From kris at kriskinc.com  Fri Jul  1 00:01:15 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 30 Jun 2011 16:01:15 -0400
Subject: [Freeswitch-users] Optimize Freeswitch build
In-Reply-To: <BANLkTik3PfW5wTA0vSkuaysqQmtc8ng30Q@mail.gmail.com>
References: <BANLkTikiD8FgPGk35EVvc5BTJNUceQhscg@mail.gmail.com>
	<BANLkTikpZb9SDjF2fz3hmCQKk3DGHe1QMQ@mail.gmail.com>
	<BANLkTime0Qq70k3Px3ZcNsg48mOTXyT-8w@mail.gmail.com>
	<BD4247AF5F8E43EFA37E14AF6CD74DF0@dell9400>
	<BANLkTik3PfW5wTA0vSkuaysqQmtc8ng30Q@mail.gmail.com>
Message-ID: <BANLkTik2H0HdsXN40-vgaFR0gwgVmiXjyw@mail.gmail.com>

Many times you will discover that these flags don't speed up your
software - they expose bugs in your compiler ;).

In all seriousness if these flags magically made FS run X faster
they'd be in the bootstrap/configure scripts by default because the
devs would put them there.

On Thu, Jun 30, 2011 at 12:23 PM, curriegrad2004
<curriegrad2004 at gmail.com> wrote:
> You enable the optimizations by this following command in the configure stage:
> "CFLAGS="-g -O2 -march=native" CXXFLAGS="-g -O2 -march=native" ./configure"
>
> after that simply run a make and you should be good.
>
> On Thu, Jun 30, 2011 at 9:17 AM, Jan Berger <jan.berger at video24.no> wrote:
>> Assuming "maximum performance" is for the purpose of getting more calls when
>> it hardly matters unless CPU is your actual bottleneck.
>>
>>
>>
>> Jan
>>
>>
>>
>> ________________________________
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stephen
>> Wilde
>> Sent: 30. juni 2011 18:11
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Optimize Freeswitch build
>>
>>
>>
>> Thank you for the info, but how I can set this flags?
>>
>> I have to change the Makefile?
>>
>> I build Freeswitch with a "make install".
>>
>>
>>
>> On Thu, Jun 30, 2011 at 5:45 PM, curriegrad2004 <curriegrad2004 at gmail.com>
>> wrote:
>>
>> You can always use the cflags and cxxflags "-g -O2 -march=proc" to
>> optimize FreeSwitch altogether during the build process.
>>
>> On Thu, Jun 30, 2011 at 8:21 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:
>>> Hi all,
>>> it's possible to optimize the Freeswitch build to maximize the
>>> performance?
>>> Or this is the default configuration?
>>> I have not found any reference to this argument.
>>> Stephen
>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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
>



-- 
Kristian Kielhofner


From msc at freeswitch.org  Fri Jul  1 00:07:18 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Jun 2011 13:07:18 -0700
Subject: [Freeswitch-users] please help!!! how to set flag 'endconf' in
 bridging conference
In-Reply-To: <4E0CB983.8060407@gosilverplus.com>
References: <E826C94A3C6BE44198FAF4D66B5AB4D81BDA7E5903@VA3DIAXVS351.RED001.local>
	<4E0CB983.8060407@gosilverplus.com>
Message-ID: <BANLkTikb-xbvLwzsSv7nQJZncx8y_v+hpw@mail.gmail.com>

Can you pastebin exactly what you are doing to establish the call? Including
any relevant dialplan entries. Also, if you have modified
conference.conf.xml we would like to see that also.

-MC

On Thu, Jun 30, 2011 at 10:59 AM, ran zhang <rzhang at gosilverplus.com> wrote:

> hi all:
>
> I'm trying to creating a conference, so when first member enters the
> conference, he has to invite another members
> and have at least 1 other member to join to have the conference
> established, so i'm
> using bridging conference.
>
> I need this conference to be terminated when the original creator of the
> conference leaves no matter how many members are still left in the
> conference.
>
> i'm trying to set 'endconf' flag in a bridging conference using
> 'bridge:confname+flag{endconf}:user/10',
> so it wil invite user extension 10, but its giving me config error while
> running.
>
> can someone tell me what to do to solve this problem or get around?  the
> key
> is i only want the original member to be able to terminate the conference
> when he leaves, not other members assuming there are at least 2 members.
>
>
> _______________________________________________
> 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 anthony.minessale at gmail.com  Fri Jul  1 00:14:19 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 30 Jun 2011 15:14:19 -0500
Subject: [Freeswitch-users] ESL Send MESSAGE_WAITING Error - Cannot find
	profile
In-Reply-To: <BANLkTimYvT-apQ7BZS3FFR8EkpEpht2ohw@mail.gmail.com>
References: <BANLkTimGcbOiDUDgkRxbHZyPYASTidbVjQ@mail.gmail.com>
	<BANLkTikHzwA0KGyBXAXxOwNid1DfMZVMKA@mail.gmail.com>
	<BANLkTimYvT-apQ7BZS3FFR8EkpEpht2ohw@mail.gmail.com>
Message-ID: <BANLkTikK5D1Cqu8bN1uJxaqGUvM552waJw@mail.gmail.com>

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-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 marketing at cluecon.com  Fri Jul  1 00:18:33 2011
From: marketing at cluecon.com (marketing at cluecon.com)
Date: Thu, 30 Jun 2011 20:18:33 +0000
Subject: [Freeswitch-users] Join Us For A ClueCon Party!
Message-ID: <00000130e233f190-29052207-b67a-4d7b-adbd-629022bdf8ec-000000@email.amazonses.com>

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From rzhang at gosilverplus.com  Fri Jul  1 01:12:57 2011
From: rzhang at gosilverplus.com (ran zhang)
Date: Thu, 30 Jun 2011 14:12:57 -0700
Subject: [Freeswitch-users] instant messaging via freeswitch
Message-ID: <4E0CE6D9.5010805@gosilverplus.com>

hi All:

        I got the instant messaging working between 2 linphone clients 
via freeswitch, but i dont see any debug info while messages are sent. 
which module enables instant messaging in freeswitch as i'm trying to 
understand more about how freeswitch is handling this.


From curriegrad2004 at gmail.com  Fri Jul  1 01:21:01 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 30 Jun 2011 14:21:01 -0700
Subject: [Freeswitch-users] instant messaging via freeswitch
In-Reply-To: <4E0CE6D9.5010805@gosilverplus.com>
References: <4E0CE6D9.5010805@gosilverplus.com>
Message-ID: <BANLkTimX14pXX+=RVk80kEa59hjd=66Cgg@mail.gmail.com>

iirc, freeswitch doesn't support SIMPLE which I can tell you're trying to use.

On Thu, Jun 30, 2011 at 2:12 PM, ran zhang <rzhang at gosilverplus.com> wrote:
> hi All:
>
> ? ? ? ?I got the instant messaging working between 2 linphone clients
> via freeswitch, but i dont see any debug info while messages are sent.
> which module enables instant messaging in freeswitch as i'm trying to
> understand more about how freeswitch is handling this.
>
> _______________________________________________
> 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 wes-fs at 499x.com  Fri Jul  1 01:33:39 2011
From: wes-fs at 499x.com (Wes)
Date: Thu, 30 Jun 2011 16:33:39 -0500
Subject: [Freeswitch-users] significant delay waiting for response after
 playAndGetDigits()
In-Reply-To: <4E0CC7F0.201@499x.com>
References: <4E0C91EF.3090200@499x.com>	<4E0CC48F.6060703@499x.com>	<BANLkTinfwDbQ+6EvhgW4jftiuZ0oV5YUOw@mail.gmail.com>
	<4E0CC7F0.201@499x.com>
Message-ID: <4E0CEBB3.1050507@499x.com>

BTW, I rewrote the test script using sayPhrase along with an input 
callback to break out when the user presses a key, and the exact same 
behavior happens, here's the script:

local numberToCall = argv[1];
invalid = "ivr/ivr-that_was_an_invalid_entry.wav"
local session = 
freeswitch.Session("{ignore_early_media=true,origination_caller_id_name='MCW 
Phone 
Transcription',origination_caller_id_number=2580}sofia/gateway/mcw/"..numberToCall);

function key_press(session, input_type, data, args)
   if input_type == "dtmf" then
     freeswitch.consoleLog("info", "Key pressed: " .. data["digit"].."\n")
     return "break"
   end
end

if session:ready() then
   session:answer()
   session:set_tts_parms("flite", "awb");
   session:setInputCallback("key_press", "")
   session:sayPhrase("count_to_ten", "", "en")
   session:speak("you pressed a digit");
   freeswitch.consoleLog("info", "the user pressed a key\n");
   session:sayPhrase("count_to_ten", "", "en")
   session:speak("you pressed a digit");
   freeswitch.consoleLog("info", "the user pressed a key\n");
   session:sayPhrase("count_to_ten", "", "en")
   session:speak("you pressed a digit");
   freeswitch.consoleLog("info", "the user pressed a key\n");
   session:sayPhrase("count_to_ten", "", "en")
   session:speak("you pressed a digit");
   freeswitch.consoleLog("info", "the user pressed a key\n");
   session:sayPhrase("count_to_ten", "", "en")
   session:speak("you pressed a digit");
   freeswitch.consoleLog("info", "the user pressed a key\n");
   session:sayPhrase("count_to_ten", "", "en")
   session:speak("you pressed a digit");
   freeswitch.consoleLog("info", "the user pressed a key\n");
end

On 6/30/2011 2:01 PM, Wes wrote:
> I'm in fs_cli, and I think the loglevel is already at debug, since I set
> it to debug and saw the same messages that I have in the pastebin
>
> http://pastebin.freeswitch.org/16635
>
> in line 12, you can see that it received DTMF 1 right after speaking "One"
>
> but then, the second time I called pagd, you can see that in line 39, it got as far as speaking the number "Three"
> And, the third time, in line 81, it gets to speaking "Six", before detecting the DTMF.
>
> So, is that what you mean by monitoring the DTMF?  Even though I press the number immediately when the message says "go", it takes longer and longer each time to respond.
>
>
> On 6/30/2011 1:50 PM, Anthony Minessale wrote:
>> have you monitored your dtmf ?
>> is it 2833?
>>
>> Try console loglevel debug and look at the message saying when you got the dtmf.
>>
>>
>> On Thu, Jun 30, 2011 at 1:46 PM, Wes<wes-fs at 499x.com>   wrote:
>>> I've narrowed it down a bit.  I think this may be a bug (memory leak maybe?)
>>>
>>> It seems that the pagd function is very responsive on the first use, but
>>> on each repeated call to this function, it takes longer to respond to
>>> the user's keypress.
>>>
>>> The following test script illustrates the problem.  It asks user to
>>> press a key, then it starts counting up to 10.  The first time, it stops
>>> right away after you press a key, telling you what key you pressed, the
>>> second time, it gets as far as the number 3, the third time, it counts
>>> up to six, and so on, until it completes the count all the way to 10,
>>> and then it says you've pressed an invalid entry, even though you indeed
>>> pressed a key at the beginning of the message.
>>>
>>> lua script:
>>> http://pastebin.freeswitch.org/16637
>>>
>>> phrase xml:
>>> http://pastebin.freeswitch.org/16636
>>>
>>> Log:
>>> http://pastebin.freeswitch.org/16635
>>>
>>> Could someone try this example on their own system to see if they could
>>> duplicate the behavior?
>>>
>>> Thanks in advance!
>>>
>>>
>>> On 6/30/2011 10:10 AM, Wes wrote:
>>>> I am experiencing a delay waiting for the system to respond in a lua
>>>> script that is using playAndGetDigits.  I press a key while the voice is
>>>> still speaking, and it continues to speak for a few seconds before
>>>> moving to the part of the script for the particular keypress.
>>>>
>>>> Also, related: sometimes it speaks the "invalid" message even though the
>>>> key pressed is a valid one, and then immediately after speaking the
>>>> invalid message, it continues with the processing for the key that was
>>>> pressed.  So I know I didn't press an invalid key.
>>>>
>>>> --        session:execute('flush_dtmf');
>>>>             session:flushDigits()
>>>>             local digits = session:playAndGetDigits(1, 1, 3, 2000, "#",
>>>> "phrase:play_submit_rerecord:1:2:3", invalid, "\\d{1}")
>>>>             if ( digits == "1") then
>>>>
>>>> ETC
>>>>
>>>> here is the phrase:
>>>>
>>>> <macro name="play_submit_rerecord">
>>>> <input pattern="^([0-9#*]):([0-9#*]):([0-9#*])$">
>>>> <match>
>>>> <action function="speak-text"
>>>>                     data="To listen to your dictation, press $1, To submit
>>>> your dictation , press $2, To discard your dictation and start over,
>>>> press $3."/>
>>>> </match>
>>>> </input>
>>>> </macro>
>>>>
>>>> Is the speak-text just too cpu intensive and causing it to be slow to
>>>> respond to the DTMF?
>>>>
>>>> Any help would be appreciated... thanks!
>>>>
>>>> _______________________________________________
>>>> 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
>>> _______________________________________________
>>> 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
>>>
>>
> _______________________________________________
> 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 anthony.minessale at gmail.com  Fri Jul  1 01:39:02 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 30 Jun 2011 16:39:02 -0500
Subject: [Freeswitch-users] significant delay waiting for response after
	playAndGetDigits()
In-Reply-To: <4E0CEBB3.1050507@499x.com>
References: <4E0C91EF.3090200@499x.com> <4E0CC48F.6060703@499x.com>
	<BANLkTinfwDbQ+6EvhgW4jftiuZ0oV5YUOw@mail.gmail.com>
	<4E0CC7F0.201@499x.com> <4E0CEBB3.1050507@499x.com>
Message-ID: <BANLkTi=x9Z4Z6+qrhs16r_bi4q3HTxYckA@mail.gmail.com>

this is inappropriate for the mailing list
you should reproduce your problem on the current GIT with "make
current" and file it to JIRA


On Thu, Jun 30, 2011 at 4:33 PM, Wes <wes-fs at 499x.com> wrote:
> BTW, I rewrote the test script using sayPhrase along with an input
> callback to break out when the user presses a key, and the exact same
> behavior happens, here's the script:
>
> local numberToCall = argv[1];
> invalid = "ivr/ivr-that_was_an_invalid_entry.wav"
> local session =
> freeswitch.Session("{ignore_early_media=true,origination_caller_id_name='MCW
> Phone
> Transcription',origination_caller_id_number=2580}sofia/gateway/mcw/"..numberToCall);
>
> function key_press(session, input_type, data, args)
> ? if input_type == "dtmf" then
> ? ? freeswitch.consoleLog("info", "Key pressed: " .. data["digit"].."\n")
> ? ? return "break"
> ? end
> end
>
> if session:ready() then
> ? session:answer()
> ? session:set_tts_parms("flite", "awb");
> ? session:setInputCallback("key_press", "")
> ? session:sayPhrase("count_to_ten", "", "en")
> ? session:speak("you pressed a digit");
> ? freeswitch.consoleLog("info", "the user pressed a key\n");
> ? session:sayPhrase("count_to_ten", "", "en")
> ? session:speak("you pressed a digit");
> ? freeswitch.consoleLog("info", "the user pressed a key\n");
> ? session:sayPhrase("count_to_ten", "", "en")
> ? session:speak("you pressed a digit");
> ? freeswitch.consoleLog("info", "the user pressed a key\n");
> ? session:sayPhrase("count_to_ten", "", "en")
> ? session:speak("you pressed a digit");
> ? freeswitch.consoleLog("info", "the user pressed a key\n");
> ? session:sayPhrase("count_to_ten", "", "en")
> ? session:speak("you pressed a digit");
> ? freeswitch.consoleLog("info", "the user pressed a key\n");
> ? session:sayPhrase("count_to_ten", "", "en")
> ? session:speak("you pressed a digit");
> ? freeswitch.consoleLog("info", "the user pressed a key\n");
> end
>
> On 6/30/2011 2:01 PM, Wes wrote:
>> I'm in fs_cli, and I think the loglevel is already at debug, since I set
>> it to debug and saw the same messages that I have in the pastebin
>>
>> http://pastebin.freeswitch.org/16635
>>
>> in line 12, you can see that it received DTMF 1 right after speaking "One"
>>
>> but then, the second time I called pagd, you can see that in line 39, it got as far as speaking the number "Three"
>> And, the third time, in line 81, it gets to speaking "Six", before detecting the DTMF.
>>
>> So, is that what you mean by monitoring the DTMF? ?Even though I press the number immediately when the message says "go", it takes longer and longer each time to respond.
>>
>>
>> On 6/30/2011 1:50 PM, Anthony Minessale wrote:
>>> have you monitored your dtmf ?
>>> is it 2833?
>>>
>>> Try console loglevel debug and look at the message saying when you got the dtmf.
>>>
>>>
>>> On Thu, Jun 30, 2011 at 1:46 PM, Wes<wes-fs at 499x.com> ? wrote:
>>>> I've narrowed it down a bit. ?I think this may be a bug (memory leak maybe?)
>>>>
>>>> It seems that the pagd function is very responsive on the first use, but
>>>> on each repeated call to this function, it takes longer to respond to
>>>> the user's keypress.
>>>>
>>>> The following test script illustrates the problem. ?It asks user to
>>>> press a key, then it starts counting up to 10. ?The first time, it stops
>>>> right away after you press a key, telling you what key you pressed, the
>>>> second time, it gets as far as the number 3, the third time, it counts
>>>> up to six, and so on, until it completes the count all the way to 10,
>>>> and then it says you've pressed an invalid entry, even though you indeed
>>>> pressed a key at the beginning of the message.
>>>>
>>>> lua script:
>>>> http://pastebin.freeswitch.org/16637
>>>>
>>>> phrase xml:
>>>> http://pastebin.freeswitch.org/16636
>>>>
>>>> Log:
>>>> http://pastebin.freeswitch.org/16635
>>>>
>>>> Could someone try this example on their own system to see if they could
>>>> duplicate the behavior?
>>>>
>>>> Thanks in advance!
>>>>
>>>>
>>>> On 6/30/2011 10:10 AM, Wes wrote:
>>>>> I am experiencing a delay waiting for the system to respond in a lua
>>>>> script that is using playAndGetDigits. ?I press a key while the voice is
>>>>> still speaking, and it continues to speak for a few seconds before
>>>>> moving to the part of the script for the particular keypress.
>>>>>
>>>>> Also, related: sometimes it speaks the "invalid" message even though the
>>>>> key pressed is a valid one, and then immediately after speaking the
>>>>> invalid message, it continues with the processing for the key that was
>>>>> pressed. ?So I know I didn't press an invalid key.
>>>>>
>>>>> -- ? ? ? ?session:execute('flush_dtmf');
>>>>> ? ? ? ? ? ? session:flushDigits()
>>>>> ? ? ? ? ? ? local digits = session:playAndGetDigits(1, 1, 3, 2000, "#",
>>>>> "phrase:play_submit_rerecord:1:2:3", invalid, "\\d{1}")
>>>>> ? ? ? ? ? ? if ( digits == "1") then
>>>>>
>>>>> ETC
>>>>>
>>>>> here is the phrase:
>>>>>
>>>>> <macro name="play_submit_rerecord">
>>>>> <input pattern="^([0-9#*]):([0-9#*]):([0-9#*])$">
>>>>> <match>
>>>>> <action function="speak-text"
>>>>> ? ? ? ? ? ? ? ? ? ? data="To listen to your dictation, press $1, To submit
>>>>> your dictation , press $2, To discard your dictation and start over,
>>>>> press $3."/>
>>>>> </match>
>>>>> </input>
>>>>> </macro>
>>>>>
>>>>> Is the speak-text just too cpu intensive and causing it to be slow to
>>>>> respond to the DTMF?
>>>>>
>>>>> Any help would be appreciated... thanks!
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>> _______________________________________________
>>>> 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
>>>>
>>>
>> _______________________________________________
>> 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
>
> _______________________________________________
> 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
>



-- 
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 wes-fs at 499x.com  Fri Jul  1 01:44:38 2011
From: wes-fs at 499x.com (Wes)
Date: Thu, 30 Jun 2011 16:44:38 -0500
Subject: [Freeswitch-users] significant delay waiting for response after
 playAndGetDigits()
In-Reply-To: <BANLkTi=x9Z4Z6+qrhs16r_bi4q3HTxYckA@mail.gmail.com>
References: <4E0C91EF.3090200@499x.com>
	<4E0CC48F.6060703@499x.com>	<BANLkTinfwDbQ+6EvhgW4jftiuZ0oV5YUOw@mail.gmail.com>	<4E0CC7F0.201@499x.com>
	<4E0CEBB3.1050507@499x.com>
	<BANLkTi=x9Z4Z6+qrhs16r_bi4q3HTxYckA@mail.gmail.com>
Message-ID: <4E0CEE46.8050304@499x.com>

will do.  I just wanted to first be sure I wasn't doing something wrong.

