITE
> ? To: <sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls>
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
> recv 499 bytes from tls/[192.168.250.11]:49179 at 10:30:51.417353:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 180 Ringing
> ? Via: SIP/2.0/TLS 192.168.250.18;branch=z9hG4bKyHm8ttBjB7Dmj;rport=5061
> ? Contact: <sip:1003 at 192.168.250.11:49179;transport=tls>
> ? From: "Extension 1002" <sip:1002 at 192.168.250.18>;tag=861g6FpZm4KHe
> ? Call-ID: 729a5dfc-62cd-122f-2d9c-001cc071031a
> ? CSeq: 18180461 INVITE
> ? To:
> <sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls>;tag=5A4771A
> 2806D8CDF36D39CDFEB4B0E50
> ? Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
> ? Supported: replaces, path
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
> 2011-09-26 11:30:51.412650 [NOTICE] sofia.c:5235 Ring-Ready
> sofia/internal/sip:1003 at 192.168.250.11:49179!
> 2011-09-26 11:30:51.412650 [NOTICE] mod_sofia.c:2391 Ring-Ready
> sofia/internal/1002 at 192.168.250.18!
> send 866 bytes to tls/[192.168.250.20]:49220 at 10:30:51.422883:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 180 Ringing
> ? Via: SIP/2.0/TLS
> 192.168.250.20:49220;branch=z9hG4bK3PhRphI56AiqPIBh;rport=49220
> ? From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
> ? To: <sip:1003 at 192.168.250.18>;tag=7X8Q4m5UQUXyj
> ? Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
> ? CSeq: 1 INVITE
> ? Contact: <sip:1003 at 192.168.250.18:5061;transport=tls>
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-6724d7a 2011-09-23 01-25-43
> +0000
> ? Accept: application/sdp
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> ? Supported: timer, precondition, path, replaces
> ? Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> ? Content-Length: 0
> ? Remote-Party-ID: "Outbound Call"
> <sip:1003 at 192.168.250.18>;party=calling;privacy=off;screen=no
>
> ? ------------------------------------------------------------------------
> 2011-09-26 11:30:51.412650 [NOTICE] switch_ivr_originate.c:481 Ring Ready
> sofia/internal/1002 at 192.168.250.18!
> recv 719 bytes from tls/[192.168.250.11]:49179 at 10:30:54.585965:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 200 OK
> ? Via: SIP/2.0/TLS 192.168.250.18;branch=z9hG4bKyHm8ttBjB7Dmj;rport=5061
> ? Contact: <sip:1003 at 192.168.250.11:49179;transport=tls>
> ? From: "Extension 1002" <sip:1002 at 192.168.250.18>;tag=861g6FpZm4KHe
> ? Call-ID: 729a5dfc-62cd-122f-2d9c-001cc071031a
> ? CSeq: 18180461 INVITE
> ? To:
> <sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls>;tag=5A4771A
> 2806D8CDF36D39CDFEB4B0E50
> ? Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
> ? Supported: replaces, path
> ? Content-Type: application/sdp
> ? Content-Length: 192
>
> ? v=0
> ? o=- 41521 15327 IN IP4 192.168.250.11
> ? s=yqsmhzj
> ? c=IN IP4 192.168.250.11
> ? t=0 0
> ? m=audio 60032 RTP/AVP 0 101
> ? a=rtpmap:101 TELEPHONE-EVENT/8000
> ? a=fmtp:101 0-15
> ? a=ptime:20
> ? a=sendrecv
> ? ------------------------------------------------------------------------
> send 459 bytes to tls/[192.168.250.11]:49179 at 10:30:54.589056:
> ? ------------------------------------------------------------------------
> ? ACK sip:1003 at 192.168.250.11:49179;transport=tls SIP/2.0
> ? Via: SIP/2.0/TLS 192.168.250.18;branch=z9hG4bKZtD1vNvN8F46D
> ? Max-Forwards: 70
> ? From: "Extension 1002" <sip:1002 at 192.168.250.18>;tag=861g6FpZm4KHe
> ? To:
> <sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls>;tag=5A4771A
> 2806D8CDF36D39CDFEB4B0E50
> ? Call-ID: 729a5dfc-62cd-122f-2d9c-001cc071031a
> ? CSeq: 18180461 ACK
> ? Contact: <sip:mod_sofia at 192.168.250.18:5061;transport=tls>
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
> 2011-09-26 11:30:54.572557 [NOTICE] sofia.c:5802 Channel
> [sofia/internal/sip:1003 at 192.168.250.11:49179] has been answered
> 2011-09-26 11:30:54.592567 [NOTICE] switch_ivr_originate.c:481 Ring Ready
> sofia/internal/1002 at 192.168.250.18!
> send 1079 bytes to tls/[192.168.250.20]:49220 at 10:30:54.596443:
> ? ------------------------------------------------------------------------
> ? SIP/2.0 200 OK
> ? Via: SIP/2.0/TLS
> 192.168.250.20:49220;branch=z9hG4bK3PhRphI56AiqPIBh;rport=49220
> ? From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
> ? To: <sip:1003 at 192.168.250.18>;tag=7X8Q4m5UQUXyj
> ? Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
> ? CSeq: 1 INVITE
> ? Contact: <sip:1003 at 192.168.250.18:5061;transport=tls>
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-6724d7a 2011-09-23 01-25-43
> +0000
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> ? Supported: timer, precondition, path, replaces
> ? Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> ? Content-Type: application/sdp
> ? Content-Disposition: session
> ? Content-Length: 180
> ? Remote-Party-ID: "Outbound Call"
> <sip:1003 at 192.168.250.18>;party=calling;privacy=off;screen=no
>
> ? v=02011-09-26 11:30:54.592567 [NOTICE] switch_ivr.c:706 Channel
> [sofia/internal/1002 at 192.168.250.18] has been answered
>
> ? o=- 41521 15327 IN IP4 192.168.250.11
> ? s=yqsmhzj
> ? c=IN IP4 192.168.250.11
> ? t=0 0
> ? m=audio 60032 RTP/AVP 0 101
> ? a=rtpmap:101 TELEPHONE-EVENT/8000
> ? a=fmtp:101 0-15
> ? a=ptime:20
> ? ------------------------------------------------------------------------
> recv 618 bytes from tls/[192.168.250.20]:49220 at 10:30:54.756293:
> ? ------------------------------------------------------------------------
> ? ACK sip:1003 at 192.168.250.18:5061;transport=tls SIP/2.0
> ? Via: SIP/2.0/TLS
> 192.168.250.20:49220;branch=z9hG4bK3PhRphI56AiqPIBh;rport
> ? Max-Forwards: 70
> ? To: <sip:1003 at 192.168.250.18>;tag=7X8Q4m5UQUXyj
> ? From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
> ? Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
> ? CSeq: 1 ACK
> ? Proxy-Authorization: Digest
> username="1002",realm="192.168.250.18",algorithm=MD5,uri="sip:1003 at 192.168.2
> 50.18",nonce="df6eed19-bd20-460a-b5ac-7e44946e729a",qop=auth,cnonce="4042a69
> 2f4f53c4479be6b2e765c37de",nc=00000005,response="bc47467ac4faac81bac59f37e4d
> df7cb"
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
> recv 825 bytes from tls/[192.168.250.20]:49221 at 10:31:00.397189:
> ? ------------------------------------------------------------------------
> ? BYE sip:1003 at 192.168.250.18:5061;transport=tls SIP/2.0
> ? Via: SIP/2.0/TCP
> 192.168.250.20:49221;branch=z9hG4bKLDntiDuXixOceTKx;rport
> ? Contact: <sips:1002 at 192.168.250.20:49221;transport=tcp>
> ? Max-Forwards: 70
> ? From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
> ? Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
> ? Supported: replaces, path
> ? User-Agent: Acrobits Softphone Business/1.8.8
> ? To: <sip:1003 at 192.168.250.18>;tag=7X8Q4m5UQUXyj
> ? Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
> ? CSeq: 2 BYE
> ? Proxy-Authorization: Digest
> username="1002",realm="192.168.250.18",algorithm=MD5,uri="sip:1003 at 192.168.2
> 50.18:5061;transport=tls",nonce="df6eed19-bd20-460a-b5ac-7e44946e729a",qop=a
> uth,cnonce="e2160338300c683de4a96cacce593376",nc=00000006,response="b1546d9b
> c75688462874c2617671f1f7"
> ? Content-Length: 0
>
> ? ------------------------------------------------------------------------
>
>
>
> Kind regards
>
> Ben
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
> Minessale
> Sent: 24 September 2011 19:59
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] TLS issue
>
> you could try using the command
>
> sofia global siptrace on
>
> to see what you are getting.
>
> On Sat, Sep 24, 2011 at 7:14 AM, Ben Naylor <bnaylor at sirran.com> wrote:
>> Hello Freeswitch community!
>>
>>
>>
>> I was wondering if someone could help me with an issue I am
>> experiencing when using TLS to secure SIP traffic?
>>
>> Our current test environment consists of a Freeswitch test server, a
>> wireless access point, and 2 x iPhones using SIP client, all connected
>> to the same closed off VLAN.
>>
>>
>>
>> The Freeswitch server is set to bypass media=true, so that the RTP
>> traffic flows directly between the iPhones.? When testing a call using
>> standard SIP/RTP, the call is successful, and terminates correctly
>> when one of the devices hangs up the call.
>>
>>
>>
>> The problem occurs when using TLS, and trying to terminate the call.
>> The call establishes successfully, but when either of the handsets
>> hangs up, the other remains connected to the call.? This happens when
>> hanging up the call from both the calling and the called handset.
>>
>>
>>
>> One thing I noticed when looking at the console output, is that two
>> messages appear along the lines of ?unknown Sip packet? (apologies for
>> the vagueness, I am not currently in front of the test server).? These
>> appear when the call is terminated on one of the hand sets.
>>
>>
>>
>> Has anyone had a similar experience when using TLS in this type of
>> scenario?
>>
>>
>>
>> Thanks for the help!
>>
>>
>>
>> Regards
>>
>>
>>
>> Ben
>>
>>
>>
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> 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
>



-- 
Kristian Kielhofner


From aromberg at gmail.com  Tue Sep 27 08:03:56 2011
From: aromberg at gmail.com (aromberg at gmail.com)
Date: Mon, 26 Sep 2011 23:03:56 -0500
Subject: [Freeswitch-users] Mitel 52xx Broadsoft SLA/SCA Issues
In-Reply-To: <CAKbxfG_zKf40FR6FnH5j7_qjGVAwyKOwE2oBZaYTeU8NGPzvOQ@mail.gmail.com>
References: <CAEC58iSKKamqXAq6pG3FxX8g6TvJwFXwbvZfAnh2JWTKM9OpvA@mail.gmail.com>
	<CAKbxfG_zKf40FR6FnH5j7_qjGVAwyKOwE2oBZaYTeU8NGPzvOQ@mail.gmail.com>
Message-ID: <CAEC58iSD2C-rAUjxsEfrD4Z1=8ytbasiBza3nJ13tE8FEOU0eQ@mail.gmail.com>

Alright, I have a new logfile dump in pastebin
http://pastebin.freeswitch.org/17403

This timeline goes a little different... Mitel 1 picks up the Shared
Line, calls the echo test, puts it on hold, and Mitel 2 tries to pick
up the shared line but gets voicemail.

I've compared the call-info line and they seem to match.... and in the
default dialplan the extension is defined... otherwise how am I
getting the voicemail?

Line 950 is where the 2nd phone tries to pick up.... as you can see
after that it has the correct appearance-id and call-info:

Line 1065 looks like Freeswitch is making a new call instead of
capturing the old one.

Hope this is of some help

Thanks,

Adam





On Mon, Sep 26, 2011 at 2:39 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Probably the barge in call is missing the replaces info or the correct
> appearance-index in the call-info or possibly you are not using the
> user/ channel in your dialplan which would mean you have to manually
> set presence_id as done in the default.xml from the stock dialplan
> template.
>
>
>
>
> On Mon, Sep 26, 2011 at 1:38 AM, ?<aromberg at gmail.com> wrote:
>> Here's the quick synopsis: ?Freeswitch on CentOS 6 freshly installed 2
>> days ago. Freeswitch was grabbed via git. ?Phones are Mitel 5220 SIP
>> phones that are in SIP mode. ?Mitel has a Broadsoft implementation
>> called SIP_SCA in the device registration. ?There is 1 Cisco 7960 but
>> its not configured at all for the shared extension.
>>
>> I can't get it to work. ?I *think* I have it narrowed down to the
>> presence ID being different on a per-phone basis. ?The LEDS show the
>> correct call status (on-hook, ringing, on hold, etc). ?However, when
>> you press the shared line from another phone, it dials the extension
>> again and goes straight to voicemail/busy instead of seizing the line.
>> All extensions are configured correctly (calling the shared extension
>> from the Cisco results in both phones ringing and able to pick up the
>> call initially)
>>
>> Logfiles are located at:
>> http://iis.powertogrow.com/fs
>>
>> Each one has the same actions:
>>
>> Mitel phone 1 (192.168.1.106/extention 1002) calls Cisco Phone 2
>> (192.168.1.103/#1000) on Shared Line 1003. ?Cisco answers. ?Mitel
>> Phone 1 puts call on hold. ?Mitel phone 2 (192.168.1.100/#1001) tries
>> to pick up Shared Line 1003 and then hangs up when it hears the
>> voicemail system. ?It normally does this twice. ?Then Mitel phone 1
>> picks up Shared Line 1003 and ends it.
>>
>> No matter what order the steps are done with whatever phone (well,
>> except the Cisco doesn't have a shared line), the results are the
>> same. ?Picking up the shared line results in a forward to the VM
>> system
>>
>> Thanks,
>>
>> Adam


From bnaylor at sirran.com  Tue Sep 27 12:59:43 2011
From: bnaylor at sirran.com (Ben Naylor)
Date: Tue, 27 Sep 2011 09:59:43 +0100
Subject: [Freeswitch-users] TLS issue
In-Reply-To: <CAKDfjgct7tp7k_Pmom8mTcz1Hq+k2xsvxOC6BV6izuzNqkVyVQ@mail.gmail.com>
References: <001e01cc7ab3$8f6f4330$ae4dc990$@sirran.com>	<CAKbxfG_mRH=7Or8=bRuixyYhDVHGiAGo37ovmShB4nocVoyrbg@mail.gmail.com>	<005701cc7c3c$7044a290$50cde7b0$@sirran.com>
	<CAKDfjgct7tp7k_Pmom8mTcz1Hq+k2xsvxOC6BV6izuzNqkVyVQ@mail.gmail.com>
Message-ID: <000e01cc7cf3$cd066fb0$67134f10$@sirran.com>

Hi Kristian

Many thanks for the information.

We did end up spotting the transport protocol change eventually, this has
been highlighted to Acrobits who say they will investigate for us.

However we didn?t spot the incrementing port number issue, so thanks for
pointing that one out!

I will drop a quick e-mail once we get it all up and running, thanks again.

Cheers

Ben

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
Kielhofner
Sent: 27 September 2011 03:26
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] TLS issue

Ben,

  This SIP trace looks pretty clean except for the final BYE (big surprise).
Especially the Contact: field generated by 192.168.250.20 and the IP:port
used by the client:

Contact: <sips:1002 at 192.168.250.20:49221;transport=tcp>

  You'll notice the Contact field in the other packets uses the SIP URI
scheme with a SIP transport=tls URI parameter:

Contact: <sip:1002 at 192.168.250.20:49220;transport=tls>

  Either URI scheme/format is considered "compliant" depending on who you
talk to, which implementation you're using, etc, etc.  It's SIP
;).   I'm almost completely certain that changing the URI scheme mid
call, however, is not compliant.  This seems to be a bug in the Acrobits
client.

  Another concern - the other packets seem to come from
192.168.250.20:49221, not 192.168.250.20:49220.  Now your client is:

1)  Switching the URI scheme and parameters mid-call.
2)  Incrementing the source TCP port number (for some reason).

  Either one (certainly both) of these conditions will most likely cause the
received BYE to not be recognized by FreeSWITCH.

On Mon, Sep 26, 2011 at 7:07 AM, Ben Naylor <bnaylor at sirran.com> wrote:
> Hi Anthony
>
> Thanks for the information, we have been using the suggested command
today.
>
> It appears that the BYE messages may not be reaching the called device 
> properly when using TLS. ?When using UDP, the BYE messages are relayed 
> to the called party as normal.
>
> One thing we did notice is that BYE messages from the calling client 
> are sent in the format '1002 at 192.168.250.18' , whereas the called 
> client sends BYE messages in the format 'mod_sofia at 192.168.250.18'. ?
> However we assumed this is normal behaviour.
>
> We have also communicated these issues to the SIP client developer to 
> see if they can shed any light on this.
>
> Here is a SIP trace for a call between the two clients -
>
>
> freeswitch at localhost.localdomain>
> freeswitch at localhost.localdomain> recv 1053 bytes from
> tls/[192.168.250.20]:49220 at 10:30:51.137867:
> ? 
> ----------------------------------------------------------------------
> --
> ? INVITE sip:1003 at 192.168.250.18 SIP/2.0
> ? Via: SIP/2.0/TLS
> 192.168.250.20:49220;branch=z9hG4bK3PhRphI56AiqPIBh;rport
> ? Contact: <sip:1002 at 192.168.250.20:49220;transport=tls>
> ? Max-Forwards: 70
> ? From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
> ? Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
> ? Supported: replaces, path
> ? User-Agent: Acrobits Softphone Business/1.8.8
> ? To: <sip:1003 at 192.168.250.18>
> ? Content-Type: application/sdp
> ? Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
> ? CSeq: 1 INVITE
> ? Proxy-Authorization: Digest
> username="1002",realm="192.168.250.18",algorithm=MD5,uri="sip:1003 at 192
> .168.2
> 50.18",nonce="df6eed19-bd20-460a-b5ac-7e44946e729a",qop=auth,cnonce="4
> 042a69 
> 2f4f53c4479be6b2e765c37de",nc=00000005,response="bc47467ac4faac81bac59
> f37e4d
> df7cb"
> ? Content-Length: 246
>
> ? v=0
> ? o=- 19191 16302 IN IP4 192.168.250.20
> ? s=epkccjl
> ? c=IN IP4 192.168.250.20
> ? t=0 0
> ? m=audio 17570 RTP/AVP 0 8 9 102 3 101
> ? a=rtpmap:101 telephone-event/8000
> ? a=rtpmap:102 ILBC/8000
> ? a=fmtp:102 mode=20
> ? a=fmtp:101 0-15
> ? a=ptime:20
> ? a=sendrecv
> ? 
> ----------------------------------------------------------------------
> -- send 372 bytes to tls/[192.168.250.20]:49220 at 10:30:51.138777:
> ? 
> ----------------------------------------------------------------------
> --
> ? SIP/2.0 100 Trying
> ? Via: SIP/2.0/TLS
> 192.168.250.20:49220;branch=z9hG4bK3PhRphI56AiqPIBh;rport=49220
> ? From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
> ? To: <sip:1003 at 192.168.250.18>
> ? Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
> ? CSeq: 1 INVITE
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-6724d7a 2011-09-23 
> 01-25-43
> +0000
> ? Content-Length: 0
>
> ? 
> ----------------------------------------------------------------------
> --
> 2011-09-26 11:30:51.132657 [NOTICE] switch_channel.c:911 New Channel
> sofia/internal/1002 at 192.168.250.18 
> [f8c38b20-5662-4c82-b364-47b316f4fd13]
> 2011-09-26 11:30:51.172669 [NOTICE] switch_channel.c:911 New Channel
> sofia/internal/sip:1003 at 192.168.250.11:49179
> [439d5373-b000-4ed0-b299-4b4808f8ddda]
> send 1252 bytes to tls/[192.168.250.11]:49179 at 10:30:51.179780:
> ? 
> ----------------------------------------------------------------------
> --
> ? INVITE 
> sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls
> SIP/2.0
> ? Via: SIP/2.0/TLS 192.168.250.18;branch=z9hG4bKyHm8ttBjB7Dmj
> ? Max-Forwards: 69
> ? From: "Extension 1002" <sip:1002 at 192.168.250.18>;tag=861g6FpZm4KHe
> ? To: <sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls>
> ? Call-ID: 729a5dfc-62cd-122f-2d9c-001cc071031a
> ? CSeq: 18180461 INVITE
> ? Contact: <sip:mod_sofia at 192.168.250.18:5061;transport=tls>
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-6724d7a 2011-09-23 
> 01-25-43
> +0000
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> ? Supported: timer, precondition, path, replaces
> ? Allow-Events: talk, hold, presence, dialog, line-seize, call-info, 
> sla, include-session-description, presence.winfo, message-summary, 
> refer
> ? Content-Type: application/sdp
> ? Content-Disposition: session
> ? Content-Length: 234
> ? X-FS-Support: update_display
> ? Remote-Party-ID: "Extension 1002"
> <sip:1002 at 192.168.250.18>;party=calling;screen=yes;privacy=off
>
> ? v=0
> ? o=- 19191 16302 IN IP4 192.168.250.20
> ? s=epkccjl
> ? c=IN IP4 192.168.250.20
> ? t=0 0
> ? m=audio 17570 RTP/AVP 0 8 9 102 3 101
> ? a=rtpmap:102 ILBC/8000
> ? a=fmtp:102 mode=20
> ? a=rtpmap:101 telephone-event/8000
> ? a=fmtp:101 0-15
> ? a=ptime:20
> ? 
> ----------------------------------------------------------------------
> -- recv 321 bytes from tls/[192.168.250.11]:49179 at 10:30:51.400065:
> ? 
> ----------------------------------------------------------------------
> --
> ? SIP/2.0 100 Trying
> ? Via: SIP/2.0/TLS 
> 192.168.250.18;branch=z9hG4bKyHm8ttBjB7Dmj;rport=5061
> ? From: "Extension 1002" <sip:1002 at 192.168.250.18>;tag=861g6FpZm4KHe
> ? Call-ID: 729a5dfc-62cd-122f-2d9c-001cc071031a
> ? CSeq: 18180461 INVITE
> ? To: <sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls>
> ? Content-Length: 0
>
> ? 
> ----------------------------------------------------------------------
> -- recv 499 bytes from tls/[192.168.250.11]:49179 at 10:30:51.417353:
> ? 
> ----------------------------------------------------------------------
> --
> ? SIP/2.0 180 Ringing
> ? Via: SIP/2.0/TLS 
> 192.168.250.18;branch=z9hG4bKyHm8ttBjB7Dmj;rport=5061
> ? Contact: <sip:1003 at 192.168.250.11:49179;transport=tls>
> ? From: "Extension 1002" <sip:1002 at 192.168.250.18>;tag=861g6FpZm4KHe
> ? Call-ID: 729a5dfc-62cd-122f-2d9c-001cc071031a
> ? CSeq: 18180461 INVITE
> ? To:
> <sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls>;tag=5
> A4771A
> 2806D8CDF36D39CDFEB4B0E50
> ? Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
> ? Supported: replaces, path
> ? Content-Length: 0
>
> ? 
> ----------------------------------------------------------------------
> --
> 2011-09-26 11:30:51.412650 [NOTICE] sofia.c:5235 Ring-Ready 
> sofia/internal/sip:1003 at 192.168.250.11:49179!
> 2011-09-26 11:30:51.412650 [NOTICE] mod_sofia.c:2391 Ring-Ready 
> sofia/internal/1002 at 192.168.250.18!
> send 866 bytes to tls/[192.168.250.20]:49220 at 10:30:51.422883:
> ? 
> ----------------------------------------------------------------------
> --
> ? SIP/2.0 180 Ringing
> ? Via: SIP/2.0/TLS
> 192.168.250.20:49220;branch=z9hG4bK3PhRphI56AiqPIBh;rport=49220
> ? From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
> ? To: <sip:1003 at 192.168.250.18>;tag=7X8Q4m5UQUXyj
> ? Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
> ? CSeq: 1 INVITE
> ? Contact: <sip:1003 at 192.168.250.18:5061;transport=tls>
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-6724d7a 2011-09-23 
> 01-25-43
> +0000
> ? Accept: application/sdp
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> ? Supported: timer, precondition, path, replaces
> ? Allow-Events: talk, hold, presence, dialog, line-seize, call-info, 
> sla, include-session-description, presence.winfo, message-summary, 
> refer
> ? Content-Length: 0
> ? Remote-Party-ID: "Outbound Call"
> <sip:1003 at 192.168.250.18>;party=calling;privacy=off;screen=no
>
> ? 
> ----------------------------------------------------------------------
> --
> 2011-09-26 11:30:51.412650 [NOTICE] switch_ivr_originate.c:481 Ring 
> Ready sofia/internal/1002 at 192.168.250.18!
> recv 719 bytes from tls/[192.168.250.11]:49179 at 10:30:54.585965:
> ? 
> ----------------------------------------------------------------------
> --
> ? SIP/2.0 200 OK
> ? Via: SIP/2.0/TLS 
> 192.168.250.18;branch=z9hG4bKyHm8ttBjB7Dmj;rport=5061
> ? Contact: <sip:1003 at 192.168.250.11:49179;transport=tls>
> ? From: "Extension 1002" <sip:1002 at 192.168.250.18>;tag=861g6FpZm4KHe
> ? Call-ID: 729a5dfc-62cd-122f-2d9c-001cc071031a
> ? CSeq: 18180461 INVITE
> ? To:
> <sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls>;tag=5
> A4771A
> 2806D8CDF36D39CDFEB4B0E50
> ? Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
> ? Supported: replaces, path
> ? Content-Type: application/sdp
> ? Content-Length: 192
>
> ? v=0
> ? o=- 41521 15327 IN IP4 192.168.250.11
> ? s=yqsmhzj
> ? c=IN IP4 192.168.250.11
> ? t=0 0
> ? m=audio 60032 RTP/AVP 0 101
> ? a=rtpmap:101 TELEPHONE-EVENT/8000
> ? a=fmtp:101 0-15
> ? a=ptime:20
> ? a=sendrecv
> ? 
> ----------------------------------------------------------------------
> -- send 459 bytes to tls/[192.168.250.11]:49179 at 10:30:54.589056:
> ? 
> ----------------------------------------------------------------------
> --
> ? ACK sip:1003 at 192.168.250.11:49179;transport=tls SIP/2.0
> ? Via: SIP/2.0/TLS 192.168.250.18;branch=z9hG4bKZtD1vNvN8F46D
> ? Max-Forwards: 70
> ? From: "Extension 1002" <sip:1002 at 192.168.250.18>;tag=861g6FpZm4KHe
> ? To:
> <sip:1003 at 192.168.250.11:49179;rinstance=69F3D1AD;transport=tls>;tag=5
> A4771A
> 2806D8CDF36D39CDFEB4B0E50
> ? Call-ID: 729a5dfc-62cd-122f-2d9c-001cc071031a
> ? CSeq: 18180461 ACK
> ? Contact: <sip:mod_sofia at 192.168.250.18:5061;transport=tls>
> ? Content-Length: 0
>
> ? 
> ----------------------------------------------------------------------
> --
> 2011-09-26 11:30:54.572557 [NOTICE] sofia.c:5802 Channel 
> [sofia/internal/sip:1003 at 192.168.250.11:49179] has been answered
> 2011-09-26 11:30:54.592567 [NOTICE] switch_ivr_originate.c:481 Ring 
> Ready sofia/internal/1002 at 192.168.250.18!
> send 1079 bytes to tls/[192.168.250.20]:49220 at 10:30:54.596443:
> ? 
> ----------------------------------------------------------------------
> --
> ? SIP/2.0 200 OK
> ? Via: SIP/2.0/TLS
> 192.168.250.20:49220;branch=z9hG4bK3PhRphI56AiqPIBh;rport=49220
> ? From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
> ? To: <sip:1003 at 192.168.250.18>;tag=7X8Q4m5UQUXyj
> ? Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
> ? CSeq: 1 INVITE
> ? Contact: <sip:1003 at 192.168.250.18:5061;transport=tls>
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-6724d7a 2011-09-23 
> 01-25-43
> +0000
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> ? Supported: timer, precondition, path, replaces
> ? Allow-Events: talk, hold, presence, dialog, line-seize, call-info, 
> sla, include-session-description, presence.winfo, message-summary, 
> refer
> ? Content-Type: application/sdp
> ? Content-Disposition: session
> ? Content-Length: 180
> ? Remote-Party-ID: "Outbound Call"
> <sip:1003 at 192.168.250.18>;party=calling;privacy=off;screen=no
>
> ? v=02011-09-26 11:30:54.592567 [NOTICE] switch_ivr.c:706 Channel 
> [sofia/internal/1002 at 192.168.250.18] has been answered
>
> ? o=- 41521 15327 IN IP4 192.168.250.11
> ? s=yqsmhzj
> ? c=IN IP4 192.168.250.11
> ? t=0 0
> ? m=audio 60032 RTP/AVP 0 101
> ? a=rtpmap:101 TELEPHONE-EVENT/8000
> ? a=fmtp:101 0-15
> ? a=ptime:20
> ? 
> ----------------------------------------------------------------------
> -- recv 618 bytes from tls/[192.168.250.20]:49220 at 10:30:54.756293:
> ? 
> ----------------------------------------------------------------------
> --
> ? ACK sip:1003 at 192.168.250.18:5061;transport=tls SIP/2.0
> ? Via: SIP/2.0/TLS
> 192.168.250.20:49220;branch=z9hG4bK3PhRphI56AiqPIBh;rport
> ? Max-Forwards: 70
> ? To: <sip:1003 at 192.168.250.18>;tag=7X8Q4m5UQUXyj
> ? From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
> ? Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
> ? CSeq: 1 ACK
> ? Proxy-Authorization: Digest
> username="1002",realm="192.168.250.18",algorithm=MD5,uri="sip:1003 at 192
> .168.2
> 50.18",nonce="df6eed19-bd20-460a-b5ac-7e44946e729a",qop=auth,cnonce="4
> 042a69 
> 2f4f53c4479be6b2e765c37de",nc=00000005,response="bc47467ac4faac81bac59
> f37e4d
> df7cb"
> ? Content-Length: 0
>
> ? 
> ----------------------------------------------------------------------
> -- recv 825 bytes from tls/[192.168.250.20]:49221 at 10:31:00.397189:
> ? 
> ----------------------------------------------------------------------
> --
> ? BYE sip:1003 at 192.168.250.18:5061;transport=tls SIP/2.0
> ? Via: SIP/2.0/TCP
> 192.168.250.20:49221;branch=z9hG4bKLDntiDuXixOceTKx;rport
> ? Contact: <sips:1002 at 192.168.250.20:49221;transport=tcp>
> ? Max-Forwards: 70
> ? From: <sip:1002 at 192.168.250.18>;tag=8614E3ED232BC82FF4EA5D313E6AF6D0
> ? Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
> ? Supported: replaces, path
> ? User-Agent: Acrobits Softphone Business/1.8.8
> ? To: <sip:1003 at 192.168.250.18>;tag=7X8Q4m5UQUXyj
> ? Call-ID: 55F45E25069A293B6E96D2EC24C30D30F4AF99B0
> ? CSeq: 2 BYE
> ? Proxy-Authorization: Digest
> username="1002",realm="192.168.250.18",algorithm=MD5,uri="sip:1003 at 192
> .168.2 
> 50.18:5061;transport=tls",nonce="df6eed19-bd20-460a-b5ac-7e44946e729a"
> ,qop=a 
> uth,cnonce="e2160338300c683de4a96cacce593376",nc=00000006,response="b1
> 546d9b
> c75688462874c2617671f1f7"
> ? Content-Length: 0
>
> ? 
> ----------------------------------------------------------------------
> --
>
>
>
> Kind regards
>
> Ben
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
> Anthony Minessale
> Sent: 24 September 2011 19:59
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] TLS issue
>
> you could try using the command
>
> sofia global siptrace on
>
> to see what you are getting.
>
> On Sat, Sep 24, 2011 at 7:14 AM, Ben Naylor <bnaylor at sirran.com> wrote:
>> Hello Freeswitch community!
>>
>>
>>
>> I was wondering if someone could help me with an issue I am 
>> experiencing when using TLS to secure SIP traffic?
>>
>> Our current test environment consists of a Freeswitch test server, a 
>> wireless access point, and 2 x iPhones using SIP client, all 
>> connected to the same closed off VLAN.
>>
>>
>>
>> The Freeswitch server is set to bypass media=true, so that the RTP 
>> traffic flows directly between the iPhones.? When testing a call 
>> using standard SIP/RTP, the call is successful, and terminates 
>> correctly when one of the devices hangs up the call.
>>
>>
>>
>> The problem occurs when using TLS, and trying to terminate the call.
>> The call establishes successfully, but when either of the handsets 
>> hangs up, the other remains connected to the call.? This happens when 
>> hanging up the call from both the calling and the called handset.
>>
>>
>>
>> One thing I noticed when looking at the console output, is that two 
>> messages appear along the lines of ?unknown Sip packet? (apologies 
>> for the vagueness, I am not currently in front of the test server).? 
>> These appear when the call is terminated on one of the hand sets.
>>
>>
>>
>> Has anyone had a similar experience when using TLS in this type of 
>> scenario?
>>
>>
>>
>> Thanks for the help!
>>
>>
>>
>> Regards
>>
>>
>>
>> Ben
>>
>>
>>
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> e
>> rs
>> 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-use
> rs
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>