On 6/30/2011 4:39 PM, Anthony Minessale wrote:
> this is inappropriate for the mailing list
> you should reproduce your problem on the current GIT with "make
> current" and file it to JIRA
>
>
> On Thu, Jun 30, 2011 at 4:33 PM, Wes<wes-fs at 499x.com>  wrote:
>> BTW, I rewrote the test script using sayPhrase along with an input
>> callback to break out when the user presses a key, and the exact same
>> behavior happens, here's the script:
>>
>> local numberToCall = argv[1];
>> invalid = "ivr/ivr-that_was_an_invalid_entry.wav"
>> local session =
>> freeswitch.Session("{ignore_early_media=true,origination_caller_id_name='MCW
>> Phone
>> Transcription',origination_caller_id_number=2580}sofia/gateway/mcw/"..numberToCall);
>>
>> function key_press(session, input_type, data, args)
>>    if input_type == "dtmf" then
>>      freeswitch.consoleLog("info", "Key pressed: " .. data["digit"].."\n")
>>      return "break"
>>    end
>> end
>>
>> if session:ready() then
>>    session:answer()
>>    session:set_tts_parms("flite", "awb");
>>    session:setInputCallback("key_press", "")
>>    session:sayPhrase("count_to_ten", "", "en")
>>    session:speak("you pressed a digit");
>>    freeswitch.consoleLog("info", "the user pressed a key\n");
>>    session:sayPhrase("count_to_ten", "", "en")
>>    session:speak("you pressed a digit");
>>    freeswitch.consoleLog("info", "the user pressed a key\n");
>>    session:sayPhrase("count_to_ten", "", "en")
>>    session:speak("you pressed a digit");
>>    freeswitch.consoleLog("info", "the user pressed a key\n");
>>    session:sayPhrase("count_to_ten", "", "en")
>>    session:speak("you pressed a digit");
>>    freeswitch.consoleLog("info", "the user pressed a key\n");
>>    session:sayPhrase("count_to_ten", "", "en")
>>    session:speak("you pressed a digit");
>>    freeswitch.consoleLog("info", "the user pressed a key\n");
>>    session:sayPhrase("count_to_ten", "", "en")
>>    session:speak("you pressed a digit");
>>    freeswitch.consoleLog("info", "the user pressed a key\n");
>> end
>>
>> On 6/30/2011 2:01 PM, Wes wrote:
>>> I'm in fs_cli, and I think the loglevel is already at debug, since I set
>>> it to debug and saw the same messages that I have in the pastebin
>>>
>>> http://pastebin.freeswitch.org/16635
>>>
>>> in line 12, you can see that it received DTMF 1 right after speaking "One"
>>>
>>> but then, the second time I called pagd, you can see that in line 39, it got as far as speaking the number "Three"
>>> And, the third time, in line 81, it gets to speaking "Six", before detecting the DTMF.
>>>
>>> So, is that what you mean by monitoring the DTMF?  Even though I press the number immediately when the message says "go", it takes longer and longer each time to respond.
>>>
>>>
>>> On 6/30/2011 1:50 PM, Anthony Minessale wrote:
>>>> have you monitored your dtmf ?
>>>> is it 2833?
>>>>
>>>> Try console loglevel debug and look at the message saying when you got the dtmf.
>>>>
>>>>
>>>> On Thu, Jun 30, 2011 at 1:46 PM, Wes<wes-fs at 499x.com>     wrote:
>>>>> I've narrowed it down a bit.  I think this may be a bug (memory leak maybe?)
>>>>>
>>>>> It seems that the pagd function is very responsive on the first use, but
>>>>> on each repeated call to this function, it takes longer to respond to
>>>>> the user's keypress.
>>>>>
>>>>> The following test script illustrates the problem.  It asks user to
>>>>> press a key, then it starts counting up to 10.  The first time, it stops
>>>>> right away after you press a key, telling you what key you pressed, the
>>>>> second time, it gets as far as the number 3, the third time, it counts
>>>>> up to six, and so on, until it completes the count all the way to 10,
>>>>> and then it says you've pressed an invalid entry, even though you indeed
>>>>> pressed a key at the beginning of the message.
>>>>>
>>>>> lua script:
>>>>> http://pastebin.freeswitch.org/16637
>>>>>
>>>>> phrase xml:
>>>>> http://pastebin.freeswitch.org/16636
>>>>>
>>>>> Log:
>>>>> http://pastebin.freeswitch.org/16635
>>>>>
>>>>> Could someone try this example on their own system to see if they could
>>>>> duplicate the behavior?
>>>>>
>>>>> Thanks in advance!
>>>>>
>>>>>
>>>>> On 6/30/2011 10:10 AM, Wes wrote:
>>>>>> I am experiencing a delay waiting for the system to respond in a lua
>>>>>> script that is using playAndGetDigits.  I press a key while the voice is
>>>>>> still speaking, and it continues to speak for a few seconds before
>>>>>> moving to the part of the script for the particular keypress.
>>>>>>
>>>>>> Also, related: sometimes it speaks the "invalid" message even though the
>>>>>> key pressed is a valid one, and then immediately after speaking the
>>>>>> invalid message, it continues with the processing for the key that was
>>>>>> pressed.  So I know I didn't press an invalid key.
>>>>>>
>>>>>> --        session:execute('flush_dtmf');
>>>>>>              session:flushDigits()
>>>>>>              local digits = session:playAndGetDigits(1, 1, 3, 2000, "#",
>>>>>> "phrase:play_submit_rerecord:1:2:3", invalid, "\\d{1}")
>>>>>>              if ( digits == "1") then
>>>>>>
>>>>>> ETC
>>>>>>
>>>>>> here is the phrase:
>>>>>>
>>>>>> <macro name="play_submit_rerecord">
>>>>>> <input pattern="^([0-9#*]):([0-9#*]):([0-9#*])$">
>>>>>> <match>
>>>>>> <action function="speak-text"
>>>>>>                      data="To listen to your dictation, press $1, To submit
>>>>>> your dictation , press $2, To discard your dictation and start over,
>>>>>> press $3."/>
>>>>>> </match>
>>>>>> </input>
>>>>>> </macro>
>>>>>>
>>>>>> Is the speak-text just too cpu intensive and causing it to be slow to
>>>>>> respond to the DTMF?
>>>>>>
>>>>>> Any help would be appreciated... thanks!
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>> _______________________________________________
>>>>> 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
>>>>>
>>> _______________________________________________
>>> 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
>> _______________________________________________
>> 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 steveayre at gmail.com  Fri Jul  1 01:49:55 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 30 Jun 2011 22:49:55 +0100
Subject: [Freeswitch-users] Optimize Freeswitch build
In-Reply-To: <BANLkTik2H0HdsXN40-vgaFR0gwgVmiXjyw@mail.gmail.com>
References: <BANLkTikiD8FgPGk35EVvc5BTJNUceQhscg@mail.gmail.com>
	<BANLkTikpZb9SDjF2fz3hmCQKk3DGHe1QMQ@mail.gmail.com>
	<BANLkTime0Qq70k3Px3ZcNsg48mOTXyT-8w@mail.gmail.com>
	<BD4247AF5F8E43EFA37E14AF6CD74DF0@dell9400>
	<BANLkTik3PfW5wTA0vSkuaysqQmtc8ng30Q@mail.gmail.com>
	<BANLkTik2H0HdsXN40-vgaFR0gwgVmiXjyw@mail.gmail.com>
Message-ID: <6E1CB389-BC11-426A-8FB8-96A82BB4FCE7@gmail.com>

It also makes debugging problems harder because you can't get a decent stack trace from a coredump.

Steve on iPhone

On 30 Jun 2011, at 21:01, Kristian Kielhofner <kris at kriskinc.com> wrote:

> Many times you will discover that these flags don't speed up your
> software - they expose bugs in your compiler ;).
> 
> In all seriousness if these flags magically made FS run X faster
> they'd be in the bootstrap/configure scripts by default because the
> devs would put them there.
> 
> On Thu, Jun 30, 2011 at 12:23 PM, curriegrad2004
> <curriegrad2004 at gmail.com> wrote:
>> You enable the optimizations by this following command in the configure stage:
>> "CFLAGS="-g -O2 -march=native" CXXFLAGS="-g -O2 -march=native" ./configure"
>> 
>> after that simply run a make and you should be good.
>> 
>> On Thu, Jun 30, 2011 at 9:17 AM, Jan Berger <jan.berger at video24.no> wrote:
>>> Assuming "maximum performance" is for the purpose of getting more calls when
>>> it hardly matters unless CPU is your actual bottleneck.
>>> 
>>> 
>>> 
>>> Jan
>>> 
>>> 
>>> 
>>> ________________________________
>>> 
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stephen
>>> Wilde
>>> Sent: 30. juni 2011 18:11
>>> To: FreeSWITCH Users Help
>>> Subject: Re: [Freeswitch-users] Optimize Freeswitch build
>>> 
>>> 
>>> 
>>> Thank you for the info, but how I can set this flags?
>>> 
>>> I have to change the Makefile?
>>> 
>>> I build Freeswitch with a "make install".
>>> 
>>> 
>>> 
>>> On Thu, Jun 30, 2011 at 5:45 PM, curriegrad2004 <curriegrad2004 at gmail.com>
>>> wrote:
>>> 
>>> You can always use the cflags and cxxflags "-g -O2 -march=proc" to
>>> optimize FreeSwitch altogether during the build process.
>>> 
>>> On Thu, Jun 30, 2011 at 8:21 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:
>>>> Hi all,
>>>> it's possible to optimize the Freeswitch build to maximize the
>>>> performance?
>>>> Or this is the default configuration?
>>>> I have not found any reference to this argument.
>>>> Stephen
>>> 
>>>> _______________________________________________
>>>> 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
>>>> 
>>>> 
>>> 
>>> _______________________________________________
>>> 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
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> 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
>>> 
>>> 
>> 
>> _______________________________________________
>> 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
>> 
> 
> 
> 
> -- 
> Kristian Kielhofner
> 
> _______________________________________________
> 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 kris at kriskinc.com  Fri Jul  1 01:51:00 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 30 Jun 2011 17:51:00 -0400
Subject: [Freeswitch-users] instant messaging via freeswitch
In-Reply-To: <BANLkTimX14pXX+=RVk80kEa59hjd=66Cgg@mail.gmail.com>
References: <4E0CE6D9.5010805@gosilverplus.com>
	<BANLkTimX14pXX+=RVk80kEa59hjd=66Cgg@mail.gmail.com>
Message-ID: <BANLkTi=vRGz0NY00sVH1YebRoGsN9v90nQ@mail.gmail.com>

FreeSWITCH supports SIMPLE.

On Thu, Jun 30, 2011 at 5:21 PM, curriegrad2004
<curriegrad2004 at gmail.com> wrote:
> iirc, freeswitch doesn't support SIMPLE which I can tell you're trying to use.
>
> On Thu, Jun 30, 2011 at 2:12 PM, ran zhang <rzhang at gosilverplus.com> wrote:
>> hi All:
>>
>> ? ? ? ?I got the instant messaging working between 2 linphone clients
>> via freeswitch, but i dont see any debug info while messages are sent.
>> which module enables instant messaging in freeswitch as i'm trying to
>> understand more about how freeswitch is handling this.
>>
>> _______________________________________________
>> 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
>>
>
> _______________________________________________
> 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
>



-- 
Kristian Kielhofner


From steveayre at gmail.com  Fri Jul  1 01:51:37 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 30 Jun 2011 22:51:37 +0100
Subject: [Freeswitch-users] Freeswitch dynamic routing of all calls
In-Reply-To: <BANLkTimuCjRcVccyrPpgQeY6N+UamyNQ=A@mail.gmail.com>
References: <BANLkTimuCjRcVccyrPpgQeY6N+UamyNQ=A@mail.gmail.com>
Message-ID: <518E6CA8-4198-4484-BF03-2DEFFA6493EA@gmail.com>

Yes it's possible and I suggest you look at esl

Steve on iPhone

On 30 Jun 2011, at 09:54, Mateusz Bartczak <netcentrica at gmail.com> wrote:

> Hi all
> 
> I'm new to FS and I would like to know is it possible to implement following scenario:
> 
> 1. User dials number
> 2. Routing script detects outgoing call event. Every call needs to be handled by routing script.
> 3. Routing script takes in input: user name, domain, dialed number. Than it query database to find best SIP trunk to route the call, it also checks destination price per minute and calculates maximum call time for prepaid user.
> 4. Routing script output is: SIP trunk to use, SIP call parameters (ie. callerid), maximum call duration
> 5. FS read output from routing script and make call using returned parameters
> 
> Preferred routing implementation technology: background running unix deamon written in Java or PHP. Connection with FS via socket.
> 
> Event routing script will be multi-threaded, must be able to deal with a lot of calls in parallel and processing of one call should not block processing of other calls (I have this problem with Yate voip server, and that's really big problem)
> 
> Is it possible to do this using FS?
> Any advices where to search for additional info? I know that there is event handler but can it return "dialstring" for outgoing call events?
> Some code examples?
> 
> I will really appreciate your help
> 
> 
> 
> 
> _______________________________________________
> 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 kris at kriskinc.com  Fri Jul  1 01:52:04 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 30 Jun 2011 17:52:04 -0400
Subject: [Freeswitch-users] instant messaging via freeswitch
In-Reply-To: <4E0CE6D9.5010805@gosilverplus.com>
References: <4E0CE6D9.5010805@gosilverplus.com>
Message-ID: <BANLkTikxVpvP4Yeud03CbmUciq1fDgiX0g@mail.gmail.com>

Don't start multiple threads on the same topic.

As I told you in another thread, FreeSWITCH supports SIP messages
natively using Sofia.  If you want to see the messages you'll need to
enable SIP trace with "sofia global siptrace on".

On Thu, Jun 30, 2011 at 5:12 PM, ran zhang <rzhang at gosilverplus.com> wrote:
> hi All:
>
> ? ? ? ?I got the instant messaging working between 2 linphone clients
> via freeswitch, but i dont see any debug info while messages are sent.
> which module enables instant messaging in freeswitch as i'm trying to
> understand more about how freeswitch is handling this.
>
> _______________________________________________
> 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
>



-- 
Kristian Kielhofner


From avi at avimarcus.net  Fri Jul  1 01:57:26 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 1 Jul 2011 00:57:26 +0300
Subject: [Freeswitch-users] Freeswitch dynamic routing of all calls
In-Reply-To: <518E6CA8-4198-4484-BF03-2DEFFA6493EA@gmail.com>
References: <BANLkTimuCjRcVccyrPpgQeY6N+UamyNQ=A@mail.gmail.com>
	<518E6CA8-4198-4484-BF03-2DEFFA6493EA@gmail.com>
Message-ID: <BANLkTim6VuMdy7dRoUv2j5jFZMfCXdBBFw@mail.gmail.com>

It doesn't sound like you're doing anything particularly complicated -
can you just do this within the normal XML dialplan?
-Avi

On Fri, Jul 1, 2011 at 12:51 AM, Steven Ayre <steveayre at gmail.com> wrote:
> Yes it's possible and I suggest you look at esl
>
> Steve on iPhone
>
> On 30 Jun 2011, at 09:54, Mateusz Bartczak <netcentrica at gmail.com> wrote:
>
>> Hi all
>>
>> I'm new to FS and I would like to know is it possible to implement following scenario:
>>
>> 1. User dials number
>> 2. Routing script detects outgoing call event. Every call needs to be handled by routing script.
>> 3. Routing script takes in input: user name, domain, dialed number. Than it query database to find best SIP trunk to route the call, it also checks destination price per minute and calculates maximum call time for prepaid user.
>> 4. Routing script output is: SIP trunk to use, SIP call parameters (ie. callerid), maximum call duration
>> 5. FS read output from routing script and make call using returned parameters
>>
>> Preferred routing implementation technology: background running unix deamon written in Java or PHP. Connection with FS via socket.
>>
>> Event routing script will be multi-threaded, must be able to deal with a lot of calls in parallel and processing of one call should not block processing of other calls (I have this problem with Yate voip server, and that's really big problem)
>>
>> Is it possible to do this using FS?
>> Any advices where to search for additional info? I know that there is event handler but can it return "dialstring" for outgoing call events?
>> Some code examples?
>>
>> I will really appreciate your help
>>
>>
>>
>>
>> _______________________________________________
>> 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
>
> _______________________________________________
> 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 anthony.minessale at gmail.com  Fri Jul  1 03:00:23 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 30 Jun 2011 18:00:23 -0500
Subject: [Freeswitch-users] NAT Traversal on SFLphone - FS not Auto
 Changing port
In-Reply-To: <BANLkTikOYsN8VLgHK7UTAYGTeri3nOpY_A@mail.gmail.com>
References: <BANLkTikOYsN8VLgHK7UTAYGTeri3nOpY_A@mail.gmail.com>
Message-ID: <BANLkTimmwvuT9eBLZprpDYyoEbdGsd6Ydw@mail.gmail.com>

you might want to include the sip-trace and the whole call setup .
Your log shows its getting audio from the phone and autodetects the correct
rtp port.
Do you have both on at once? maybe one already owns your soundcard so it
breaks the other, try turning them both off and only start the one.
while the call is up, get a pcap on both sides and see if audio is going to
the right place.