--
Kristian Kielhofner


FreeSWITCH-users mailing list
FreeSWITCH-users at 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 asilva at wirelessmundi.com  Tue Sep 27 13:42:31 2011
From: asilva at wirelessmundi.com (Antonio)
Date: Tue, 27 Sep 2011 11:42:31 +0200
Subject: [Freeswitch-users] Problem with Sangoma D100 Transconding card
Message-ID: <1317116551.12197.21.camel@marces.madrid.commsmundi.com>

Hi,

I'm using a sangoma D100 transconding card for the codec g729, it works
perfectly until recently that it stops working with the following error:


2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:1095
sngtc_get_existing_rtp_session
2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:1095
sngtc_get_free_rtp_session
2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:224 New allocated
port 13284 for IP 10.254.254.1/10.254.254.1
2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:224 New allocated
port 13286 for IP 10.254.254.1/10.254.254.1
2011-09-27 10:32:45.192555 [ERR] mod_sangoma_codec.c:1106 Create
Transcoding Session Error
2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:243 Released port
13284 for IP 10.254.254.1/10.254.254.1
2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:243 Released port
13286 for IP 10.254.254.1/10.254.254.1
2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:1095
sngtc_release_rtp_session
2011-09-27 10:32:45.192555 [ERR] mod_sangoma_codec.c:486 Failed to
create Sangoma encoding session.
2011-09-27 10:32:45.192555 [ERR] switch_core_io.c:1060 Codec Sangoma
G729 encoder error!


This error has trigger by the stop/start of freeswitch. 

I had to restart the sangoma daemon (sngtc_server) to make it work
again.


Do i have to restart the sangoma daemon every time i restart the
freeswitch? shouldn't they be independent? 





-- 

Best regards, 

 _________________________

Ant?nio Silva

E-mail:asilva at wirelessmundi.com
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From irwin.billy at gmail.com  Tue Sep 27 05:55:35 2011
From: irwin.billy at gmail.com (Billy L. Irwin)
Date: Mon, 26 Sep 2011 21:55:35 -0400
Subject: [Freeswitch-users] Any DID
Message-ID: <4e812d15.2888ec0a.2d79.ffffec4b@mx.google.com>

Hi All,

I am running Freeswitch with Blue.Box as a front end. Is there a way to add
an inbound route that would support any DID as I did within Asterisk? I
don't want to have to add all our DIDs to users right off the bat? 

Thanks,

Billy




From jmesquita at freeswitch.org  Tue Sep 27 19:41:28 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 27 Sep 2011 12:41:28 -0300
Subject: [Freeswitch-users] FSGUI
In-Reply-To: <CAGRr05_efk_i211d8eNBWm_2tDdVxRKzCjcHMwJg6ErRX8Uoxg@mail.gmail.com>
References: <CAGRr05_efk_i211d8eNBWm_2tDdVxRKzCjcHMwJg6ErRX8Uoxg@mail.gmail.com>
Message-ID: <CABH8QX3E7UnSb5E1=s4uz=8u0j1uBNyk19wnarJq--zwgJ4xLA@mail.gmail.com>

Hello Brad,

First of all, let me say that I am flattered that someone actually uses
FSGui. I thought there simply was no interest in it whatsoever.

The linux version is not that easy to "package" as the Windows version. The
best way is to compile it yourself. I can even do it for you. What arch are
you running on? 64 or 32?

Regards,
Jo?o Mesquita (aka jmesquita)



On Mon, Sep 26, 2011 at 6:34 PM, Brad Mina <brad at tech21.com> wrote:

> I've stumbled across fsgui a few times, (
> http://wiki.freeswitch.org/wiki/Fsgui ), and am happy to report the
> windows release of this program works flawlessly!
>
> After toying with the windows version myself and a few other community
> members wanted to see this application running in linux as well.
> I downloaded the fsgui.tar.gz and unzipped it to /tmp then installed the
> following packages on Ubuntu 10.04, which also installed related
> dependencies:
> libqt4-dev
> libqt4-core
>
> I tried to chmod+x and execute the application with ./fsgui - however bash
> reports the following:
>
> fsgui: cannot execute binary file
>
>
> Is there something I'm missing here? I'm hoping jmesquita or a fellow
> community member can respond with insight.
>
> Thanks in advance.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Tue Sep 27 19:51:23 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 27 Sep 2011 18:51:23 +0300
Subject: [Freeswitch-users] FSGUI
In-Reply-To: <CABH8QX3E7UnSb5E1=s4uz=8u0j1uBNyk19wnarJq--zwgJ4xLA@mail.gmail.com>
References: <CAGRr05_efk_i211d8eNBWm_2tDdVxRKzCjcHMwJg6ErRX8Uoxg@mail.gmail.com>
	<CABH8QX3E7UnSb5E1=s4uz=8u0j1uBNyk19wnarJq--zwgJ4xLA@mail.gmail.com>
Message-ID: <CALNZuwng=J4PO+QwqBigJxbbp3jQBkBOeHqESmU=KRuAQmP_Cg@mail.gmail.com>

I'm running on ubuntu (11.04) 32bit.
I'm no stranger to compiling (thank FS!) but I simply didn't see how,
given the links/info on the wiki.

-Avi


2011/9/27 Jo?o Mesquita <jmesquita at freeswitch.org>:
> Hello Brad,
> First of all, let me say that I am flattered that someone actually uses
> FSGui. I thought there simply was no interest in it whatsoever.
> The linux version is not that easy to "package" as the Windows version. The
> best way is to compile it yourself. I can even do it for you. What arch are
> you running on? 64 or 32?
> Regards,
> Jo?o Mesquita (aka jmesquita)
>
>
>
> On Mon, Sep 26, 2011 at 6:34 PM, Brad Mina <brad at tech21.com> wrote:
>>
>> I've stumbled across fsgui a few times, (
>> http://wiki.freeswitch.org/wiki/Fsgui ), and am happy to report the windows
>> release of this program works flawlessly!
>>
>> After toying with the windows version myself and a few other community
>> members wanted to see this application running in linux as well.
>> I downloaded the fsgui.tar.gz and unzipped it to /tmp then installed the
>> following packages on Ubuntu 10.04, which also installed related
>> dependencies:
>> libqt4-dev
>> libqt4-core
>> I tried to chmod+x and execute the application with ./fsgui - however bash
>> reports the following:
>>
>> fsgui: cannot execute binary file
>>
>> Is there something I'm missing here? I'm hoping jmesquita or a fellow
>> community member can respond with insight.
>>
>> 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
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Tue Sep 27 19:54:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 27 Sep 2011 08:54:02 -0700
Subject: [Freeswitch-users] Any DID
In-Reply-To: <4e812d15.2888ec0a.2d79.ffffec4b@mx.google.com>
References: <4e812d15.2888ec0a.2d79.ffffec4b@mx.google.com>
Message-ID: <CAKzWOxWnXEvUr8JPt+s18CoFXGeh3GHuTTucLer=vMA37qUtPA@mail.gmail.com>

I would ask in the #2600hz IRC channel or on their mailing lists. BB uses a
database to store dialplan information so it would have to be configured
somehow from within BB. I know the stock FS XML dialplan can easily do this,
but when you add a GUI layer in between the admin and the configuration it
can make stuff like this a bit more troublesome.

-MC

On Mon, Sep 26, 2011 at 6:55 PM, Billy L. Irwin <irwin.billy at gmail.com>wrote:

> Hi All,
>
> I am running Freeswitch with Blue.Box as a front end. Is there a way to add
> an inbound route that would support any DID as I did within Asterisk? I
> don't want to have to add all our DIDs to users right off the bat?
>
> Thanks,
>
> Billy
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Sep 27 19:55:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 27 Sep 2011 08:55:44 -0700
Subject: [Freeswitch-users] Error loading mod_shout
In-Reply-To: <4E7BD3E4.7090200@statirasystems.com>
References: <4E7BD3E4.7090200@statirasystems.com>
Message-ID: <CAKzWOxV8xrwFjUzzK-m+F43kR=jkuS5e8hNq4Bi7sBMA-rip-g@mail.gmail.com>

Were you able to resolve this?
-MC

On Thu, Sep 22, 2011 at 5:33 PM, William Moore <bmoore at statirasystems.com>wrote:

> I get the following error when I load mod_shout through the cli.
>
> freeswitch at internal> load mod_shout
> +OK Reloading XML
> -ERR [module load file routine returned an error]
>
> 2011-09-23 00:31:10.654023 [INFO] switch_time.c:1028 Timezone reloaded
> 530 definitions
> freeswitch at internal> 2011-09-23 00:31:10.654023 [CRIT]
> switch_loadable_module.c:1271 Error Loading module
> /usr/local/freeswitch/mod/mod_shout.so
> **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol:
> curl_easy_setopt**
>
> Any ideas?
>
> --
> William J. Moore
> Statira Systems
> 611 Caroline St
> Fredericksburg, VA
> http://www.statirasystems.com
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Sep 27 20:02:32 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 27 Sep 2011 09:02:32 -0700
Subject: [Freeswitch-users] Gateway Authentication
In-Reply-To: <D2DD0DB0-1560-459E-9194-9ADB754B534B@lyonl.com>
References: <D2DD0DB0-1560-459E-9194-9ADB754B534B@lyonl.com>
Message-ID: <CAKzWOxVNv5h1V3MMBBirib52SUo_zDbJOTCj9t3y7RNuQpBmmw@mail.gmail.com>

Were you able to get this working? If not, get a console debug log with a
siptrace and drop it on pastebin.freeswitch.org. Be sure to use "FreeSWITCH
Log" as the syntax highlighting.

at fs_cli you need to do:

console loglevel debug
sofia global siptrace on

That will turn on all the debug stuff you need. From there make the call
attempt and capture the output, then drop on pastebin. Give us the pb URL in
this thread.

Thanks,
MC

On Tue, Sep 20, 2011 at 9:34 AM, Chad Vogel <cvogel at lyonl.com> wrote:

>  Hello, I'm trying to make the move from Asterix, but I'm running into
> some difficulties. I'm try to bridge a call using our gateway however it
> doesn't work. In wireshark I can see I'm getting an SIP
> 401 Unauthorized error with a WWW-Authenticate header, after FS send the
> INVITE message to the gateway. However FS doesnt seem to respond to the
> request for Authentication.  Asterix responds correctly however I cant seem
> to make FS to do the same. Any help would be appreciated
>
>
>  INVITE sip:+15618911806 at 4.55.35.60:5070 SIP/2.0
> Via: SIP/2.0/UDP 207.67.30.226;rport;branch=z9hG4bKB49SZQHrgaaKc
> Max-Forwards: 8
> From: "LyonL" <sip:+14142211800 at 207.67.30.226:5060>;tag=eZe8gcQgXXv5c
> To: <sip:+15618911806 at 4.55.35.60:5070>
> Call-ID: 4f8edae0-5e45-122f-6399-07d4dbeff43f
> CSeq: 17931324 INVITE
> Contact: <sip:1-F2la9 at 207.67.30.226:5060;transport=udp;gw=level3>
> User-Agent: FreeSWITCH
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 227
> X-FS-Support: update_display
> Remote-Party-ID: "LyonL" <sip:+14142211800 at 207.67.30.226:5060
> >;party=calling;screen=yes;privacy=off
>
>  v=0
> o=FreeSWITCH 1316516832 1316516833 IN IP4 10.126.200.6
> s=FreeSWITCH
> c=IN IP4 10.126.200.6
> t=0 0
> m=audio 17944 RTP/AVP 0 8 18 101 13
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
>
>  SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 207.67.30.226;received=207.67.30.226
> ;branch=z9hG4bKB49SZQHrgaaKc;rport=42534
> From: "LyonL" <sip:+14142211800 at 207.67.30.226:5060>;tag=eZe8gcQgXXv5c
> To: <sip:+15618911806 at 4.55.35.60:5070
> >;tag=SD6soqf99-1367649635-1316534779161
> Call-ID: 4f8edae0-5e45-122f-6399-07d4dbeff43f
> CSeq: 17931324 INVITE
> WWW-Authenticate: DIGEST
> qop="auth",nonce="BroadWorksXgst2td09Tbihi2qBW",algorithm=MD5,realm="BroadWorks"
> Content-Length: 0
>
>     <include>
>      <extension name="4142211800">
>         <condition field="destination_number" expression="^(\+?1)?
> (4142211800)$">
>          <action application="set" data="effective_caller_id_name=LyonL"/>
>          <action application="set" data="effective_caller_id_number=
> +14142211800"/>
>          <action application="bridge" data="sofia/gateway/level3/
> +15618911806"/>
>        </condition>
>      </extension>
>    </include>
>
>   <include>
>    <gateway name="level3">
>      <param name="apply-inbound-acl" value="level3"/>
>      <param name="username" value="1-F2la9"/>
>      <param name="password" value="password"/>>
>      <param name="realm" value="BroadWorks"/>
>      <param name="proxy" value="4.55.35.60:5070"/>
>      <param name="from-domain" value="207.67.30.226:5060"/>
>      <param name="dtmf-type" value="rfc2833"/>
>      <param name="extension-in-contact" value="true"/>
>      <param name="caller-id-in-from" value="true"/>
>      <param name="register" value="false"/>
>    </gateway>
>  </include>
>
>   <profile name="external">
>    <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
>    <!-- This profile is only for outbound registrations to providers -->
>    <gateways>
>      <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
>    </gateways>
>
>    <aliases>
>      <!--
>      <alias name="outbound"/>
>      <alias name="nat"/>
>      -->
>    </aliases>
>
>    <domains>
>      <domain name="all" alias="false" parse="true"/>
>    </domains>
>
>    <settings>
>      <param name="debug" value="0"/>
>  <!-- If you want FreeSWITCH to shutdown if this profile fails to load,
> uncomment the next line. -->
>  <!-- <param name="shutdown-on-fail" value="true"/> -->
>      <param name="sip-trace" value="no"/>
>      <param name="sip-capture" value="no"/>
>      <param name="rfc2833-pt" value="101"/>
>      <param name="sip-port" value="$${external_sip_port}"/>
>      <param name="dialplan" value="XML"/>
>      <param name="context" value="public"/>
>      <param name="dtmf-duration" value="2000"/>
>      <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>      <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
>      <param name="hold-music" value="$${hold_music}"/>
>      <param name="rtp-timer-name" value="soft"/>
>      <!--<param name="enable-100rel" value="true"/>-->
>      <!--<param name="disable-srv503" value="true"/>-->
>      <!-- This could be set to "passive" -->
>      <param name="local-network-acl" value="localnet.auto"/>
>      <param name="manage-presence" value="false"/>
>
>      <!-- used to share presence info across sofia profiles
>  manage-presence needs to be set to passive on this profile
>  if you want it to behave as if it were the internal profile
>  for presence.
>      -->
>      <!-- Name of the db to use for this profile -->
>      <!--<param name="dbname" value="share_presence"/>-->
>      <!--<param name="presence-hosts" value="$${domain}"/>-->
>      <!--<param name="force-register-domain" value="$${domain}"/>-->
>      <!--all inbound reg will stored in the db using this domain -->
>      <!--<param name="force-register-db-domain" value="$${domain}"/>-->
>      <!-- ************************************************* -->
>
>      <!--<param name="aggressive-nat-detection" value="true"/>-->
>      <param name="inbound-codec-negotiation" value="generous"/>
>      <param name="nonce-ttl" value="60"/>
>      <param name="auth-calls" value="false"/>
>      <!--<param name="challenge-realm" value="auto_from"/>-->
>      <param name="user-agent-string" value="FreeSWITCH"/>
>      <!--
>  DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
>      -->
>      <param name="rtp-ip" value="$${local_ip_v4}"/>
>      <param name="sip-ip" value="$${local_ip_v4}"/>
>      <param name="ext-rtp-ip" value="auto-nat"/>
>      <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>      <param name="rtp-timeout-sec" value="300"/>
>      <param name="rtp-hold-timeout-sec" value="1800"/>
>      <!--<param name="enable-3pcc" value="true"/>-->
>
>      <!-- TLS: disabled by default, set to "true" to enable -->
>      <param name="tls" value="$${external_ssl_enable}"/>
>      <!-- additional bind parameters for TLS -->
>      <param name="tls-bind-params" value="transport=tls"/>
>      <!-- Port to listen on for TLS requests. (5081 will be used if
> unspecified) -->
>      <param name="tls-sip-port" value="$${external_tls_port}"/>
>      <!-- Location of the agent.pem and cafile.pem ssl certificates
> (needed for TLS server) -->
>      <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
>      <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not
> work with TLSv1 -->
>      <param name="tls-version" value="$${sip_tls_version}"/>
>
>    </settings>
>  </profile>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From robert.hadley at teotech.com  Tue Sep 27 20:15:11 2011
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 27 Sep 2011 09:15:11 -0700
Subject: [Freeswitch-users] large block of DID's
In-Reply-To: <CAGRr05-3GjyrEU+74oQbprayxo+K2eKPzrkpDgqaSFtRZqMp9A@mail.gmail.com>
References: <D25CC3E5-C0D2-41D7-87C3-843C24A0CA8C@lyonl.com>
	<801470822-1316057954-cardhu_decombobulator_blackberry.rim.net-146923873-@b5.c27.bise6.blackberry>
	<2C4ADC11-8559-45ED-94E9-435F1D60CA78@lyonl.com>
	<CALNZuwnhT26xOEDAR9iGtcSicOtx6L8F1O6vzEBBCAwAX4ziig@mail.gmail.com>
	<E826C94A3C6BE44198FAF4D66B5AB4D8245DAAA0A6@VA3DIAXVS351.RED001.local>
	<465713D3-D7A4-4CD5-99B8-72F583836410@lyonl.com>
	<E826C94A3C6BE44198FAF4D66B5AB4D8245DAAA191@VA3DIAXVS351.RED001.local>
	<CAGRr05-3GjyrEU+74oQbprayxo+K2eKPzrkpDgqaSFtRZqMp9A@mail.gmail.com>
Message-ID: <E826C94A3C6BE44198FAF4D66B5AB4D8245DAAA99B@VA3DIAXVS351.RED001.local>

Hi Brad,

I submitted it via improvement Jira today.

http://jira.freeswitch.org/browse/FS-3584

Thanks for asking,
Robert


From: Brad Mina [mailto:brad at tech21.com]
Sent: Saturday, September 24, 2011 11:28 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] large block of DID's

Robert,

Is your module in the git contrib? Is there a public place we can download it?
On Fri, Sep 23, 2011 at 1:13 PM, Robert Hadley <robert.hadley at teotech.com<mailto:robert.hadley at teotech.com>> wrote:

No, the custom didmap module doesn't manage users, another app does.  I call the didmap app from public.xml and transfer to whatever destination it returns in defaults.xml, e.g. a local_extension, faxserver, or caller_blocked.

    <extension name="didmap_lookup_transfer" continue="true">
      <condition field="destination_number" expression="^(\d+)$">
        <action application="set" data="did_number=$1"/>
        <action application="didmap" data="${did_number} ${caller_id_number}"/>
        <action application="transfer" data="${didmaplookup_dest} XML default"/>
      </condition>
    </extension>

DID map DB table:
CREATE TABLE inbounddidnumbermap
(
  id bigserial NOT NULL,
  phonenumber character varying(255),
  fax boolean NOT NULL DEFAULT false,
  mobile boolean NOT NULL DEFAULT false,
  extension_id bigint,
  CONSTRAINT inbounddidnumbermap_pkey PRIMARY KEY (id),
)


From: Chad Vogel [mailto:cvogel at lyonl.com<mailto:cvogel at lyonl.com>]
Sent: Friday, September 23, 2011 12:31 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] large block of DID's

I would love to see your module because I'm are thinking about writing one c or c# to do almost the same thing. are you then routing the DID to an application server to manage the users?


On Sep 23, 2011, at 11:26 AM, Robert Hadley wrote:


I created a C didmap module I can share that uses ODBC to lookup a variable length DID number and return the user's extension, plus flags for fax server and mobile DIDs.  If you pass in the caller_id_number it also checks for blocked callers.

Robert


From: Avi Marcus [mailto:avi at avimarcus.net]<mailto:[mailto:avi at avimarcus.net]>
Sent: Thursday, September 22, 2011 12:54 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] large block of DID's

I'd be interested in seeing the Lua script that does the users/directory. I don't think I've seen one yet, I'm not quite sure how that would work. I've only used xml_curl for a dialplan with a web server..
-Avi


On Thu, Sep 22, 2011 at 10:00 PM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:
Are you also using a lua script via odbc to manage users?


On Sep 14, 2011, at 10:39 PM, <ljjimenez at gmail.com<mailto:ljjimenez at gmail.com>>
 <ljjimenez at gmail.com<mailto:ljjimenez at gmail.com>> wrote:

> We do this with about 10k DIDs, we use MySQL and a Lua script via ODBC
>
> Luis Jimenez
>
> -----Original Message-----
> From: Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>>
> Sender: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>
> Date: Wed, 14 Sep 2011 15:44:23
> To: FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org><FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: [Freeswitch-users] large block of DID's
>
> I was wondering if anyone has any thoughts what is the best way to manage a large blocks of DID? (we have about 1600 DID's that we are moving off of our asterisk servers)
>
> Chad
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




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


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<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>
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From brad at tech21.com  Tue Sep 27 20:25:16 2011
From: brad at tech21.com (Brad Mina)
Date: Tue, 27 Sep 2011 09:25:16 -0700
Subject: [Freeswitch-users] FSGUI
In-Reply-To: <CALNZuwng=J4PO+QwqBigJxbbp3jQBkBOeHqESmU=KRuAQmP_Cg@mail.gmail.com>
References: <CAGRr05_efk_i211d8eNBWm_2tDdVxRKzCjcHMwJg6ErRX8Uoxg@mail.gmail.com>
	<CABH8QX3E7UnSb5E1=s4uz=8u0j1uBNyk19wnarJq--zwgJ4xLA@mail.gmail.com>
	<CALNZuwng=J4PO+QwqBigJxbbp3jQBkBOeHqESmU=KRuAQmP_Cg@mail.gmail.com>
Message-ID: <CAGRr05-Rgu-a58LnaxqjveE5gWB=834f_8OdB+zkejkt07dCAg@mail.gmail.com>

First, I really do like this application. It reallllly helps when I manage a
couple test servers and a couple production servers. The custom hilighting
and search functions are awesome, to say the least. (Built in regex search
 <3333)

I run various linux distros - however generally on 32 bit.

I noticed it keeps in memory the log it receives - and after leaving an
active switch running for an hour or so the program will become unresponsive
and crash (on windows). Is there a way you could have it instead write to a
textfile upon each connect and read/write from/to it instead of storing it
instead to ease on the resource usage?

Thanks again!

On Tue, Sep 27, 2011 at 8:51 AM, Avi Marcus <avi at avimarcus.net> wrote:

> I'm running on ubuntu (11.04) 32bit.
> I'm no stranger to compiling (thank FS!) but I simply didn't see how,
> given the links/info on the wiki.
>
> -Avi
>
>
> 2011/9/27 Jo?o Mesquita <jmesquita at freeswitch.org>:
> > Hello Brad,
> > First of all, let me say that I am flattered that someone actually uses
> > FSGui. I thought there simply was no interest in it whatsoever.
> > The linux version is not that easy to "package" as the Windows version.
> The
> > best way is to compile it yourself. I can even do it for you. What arch
> are
> > you running on? 64 or 32?
> > Regards,
> > Jo?o Mesquita (aka jmesquita)
> >
> >
> >
> > On Mon, Sep 26, 2011 at 6:34 PM, Brad Mina <brad at tech21.com> wrote:
> >>
> >> I've stumbled across fsgui a few times, (
> >> http://wiki.freeswitch.org/wiki/Fsgui ), and am happy to report the
> windows
> >> release of this program works flawlessly!
> >>
> >> After toying with the windows version myself and a few other community
> >> members wanted to see this application running in linux as well.
> >> I downloaded the fsgui.tar.gz and unzipped it to /tmp then installed the
> >> following packages on Ubuntu 10.04, which also installed related
> >> dependencies:
> >> libqt4-dev
> >> libqt4-core
> >> I tried to chmod+x and execute the application with ./fsgui - however
> bash
> >> reports the following:
> >>
> >> fsgui: cannot execute binary file
> >>
> >> Is there something I'm missing here? I'm hoping jmesquita or a fellow
> >> community member can respond with insight.
> >>
> >> 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
> >>
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From cvogel at lyonl.com  Tue Sep 27 20:30:34 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Tue, 27 Sep 2011 16:30:34 +0000
Subject: [Freeswitch-users] Gateway Authentication
In-Reply-To: <CAKzWOxVNv5h1V3MMBBirib52SUo_zDbJOTCj9t3y7RNuQpBmmw@mail.gmail.com>
References: <D2DD0DB0-1560-459E-9194-9ADB754B534B@lyonl.com>
	<CAKzWOxVNv5h1V3MMBBirib52SUo_zDbJOTCj9t3y7RNuQpBmmw@mail.gmail.com>
Message-ID: <41368772-7534-4021-96FD-C03998CCC7AC@lyonl.com>

I was able to get it to work by hardcoding the via rport to 5060 in sofia - our upstream provider uses a checkpont firewall in their network that is changing the source port which made level3 respond on the wrong port... I don't like this fix but it works. I really wish there was an option in fs to enable this because the rfc for sip requires the receiver respond on the port they receive the UDP message on if rport in the via header isn't set.

On Sep 27, 2011, at 11:02 AM, Michael Collins wrote:

Were you able to get this working? If not, get a console debug log with a siptrace and drop it on pastebin.freeswitch.org<http://pastebin.freeswitch.org/>. Be sure to use "FreeSWITCH Log" as the syntax highlighting.

at fs_cli you need to do:

console loglevel debug
sofia global siptrace on

That will turn on all the debug stuff you need. From there make the call attempt and capture the output, then drop on pastebin. Give us the pb URL in this thread.

Thanks,
MC

On Tue, Sep 20, 2011 at 9:34 AM, Chad Vogel <cvogel at lyonl.com<mailto:cvogel at lyonl.com>> wrote:
Hello, I'm trying to make the move from Asterix, but I'm running into some difficulties. I'm try to bridge a call using our gateway however it doesn't work. In wireshark I can see I'm getting an SIP 401 Unauthorized error with a WWW-Authenticate header, after FS send the INVITE message to the gateway. However FS doesnt seem to respond to the request for Authentication.  Asterix responds correctly however I cant seem to make FS to do the same. Any help would be appreciated


INVITE sip:+15618911806 at 4.55.35.60:5070 SIP/2.0
Via: SIP/2.0/UDP 207.67.30.226<tel:207.67.30.226>;rport;branch=z9hG4bKB49SZQHrgaaKc
Max-Forwards: 8
From: "LyonL" <sip:+14142211800 at 207.67.30.226:5060>;tag=eZe8gcQgXXv5c
To: <sip:+15618911806 at 4.55.35.60:5070>
Call-ID: 4f8edae0-5e45-122f-6399-07d4dbeff43f
CSeq: 17931324 INVITE
Contact: <sip:1-F2la9 at 207.67.30.226:5060;transport=udp;gw=level3>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 227
X-FS-Support: update_display
Remote-Party-ID: "LyonL" <sip:+14142211800 at 207.67.30.226:5060>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1316516832 1316516833 IN IP4 10.126.200.6
s=FreeSWITCH
c=IN IP4 10.126.200.6
t=0 0
m=audio 17944 RTP/AVP 0 8 18 101 13
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20


SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 207.67.30.226<tel:207.67.30.226>;received=207.67.30.226<tel:207.67.30.226>;branch=z9hG4bKB49SZQHrgaaKc;rport=42534
From: "LyonL" <sip:+14142211800 at 207.67.30.226:5060>;tag=eZe8gcQgXXv5c
To: <sip:+15618911806 at 4.55.35.60:5070>;tag=SD6soqf99-1367649635-1316534779161
Call-ID: 4f8edae0-5e45-122f-6399-07d4dbeff43f
CSeq: 17931324 INVITE
WWW-Authenticate: DIGEST qop="auth",nonce="BroadWorksXgst2td09Tbihi2qBW",algorithm=MD5,realm="BroadWorks"
Content-Length: 0

  <include>
    <extension name="4142211800<tel:4142211800>">
      <condition field="destination_number" expression="^(\+?1)?(4142211800<tel:%284142211800>)$">
        <action application="set" data="effective_caller_id_name=LyonL"/>
        <action application="set" data="effective_caller_id_number=+14142211800<tel:%2B14142211800>"/>
        <action application="bridge" data="sofia/gateway/level3/+15618911806<tel:%2B15618911806>"/>
      </condition>
    </extension>
  </include>

<include>
  <gateway name="level3">
    <param name="apply-inbound-acl" value="level3"/>
    <param name="username" value="1-F2la9"/>
    <param name="password" value="password"/>>
    <param name="realm" value="BroadWorks"/>
    <param name="proxy" value="4.55.35.60:5070<http://4.55.35.60:5070/>"/>
    <param name="from-domain" value="207.67.30.226<tel:207.67.30.226>:5060"/>
    <param name="dtmf-type" value="rfc2833"/>
    <param name="extension-in-contact" value="true"/>
    <param name="caller-id-in-from" value="true"/>
    <param name="register" value="false"/>
  </gateway>
</include>

<profile name="external">
  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
  <!-- This profile is only for outbound registrations to providers -->
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>

  <aliases>
    <!--
    <alias name="outbound"/>
    <alias name="nat"/>
    -->
  </aliases>

  <domains>
    <domain name="all" alias="false" parse="true"/>
  </domains>

  <settings>
    <param name="debug" value="0"/>
<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
<!-- <param name="shutdown-on-fail" value="true"/> -->
    <param name="sip-trace" value="no"/>
    <param name="sip-capture" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="$${external_sip_port}"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="rtp-timer-name" value="soft"/>
    <!--<param name="enable-100rel" value="true"/>-->
    <!--<param name="disable-srv503" value="true"/>-->
    <!-- This could be set to "passive" -->
    <param name="local-network-acl" value="localnet.auto"/>
    <param name="manage-presence" value="false"/>

    <!-- used to share presence info across sofia profiles
manage-presence needs to be set to passive on this profile
if you want it to behave as if it were the internal profile
for presence.
    -->
    <!-- Name of the db to use for this profile -->
    <!--<param name="dbname" value="share_presence"/>-->
    <!--<param name="presence-hosts" value="$${domain}"/>-->
    <!--<param name="force-register-domain" value="$${domain}"/>-->
    <!--all inbound reg will stored in the db using this domain -->
    <!--<param name="force-register-db-domain" value="$${domain}"/>-->
    <!-- ************************************************* -->

    <!--<param name="aggressive-nat-detection" value="true"/>-->
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="false"/>
    <!--<param name="challenge-realm" value="auto_from"/>-->
    <param name="user-agent-string" value="FreeSWITCH"/>
    <!--
DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
    -->
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
    <!--<param name="enable-3pcc" value="true"/>-->

    <!-- TLS: disabled by default, set to "true" to enable -->
    <param name="tls" value="$${external_ssl_enable}"/>
    <!-- additional bind parameters for TLS -->
    <param name="tls-bind-params" value="transport=tls"/>
    <!-- Port to listen on for TLS requests. (5081 will be used if unspecified) -->
    <param name="tls-sip-port" value="$${external_tls_port}"/>
    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
    <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
    <param name="tls-version" value="$${sip_tls_version}"/>

  </settings>
</profile>



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From chrisg.lists at gmail.com  Tue Sep 27 20:36:43 2011
From: chrisg.lists at gmail.com (Chris Graham)
Date: Tue, 27 Sep 2011 18:36:43 +0200
Subject: [Freeswitch-users] FSGUI
In-Reply-To: <CAGRr05-Rgu-a58LnaxqjveE5gWB=834f_8OdB+zkejkt07dCAg@mail.gmail.com>
References: <CAGRr05_efk_i211d8eNBWm_2tDdVxRKzCjcHMwJg6ErRX8Uoxg@mail.gmail.com>
	<CABH8QX3E7UnSb5E1=s4uz=8u0j1uBNyk19wnarJq--zwgJ4xLA@mail.gmail.com>
	<CALNZuwng=J4PO+QwqBigJxbbp3jQBkBOeHqESmU=KRuAQmP_Cg@mail.gmail.com>
	<CAGRr05-Rgu-a58LnaxqjveE5gWB=834f_8OdB+zkejkt07dCAg@mail.gmail.com>
Message-ID: <CAMKRqpgncFGL_w_x5u7Z8KfL0oktfuRARS36bKyOM_SQpTL8AQ@mail.gmail.com>

On Tue, Sep 27, 2011 at 6:25 PM, Brad Mina <brad at tech21.com> wrote:

> First, I really do like this application. It reallllly helps when I manage
> a couple test servers and a couple production servers. The custom hilighting
> and search functions are awesome, to say the least. (Built in regex search
>  <3333)
>
> I run various linux distros - however generally on 32 bit.
>
> I noticed it keeps in memory the log it receives - and after leaving an
> active switch running for an hour or so the program will become unresponsive
> and crash (on windows). Is there a way you could have it instead write to a
> textfile upon each connect and read/write from/to it instead of storing it
> instead to ease on the resource usage?
>
> Thanks again!
>
> On Tue, Sep 27, 2011 at 8:51 AM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> I'm running on ubuntu (11.04) 32bit.
>> I'm no stranger to compiling (thank FS!) but I simply didn't see how,
>> given the links/info on the wiki.
>>
>> -Avi
>>
>>
>> 2011/9/27 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> > Hello Brad,
>> > First of all, let me say that I am flattered that someone actually uses
>> > FSGui. I thought there simply was no interest in it whatsoever.
>> > The linux version is not that easy to "package" as the Windows version.
>> The
>> > best way is to compile it yourself. I can even do it for you. What arch
>> are
>> > you running on? 64 or 32?
>> > Regards,
>> > Jo?o Mesquita (aka jmesquita)
>> >
>> >
>> >
>> > On Mon, Sep 26, 2011 at 6:34 PM, Brad Mina <brad at tech21.com> wrote:
>> >>
>> >> I've stumbled across fsgui a few times, (
>> >> http://wiki.freeswitch.org/wiki/Fsgui ), and am happy to report the
>> windows
>> >> release of this program works flawlessly!
>> >>
>> >> After toying with the windows version myself and a few other community
>> >> members wanted to see this application running in linux as well.
>> >> I downloaded the fsgui.tar.gz and unzipped it to /tmp then installed
>> the
>> >> following packages on Ubuntu 10.04, which also installed related
>> >> dependencies:
>> >> libqt4-dev
>> >> libqt4-core
>> >> I tried to chmod+x and execute the application with ./fsgui - however
>> bash
>> >> reports the following:
>> >>
>> >> fsgui: cannot execute binary file
>> >>
>> >> Is there something I'm missing here? I'm hoping jmesquita or a fellow
>> >> community member can respond with insight.
>> >>
>> >> 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
>> >>
>> >
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
Hi Brad/Joao,

Apologies for hijacking this thread.. Let me know if I should rather start
another. I have been using FSGUI on snow leopard, but have zero luck with
Lion. I would be happy to do beta testing or even some extra work to get it
going. Before that, if there is a lion .dmg out there already I would rather
not re-invent the wheel.

Thanks in advance,
Chris G
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From jmesquita at freeswitch.org  Tue Sep 27 20:36:49 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 27 Sep 2011 13:36:49 -0300
Subject: [Freeswitch-users] FSGUI
In-Reply-To: <CALNZuwng=J4PO+QwqBigJxbbp3jQBkBOeHqESmU=KRuAQmP_Cg@mail.gmail.com>
References: <CAGRr05_efk_i211d8eNBWm_2tDdVxRKzCjcHMwJg6ErRX8Uoxg@mail.gmail.com>
	<CABH8QX3E7UnSb5E1=s4uz=8u0j1uBNyk19wnarJq--zwgJ4xLA@mail.gmail.com>
	<CALNZuwng=J4PO+QwqBigJxbbp3jQBkBOeHqESmU=KRuAQmP_Cg@mail.gmail.com>
Message-ID: <CABH8QX0qS7Uaxb69nrCg9K-F7W_sQ=QZvY2xSr_Kj_N9WmZEaQ@mail.gmail.com>

Avi, you are right. Let me resurrect an Ubuntu here and compile it to make
the docs. I will post to the wiki and let you know.

Regards,
Jo?o Mesquita



On Tue, Sep 27, 2011 at 12:51 PM, Avi Marcus <avi at avimarcus.net> wrote:

> I'm running on ubuntu (11.04) 32bit.
> I'm no stranger to compiling (thank FS!) but I simply didn't see how,
> given the links/info on the wiki.
>
> -Avi
>
>
> 2011/9/27 Jo?o Mesquita <jmesquita at freeswitch.org>:
> > Hello Brad,
> > First of all, let me say that I am flattered that someone actually uses
> > FSGui. I thought there simply was no interest in it whatsoever.
> > The linux version is not that easy to "package" as the Windows version.
> The
> > best way is to compile it yourself. I can even do it for you. What arch
> are
> > you running on? 64 or 32?
> > Regards,
> > Jo?o Mesquita (aka jmesquita)
> >
> >
> >
> > On Mon, Sep 26, 2011 at 6:34 PM, Brad Mina <brad at tech21.com> wrote:
> >>
> >> I've stumbled across fsgui a few times, (
> >> http://wiki.freeswitch.org/wiki/Fsgui ), and am happy to report the
> windows
> >> release of this program works flawlessly!
> >>
> >> After toying with the windows version myself and a few other community
> >> members wanted to see this application running in linux as well.
> >> I downloaded the fsgui.tar.gz and unzipped it to /tmp then installed the
> >> following packages on Ubuntu 10.04, which also installed related
> >> dependencies:
> >> libqt4-dev
> >> libqt4-core
> >> I tried to chmod+x and execute the application with ./fsgui - however
> bash
> >> reports the following:
> >>
> >> fsgui: cannot execute binary file
> >>
> >> Is there something I'm missing here? I'm hoping jmesquita or a fellow
> >> community member can respond with insight.
> >>
> >> 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
> >>
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Tue Sep 27 20:40:28 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 27 Sep 2011 13:40:28 -0300
Subject: [Freeswitch-users] FSGUI
In-Reply-To: <CAGRr05-Rgu-a58LnaxqjveE5gWB=834f_8OdB+zkejkt07dCAg@mail.gmail.com>
References: <CAGRr05_efk_i211d8eNBWm_2tDdVxRKzCjcHMwJg6ErRX8Uoxg@mail.gmail.com>
	<CABH8QX3E7UnSb5E1=s4uz=8u0j1uBNyk19wnarJq--zwgJ4xLA@mail.gmail.com>
	<CALNZuwng=J4PO+QwqBigJxbbp3jQBkBOeHqESmU=KRuAQmP_Cg@mail.gmail.com>
	<CAGRr05-Rgu-a58LnaxqjveE5gWB=834f_8OdB+zkejkt07dCAg@mail.gmail.com>
Message-ID: <CABH8QX1BgCkygrUhEej-HwTrizXZUPaJF_CNrs5onESB1Y=A9A@mail.gmail.com>

Brad,

You are right, we do keep all log messages on the buffer and don't discard
any of them.

Here are the steps I can take to solve this. Will do it in phases so that

1. Have a configuration that specifies the console buffer size. Meaning that
I will simply discard X amount of messages. That will prevent it from
locking up.
2. Save the discarded messages to a log file on disk.
3. Read that log file when the user requests messages from a buffer already
discarded.

How does that sound?

Regards,
Jo?o Mesquita



On Tue, Sep 27, 2011 at 1:25 PM, Brad Mina <brad at tech21.com> wrote:

> First, I really do like this application. It reallllly helps when I manage
> a couple test servers and a couple production servers. The custom hilighting
> and search functions are awesome, to say the least. (Built in regex search
>  <3333)
>
> I run various linux distros - however generally on 32 bit.
>
> I noticed it keeps in memory the log it receives - and after leaving an
> active switch running for an hour or so the program will become unresponsive
> and crash (on windows). Is there a way you could have it instead write to a
> textfile upon each connect and read/write from/to it instead of storing it
> instead to ease on the resource usage?
>
> Thanks again!
>
> On Tue, Sep 27, 2011 at 8:51 AM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> I'm running on ubuntu (11.04) 32bit.
>> I'm no stranger to compiling (thank FS!) but I simply didn't see how,
>> given the links/info on the wiki.
>>
>> -Avi
>>
>>
>> 2011/9/27 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> > Hello Brad,
>> > First of all, let me say that I am flattered that someone actually uses
>> > FSGui. I thought there simply was no interest in it whatsoever.
>> > The linux version is not that easy to "package" as the Windows version.
>> The
>> > best way is to compile it yourself. I can even do it for you. What arch
>> are
>> > you running on? 64 or 32?
>> > Regards,
>> > Jo?o Mesquita (aka jmesquita)
>> >
>> >
>> >
>> > On Mon, Sep 26, 2011 at 6:34 PM, Brad Mina <brad at tech21.com> wrote:
>> >>
>> >> I've stumbled across fsgui a few times, (
>> >> http://wiki.freeswitch.org/wiki/Fsgui ), and am happy to report the
>> windows
>> >> release of this program works flawlessly!
>> >>
>> >> After toying with the windows version myself and a few other community
>> >> members wanted to see this application running in linux as well.
>> >> I downloaded the fsgui.tar.gz and unzipped it to /tmp then installed
>> the
>> >> following packages on Ubuntu 10.04, which also installed related
>> >> dependencies:
>> >> libqt4-dev
>> >> libqt4-core
>> >> I tried to chmod+x and execute the application with ./fsgui - however
>> bash
>> >> reports the following:
>> >>
>> >> fsgui: cannot execute binary file
>> >>
>> >> Is there something I'm missing here? I'm hoping jmesquita or a fellow
>> >> community member can respond with insight.
>> >>
>> >> 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
>> >>
>> >
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Tue Sep 27 20:41:55 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 27 Sep 2011 13:41:55 -0300
Subject: [Freeswitch-users] FSGUI
In-Reply-To: <CAMKRqpgncFGL_w_x5u7Z8KfL0oktfuRARS36bKyOM_SQpTL8AQ@mail.gmail.com>
References: <CAGRr05_efk_i211d8eNBWm_2tDdVxRKzCjcHMwJg6ErRX8Uoxg@mail.gmail.com>
	<CABH8QX3E7UnSb5E1=s4uz=8u0j1uBNyk19wnarJq--zwgJ4xLA@mail.gmail.com>
	<CALNZuwng=J4PO+QwqBigJxbbp3jQBkBOeHqESmU=KRuAQmP_Cg@mail.gmail.com>
	<CAGRr05-Rgu-a58LnaxqjveE5gWB=834f_8OdB+zkejkt07dCAg@mail.gmail.com>
	<CAMKRqpgncFGL_w_x5u7Z8KfL0oktfuRARS36bKyOM_SQpTL8AQ@mail.gmail.com>
Message-ID: <CABH8QX0q4tZWqzo4iswZMD3UKc0giwZe+2E1KEf0WOrGWL0iyA@mail.gmail.com>

Chris, not at all. It is OK to put it here. I am amazed how many ppl have
been using this application! Thank you all!

I use Lion as well and haven't tried to build a dmg for it yet. I will add
it to the queue. It might take me a few days tho. I don't see why it
shouldn't be running tho...

Regards,
Jo?o Mesquita



On Tue, Sep 27, 2011 at 1:36 PM, Chris Graham <chrisg.lists at gmail.com>wrote:

> On Tue, Sep 27, 2011 at 6:25 PM, Brad Mina <brad at tech21.com> wrote:
>
>> First, I really do like this application. It reallllly helps when I manage
>> a couple test servers and a couple production servers. The custom hilighting
>> and search functions are awesome, to say the least. (Built in regex search
>>  <3333)
>>
>> I run various linux distros - however generally on 32 bit.
>>
>> I noticed it keeps in memory the log it receives - and after leaving an
>> active switch running for an hour or so the program will become unresponsive
>> and crash (on windows). Is there a way you could have it instead write to a
>> textfile upon each connect and read/write from/to it instead of storing it
>> instead to ease on the resource usage?
>>
>> Thanks again!
>>
>> On Tue, Sep 27, 2011 at 8:51 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> I'm running on ubuntu (11.04) 32bit.
>>> I'm no stranger to compiling (thank FS!) but I simply didn't see how,
>>> given the links/info on the wiki.
>>>
>>> -Avi
>>>
>>>
>>> 2011/9/27 Jo?o Mesquita <jmesquita at freeswitch.org>:
>>> > Hello Brad,
>>> > First of all, let me say that I am flattered that someone actually uses
>>> > FSGui. I thought there simply was no interest in it whatsoever.
>>> > The linux version is not that easy to "package" as the Windows version.
>>> The
>>> > best way is to compile it yourself. I can even do it for you. What arch
>>> are
>>> > you running on? 64 or 32?
>>> > Regards,
>>> > Jo?o Mesquita (aka jmesquita)
>>> >
>>> >
>>> >
>>> > On Mon, Sep 26, 2011 at 6:34 PM, Brad Mina <brad at tech21.com> wrote:
>>> >>
>>> >> I've stumbled across fsgui a few times, (
>>> >> http://wiki.freeswitch.org/wiki/Fsgui ), and am happy to report the
>>> windows
>>> >> release of this program works flawlessly!
>>> >>
>>> >> After toying with the windows version myself and a few other community
>>> >> members wanted to see this application running in linux as well.
>>> >> I downloaded the fsgui.tar.gz and unzipped it to /tmp then installed
>>> the
>>> >> following packages on Ubuntu 10.04, which also installed related
>>> >> dependencies:
>>> >> libqt4-dev
>>> >> libqt4-core
>>> >> I tried to chmod+x and execute the application with ./fsgui - however
>>> bash
>>> >> reports the following:
>>> >>
>>> >> fsgui: cannot execute binary file
>>> >>
>>> >> Is there something I'm missing here? I'm hoping jmesquita or a fellow
>>> >> community member can respond with insight.
>>> >>
>>> >> 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
>>> >>
>>> >
>>> >
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> Hi Brad/Joao,
>
> Apologies for hijacking this thread.. Let me know if I should rather start
> another. I have been using FSGUI on snow leopard, but have zero luck with
> Lion. I would be happy to do beta testing or even some extra work to get it
> going. Before that, if there is a lion .dmg out there already I would rather
> not re-invent the wheel.
>
> Thanks in advance,
> Chris G
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From irwin.billy at gmail.com  Tue Sep 27 20:44:14 2011
From: irwin.billy at gmail.com (Billy L. Irwin)
Date: Tue, 27 Sep 2011 12:44:14 -0400
Subject: [Freeswitch-users] Any DID
In-Reply-To: <CAKzWOxWnXEvUr8JPt+s18CoFXGeh3GHuTTucLer=vMA37qUtPA@mail.gmail.com>
References: <4e812d15.2888ec0a.2d79.ffffec4b@mx.google.com>
	<CAKzWOxWnXEvUr8JPt+s18CoFXGeh3GHuTTucLer=vMA37qUtPA@mail.gmail.com>
Message-ID: <4e81fd60.0f42960a.18c6.2263@mx.google.com>

I will definitely go there and see, but in the mean time can you tell me how
you do it with FS alone? That will likely point me in the right direction
plus I am trying to be familiar with both ways just in case. Thanks, Billy

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, September 27, 2011 11:54
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Any DID

I would ask in the #2600hz IRC channel or on their mailing lists. BB uses a
database to store dialplan information so it would have to be configured
somehow from within BB. I know the stock FS XML dialplan can easily do this,
but when you add a GUI layer in between the admin and the configuration it
can make stuff like this a bit more troublesome.

-MC


On Mon, Sep 26, 2011 at 6:55 PM, Billy L. Irwin <irwin.billy at gmail.com>
wrote:


	Hi All,
	
	I am running Freeswitch with Blue.Box as a front end. Is there a way
to add
	an inbound route that would support any DID as I did within
Asterisk? I
	don't want to have to add all our DIDs to users right off the bat?
	
	Thanks,
	
	Billy
	
	
	
	
	FreeSWITCH-users mailing list
	FreeSWITCH-users at 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 brad at tech21.com  Tue Sep 27 21:02:03 2011
From: brad at tech21.com (Brad Mina)
Date: Tue, 27 Sep 2011 10:02:03 -0700
Subject: [Freeswitch-users] FSGUI
In-Reply-To: <CABH8QX1BgCkygrUhEej-HwTrizXZUPaJF_CNrs5onESB1Y=A9A@mail.gmail.com>
References: <CAGRr05_efk_i211d8eNBWm_2tDdVxRKzCjcHMwJg6ErRX8Uoxg@mail.gmail.com>
	<CABH8QX3E7UnSb5E1=s4uz=8u0j1uBNyk19wnarJq--zwgJ4xLA@mail.gmail.com>
	<CALNZuwng=J4PO+QwqBigJxbbp3jQBkBOeHqESmU=KRuAQmP_Cg@mail.gmail.com>
	<CAGRr05-Rgu-a58LnaxqjveE5gWB=834f_8OdB+zkejkt07dCAg@mail.gmail.com>
	<CABH8QX1BgCkygrUhEej-HwTrizXZUPaJF_CNrs5onESB1Y=A9A@mail.gmail.com>
Message-ID: <CAGRr05_brgN2VNP4c=skP5aRWagzxVYd83rrj=T82=fe5d1s1Q@mail.gmail.com>

Awesome!

This tool is very helpful. I can see a lot of addition able to be made on
top of it's already very practical functionality and I'm sure with a little
word of mouth and these improvements, people will come around to using your
program :D

2011/9/27 Jo?o Mesquita <jmesquita at freeswitch.org>

> Brad,
>
> You are right, we do keep all log messages on the buffer and don't discard
> any of them.
>
> Here are the steps I can take to solve this. Will do it in phases so that
>
> 1. Have a configuration that specifies the console buffer size. Meaning
> that I will simply discard X amount of messages. That will prevent it from
> locking up.
> 2. Save the discarded messages to a log file on disk.
> 3. Read that log file when the user requests messages from a buffer already
> discarded.
>
> How does that sound?
>
> Regards,
> Jo?o Mesquita
>
>
>
>
> On Tue, Sep 27, 2011 at 1:25 PM, Brad Mina <brad at tech21.com> wrote:
>
>> First, I really do like this application. It reallllly helps when I manage
>> a couple test servers and a couple production servers. The custom hilighting
>> and search functions are awesome, to say the least. (Built in regex search
>>  <3333)
>>
>> I run various linux distros - however generally on 32 bit.
>>
>> I noticed it keeps in memory the log it receives - and after leaving an
>> active switch running for an hour or so the program will become unresponsive
>> and crash (on windows). Is there a way you could have it instead write to a
>> textfile upon each connect and read/write from/to it instead of storing it
>> instead to ease on the resource usage?
>>
>> Thanks again!
>>
>> On Tue, Sep 27, 2011 at 8:51 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> I'm running on ubuntu (11.04) 32bit.
>>> I'm no stranger to compiling (thank FS!) but I simply didn't see how,
>>> given the links/info on the wiki.
>>>
>>> -Avi
>>>
>>>
>>> 2011/9/27 Jo?o Mesquita <jmesquita at freeswitch.org>:
>>> > Hello Brad,
>>> > First of all, let me say that I am flattered that someone actually uses
>>> > FSGui. I thought there simply was no interest in it whatsoever.
>>> > The linux version is not that easy to "package" as the Windows version.
>>> The
>>> > best way is to compile it yourself. I can even do it for you. What arch
>>> are
>>> > you running on? 64 or 32?
>>> > Regards,
>>> > Jo?o Mesquita (aka jmesquita)
>>> >
>>> >
>>> >
>>> > On Mon, Sep 26, 2011 at 6:34 PM, Brad Mina <brad at tech21.com> wrote:
>>> >>
>>> >> I've stumbled across fsgui a few times, (
>>> >> http://wiki.freeswitch.org/wiki/Fsgui ), and am happy to report the
>>> windows
>>> >> release of this program works flawlessly!
>>> >>
>>> >> After toying with the windows version myself and a few other community
>>> >> members wanted to see this application running in linux as well.
>>> >> I downloaded the fsgui.tar.gz and unzipped it to /tmp then installed
>>> the
>>> >> following packages on Ubuntu 10.04, which also installed related
>>> >> dependencies:
>>> >> libqt4-dev
>>> >> libqt4-core
>>> >> I tried to chmod+x and execute the application with ./fsgui - however
>>> bash
>>> >> reports the following:
>>> >>
>>> >> fsgui: cannot execute binary file
>>> >>
>>> >> Is there something I'm missing here? I'm hoping jmesquita or a fellow
>>> >> community member can respond with insight.
>>> >>
>>> >> 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
>>> >>
>>> >
>>> >
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brad at tech21.com  Tue Sep 27 21:09:15 2011
From: brad at tech21.com (Brad Mina)
Date: Tue, 27 Sep 2011 10:09:15 -0700
Subject: [Freeswitch-users] Any DID
In-Reply-To: <4e81fd60.0f42960a.18c6.2263@mx.google.com>
References: <4e812d15.2888ec0a.2d79.ffffec4b@mx.google.com>
	<CAKzWOxWnXEvUr8JPt+s18CoFXGeh3GHuTTucLer=vMA37qUtPA@mail.gmail.com>
	<4e81fd60.0f42960a.18c6.2263@mx.google.com>
Message-ID: <CAGRr05-0FSzVKL9JvAxE1=t1Zpw_Zvc-Q3yXbvcnS0Yzh1nwJw@mail.gmail.com>

>
> <condition field="destination_number" expression="(\d{11})"/>

Or however many digits your carrier sends it to you as.