On Thu, Jun 30, 2011 at 1:23 AM, Avi Marcus <avi at avimarcus.net> wrote:

> http://pastebin.freeswitch.org/16627
> I've got a Linksys ATA behind NAT and a softphone behind NAT. Both seem to
> register with the same UDP-NAT string and both have the same type of contact
> string.. but on one FS is rewriting the RTP IP to work properly - the 1102
> works, but on 1000 I don't seem to be getting any audio.
>
> Is there some hidden parameter I can't see affecting this? Both are dialing
> the same extension 9664 default MOH stuff...
> This softphone actually seems to have a responsive gui in linux. If I can
> get the audio to actually work, that would great!
> (Oh, it has tls/srtp too, it seems)
>
> Thanks!
> <http://pastebin.freeswitch.org/16627>
> -Avi Marcus
>
>
>
> _______________________________________________
> 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
>
>


-- 
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
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From rzhang at gosilverplus.com  Fri Jul  1 03:28:17 2011
From: rzhang at gosilverplus.com (ran zhang)
Date: Thu, 30 Jun 2011 16:28:17 -0700
Subject: [Freeswitch-users] how to send messages to all members in a
	conference via freeswitch?
In-Reply-To: <BANLkTimmwvuT9eBLZprpDYyoEbdGsd6Ydw@mail.gmail.com>
References: <BANLkTikOYsN8VLgHK7UTAYGTeri3nOpY_A@mail.gmail.com>
	<BANLkTimmwvuT9eBLZprpDYyoEbdGsd6Ydw@mail.gmail.com>
Message-ID: <4E0D0691.8080503@gosilverplus.com>

Hi All:

            I'm using Linphone as my VOIP client, i want 1 member to 
send instant messages to the conference and all other members in the 
conference should receive them.

           I can't figure out a way to do this,  Linphone only allows to 
send messages to other members on the contact list, even if Linphone can 
do it, can freeswitch handle this?


From tomasz at kopacki.eu  Fri Jul  1 00:51:32 2011
From: tomasz at kopacki.eu (Tomasz Kopacki)
Date: Thu, 30 Jun 2011 20:51:32 +0000
Subject: [Freeswitch-users] mod_limit
Message-ID: <443EC226AAEABB48B58CAF9D56D80AB41604F0@hektor.dom.local>

  Hi,
Im trying to set mod_limit but i dont understand what are realm and id parameters for.
I have 2 users groups - Group A and Group B. Group A has two domains and Group B has three domains.
I want to set limit 100cc on group A and 150cc on group B.
I did this:

<action application="limit" data="hash realm groupA 100 />

<action application="limit" data="hash realm groupB 150 />

Will it works ?
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From rzhang at gosilverplus.com  Fri Jul  1 05:29:19 2011
From: rzhang at gosilverplus.com (ran zhang)
Date: Thu, 30 Jun 2011 18:29:19 -0700
Subject: [Freeswitch-users] please help!!! how to set flag 'endconf' in
 bridging conference
In-Reply-To: <BANLkTi=Cmuz6a3K8smK3Nr46E2ptg=pZdQ@mail.gmail.com>
References: <E826C94A3C6BE44198FAF4D66B5AB4D81BDA7E5903@VA3DIAXVS351.RED001.local>	<4E0CB983.8060407@gosilverplus.com>	<BANLkTikb-xbvLwzsSv7nQJZncx8y_v+hpw@mail.gmail.com>	<4E0CE1B7.60609@gosilverplus.com>
	<BANLkTi=Cmuz6a3K8smK3Nr46E2ptg=pZdQ@mail.gmail.com>
Message-ID: <4E0D22EF.2070009@gosilverplus.com>

        I attached the section in my dialplan that handles the bridging 
conference, when the first user (just say user 99)  dials '20', he will 
invite user 10 to join conference, the conference will only be 
established if user 10 accepts the invitation, after this, other users 
can join the conference by dialing '20'.

        I want that if the first user (user 99 in this case)  or the 
user been invited (user 10 in this case)  leaves the conference, no 
matter how many people are still in the conference, it will close down 
the conference.   so i'm trying to use the 'endconf' flag, but 
apparently it is not valid syntax for bridging conference as I get a 
config error while running.  If I take out the '+flags{endconf} ', i 
wont get a config error while running, but then conference will only 
close down when there is 1 person left.

       I have also tried creating 2 difference conference profiles,  one 
profile has the 'endconf' set in the 'member-flags', one profiles doesnt 
have 'endconf' set, so user99 and user20 joins the 20 at profile1 
conference, and other users joins the 20 at profile2 conference, that 
doesnt seem to work neither.   I have pasted conference.conf.xml file as 
well for ur review.

<include>
   <context name="test">
     <extension name="global" continue="true">
       <condition>
       </condition>
     </extension>

     <extension name="conf">
       <condition field="destination_number" expression="^(20)$">  </condition>
         <action application="answer"/>
         <action application="conference" data="bridge:20 at default+flags{endconf}:user/10"/>
         <action application="conference" data="20 at default"/>
       </condition>
     </extension>
   </context>
</include>





On 6/30/2011 6:16 PM, Michael Collins wrote:
> Please put this information on pastebin and reply to the list so that 
> we can all discuss it.
> -MC
>
> On Thu, Jun 30, 2011 at 1:51 PM, ran zhang <rzhang at gosilverplus.com 
> <mailto:rzhang at gosilverplus.com>> wrote:
>
>     Mr Collins:
>
>            I attached the section in my dialplan that handles the
>     bridging conference, when the first user (just say user 99)  dials
>     '20', he will invite user 10 to join conference, the conference
>     will only be established if user 10 accepts the invitation, after
>     this, other users can join the conference by dialing '20'.
>
>            I want that if the first user (user 99 in this case)  or
>     the user been invited (user 10 in this case)  leaves the
>     conference, no matter how many people are still in the conference,
>     it will close down the conference.   so i'm trying to use the
>     'endconf' flag, but apparently it is not valid syntax for bridging
>     conference as I get a config error while running.  If I take out
>     the '+flags{endconf} ', i wont get a config error while running,
>     but then conference will only close down when there is 1 person left.
>
>           I have also tried creating 2 difference conference
>     profiles,  one profile has the 'endconf' set in the
>     'member-flags', one profiles doesnt have 'endconf' set, so user99
>     and user20 joins the 20 at profile1 conference, and other users joins
>     the 20 at profile2 conference, that doesnt seem to work neither.   I
>     have pasted conference.conf.xml file as well for ur review.
>
>     <include>
>        <context name="test">
>          <extension name="global" continue="true">
>            <condition>
>            </condition>
>          </extension>
>
>          <extension name="conf">
>            <condition field="destination_number" expression="^(20)$">  </condition>
>              <action application="answer"/>
>              <action application="conference" data="bridge:20 at default+flags{endconf}:user/10"/>
>              <action application="conference" data="20 at default"/>
>            </condition>
>          </extension>
>        </context>
>     </include>
>
>
>     <configuration name="conference.conf" description="Audio Conference">
>        <!-- Advertise certain presence on startup . -->
>        <advertise>
>          <room name="3001@$${domain}" status="FreeSWITCH"/>
>        </advertise>
>
>        <!-- These are the default keys that map when you do not specify a caller control group -->	
>        <!-- Note: none and default are reserved names for group names.  Disabled if dist-dtmf member flag is set. -->	
>        <caller-controls>
>          <group name="default">
>            <control action="mute" digits="0"/>
>            <control action="deaf mute" digits="*"/>
>            <control action="energy up" digits="9"/>
>            <control action="energy equ" digits="8"/>
>            <control action="energy dn" digits="7"/>
>            <control action="vol talk up" digits="3"/>
>            <control action="vol talk zero" digits="2"/>
>            <control action="vol talk dn" digits="1"/>
>            <control action="vol listen up" digits="6"/>
>            <control action="vol listen zero" digits="5"/>
>            <control action="vol listen dn" digits="4"/>
>            <control action="hangup" digits="#"/>
>          </group>
>        </caller-controls>
>
>        <!-- Profiles are collections of settings you can reference by name. -->
>        <profiles>
>          <!--If no profile is specified it will default to "default"-->
>          <profile name="default">
>            <!-- Domain (for presence) -->
>            <param name="domain" value="$${domain}"/>
>            <!-- Sample Rate-->
>            <param name="rate" value="8000"/>
>            <!-- Number of milliseconds per frame -->
>            <param name="interval" value="20"/>
>            <!-- Energy level required for audio to be sent to the other users -->
>            <param name="energy-level" value="300"/>
>
>            <!--Can be | delim of waste|mute|deaf|dist-dtmf waste will always transmit data to each channel
>                even during silence.  dist-dtmf propagates dtmfs to all other members, but channel controls
>     	  via dtmf will be disabled. -->
>            <param name="member-flags" value="dist-dtmf"/>
>
>            <!-- Name of the caller control group to use for this profile -->
>            <!--<param name="caller-controls" value="some name"/>  -->
>            <!-- TTS Engine to use -->
>            <!--<param name="tts-engine" value="cepstral"/>-->
>            <!-- TTS Voice to use -->
>            <!--<param name="tts-voice" value="david"/>-->
>
>            <!-- If TTS is enabled all audio-file params beginning with -->
>            <!-- 'say:' will be considered text to say with TTS -->
>            <!-- Override the default path here, after which you use relative paths in the other sound params -->
>            <!-- Note: The default path is the conference's first caller's sound_prefix -->
>            <!--<param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/>-->
>            <!-- File to play to acknowledge succees -->
>            <!--<param name="ack-sound" value="beep.wav"/>-->
>            <!-- File to play to acknowledge failure -->
>            <!--<param name="nack-sound" value="beeperr.wav"/>-->
>            <!-- File to play to acknowledge muted -->
>            <!-- Conference pin -->
>            <!--<param name="pin" value="12345"/>-->
>            <!-- Default Caller ID Name for outbound calls -->
>            <param name="caller-id-name" value="$${outbound_caller_name}"/>
>            <!-- Default Caller ID Number for outbound calls -->
>            <param name="caller-id-number" value="1234"/>
>            <!-- Suppress start and stop talking events -->
>            <!--<param name="suppress-events" value="start-talking,stop-talking"/>  -->
>            <!-- enable comfort noise generation -->
>            <param name="comfort-noise" value="true"/>
>            <!-- Uncomment auto-record to toggle recording every conference call. -->
>            <!-- Another valid value isshout://user:pass at server.com/live.mp3  <mailto:shout://user:pass at server.com/live.mp3>    -->
>            <!--
>            <param name="auto-record" value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>            -->
>          </profile>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>     On 6/30/2011 1:07 PM, Michael Collins wrote:
>>     Can you pastebin exactly what you are doing to establish the
>>     call? Including any relevant dialplan entries. Also, if you have
>>     modified conference.conf.xml we would like to see that also.
>>
>>     -MC
>>
>>     On Thu, Jun 30, 2011 at 10:59 AM, ran zhang
>>     <rzhang at gosilverplus.com <mailto:rzhang at gosilverplus.com>> wrote:
>>
>>         hi all:
>>
>>         I'm trying to creating a conference, so when first member
>>         enters the
>>         conference, he has to invite another members
>>         and have at least 1 other member to join to have the
>>         conference established, so i'm
>>         using bridging conference.
>>
>>         I need this conference to be terminated when the original
>>         creator of the
>>         conference leaves no matter how many members are still left
>>         in the conference.
>>
>>         i'm trying to set 'endconf' flag in a bridging conference using
>>         'bridge:confname+flag{endconf}:user/10',
>>         so it wil invite user extension 10, but its giving me config
>>         error while
>>         running.
>>
>>         can someone tell me what to do to solve this problem or get
>>         around?  the key
>>         is i only want the original member to be able to terminate
>>         the conference
>>         when he leaves, not other members assuming there are at least
>>         2 members.
>>
>>
>>         _______________________________________________
>>         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
>>         <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
>>
>>
>>
>>     _______________________________________________
>>     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  <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 gcd at i.ph  Fri Jul  1 06:04:37 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Fri, 1 Jul 2011 10:04:37 +0800
Subject: [Freeswitch-users] please help!!! how to set flag 'endconf' in
 bridging conference
In-Reply-To: <4E0D22EF.2070009@gosilverplus.com>
References: <E826C94A3C6BE44198FAF4D66B5AB4D81BDA7E5903@VA3DIAXVS351.RED001.local>
	<4E0CB983.8060407@gosilverplus.com>
	<BANLkTikb-xbvLwzsSv7nQJZncx8y_v+hpw@mail.gmail.com>
	<4E0CE1B7.60609@gosilverplus.com>
	<BANLkTi=Cmuz6a3K8smK3Nr46E2ptg=pZdQ@mail.gmail.com>
	<4E0D22EF.2070009@gosilverplus.com>
Message-ID: <BANLkTinSmZEU3-zzqDQC2EnoMQVY+LWGfA@mail.gmail.com>

hi michael,

i'd like to have that feature, too, because our telco has an unusual line
signalling - it only tear down the connection when the A-party hangs up
first (Clear Forward).  if i transfer the callee to the conference, the
telco line remains off-hook forever. as an interim solution, i'm using the
terminate_on_silence parameter.

-nandy

On Fri, Jul 1, 2011 at 9:29 AM, ran zhang <rzhang at gosilverplus.com> wrote:

>         I attached the section in my dialplan that handles the bridging
> conference, when the first user (just say user 99)  dials '20', he will
> invite user 10 to join conference, the conference will only be established
> if user 10 accepts the invitation, after this, other users can join the
> conference by dialing '20'.
>
>        I want that if the first user (user 99 in this case)  or the user
> been invited (user 10 in this case)  leaves the conference, no matter how
> many people are still in the conference, it will close down the
> conference.   so i'm trying to use the 'endconf' flag, but apparently it is
> not valid syntax for bridging conference as I get a config error while
> running.  If I take out the '+flags{endconf} ', i wont get a config error
> while running, but then conference will only close down when there is 1
> person left.
>
>       I have also tried creating 2 difference conference profiles,  one
> profile has the 'endconf' set in the 'member-flags', one profiles doesnt
> have 'endconf' set, so user99 and user20 joins the 20 at profile1 conference,
> and other users joins the 20 at profile2 conference, that doesnt seem to work
> neither.   I have pasted conference.conf.xml file as well for ur review.
>
>
> <include>
>   <context name="test">
>     <extension name="global" continue="true">
>       <condition>
>       </condition>
>     </extension>
>
>     <extension name="conf">
>       <condition field="destination_number" expression="^(20)$" > </condition>
>         <action application="answer"/>
>         <action application="conference" data="bridge:20 at default+flags{endconf}:user/10"/>
>         <action application="conference" data="20 at default"/>
>       </condition>
>     </extension>
>   </context>
> </include>
>
>
>
>
>
> On 6/30/2011 6:16 PM, Michael Collins wrote:
>
> Please put this information on pastebin and reply to the list so that we
> can all discuss it.
> -MC
>
> On Thu, Jun 30, 2011 at 1:51 PM, ran zhang <rzhang at gosilverplus.com>wrote:
>
>>  Mr Collins:
>>
>>        I attached the section in my dialplan that handles the bridging
>> conference, when the first user (just say user 99)  dials '20', he will
>> invite user 10 to join conference, the conference will only be established
>> if user 10 accepts the invitation, after this, other users can join the
>> conference by dialing '20'.
>>
>>        I want that if the first user (user 99 in this case)  or the user
>> been invited (user 10 in this case)  leaves the conference, no matter how
>> many people are still in the conference, it will close down the
>> conference.   so i'm trying to use the 'endconf' flag, but apparently it is
>> not valid syntax for bridging conference as I get a config error while
>> running.  If I take out the '+flags{endconf} ', i wont get a config error
>> while running, but then conference will only close down when there is 1
>> person left.
>>
>>       I have also tried creating 2 difference conference profiles,  one
>> profile has the 'endconf' set in the 'member-flags', one profiles doesnt
>> have 'endconf' set, so user99 and user20 joins the 20 at profile1conference, and other users joins the 20 at profile2conference, that doesnt seem to work neither.   I have pasted
>> conference.conf.xml file as well for ur review.
>>
>>
>> <include>
>>   <context name="test">
>>     <extension name="global" continue="true">
>>       <condition>
>>       </condition>
>>     </extension>
>>
>>     <extension name="conf">
>>       <condition field="destination_number" expression="^(20)$" > </condition>
>>         <action application="answer"/>
>>         <action application="conference" data="bridge:20 at default+flags{endconf}:user/10"/>
>>         <action application="conference" data="20 at default"/>
>>       </condition>
>>     </extension>
>>   </context>
>> </include>
>>
>>
>> <configuration name="conference.conf" description="Audio Conference">
>>   <!-- Advertise certain presence on startup . -->
>>   <advertise>
>>     <room name="3001@$${domain}" status="FreeSWITCH"/>
>>   </advertise>
>>
>>   <!-- These are the default keys that map when you do not specify a caller control group -->	
>>   <!-- Note: none and default are reserved names for group names.  Disabled if dist-dtmf member flag is set. -->	
>>   <caller-controls>
>>     <group name="default">
>>       <control action="mute" digits="0"/>
>>       <control action="deaf mute" digits="*"/>
>>       <control action="energy up" digits="9"/>
>>       <control action="energy equ" digits="8"/>
>>       <control action="energy dn" digits="7"/>
>>       <control action="vol talk up" digits="3"/>
>>       <control action="vol talk zero" digits="2"/>
>>       <control action="vol talk dn" digits="1"/>
>>       <control action="vol listen up" digits="6"/>
>>       <control action="vol listen zero" digits="5"/>
>>       <control action="vol listen dn" digits="4"/>
>>       <control action="hangup" digits="#"/>
>>     </group>
>>   </caller-controls>
>>
>>   <!-- Profiles are collections of settings you can reference by name. -->
>>   <profiles>
>>     <!--If no profile is specified it will default to "default"-->
>>     <profile name="default">
>>       <!-- Domain (for presence) -->
>>       <param name="domain" value="$${domain}"/>
>>       <!-- Sample Rate-->
>>       <param name="rate" value="8000"/>
>>       <!-- Number of milliseconds per frame -->
>>       <param name="interval" value="20"/>
>>       <!-- Energy level required for audio to be sent to the other users -->
>>       <param name="energy-level" value="300"/>
>>
>>       <!--Can be | delim of waste|mute|deaf|dist-dtmf waste will always transmit data to each channel
>>           even during silence.  dist-dtmf propagates dtmfs to all other members, but channel controls
>> 	  via dtmf will be disabled. -->
>>       <param name="member-flags" value="dist-dtmf"/>
>>
>>       <!-- Name of the caller control group to use for this profile -->
>>       <!-- <param name="caller-controls" value="some name"/> -->
>>       <!-- TTS Engine to use -->
>>       <!--<param name="tts-engine" value="cepstral"/>-->
>>       <!-- TTS Voice to use -->
>>       <!--<param name="tts-voice" value="david"/>-->
>>
>>       <!-- If TTS is enabled all audio-file params beginning with -->
>>       <!-- 'say:' will be considered text to say with TTS -->
>>       <!-- Override the default path here, after which you use relative paths in the other sound params -->
>>       <!-- Note: The default path is the conference's first caller's sound_prefix -->
>>       <!--<param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/>-->
>>       <!-- File to play to acknowledge succees -->
>>       <!--<param name="ack-sound" value="beep.wav"/>-->
>>       <!-- File to play to acknowledge failure -->
>>       <!--<param name="nack-sound" value="beeperr.wav"/>-->
>>       <!-- File to play to acknowledge muted -->
>>       <!-- Conference pin -->
>>       <!--<param name="pin" value="12345"/>-->
>>       <!-- Default Caller ID Name for outbound calls -->
>>       <param name="caller-id-name" value="$${outbound_caller_name}"/>
>>       <!-- Default Caller ID Number for outbound calls -->
>>       <param name="caller-id-number" value="1234"/>
>>       <!-- Suppress start and stop talking events -->
>>       <!-- <param name="suppress-events" value="start-talking,stop-talking"/> -->
>>       <!-- enable comfort noise generation -->
>>       <param name="comfort-noise" value="true"/>
>>       <!-- Uncomment auto-record to toggle recording every conference call. -->
>>       <!-- Another valid value is   shout://user:pass at server.com/live.mp3   -->
>>       <!--
>>       <param name="auto-record" value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>>       -->
>>     </profile>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On 6/30/2011 1:07 PM, Michael Collins wrote:
>>
>> Can you pastebin exactly what you are doing to establish the call?
>> Including any relevant dialplan entries. Also, if you have modified
>> conference.conf.xml we would like to see that also.
>>
>> -MC
>>
>>  On Thu, Jun 30, 2011 at 10:59 AM, ran zhang <rzhang at gosilverplus.com>wrote:
>>
>>> hi all:
>>>
>>> I'm trying to creating a conference, so when first member enters the
>>> conference, he has to invite another members
>>> and have at least 1 other member to join to have the conference
>>> established, so i'm
>>> using bridging conference.
>>>
>>> I need this conference to be terminated when the original creator of the
>>> conference leaves no matter how many members are still left in the
>>> conference.
>>>
>>> i'm trying to set 'endconf' flag in a bridging conference using
>>> 'bridge:confname+flag{endconf}:user/10',
>>> so it wil invite user extension 10, but its giving me config error while
>>> running.
>>>
>>> can someone tell me what to do to solve this problem or get around?  the
>>> key
>>> is i only want the original member to be able to terminate the conference
>>> when he leaves, not other members assuming there are at least 2 members.
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>>
>>
>>
>
>
> _______________________________________________
> 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 gcd at i.ph  Fri Jul  1 06:12:11 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Fri, 1 Jul 2011 10:12:11 +0800
Subject: [Freeswitch-users] please help!!! how to set flag 'endconf' in
 bridging conference
In-Reply-To: <BANLkTinSmZEU3-zzqDQC2EnoMQVY+LWGfA@mail.gmail.com>
References: <E826C94A3C6BE44198FAF4D66B5AB4D81BDA7E5903@VA3DIAXVS351.RED001.local>
	<4E0CB983.8060407@gosilverplus.com>
	<BANLkTikb-xbvLwzsSv7nQJZncx8y_v+hpw@mail.gmail.com>
	<4E0CE1B7.60609@gosilverplus.com>
	<BANLkTi=Cmuz6a3K8smK3Nr46E2ptg=pZdQ@mail.gmail.com>
	<4E0D22EF.2070009@gosilverplus.com>
	<BANLkTinSmZEU3-zzqDQC2EnoMQVY+LWGfA@mail.gmail.com>
Message-ID: <BANLkTim6mjRv7sf8d8kCJnqNSZGxenDgEQ@mail.gmail.com>