On Tue, Sep 27, 2011 at 9:44 AM, Billy L. Irwin <irwin.billy at gmail.com>wrote:

> I will definitely go there and see, but in the mean time can you tell me
> how
> you do it with FS alone? That will likely point me in the right direction
> plus I am trying to be familiar with both ways just in case. Thanks, Billy
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael
> Collins
> Sent: Tuesday, September 27, 2011 11:54
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Any DID
>
> I would ask in the #2600hz IRC channel or on their mailing lists. BB uses a
> database to store dialplan information so it would have to be configured
> somehow from within BB. I know the stock FS XML dialplan can easily do
> this,
> but when you add a GUI layer in between the admin and the configuration it
> can make stuff like this a bit more troublesome.
>
> -MC
>
>
> On Mon, Sep 26, 2011 at 6:55 PM, Billy L. Irwin <irwin.billy at gmail.com>
> wrote:
>
>
>        Hi All,
>
>        I am running Freeswitch with Blue.Box as a front end. Is there a way
> to add
>        an inbound route that would support any DID as I did within
> Asterisk? I
>        don't want to have to add all our DIDs to users right off the bat?
>
>        Thanks,
>
>        Billy
>
>
>
>
>        FreeSWITCH-users mailing list
>        FreeSWITCH-users at lists.freeswitch.org
>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>        http://www.freeswitch.org
>
>
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Sep 27 21:10:59 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 27 Sep 2011 10:10:59 -0700
Subject: [Freeswitch-users] Any DID
In-Reply-To: <4e81fd60.0f42960a.18c6.2263@mx.google.com>
References: <4e812d15.2888ec0a.2d79.ffffec4b@mx.google.com>
	<CAKzWOxWnXEvUr8JPt+s18CoFXGeh3GHuTTucLer=vMA37qUtPA@mail.gmail.com>
	<4e81fd60.0f42960a.18c6.2263@mx.google.com>
Message-ID: <CAKzWOxXT3ouN=99UF9mjvs68citRJJrmQ5vRaDyKuAXca7hr5w@mail.gmail.com>

In FreeSWITCH you'd create a condition and match on a regex that represents
all your DIDs. So, depending on your DID range, you could do something like
this:

<extension name="inbound DIDs">
  <condition field="destination_number" expression="^1?(21299935\d\d)$">
    <action application="transfer" data="1000 XML default"/>
  </condition>
</extension>

So if your DIDs are all in area code 213 and prefix 999 and you have the
block of 3500-3599 then the above expression would match and then send any
of those inbound calls to user 1000.

-MC

On Tue, Sep 27, 2011 at 9:44 AM, Billy L. Irwin <irwin.billy at gmail.com>wrote:

> I will definitely go there and see, but in the mean time can you tell me
> how
> you do it with FS alone? That will likely point me in the right direction
> plus I am trying to be familiar with both ways just in case. Thanks, Billy
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael
> Collins
> Sent: Tuesday, September 27, 2011 11:54
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Any DID
>
> I would ask in the #2600hz IRC channel or on their mailing lists. BB uses a
> database to store dialplan information so it would have to be configured
> somehow from within BB. I know the stock FS XML dialplan can easily do
> this,
> but when you add a GUI layer in between the admin and the configuration it
> can make stuff like this a bit more troublesome.
>
> -MC
>
>
> On Mon, Sep 26, 2011 at 6:55 PM, Billy L. Irwin <irwin.billy at gmail.com>
> wrote:
>
>
>        Hi All,
>
>        I am running Freeswitch with Blue.Box as a front end. Is there a way
> to add
>        an inbound route that would support any DID as I did within
> Asterisk? I
>        don't want to have to add all our DIDs to users right off the bat?
>
>        Thanks,
>
>        Billy
>
>
>
>
>        FreeSWITCH-users mailing list
>        FreeSWITCH-users at lists.freeswitch.org
>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>        http://www.freeswitch.org
>
>
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brad at tech21.com  Tue Sep 27 22:04:52 2011
From: brad at tech21.com (Brad Mina)
Date: Tue, 27 Sep 2011 11:04:52 -0700
Subject: [Freeswitch-users] Any DID
In-Reply-To: <CAKzWOxXT3ouN=99UF9mjvs68citRJJrmQ5vRaDyKuAXca7hr5w@mail.gmail.com>
References: <4e812d15.2888ec0a.2d79.ffffec4b@mx.google.com>
	<CAKzWOxWnXEvUr8JPt+s18CoFXGeh3GHuTTucLer=vMA37qUtPA@mail.gmail.com>
	<4e81fd60.0f42960a.18c6.2263@mx.google.com>
	<CAKzWOxXT3ouN=99UF9mjvs68citRJJrmQ5vRaDyKuAXca7hr5w@mail.gmail.com>
Message-ID: <CAGRr05-EnHZOQhJSt503M02KexxgabfJ4CHQ08n8C+b12wnPqg@mail.gmail.com>

Just for clarification, the method I mentioned will take any 11 digit invite
and try and process it - Michael's will match all invites for 21299935xx.

On Tue, Sep 27, 2011 at 10:10 AM, Michael Collins <msc at freeswitch.org>wrote:

> In FreeSWITCH you'd create a condition and match on a regex that represents
> all your DIDs. So, depending on your DID range, you could do something like
> this:
>
> <extension name="inbound DIDs">
>   <condition field="destination_number" expression="^1?(21299935\d\d)$">
>     <action application="transfer" data="1000 XML default"/>
>   </condition>
> </extension>
>
> So if your DIDs are all in area code 213 and prefix 999 and you have the
> block of 3500-3599 then the above expression would match and then send any
> of those inbound calls to user 1000.
>
> -MC
>
>
> On Tue, Sep 27, 2011 at 9:44 AM, Billy L. Irwin <irwin.billy at gmail.com>wrote:
>
>> I will definitely go there and see, but in the mean time can you tell me
>> how
>> you do it with FS alone? That will likely point me in the right direction
>> plus I am trying to be familiar with both ways just in case. Thanks, Billy
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Michael
>> Collins
>> Sent: Tuesday, September 27, 2011 11:54
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Any DID
>>
>> I would ask in the #2600hz IRC channel or on their mailing lists. BB uses
>> a
>> database to store dialplan information so it would have to be configured
>> somehow from within BB. I know the stock FS XML dialplan can easily do
>> this,
>> but when you add a GUI layer in between the admin and the configuration it
>> can make stuff like this a bit more troublesome.
>>
>> -MC
>>
>>
>> On Mon, Sep 26, 2011 at 6:55 PM, Billy L. Irwin <irwin.billy at gmail.com>
>> wrote:
>>
>>
>>        Hi All,
>>
>>        I am running Freeswitch with Blue.Box as a front end. Is there a
>> way
>> to add
>>        an inbound route that would support any DID as I did within
>> Asterisk? I
>>        don't want to have to add all our DIDs to users right off the bat?
>>
>>        Thanks,
>>
>>        Billy
>>
>>
>>
>>
>>        FreeSWITCH-users mailing list
>>        FreeSWITCH-users at lists.freeswitch.org
>>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>        http://www.freeswitch.org
>>
>>
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Sep 27 22:08:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 27 Sep 2011 11:08:10 -0700
Subject: [Freeswitch-users] A Chance To "Give Back" To FreeSWITCH
Message-ID: <CAKzWOxUvBA4WC96wiRXgGKy=54=JDPQEe1ry7O0pYVrFWbVfcw@mail.gmail.com>

Hello all,

I have a few FreeSWITCH-related projects with which I could use a hand. We
have some non-glamorous janitorial items that need some attention.
Specifically I could use assistance with a few freeswitch.org and
wiki.freeswitch.org items. Some research/digging for answers is involved. If
you have PHP skills those are a plus.

Please email me off list if you are able to devote a few hours to some
projects that I have pending. We will also discuss some of these on the
FreeSWITCH conference call tomorrow.

Thanks for giving back <http://www.freeswitch.org/node/339>.
-Michael Collins
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From henrikaagaardsorensen at gmail.com  Tue Sep 27 22:46:03 2011
From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=)
Date: Tue, 27 Sep 2011 20:46:03 +0200
Subject: [Freeswitch-users] Change voicemail, example code available?
Message-ID: <CAGH8SeZeLiFdyyE32an2=-JRC==nHgs7V=YFx_fTG_wAujAOyQ@mail.gmail.com>

I would like to change how the voicemail acts in FreeSwitch. But as I've
figured out it's hardcoded into Mod_voicemail.
So I would like to make my own via an IVR.

Does anyone know any sample code for a simple voicemail, that I can use for
reference?

One thing I would like to change is for example that it shouldn't hang up on
wrong ID's, but give one 3 tries to enter a correct ID.
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From fs-list at communicatefreely.net  Tue Sep 27 23:01:47 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Tue, 27 Sep 2011 15:01:47 -0400
Subject: [Freeswitch-users] fifo timeout?
Message-ID: <4E821D9B.30204@communicatefreely.net>

Hello,

I'm building a very simple queue system with mod_fifo.  For the most 
part, things are working the way I want, but I can't figure out how to 
limit the amount of time someone can sit in the queue.  I'm trying to 
implement something like max_wait would do in mod_callcenter.

I can use the orbit_exten option, but it wants to transfer to a 
different extension.  I really just want the fifo application to exit so 
the caller can progress to the next dialplan action.  Is there a way to 
make the caller exit the fifo if they are unanswered after a set amount 
of time?

I did give mod_callcenter a try, but I had a lot of problems with calls 
getting stuck on hold, with no agents ringing, so I went to mod_fifo as 
a simpler solution.

Thanks for any suggestions.

-Tim


From fs-list at communicatefreely.net  Tue Sep 27 23:04:58 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Tue, 27 Sep 2011 15:04:58 -0400
Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re:
 Is there any license G729?
In-Reply-To: <CAMWiMy9dE72qi8ph8DqXpuFwtLHj4q+C0u0jPPV5HB1RegZQmg@mail.gmail.com>
References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com>	<4E80A921.8070600@coppice.org>
	<CAMWiMy9dE72qi8ph8DqXpuFwtLHj4q+C0u0jPPV5HB1RegZQmg@mail.gmail.com>
Message-ID: <4E821E5A.1000604@communicatefreely.net>

Even if you did have a phone that could do it, you would have to design 
a completely different handset with high end audio components to hear 
any difference.

Having said that, I can see an application for a hardware "phone" that 
has balanced line in and out connections, so you could use it for 
broadcast applications.  This could replace all the hybrids you normally 
have in an audio control room, and allow for full-quality phone links if 
both sides support it.



curriegrad2004 wrote:
> Unless you want to spend loads of money on prototyping an VoIP IC that
> does 48KHz, then be my guest and build one
>
> On Mon, Sep 26, 2011 at 9:32 AM, Steve Underwood <steveu at coppice.org> wrote:
>   
>> On 06/30/2009 09:26 AM, Craig Askings wrote:
>>     
>>> Are there any hardware phones that support 48 Khz Celt and
>>> automated/mass deployment?
>>>
>>> Craig.
>>>       
>> Most of the pure VoIP chipsets aren't capable of running at 48k
>> samples/second, so off the shelf phone hardware platforms probably won't
>> cut it. The phones with video capabilities might have 48k sample/second
>> facilities.
>>     
>>> 2009/6/30 Jason White<jason at jasonjgw.net>:
>>>       
>>>> Brian West<brian at freeswitch.org>  wrote:
>>>>         
>>>>> Everyone has this need for lower bandwidth calls... I tend to march the
>>>>> other way. 48kHz baby!  (btw you can do 48kHz in the same bandwidth as a
>>>>> single ulaw call)
>>>>>           
>>>> 48khz Celt (celt at 48000 in your codec preferences) sounds wonderful with
>>>> FreeSWITCH. To test, run two FreeSWITCH instances, both with mod_portaudio.
>>>> This also works well in 48khz conferences.
>>>>
>>>> I wouldn't use G.729 even if it weren't encumbered by patents - it's G.711,
>>>> G.722, G.722.1 and (my current favourite) Celt all the way.
>>>>         
>> 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
>>
>>     
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 freeswitch-list at puzzled.xs4all.nl  Tue Sep 27 23:40:44 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Tue, 27 Sep 2011 21:40:44 +0200
Subject: [Freeswitch-users] Change voicemail, example code available?
In-Reply-To: <CAGH8SeZeLiFdyyE32an2=-JRC==nHgs7V=YFx_fTG_wAujAOyQ@mail.gmail.com>
References: <CAGH8SeZeLiFdyyE32an2=-JRC==nHgs7V=YFx_fTG_wAujAOyQ@mail.gmail.com>
Message-ID: <4E8226BC.1050404@puzzled.xs4all.nl>

On 09/27/2011 08:46 PM, Henrik Aagaard S?rensen wrote:
> I would like to change how the voicemail acts in FreeSwitch. But as I've
> figured out it's hardcoded into Mod_voicemail.
> So I would like to make my own via an IVR.
>
> Does anyone know any sample code for a simple voicemail, that I can use
> for reference?
>
> One thing I would like to change is for example that it shouldn't hang
> up on wrong ID's, but give one 3 tries to enter a correct ID.

Have a look at Jester. It's another voicemail module written in Lua and 
is perhaps easier to change than the original one. Source here:
http://fisheye.freeswitch.org/browse/freeswitch-contrib/hunmonk

Regards,
Patrick


From davidwaf at gmail.com  Wed Sep 28 00:22:36 2011
From: davidwaf at gmail.com (David Wafula)
Date: Tue, 27 Sep 2011 22:22:36 +0200
Subject: [Freeswitch-users] mod_rtmp and flex client
Message-ID: <CAK9Ws2fPY_Gy-tzSrcBqeneNs6msOCBhYejqQ3YFjT7U=OfNUA@mail.gmail.com>

Am finally trying out flex client with mod_rtmp and i just realize how
little i know of this module. I have installed mod_rtmp  and loaded it (or
so i think). On the server, this is what i get:

 freeswitch at internal>rtmp status
-ERR no reply

This doesn't look right for starters..what do i expect for a valid reply?


To the client.
It looks like the basic config on the flex client is to  update
var flashvars = { xxxxxxxx} to point to my freeswitch server, no ?

So, first i try:
var flashvars = {
rtmp_url: 'rtmp://conference.freeswitch.org/phone'
 };

And it does connect.

Say my freeswitch server is running at 1.2.3.4

So, naturally i end up with

var flashvars = {
rtmp_url: 'rtmp://1.2.3.4'
};

Is that the way to do it? Because this gives me:

netStatus: NetConnection.Connect.InvalidApp
netStatus: NetConnection.Connect.Closed

Any pointers will be appreciated.


Thank you
-- 


David Wafula
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From henrikaagaardsorensen at gmail.com  Wed Sep 28 01:25:58 2011
From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=)
Date: Tue, 27 Sep 2011 23:25:58 +0200
Subject: [Freeswitch-users] mod_conference.c:6979 Open of conference.conf
	failed
Message-ID: <CAGH8SeYA8U94Uc=CZ_ij4LftigvC2ND7VwfHHULYJzH4oDH4Hg@mail.gmail.com>

I've installed FS via
http://wiki.freeswitch.org/wiki/Linux_Quick_Install_Guide

Everything seems to run just fine, but when starting up I get the error:
2011-09-27 14:03:16.336176 [ERR] mod_conference.c:6979 Open of
conference.conf failed

Can anyone help me out?

I guess I don't have a conference.conf(.xml) file. Should I? Should it be
installed via the guide or...?
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From yungwei at resolvity.com  Wed Sep 28 01:52:24 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Tue, 27 Sep 2011 17:52:24 -0400
Subject: [Freeswitch-users] fifo count doesn't indicate the number of agents.
Message-ID: <33095823FD21DF429B481B5163264B79515BF38FA6@VMBX102.ihostexchange.net>

Hi,

I defined a queue called support in fifo.conf.xml and added an agent to the queue as show below.
I am able to send a call to the queue first and then take the call to the agent 1004.
However, "fifo count support" always return "support:0:0:0" when an agent is available and nothing in the queue.
The same command returns "support:0:1:1" when an agent is available and one call in the queue.
What am I missing here? Thanks.

<configuration name="fifo.conf" description="FIFO Configuration">
  <settings>
    <param name="delete-all-outbound-member-on-startup" value="false"/>
  </settings>
  <fifos>
    <fifo name="support" importance="0">
        <member timeout="60" simo="1" lag="10">{member_wait=nowait}sofia/user/1004 at 192.168.6.8</member>
    </fifo>
  </fifos>
</configuration>




From steveu at coppice.org  Wed Sep 28 09:43:57 2011
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 28 Sep 2011 13:43:57 +0800
Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re:
 Is there any license G729?
In-Reply-To: <4E821E5A.1000604@communicatefreely.net>
References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com>	<4E80A921.8070600@coppice.org>	<CAMWiMy9dE72qi8ph8DqXpuFwtLHj4q+C0u0jPPV5HB1RegZQmg@mail.gmail.com>
	<4E821E5A.1000604@communicatefreely.net>
Message-ID: <4E82B41D.9010309@coppice.org>

On 09/28/2011 03:04 AM, Tim St. Pierre wrote:
> Even if you did have a phone that could do it, you would have to design
> a completely different handset with high end audio components to hear
> any difference.
You would certainly need a better mic and earpiece, but they don't need 
to be very expensive parts to get pretty good results, and making them 
fit the existing mouldings is unlikely to be that hard. The killer would 
be if the plastic moulding has a mass of resonances. You might need to 
re-engineer the entire thing in that case.
> Having said that, I can see an application for a hardware "phone" that
> has balanced line in and out connections, so you could use it for
> broadcast applications.  This could replace all the hybrids you normally
> have in an audio control room, and allow for full-quality phone links if
> both sides support it.
There are video conferencing platforms which are already like that.

Steve



From xing2kin at yahoo.com  Wed Sep 28 10:47:45 2011
From: xing2kin at yahoo.com (king2kin)
Date: Tue, 27 Sep 2011 23:47:45 -0700 (PDT)
Subject: [Freeswitch-users] FsGUI.exe on windows 2003 server: no available
	plugin
Message-ID: <1317192465.66397.YahooMailNeo@web39701.mail.mud.yahoo.com>

Hi, 
?
I downloaded windows version of FsGui.exe on http://wiki.freeswitch.org/wiki/Fsgui?.
?
After installing it, I tried to run it on windows 2003 server, it poped up an error dialog saying
"no available plugins. Install conpatible plugins before running fsgui". Here I wonder what else I should install to run fsgui.
?
Thanks.
?
x.k.
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From beffa at ieee.org  Wed Sep 28 12:14:53 2011
From: beffa at ieee.org (Federico Beffa)
Date: Wed, 28 Sep 2011 10:14:53 +0200
Subject: [Freeswitch-users] google voice connection going offline
	(mod_dingaling)
Message-ID: <CAKrPhPMsaXaTPi498hS-=Xo2Y3A6tZRCcx4p=waV0Sfvw7W8ug@mail.gmail.com>

Hi All,

I've setup mod_dingaling to connect to my google voice account
according to the instructions on the wiki
http://wiki.freeswitch.org/wiki/Google_Voice#Setup_FreeSWITCH_-_Dingaling_to_work_with_your_Gmail_account
with two modifications:
<param name="rtp-ip" value="auto"/>
<param name="ext-rtp-ip" value="auto-nat"/>

I use the same IP settings for SIP (sofia) with no problem.

The setup works fine for the first hour or two, but then I can see
from another google account that the connection established by
freeswitch goes offline. If I try anyway to make an outbound call,
then the other end does not ring (even if I wait for a long time).
However, after some minutes the freeswitch account returns online and
works again for a couple of hours.

I'm behind a nat and firewall. Is there any port which I have to open
to keep freeswitch online? As far as I understand only the google XMMP
server needs to listen to port 5222 and port 5269.
Or, is there any kind of "keep alive" timer available?

The google talk client running in the same environment works flawlessly.

Thanks for any advise.
Fede


From jmesquita at freeswitch.org  Wed Sep 28 17:19:25 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 28 Sep 2011 10:19:25 -0300
Subject: [Freeswitch-users] FsGUI.exe on windows 2003 server: no
	available plugin
In-Reply-To: <1317192465.66397.YahooMailNeo@web39701.mail.mud.yahoo.com>
References: <1317192465.66397.YahooMailNeo@web39701.mail.mud.yahoo.com>
Message-ID: <CABH8QX3tmNTEhhQq40bvM8cHZa6TgorOKJKN9vJNhTkiaxhdCA@mail.gmail.com>

Do you have the plugins folder inside the folder where you have the .exe?

Regards,
Jo?o Mesquita



On Wed, Sep 28, 2011 at 3:47 AM, king2kin <xing2kin at yahoo.com> wrote:

> Hi,
>
> I downloaded windows version of FsGui.exe on
> http://wiki.freeswitch.org/wiki/Fsgui .
>
> After installing it, I tried to run it on windows 2003 server, it poped up
> an error dialog saying
> "no available plugins. Install conpatible plugins before running fsgui".
> Here I wonder what else I should install to run fsgui.
>
> 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
>
>
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From moises.silva at gmail.com  Wed Sep 28 19:33:01 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 28 Sep 2011 11:33:01 -0400
Subject: [Freeswitch-users] Problem with Sangoma D100 Transconding card
In-Reply-To: <1317116551.12197.21.camel@marces.madrid.commsmundi.com>
References: <1317116551.12197.21.camel@marces.madrid.commsmundi.com>
Message-ID: <CAA4nhyBM0cRr472xe+-XZ_2fsZieP1cWWGPGgnRbPc_qHoscsA@mail.gmail.com>

On Tue, Sep 27, 2011 at 5:42 AM, Antonio <asilva at wirelessmundi.com> wrote:

> **
> Hi,
>
> I'm using a sangoma D100 transconding card for the codec g729, it works
> perfectly until recently that it stops working with the following error:
>
>
> 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:1095
> sngtc_get_existing_rtp_session
> 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:1095
> sngtc_get_free_rtp_session
> 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:224 New allocated
> port 13284 for IP 10.254.254.1/10.254.254.1
> 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:224 New allocated
> port 13286 for IP 10.254.254.1/10.254.254.1
> 2011-09-27 10:32:45.192555 [ERR] mod_sangoma_codec.c:1106 Create
> Transcoding Session Error
> 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:243 Released port
> 13284 for IP 10.254.254.1/10.254.254.1
> 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:243 Released port
> 13286 for IP 10.254.254.1/10.254.254.1
> 2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:1095
> sngtc_release_rtp_session
> 2011-09-27 10:32:45.192555 [ERR] mod_sangoma_codec.c:486 Failed to create
> Sangoma encoding session.
> 2011-09-27 10:32:45.192555 [ERR] switch_core_io.c:1060 Codec Sangoma G729
> encoder error!
>
>
> This error has trigger by the stop/start of freeswitch.
>
> I had to restart the sangoma daemon (sngtc_server) to make it work again.
>
>
> Do i have to restart the sangoma daemon every time i restart the
> freeswitch? shouldn't they be independent?
>

They are independent. If you can reproduce this problem, pastebin the log
/var/log/sngtc_server.log output when the error happens.

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
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From msc at freeswitch.org  Wed Sep 28 20:12:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Sep 2011 09:12:36 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxWy3w78Y2GAWsS=nMY220HsB3wfHaGT4kJRKgT5qOHHQQ@mail.gmail.com>

Hello folks!

We have a few items to discuss today. Here's the agenda:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_09_28

We have a few updates to share with everyone and we also would like to hear
from you if you have been using mod_sms that just came out last week.

Talk to you soon,
Michael
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From anton.vazir at gmail.com  Wed Sep 28 21:01:27 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 28 Sep 2011 22:01:27 +0500
Subject: [Freeswitch-users] How to distinguish between Authenticated sip and
	not authenticated users
Message-ID: <CAGX+XzsxHocAhqHf_D4Sd-Jj_OCqLq5z7NKz6LZQV0_gtmhyRg@mail.gmail.com>

Guys,

trying to distinguish between authenticated SIP user and not
authenticated, still no luck

I would like to allow anyone to call my FS users via SIP, so there are
registered users and everyone else

so I have allowed

<param name="auth-calls" value="true"/>
<param name="apply-inbound-acl" value="allow_all"/>

acl.xml:
    <list name="allow_all" default="allow">
      <node type="allow" cidr="0.0.0.0/0"/>
    </list>
Any call passes through, by always have

[CHANNEL_CREATE] event
variable_sip_authorized: true

in event headers. But I would like to know if user is authorized or not.

if I disable authentication by setting this to false

<param name="auth-calls" value="false"/>

all calls do not have any auth headers.

Any clue?


From dave at clancysystems.com  Wed Sep 28 21:35:14 2011
From: dave at clancysystems.com (Dave)
Date: Wed, 28 Sep 2011 11:35:14 -0600
Subject: [Freeswitch-users] Outbound calling with recorded message.
Message-ID: <F91D0399DF064C4193AA40A6BB059076@clancysystems.com>


Hi all, 

I have been using freeSWITCH for a while for inbound calls in which a person registers for a seminar via IVR.  I simply use the DID, CID number and name to identify the person and put them in the Database. That part of the application is working wonderfully.

The issue I'm presented with now is that we need automate making calls to a few of these registrants, after each event, whos caller_id_name comes in as "Unknown" or "Wireless Caller", Play a recording to let them know we need the name, and that they can record the information right then, or call the office.

I am limited to using a Windows 7 machine.

What would be the best tool to use to automate these calls with the use of an IVR?

Dave Goodwin

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From tyler at phone.com  Wed Sep 28 01:58:17 2011
From: tyler at phone.com (Tyler Winter)
Date: Tue, 27 Sep 2011 14:58:17 -0700
Subject: [Freeswitch-users] Problem 'adding' video to an audio initiated
	call, fmtp
Message-ID: <539AC2B9-E137-4CAF-BFDC-DED583F66D7F@phone.com>

Hi,

I'm trying to add video to a non-video initiated call. I should mention that I have no problems initiated and answering a call as video (using Bria softphone and H.264 codec). I captured several test calls to compare the SDP. I found that adding video doesn't work only when initiating the call as audio. In other words, I can initiate the call as video on the A-leg, and answer the call as audio on the B-leg and then add video on the B-leg and video starts working. I think problem is due to there being no fmtp "media attribute" on the 200 OK w/SDP response from FS to the A-leg when I initialize the call as audio, then add video to it.