hi michael,

pls disregard me last post. i just digged the "endconf" parameter. i'll try
it. tks.


On Fri, Jul 1, 2011 at 10:04 AM, Nandy Dagondon <gcd at i.ph> wrote:

> hi michael,
>
> i'd like to have that feature, too, because our telco has an unusual line
> signalling - it only tear down the connection when the A-party hangs up
> first (Clear Forward).  if i transfer the callee to the conference, the
> telco line remains off-hook forever. as an interim solution, i'm using the
> terminate_on_silence parameter.
>
> -nandy
>
> On Fri, Jul 1, 2011 at 9:29 AM, ran zhang <rzhang at gosilverplus.com> wrote:
>
>>         I attached the section in my dialplan that handles the bridging
>> conference, when the first user (just say user 99)  dials '20', he will
>> invite user 10 to join conference, the conference will only be established
>> if user 10 accepts the invitation, after this, other users can join the
>> conference by dialing '20'.
>>
>>        I want that if the first user (user 99 in this case)  or the user
>> been invited (user 10 in this case)  leaves the conference, no matter how
>> many people are still in the conference, it will close down the
>> conference.   so i'm trying to use the 'endconf' flag, but apparently it is
>> not valid syntax for bridging conference as I get a config error while
>> running.  If I take out the '+flags{endconf} ', i wont get a config error
>> while running, but then conference will only close down when there is 1
>> person left.
>>
>>       I have also tried creating 2 difference conference profiles,  one
>> profile has the 'endconf' set in the 'member-flags', one profiles doesnt
>> have 'endconf' set, so user99 and user20 joins the 20 at profile1conference, and other users joins the 20 at profile2conference, that doesnt seem to work neither.   I have pasted
>> conference.conf.xml file as well for ur review.
>>
>>
>> <include>
>>   <context name="test">
>>     <extension name="global" continue="true">
>>       <condition>
>>       </condition>
>>     </extension>
>>
>>     <extension name="conf">
>>       <condition field="destination_number" expression="^(20)$" > </condition>
>>         <action application="answer"/>
>>         <action application="conference" data="bridge:20 at default+flags{endconf}:user/10"/>
>>         <action application="conference" data="20 at default"/>
>>       </condition>
>>     </extension>
>>   </context>
>> </include>
>>
>>
>>
>>
>>
>> On 6/30/2011 6:16 PM, Michael Collins wrote:
>>
>> Please put this information on pastebin and reply to the list so that we
>> can all discuss it.
>> -MC
>>
>> On Thu, Jun 30, 2011 at 1:51 PM, ran zhang <rzhang at gosilverplus.com>wrote:
>>
>>>  Mr Collins:
>>>
>>>        I attached the section in my dialplan that handles the bridging
>>> conference, when the first user (just say user 99)  dials '20', he will
>>> invite user 10 to join conference, the conference will only be established
>>> if user 10 accepts the invitation, after this, other users can join the
>>> conference by dialing '20'.
>>>
>>>        I want that if the first user (user 99 in this case)  or the user
>>> been invited (user 10 in this case)  leaves the conference, no matter how
>>> many people are still in the conference, it will close down the
>>> conference.   so i'm trying to use the 'endconf' flag, but apparently it is
>>> not valid syntax for bridging conference as I get a config error while
>>> running.  If I take out the '+flags{endconf} ', i wont get a config error
>>> while running, but then conference will only close down when there is 1
>>> person left.
>>>
>>>       I have also tried creating 2 difference conference profiles,  one
>>> profile has the 'endconf' set in the 'member-flags', one profiles doesnt
>>> have 'endconf' set, so user99 and user20 joins the 20 at profile1conference, and other users joins the 20 at profile2conference, that doesnt seem to work neither.   I have pasted
>>> conference.conf.xml file as well for ur review.
>>>
>>>
>>> <include>
>>>   <context name="test">
>>>     <extension name="global" continue="true">
>>>       <condition>
>>>       </condition>
>>>     </extension>
>>>
>>>     <extension name="conf">
>>>       <condition field="destination_number" expression="^(20)$" > </condition>
>>>         <action application="answer"/>
>>>         <action application="conference" data="bridge:20 at default+flags{endconf}:user/10"/>
>>>         <action application="conference" data="20 at default"/>
>>>       </condition>
>>>     </extension>
>>>   </context>
>>> </include>
>>>
>>>
>>> <configuration name="conference.conf" description="Audio Conference">
>>>   <!-- Advertise certain presence on startup . -->
>>>   <advertise>
>>>     <room name="3001@$${domain}" status="FreeSWITCH"/>
>>>   </advertise>
>>>
>>>   <!-- These are the default keys that map when you do not specify a caller control group -->	
>>>   <!-- Note: none and default are reserved names for group names.  Disabled if dist-dtmf member flag is set. -->	
>>>   <caller-controls>
>>>     <group name="default">
>>>       <control action="mute" digits="0"/>
>>>       <control action="deaf mute" digits="*"/>
>>>       <control action="energy up" digits="9"/>
>>>       <control action="energy equ" digits="8"/>
>>>       <control action="energy dn" digits="7"/>
>>>       <control action="vol talk up" digits="3"/>
>>>       <control action="vol talk zero" digits="2"/>
>>>       <control action="vol talk dn" digits="1"/>
>>>       <control action="vol listen up" digits="6"/>
>>>       <control action="vol listen zero" digits="5"/>
>>>       <control action="vol listen dn" digits="4"/>
>>>       <control action="hangup" digits="#"/>
>>>     </group>
>>>   </caller-controls>
>>>
>>>   <!-- Profiles are collections of settings you can reference by name. -->
>>>   <profiles>
>>>     <!--If no profile is specified it will default to "default"-->
>>>     <profile name="default">
>>>       <!-- Domain (for presence) -->
>>>       <param name="domain" value="$${domain}"/>
>>>       <!-- Sample Rate-->
>>>       <param name="rate" value="8000"/>
>>>       <!-- Number of milliseconds per frame -->
>>>       <param name="interval" value="20"/>
>>>       <!-- Energy level required for audio to be sent to the other users -->
>>>       <param name="energy-level" value="300"/>
>>>
>>>       <!--Can be | delim of waste|mute|deaf|dist-dtmf waste will always transmit data to each channel
>>>           even during silence.  dist-dtmf propagates dtmfs to all other members, but channel controls
>>> 	  via dtmf will be disabled. -->
>>>       <param name="member-flags" value="dist-dtmf"/>
>>>
>>>       <!-- Name of the caller control group to use for this profile -->
>>>       <!-- <param name="caller-controls" value="some name"/> -->
>>>       <!-- TTS Engine to use -->
>>>       <!--<param name="tts-engine" value="cepstral"/>-->
>>>       <!-- TTS Voice to use -->
>>>       <!--<param name="tts-voice" value="david"/>-->
>>>
>>>       <!-- If TTS is enabled all audio-file params beginning with -->
>>>       <!-- 'say:' will be considered text to say with TTS -->
>>>       <!-- Override the default path here, after which you use relative paths in the other sound params -->
>>>       <!-- Note: The default path is the conference's first caller's sound_prefix -->
>>>       <!--<param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/>-->
>>>       <!-- File to play to acknowledge succees -->
>>>       <!--<param name="ack-sound" value="beep.wav"/>-->
>>>       <!-- File to play to acknowledge failure -->
>>>       <!--<param name="nack-sound" value="beeperr.wav"/>-->
>>>       <!-- File to play to acknowledge muted -->
>>>       <!-- Conference pin -->
>>>       <!--<param name="pin" value="12345"/>-->
>>>       <!-- Default Caller ID Name for outbound calls -->
>>>       <param name="caller-id-name" value="$${outbound_caller_name}"/>
>>>       <!-- Default Caller ID Number for outbound calls -->
>>>       <param name="caller-id-number" value="1234"/>
>>>       <!-- Suppress start and stop talking events -->
>>>       <!-- <param name="suppress-events" value="start-talking,stop-talking"/> -->
>>>       <!-- enable comfort noise generation -->
>>>       <param name="comfort-noise" value="true"/>
>>>       <!-- Uncomment auto-record to toggle recording every conference call. -->
>>>       <!-- Another valid value is   shout://user:pass at server.com/live.mp3   -->
>>>       <!--
>>>       <param name="auto-record" value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>>>       -->
>>>     </profile>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On 6/30/2011 1:07 PM, Michael Collins wrote:
>>>
>>> Can you pastebin exactly what you are doing to establish the call?
>>> Including any relevant dialplan entries. Also, if you have modified
>>> conference.conf.xml we would like to see that also.
>>>
>>> -MC
>>>
>>>  On Thu, Jun 30, 2011 at 10:59 AM, ran zhang <rzhang at gosilverplus.com>wrote:
>>>
>>>> hi all:
>>>>
>>>> I'm trying to creating a conference, so when first member enters the
>>>> conference, he has to invite another members
>>>> and have at least 1 other member to join to have the conference
>>>> established, so i'm
>>>> using bridging conference.
>>>>
>>>> I need this conference to be terminated when the original creator of the
>>>> conference leaves no matter how many members are still left in the
>>>> conference.
>>>>
>>>> i'm trying to set 'endconf' flag in a bridging conference using
>>>> 'bridge:confname+flag{endconf}:user/10',
>>>> so it wil invite user extension 10, but its giving me config error while
>>>> running.
>>>>
>>>> can someone tell me what to do to solve this problem or get around?  the
>>>> key
>>>> is i only want the original member to be able to terminate the
>>>> conference
>>>> when he leaves, not other members assuming there are at least 2 members.
>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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
>>>
>>>
>>>
>>
>>
>> _______________________________________________
>> 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 lists at ticm.com  Fri Jul  1 07:34:32 2011
From: lists at ticm.com (Bret Watson)
Date: Fri, 01 Jul 2011 11:34:32 +0800
Subject: [Freeswitch-users] inbound trunk from Linksys 3102?
Message-ID: <4e0d4052.869d2a0a.4a79.ffffbc07@mx.google.com>

Hi All,
one of the few things that really worked on my * setup was the 
inbound trunk from the 3102...

* config was
trunk name 1-pstn
disallow=all
allow=ulaw
canreinvite=no
context=from-trunk
dtmfmode=rfc2833
host=dynamic
incominglimit=1
nat=never
port=5061
qualify=yes
secret=xxxxxxxx
type=friend
username=1-pstn

How do I translate this into freeswitch?

Thanks!

Bret




From max.asterisk at gmail.com  Fri Jul  1 08:12:15 2011
From: max.asterisk at gmail.com (Max Alex)
Date: Fri, 1 Jul 2011 09:42:15 +0530
Subject: [Freeswitch-users] Answer issue on inbound call
In-Reply-To: <BANLkTin5zjPmdaV6Qa9ZM8dYtBGBUmcKOg@mail.gmail.com>
References: <BANLkTin+f=uDMiOouyLWAorPvNTd1Z6xaA@mail.gmail.com>
	<BANLkTim=TOQD1tUvmHjeRuhAR-eKkGYgsw@mail.gmail.com>
	<BANLkTim3pTPV804t_Usk59eG2C56HAyFdA@mail.gmail.com>
	<BANLkTi=Q3qRBCDRARDWp15=Guwyq-jtNLQ@mail.gmail.com>
	<BANLkTinMokGeJU+wVVA0U-oxB97z+c+ZGg@mail.gmail.com>
	<BANLkTinEp1Cu5mB_UHw-Y+mitGsNByR3sQ@mail.gmail.com>
	<BANLkTikcWDbCkTPRow-TMqPN=D3FV+LjWw@mail.gmail.com>
	<BANLkTinE1292_p_VZd+D4N6vvEEot8LXRw@mail.gmail.com>
	<BANLkTin5zjPmdaV6Qa9ZM8dYtBGBUmcKOg@mail.gmail.com>
Message-ID: <BANLkTim5jOswqjM9DwpH99UpjCm5MmVXjw@mail.gmail.com>

Hi,
Can any one provide their suggestions and help for this issue,
I really need to resolve and get this working.

Thanks,
Max Alex
Voip Developer



On Tue, Jun 28, 2011 at 6:01 PM, Max Alex <max.asterisk at gmail.com> wrote:

> Hi,
> Thanks for your reply.
> I have enabled logger as per your help.
> I have given completed log on following link of pastebin.
> http://pastebin.freeswitch.org/16616
>
> You can see this line as it is showing answered and this call is answered
> on my cell phone but on softphone it is ringing and i have rejected from
> there.
> 2011-06-28 17:47:52.909657 [NOTICE] mod_freetdm.c:1953 Channel [FreeTDM/2:
> 1/01234567890] has been answered
>
> It is pre answering the cell phone when it is ringing on phone
> Waiting for your help.
>
> Thanks,
> Max Alex
> Voip Developer
>
>
>
> On Mon, Jun 27, 2011 at 9:15 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Get a complete, unedited, unabridged console debug log w/ siptrace and put
>> it on pastebin w/ "FreeSWITCH Log" for the syntax highlighting. Use "sofia
>> global siptrace on" to make sure you can all SIP traffic.
>>
>> -MC
>>
>>
>> On Mon, Jun 27, 2011 at 6:05 AM, Max Alex <max.asterisk at gmail.com> wrote:
>>
>>> Hi,
>>> Thanks for reply,
>>> I have tried the same way and reloaded freeswitch, but still it is
>>> answered on first ring of the call.
>>> When it is ringing the call on 1001 and the same time it is answered on
>>> cell phone, so something is done when it is ringing on 1001.
>>>
>>> Here is the dialplan for the same
>>>
>>>     <extension name="Local_Extension">
>>>       <condition field="destination_number" expression="^(10[01][0-9])$">
>>>         <action application="set" data="dialed_extension=$1"/>
>>>         <action application="export" data="dialed_extension=$1"/>
>>>         <action application="bind_meta_app" data="2 b s
>>> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>>>         <action application="bind_meta_app" data="3 b s
>>> execute_extension::cf XML features"/>-->
>>>         <action application="set" data="ringback=${us-ring}"/>
>>>         <action application="set"
>>> data="transfer_ringback=$${hold_music}"/>
>>>         <action application="set" data="call_timeout=30"/>
>>>           <action application="set" data="hangup_after_bridge=true"/>
>>>          <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="set"
>>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>>> var callgroup)}"/>
>>>         <action application="hash"
>>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>>>
>>>        <action application="bridge"
>>> data="{ignore_early_media=true}user/${dialed_extension}@
>>> ${domain_name}"/>
>>>         <action application="sleep" data="1000"/>
>>>         <action application="voicemail" data="default ${domain_name}
>>> ${dialed_extension}"/>
>>>       </condition>
>>>     </extension>
>>>
>>>
>>> Please help me for this issue.
>>>
>>>
>>> Thanks,
>>> Max Alex
>>> Voip Developer
>>>
>>>
>>>
>>> On Fri, Jun 24, 2011 at 11:38 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Are you using the default dialplan? I think you might just need to
>>>> ignore early media on your bridge app. If you are using the default.xml file
>>>> then locate "Local_Extension" and the bridge line:
>>>>
>>>>     <action application="bridge" data="user/${dialed_extension}@
>>>> ${domain_name}"/>
>>>>
>>>> Change it to this, then try again:
>>>>
>>>>     <action application="bridge"
>>>> data="{ignore_early_media=true}user/${dialed_extension}@
>>>> ${domain_name}"/>
>>>>
>>>> If I understand correctly, the "symptom" you are experiencing is the
>>>> normal operation of the bridge app (and it's cousin, the originate API
>>>> command). When the b-leg sends back media, including ringing, the bridge (or
>>>> the originate) will be considered "successful," and in the case of bridge,
>>>> the b-leg's audio (the early media) will be connected to the a-leg. If you
>>>> set ignore_early_media=true then you are explicitly telling the bridge app
>>>> that you only want to connect the b-leg to the a-leg if the b-leg actually
>>>> answers.
>>>>
>>>> I hope that made sense...
>>>>
>>>> -MC
>>>>
>>>>
>>>>
>>>> On Thu, Jun 23, 2011 at 9:32 PM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>
>>>>> Hi,
>>>>> Thanks for reply.
>>>>> Current scenario is below.
>>>>>
>>>>> PSTN call -> sangoma device -> freeswitch incoming context -> routed to
>>>>> 1001 -> ringing (Answered on cellphone)
>>>>> Here when it is routed to 1001 the call it is started ringing, but on
>>>>> phone that call is answered and hearding sound of ringing.
>>>>>
>>>>> Required flow:
>>>>> PSTN call -> sangoma device -> freeswitch incoming context -> routed to
>>>>> 1001 -> ringing (Ringing on cellphone)
>>>>>
>>>>> I hope it is understable, the call should not be answered until 1001
>>>>> answer it, right not when it is started ring it is answered on cell phone.
>>>>> that should not be happen as it is not answered yet.
>>>>>
>>>>> Waiting for your reply.
>>>>>
>>>>>
>>>>> Thanks,
>>>>> Max Alex
>>>>> Voip Developer
>>>>>
>>>>>
>>>>>
>>>>> On Fri, Jun 24, 2011 at 5:48 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> I'm not sure I understand the problem. What is happening vs. what you
>>>>>> believe should be happening?
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>> On Thu, Jun 23, 2011 at 3:31 AM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>> Thanks for your reply.
>>>>>>> Here is my configuration and log
>>>>>>> http://pastebin.freeswitch.org/16571
>>>>>>>
>>>>>>> I am using A200 analog sangoma device with freeswitch, it is working
>>>>>>> fine but when it is routing call to 1001 then it is answered.
>>>>>>> Please provider your suggestions.
>>>>>>>
>>>>>>> Thanks,
>>>>>>> Max Alex
>>>>>>> Voip Developer
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Jun 22, 2011 at 8:51 PM, Michael Collins <msc at freeswitch.org
>>>>>>> > wrote:
>>>>>>>
>>>>>>>> I thought the A200 was an analog card? Maybe I have my numbers mixed
>>>>>>>> up...
>>>>>>>>
>>>>>>>> Go ahead and collect a debug log of this call. It might help to have
>>>>>>>> your configs posted as well. Use pastebin.freeswitch.org. See this
>>>>>>>> wiki article for tips on how to collect information:
>>>>>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>>>>>>
>>>>>>>> -MC
>>>>>>>>
>>>>>>>> On Wed, Jun 22, 2011 at 3:23 AM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>>>>>
>>>>>>>>> Hi,
>>>>>>>>> I have installed freeswitch latest version with sangoma card A200
>>>>>>>>> as well,
>>>>>>>>> I have installed and configured freetdm module with wanpipe drivers
>>>>>>>>> for freeswitch,
>>>>>>>>> We are properly receiving the inbound calls in public context and
>>>>>>>>> then we are routing that call to 1001 extension,
>>>>>>>>> it is properly routing to 1001 as well, but we have one issue while
>>>>>>>>> routing on 1001.
>>>>>>>>>
>>>>>>>>> Here is the issue description.
>>>>>>>>> I am calling from my cell phone to that DID number of pri line, and
>>>>>>>>> then it will start ringing on 1001 extension,
>>>>>>>>> When 1001 extension start ringing the call is answered on my cell
>>>>>>>>> phone,
>>>>>>>>> it is something like freeswitch preanswer or autoanswer the call,
>>>>>>>>> how can i stop this answer call when it is ringing on 1001 extension,
>>>>>>>>> Waiting for good reply.
>>>>>>>>>
>>>>>>>>> Thanks,
>>>>>>>>> Max Alex
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> 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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> 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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> 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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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 ankitwalia4u at gmail.com  Fri Jul  1 08:26:43 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Fri, 1 Jul 2011 09:56:43 +0530
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <SNT126-W63C9737241DBFF43ACCCCEC6580@phx.gbl>
References: <SNT126-W63C9737241DBFF43ACCCCEC6580@phx.gbl>
Message-ID: <BANLkTinRocHS0-QQbU55dqvBM+0ro_5XJQ@mail.gmail.com>