I believe I narrowed down the problem to be the following:
- Caller (a-leg) initiated the call as audio.
- Callee (b-leg) answered the call as video
- Caller (a-leg) "adds" video which sends an INVITE to the switch to negotiate the video codecs
- FS responds to Caller (a-leg) with 200 OK (SDP) but the "fmtp" media attribute is missing. 

On a working video call the fmtp attribute is appended to the SDP along with the rest of the video codec details. 

Here's an example:

this is the INVITEs SDP information to FS from the caller (a-leg) on a  working video call (initiated as video):
Media Description, name and address (m): video 55752 RTP/AVP 123 124
Bandwidth Information (b): TIAS:1574400
Media Attribute (a): rtpmap:123 H264/90000
Media Attribute (a): fmtp:123 profile-level-id=42801f;packetization-mode=0;level-asymmetry-allowed=1
Media Attribute (a): rtpmap:124 H264/90000
Media Attribute (a): fmtp:124 profile-level-id=42801f;packetization-mode=1;level-asymmetry-allowed=1
Media Attribute (a): sendrecv

200 OK from FS to the Caller (a-leg):
Media Description, name and address (m): video 16454 RTP/AVP 123
Media Attribute (a): rtpmap:123 H264/90000
Media Attribute (a): fmtp:123 profile-level-id=42801f;packetization-mode=0;level-asymmetry-allowed=1

Now, when I try to initiate a call as audio, then add video:

INVITE from the caller (a-leg) to FS ("adding" video during call):
Media Description, name and address (m): video 64698 RTP/AVP 123 124
Bandwidth Information (b): TIAS:1574400
Media Attribute (a): rtpmap:123 H264/90000
Media Attribute (a): fmtp:123 profile-level-id=42801f;packetization-mode=0;level-asymmetry-allowed=1
Media Attribute (a): rtpmap:124 H264/90000
Media Attribute (a): fmtp:124 profile-level-id=42801f;packetization-mode=1;level-asymmetry-allowed=1
Media Attribute (a): sendrecv

200 OK from FS to the caller (a-leg):
Media Description, name and address (m): video 23678 RTP/AVP 123
Media Attribute (a): rtpmap:123 H264/90000

So, it appears that the "fmtp" attribute is missing on the 200 OK response:
Media Attribute (a): fmtp:123 profile-level-id=42801f;packetization-mode=0;level-asymmetry-allowed=1

I can upload the captures somewhere if the full details will be helpful to anyone. If anyone has had any luck with this feature, I'm interested in how you accomplished it. I also tried using the setting "sip_force_video_fmtp" to no avail.

I appreciate any help.

Regards,

Tyler

From pkubat.lists at kubat.com  Wed Sep 28 16:58:56 2011
From: pkubat.lists at kubat.com (Kubes)
Date: Wed, 28 Sep 2011 08:58:56 -0400
Subject: [Freeswitch-users] Alarm System Central Office over VoIP
	(FreeSwitch)
Message-ID: <CANzfhcdvwcJwc-zWtmZE2wqG=XJtuhaFoHh40Y=QAfZk-kr_4g@mail.gmail.com>

All,
I am trying to get an alarm system which calls a central monitoring center
to work via FreeSwitch.  I am using a Linksys SPA-1001 and a SIP trunk
(Vitelity).   Even though the call is answered the "modems" never connect,
from what I can tell.  I see the call being answered in Freeswtich, but the
alarm system reports a "comm error".   PCMU is being used for the call and
it does hang-up with "NORMAL_CLEARING".

Does anyone have any ideas how to get "modems" to work over VoIP?  Any
suggestions would be appreciated.

Thanks
Kubes
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From darcy at primrose.ws  Wed Sep 28 21:50:27 2011
From: darcy at primrose.ws (Darcy)
Date: Wed, 28 Sep 2011 13:50:27 -0400
Subject: [Freeswitch-users] Alarm System Central Office over
	VoIP(FreeSwitch)
In-Reply-To: <CANzfhcdvwcJwc-zWtmZE2wqG=XJtuhaFoHh40Y=QAfZk-kr_4g@mail.gmail.com>
References: <CANzfhcdvwcJwc-zWtmZE2wqG=XJtuhaFoHh40Y=QAfZk-kr_4g@mail.gmail.com>
Message-ID: <7B9A08B704C549DEBA1F24E1E5033C8D@DWP>

We, the voiphighway, have about 50 alarms running over voip, it is a combination of echo cancellation, silent suppression, force g711 and the carrier you use. The setting can be different for each site, but you should disable echo cancellation and silent suppression, that is how we usually get it to function.

Otherwise it is a shot in the dark.  They are all running thru a freeswitch setup in what we call a tandem mode, the calls come from either another freeswitch pbx or pbxnsip hosted server and we relay the calls onwards.  Be sure to use bypass media, the less times the rtp touches something, the better chance you have of it working.
Darcy

From: Kubes 
Sent: Wednesday, September 28, 2011 8:58 AM
To: freeswitch-users at lists.freeswitch.org 
Subject: [Freeswitch-users] Alarm System Central Office over VoIP(FreeSwitch)

All,
I am trying to get an alarm system which calls a central monitoring center to work via FreeSwitch.  I am using a Linksys SPA-1001 and a SIP trunk (Vitelity).   Even though the call is answered the "modems" never connect, from what I can tell.  I see the call being answered in Freeswtich, but the alarm system reports a "comm error".   PCMU is being used for the call and it does hang-up with "NORMAL_CLEARING".

Does anyone have any ideas how to get "modems" to work over VoIP?  Any suggestions would be appreciated.

Thanks
Kubes




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

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From nbhatti at gmail.com  Wed Sep 28 22:49:33 2011
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Wed, 28 Sep 2011 21:49:33 +0300
Subject: [Freeswitch-users] How to distinguish between Authenticated sip
 and not authenticated users
In-Reply-To: <CAGX+XzsxHocAhqHf_D4Sd-Jj_OCqLq5z7NKz6LZQV0_gtmhyRg@mail.gmail.com>
References: <CAGX+XzsxHocAhqHf_D4Sd-Jj_OCqLq5z7NKz6LZQV0_gtmhyRg@mail.gmail.com>
Message-ID: <CACak_yzV0oG5MzAKB2NXud8ikuZa6AvxTwYxiqPmMKaTs_qU3g@mail.gmail.com>

Hello Anton,
I tried this before and ended up in the same situation. the event
variable_sip_authorized, for some reason always return true. What I
did in my cause, since I am using the variable_sip_auth_username to
match the username for my billing, this does the trick. I don't know
if this is a bug or something, but it works this way.

Thanks

On Wed, Sep 28, 2011 at 8:01 PM, Anton VG <anton.vazir at gmail.com> wrote:
> Guys,
>
> trying to distinguish between authenticated SIP user and not
> authenticated, still no luck
>
> I would like to allow anyone to call my FS users via SIP, so there are
> registered users and everyone else
>
> so I have allowed
>
> <param name="auth-calls" value="true"/>
> <param name="apply-inbound-acl" value="allow_all"/>
>
> acl.xml:
> ? ?<list name="allow_all" default="allow">
> ? ? ?<node type="allow" cidr="0.0.0.0/0"/>
> ? ?</list>
> Any call passes through, by always have
>
> [CHANNEL_CREATE] event
> variable_sip_authorized: true
>
> in event headers. But I would like to know if user is authorized or not.
>
> if I disable authentication by setting this to false
>
> <param name="auth-calls" value="false"/>
>
> all calls do not have any auth headers.
>
> Any clue?
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jack at livecall.com  Wed Sep 28 22:58:40 2011
From: jack at livecall.com (Jack)
Date: Wed, 28 Sep 2011 11:58:40 -0700
Subject: [Freeswitch-users] mod_rtmp and flex client
In-Reply-To: <CAK9Ws2fPY_Gy-tzSrcBqeneNs6msOCBhYejqQ3YFjT7U=OfNUA@mail.gmail.com>
References: <CAK9Ws2fPY_Gy-tzSrcBqeneNs6msOCBhYejqQ3YFjT7U=OfNUA@mail.gmail.com>
Message-ID: <4E836E60.4090901@livecall.com>

David,
I have mod_rtmp installed in a Win2003 environment this is my return 
when I do rtmp status:
default tcp:0.0.0.0:1935        profile

Make sure you include mod_rtmp in your build, it is not included by 
default.

Jack

On 9/27/2011 1:22 PM, David Wafula wrote:
> Am finally trying out flex client with mod_rtmp and i just realize how 
> little i know of this module. I have installed mod_rtmp  and loaded it 
> (or so i think). On the server, this is what i get:
>
>  freeswitch at internal>rtmp status
> -ERR no reply
>
> This doesn't look right for starters..what do i expect for a valid reply?
>
>
> To the client.
> It looks like the basic config on the flex client is to  update
> var flashvars = { xxxxxxxx} to point to my freeswitch server, no ?
>
> So, first i try:
> var flashvars = {
> rtmp_url: 'rtmp://conference.freeswitch.org/phone 
> <http://conference.freeswitch.org/phone>'
> };
>
> And it does connect.
>
> Say my freeswitch server is running at 1.2.3.4
>
> So, naturally i end up with
>
> var flashvars = {
> rtmp_url: 'rtmp://1.2.3.4 <http://1.2.3.4>'
> };
>
> Is that the way to do it? Because this gives me:
>
> netStatus: NetConnection.Connect.InvalidApp
> netStatus: NetConnection.Connect.Closed
>
> Any pointers will be appreciated.
>
>
> Thank you
> -- 
>
>
> 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
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From infos at madovsky.org  Wed Sep 28 23:48:03 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 28 Sep 2011 15:48:03 -0400
Subject: [Freeswitch-users] SIP client with H264
Message-ID: <38E21C7245CF47C78A27C10B6A81C0C0@e1705>

Hi all,

is anyone know which free SIP client is the best to
test H264 video ?

thanks

Franck
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From anthony.minessale at gmail.com  Wed Sep 28 23:49:36 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 28 Sep 2011 14:49:36 -0500
Subject: [Freeswitch-users] How to distinguish between Authenticated sip
 and not authenticated users
In-Reply-To: <CACak_yzV0oG5MzAKB2NXud8ikuZa6AvxTwYxiqPmMKaTs_qU3g@mail.gmail.com>
References: <CAGX+XzsxHocAhqHf_D4Sd-Jj_OCqLq5z7NKz6LZQV0_gtmhyRg@mail.gmail.com>
	<CACak_yzV0oG5MzAKB2NXud8ikuZa6AvxTwYxiqPmMKaTs_qU3g@mail.gmail.com>
Message-ID: <CAKbxfG9xs5Q94xNJXv6iRzFz4cp8PUPmQc2ZbpS5b5N-pnsRHg@mail.gmail.com>

This seems odd....

So you are making a global ACL to 0.0.0.0 which will let everyone in.
This takes precedence over the auth calls because you are passing the
ACL which is a means of authorization so you are never using the
actual challenge based auth at all.   Of course they will all appear
as sip_authorized=true

you do have the variable sip_acl_authed_by to tell which acl you passed.

This has come up today in irc too, its daft to try to mix authed and
non-auth calls on the same profile.
Just make your authenticated users use a dedicated profile and the
non-authenticated ones use another.



On Wed, Sep 28, 2011 at 1:49 PM, Muhammad Naseer Bhatti
<nbhatti at gmail.com> wrote:
> Hello Anton,
> I tried this before and ended up in the same situation. the event
> variable_sip_authorized, for some reason always return true. What I
> did in my cause, since I am using the variable_sip_auth_username to
> match the username for my billing, this does the trick. I don't know
> if this is a bug or something, but it works this way.
>
> Thanks
>
> On Wed, Sep 28, 2011 at 8:01 PM, Anton VG <anton.vazir at gmail.com> wrote:
>> Guys,
>>
>> trying to distinguish between authenticated SIP user and not
>> authenticated, still no luck
>>
>> I would like to allow anyone to call my FS users via SIP, so there are
>> registered users and everyone else
>>
>> so I have allowed
>>
>> <param name="auth-calls" value="true"/>
>> <param name="apply-inbound-acl" value="allow_all"/>
>>
>> acl.xml:
>> ? ?<list name="allow_all" default="allow">
>> ? ? ?<node type="allow" cidr="0.0.0.0/0"/>
>> ? ?</list>
>> Any call passes through, by always have
>>
>> [CHANNEL_CREATE] event
>> variable_sip_authorized: true
>>
>> in event headers. But I would like to know if user is authorized or not.
>>
>> if I disable authentication by setting this to false
>>
>> <param name="auth-calls" value="false"/>
>>
>> all calls do not have any auth headers.
>>
>> Any clue?
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 davidwaf at gmail.com  Thu Sep 29 00:39:15 2011
From: davidwaf at gmail.com (David Wafula)
Date: Wed, 28 Sep 2011 22:39:15 +0200
Subject: [Freeswitch-users] mod_rtmp and flex client
In-Reply-To: <4E836E60.4090901@livecall.com>
References: <CAK9Ws2fPY_Gy-tzSrcBqeneNs6msOCBhYejqQ3YFjT7U=OfNUA@mail.gmail.com>
	<4E836E60.4090901@livecall.com>
Message-ID: <CAK9Ws2fSMukqqBV2zc4mP+VBU_QyM5oCB5UZtn1is5Z=X_+hig@mail.gmail.com>

Thanks Jack. So found this in the logs:
2011-09-28 20:22:29.634103 [CRIT] rtmp_tcp.c:355 Socket error. Couldn't
listen on 0.0.0.0:1935

And turns out i was running red5 on same server, hence the port conflict. I
temporarily stopped red5 and can connect now...will sort out the conflict
later.
Regards.


On Wed, Sep 28, 2011 at 8:58 PM, Jack <jack at livecall.com> wrote:

>  David,
> I have mod_rtmp installed in a Win2003 environment this is my return when I
> do rtmp status:
> default tcp:0.0.0.0:1935        profile
>
> Make sure you include mod_rtmp in your build, it is not included by
> default.
>
> Jack
>
>
> On 9/27/2011 1:22 PM, David Wafula wrote:
>
> Am finally trying out flex client with mod_rtmp and i just realize how
> little i know of this module. I have installed mod_rtmp  and loaded it (or
> so i think). On the server, this is what i get:
>
>   freeswitch at internal>rtmp status
> -ERR no reply
>
>  This doesn't look right for starters..what do i expect for a valid reply?
>
>
> To the client.
> It looks like the basic config on the flex client is to  update
> var flashvars = { xxxxxxxx} to point to my freeswitch server, no ?
>
>  So, first i try:
>  var flashvars = {
>  rtmp_url: 'rtmp://conference.freeswitch.org/phone'
>  };
>
>  And it does connect.
>
>  Say my freeswitch server is running at 1.2.3.4
>
>  So, naturally i end up with
>
>  var flashvars = {
>  rtmp_url: 'rtmp://1.2.3.4'
>  };
>
>  Is that the way to do it? Because this gives me:
>
>  netStatus: NetConnection.Connect.InvalidApp
> netStatus: NetConnection.Connect.Closed
>
> Any pointers will be appreciated.
>
>
>  Thank you
> --
>
>
> David Wafula
>
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
David Wafula
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From anton.vazir at gmail.com  Thu Sep 29 00:55:33 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Thu, 29 Sep 2011 01:55:33 +0500
Subject: [Freeswitch-users] How to distinguish between Authenticated sip
 and not authenticated users
In-Reply-To: <CAKbxfG9xs5Q94xNJXv6iRzFz4cp8PUPmQc2ZbpS5b5N-pnsRHg@mail.gmail.com>
References: <CAGX+XzsxHocAhqHf_D4Sd-Jj_OCqLq5z7NKz6LZQV0_gtmhyRg@mail.gmail.com>
	<CACak_yzV0oG5MzAKB2NXud8ikuZa6AvxTwYxiqPmMKaTs_qU3g@mail.gmail.com>
	<CAKbxfG9xs5Q94xNJXv6iRzFz4cp8PUPmQc2ZbpS5b5N-pnsRHg@mail.gmail.com>
Message-ID: <CAGX+XzvCg0GR8Ph5-Lykhe3Fs_EWqfu7K7M1s3S=9AO-7Ah_bg@mail.gmail.com>

Brian have suggested on IRC to use dialplan auth via

<condition field="${sip_authorized}" expression="^true$" break="never">
     <anti-action application="respond" data="407"/>
</condition>

seems it can do the trick.

2011/9/29 Anthony Minessale <anthony.minessale at gmail.com>:
> This seems odd....
>
> So you are making a global ACL to 0.0.0.0 which will let everyone in.
> This takes precedence over the auth calls because you are passing the
> ACL which is a means of authorization so you are never using the
> actual challenge based auth at all. ? Of course they will all appear
> as sip_authorized=true
>
> you do have the variable sip_acl_authed_by to tell which acl you passed.
>
> This has come up today in irc too, its daft to try to mix authed and
> non-auth calls on the same profile.
> Just make your authenticated users use a dedicated profile and the
> non-authenticated ones use another.
>
>
>
> On Wed, Sep 28, 2011 at 1:49 PM, Muhammad Naseer Bhatti
> <nbhatti at gmail.com> wrote:
>> Hello Anton,
>> I tried this before and ended up in the same situation. the event
>> variable_sip_authorized, for some reason always return true. What I
>> did in my cause, since I am using the variable_sip_auth_username to
>> match the username for my billing, this does the trick. I don't know
>> if this is a bug or something, but it works this way.
>>
>> Thanks
>>
>> On Wed, Sep 28, 2011 at 8:01 PM, Anton VG <anton.vazir at gmail.com> wrote:
>>> Guys,
>>>
>>> trying to distinguish between authenticated SIP user and not
>>> authenticated, still no luck
>>>
>>> I would like to allow anyone to call my FS users via SIP, so there are
>>> registered users and everyone else
>>>
>>> so I have allowed
>>>
>>> <param name="auth-calls" value="true"/>
>>> <param name="apply-inbound-acl" value="allow_all"/>
>>>
>>> acl.xml:
>>> ? ?<list name="allow_all" default="allow">
>>> ? ? ?<node type="allow" cidr="0.0.0.0/0"/>
>>> ? ?</list>
>>> Any call passes through, by always have
>>>
>>> [CHANNEL_CREATE] event
>>> variable_sip_authorized: true
>>>
>>> in event headers. But I would like to know if user is authorized or not.
>>>
>>> if I disable authentication by setting this to false
>>>
>>> <param name="auth-calls" value="false"/>
>>>
>>> all calls do not have any auth headers.
>>>
>>> Any clue?
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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  Thu Sep 29 04:23:43 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Sep 2011 17:23:43 -0700
Subject: [Freeswitch-users] mod_conference.c:6979 Open of
	conference.conf failed
In-Reply-To: <CAGH8SeYA8U94Uc=CZ_ij4LftigvC2ND7VwfHHULYJzH4oDH4Hg@mail.gmail.com>
References: <CAGH8SeYA8U94Uc=CZ_ij4LftigvC2ND7VwfHHULYJzH4oDH4Hg@mail.gmail.com>
Message-ID: <CAKzWOxUnaiV6rbu+yNdcpDSmkV0b+H8-p35C-2x6szK9wKbH5w@mail.gmail.com>

Look in your config directory, which should be /usr/local/freeswitch/conf
You should have a bunch of stuff in there. If not, go back to your source
directory and do "make install" or "make samples" (the first one does all of
freeswitch, the second one does only the default install)

Let us know how it goes.

-MC

2011/9/27 Henrik Aagaard S?rensen <henrikaagaardsorensen at gmail.com>

> I've installed FS via
> http://wiki.freeswitch.org/wiki/Linux_Quick_Install_Guide
>
> Everything seems to run just fine, but when starting up I get the error:
> 2011-09-27 14:03:16.336176 [ERR] mod_conference.c:6979 Open of
> conference.conf failed
>
> Can anyone help me out?
>
> I guess I don't have a conference.conf(.xml) file. Should I? Should it be
> installed via the guide or...?
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Sep 29 04:26:40 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Sep 2011 17:26:40 -0700
Subject: [Freeswitch-users] fifo timeout?
In-Reply-To: <4E821D9B.30204@communicatefreely.net>
References: <4E821D9B.30204@communicatefreely.net>
Message-ID: <CAKzWOxXBi2dfQD11=+=1whpAm4eSNvJj_Ok-EW4Nu+4K=A+5eQ@mail.gmail.com>

You probably want the fifo_orbit_exten chan var. See here:
http://wiki.freeswitch.org/wiki/Mod_fifo#This_channel_uses_or_sets.2C_for_the_caller_leg_entering_the_FIFO

-MC

On Tue, Sep 27, 2011 at 12:01 PM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> Hello,
>
> I'm building a very simple queue system with mod_fifo.  For the most
> part, things are working the way I want, but I can't figure out how to
> limit the amount of time someone can sit in the queue.  I'm trying to
> implement something like max_wait would do in mod_callcenter.
>
> I can use the orbit_exten option, but it wants to transfer to a
> different extension.  I really just want the fifo application to exit so
> the caller can progress to the next dialplan action.  Is there a way to
> make the caller exit the fifo if they are unanswered after a set amount
> of time?
>
> I did give mod_callcenter a try, but I had a lot of problems with calls
> getting stuck on hold, with no agents ringing, so I went to mod_fifo as
> a simpler solution.
>
> Thanks for any suggestions.
>
> -Tim
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Sep 29 05:18:56 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Sep 2011 18:18:56 -0700
Subject: [Freeswitch-users] Outbound calling with recorded message.
In-Reply-To: <F91D0399DF064C4193AA40A6BB059076@clancysystems.com>
References: <F91D0399DF064C4193AA40A6BB059076@clancysystems.com>
Message-ID: <CAKzWOxXOXM_Y7X2QcETKUpr7_yQ9mQYOBzkWAVonsH-8XYNrkA@mail.gmail.com>

Once you have the calls logged into a database then it is a relatively
simple matter to generate the outbound calls using the event socket. The
real challenge (IMHO) is accounting for the outbound calls that don't
actually reach the target, or that go to the target's voicemail, etc.

For the sake of simplicity, let's assume that each person you call will
answer. From there you just need a simple dialplan extension that does a
record app with a specific filename. (You need to match up the filename
recorded with the person you called. You could use <phone_num>.wav I
suppose.) From there it's a matter of launching the calls. I don't do much
with Windows but there are plenty of folks here who do. If you can establish
an event socket connection then you can execute a bunch of "originate" API
calls to generate your outbound calls.

Let's say your dialplan extension for recording the name is this:
<extension name="get their name">
  <condition field="destination_number"
expression="^OB_IVR_Record_Name_(\d+)$">
    <action application="answer"/>
    <action application="sleep" data="1000"/>
    <action application="playback" data="please_record_your_name.wav"/>
    <action application="playback" data="tone_stream://%(1000,0,1500)"/>
    <action application="record" data="/tmp/$1.wav 25 200"/>
  </condition>
</extension>

You can generate a call with this API:

originate sofia/gw/gwname/18005551212 OB_IVR_Record_Name_8005551212

If it works then you'll end up with /tmp/8005551212.wav and hopefully
they'll have given you the info you need.

As for generating these calls, if you don't need something too fancy you
could just use a program written in the scripting language of your choice.
(Perl, Python, Ruby, and PHP are all suitable for this task.) You could also
write "real" program with Visual Studio if that suits you. The key is that
you will need to keep track of what happens when you make all these calls
and be sure not to keep calling them over and over again. :)

I've done this sort of thing with just Perl scripts and it works really
well.

-MC

On Wed, Sep 28, 2011 at 10:35 AM, Dave <dave at clancysystems.com> wrote:

> **
>
> Hi all,
>
> I have been using freeSWITCH for a while for inbound calls in which a
> person registers for a seminar via IVR.  I simply use the DID, CID number
> and name to identify the person and put them in the Database. That part of
> the application is working wonderfully.
>
> The issue I'm presented with now is that we need automate making calls to a
> few of these registrants, after each event, whos caller_id_name comes in as
> "Unknown" or "Wireless Caller", Play a recording to let them know we need
> the name, and that they can record the information right then, or call the
> office.
>
> I am limited to using a Windows 7 machine.
>
> What would be the best tool to use to automate these calls with the use of
> an IVR?
>
> Dave Goodwin
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From henrikaagaardsorensen at gmail.com  Thu Sep 29 06:45:57 2011
From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=)
Date: Thu, 29 Sep 2011 04:45:57 +0200
Subject: [Freeswitch-users] mod_conference.c:6979 Open of
	conference.conf failed
In-Reply-To: <CAKzWOxUnaiV6rbu+yNdcpDSmkV0b+H8-p35C-2x6szK9wKbH5w@mail.gmail.com>
References: <CAGH8SeYA8U94Uc=CZ_ij4LftigvC2ND7VwfHHULYJzH4oDH4Hg@mail.gmail.com>
	<CAKzWOxUnaiV6rbu+yNdcpDSmkV0b+H8-p35C-2x6szK9wKbH5w@mail.gmail.com>
Message-ID: <CAGH8SeZ5oTSFOK84YjQb_w7y=Sjpohf9j7BQa2A8X6tyW3XtGw@mail.gmail.com>

I just created a conference.conf.xml and everything works now.
On Sep 28, 2011 8:29 PM, "Michael Collins" <msc at freeswitch.org> wrote:
> Look in your config directory, which should be /usr/local/freeswitch/conf
> You should have a bunch of stuff in there. If not, go back to your source
> directory and do "make install" or "make samples" (the first one does all
of
> freeswitch, the second one does only the default install)
>
> Let us know how it goes.
>
> -MC
>
> 2011/9/27 Henrik Aagaard S?rensen <henrikaagaardsorensen at gmail.com>
>
>> I've installed FS via
>> http://wiki.freeswitch.org/wiki/Linux_Quick_Install_Guide
>>
>> Everything seems to run just fine, but when starting up I get the error:
>> 2011-09-27 14:03:16.336176 [ERR] mod_conference.c:6979 Open of
>> conference.conf failed
>>
>> Can anyone help me out?
>>
>> I guess I don't have a conference.conf(.xml) file. Should I? Should it be
>> installed via the guide or...?
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
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From xing2kin at yahoo.com  Thu Sep 29 06:52:42 2011
From: xing2kin at yahoo.com (king2kin)
Date: Wed, 28 Sep 2011 19:52:42 -0700 (PDT)
Subject: [Freeswitch-users] FsGUI.exe on windows 2003 server: no
	available plugin
In-Reply-To: <CABH8QX3tmNTEhhQq40bvM8cHZa6TgorOKJKN9vJNhTkiaxhdCA@mail.gmail.com>
References: <1317192465.66397.YahooMailNeo@web39701.mail.mud.yahoo.com>
	<CABH8QX3tmNTEhhQq40bvM8cHZa6TgorOKJKN9vJNhTkiaxhdCA@mail.gmail.com>
Message-ID: <1317264762.4235.YahooMailNeo@web39702.mail.mud.yahoo.com>

Hi Joao,
?
Yes, I do have the folder [plugins] inside the folder where I have the fsgui.exe. The directory structure where I installed FsGui is as follows:
?
[C:\FreeSWITCH]: this is the root directory where [FsGui] was installed, and ?those exe files are located;
[C:\FreeSWITCH\plugins]: it contains only one dll file "consoleplugin.dll".
?
x.k. 

From: Jo?o Mesquita <jmesquita at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, September 28, 2011 9:19 PM
Subject: Re: [Freeswitch-users] FsGUI.exe on windows 2003 server: no available plugin


Do you have the plugins folder inside the folder where you have the .exe?