Hi Sharad,

Appreciating your efforts, I have few questions.


   1. Can I schedule the reminders at specific time?
   2. The reminder calls will be IVR and these IVRs Menu will have to be
   configured in FreeSwitch XML or any GUI Fusionpbx, Freepbx or scripts.
   3. As per the configuration you have used, if we need to send several
   reminder calls at let us say on 100 mobile numbers, how is the performance?
   4. Last but not the least, under what license, your application is
   available?

Thanks
Ankit

On Thu, Jun 30, 2011 at 10:05 AM, sharad garg <sharad2710 at hotmail.com>wrote:

>  Hi to all,
>
> This is just let to know this forum that we have completed the work on
> Alarm Service. All this is done using our great Freeswitch only.
>
> The GUI also is designed for the same.
>
> You can have a view at the attached PPT for the details.
>
> Thanks to Freeswitch developers & community off course who helped us in
> this project directly or indirectly.
>
> Regards
> Sharad
>
>
>
> _______________________________________________
> 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
>
>


-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From bryansmart at bryansmart.com  Fri Jul  1 09:13:20 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Fri, 1 Jul 2011 01:13:20 -0400
Subject: [Freeswitch-users] Playing multiple files simultaneously
Message-ID: <2DB72AF2-2394-419A-ADC5-0B76FE594662@bryansmart.com>

Is it possible for Freeswitch to play more than one file to a channel at a time? What I've seen and tried from the dialplan and scripts either queues files to play, or will stop a currently playing file so that the newly requested file will play. This also seems to be the case in conferences. When I send multiple play commands to conferences, the files are queued.

As for how this might be used, think of an IVR that plays queued prompts, yet continuously plays looping music or a Shoutcast stream in the background. I also want to be able to play short cue tones that start at the same time as a prompt (don't want to pre-mix them in to a single file, though).

Is this currently possible through any means? Perhaps with the event socket?

Bryan



From david.ponzone at ipeva.fr  Fri Jul  1 09:39:14 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 1 Jul 2011 07:39:14 +0200
Subject: [Freeswitch-users] t38-passthru sofia param
In-Reply-To: <A9E1BFB8-8BE0-4ED1-86C9-E84A3ACC4AD2@5ninesolutions.com>
References: <724715C2-4970-4594-9346-167EA8B89C0F@5ninesolutions.com>
	<A9E1BFB8-8BE0-4ED1-86C9-E84A3ACC4AD2@5ninesolutions.com>
Message-ID: <EF2625C5-9EB6-41E7-8944-4A6021B072FC@ipeva.fr>

You need to enable t38-passthru when you want FS to forward the T38 reinvite from a leg to anothern.

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

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

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




Le 30/06/2011 ? 21:25, Spencer Thomason a ?crit :

> Just a quick follow up.  The fact that the reINVITE wasn't being passed correctly across the bridge was pilot error.. I had enable-soa=false on the profile.  I am however a little confused as when to use the t38-passthru parameter.    Is it needed even when doing bypass media or proxy media?
> 
> On Jun 30, 2011, at 11:40 AM, Spencer Thomason wrote:
> 
>> Hello all,
>> I have a few questions regarding the t38-passthru sofia param.  One of the ways we are using FreeSWITCH is as a simple B2BUA in our SBCs.  The topology looks like this:
>> 
>>                                            Freeswitch
>>                                               /\          \/
>> FS and Asterisk boxen -> OpenSIPS -> PSTN
>> 
>> The Freeswitch instance is setup with inbound late negotiation and bypass media = true and the dialplan consists of a very simple bridge statement that sends a call back to the proxy.  
>> 
>> The problem is that when someone sends a fax and the GW send a t.38 reINVITE, Freeswitch is sending the original SDP (PCMU in this case) back to Leg A and the fax then fails using PCMU.  Is this parameter needed when bypass media is already enabled?
>> 
>> Thanks,
>> Spencer
>> 
>> 
>> 
>> _______________________________________________
>> 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
>> 
> 
> 
> _______________________________________________
> 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 david.ponzone at ipeva.fr  Fri Jul  1 09:42:25 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 1 Jul 2011 07:42:25 +0200
Subject: [Freeswitch-users] mod_limit
In-Reply-To: <443EC226AAEABB48B58CAF9D56D80AB41604F0@hektor.dom.local>
References: <443EC226AAEABB48B58CAF9D56D80AB41604F0@hektor.dom.local>
Message-ID: <1D16ED30-9647-4A60-BEAA-20AEACC4B3B6@ipeva.fr>

When you wonder if something would work, just try it (with lower values).
real and id are arbitrary parameters.
Think of them as the 2 coordinates of a dual-entry table.

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

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

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




Le 30/06/2011 ? 22:51, Tomasz Kopacki a ?crit :

>   Hi,
> Im trying to set mod_limit but i dont understand what are realm and id parameters for.
> I have 2 users groups - Group A and Group B. Group A has two domains and Group B has three domains.
> I want to set limit 100cc on group A and 150cc on group B.
> I did this:
> <action application="limit" data="hash realm groupA 100 />
> <action application="limit" data="hash realm groupB 150 />
> Will it works ?
> _______________________________________________
> 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 jan.berger at video24.no  Fri Jul  1 10:49:23 2011
From: jan.berger at video24.no (Jan Berger)
Date: Fri, 1 Jul 2011 08:49:23 +0200
Subject: [Freeswitch-users] Playing multiple files simultaneously
In-Reply-To: <2DB72AF2-2394-419A-ADC5-0B76FE594662@bryansmart.com>
References: <2DB72AF2-2394-419A-ADC5-0B76FE594662@bryansmart.com>
Message-ID: <EB5BFD1EA1F444079A8BFBAC95976FE1@dell9400>

If you play several files FS will queue them if it is the same stream - you
need to somehow set up a conference with multiple streams (callers).

I would check options around music on hold.

But, another way is to use FS to play several outgoing SIP streams and loop
them back into a conference.


Jan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bryan
Smart
Sent: 1. juli 2011 07:13
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Playing multiple files simultaneously

Is it possible for Freeswitch to play more than one file to a channel at a
time? What I've seen and tried from the dialplan and scripts either queues
files to play, or will stop a currently playing file so that the newly
requested file will play. This also seems to be the case in conferences.
When I send multiple play commands to conferences, the files are queued.

As for how this might be used, think of an IVR that plays queued prompts,
yet continuously plays looping music or a Shoutcast stream in the
background. I also want to be able to play short cue tones that start at the
same time as a prompt (don't want to pre-mix them in to a single file,
though).

Is this currently possible through any means? Perhaps with the event socket?

Bryan


_______________________________________________
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 mbrancaleoni at voismart.it  Fri Jul  1 16:10:48 2011
From: mbrancaleoni at voismart.it (Matteo)
Date: Fri, 01 Jul 2011 14:10:48 +0200 (CEST)
Subject: [Freeswitch-users] Generating ringback if 180 arrives after 183
In-Reply-To: <0503e1d1-37d5-4087-b198-880cf316702e@mx.voismart.com>
Message-ID: <ec66aa6d-1675-40c1-904f-a44e3c51cd7f@mx.voismart.com>

Hi,

I've made up an external perl client for ESL which
monitors for events and ask to generate ringback
on caller channel if certain conditions happens.

I had to insert two new custom events in sofia.c 
to trigger this situation.

Can be usefule for general use?
If yes, I'll be glad to submit the patch to jira.

regards,
matteo.


From rajesh.npnr at yahoo.com  Fri Jul  1 16:41:16 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Fri, 1 Jul 2011 05:41:16 -0700 (PDT)
Subject: [Freeswitch-users] Inband DTMF set in Event Socket
Message-ID: <1309524076282-6537620.post@n2.nabble.com>

Hello,

How to set the DTMF type as inband when I dial through event socket using
api originate command?

Thanks,
Rex

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Inband-DTMF-set-in-Event-Socket-tp6537620p6537620.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From avi at avimarcus.net  Fri Jul  1 16:44:31 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 1 Jul 2011 15:44:31 +0300
Subject: [Freeswitch-users] Inband DTMF set in Event Socket
In-Reply-To: <1309524076282-6537620.post@n2.nabble.com>
References: <1309524076282-6537620.post@n2.nabble.com>
Message-ID: <BANLkTinHT_o-_ytPr7gjL8hAoSzjzh3-7A@mail.gmail.com>

I think you want to turn on this in the originate string?
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate

-Avi


On Fri, Jul 1, 2011 at 3:41 PM, rex.alex <rajesh.npnr at yahoo.com> wrote:
> Hello,
>
> How to set the DTMF type as inband when I dial through event socket using
> api originate command?
>
> Thanks,
> Rex
>
> --
> View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Inband-DTMF-set-in-Event-Socket-tp6537620p6537620.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> 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 netcentrica at gmail.com  Fri Jul  1 16:55:49 2011
From: netcentrica at gmail.com (Mateusz Bartczak)
Date: Fri, 1 Jul 2011 14:55:49 +0200
Subject: [Freeswitch-users] Freeswitch dynamic routing of all calls
In-Reply-To: <518E6CA8-4198-4484-BF03-2DEFFA6493EA@gmail.com>
References: <BANLkTimuCjRcVccyrPpgQeY6N+UamyNQ=A@mail.gmail.com>
	<518E6CA8-4198-4484-BF03-2DEFFA6493EA@gmail.com>
Message-ID: <BANLkTi=UJ0+GKwd3T2uom=MuS4aLqudDmA@mail.gmail.com>

OK but how can I respond for received events?

I subscribe to receive events using following event socket command:

event plain CHANNEL_CREATE

then I got all events of that type and that's great but the question is how
can I handle those events, what's the syntax and where to put it?

I will simply explain what I'm trying to achieve:

1. User dialed number
2. CHANNEL_CREATE event is created
3. I got this event using socket
4. What to do now? How to respond for that event? For example I would like
to respond with dialstring to use, user called number 123, I would like to
return something like sofia/gateway123/00123



2011/6/30 Steven Ayre <steveayre at gmail.com>

> Yes it's possible and I suggest you look at esl
>
> Steve on iPhone
>
> On 30 Jun 2011, at 09:54, Mateusz Bartczak <netcentrica at gmail.com> wrote:
>
> > Hi all
> >
> > I'm new to FS and I would like to know is it possible to implement
> following scenario:
> >
> > 1. User dials number
> > 2. Routing script detects outgoing call event. Every call needs to be
> handled by routing script.
> > 3. Routing script takes in input: user name, domain, dialed number. Than
> it query database to find best SIP trunk to route the call, it also checks
> destination price per minute and calculates maximum call time for prepaid
> user.
> > 4. Routing script output is: SIP trunk to use, SIP call parameters (ie.
> callerid), maximum call duration
> > 5. FS read output from routing script and make call using returned
> parameters
> >
> > Preferred routing implementation technology: background running unix
> deamon written in Java or PHP. Connection with FS via socket.
> >
> > Event routing script will be multi-threaded, must be able to deal with a
> lot of calls in parallel and processing of one call should not block
> processing of other calls (I have this problem with Yate voip server, and
> that's really big problem)
> >
> > Is it possible to do this using FS?
> > Any advices where to search for additional info? I know that there is
> event handler but can it return "dialstring" for outgoing call events?
> > Some code examples?
> >
> > I will really appreciate your help
> >
> >
> >
> >
> > _______________________________________________
> > 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
>
> _______________________________________________
> 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 ankitwalia4u at gmail.com  Fri Jul  1 18:18:09 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Fri, 1 Jul 2011 19:48:09 +0530
Subject: [Freeswitch-users] Could not connect thru GSMOpen
In-Reply-To: <BANLkTimkhpspbUu+JiqNTrWtSZaRvRdWJg@mail.gmail.com>
References: <BANLkTimk9mc9M66kE0_h5bF5bKf2S0NReg@mail.gmail.com>
	<BANLkTimkhpspbUu+JiqNTrWtSZaRvRdWJg@mail.gmail.com>
Message-ID: <BANLkTim3EzVvkPq5Q_zk-JaOXOcFJ7RiTQ@mail.gmail.com>

Ok, Could you please guide me how can I uninstall the alsa on my Ubuntu
system.

I tried installing on other system having Centos (this system is fresh
freeswitch install).  I followed the steps

cd gsmlib/gsmlib-1.10-patched-12ubuntu1
./configure
make
make install

Now, I was confused at this below step. What does the below lines mean on wiki.
*Without audio support (SMSs only, no voice calls)*


   - * With gsmlib is in noaudio_gsmlib_cplusplus/mod_gsmopen *
   - * Without gsmlib is in noaudio_nogsmlib_nocplusplus/mod_gsmopen *

Does it mean that I need to add one more layer of directory under
endpoints/noaudio_gsmlib_cplusplus/mod_gsmopen ?

Although I was not sure, Still I did install under normal mod_gsmopen. Now
when I am trying to load mod_gsmopen. It is giving me the error
2011-07-01 09:59:53.689233 [CRIT] switch_loadable_module.c:928 Error
Loading module /usr/local/freeswitch/mod/mod_gsmopen.so

Though the file mod_gsmopen.so exist and I logged into root which has all
the permission.

Please help.

Regards,
Ankit



On Thu, Jun 30, 2011 at 7:34 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> the "plughw:1" alsa device does not exists (or is not read-write to
> the FS user).
>
> Are you using a soundcard connected to the cellphone? To be audio
> enabled, gsmopen need a soundcard connected to the cellphone (as
> explained in the wiki page).
>
> If you're not interested in audio, but only in SMS, then compile
> gsmopen without audio, so it will not check for the soundcard.
>
> Also, I don't know if gsmopen will work with the kind of cellphone
> you're using. For sure it works with mobigater embedded devices, with
> motorolas, with ericssons.
>
> -giovanni
> -giovanni
>
> On Thu, Jun 30, 2011 at 3:37 PM, ankIT WALiA <ankitwalia4u at gmail.com>
> wrote:
> > Hi all,
> >
> > I have properly compiled the gsmopen module.
> >
> > Added the Dialplan
> >
> >     <extension name="gsmopen">
> >       <condition field="destination_number" expression="^1002$">
> >         <action application="bridge"
> data="gsmopen/interface1/xxxxxxxxx"/>
> >       </condition>
> >     </extension>
> > ----------------------------------------------------
> > And the gsm conf file
> > <configuration name="gsmopen.conf" description="GSMopen Configuration">
> > -
> > <global_settings>
> >
> > <param name="debug" value="8"/>
> > <param name="dialplan" value="XML"/>
> > <param name="context" value="default"/>
> > <param name="hold-music" value="$${moh_uri}"/>
> >
> > <param name="destination" value="9999"/>
> > </global_settings>
> > <!-- one entry here per gsmopen interface -->
> > -
> > <per_interface_settings>
> > -
> > <interface id="1" name="interface1">
> >
> > <param name="hold-music" value="$${moh_uri}"/>
> > <param name="dialplan" value="XML"/>
> > <param name="context" value="default"/>
> > <param name="destination" value="5000"/>
> >
> > <param name="alsacname" value="plughw:1"/>
> > <param name="alsapname" value="plughw:1"/>
> > <param name="controldevice_name" value="/dev/ttyACM1"/>
> >
> > </interface>
> > <!-- following interfaces are commented out!
> >     -->
> > </per_interface_settings>
> > </configuration>
> > ------------------------------------------------------------------
> > I am using a Nokia handset in India
> >
> > ---------------------------------------------------------------------
> > I am getting the below error on gsm_load.
> >
> > gsm reload
> > 2011-06-30 19:03:52.716491 [WARNING] mod_gsmopen.cpp:1860 rev
> > exported[(nil)|37     ][WARNINGA  1860 ][interface1][-1, 0, 0] STARTING
> > interface_id=1
> >
> > 2011-06-30 19:03:56.616460 [ERR] gsmopen_protocol.cpp:2999 rev
> > exported[(nil)|37     ][ERRORA  2999 ][interface1][-1, 0, 0] snd_pcm_open
> > failed with error 'No such file or directory' on device 'plughw:1', if
> you
> > are using a plughw:n device please change it to be a default:n device (so
> to
> > allow it to be shared with other concurrent programs), or maybe you are
> > using an ALSA voicemodem and slmodemd is running?
> >
> > 2011-06-30 19:03:56.616460 [ERR] gsmopen_protocol.cpp:2891 rev
> > exported[(nil)|37     ][ERRORA  2891 ][interface1][-1, 0, 0] Failed
> opening
> > ALSA capture device: plughw:1
> > 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:1931 rev
> exported[(nil)|37
> > ][ERRORA  1931 ][interface1][-1, 0, 0] alsa_init failed
> >
> > 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:1932 rev
> exported[(nil)|37
> > ][ERRORA  1932 ][interface1][-1, 0, 0] STARTING interface_id=1 FAILED
> > 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:3146 rev
> exported[(nil)|37
> > ][ERRORA  3146 ][interface1][-1, 0, 0] ALARM on interface interface1:
> >
> >
> >
> -----------------------------------------------------------------------------------------------------------------
> > I could not understand what does the error message trying to say? I am
> > connected to friend's mobile with datacable on com1 port. I jsut wanted
> to
> > check SMS. Please help.
> >
> > Thanks
> > Ankit
> >
> > --
> > Life is like a rose its upto u feel it as its fragrance or thorns
> >
> > _______________________________________________
> > 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
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> 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
>



-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From steveu at coppice.org  Fri Jul  1 18:21:24 2011
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 01 Jul 2011 22:21:24 +0800
Subject: [Freeswitch-users] change sounds file format of voicemail
In-Reply-To: <EAC9775B572D499CA7869A2FA228C517@stor1>
References: <89F3619D8E6F437F82F3B1C00FA69725@e1705>
	<EAC9775B572D499CA7869A2FA228C517@stor1>
Message-ID: <4E0DD7E4.9050102@coppice.org>

On 06/30/2011 03:05 AM, Kris wrote:
> I am doing testing and I am about to submit changes to the GIT to allow
> 8000x8 bit wav and 8000x4bit vox recording. What formats would you like to
8000x8 bit is a really stupid choice. A-law or u-law would do a far 
better job, despite being lossy codecs. The dynamic range of 8 bit line 
audio just isn't enough for speech. 8000x4bit vox made sense in the 90s, 
because Dialogic used it, and they were big in IVRs. It seems a very 
obsolete choice these days.