Regards,Jo?o Mesquita




On Wed, Sep 28, 2011 at 3:47 AM, king2kin <xing2kin at yahoo.com> wrote:

Hi, 
>
>I downloaded windows version of FsGui.exe on http://wiki.freeswitch.org/wiki/Fsgui?.
>
>After installing it, I tried to run it on windows 2003 server, it poped up an error dialog saying
>"no available plugins. Install conpatible plugins before running fsgui". Here I wonder what else I should install to run fsgui.
>
>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
>
>


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From xing2kin at yahoo.com  Thu Sep 29 07:11:16 2011
From: xing2kin at yahoo.com (king2kin)
Date: Wed, 28 Sep 2011 20:11:16 -0700 (PDT)
Subject: [Freeswitch-users] google voice connection going offline
	(mod_dingaling)
In-Reply-To: <CAKrPhPMsaXaTPi498hS-=Xo2Y3A6tZRCcx4p=waV0Sfvw7W8ug@mail.gmail.com>
References: <CAKrPhPMsaXaTPi498hS-=Xo2Y3A6tZRCcx4p=waV0Sfvw7W8ug@mail.gmail.com>
Message-ID: <1317265876.9229.YahooMailNeo@web39703.mail.mud.yahoo.com>

I also had problem in making FS mod_dingaling work with google voice: 

- enable mod_dingaling in modules.conf.xml at first;

- copy the following content into?disk file?[freeswitch/conf/jingle_profiles/xk_client.xml]?according to wiki page ?[http://wiki.freeswitch.org/wiki/Google_Voice]:

<include>
? <profile type="client">
??? <param name="name" value="gtalk"/>
??? <param name="login" value="GV-ACCOUNT-NAME at gmail.com/talk"/>
??? <param name="password" value="MY-PASSWORD"/>
??? <param name="server" value="talk.google.com"/>
??? <param name="dialplan" value="XML"/>
??? <param name="context" value="public"/>
??? <param name="message" value="Hello from Freeswitch!"/>
??? <param name="rtp-ip" value="$${bind_server_ip}"/>
??? <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
??? <param name="auto-login" value="true"/>
??? <!-- SASL "plain" or "md5? -->
??? <param name="sasl" value="plain"/>
??? <!-- Enable TLS or not -->
??? <param name="tls" value="true"/>
??? <!-- disable to trade async for more calls -->
??? <param name="use-rtp-timer" value="true"/>
??? <param name="rtp-timer-name" value="none"/>
??? <!-- default extension -->
??? <param name="exten" value="2001"/>
??? <param name="vad" value="both"/>
??? <param name="candidate-acl" value="wan.auto"/>?????????????????????? 
??? <param name="local-network-acl" value="localnet.auto"/>
? </profile>
</include>

- then, since [freeswitch/conf/jingle_profiles/] already contains cleint.xml and server.xml, ?I made four changes to the above content of xk_client.xml:

<profile type="client">? ===>? <profile type="xk_client">

??? <param name="login" value="GV-ACCOUNT-NAME at gmail.com/talk"/>? ===>?update it with my own gmail account
??? <param name="password" value="MY-PASSWORD"/> ===> update it with my own password to gmail account.
<param name="exten" value="2001"/> ===> <param name="exten" value="1006"/>

- add a dialplan extension for default context:

?<extension name="gvoice_out">
?? <condition field="destination_number" expression="^50(1\d{10})$">
???? <action application="set" data="hangup_after_bridge=true"/>
???? <action application="bridge" data="dingaling/gtalk/+$1 at voice.google.com"/>
?? </condition>
</extension>

-?also, set up my google voice settings as required by FS

- finally, dial out "50 1 408 543 xxxx" on a sip client with internal user "1006",??the call? was never completed.?
?
?


----- Original Message -----
From: Federico Beffa <beffa at ieee.org>
To: freeswitch-users at lists.freeswitch.org
Cc: 
Sent: Wednesday, September 28, 2011 4:14 PM
Subject: [Freeswitch-users] google voice connection going offline (mod_dingaling)

Hi All,

I've setup mod_dingaling to connect to my google voice account
according to the instructions on the wiki
http://wiki.freeswitch.org/wiki/Google_Voice#Setup_FreeSWITCH_-_Dingaling_to_work_with_your_Gmail_account
with two modifications:
<param name="rtp-ip" value="auto"/>
<param name="ext-rtp-ip" value="auto-nat"/>

I use the same IP settings for SIP (sofia) with no problem.

The setup works fine for the first hour or two, but then I can see
from another google account that the connection established by
freeswitch goes offline. If I try anyway to make an outbound call,
then the other end does not ring (even if I wait for a long time).
However, after some minutes the freeswitch account returns online and
works again for a couple of hours.

I'm behind a nat and firewall. Is there any port which I have to open
to keep freeswitch online? As far as I understand only the google XMMP
server needs to listen to port 5222 and port 5269.
Or, is there any kind of "keep alive" timer available?

The google talk client running in the same environment works flawlessly.

Thanks for any advise.
Fede


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From jmesquita at freeswitch.org  Thu Sep 29 07:54:40 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 29 Sep 2011 00:54:40 -0300
Subject: [Freeswitch-users] FsGUI.exe on windows 2003 server: no
	available plugin
In-Reply-To: <1317264762.4235.YahooMailNeo@web39702.mail.mud.yahoo.com>
References: <1317192465.66397.YahooMailNeo@web39701.mail.mud.yahoo.com>
	<CABH8QX3tmNTEhhQq40bvM8cHZa6TgorOKJKN9vJNhTkiaxhdCA@mail.gmail.com>
	<1317264762.4235.YahooMailNeo@web39702.mail.mud.yahoo.com>
Message-ID: <CABH8QX1bUJ79zE7q4s4c9qxJY13Zke6z-tnVcQNHFkOoTep_bg@mail.gmail.com>

This is really strange. The way the app is designed is that we can develop
more plugins to the console and load them on runtime. Much like what
FreeSWITCH does with its modules (but in a muuuuch more rudimentary way).

When you open the app, you should see the directory in which the application
is looking for its plugins. Do you see the correct path?

If you don't, try setting the correct path. Please let me know the results
of that.

I am adding logging to the application to help me debug stuff like that.

Regards,
Jo?o Mesquita



On Wed, Sep 28, 2011 at 11:52 PM, king2kin <xing2kin at yahoo.com> wrote:

> Hi Joao,
>
> Yes, I do have the folder [plugins] inside the folder where I have the
> fsgui.exe. The directory structure where I installed FsGui is as follows:
>
> [C:\FreeSWITCH]: this is the root directory where [FsGui] was installed,
> and  those exe files are located;
> [C:\FreeSWITCH\plugins]: it contains only one dll file "consoleplugin.dll".
>
> x.k.
>
>  *From:* Jo?o Mesquita <jmesquita at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, September 28, 2011 9:19 PM
> *Subject:* Re: [Freeswitch-users] FsGUI.exe on windows 2003 server: no
> available plugin
>
>  Do you have the plugins folder inside the folder where you have the .exe?
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Wed, Sep 28, 2011 at 3:47 AM, king2kin <xing2kin at yahoo.com> wrote:
>
>  Hi,
>
> I downloaded windows version of FsGui.exe on
> http://wiki.freeswitch.org/wiki/Fsgui .
>
> After installing it, I tried to run it on windows 2003 server, it poped up
> an error dialog saying
> "no available plugins. Install conpatible plugins before running fsgui".
> Here I wonder what else I should install to run fsgui.
>
> 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
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Thu Sep 29 07:57:26 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 29 Sep 2011 00:57:26 -0300
Subject: [Freeswitch-users] FSGUI
In-Reply-To: <CALNZuwng=J4PO+QwqBigJxbbp3jQBkBOeHqESmU=KRuAQmP_Cg@mail.gmail.com>
References: <CAGRr05_efk_i211d8eNBWm_2tDdVxRKzCjcHMwJg6ErRX8Uoxg@mail.gmail.com>
	<CABH8QX3E7UnSb5E1=s4uz=8u0j1uBNyk19wnarJq--zwgJ4xLA@mail.gmail.com>
	<CALNZuwng=J4PO+QwqBigJxbbp3jQBkBOeHqESmU=KRuAQmP_Cg@mail.gmail.com>
Message-ID: <CABH8QX3cEo5wPnzaW_oJ_2C847VMQmtwwuWJjWvnOwO7WbgFrQ@mail.gmail.com>

Avi, the I have not forgotten on adding wiki details on how to compile the
app. It is just that the build system was using SVN still!! Long time since
I haven't touched the app.

So, I need to fix some (several things actually) to get it to build
correctly without having to mangle around qmake files. Hold on tight, I will
get it sometime soon.

I did compile a version for Ubuntu 11.04 and sent to Brad Mina to test. It
failed with loading the plugins. I am still trying to figure out why. Once I
do, I will send you that version meanwhile.

Thanks for the support!

Regards,
Jo?o Mesquita



On Tue, Sep 27, 2011 at 12:51 PM, Avi Marcus <avi at avimarcus.net> wrote:

> I'm running on ubuntu (11.04) 32bit.
> I'm no stranger to compiling (thank FS!) but I simply didn't see how,
> given the links/info on the wiki.
>
> -Avi
>
>
> 2011/9/27 Jo?o Mesquita <jmesquita at freeswitch.org>:
> > Hello Brad,
> > First of all, let me say that I am flattered that someone actually uses
> > FSGui. I thought there simply was no interest in it whatsoever.
> > The linux version is not that easy to "package" as the Windows version.
> The
> > best way is to compile it yourself. I can even do it for you. What arch
> are
> > you running on? 64 or 32?
> > Regards,
> > Jo?o Mesquita (aka jmesquita)
> >
> >
> >
> > On Mon, Sep 26, 2011 at 6:34 PM, Brad Mina <brad at tech21.com> wrote:
> >>
> >> I've stumbled across fsgui a few times, (
> >> http://wiki.freeswitch.org/wiki/Fsgui ), and am happy to report the
> windows
> >> release of this program works flawlessly!
> >>
> >> After toying with the windows version myself and a few other community
> >> members wanted to see this application running in linux as well.
> >> I downloaded the fsgui.tar.gz and unzipped it to /tmp then installed the
> >> following packages on Ubuntu 10.04, which also installed related
> >> dependencies:
> >> libqt4-dev
> >> libqt4-core
> >> I tried to chmod+x and execute the application with ./fsgui - however
> bash
> >> reports the following:
> >>
> >> fsgui: cannot execute binary file
> >>
> >> Is there something I'm missing here? I'm hoping jmesquita or a fellow
> >> community member can respond with insight.
> >>
> >> 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
> >>
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From xing2kin at yahoo.com  Thu Sep 29 10:24:44 2011
From: xing2kin at yahoo.com (king2kin)
Date: Wed, 28 Sep 2011 23:24:44 -0700 (PDT)
Subject: [Freeswitch-users] FsGUI.exe on windows 2003 server: no
	available plugin
In-Reply-To: <CABH8QX1bUJ79zE7q4s4c9qxJY13Zke6z-tnVcQNHFkOoTep_bg@mail.gmail.com>
References: <1317192465.66397.YahooMailNeo@web39701.mail.mud.yahoo.com>
	<CABH8QX3tmNTEhhQq40bvM8cHZa6TgorOKJKN9vJNhTkiaxhdCA@mail.gmail.com>
	<1317264762.4235.YahooMailNeo@web39702.mail.mud.yahoo.com>
	<CABH8QX1bUJ79zE7q4s4c9qxJY13Zke6z-tnVcQNHFkOoTep_bg@mail.gmail.com>
Message-ID: <1317277484.60423.YahooMailNeo@web39706.mail.mud.yahoo.com>

As I recall, the directory in which the application is looking for its plugins seems to point to the root directory (e.g. C:\FreeSWITCH) ?of FsGUI when I started it first time after installation. 
?
Then, I forced it to point to the folder "C:\FreeSWITCH\plugins"; so now it always points to "C:\FreeSWITCH\plugins" whenever it's started, and no error dialog pops up.? However, if I click on button [ok] at the bottom, it asks me "Do you really want to close the FsGUI? yes or no" inside a separate pop-up window. 
Why doesn't FsGUI start to run when clicking on button [ok] at the bottom of main window?
?
I have to click on item [console, ?the application will do the same ...] in order to enable the button [Run].
Once I click on button [Run], I can add or edit server entry inside Server Manager window. 
Question:?how do I fill the blank [Password] to connect my FreeSwitch server? 
I got error: connection failed! socket connection error.
?

From: Jo?o Mesquita <jmesquita at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, September 29, 2011 11:54 AM
Subject: Re: [Freeswitch-users] FsGUI.exe on windows 2003 server: no available plugin


This is really strange. The way the app is designed is that we can develop more plugins to the console and load them on runtime. Much like what FreeSWITCH does with its modules (but in a muuuuch more rudimentary way). 

When you open the app, you should see the directory in which the application is looking for its plugins. Do you see the correct path?

If you don't, try setting the correct path. Please let me know the results of that.

I am adding logging to the application to help me debug stuff like that.

Regards,
Jo?o Mesquita




On Wed, Sep 28, 2011 at 11:52 PM, king2kin <xing2kin at yahoo.com> wrote:

Hi Joao,
>?
>Yes, I do have the folder [plugins] inside the folder where I have the fsgui.exe. The directory structure where I installed FsGui is as follows:
>?
>[C:\FreeSWITCH]: this is the root directory where [FsGui] was installed, and ?those exe files are located;
>[C:\FreeSWITCH\plugins]: it contains only one dll file "consoleplugin.dll".
>?
>x.k. 
>
>
>From: Jo?o Mesquita <jmesquita at freeswitch.org>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>Sent: Wednesday, September 28, 2011 9:19 PM
>Subject: Re: [Freeswitch-users] FsGUI.exe on windows 2003 server: no available plugin
>
>
>
>Do you have the plugins folder inside the folder where you have the .exe?
>
>
>Regards,Jo?o Mesquita
>
>
>
>
>On Wed, Sep 28, 2011 at 3:47 AM, king2kin <xing2kin at yahoo.com> wrote:
>
>Hi, 
>>
>>I downloaded windows version of FsGui.exe on http://wiki.freeswitch.org/wiki/Fsgui?.
>>
>>After installing it, I tried to run it on windows 2003 server, it poped up an error dialog saying
>>"no available plugins. Install conpatible plugins before running fsgui". Here I wonder what else I should install to run fsgui.
>>
>>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
>>
>>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From beffa at ieee.org  Thu Sep 29 11:08:40 2011
From: beffa at ieee.org (Federico Beffa)
Date: Thu, 29 Sep 2011 09:08:40 +0200
Subject: [Freeswitch-users] google voice connection going offline
	(mod_dingaling)
In-Reply-To: <1317265876.9229.YahooMailNeo@web39703.mail.mud.yahoo.com>
References: <CAKrPhPMsaXaTPi498hS-=Xo2Y3A6tZRCcx4p=waV0Sfvw7W8ug@mail.gmail.com>
	<1317265876.9229.YahooMailNeo@web39703.mail.mud.yahoo.com>
Message-ID: <CAKrPhPPxA648-Gq=_56wQN94-c+XDGHvHnTE=pxvWgSD7s1y8A@mail.gmail.com>

Thanks for the suggestion. That is about what I did and calls work in
both directions: outbound and inbound. This however only for a couple
of hours. Then the google account assigned to FS goes offline and
calls stop working.

I have tried to track events and noticed the following:

*) fs_cli with log to 7 does not report anything when the account goes offline.

*) my NAT/Firewall sends the following message "Firewall session time
out, sent TCP RST: TCP" to both my FS host and the google server. This
happens to the google talk client from google as well (running on
another machine on the same network). However, while the latter
handles the situation well and is able to keep the account online, it
appears that FS fails to act properly.

Is it possible to set a "keep alive" kind of timer in FS?

Thanks,
Fede

On Thu, Sep 29, 2011 at 5:11 AM, king2kin <xing2kin at yahoo.com> wrote:
> I also had problem in making FS mod_dingaling work with google voice:
>
> - enable mod_dingaling in modules.conf.xml at first;
>
> - copy the following content into?disk
> file?[freeswitch/conf/jingle_profiles/xk_client.xml]?according to wiki page
> ?[http://wiki.freeswitch.org/wiki/Google_Voice]:
>
> <include>
> ? <profile type="client">
> ??? <param name="name" value="gtalk"/>
> ??? <param name="login" value="GV-ACCOUNT-NAME at gmail.com/talk"/>
> ??? <param name="password" value="MY-PASSWORD"/>
> ??? <param name="server" value="talk.google.com"/>
> ??? <param name="dialplan" value="XML"/>
> ??? <param name="context" value="public"/>
> ??? <param name="message" value="Hello from Freeswitch!"/>
> ??? <param name="rtp-ip" value="$${bind_server_ip}"/>
> ??? <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> ??? <param name="auto-login" value="true"/>
> ??? <!-- SASL "plain" or "md5? -->
> ??? <param name="sasl" value="plain"/>
> ??? <!-- Enable TLS or not -->
> ??? <param name="tls" value="true"/>
> ??? <!-- disable to trade async for more calls -->
> ??? <param name="use-rtp-timer" value="true"/>
> ??? <param name="rtp-timer-name" value="none"/>
> ??? <!-- default extension -->
> ??? <param name="exten" value="2001"/>
> ??? <param name="vad" value="both"/>
> ??? <param name="candidate-acl" value="wan.auto"/>
> ??? <param name="local-network-acl" value="localnet.auto"/>
> ? </profile>
> </include>
> - then, since [freeswitch/conf/jingle_profiles/] already contains cleint.xml
> and server.xml, ?I made four changes to the above content of xk_client.xml:
>
> <profile type="client">? ===>? <profile type="xk_client">
>
> ??? <param name="login" value="GV-ACCOUNT-NAME at gmail.com/talk"/>
> ===>?update it with my own gmail account
> ??? <param name="password" value="MY-PASSWORD"/> ===> update it with my own
> password to gmail account.
> <param name="exten" value="2001"/> ===> <param name="exten" value="1006"/>
> - add a dialplan extension for default context:
>
> ?<extension name="gvoice_out">
> ?? <condition field="destination_number" expression="^50(1\d{10})$">
> ???? <action application="set" data="hangup_after_bridge=true"/>
> ???? <action application="bridge"
> data="dingaling/gtalk/+$1 at voice.google.com"/>
> ?? </condition>
> </extension>
> -?also, set up my google voice settings as required by FS
>
> - finally, dial out "50 1 408 543 xxxx" on a sip client with internal user
> "1006",??the call? was never completed.
>
>
>
>
> ----- Original Message -----
> From: Federico Beffa <beffa at ieee.org>
> To: freeswitch-users at lists.freeswitch.org
> Cc:
> Sent: Wednesday, September 28, 2011 4:14 PM
> Subject: [Freeswitch-users] google voice connection going offline
> (mod_dingaling)
>
> Hi All,
>
> I've setup mod_dingaling to connect to my google voice account
> according to the instructions on the wiki
> http://wiki.freeswitch.org/wiki/Google_Voice#Setup_FreeSWITCH_-_Dingaling_to_work_with_your_Gmail_account
> with two modifications:
> <param name="rtp-ip" value="auto"/>
> <param name="ext-rtp-ip" value="auto-nat"/>
>
> I use the same IP settings for SIP (sofia) with no problem.
>
> The setup works fine for the first hour or two, but then I can see
> from another google account that the connection established by
> freeswitch goes offline. If I try anyway to make an outbound call,
> then the other end does not ring (even if I wait for a long time).
> However, after some minutes the freeswitch account returns online and
> works again for a couple of hours.
>
> I'm behind a nat and firewall. Is there any port which I have to open
> to keep freeswitch online? As far as I understand only the google XMMP
> server needs to listen to port 5222 and port 5269.
> Or, is there any kind of "keep alive" timer available?
>
> The google talk client running in the same environment works flawlessly.
>
> Thanks for any advise.
> Fede
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 asilva at wirelessmundi.com  Thu Sep 29 13:44:43 2011
From: asilva at wirelessmundi.com (Antonio)
Date: Thu, 29 Sep 2011 11:44:43 +0200
Subject: [Freeswitch-users] Problem with Sangoma D100 Transconding card
In-Reply-To: <CAA4nhyBM0cRr472xe+-XZ_2fsZieP1cWWGPGgnRbPc_qHoscsA@mail.gmail.com>
References: <1317116551.12197.21.camel@marces.madrid.commsmundi.com>
	<CAA4nhyBM0cRr472xe+-XZ_2fsZieP1cWWGPGgnRbPc_qHoscsA@mail.gmail.com>
Message-ID: <1317289483.4860.20.camel@marces.madrid.commsmundi.com>

Hi,

The log when this error occurs can be found in
http://pastebin.freeswitch.org/17431 . 



On Wed, 2011-09-28 at 11:33 -0400, Moises Silva wrote: 

> On Tue, Sep 27, 2011 at 5:42 AM, Antonio <asilva at wirelessmundi.com>
> wrote:
> 
>         Hi,
>         
>         I'm using a sangoma D100 transconding card for the codec g729,
>         it works perfectly until recently that it stops working with
>         the following error:
>         
>         
>         2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:1095
>         sngtc_get_existing_rtp_session
>         2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:1095
>         sngtc_get_free_rtp_session
>         2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:224 New
>         allocated port 13284 for IP 10.254.254.1/10.254.254.1
>         2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:224 New
>         allocated port 13286 for IP 10.254.254.1/10.254.254.1
>         2011-09-27 10:32:45.192555 [ERR] mod_sangoma_codec.c:1106
>         Create Transcoding Session Error
>         2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:243
>         Released port 13284 for IP 10.254.254.1/10.254.254.1
>         2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:243
>         Released port 13286 for IP 10.254.254.1/10.254.254.1
>         2011-09-27 10:32:45.192555 [DEBUG] mod_sangoma_codec.c:1095
>         sngtc_release_rtp_session
>         2011-09-27 10:32:45.192555 [ERR] mod_sangoma_codec.c:486
>         Failed to create Sangoma encoding session.
>         2011-09-27 10:32:45.192555 [ERR] switch_core_io.c:1060 Codec
>         Sangoma G729 encoder error!
>         
>         
>         This error has trigger by the stop/start of freeswitch. 
>         
>         I had to restart the sangoma daemon (sngtc_server) to make it
>         work again.
>         
>         
>         Do i have to restart the sangoma daemon every time i restart
>         the freeswitch? shouldn't they be independent? 
>         
> 
> 
> They are independent. If you can reproduce this problem, pastebin the
> log /var/log/sngtc_server.log output when the error happens.
> 
> 
> 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
> 
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From ocset at the800group.com  Thu Sep 29 17:43:10 2011
From: ocset at the800group.com (ocset)
Date: Thu, 29 Sep 2011 21:43:10 +0800
Subject: [Freeswitch-users] Hardware - PC vs blackbox vs server
Message-ID: <4E8475EE.2080408@the800group.com>

Hi

I have successfully built a FreeSwitch system and just wanted to get 
some input from those who have already deployed a system as to your 
hardware of choice (just the OS hardware - will be running Linux)?

Is there someone who makes a good slimline blackbox solution or should I 
just put together my own machine. I will be using all external devices 
(no internal FXO/FXS cards) so there should be no need to worry about 
which is the best mobo etc.

My concern is that a standard PC is not meant to be left running 24/7 
(reliability) while a server may be too expensive and large for the task?

All comments greatly appreciated.


From gmaruzz at gmail.com  Thu Sep 29 17:49:05 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 29 Sep 2011 15:49:05 +0200
Subject: [Freeswitch-users] Hardware - PC vs blackbox vs server
In-Reply-To: <4E8475EE.2080408@the800group.com>
References: <4E8475EE.2080408@the800group.com>
Message-ID: <CALXCt0rLjnc+E0CvzwkoASHhPsZ7=H28gJ=nATHJKFaWdnsCkQ@mail.gmail.com>

a WRT device is probably what you're looking for.

Search the mailing list and the wiki for info and instructions.

-giovanni

On Thu, Sep 29, 2011 at 3:43 PM, ocset <ocset at the800group.com> wrote:
> Hi
>
> I have successfully built a FreeSwitch system and just wanted to get
> some input from those who have already deployed a system as to your
> hardware of choice (just the OS hardware - will be running Linux)?
>
> Is there someone who makes a good slimline blackbox solution or should I
> just put together my own machine. I will be using all external devices
> (no internal FXO/FXS cards) so there should be no need to worry about
> which is the best mobo etc.
>
> My concern is that a standard PC is not meant to be left running 24/7
> (reliability) while a server may be too expensive and large for the task?
>
> All comments greatly appreciated.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From roger.castaldo at gmail.com  Thu Sep 29 17:58:50 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Thu, 29 Sep 2011 09:58:50 -0400
Subject: [Freeswitch-users] Hardware - PC vs blackbox vs server
In-Reply-To: <CALXCt0rLjnc+E0CvzwkoASHhPsZ7=H28gJ=nATHJKFaWdnsCkQ@mail.gmail.com>
References: <4E8475EE.2080408@the800group.com>
	<CALXCt0rLjnc+E0CvzwkoASHhPsZ7=H28gJ=nATHJKFaWdnsCkQ@mail.gmail.com>
Message-ID: <CAAYf2Lvb_De19zcMY++ivpZzRGXSU4XtsbHfAVgaN49poJuF8A@mail.gmail.com>

Depending on the throughput ie number of calls an Alix board can be a
pretty solid solution too.

On Thu, Sep 29, 2011 at 9:49 AM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> a WRT device is probably what you're looking for.
>
> Search the mailing list and the wiki for info and instructions.
>
> -giovanni
>
> On Thu, Sep 29, 2011 at 3:43 PM, ocset <ocset at the800group.com> wrote:
>> Hi
>>
>> I have successfully built a FreeSwitch system and just wanted to get
>> some input from those who have already deployed a system as to your
>> hardware of choice (just the OS hardware - will be running Linux)?
>>
>> Is there someone who makes a good slimline blackbox solution or should I
>> just put together my own machine. I will be using all external devices
>> (no internal FXO/FXS cards) so there should be no need to worry about
>> which is the best mobo etc.
>>
>> My concern is that a standard PC is not meant to be left running 24/7
>> (reliability) while a server may be too expensive and large for the task?
>>
>> All comments greatly appreciated.
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From Nabble_01394 at slickdeals.endjunk.com  Thu Sep 29 18:57:22 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Thu, 29 Sep 2011 07:57:22 -0700 (PDT)
Subject: [Freeswitch-users] Hardware - PC vs blackbox vs server
In-Reply-To: <4E8475EE.2080408@the800group.com>
References: <4E8475EE.2080408@the800group.com>
Message-ID: <1317308242076-6844184.post@n2.nabble.com>


ocset wrote:
> My concern is that a standard PC is not meant to be left running 24/7 
> (reliability) while a server may be too expensive and large for the task?
These days, a standard PC is very reliable to run on 24/7, AFAICT. The
problem using a standard PC as a server for personal use is nothing, but a
waste in electricity, especially if the standard PC is a multi-core CPUs
system and the services it provides use less than 2% CPU + system resources.
These days, an ARM driven Linux embedded system clocked @800+ MHz with at
least 64MB RAM + a USB2 port can easily be turned into a server to host FS +
among other things. In your case, you can setup any inexpensive 
http://www.google.com/search?aq=f&gcx=c&sourceid=chrome&ie=UTF-8&q=PogoPlug#q=PogoPlug&hl=en&prmd=imvnsr&source=univ&tbm=shop&tbo=u&sa=X&ei=T4SEToGyBcmftge40pEs&ved=0CGYQrQQ&bav=on.2,or.r_gc.r_pw.&fp=6262a56e37f57306&biw=1050&bih=869
PogoPlug  device to host your FS and it will easily handle 20 concurrent
calls sans transcodings, AFAICT. Adding some 
http://www.amazon.com/OBi110-Service-Bridge-Telephone-Adapter/dp/B0045RMEPI
Obi-110  devices as extensions to your FS, you can have a bridge between
your VoIP and PSTN lines. For me, my FS is for personal use with no more
than five concurrent calls sans transcodings.