Steve



From rajesh.npnr at yahoo.com  Fri Jul  1 18:46:19 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Fri, 1 Jul 2011 07:46:19 -0700 (PDT)
Subject: [Freeswitch-users] Inband DTMF set in Event Socket
In-Reply-To: <BANLkTinHT_o-_ytPr7gjL8hAoSzjzh3-7A@mail.gmail.com>
References: <1309524076282-6537620.post@n2.nabble.com>
	<BANLkTinHT_o-_ytPr7gjL8hAoSzjzh3-7A@mail.gmail.com>
Message-ID: <1309531579711-6538036.post@n2.nabble.com>

I am able to see "<action application="start_dtmf_generate" data="true"/>"
which is to enable inband DTMF when I dial through dialplan. But how to
enable the same when I dial through api commands like

api originate {origination_caller_id_name='XXX'}user/1000
&bridge({accountcode='XXX'}sofia/gateway/gw_name/phno)

How to prefix start_dtmf_generate=true in originate command?

Thanks,
Rex

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Inband-DTMF-set-in-Event-Socket-tp6537620p6538036.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From avi at avimarcus.net  Fri Jul  1 18:51:44 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 1 Jul 2011 17:51:44 +0300
Subject: [Freeswitch-users] Inband DTMF set in Event Socket
In-Reply-To: <1309531579711-6538036.post@n2.nabble.com>
References: <1309524076282-6537620.post@n2.nabble.com>
	<BANLkTinHT_o-_ytPr7gjL8hAoSzjzh3-7A@mail.gmail.com>
	<1309531579711-6538036.post@n2.nabble.com>
Message-ID: <BANLkTinmbb5dwN64EjFe+vP50nxR5-C5rw@mail.gmail.com>

... the same way you set the origination_caller_id_name or the accountcode -
they are both setting channel variables.
-Avi



On Fri, Jul 1, 2011 at 5:46 PM, rex.alex <rajesh.npnr at yahoo.com> wrote:

> I am able to see "<action application="start_dtmf_generate" data="true"/>"
> which is to enable inband DTMF when I dial through dialplan. But how to
> enable the same when I dial through api commands like
>
> api originate {origination_caller_id_name='XXX'}user/1000
> &bridge({accountcode='XXX'}sofia/gateway/gw_name/phno)
>
> How to prefix start_dtmf_generate=true in originate command?
>
> Thanks,
> Rex
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Inband-DTMF-set-in-Event-Socket-tp6537620p6538036.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> 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 daniel at danielknoll.de  Fri Jul  1 11:02:05 2011
From: daniel at danielknoll.de (Daniel Knoll)
Date: Fri, 1 Jul 2011 09:02:05 +0200
Subject: [Freeswitch-users] curl.run and curl parameter
Message-ID: <BANLkTimgBJHhSw=2xmQWwdqyFoUWXeAc5Q@mail.gmail.com>

Hey Freeswitch and Javascript Guys,

i would like to use curl.run to post a file to googles speech api. How
can i add curl parameters like "-F filename" to pass the filename.

curl.run("POST", "https://www.google.com/speech-api/v1/",
"recognize?xjerr=1&client=chromium&lang=de-DE&lm=builtin:dictation",
my_callback, "my callback_arg\n",, Content-Type: audio/x-flac;
rate=16000);


Has anyone an idea ?

Thanx a lot for answers.
Daniel


From michal.zubac at comgate.cz  Fri Jul  1 16:08:09 2011
From: michal.zubac at comgate.cz (=?UTF-8?B?TWljaGFsIFp1YsOhxI0=?=)
Date: Fri, 01 Jul 2011 14:08:09 +0200
Subject: [Freeswitch-users] How to send Outbound Caller ID Name out
 Sangoma A101 PRI
In-Reply-To: <E826C94A3C6BE44198FAF4D66B5AB4D81BDA7E5903@VA3DIAXVS351.RED001.local>
References: <E826C94A3C6BE44198FAF4D66B5AB4D81BDA7E5903@VA3DIAXVS351.RED001.local>
Message-ID: <4E0DB8A9.5050803@comgate.cz>

Hi.

Are you using ftmod_sangoma_isdn? If so, I'm filing bugreport to sangoma 
developers, because there's bug in ftmod_sangoma_isdn. I have the same 
problem. Library isn't filling caller id fields of ISDN SETUP message 
properly in outbound calls.

Also try to setup correctly these values in "freetdm.conf.xml":
<param name="outbound-calling-ton" value="international" />
<param name="outbound-calling-npi" value="isdn" />

Michal Zubac

On 30.6.2011 19:44, Robert Hadley wrote:
>
> Hi FS Users,
>
> Is there a way to send the outbound caller ID name when using FreeTDM, 
> wanpipe, and a Sangoma A101 PRI?
>
> I am setting the effective_caller_id_name and _number and the number 
> goes out but not the name.
>
> I also tried setting the origination_caller_id_name on the bridge but 
> that didn't work either.
>
> I receive the incoming caller_id_name in a separate Facility message.
>
> Thanks,
>
> Robert
>
>
> _______________________________________________
> 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 gmaruzz at celliax.org  Fri Jul  1 19:18:42 2011
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 1 Jul 2011 17:18:42 +0200
Subject: [Freeswitch-users] Could not connect thru GSMOpen
In-Reply-To: <BANLkTim3EzVvkPq5Q_zk-JaOXOcFJ7RiTQ@mail.gmail.com>
References: <BANLkTimk9mc9M66kE0_h5bF5bKf2S0NReg@mail.gmail.com>
	<BANLkTimkhpspbUu+JiqNTrWtSZaRvRdWJg@mail.gmail.com>
	<BANLkTim3EzVvkPq5Q_zk-JaOXOcFJ7RiTQ@mail.gmail.com>
Message-ID: <BANLkTik+pjc_JzaSWde2eA-d-RSC5dSXdg@mail.gmail.com>

Ankit,

the error is just that you have not installed the module.

For what I can see, you are just beginning to use FreeSWIT>CH, and
mod_gsmopen is an experimental module that requires a fair knowledge of
FreeSWITCH and Linux to be used.

I would counseil you to not use it, and go back to it in the future.

Also, I highly doubt it will work with your nokia.

-giovanni

On Fri, Jul 1, 2011 at 4:18 PM, ankIT WALiA <ankitwalia4u at gmail.com> wrote:

> Ok, Could you please guide me how can I uninstall the alsa on my Ubuntu
> system.
>
> I tried installing on other system having Centos (this system is fresh
> freeswitch install).  I followed the steps
>
> cd gsmlib/gsmlib-1.10-patched-12ubuntu1
> ./configure
> make
> make install
>
> Now, I was confused at this below step. What does the below lines mean on wiki.
> *Without audio support (SMSs only, no voice calls)*
>
>
>    - * With gsmlib is in noaudio_gsmlib_cplusplus/mod_gsmopen *
>    - * Without gsmlib is in noaudio_nogsmlib_nocplusplus/mod_gsmopen *
>
> Does it mean that I need to add one more layer of directory under
> endpoints/noaudio_gsmlib_cplusplus/mod_gsmopen ?
>
> Although I was not sure, Still I did install under normal mod_gsmopen. Now
> when I am trying to load mod_gsmopen. It is giving me the error
> 2011-07-01 09:59:53.689233 [CRIT] switch_loadable_module.c:928 Error
> Loading module /usr/local/freeswitch/mod/mod_gsmopen.so
>
> Though the file mod_gsmopen.so exist and I logged into root which has all
> the permission.
>
> Please help.
>
> Regards,
>  Ankit
>
>
>
>
> On Thu, Jun 30, 2011 at 7:34 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>
>> the "plughw:1" alsa device does not exists (or is not read-write to
>> the FS user).
>>
>> Are you using a soundcard connected to the cellphone? To be audio
>> enabled, gsmopen need a soundcard connected to the cellphone (as
>> explained in the wiki page).
>>
>> If you're not interested in audio, but only in SMS, then compile
>> gsmopen without audio, so it will not check for the soundcard.
>>
>> Also, I don't know if gsmopen will work with the kind of cellphone
>> you're using. For sure it works with mobigater embedded devices, with
>> motorolas, with ericssons.
>>
>> -giovanni
>> -giovanni
>>
>> On Thu, Jun 30, 2011 at 3:37 PM, ankIT WALiA <ankitwalia4u at gmail.com>
>> wrote:
>> > Hi all,
>> >
>> > I have properly compiled the gsmopen module.
>> >
>> > Added the Dialplan
>> >
>> >     <extension name="gsmopen">
>> >       <condition field="destination_number" expression="^1002$">
>> >         <action application="bridge"
>> data="gsmopen/interface1/xxxxxxxxx"/>
>> >       </condition>
>> >     </extension>
>> > ----------------------------------------------------
>> > And the gsm conf file
>> > <configuration name="gsmopen.conf" description="GSMopen Configuration">
>> > -
>> > <global_settings>
>> >
>> > <param name="debug" value="8"/>
>> > <param name="dialplan" value="XML"/>
>> > <param name="context" value="default"/>
>> > <param name="hold-music" value="$${moh_uri}"/>
>> >
>> > <param name="destination" value="9999"/>
>> > </global_settings>
>> > <!-- one entry here per gsmopen interface -->
>> > -
>> > <per_interface_settings>
>> > -
>> > <interface id="1" name="interface1">
>> >
>> > <param name="hold-music" value="$${moh_uri}"/>
>> > <param name="dialplan" value="XML"/>
>> > <param name="context" value="default"/>
>> > <param name="destination" value="5000"/>
>> >
>> > <param name="alsacname" value="plughw:1"/>
>> > <param name="alsapname" value="plughw:1"/>
>> > <param name="controldevice_name" value="/dev/ttyACM1"/>
>> >
>> > </interface>
>> > <!-- following interfaces are commented out!
>> >     -->
>> > </per_interface_settings>
>> > </configuration>
>> > ------------------------------------------------------------------
>> > I am using a Nokia handset in India
>> >
>> > ---------------------------------------------------------------------
>> > I am getting the below error on gsm_load.
>> >
>> > gsm reload
>> > 2011-06-30 19:03:52.716491 [WARNING] mod_gsmopen.cpp:1860 rev
>> > exported[(nil)|37     ][WARNINGA  1860 ][interface1][-1, 0, 0] STARTING
>> > interface_id=1
>> >
>> > 2011-06-30 19:03:56.616460 [ERR] gsmopen_protocol.cpp:2999 rev
>> > exported[(nil)|37     ][ERRORA  2999 ][interface1][-1, 0, 0]
>> snd_pcm_open
>> > failed with error 'No such file or directory' on device 'plughw:1', if
>> you
>> > are using a plughw:n device please change it to be a default:n device
>> (so to
>> > allow it to be shared with other concurrent programs), or maybe you are
>> > using an ALSA voicemodem and slmodemd is running?
>> >
>> > 2011-06-30 19:03:56.616460 [ERR] gsmopen_protocol.cpp:2891 rev
>> > exported[(nil)|37     ][ERRORA  2891 ][interface1][-1, 0, 0] Failed
>> opening
>> > ALSA capture device: plughw:1
>> > 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:1931 rev
>> exported[(nil)|37
>> > ][ERRORA  1931 ][interface1][-1, 0, 0] alsa_init failed
>> >
>> > 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:1932 rev
>> exported[(nil)|37
>> > ][ERRORA  1932 ][interface1][-1, 0, 0] STARTING interface_id=1 FAILED
>> > 2011-06-30 19:03:56.616460 [ERR] mod_gsmopen.cpp:3146 rev
>> exported[(nil)|37
>> > ][ERRORA  3146 ][interface1][-1, 0, 0] ALARM on interface interface1:
>> >
>> >
>> >
>> -----------------------------------------------------------------------------------------------------------------
>> > I could not understand what does the error message trying to say? I am
>> > connected to friend's mobile with datacable on com1 port. I jsut wanted
>> to
>> > check SMS. Please help.
>> >
>> > Thanks
>> > Ankit
>> >
>> > --
>> > Life is like a rose its upto u feel it as its fragrance or thorns
>> >
>> > _______________________________________________
>> > 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
>> >
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> 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
>>
>
>
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>
> _______________________________________________
> 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
>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From rajesh.npnr at yahoo.com  Fri Jul  1 19:48:09 2011
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Fri, 1 Jul 2011 08:48:09 -0700 (PDT)
Subject: [Freeswitch-users] Inband DTMF set in Event Socket
In-Reply-To: <BANLkTinmbb5dwN64EjFe+vP50nxR5-C5rw@mail.gmail.com>
References: <1309524076282-6537620.post@n2.nabble.com>
	<BANLkTinHT_o-_ytPr7gjL8hAoSzjzh3-7A@mail.gmail.com>
	<1309531579711-6538036.post@n2.nabble.com>
	<BANLkTinmbb5dwN64EjFe+vP50nxR5-C5rw@mail.gmail.com>
Message-ID: <1309535289627-6538248.post@n2.nabble.com>

similar way I tried both

api originate {origination_caller_id_name='XXX',
start_dtmf_generate=true}user/1000
&bridge({accountcode='XXX'}sofia/gateway/gw_name/phno)

and 

api originate {origination_caller_id_name='XXX'}user/1000
&bridge({accountcode='XXX',
start_dtmf_generate=true}sofia/gateway/gw_name/phno)

it's not working, but the inband type is working when i use it in dialplan.

Please assist.

Thanks,
Rex

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Inband-DTMF-set-in-Event-Socket-tp6537620p6538248.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From abid_freeswitch at live.com  Fri Jul  1 19:54:04 2011
From: abid_freeswitch at live.com (Abid Saleem)
Date: Fri, 1 Jul 2011 21:54:04 +0600
Subject: [Freeswitch-users] Load Balance Trunks
In-Reply-To: <C71A4485-E533-48AE-AC3F-C59D7A1E79CA@ipeva.fr>
References: <col109-w22D1131E50A2495C1521C9F4590@phx.gbl>,
	<C71A4485-E533-48AE-AC3F-C59D7A1E79CA@ipeva.fr>
Message-ID: <COL109-W612603C4C7BCC734CBD974F45B0@phx.gbl>


Dear David,
Thanks for answer. This is the only provider choice I have. Also I am not good at programming, so can you help me writing such a script that can do this please. 


From: david.ponzone at ipeva.fr
Date: Wed, 29 Jun 2011 17:41:41 +0200
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Load Balance Trunks



1- I have 100 Trunks from my SIP Provider. My provider restricts me to send only 120 minutes call duration per Trunk per each day. This means around 30 calls per Trunk per day with an ACD of 3 minutes. Please help me how to configure this if it is possible?
Can't you find another provider with no such ... limitations ?
You can do what Avi told you, or you can write your own piece of LUA script which would check in a DB which trunk to use, and then would bridge using this trunk, and will update the DB after hangup to add the call duration of the last call to the total amount for trunk X and current day (that would have to be done with the hangup_hook thingy, probably).
2- I need to send P-Preferred-Identity<sip:xxxxxxxxx at domain.com> in SIP header for each Trunk while dialing out to provider. Please help how to configure this.
bridge {sip_cid_type=none,sip_h_P-Preferred-Identity=XXXXXXX}sofia/.....  -> I am sure of this oneorbridge {sip_cid_type=pid}sofia/.... -> never tested
Thanks in Advance.
Rgrds-----------Abid SaleemTechnical Manager NGNTerminus Technologies_______________________________________________
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


_______________________________________________
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
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From brian at freeswitch.org  Fri Jul  1 19:59:57 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 1 Jul 2011 11:59:57 -0400
Subject: [Freeswitch-users] Inband DTMF set in Event Socket
In-Reply-To: <1309535289627-6538248.post@n2.nabble.com>
References: <1309524076282-6537620.post@n2.nabble.com>
	<BANLkTinHT_o-_ytPr7gjL8hAoSzjzh3-7A@mail.gmail.com>
	<1309531579711-6538036.post@n2.nabble.com>
	<BANLkTinmbb5dwN64EjFe+vP50nxR5-C5rw@mail.gmail.com>
	<1309535289627-6538248.post@n2.nabble.com>
Message-ID: <6D420599-38CB-4CD8-9070-337723DE89F2@freeswitch.org>

/me shakes his head.... start_dtmf_generate is an application not a variable :P

use it with execute_on_answer usually...

/b

On Jul 1, 2011, at 11:48 AM, rex.alex wrote:

> similar way I tried both
> 
> api originate {origination_caller_id_name='XXX',
> start_dtmf_generate=true}user/1000
> &bridge({accountcode='XXX'}sofia/gateway/gw_name/phno)
> 
> and 
> 
> api originate {origination_caller_id_name='XXX'}user/1000
> &bridge({accountcode='XXX',
> start_dtmf_generate=true}sofia/gateway/gw_name/phno)
> 
> it's not working, but the inband type is working when i use it in dialplan.
> 
> Please assist.
> 
> Thanks,
> Rex
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Inband-DTMF-set-in-Event-Socket-tp6537620p6538248.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> 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 msc at freeswitch.org  Fri Jul  1 20:07:59 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Jul 2011 09:07:59 -0700
Subject: [Freeswitch-users] Inband DTMF set in Event Socket
In-Reply-To: <1309535289627-6538248.post@n2.nabble.com>
References: <1309524076282-6537620.post@n2.nabble.com>
	<BANLkTinHT_o-_ytPr7gjL8hAoSzjzh3-7A@mail.gmail.com>
	<1309531579711-6538036.post@n2.nabble.com>
	<BANLkTinmbb5dwN64EjFe+vP50nxR5-C5rw@mail.gmail.com>
	<1309535289627-6538248.post@n2.nabble.com>
Message-ID: <BANLkTikkNSr9wf-SfhNQBW6ERH+sj=z=8w@mail.gmail.com>

On Fri, Jul 1, 2011 at 8:48 AM, rex.alex <rajesh.npnr at yahoo.com> wrote:

> similar way I tried both
>
> api originate {origination_caller_id_name='XXX',
> start_dtmf_generate=true}user/1000
> &bridge({accountcode='XXX'}sofia/gateway/gw_name/phno)
>
> and
>
> api originate {origination_caller_id_name='XXX'}user/1000
> &bridge({accountcode='XXX',
> start_dtmf_generate=true}sofia/gateway/gw_name/phno)
>
>
it's not working, but the inband type is working when i use it in dialplan.
>
> Please assist.
>
> Which leg needs to have inband DTMF - the calling or the called?

You can use these variables to do a pre-execute on either leg:
http://wiki.freeswitch.org/wiki/Variable_bridge_pre_execute_aleg_app
http://wiki.freeswitch.org/wiki/Variable_bridge_pre_execute_aleg_data
http://wiki.freeswitch.org/wiki/Variable_bridge_pre_execute_bleg_app
http://wiki.freeswitch.org/wiki/Variable_bridge_pre_execute_bleg_data

start_dtmf_generate is a dialplan app, not a chan var, so it needs to be
executed, not "set".

-MC
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From msc at freeswitch.org  Fri Jul  1 20:09:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Jul 2011 09:09:38 -0700
Subject: [Freeswitch-users] Load Balance Trunks
In-Reply-To: <COL109-W612603C4C7BCC734CBD974F45B0@phx.gbl>
References: <col109-w22D1131E50A2495C1521C9F4590@phx.gbl>
	<C71A4485-E533-48AE-AC3F-C59D7A1E79CA@ipeva.fr>
	<COL109-W612603C4C7BCC734CBD974F45B0@phx.gbl>
Message-ID: <BANLkTinT9O5+hzT1YczsmQV5ouL9EQ_2TQ@mail.gmail.com>

How does the provider notify you that each trunk has used its allotted time?
-MC

On Fri, Jul 1, 2011 at 8:54 AM, Abid Saleem <abid_freeswitch at live.com>wrote:

>  Dear David,
>
> Thanks for answer. This is the only provider choice I have. Also I am not
> good at programming, so can you help me writing such a script that can do
> this please.
>
>
>
> ------------------------------
> From: david.ponzone at ipeva.fr
> Date: Wed, 29 Jun 2011 17:41:41 +0200
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Load Balance Trunks
>
> 1- I have 100 Trunks from my SIP Provider. My provider restricts me to send
> only 120 minutes call duration per Trunk per each day. This means around 30
> calls per Trunk per day with an ACD of 3 minutes. Please help me how to
> configure this if it is possible?
>
>
> Can't you find another provider with no such ... limitations ?
>
> You can do what Avi told you, or you can write your own piece of LUA script
> which would check in a DB which trunk to use, and then would bridge using
> this trunk, and will update the DB after hangup to add the call duration of
> the last call to the total amount for trunk X and current day (that would
> have to be done with the hangup_hook thingy, probably).
>
> 2- I need to send P-Preferred-Identity<sip:xxxxxxxxx at domain.com> in SIP
> header for each Trunk while dialing out to provider. Please help how to
> configure this.
>
> bridge {sip_cid_type=none,sip_h_P-Preferred-Identity=XXXXXXX}sofia/.....
>  -> I am sure of this one
> or
> bridge {sip_cid_type=pid}sofia/.... -> never tested
>
> Thanks in Advance.
>
> Rgrds
> -----------
> Abid Saleem
> Technical Manager NGN
> Terminus Technologies
> _______________________________________________
> 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
>
>
>
> _______________________________________________ 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
>
> _______________________________________________
> 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
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From hmkias at gmail.com  Fri Jul  1 20:18:07 2011
From: hmkias at gmail.com (hmkias at gmail.com)
Date: Fri, 1 Jul 2011 16:18:07 +0000
Subject: [Freeswitch-users] Load Balance Trunks
In-Reply-To: <BANLkTinT9O5+hzT1YczsmQV5ouL9EQ_2TQ@mail.gmail.com>
References: <col109-w22D1131E50A2495C1521C9F4590@phx.gbl><C71A4485-E533-48AE-AC3F-C59D7A1E79CA@ipeva.fr><COL109-W612603C4C7BCC734CBD974F45B0@phx.gbl><BANLkTinT9O5+hzT1YczsmQV5ouL9EQ_2TQ@mail.gmail.com>
Message-ID: <233898962-1309536985-cardhu_decombobulator_blackberry.rim.net-2106826141-@b13.c4.bise7.blackberry>

What happens when a  call extends more than 120 mins on a trunk, would the call be disconnected.
Sent from BSNL with my BlackBerry? smartphone

-----Original Message-----
From: Michael Collins <msc at freeswitch.org>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Fri, 1 Jul 2011 09:09:38 
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] Load Balance Trunks

_______________________________________________
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 msc at freeswitch.org  Fri Jul  1 20:21:06 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Jul 2011 09:21:06 -0700
Subject: [Freeswitch-users] Answer issue on inbound call
In-Reply-To: <BANLkTim5jOswqjM9DwpH99UpjCm5MmVXjw@mail.gmail.com>
References: <BANLkTin+f=uDMiOouyLWAorPvNTd1Z6xaA@mail.gmail.com>
	<BANLkTim=TOQD1tUvmHjeRuhAR-eKkGYgsw@mail.gmail.com>
	<BANLkTim3pTPV804t_Usk59eG2C56HAyFdA@mail.gmail.com>
	<BANLkTi=Q3qRBCDRARDWp15=Guwyq-jtNLQ@mail.gmail.com>
	<BANLkTinMokGeJU+wVVA0U-oxB97z+c+ZGg@mail.gmail.com>
	<BANLkTinEp1Cu5mB_UHw-Y+mitGsNByR3sQ@mail.gmail.com>
	<BANLkTikcWDbCkTPRow-TMqPN=D3FV+LjWw@mail.gmail.com>
	<BANLkTinE1292_p_VZd+D4N6vvEEot8LXRw@mail.gmail.com>
	<BANLkTin5zjPmdaV6Qa9ZM8dYtBGBUmcKOg@mail.gmail.com>
	<BANLkTim5jOswqjM9DwpH99UpjCm5MmVXjw@mail.gmail.com>
Message-ID: <BANLkTinvomsqyR3JMn0Z58Hoq4XJx=oRPw@mail.gmail.com>

What's with the errors on the voicemail sound files?
-MC

On Thu, Jun 30, 2011 at 9:12 PM, Max Alex <max.asterisk at gmail.com> wrote:

> Hi,
> Can any one provide their suggestions and help for this issue,
> I really need to resolve and get this working.
>
> Thanks,
> Max Alex
> Voip Developer
>
>
>
> On Tue, Jun 28, 2011 at 6:01 PM, Max Alex <max.asterisk at gmail.com> wrote:
>
>> Hi,
>> Thanks for your reply.
>> I have enabled logger as per your help.
>> I have given completed log on following link of pastebin.
>> http://pastebin.freeswitch.org/16616
>>
>> You can see this line as it is showing answered and this call is answered
>> on my cell phone but on softphone it is ringing and i have rejected from
>> there.
>> 2011-06-28 17:47:52.909657 [NOTICE] mod_freetdm.c:1953 Channel [FreeTDM/2
>> :1/01234567890] has been answered
>>
>> It is pre answering the cell phone when it is ringing on phone
>> Waiting for your help.
>>
>> Thanks,
>> Max Alex
>> Voip Developer
>>
>>
>>
>> On Mon, Jun 27, 2011 at 9:15 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Get a complete, unedited, unabridged console debug log w/ siptrace and
>>> put it on pastebin w/ "FreeSWITCH Log" for the syntax highlighting. Use
>>> "sofia global siptrace on" to make sure you can all SIP traffic.
>>>
>>> -MC
>>>
>>>
>>> On Mon, Jun 27, 2011 at 6:05 AM, Max Alex <max.asterisk at gmail.com>wrote:
>>>
>>>> Hi,
>>>> Thanks for reply,
>>>> I have tried the same way and reloaded freeswitch, but still it is
>>>> answered on first ring of the call.
>>>> When it is ringing the call on 1001 and the same time it is answered on
>>>> cell phone, so something is done when it is ringing on 1001.
>>>>
>>>> Here is the dialplan for the same
>>>>
>>>>     <extension name="Local_Extension">
>>>>       <condition field="destination_number"
>>>> expression="^(10[01][0-9])$">
>>>>         <action application="set" data="dialed_extension=$1"/>
>>>>         <action application="export" data="dialed_extension=$1"/>
>>>>         <action application="bind_meta_app" data="2 b s
>>>> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>>>>         <action application="bind_meta_app" data="3 b s
>>>> execute_extension::cf XML features"/>-->
>>>>         <action application="set" data="ringback=${us-ring}"/>
>>>>         <action application="set"
>>>> data="transfer_ringback=$${hold_music}"/>
>>>>         <action application="set" data="call_timeout=30"/>
>>>>           <action application="set" data="hangup_after_bridge=true"/>
>>>>          <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="set"
>>>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>>>> var callgroup)}"/>
>>>>         <action application="hash"
>>>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>>>>
>>>>        <action application="bridge"
>>>> data="{ignore_early_media=true}user/${dialed_extension}@
>>>> ${domain_name}"/>
>>>>         <action application="sleep" data="1000"/>
>>>>         <action application="voicemail" data="default ${domain_name}
>>>> ${dialed_extension}"/>
>>>>       </condition>
>>>>     </extension>
>>>>
>>>>
>>>> Please help me for this issue.
>>>>
>>>>
>>>> Thanks,
>>>> Max Alex
>>>> Voip Developer
>>>>
>>>>
>>>>
>>>> On Fri, Jun 24, 2011 at 11:38 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> Are you using the default dialplan? I think you might just need to
>>>>> ignore early media on your bridge app. If you are using the default.xml file
>>>>> then locate "Local_Extension" and the bridge line:
>>>>>
>>>>>     <action application="bridge" data="user/${dialed_extension}@
>>>>> ${domain_name}"/>
>>>>>
>>>>> Change it to this, then try again:
>>>>>
>>>>>     <action application="bridge"
>>>>> data="{ignore_early_media=true}user/${dialed_extension}@
>>>>> ${domain_name}"/>
>>>>>
>>>>> If I understand correctly, the "symptom" you are experiencing is the
>>>>> normal operation of the bridge app (and it's cousin, the originate API
>>>>> command). When the b-leg sends back media, including ringing, the bridge (or
>>>>> the originate) will be considered "successful," and in the case of bridge,
>>>>> the b-leg's audio (the early media) will be connected to the a-leg. If you
>>>>> set ignore_early_media=true then you are explicitly telling the bridge app
>>>>> that you only want to connect the b-leg to the a-leg if the b-leg actually
>>>>> answers.
>>>>>
>>>>> I hope that made sense...
>>>>>
>>>>> -MC
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Jun 23, 2011 at 9:32 PM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>>
>>>>>> Hi,
>>>>>> Thanks for reply.
>>>>>> Current scenario is below.
>>>>>>
>>>>>> PSTN call -> sangoma device -> freeswitch incoming context -> routed
>>>>>> to 1001 -> ringing (Answered on cellphone)
>>>>>> Here when it is routed to 1001 the call it is started ringing, but on
>>>>>> phone that call is answered and hearding sound of ringing.
>>>>>>
>>>>>> Required flow:
>>>>>> PSTN call -> sangoma device -> freeswitch incoming context -> routed
>>>>>> to 1001 -> ringing (Ringing on cellphone)
>>>>>>
>>>>>> I hope it is understable, the call should not be answered until 1001
>>>>>> answer it, right not when it is started ring it is answered on cell phone.
>>>>>> that should not be happen as it is not answered yet.
>>>>>>
>>>>>> Waiting for your reply.
>>>>>>
>>>>>>
>>>>>> Thanks,
>>>>>> Max Alex
>>>>>> Voip Developer
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Fri, Jun 24, 2011 at 5:48 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>>
>>>>>>> I'm not sure I understand the problem. What is happening vs. what you
>>>>>>> believe should be happening?
>>>>>>> -MC
>>>>>>>
>>>>>>>
>>>>>>> On Thu, Jun 23, 2011 at 3:31 AM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>>>>
>>>>>>>> Hi,
>>>>>>>> Thanks for your reply.
>>>>>>>> Here is my configuration and log
>>>>>>>> http://pastebin.freeswitch.org/16571
>>>>>>>>
>>>>>>>> I am using A200 analog sangoma device with freeswitch, it is working
>>>>>>>> fine but when it is routing call to 1001 then it is answered.
>>>>>>>> Please provider your suggestions.
>>>>>>>>
>>>>>>>> Thanks,
>>>>>>>> Max Alex
>>>>>>>> Voip Developer
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Wed, Jun 22, 2011 at 8:51 PM, Michael Collins <
>>>>>>>> msc at freeswitch.org> wrote:
>>>>>>>>
>>>>>>>>> I thought the A200 was an analog card? Maybe I have my numbers
>>>>>>>>> mixed up...
>>>>>>>>>
>>>>>>>>> Go ahead and collect a debug log of this call. It might help to
>>>>>>>>> have your configs posted as well. Use pastebin.freeswitch.org. See
>>>>>>>>> this wiki article for tips on how to collect information:
>>>>>>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>>>>>>>
>>>>>>>>> -MC
>>>>>>>>>
>>>>>>>>> On Wed, Jun 22, 2011 at 3:23 AM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>>>>>>
>>>>>>>>>> Hi,
>>>>>>>>>> I have installed freeswitch latest version with sangoma card A200
>>>>>>>>>> as well,
>>>>>>>>>> I have installed and configured freetdm module with wanpipe
>>>>>>>>>> drivers for freeswitch,
>>>>>>>>>> We are properly receiving the inbound calls in public context and
>>>>>>>>>> then we are routing that call to 1001 extension,
>>>>>>>>>> it is properly routing to 1001 as well, but we have one issue
>>>>>>>>>> while routing on 1001.
>>>>>>>>>>
>>>>>>>>>> Here is the issue description.
>>>>>>>>>> I am calling from my cell phone to that DID number of pri line,
>>>>>>>>>> and then it will start ringing on 1001 extension,
>>>>>>>>>> When 1001 extension start ringing the call is answered on my cell
>>>>>>>>>> phone,
>>>>>>>>>> it is something like freeswitch preanswer or autoanswer the call,
>>>>>>>>>> how can i stop this answer call when it is ringing on 1001 extension,
>>>>>>>>>> Waiting for good reply.
>>>>>>>>>>
>>>>>>>>>> Thanks,
>>>>>>>>>> Max Alex
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> 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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> 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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> 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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> 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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>
> _______________________________________________
> 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 msc at freeswitch.org  Fri Jul  1 20:21:51 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Jul 2011 09:21:51 -0700
Subject: [Freeswitch-users] Load Balance Trunks
In-Reply-To: <233898962-1309536985-cardhu_decombobulator_blackberry.rim.net-2106826141-@b13.c4.bise7.blackberry>
References: <col109-w22D1131E50A2495C1521C9F4590@phx.gbl>
	<C71A4485-E533-48AE-AC3F-C59D7A1E79CA@ipeva.fr>
	<COL109-W612603C4C7BCC734CBD974F45B0@phx.gbl>
	<BANLkTinT9O5+hzT1YczsmQV5ouL9EQ_2TQ@mail.gmail.com>
	<233898962-1309536985-cardhu_decombobulator_blackberry.rim.net-2106826141-@b13.c4.bise7.blackberry>
Message-ID: <BANLkTikjYw=0UDyGdipBzadj0_kSoOVS2Q@mail.gmail.com>

And then no more calls that day on that trunk?
-MC

On Fri, Jul 1, 2011 at 9:18 AM, <hmkias at gmail.com> wrote:

> What happens when a  call extends more than 120 mins on a trunk, would the
> call be disconnected.
> Sent from BSNL with my BlackBerry? smartphone
>
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From henk at oegema.com  Fri Jul  1 20:29:32 2011
From: henk at oegema.com (Henk Oegema)
Date: Fri, 01 Jul 2011 18:29:32 +0200
Subject: [Freeswitch-users] No voice channels to mobile.
Message-ID: <1309537772.2523.124.camel@DELL>

I'm facing the following problem:

External calls come in via an ITSP (Localphone) and then routed to
internal extension 1000.
That works OK.

If there's no answer, then the call is routed outbound via ITSP
(Powervoip) to my mobile +316xxxxxxxxx)
In this case there is NO voice communication at all. 
(a direct call from 1000 to a mobile is OK)


<extension name="1000">
    <condition field="destination_number" expression="^1000$">
      ..........................................................................................
      ..........................................................................................
      <action application="set" data="continue_on_fail=true"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="bridge"
data="{originate_timeout=20}sofia/internal/1000%$${domain}"/>
<----------audio OK
      <action application="bridge"
data="{originate_timeout=20}sofia/gateway/powervoip/003163086xxxx"/>
<------------NO audio both ways
      <action application="voicemail" data="default $${domain} 1000"/>
    </condition>
  </extension>

How do I approach this problem?