For me, I have my FS + among other things hosted on a Seagate 
http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
DockStar  running on an  http://openwrt.org OpenWRT  OS and it works
flawlessly. Friends say their  http://en.wikipedia.org/wiki/Kill_A_Watt
Kill-A-Watt  device shows a Seagate 
http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
DockStar  (with an external USB card reader + an 8GB SDHC card) alone
consumes a little less than 3 Watts of electricity. On idle, FS uses no more
than 6% CPU resources on my server. This got me thinking, perhaps I should
let my main NAT/Firewall router (an old and discontinued Netgear 
http://support.netgear.com/app/products/model/a_id/2598 WGT634U ) to host my
FS. The only problem I found is FS will take about 24 minutes from the
moment I flip the switch to turn on my router before it is ready to serve
calls.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Hardware-PC-vs-blackbox-vs-server-tp6843957p6844184.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble_01394 at slickdeals.endjunk.com  Thu Sep 29 19:04:14 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Thu, 29 Sep 2011 08:04:14 -0700 (PDT)
Subject: [Freeswitch-users] google voice connection going offline
	(mod_dingaling)
In-Reply-To: <CAKrPhPPxA648-Gq=_56wQN94-c+XDGHvHnTE=pxvWgSD7s1y8A@mail.gmail.com>
References: <CAKrPhPMsaXaTPi498hS-=Xo2Y3A6tZRCcx4p=waV0Sfvw7W8ug@mail.gmail.com>
	<1317265876.9229.YahooMailNeo@web39703.mail.mud.yahoo.com>
	<CAKrPhPPxA648-Gq=_56wQN94-c+XDGHvHnTE=pxvWgSD7s1y8A@mail.gmail.com>
Message-ID: <1317308654166-6844212.post@n2.nabble.com>

The only thing I can say is to update your FS with the latest git commit.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/google-voice-connection-going-offline-mod-dingaling-tp6839355p6844212.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kris at kriskinc.com  Thu Sep 29 19:08:40 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 29 Sep 2011 11:08:40 -0400
Subject: [Freeswitch-users] Hardware - PC vs blackbox vs server
In-Reply-To: <1317308242076-6844184.post@n2.nabble.com>
References: <4E8475EE.2080408@the800group.com>
	<1317308242076-6844184.post@n2.nabble.com>
Message-ID: <CAKDfjgd-N+kDwnJYVgpt-b0nP2sREWvHHmQXN93wwHJMTMb7RQ@mail.gmail.com>

http://www.amazon.com/gp/product/B004UAL5AU/ref=ox_ya_os_product

680MHz CPU, 128MB RAM, 32MB flash for under $100.  It will start out
life at my house as a router+AP but I'm sure I'll be tempted to throw
FS on it with OpenWRT at some point...

On Thu, Sep 29, 2011 at 10:57 AM, mazilo
<Nabble_01394 at slickdeals.endjunk.com> wrote:
>
> ocset wrote:
>> My concern is that a standard PC is not meant to be left running 24/7
>> (reliability) while a server may be too expensive and large for the task?
> These days, a standard PC is very reliable to run on 24/7, AFAICT. The
> problem using a standard PC as a server for personal use is nothing, but a
> waste in electricity, especially if the standard PC is a multi-core CPUs
> system and the services it provides use less than 2% CPU + system resources.
> These days, an ARM driven Linux embedded system clocked @800+ MHz with at
> least 64MB RAM + a USB2 port can easily be turned into a server to host FS +
> among other things. In your case, you can setup any inexpensive
> http://www.google.com/search?aq=f&gcx=c&sourceid=chrome&ie=UTF-8&q=PogoPlug#q=PogoPlug&hl=en&prmd=imvnsr&source=univ&tbm=shop&tbo=u&sa=X&ei=T4SEToGyBcmftge40pEs&ved=0CGYQrQQ&bav=on.2,or.r_gc.r_pw.&fp=6262a56e37f57306&biw=1050&bih=869
> PogoPlug ?device to host your FS and it will easily handle 20 concurrent
> calls sans transcodings, AFAICT. Adding some
> http://www.amazon.com/OBi110-Service-Bridge-Telephone-Adapter/dp/B0045RMEPI
> Obi-110 ?devices as extensions to your FS, you can have a bridge between
> your VoIP and PSTN lines. For me, my FS is for personal use with no more
> than five concurrent calls sans transcodings.
>
> For me, I have my FS + among other things hosted on a Seagate
> http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
> DockStar ?running on an ?http://openwrt.org OpenWRT ?OS and it works
> flawlessly. Friends say their ?http://en.wikipedia.org/wiki/Kill_A_Watt
> Kill-A-Watt ?device shows a Seagate
> http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
> DockStar ?(with an external USB card reader + an 8GB SDHC card) alone
> consumes a little less than 3 Watts of electricity. On idle, FS uses no more
> than 6% CPU resources on my server. This got me thinking, perhaps I should
> let my main NAT/Firewall router (an old and discontinued Netgear
> http://support.netgear.com/app/products/model/a_id/2598 WGT634U ) to host my
> FS. The only problem I found is FS will take about 24 minutes from the
> moment I flip the switch to turn on my router before it is ready to serve
> calls.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Hardware-PC-vs-blackbox-vs-server-tp6843957p6844184.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
>



-- 
Kristian Kielhofner


From Nabble_01394 at slickdeals.endjunk.com  Thu Sep 29 19:11:13 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Thu, 29 Sep 2011 08:11:13 -0700 (PDT)
Subject: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was
 Re:	Is there any license G729?
In-Reply-To: <2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org>
References: <8cc991dd0906291826u6ecf131fq66167b1928e5e820@mail.gmail.com>
	<20090630014115.GA23733@jdc.jasonjgw.net>
	<2703B080-168C-4D83-89E9-0A5FCDB162BB@freeswitch.org>
Message-ID: <1317309073514-6844247.post@n2.nabble.com>

One word only: *CHINESE*!

Brian West wrote:
> 
> Now who can we con into building an open platform thats nothing more  
> than a linux box shaped like a nice looking phone?  :P


-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Hardware-phones-that-do-48Khz-Celt-Was-Re-Is-there-any-license-G729-tp3178471p6844247.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jmesquita at freeswitch.org  Thu Sep 29 19:56:03 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 29 Sep 2011 12:56:03 -0300
Subject: [Freeswitch-users] FsGUI.exe on windows 2003 server: no
	available plugin
In-Reply-To: <1317277484.60423.YahooMailNeo@web39706.mail.mud.yahoo.com>
References: <1317192465.66397.YahooMailNeo@web39701.mail.mud.yahoo.com>
	<CABH8QX3tmNTEhhQq40bvM8cHZa6TgorOKJKN9vJNhTkiaxhdCA@mail.gmail.com>
	<1317264762.4235.YahooMailNeo@web39702.mail.mud.yahoo.com>
	<CABH8QX1bUJ79zE7q4s4c9qxJY13Zke6z-tnVcQNHFkOoTep_bg@mail.gmail.com>
	<1317277484.60423.YahooMailNeo@web39706.mail.mud.yahoo.com>
Message-ID: <CABH8QX296ej17x2GuY73YEbmdLdZTB+AwHJTyv3AntSWSD0MvA@mail.gmail.com>

FSGui was originally designed to be working as a plugin based application.
Console was one of the plugins that the application was using but my
original idea was to make it extensible and improve the plugin machinery as
we go so that we can support plugin dependencies and such. Obviously this
was a bad idea as FSGui never really went forward and the only plugin really
being requested and used is the Console. That being said, I need to
restructure the application to drop the plugin idea.

2 side effects of this description are that the first window you see is a
list of plugins detected. You necessarily need to select a plugin to run the
application. This very same dialog is using when you are in the application
and you want to change the plugin being used or start a new one. So the OK
button will quit the application if no plugin is currently running. This
could be seen as a bug because the OK button shouldn't be appearing in that
instance in the first place. Maybe a Quit button at best.

The other side effect is that I am failing to make the plugin system to work
well. So users are having trouble because the plugin search path is wrong on
some systems. I can't understand why, yet and will only know for sure when I
add internal logging to the application (which I intend to get done today).

Now, as for the error you are getting when trying to connect is because you
don't have the mod_event_socket properly configured on the FreeSWITCH you
are trying to connect to. It needs to bind to something else then localhost
and/or have the ACLs properly configured. Take a look at the wiki here:
http://wiki.freeswitch.org/wiki/Mod_event_socket#Configuration

Regards,
Jo?o Mesquita



On Thu, Sep 29, 2011 at 3:24 AM, king2kin <xing2kin at yahoo.com> wrote:

> As I recall, the directory in which the application is looking for its
> plugins seems to point to the root directory (e.g. C:\FreeSWITCH)  of FsGUI
> when I started it first time after installation.
>
> Then, I forced it to point to the folder "C:\FreeSWITCH\plugins"; so now it
> always points to "C:\FreeSWITCH\plugins" whenever it's started, and no error
> dialog pops up.  However, if I click on button [ok] at the bottom, it asks
> me "Do you really want to close the FsGUI? yes or no" inside a separate
> pop-up window.
> Why doesn't FsGUI start to run when clicking on button [ok] at the bottom
> of main window?
>
> I have to click on item [console,  the application will do the same ...] in
> order to enable the button [Run].
> Once I click on button [Run], I can add or edit server entry inside Server
> Manager window.
> Question: how do I fill the blank [Password] to connect my FreeSwitch
> server?
> I got error: connection failed! socket connection error.
>
>
>  *From:* Jo?o Mesquita <jmesquita at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Thursday, September 29, 2011 11:54 AM
>
> *Subject:* Re: [Freeswitch-users] FsGUI.exe on windows 2003 server: no
> available plugin
>
> This is really strange. The way the app is designed is that we can develop
> more plugins to the console and load them on runtime. Much like what
> FreeSWITCH does with its modules (but in a muuuuch more rudimentary way).
>
> When you open the app, you should see the directory in which the
> application is looking for its plugins. Do you see the correct path?
>
> If you don't, try setting the correct path. Please let me know the results
> of that.
>
> I am adding logging to the application to help me debug stuff like that.
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Wed, Sep 28, 2011 at 11:52 PM, king2kin <xing2kin at yahoo.com> wrote:
>
>  Hi Joao,
>
> Yes, I do have the folder [plugins] inside the folder where I have the
> fsgui.exe. The directory structure where I installed FsGui is as follows:
>
> [C:\FreeSWITCH]: this is the root directory where [FsGui] was installed,
> and  those exe files are located;
> [C:\FreeSWITCH\plugins]: it contains only one dll file "consoleplugin.dll".
>
> x.k.
>
>  *From:* Jo?o Mesquita <jmesquita at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, September 28, 2011 9:19 PM
> *Subject:* Re: [Freeswitch-users] FsGUI.exe on windows 2003 server: no
> available plugin
>
>  Do you have the plugins folder inside the folder where you have the .exe?
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Wed, Sep 28, 2011 at 3:47 AM, king2kin <xing2kin at yahoo.com> wrote:
>
>  Hi,
>
> I downloaded windows version of FsGui.exe on
> http://wiki.freeswitch.org/wiki/Fsgui .
>
> After installing it, I tried to run it on windows 2003 server, it poped up
> an error dialog saying
> "no available plugins. Install conpatible plugins before running fsgui".
> Here I wonder what else I should install to run fsgui.
>
> 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
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dave at clancysystems.com  Thu Sep 29 19:58:31 2011
From: dave at clancysystems.com (Dave)
Date: Thu, 29 Sep 2011 09:58:31 -0600
Subject: [Freeswitch-users] Outbound calling with recorded message.
References: <F91D0399DF064C4193AA40A6BB059076@clancysystems.com>
	<CAKzWOxXOXM_Y7X2QcETKUpr7_yQ9mQYOBzkWAVonsH-8XYNrkA@mail.gmail.com>
Message-ID: <9AD65D9A5C774D98B07C8A46D3D9E535@clancysystems.com>

Thank you very much for the reply.

Good point about the calls not reaching the target. In thinking about it it might be better to send a text or email to them.

I do want to learn about the event socket though, so I'll study up on that. After I sent the question I noticed the freeSWITCH book has a chapter on that and there's a lot online about it.

I really appreciate the help everyone provides.

Dave



  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, September 28, 2011 7:18 PM
  Subject: Re: [Freeswitch-users] Outbound calling with recorded message.


  Once you have the calls logged into a database then it is a relatively simple matter to generate the outbound calls using the event socket. The real challenge (IMHO) is accounting for the outbound calls that don't actually reach the target, or that go to the target's voicemail, etc. 


  For the sake of simplicity, let's assume that each person you call will answer. From there you just need a simple dialplan extension that does a record app with a specific filename. (You need to match up the filename recorded with the person you called. You could use <phone_num>.wav I suppose.) From there it's a matter of launching the calls. I don't do much with Windows but there are plenty of folks here who do. If you can establish an event socket connection then you can execute a bunch of "originate" API calls to generate your outbound calls.


  Let's say your dialplan extension for recording the name is this:
  <extension name="get their name">
    <condition field="destination_number" expression="^OB_IVR_Record_Name_(\d+)$">
      <action application="answer"/>
      <action application="sleep" data="1000"/>
      <action application="playback" data="please_record_your_name.wav"/>
      <action application="playback" data="tone_stream://%(1000,0,1500)"/>
      <action application="record" data="/tmp/$1.wav 25 200"/>
    </condition>
  </extension>


  You can generate a call with this API:


  originate sofia/gw/gwname/18005551212 OB_IVR_Record_Name_8005551212


  If it works then you'll end up with /tmp/8005551212.wav and hopefully they'll have given you the info you need. 


  As for generating these calls, if you don't need something too fancy you could just use a program written in the scripting language of your choice. (Perl, Python, Ruby, and PHP are all suitable for this task.) You could also write "real" program with Visual Studio if that suits you. The key is that you will need to keep track of what happens when you make all these calls and be sure not to keep calling them over and over again. :)


  I've done this sort of thing with just Perl scripts and it works really well. 


  -MC


  On Wed, Sep 28, 2011 at 10:35 AM, Dave <dave at clancysystems.com> wrote:


    Hi all, 

    I have been using freeSWITCH for a while for inbound calls in which a person registers for a seminar via IVR.  I simply use the DID, CID number and name to identify the person and put them in the Database. That part of the application is working wonderfully.

    The issue I'm presented with now is that we need automate making calls to a few of these registrants, after each event, whos caller_id_name comes in as "Unknown" or "Wireless Caller", Play a recording to let them know we need the name, and that they can record the information right then, or call the office.

    I am limited to using a Windows 7 machine.

    What would be the best tool to use to automate these calls with the use of an IVR?

    Dave Goodwin




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



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From anthony.minessale at gmail.com  Thu Sep 29 20:07:40 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 29 Sep 2011 11:07:40 -0500
Subject: [Freeswitch-users] fifo timeout?
In-Reply-To: <CAKzWOxXBi2dfQD11=+=1whpAm4eSNvJj_Ok-EW4Nu+4K=A+5eQ@mail.gmail.com>
References: <4E821D9B.30204@communicatefreely.net>
	<CAKzWOxXBi2dfQD11=+=1whpAm4eSNvJj_Ok-EW4Nu+4K=A+5eQ@mail.gmail.com>
Message-ID: <CAKbxfG9XQaCYrj4LfpNR3rEb=gNaPLk-7Qed421RmQKSu4QpzQ@mail.gmail.com>

I added a patch just now so you can use "_continue_" as the orbit
exten value which tells it to just exit the app back to the existing
dialplan.


On Wed, Sep 28, 2011 at 7:26 PM, Michael Collins <msc at freeswitch.org> wrote:
> You probably want the fifo_orbit_exten chan var. See here:
> http://wiki.freeswitch.org/wiki/Mod_fifo#This_channel_uses_or_sets.2C_for_the_caller_leg_entering_the_FIFO
> -MC
>
> On Tue, Sep 27, 2011 at 12:01 PM, Tim St. Pierre
> <fs-list at communicatefreely.net> wrote:
>>
>> Hello,
>>
>> I'm building a very simple queue system with mod_fifo. ?For the most
>> part, things are working the way I want, but I can't figure out how to
>> limit the amount of time someone can sit in the queue. ?I'm trying to
>> implement something like max_wait would do in mod_callcenter.
>>
>> I can use the orbit_exten option, but it wants to transfer to a
>> different extension. ?I really just want the fifo application to exit so
>> the caller can progress to the next dialplan action. ?Is there a way to
>> make the caller exit the fifo if they are unanswered after a set amount
>> of time?
>>
>> I did give mod_callcenter a try, but I had a lot of problems with calls
>> getting stuck on hold, with no agents ringing, so I went to mod_fifo as
>> a simpler solution.
>>
>> Thanks for any suggestions.
>>
>> -Tim
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 29 20:42:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 29 Sep 2011 09:42:02 -0700
Subject: [Freeswitch-users] fifo timeout?
In-Reply-To: <CAKbxfG9XQaCYrj4LfpNR3rEb=gNaPLk-7Qed421RmQKSu4QpzQ@mail.gmail.com>
References: <4E821D9B.30204@communicatefreely.net>
	<CAKzWOxXBi2dfQD11=+=1whpAm4eSNvJj_Ok-EW4Nu+4K=A+5eQ@mail.gmail.com>
	<CAKbxfG9XQaCYrj4LfpNR3rEb=gNaPLk-7Qed421RmQKSu4QpzQ@mail.gmail.com>
Message-ID: <CAKzWOxV1eyr5mGLR=mVZCaDRK+dNE1qJfzTyrJ+SFp1Yp6S0OQ@mail.gmail.com>

I added this to the wiki; i will expand upon it shortly.
-MC

On Thu, Sep 29, 2011 at 9:07 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I added a patch just now so you can use "_continue_" as the orbit
> exten value which tells it to just exit the app back to the existing
> dialplan.
>
>
> On Wed, Sep 28, 2011 at 7:26 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > You probably want the fifo_orbit_exten chan var. See here:
> >
> http://wiki.freeswitch.org/wiki/Mod_fifo#This_channel_uses_or_sets.2C_for_the_caller_leg_entering_the_FIFO
> > -MC
> >
> > On Tue, Sep 27, 2011 at 12:01 PM, Tim St. Pierre
> > <fs-list at communicatefreely.net> wrote:
> >>
> >> Hello,
> >>
> >> I'm building a very simple queue system with mod_fifo.  For the most
> >> part, things are working the way I want, but I can't figure out how to
> >> limit the amount of time someone can sit in the queue.  I'm trying to
> >> implement something like max_wait would do in mod_callcenter.
> >>
> >> I can use the orbit_exten option, but it wants to transfer to a
> >> different extension.  I really just want the fifo application to exit so
> >> the caller can progress to the next dialplan action.  Is there a way to
> >> make the caller exit the fifo if they are unanswered after a set amount
> >> of time?
> >>
> >> I did give mod_callcenter a try, but I had a lot of problems with calls
> >> getting stuck on hold, with no agents ringing, so I went to mod_fifo as
> >> a simpler solution.
> >>
> >> Thanks for any suggestions.
> >>
> >> -Tim
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From henrikaagaardsorensen at gmail.com  Fri Sep 30 02:59:25 2011
From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=)
Date: Fri, 30 Sep 2011 00:59:25 +0200
Subject: [Freeswitch-users] FreeSwitch fails with Kamailio dispatcher as
	front.
Message-ID: <CAGH8SeaBU=G09wodOjGGSUaP7khX91P+cjDBGDLH3DU3Z4VjAw@mail.gmail.com>

I have a setup with 2 FreeSwitch servers and a single Kamailio sevrer in
front with the dispatcher module.

Everything works when directly connected to FreeSwitch. When I connect to
Kamailio I'm only able to call one way between 2 extensions.

The log from FreeSwitch says:
2011-09-29 15:51:53.625761 [ERR] sofia_reg.c:2136 Cannot locate any
authentication credentials to complete an authentication request for realm
'"10.0.1.7"'

I guess it because, somehow, that one of the extension has a local IP.

Both extensions can register and call into FreeSwitch and hear hold music
etc. It's just one of the extension who can't call the other extension.

I don't know if the error should be fixed on FreeSwitch or on Kamailio. And
how...

Can anyone help me out?
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From djbinter at gmail.com  Fri Sep 30 03:59:07 2011
From: djbinter at gmail.com (DJB International)
Date: Thu, 29 Sep 2011 16:59:07 -0700
Subject: [Freeswitch-users] FreeSwitch fails with Kamailio dispatcher as
	front.
In-Reply-To: <CAGH8SeaBU=G09wodOjGGSUaP7khX91P+cjDBGDLH3DU3Z4VjAw@mail.gmail.com>
References: <CAGH8SeaBU=G09wodOjGGSUaP7khX91P+cjDBGDLH3DU3Z4VjAw@mail.gmail.com>
Message-ID: <CAAB3jWs+N1wh_BiuNqWxQPSGA+2eDQKGRDzd1aL18=D_imPfwQ@mail.gmail.com>

First thing I would check ACL probably to allow the ip address in
acl.conf.xml, but cannot tell you much since there's not much info here in
your email.  You can also read from here, but I am not sure whether it's how
you set up the network:
http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc

-djbinter

2011/9/29 Henrik Aagaard S?rensen <henrikaagaardsorensen at gmail.com>

> I have a setup with 2 FreeSwitch servers and a single Kamailio sevrer in
> front with the dispatcher module.
>
> Everything works when directly connected to FreeSwitch. When I connect to
> Kamailio I'm only able to call one way between 2 extensions.
>
> The log from FreeSwitch says:
> 2011-09-29 15:51:53.625761 [ERR] sofia_reg.c:2136 Cannot locate any
> authentication credentials to complete an authentication request for realm
> '"10.0.1.7"'
>
> I guess it because, somehow, that one of the extension has a local IP.
>
> Both extensions can register and call into FreeSwitch and hear hold music
> etc. It's just one of the extension who can't call the other extension.
>
> I don't know if the error should be fixed on FreeSwitch or on Kamailio. And
> how...
>
> Can anyone help me out?
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cvogel at lyonl.com  Fri Sep 30 04:47:59 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Fri, 30 Sep 2011 00:47:59 +0000
Subject: [Freeswitch-users] User management
Message-ID: <ACCA0268-D0F0-42A0-A755-43745D9E1570@lyonl.com>

Hello,

I am wondering if anyone has created any scripts to manage user authentication using a database?

Chad


From xing2kin at yahoo.com  Fri Sep 30 05:16:37 2011
From: xing2kin at yahoo.com (king2kin)
Date: Thu, 29 Sep 2011 18:16:37 -0700 (PDT)
Subject: [Freeswitch-users] FsGUI.exe on windows 2003 server: no
	available plugin
In-Reply-To: <CABH8QX296ej17x2GuY73YEbmdLdZTB+AwHJTyv3AntSWSD0MvA@mail.gmail.com>
References: <1317192465.66397.YahooMailNeo@web39701.mail.mud.yahoo.com>
	<CABH8QX3tmNTEhhQq40bvM8cHZa6TgorOKJKN9vJNhTkiaxhdCA@mail.gmail.com>
	<1317264762.4235.YahooMailNeo@web39702.mail.mud.yahoo.com>
	<CABH8QX1bUJ79zE7q4s4c9qxJY13Zke6z-tnVcQNHFkOoTep_bg@mail.gmail.com>
	<1317277484.60423.YahooMailNeo@web39706.mail.mud.yahoo.com>
	<CABH8QX296ej17x2GuY73YEbmdLdZTB+AwHJTyv3AntSWSD0MvA@mail.gmail.com>
Message-ID: <1317345397.16828.YahooMailNeo@web39707.mail.mud.yahoo.com>

Thank you! yes, now it works after I set correct passord and host to server.
?
By the way,?is your FsGui application open source project? if so, where?are your source codes?located? I am intersted to take a look at it.

From: Jo?o Mesquita <jmesquita at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, September 29, 2011 11:56 PM
Subject: Re: [Freeswitch-users] FsGUI.exe on windows 2003 server: no available plugin




FSGui was originally designed to be working as a plugin based application. Console was one of the plugins that the application was using but my original idea was to make it extensible and improve the plugin machinery as we go so that we can support plugin dependencies and such. Obviously this was a bad idea as FSGui never really went forward and the only plugin really being requested and used is the Console.?That being said, I need to restructure the application to drop the plugin idea.

2 side effects of this description are that the first window you see is a list of plugins detected. You necessarily need to select a plugin to run the application. This very same dialog is using when you are in the application and you want to change the plugin being used or start a new one. So the OK button will quit the application if no plugin is currently running. This could be seen as a bug because the OK button shouldn't be appearing in that instance in the first place. Maybe a Quit button at best.

The other side effect is that I am failing to make the plugin system to work well. So users are having trouble because the plugin search path is wrong on some systems. I can't understand why, yet and will only know for sure when I add internal logging to the application (which I intend to get done today).

Now, as for the error you are getting when trying to connect is because you don't have the mod_event_socket properly configured on the FreeSWITCH you are trying to connect to. It needs to bind to something else then localhost and/or have the ACLs properly configured. Take a look at the wiki here:?http://wiki.freeswitch.org/wiki/Mod_event_socket#Configuration

Regards,Jo?o Mesquita




On Thu, Sep 29, 2011 at 3:24 AM, king2kin <xing2kin at yahoo.com> wrote:

As I recall, the directory in which the application is looking for its plugins seems to point to the root directory (e.g. C:\FreeSWITCH) ?of FsGUI when I started it first time after installation. 
>?
>Then, I forced it to point to the folder "C:\FreeSWITCH\plugins"; so now it always points to "C:\FreeSWITCH\plugins" whenever it's started, and no error dialog pops up.? However, if I click on button [ok] at the bottom, it asks me "Do you really want to close the FsGUI? yes or no" inside a separate pop-up window. 
>Why doesn't FsGUI start to run when clicking on button [ok] at the bottom of main window?
>?
>I have to click on item [console, ?the application will do the same ...] in order to enable the button [Run].
>Once I click on button [Run], I can add or edit server entry inside Server Manager window. 
>Question:?how do I fill the blank [Password] to connect my FreeSwitch server? 
>I got error: connection failed! socket connection error.
>?
>
>
>From: Jo?o Mesquita <jmesquita at freeswitch.org>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>Sent: Thursday, September 29, 2011 11:54 AM 
>
>Subject: Re: [Freeswitch-users] FsGUI.exe on windows 2003 server: no available plugin
>
>
>
>This is really strange. The way the app is designed is that we can develop more plugins to the console and load them on runtime. Much like what FreeSWITCH does with its modules (but in a muuuuch more rudimentary way). 
>
>
>When you open the app, you should see the directory in which the application is looking for its plugins. Do you see the correct path?
>
>
>If you don't, try setting the correct path. Please let me know the results of that.
>
>
>I am adding logging to the application to help me debug stuff like that.
>
>
>Regards,
>Jo?o Mesquita
>
>
>
>
>On Wed, Sep 28, 2011 at 11:52 PM, king2kin <xing2kin at yahoo.com> wrote:
>
>Hi Joao,
>>?
>>Yes, I do have the folder [plugins] inside the folder where I have the fsgui.exe. The directory structure where I installed FsGui is as follows:
>>?
>>[C:\FreeSWITCH]: this is the root directory where [FsGui] was installed, and ?those exe files are located;
>>[C:\FreeSWITCH\plugins]: it contains only one dll file "consoleplugin.dll".
>>?
>>x.k. 
>>
>>
>>From: Jo?o Mesquita <jmesquita at freeswitch.org>
>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>Sent: Wednesday, September 28, 2011 9:19 PM
>>Subject: Re: [Freeswitch-users] FsGUI.exe on windows 2003 server: no available plugin
>>
>>
>>
>>Do you have the plugins folder inside the folder where you have the .exe?
>>
>>
>>Regards,Jo?o Mesquita
>>
>>
>>
>>
>>On Wed, Sep 28, 2011 at 3:47 AM, king2kin <xing2kin at yahoo.com> wrote:
>>
>>Hi, 
>>>
>>>I downloaded windows version of FsGui.exe on http://wiki.freeswitch.org/wiki/Fsgui?.
>>>
>>>After installing it, I tried to run it on windows 2003 server, it poped up an error dialog saying
>>>"no available plugins. Install conpatible plugins before running fsgui". Here I wonder what else I should install to run fsgui.
>>>
>>>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
>>>
>>>
>>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From Nabble_01394 at slickdeals.endjunk.com  Fri Sep 30 06:37:08 2011
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Thu, 29 Sep 2011 19:37:08 -0700 (PDT)
Subject: [Freeswitch-users] Hardware - PC vs blackbox vs server
In-Reply-To: <CAKDfjgd-N+kDwnJYVgpt-b0nP2sREWvHHmQXN93wwHJMTMb7RQ@mail.gmail.com>
References: <4E8475EE.2080408@the800group.com>
	<1317308242076-6844184.post@n2.nabble.com>
	<CAKDfjgd-N+kDwnJYVgpt-b0nP2sREWvHHmQXN93wwHJMTMb7RQ@mail.gmail.com>
Message-ID: <1317350228417-6846160.post@n2.nabble.com>

That's is not bad especially if OP is also looking for a NAT/Firewall router.
However, I would rather spend less for a more powerful device, i.e. 
http://www.google.com/products/catalog?q=pogoplug+pro&um=1&ie=UTF-8&tbm=shop&cid=8167861145955058396&sa=X&ei=rSeFTvSeOMqgtwfX0eU8&ved=0CHAQ8wIwAg
PogoPlug Pro , etc., as my personal server to host FS + among other things.
Honestly, I don't know if debian ARM has already supported this device. That
said, the device is built on a multi-core ARM6 (PLX NAS7820) CPU with 128MB
RAM clocked @1.2GHz. This   http://archlinuxarm.org/forum ArchLinux forum 
has a lot of information on how to hack this device.


Kristian Kielhofner-3 wrote:
> 
> http://www.amazon.com/gp/product/B004UAL5AU/ref=ox_ya_os_product
> 
> 680MHz CPU, 128MB RAM, 32MB flash for under $100.  It will start out
> life at my house as a router+AP but I'm sure I'll be tempted to throw
> FS on it with OpenWRT at some point...


-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Hardware-PC-vs-blackbox-vs-server-tp6843957p6846160.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jmesquita at freeswitch.org  Fri Sep 30 08:14:50 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 30 Sep 2011 01:14:50 -0300
Subject: [Freeswitch-users] FsGUI.exe on windows 2003 server: no
	available plugin
In-Reply-To: <1317345397.16828.YahooMailNeo@web39707.mail.mud.yahoo.com>
References: <1317192465.66397.YahooMailNeo@web39701.mail.mud.yahoo.com>
	<CABH8QX3tmNTEhhQq40bvM8cHZa6TgorOKJKN9vJNhTkiaxhdCA@mail.gmail.com>
	<1317264762.4235.YahooMailNeo@web39702.mail.mud.yahoo.com>
	<CABH8QX1bUJ79zE7q4s4c9qxJY13Zke6z-tnVcQNHFkOoTep_bg@mail.gmail.com>
	<1317277484.60423.YahooMailNeo@web39706.mail.mud.yahoo.com>
	<CABH8QX296ej17x2GuY73YEbmdLdZTB+AwHJTyv3AntSWSD0MvA@mail.gmail.com>
	<1317345397.16828.YahooMailNeo@web39707.mail.mud.yahoo.com>
Message-ID: <CABH8QX2EojMOufmWK2OtXXxdezNWZi+GRHYkMzJDkjRz0B-wPw@mail.gmail.com>

It is MPL just like the rest of FreeSWITCH. The source code for it can be
found on my contrib on freeswitch's git contrib repo.

I am glad it worked for you.

There are improvements coming on the next few days.

Regards,
Jo?o Mesquita



On Thu, Sep 29, 2011 at 10:16 PM, king2kin <xing2kin at yahoo.com> wrote:

> Thank you! yes, now it works after I set correct passord and host to
> server.
>
> By the way, is your FsGui application open source project? if so, where are
> your source codes located? I am intersted to take a look at it.
>
>  *From:* Jo?o Mesquita <jmesquita at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Thursday, September 29, 2011 11:56 PM
>
> *Subject:* Re: [Freeswitch-users] FsGUI.exe on windows 2003 server: no
> available plugin
>
>
> FSGui was originally designed to be working as a plugin based application.
> Console was one of the plugins that the application was using but my
> original idea was to make it extensible and improve the plugin machinery as
> we go so that we can support plugin dependencies and such. Obviously this
> was a bad idea as FSGui never really went forward and the only plugin really
> being requested and used is the Console. That being said, I need to
> restructure the application to drop the plugin idea.
>
> 2 side effects of this description are that the first window you see is a
> list of plugins detected. You necessarily need to select a plugin to run the
> application. This very same dialog is using when you are in the application
> and you want to change the plugin being used or start a new one. So the OK
> button will quit the application if no plugin is currently running. This
> could be seen as a bug because the OK button shouldn't be appearing in that
> instance in the first place. Maybe a Quit button at best.
>
> The other side effect is that I am failing to make the plugin system to
> work well. So users are having trouble because the plugin search path is
> wrong on some systems. I can't understand why, yet and will only know for
> sure when I add internal logging to the application (which I intend to get
> done today).
>
> Now, as for the error you are getting when trying to connect is because you
> don't have the mod_event_socket properly configured on the FreeSWITCH you
> are trying to connect to. It needs to bind to something else then localhost
> and/or have the ACLs properly configured. Take a look at the wiki here:
> http://wiki.freeswitch.org/wiki/Mod_event_socket#Configuration
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Thu, Sep 29, 2011 at 3:24 AM, king2kin <xing2kin at yahoo.com> wrote:
>
>  As I recall, the directory in which the application is looking for its
> plugins seems to point to the root directory (e.g. C:\FreeSWITCH)  of FsGUI
> when I started it first time after installation.
>
> Then, I forced it to point to the folder "C:\FreeSWITCH\plugins"; so now it
> always points to "C:\FreeSWITCH\plugins" whenever it's started, and no error
> dialog pops up.  However, if I click on button [ok] at the bottom, it asks
> me "Do you really want to close the FsGUI? yes or no" inside a separate
> pop-up window.
> Why doesn't FsGUI start to run when clicking on button [ok] at the bottom
> of main window?
>
> I have to click on item [console,  the application will do the same ...] in
> order to enable the button [Run].
> Once I click on button [Run], I can add or edit server entry inside Server
> Manager window.
> Question: how do I fill the blank [Password] to connect my FreeSwitch
> server?
> I got error: connection failed! socket connection error.
>
>
>   *From:* Jo?o Mesquita <jmesquita at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Thursday, September 29, 2011 11:54 AM
>
> *Subject:* Re: [Freeswitch-users] FsGUI.exe on windows 2003 server: no
> available plugin
>
> This is really strange. The way the app is designed is that we can develop
> more plugins to the console and load them on runtime. Much like what
> FreeSWITCH does with its modules (but in a muuuuch more rudimentary way).
>
> When you open the app, you should see the directory in which the
> application is looking for its plugins. Do you see the correct path?
>
> If you don't, try setting the correct path. Please let me know the results
> of that.
>
> I am adding logging to the application to help me debug stuff like that.
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Wed, Sep 28, 2011 at 11:52 PM, king2kin <xing2kin at yahoo.com> wrote:
>
>  Hi Joao,
>
> Yes, I do have the folder [plugins] inside the folder where I have the
> fsgui.exe. The directory structure where I installed FsGui is as follows:
>
> [C:\FreeSWITCH]: this is the root directory where [FsGui] was installed,
> and  those exe files are located;
> [C:\FreeSWITCH\plugins]: it contains only one dll file "consoleplugin.dll".
>
> x.k.
>
>  *From:* Jo?o Mesquita <jmesquita at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, September 28, 2011 9:19 PM
> *Subject:* Re: [Freeswitch-users] FsGUI.exe on windows 2003 server: no
> available plugin
>
>  Do you have the plugins folder inside the folder where you have the .exe?
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Wed, Sep 28, 2011 at 3:47 AM, king2kin <xing2kin at yahoo.com> wrote:
>
>  Hi,
>
> I downloaded windows version of FsGui.exe on
> http://wiki.freeswitch.org/wiki/Fsgui .
>
> After installing it, I tried to run it on windows 2003 server, it poped up
> an error dialog saying
> "no available plugins. Install conpatible plugins before running fsgui".
> Here I wonder what else I should install to run fsgui.
>
> 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
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jack at livecall.com  Fri Sep 30 08:22:35 2011
From: jack at livecall.com (Jack)
Date: Thu, 29 Sep 2011 21:22:35 -0700
Subject: [Freeswitch-users] User management
In-Reply-To: <ACCA0268-D0F0-42A0-A755-43745D9E1570@lyonl.com>
References: <ACCA0268-D0F0-42A0-A755-43745D9E1570@lyonl.com>
Message-ID: <4E85440B.4020203@livecall.com>

Hi Chad,
  mod_curl works great for that....
Jack

On 9/29/2011 5:47 PM, Chad Vogel wrote:
> Hello,
>
> I am wondering if anyone has created any scripts to manage user authentication using a database?
>
> Chad
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 lakersman2006 at yahoo.com  Fri Sep 30 08:26:43 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Thu, 29 Sep 2011 21:26:43 -0700 (PDT)
Subject: [Freeswitch-users] Need help
Message-ID: <1317356803.21942.YahooMailNeo@web161017.mail.bf1.yahoo.com>

Hi,

I need some help with an idea I am trying to implement with Freeswitch. I am trying to work with a partner company which will forward a sip call to my freeswitch server and my freeswitch server is suppose to playback annoucements to the caller and then once they are finished listening they should be forwarded back to the partner company so that the call can continue without my freeswitch being in the call path. So does any one have any suggestions/ideas on the best way to implement this?
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From moises.silva at gmail.com  Fri Sep 30 09:15:29 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 30 Sep 2011 01:15:29 -0400
Subject: [Freeswitch-users] Problem with Sangoma D100 Transconding card
In-Reply-To: <1317289483.4860.20.camel@marces.madrid.commsmundi.com>
References: <1317116551.12197.21.camel@marces.madrid.commsmundi.com>
	<CAA4nhyBM0cRr472xe+-XZ_2fsZieP1cWWGPGgnRbPc_qHoscsA@mail.gmail.com>
	<1317289483.4860.20.camel@marces.madrid.commsmundi.com>
Message-ID: <CAA4nhyCdR9mbXS=_q3UN2Hnrm+M=eySBwN3+31-4epTbjHYbmw@mail.gmail.com>

On Thu, Sep 29, 2011 at 5:44 AM, Antonio <asilva at wirelessmundi.com> wrote:

> **
> Hi,
>
> The log when this error occurs can be found in
> http://pastebin.freeswitch.org/17431 .
>
>
Can you reproduce this problem easily?

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
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From govoiper at gmail.com  Fri Sep 30 09:15:41 2011
From: govoiper at gmail.com (Sam Govind)
Date: Fri, 30 Sep 2011 10:15:41 +0500
Subject: [Freeswitch-users] Need help
In-Reply-To: <1317356803.21942.YahooMailNeo@web161017.mail.bf1.yahoo.com>
References: <1317356803.21942.YahooMailNeo@web161017.mail.bf1.yahoo.com>
Message-ID: <CAJUJwtjcALgprz=y-eSuf3HchyyfTSbg+=r_MRJN1aRNa40TTA@mail.gmail.com>

I could only think of these two. I was almost trying the same thing on my
setup some time ago.

1- Use redirect application at the end of your prompts back to the same
server. So A-party and C-party being the same server should bridge the same
call back to one.
2- Use hangup application with some custom header as flag and tell your
partner company to proceed on the basis of that flag.

hope anybody else comes up with some brilliantly simple idea.

Regards,
-Sammy

On Fri, Sep 30, 2011 at 9:26 AM, Sam <lakersman2006 at yahoo.com> wrote:

> Hi,
>
> I need some help with an idea I am trying to implement with Freeswitch. I
> am trying to work with a partner company which will forward a sip call to my
> freeswitch server and my freeswitch server is suppose to playback
> annoucements to the caller and then once they are finished listening they
> should be forwarded back to the partner company so that the call can
> continue without my freeswitch being in the call path. So does any one have
> any suggestions/ideas on the best way to implement this?
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From cvogel at lyonl.com  Fri Sep 30 09:31:31 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Fri, 30 Sep 2011 05:31:31 +0000
Subject: [Freeswitch-users] User management
In-Reply-To: <4E85440B.4020203@livecall.com>
References: <ACCA0268-D0F0-42A0-A755-43745D9E1570@lyonl.com>
	<4E85440B.4020203@livecall.com>
Message-ID: <0EDCF8FF-EAD8-4F3B-B974-0A19FD953379@lyonl.com>

I'm not looking to use curl, I would like to use some kind of scripting so i can apply custom rules to track and reject authentication based on rate and origin of authentication from source ip address

On Sep 29, 2011, at 11:22 PM, Jack wrote:

> Hi Chad,
>  mod_curl works great for that....
> Jack
> 
> On 9/29/2011 5:47 PM, Chad Vogel wrote:
>> Hello,
>> 
>> I am wondering if anyone has created any scripts to manage user authentication using a database?
>> 
>> Chad
>> 
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 asilva at wirelessmundi.com  Fri Sep 30 12:24:29 2011
From: asilva at wirelessmundi.com (Antonio)
Date: Fri, 30 Sep 2011 10:24:29 +0200
Subject: [Freeswitch-users] Problem with Sangoma D100 Transconding card
In-Reply-To: <CAA4nhyCdR9mbXS=_q3UN2Hnrm+M=eySBwN3+31-4epTbjHYbmw@mail.gmail.com>
References: <1317116551.12197.21.camel@marces.madrid.commsmundi.com>
	<CAA4nhyBM0cRr472xe+-XZ_2fsZieP1cWWGPGgnRbPc_qHoscsA@mail.gmail.com>
	<1317289483.4860.20.camel@marces.madrid.commsmundi.com>
	<CAA4nhyCdR9mbXS=_q3UN2Hnrm+M=eySBwN3+31-4epTbjHYbmw@mail.gmail.com>
Message-ID: <1317371069.8959.15.camel@marces.madrid.commsmundi.com>

Hi Moises,

the card is in a produce system and i can execute as many tests as i
wish....

I'm suspecting that is when i freeswitch stops, and then start it. Once
you place a call with g729 it doesn't work.

Next monday i will have the opportunity to do more extensive tests in
this server.

I post here the configuration and results.



On Fri, 2011-09-30 at 01:15 -0400, Moises Silva wrote: 

> On Thu, Sep 29, 2011 at 5:44 AM, Antonio <asilva at wirelessmundi.com>
> wrote:
> 
>         Hi,
>         
>         The log when this error occurs can be found in
>         http://pastebin.freeswitch.org/17431 . 
>         
>         
> 
> 
> 
> Can you reproduce this problem easily?
> 
> 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
> 
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From henrikaagaardsorensen at gmail.com  Fri Sep 30 17:52:13 2011
From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=)
Date: Fri, 30 Sep 2011 15:52:13 +0200
Subject: [Freeswitch-users] FreeSwitch fails with Kamailio dispatcher as
	front.
In-Reply-To: <CAAB3jWs+N1wh_BiuNqWxQPSGA+2eDQKGRDzd1aL18=D_imPfwQ@mail.gmail.com>
References: <CAGH8SeaBU=G09wodOjGGSUaP7khX91P+cjDBGDLH3DU3Z4VjAw@mail.gmail.com>
	<CAAB3jWs+N1wh_BiuNqWxQPSGA+2eDQKGRDzd1aL18=D_imPfwQ@mail.gmail.com>
Message-ID: <CAGH8SebUDT0Ly96U2TK9YDvask7EA8Hp5q4S3MO_jfLGENCQrg@mail.gmail.com>

I'm running FreeSwitch with blue.box (from www.2600hz.org).

The acl.conf.xml looks like this:

<?xml version="1.0"?>
<include>
  <configuration name="acl.conf">
    <network-lists>
      <list name="net_list_1" default="deny"/>
      <list name="net_list_2" default="deny"/>
      <list name="net_list_3" default="deny"/>
      <list name="net_list_4" default="deny"/>
      <list name="net_list_5" default="deny"/>
      <list name="net_list_6" default="allow">
        <node bluebox="net_list_item_1" type="allow" cidr="0.0.0.0/0"/>
      </list>
    </network-lists>
  </configuration>
</include>

On Fri, Sep 30, 2011 at 1:59 AM, DJB International <djbinter at gmail.com>wrote:

> First thing I would check ACL probably to allow the ip address in
> acl.conf.xml, but cannot tell you much since there's not much info here in
> your email.  You can also read from here, but I am not sure whether it's how
> you set up the network:
> http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc
>
> -djbinter
>
> 2011/9/29 Henrik Aagaard S?rensen <henrikaagaardsorensen at gmail.com>
>
>> I have a setup with 2 FreeSwitch servers and a single Kamailio sevrer in
>> front with the dispatcher module.
>>
>> Everything works when directly connected to FreeSwitch. When I connect to
>> Kamailio I'm only able to call one way between 2 extensions.
>>
>> The log from FreeSwitch says:
>> 2011-09-29 15:51:53.625761 [ERR] sofia_reg.c:2136 Cannot locate any
>> authentication credentials to complete an authentication request for realm
>> '"10.0.1.7"'
>>
>> I guess it because, somehow, that one of the extension has a local IP.
>>
>> Both extensions can register and call into FreeSwitch and hear hold music
>> etc. It's just one of the extension who can't call the other extension.
>>
>> I don't know if the error should be fixed on FreeSwitch or on Kamailio.
>> And how...
>>
>> Can anyone help me out?
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yungwei at resolvity.com  Fri Sep 30 18:21:21 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Fri, 30 Sep 2011 10:21:21 -0400
Subject: [Freeswitch-users] mod_callcenter doesn't send calls from a queue
	to an agent
Message-ID: <33095823FD21DF429B481B5163264B79515BF39323@VMBX102.ihostexchange.net>

Hi,

I am experimenting with mod_callcenter. And I notice that mod_callcenter doesn't send calls from a queue to an agent.
In my test, agent 1005 is registered with FS using phoner-lite.

When freeswitch is restarted, "callcenter_config queue list" returns the following:
name|strategy|moh_sound|time_base_score|tier_rules_apply|tier_rule_wait_second|tier_rule_wait_multiply_level|tier_rule_no_agent_no_wait|discard_abandoned_after|abandoned_resume_allowed|max_wait_time|max_wait_time_with_no_agent|max_wait_time_with_no_agent_time_reached|record_template
support|longest-idle-agent|local_stream://moh|system|false|300|true|false|60|false|0|0|5|
+OK

and "callcenter_config agent get status 1005" returns the following:
Available
2011-09-30 09:09:14.143579 [DEBUG] mod_callcenter.c:886 Get Info Agent 1005 status = Available

Now when I dial cc_queue from another sip client, I can hear MOH fine, but this call is not passed to agent 1005.
The following shows up in the log.
2011-09-30 09:09:23.442165 [DEBUG] mod_callcenter.c:1043 Updated Agent 1005 set state = Receiving
2011-09-30 09:09:23.442165 [DEBUG] mod_callcenter.c:1043 Updated Agent 1005 set status = Logged Out
2011-09-30 09:09:23.442165 [DEBUG] mod_callcenter.c:1043 Updated Agent 1005 set uuid =
2011-09-30 09:09:23.442165 [DEBUG] mod_callcenter.c:1654 Agent 1005 Origination Canceled : NONE
2011-09-30 09:09:23.442165 [DEBUG] mod_callcenter.c:1043 Updated Agent 1005 set state = Waiting
and "callcenter_config agent get status 1005" now returns the following:
Logged Out
2011-09-30 09:12:11.092643 [DEBUG] mod_callcenter.c:886 Get Info Agent 1005 status = Logged Out

What am I missing here? Thanks.

Here's the content of callcenter.conf.xml, in which I define a queue, an agent, and assign the agent to the queue.
<configuration name="callcenter.conf" description="CallCenter">
  <settings></settings>

  <queues>
    <queue name="support">
      <param name="strategy" value="longest-idle-agent"/>
      <param name="moh-sound" value="$${hold_music}"/>
      <param name="time-base-score" value="system"/>
      <param name="max-wait-time" value="0"/>
      <param name="max-wait-time-with-no-agent" value="0"/>
      <param name="max-wait-time-with-no-agent-time-reached" value="5"/>
      <param name="tier-rules-apply" value="false"/>
      <param name="tier-rule-wait-second" value="300"/>
      <param name="tier-rule-wait-multiply-level" value="true"/>
      <param name="tier-rule-no-agent-no-wait" value="false"/>
      <param name="discard-abandoned-after" value="60"/>
      <param name="abandoned-resume-allowed" value="false"/>
    </queue>
  </queues>
  <agents>
    <agent name="1005" type="uuid-standby" contact="[call_timeout=10]user/1005 at 192.168.216.18" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />
  </agents>
  <tiers>
    <tier agent="1005" queue="support" level="1" position="1"/>
  </tiers>
</configuration>

Here's how I send a call to the queue. This works fine.
  <extension name="sending-calls-to-queue">
      <condition field="destination_number" expression="^cc_queue$">
        <action application="answer" />
        <action application="callcenter" data="support"/>
      </condition>
  </extension>


From lakersman2006 at yahoo.com  Fri Sep 30 20:39:29 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Fri, 30 Sep 2011 09:39:29 -0700 (PDT)
Subject: [Freeswitch-users] Need help
In-Reply-To: <CAJUJwtjcALgprz=y-eSuf3HchyyfTSbg+=r_MRJN1aRNa40TTA@mail.gmail.com>
References: <1317356803.21942.YahooMailNeo@web161017.mail.bf1.yahoo.com>
	<CAJUJwtjcALgprz=y-eSuf3HchyyfTSbg+=r_MRJN1aRNa40TTA@mail.gmail.com>
Message-ID: <1317400769.64284.YahooMailNeo@web161012.mail.bf1.yahoo.com>

Thanks for the two ideas, I have a question will method #1 work if A-party and C-party are on seperate servers?

Also for method #2, is it possible to use the hangup application to send a custom sip header? How exactly would that work?



________________________________
From: Sam Govind <govoiper at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, September 29, 2011 10:15 PM
Subject: Re: [Freeswitch-users] Need help


I could only think of these two. I was almost trying the same thing on my setup some time ago.


1- Use redirect application at the end of your prompts back to the same server. So A-party and C-party being the same server should bridge the same call back to one.
2- Use hangup application with some custom header as flag and tell your partner company to proceed on the basis of that flag.

hope anybody else comes up with some?brilliantly?simple idea.

Regards,
-Sammy


On Fri, Sep 30, 2011 at 9:26 AM, Sam <lakersman2006 at yahoo.com> wrote:

Hi,
>
>
>I need some help with an idea I am trying to implement with Freeswitch. I am trying to work with a partner company which will forward a sip call to my freeswitch server and my freeswitch server is suppose to playback annoucements to the caller and then once they are finished listening they should be forwarded back to the partner company so that the call can continue without my freeswitch being in the call path. So does any one have any suggestions/ideas on the best way to implement this?
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From curriegrad2004 at gmail.com  Fri Sep 30 22:24:39 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 30 Sep 2011 11:24:39 -0700
Subject: [Freeswitch-users] Need help
In-Reply-To: <1317400769.64284.YahooMailNeo@web161012.mail.bf1.yahoo.com>
References: <1317356803.21942.YahooMailNeo@web161017.mail.bf1.yahoo.com>
	<CAJUJwtjcALgprz=y-eSuf3HchyyfTSbg+=r_MRJN1aRNa40TTA@mail.gmail.com>
	<1317400769.64284.YahooMailNeo@web161012.mail.bf1.yahoo.com>
Message-ID: <CAMWiMy8+yqMt5Rwv44Mq1gScHf3-L+kP3AB5FW2oELRnJkuHsg@mail.gmail.com>

Most of the stuff you're trying to implement are on the FreeSWITCH Wiki. If
you are looking for professional consulting there's
consulting at FreeSWITCH.org
On 2011-09-30 9:40 AM, "Sam" <lakersman2006 at yahoo.com> wrote:
>
> Thanks for the two ideas, I have a question will method #1 work if A-party
and C-party are on seperate servers?
>
> Also for method #2, is it possible to use the hangup application to send a
custom sip header? How exactly would that work?
>
> ________________________________
> From: Sam Govind <govoiper at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, September 29, 2011 10:15 PM
> Subject: Re: [Freeswitch-users] Need help
>
> I could only think of these two. I was almost trying the same thing on my
setup some time ago.
>
> 1- Use redirect application at the end of your prompts back to the same
server. So A-party and C-party being the same server should bridge the same
call back to one.
> 2- Use hangup application with some custom header as flag and tell your
partner company to proceed on the basis of that flag.
>
> hope anybody else comes up with some brilliantly simple idea.
>
> Regards,
> -Sammy
>
> On Fri, Sep 30, 2011 at 9:26 AM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>> Hi,
>>
>> I need some help with an idea I am trying to implement with Freeswitch. I
am trying to work with a partner company which will forward a sip call to my
freeswitch server and my freeswitch server is suppose to playback
annoucements to the caller and then once they are finished listening they
should be forwarded back to the partner company so that the call can
continue without my freeswitch being in the call path. So does any one have
any suggestions/ideas on the best way to implement this?
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anton.vazir at gmail.com  Fri Sep 30 22:53:52 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Fri, 30 Sep 2011 23:53:52 +0500
Subject: [Freeswitch-users] How to distinguish between Authenticated sip
 and not authenticated users
In-Reply-To: <CAGX+XzvCg0GR8Ph5-Lykhe3Fs_EWqfu7K7M1s3S=9AO-7Ah_bg@mail.gmail.com>
References: <CAGX+XzsxHocAhqHf_D4Sd-Jj_OCqLq5z7NKz6LZQV0_gtmhyRg@mail.gmail.com>
	<CACak_yzV0oG5MzAKB2NXud8ikuZa6AvxTwYxiqPmMKaTs_qU3g@mail.gmail.com>
	<CAKbxfG9xs5Q94xNJXv6iRzFz4cp8PUPmQc2ZbpS5b5N-pnsRHg@mail.gmail.com>
	<CAGX+XzvCg0GR8Ph5-Lykhe3Fs_EWqfu7K7M1s3S=9AO-7Ah_bg@mail.gmail.com>
Message-ID: <CAGX+XztzZ_MwFZWb41LkoiW9iSgKjX1WXC3f87J2BLk8MmNtuw@mail.gmail.com>

Brian, thanks for advice on IRC, it helped to do what I wanted.

Regards,
Anton

2011/9/29 Anton VG <anton.vazir at gmail.com>:
> Brian have suggested on IRC to use dialplan auth via
>
> <condition field="${sip_authorized}" expression="^true$" break="never">
> ? ? <anti-action application="respond" data="407"/>
> </condition>
>
> seems it can do the trick.
>
> 2011/9/29 Anthony Minessale <anthony.minessale at gmail.com>:
>> This seems odd....
>>
>> So you are making a global ACL to 0.0.0.0 which will let everyone in.
>> This takes precedence over the auth calls because you are passing the
>> ACL which is a means of authorization so you are never using the
>> actual challenge based auth at all. ? Of course they will all appear
>> as sip_authorized=true
>>
>> you do have the variable sip_acl_authed_by to tell which acl you passed.
>>
>> This has come up today in irc too, its daft to try to mix authed and
>> non-auth calls on the same profile.
>> Just make your authenticated users use a dedicated profile and the
>> non-authenticated ones use another.
>>
>>
>>
>> On Wed, Sep 28, 2011 at 1:49 PM, Muhammad Naseer Bhatti
>> <nbhatti at gmail.com> wrote:
>>> Hello Anton,
>>> I tried this before and ended up in the same situation. the event
>>> variable_sip_authorized, for some reason always return true. What I
>>> did in my cause, since I am using the variable_sip_auth_username to
>>> match the username for my billing, this does the trick. I don't know
>>> if this is a bug or something, but it works this way.
>>>
>>> Thanks
>>>
>>> On Wed, Sep 28, 2011 at 8:01 PM, Anton VG <anton.vazir at gmail.com> wrote:
>>>> Guys,
>>>>
>>>> trying to distinguish between authenticated SIP user and not
>>>> authenticated, still no luck
>>>>
>>>> I would like to allow anyone to call my FS users via SIP, so there are
>>>> registered users and everyone else
>>>>
>>>> so I have allowed
>>>>
>>>> <param name="auth-calls" value="true"/>
>>>> <param name="apply-inbound-acl" value="allow_all"/>
>>>>
>>>> acl.xml:
>>>> ? ?<list name="allow_all" default="allow">
>>>> ? ? ?<node type="allow" cidr="0.0.0.0/0"/>
>>>> ? ?</list>
>>>> Any call passes through, by always have
>>>>
>>>> [CHANNEL_CREATE] event
>>>> variable_sip_authorized: true
>>>>
>>>> in event headers. But I would like to know if user is authorized or not.
>>>>
>>>> if I disable authentication by setting this to false
>>>>
>>>> <param name="auth-calls" value="false"/>
>>>>
>>>> all calls do not have any auth headers.
>>>>
>>>> Any clue?
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 cvogel at lyonl.com  Fri Sep 30 23:44:03 2011
From: cvogel at lyonl.com (Chad Vogel)
Date: Fri, 30 Sep 2011 19:44:03 +0000
Subject: [Freeswitch-users] Customer SIP headers
Message-ID: <335D0A35-929D-4CAB-9E68-99FDE829C489@lyonl.com>

hello,

I'm looking to do custom sip headers. I wan't to send a 302 message and set the content header to something like this: Contact: <tel:+17208886435> ;expires=3600 Also on the From header I would like to add a isup-oli param to set ISUP cause codes.  how can i do this?

Chad
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