Rgds,
Henk
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From msc at freeswitch.org  Fri Jul  1 20:39:35 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Jul 2011 09:39:35 -0700
Subject: [Freeswitch-users] Call Dropped When Answered If Bridged To
 Internal Extension And PSTN Number Via
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D69624F481C617@VA3DIAXVS351.RED001.local>
References: <2BF7FB90DF25EA4485949F3AF2B9D69624F481C617@VA3DIAXVS351.RED001.local>
Message-ID: <BANLkTikMpW_C3YuJ7LBzyfEsFt5fdZGX4Q@mail.gmail.com>

Wait, can you explain that again? Step us through the call flow. Who
connects to whom and in what order, etc.
-MC

On Thu, Jun 30, 2011 at 11:42 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

> Hello,****
>
> ** **
>
> If I bridge a call to both an internal extension and a PSTN number (via
> FreeTDM), and then answer the call at the internal extension, the call is
> dropped.  I posted the log at ****
>
> ** **
>
> http://pastebin.freeswitch.org/16634****
>
> ** **
>
> The log shows many errors of the type:****
>
> ** **
>
> reading on a session with no media!****
>
> ** **
>
> I have bypass-media 'true' (not sure if this has anything to do with it).
> Any clue why this is happening?****
>
> ** **
>
> Thanks,****
>
> Jerry ****
>
> _______________________________________________
> 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 hmkias at gmail.com  Fri Jul  1 20:46:15 2011
From: hmkias at gmail.com (hmkias at gmail.com)
Date: Fri, 1 Jul 2011 16:46:15 +0000
Subject: [Freeswitch-users] Load Balance Trunks
In-Reply-To: <BANLkTikjYw=0UDyGdipBzadj0_kSoOVS2Q@mail.gmail.com>
References: <col109-w22D1131E50A2495C1521C9F4590@phx.gbl><C71A4485-E533-48AE-AC3F-C59D7A1E79CA@ipeva.fr><COL109-W612603C4C7BCC734CBD974F45B0@phx.gbl><BANLkTinT9O5+hzT1YczsmQV5ouL9EQ_2TQ@mail.gmail.com><233898962-1309536985-cardhu_decombobulator_blackberry.rim.net-2106826141-@b13.c4.bise7.blackberry><BANLkTikjYw=0UDyGdipBzadj0_kSoOVS2Q@mail.gmail.com>
Message-ID: <1637192224-1309538673-cardhu_decombobulator_blackberry.rim.net-1258156530-@b13.c4.bise7.blackberry>

Lcr entry could be activated on the the trunks. A cron job could monitor the usage and add or remove the trunk for the day.
Sent from BSNL with my BlackBerry? smartphone

-----Original Message-----
From: Michael Collins <msc at freeswitch.org>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Fri, 1 Jul 2011 09:21:51 
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] Load Balance Trunks

_______________________________________________
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 msc at freeswitch.org  Fri Jul  1 20:49:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Jul 2011 09:49:02 -0700
Subject: [Freeswitch-users] Load Balance Trunks
In-Reply-To: <1637192224-1309538673-cardhu_decombobulator_blackberry.rim.net-1258156530-@b13.c4.bise7.blackberry>
References: <col109-w22D1131E50A2495C1521C9F4590@phx.gbl>
	<C71A4485-E533-48AE-AC3F-C59D7A1E79CA@ipeva.fr>
	<COL109-W612603C4C7BCC734CBD974F45B0@phx.gbl>
	<BANLkTinT9O5+hzT1YczsmQV5ouL9EQ_2TQ@mail.gmail.com>
	<233898962-1309536985-cardhu_decombobulator_blackberry.rim.net-2106826141-@b13.c4.bise7.blackberry>
	<BANLkTikjYw=0UDyGdipBzadj0_kSoOVS2Q@mail.gmail.com>
	<1637192224-1309538673-cardhu_decombobulator_blackberry.rim.net-1258156530-@b13.c4.bise7.blackberry>
Message-ID: <BANLkTineHeUHCMpKNJS7msb6xCZzBqmkQg@mail.gmail.com>

On Fri, Jul 1, 2011 at 9:46 AM, <hmkias at gmail.com> wrote:

> Lcr entry could be activated on the the trunks. A cron job could monitor
> the usage and add or remove the trunk for the day.
>

Are these trunks inbound only?
-MC
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From rzhang at gosilverplus.com  Fri Jul  1 20:59:26 2011
From: rzhang at gosilverplus.com (ran zhang)
Date: Fri, 01 Jul 2011 09:59:26 -0700
Subject: [Freeswitch-users] please help!!! how to set flag 'endconf' in
 bridging conference
In-Reply-To: <BANLkTimpd--c8JabTrd1ki3rFFt+tZY3bg@mail.gmail.com>
References: <E826C94A3C6BE44198FAF4D66B5AB4D81BDA7E5903@VA3DIAXVS351.RED001.local>	<4E0CB983.8060407@gosilverplus.com>	<BANLkTikb-xbvLwzsSv7nQJZncx8y_v+hpw@mail.gmail.com>	<4E0CE1B7.60609@gosilverplus.com>	<BANLkTi=Cmuz6a3K8smK3Nr46E2ptg=pZdQ@mail.gmail.com>	<4E0D233B.6010609@gosilverplus.com>	<BANLkTinmHti_769mQ5Xbs4sV6U0em-VZaw@mail.gmail.com>	<4E0DF997.2060105@gosilverplus.com>
	<BANLkTimpd--c8JabTrd1ki3rFFt+tZY3bg@mail.gmail.com>
Message-ID: <4E0DFCEE.5010203@gosilverplus.com>

I think most people don't understand my problem, let me just clarify:

I want a way so only when either one of the  first 2 members of the 
bridging conference leaves the conference,
the conference will be terminated even if there are 2 or more members in it.

i'm trying to set 'endconf' flag in a bridging conference using
'bridge:confname+flag{endconf}:user/10',
so the first member entering the conference will invite user extension 
10, but its giving me config error while
running.

I only want to use 'bridging conference' since i want at least 2 people 
in the conference to start with.








On 7/1/2011 9:46 AM, Michael Collins wrote:
>
>
> On Fri, Jul 1, 2011 at 9:45 AM, ran zhang <rzhang at gosilverplus.com 
> <mailto:rzhang at gosilverplus.com>> wrote:
>
>     do u understand my problem with the bridging conference?  pl
>     suggest on what to do.
>
> Honestly, no. But I want this discussion on the public mailing list so 
> that we all can talk about it.
> -MC

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From msc at freeswitch.org  Fri Jul  1 21:28:34 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Jul 2011 10:28:34 -0700
Subject: [Freeswitch-users] please help!!! how to set flag 'endconf' in
 bridging conference
In-Reply-To: <4E0DFCEE.5010203@gosilverplus.com>
References: <E826C94A3C6BE44198FAF4D66B5AB4D81BDA7E5903@VA3DIAXVS351.RED001.local>
	<4E0CB983.8060407@gosilverplus.com>
	<BANLkTikb-xbvLwzsSv7nQJZncx8y_v+hpw@mail.gmail.com>
	<4E0CE1B7.60609@gosilverplus.com>
	<BANLkTi=Cmuz6a3K8smK3Nr46E2ptg=pZdQ@mail.gmail.com>
	<4E0D233B.6010609@gosilverplus.com>
	<BANLkTinmHti_769mQ5Xbs4sV6U0em-VZaw@mail.gmail.com>
	<4E0DF997.2060105@gosilverplus.com>
	<BANLkTimpd--c8JabTrd1ki3rFFt+tZY3bg@mail.gmail.com>
	<4E0DFCEE.5010203@gosilverplus.com>
Message-ID: <BANLkTi=URVjLSShw+gmnijEkRNfrGiy0XQ@mail.gmail.com>

On Fri, Jul 1, 2011 at 9:59 AM, ran zhang <rzhang at gosilverplus.com> wrote:

> **
> I think most people don't understand my problem, let me just clarify:
>
> I want a way so only when either one of the  first 2 members of the
> bridging conference leaves the conference,
> the conference will be terminated even if there are 2 or more members in
> it.
>
>
The problem with using the endconf flag is that *all* members with that flag
must leave before the conference is torn down. I don't believe that this
feature is presently in FS but with a feature request and possibly a modest
bounty it could possibly be added. Something like a flag named
"endconfabsolute" that would end the conference no matter what.

Now your other problem was trying to do a "bridging" conference and also set
the endconf flag and you were getting errors. I was able to reproduce the
symptoms you saw, but I have not investigated any further at this point.

If anyone knows the trick to making the members added to a "bridging"
conference also have the endconf flag set (or any flags for that matter)
please let me know and we can get it documented.

-MC
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From jerry.richards at teotech.com  Fri Jul  1 21:50:44 2011
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 1 Jul 2011 10:50:44 -0700
Subject: [Freeswitch-users] Call Dropped When Answered If Bridged To
 Internal Extension And PSTN Number Via
In-Reply-To: <BANLkTikMpW_C3YuJ7LBzyfEsFt5fdZGX4Q@mail.gmail.com>
References: <2BF7FB90DF25EA4485949F3AF2B9D69624F481C617@VA3DIAXVS351.RED001.local>
	<BANLkTikMpW_C3YuJ7LBzyfEsFt5fdZGX4Q@mail.gmail.com>
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D69624F481C95A@VA3DIAXVS351.RED001.local>

MC,

I have two internal extensions (2001 and 2002) and a PRI connected to the PSTN.  Additionally, the dialplan is configured so when 2001 is called, it will bridge to two destinations (internal extension 2001 and PSTN number 4253491028).

Here is the detailed scenario of http://pastebin.freeswitch.org/16634:

Line: Description
0005: At internal 2002, call internal 2001
0490: Execute this bridge statement:
bridge({ignore_early_media=ring_ready}{presence_id=2001 at 192.168.72.79}sofia/internal/sip:2001 at 192.168.73.120:5060;transport=udp,sofia/internal/sip:2001 at 192.168.72.87:5060;transport=udp,sofia/internal/sip:2001 at 192.168.73.125:5060;transport=udp,[lcr_carrier=Carrier / Location 1/INTERNAL PRI,lcr_rate=0.00200,origination_caller_id_number=2002]freetdm/grp1/a/4253491028)
1098: At internal 2001, answer call
1290: A long series of "reading on a session with no media" logs starts
1720: Freeswitch sends BYE to 2001 with reason "INCOMPATIBLE DESTINATION"
1752: Freeswitch sends BYE to 2002 with reason "INCOMPATIBLE DESTINATION"

Thanks Much,
Jerry



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, July 01, 2011 9:40 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call Dropped When Answered If Bridged To Internal Extension And PSTN Number Via

Wait, can you explain that again? Step us through the call flow. Who connects to whom and in what order, etc.
-MC
On Thu, Jun 30, 2011 at 11:42 AM, Jerry Richards <jerry.richards at teotech.com<mailto:jerry.richards at teotech.com>> wrote:
Hello,

If I bridge a call to both an internal extension and a PSTN number (via FreeTDM), and then answer the call at the internal extension, the call is dropped.  I posted the log at

http://pastebin.freeswitch.org/16634

The log shows many errors of the type:

reading on a session with no media!

I have bypass-media 'true' (not sure if this has anything to do with it).  Any clue why this is happening?

Thanks,
Jerry

_______________________________________________
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<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 dave at dchorton.com  Fri Jul  1 23:37:50 2011
From: dave at dchorton.com (Dave Horton)
Date: Fri, 1 Jul 2011 15:37:50 -0400
Subject: [Freeswitch-users] REGISTER response sent to the wrong por
Message-ID: <A66F955C-EA7C-4355-83C1-38FF53011F44@dchorton.com>


I have a phone behind a firewall registering with the internal profile.  The address in the Via header is not reachable, and the FS (correctly, IMO) sends the response back to the actual sending IP address.  However, rather than sending it to the actual sending udp port, it incorrectly (again, IMO) sends it to the port advertised in the Via header.  Of course the message does not get through the firewall, and the phone keeps retransmitting the REGISTER.

Is this a bug, or am I missing something?  And is there a setting where I can force the behavior that I want (which is: send responses to the sending address:port if the sending address:port is different than the Via header).

Trace below..


recv 481 bytes from udp/[66.37.97.53]:1024 at 19:28:12.445081:
   ------------------------------------------------------------------------
   REGISTER sip:65.162.239.204 SIP/2.0
   Via: SIP/2.0/UDP 10.93.14.119:5064;branch=z9hG4bK-e73de4bf
   From: "1002" <sip:1002 at 65.162.239.204>;tag=7718f2af7f3dcc73o2
   To: "1002" <sip:1002 at 65.162.239.204>
   Call-ID: 930c7e31-648ae3db at 10.93.14.119
   CSeq: 53622 REGISTER
   Max-Forwards: 70
   Contact: "1002" <sip:1002 at 10.93.14.119:5064>;expires=3600
   User-Agent: Linksys/SPA962-5.2.8(SC)
   Content-Length: 0
   Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
   Supported: replaces
   
   ------------------------------------------------------------------------
send 671 bytes to udp/[66.37.97.53]:5064 at 19:28:12.445259:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP 10.93.14.119:5064;branch=z9hG4bK-e73de4bf;received=66.37.97.53
   From: "1002" <sip:1002 at 65.162.239.204>;tag=7718f2af7f3dcc73o2
   To: "1002" <sip:1002 at 65.162.239.204>;tag=t1m9emDKB4trF
   Call-ID: 930c7e31-648ae3db at 10.93.14.119
   CSeq: 53622 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-25c725c 2011-06-30 18-30-24 -0500
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   WWW-Authenticate: Digest realm="65.162.239.204", nonce="7e426985-2f7a-40ff-a817-6c167207853f", algorithm=MD5, qop="auth"
   Content-Length: 0




From kris at livecall.com  Fri Jul  1 23:59:33 2011
From: kris at livecall.com (Kris)
Date: Fri, 1 Jul 2011 12:59:33 -0700
Subject: [Freeswitch-users] change sounds file format of voicemail
References: <89F3619D8E6F437F82F3B1C00FA69725@e1705><EAC9775B572D499CA7869A2FA228C517@stor1>
	<4E0DD7E4.9050102@coppice.org>
Message-ID: <436BD03F7958410D97EFB7D79002E3FE@stor1>

There are a lot of Dialogics out there and I just needed compativility with 
it's 8 bit wav and 4bit vox and I will submit it to the GIT so others who 
need that compatibility can use it. When I first started with 2 phone lines, 
I had a 286 with 40mb disk. When you have a few thousand files you can copy, 
convert and not worry about it. When you have a million, not so fast... I 
want to have both Dialogic and FS running side by side and sharing messages. 
I am just not a fan of converting gigabytes of files, then find that I am up 
the creek without a paddle-something goes wrong..I wanted to rewrite my 
program in a modern language like C#, instead of continuing with ancient 
C,that  I started with 20 years ago. 16 bit wav, do sound better at 4x the 
storage.


----- Original Message ----- 
From: "Steve Underwood" <steveu at coppice.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, July 01, 2011 7:21 AM
Subject: Re: [Freeswitch-users] change sounds file format of voicemail


> On 06/30/2011 03:05 AM, Kris wrote:
>> I am doing testing and I am about to submit changes to the GIT to allow
>> 8000x8 bit wav and 8000x4bit vox recording. What formats would you like 
>> to
> 8000x8 bit is a really stupid choice. A-law or u-law would do a far better 
> job, despite being lossy codecs. The dynamic range of 8 bit line audio 
> just isn't enough for speech. 8000x4bit vox made sense in the 90s, because 
> Dialogic used it, and they were big in IVRs. It seems a very obsolete 
> choice these days.
>
> Steve
>
>
>
> 



From bryansmart at bryansmart.com  Sat Jul  2 00:21:03 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Fri, 1 Jul 2011 16:21:03 -0400
Subject: [Freeswitch-users] change sounds file format of voicemail
In-Reply-To: <EAC9775B572D499CA7869A2FA228C517@stor1>
References: <89F3619D8E6F437F82F3B1C00FA69725@e1705>
	<EAC9775B572D499CA7869A2FA228C517@stor1>
Message-ID: <08FCC436-921A-4722-BF2B-419CB997345E@bryansmart.com>

Would this work with general recording to a file, or is it just for voicemail?

Can we record to, or have voicemail record to, compressed formats?

Bryan

On Jun 29, 2011, at 3:05 PM, Kris wrote:

> I am doing testing and I am about to submit changes to the GIT to allow 
> 8000x8 bit wav and 8000x4bit vox recording. What formats would you like to 
> use?..maybe I can add them before I submit. This is my test extension and it 
> works with my modifications:
> Kris
> <extension name="playtest">
> 
> <condition field="destination_number" expression="^playtest$">
> 
> <action application="answer" data=""/>
> 
> <action application="set" data="tts_engine=flite"/>
> 
> <action application="set" data="tts_voice=kal"/>
> 
> <!--
> 
> -->
> 
> <action application="set" data="record_sample_rate=8000"/>
> 
> <action application="set" data="record_format= major=WAV,subtype=PCM_U8"/>
> 
> <action application="record" data="Test1.wav 20"/>
> 
> <action application="playback" data="tone_stream://%(100,0,1004);loops=5"/>
> 
> <action application="set" data="record_format= major=WAV,subtype=PCM_16"/>
> 
> <action application="record" data="Test2.wav 20"/>
> 
> <action application="playback" data="tone_stream://%(100,0,1004);loops=5"/>
> 
> <action application="set" data="record_format= 
> major=RAW,subtype=VOX_ADPCM"/>
> 
> <action application="sleep" data="200"/>
> 
> <action application="record" data="Test3.vox 20"/>
> 
> <action application="set" data="playback_delimiter=!"/>
> 
> 
> <action application="read" data="0 1 'say:this starts at 5 seconds and plays 
> for 10 at volume 
> 2!Test1.wav{start_ms=5000,timeout_ms=10000,speed=0,vol=2}!say:this is at 
> volume -1!Test2.wav{start_ms=5000,timeout_ms=10000,speed=0,vol=-1}!say:this 
> is vox!Test3.vox{start_ms=0,timeout_ms=8000,speed=2,vol=4}' digits 5000 #"/>
> 
> <action application="hangup"/>
> 
> </condition>
> 
> </extension>
> 
> ----- Original Message ----- 
> From: "Madovsky" <infos at madovsky.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, June 29, 2011 9:31 AM
> Subject: [Freeswitch-users] change sounds file format of voicemail
> 
> 
> Where can I force voicemail to use sounds other than .wav ?
> 
> Thanks 
> 
> 
> _______________________________________________
> 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 msc at freeswitch.org  Sat Jul  2 00:54:29 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Jul 2011 13:54:29 -0700
Subject: [Freeswitch-users] Call Dropped When Answered If Bridged To
 Internal Extension And PSTN Number Via
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D69624F481C95A@VA3DIAXVS351.RED001.local>
References: <2BF7FB90DF25EA4485949F3AF2B9D69624F481C617@VA3DIAXVS351.RED001.local>
	<BANLkTikMpW_C3YuJ7LBzyfEsFt5fdZGX4Q@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D69624F481C95A@VA3DIAXVS351.RED001.local>
Message-ID: <BANLkTimkYK-FBC4PVVfsAOkCYJ1TcWVaOg@mail.gmail.com>

So it looks like 2001 has multiple registrations, is that correct? It looks
like it is calling 3 endpoints:
2001 at 192.168.73.120
2001 at 192.168.72.87
2001 at 192.168.73.125

I'm just curious - what happens if either of the other SIP endpoints answers
this call?
-MC

On Fri, Jul 1, 2011 at 10:50 AM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> MC,****
>
> ** **
>
> I have two internal extensions (2001 and 2002) and a PRI connected to the
> PSTN.  Additionally, the dialplan is configured so when 2001 is called, it
> will bridge to two destinations (internal extension 2001 and PSTN number
> 4253491028).****
>
> ** **
>
> Here is the detailed scenario of http://pastebin.freeswitch.org/16634:****
>
> ** **
>
> Line: Description****
>
> 0005: At internal 2002, call internal 2001****
>
> 0490: Execute this bridge statement:****
>
> bridge({ignore_early_media=ring_ready}{presence_id=2001 at 192.168.72.79
> }sofia/internal/sip:2001 at 192.168.73.120:5060
> ;transport=udp,sofia/internal/sip:2001 at 192.168.72.87:5060
> ;transport=udp,sofia/internal/sip:2001 at 192.168.73.125:5060;transport=udp,[lcr_carrier=Carrier
> / Location 1/INTERNAL
> PRI,lcr_rate=0.00200,origination_caller_id_number=2002]freetdm/grp1/a/4253491028)
> ****
>
> 1098: At internal 2001, answer call****
>
> 1290: A long series of "reading on a session with no media" logs starts***
> *
>
> 1720: Freeswitch sends BYE to 2001 with reason "INCOMPATIBLE DESTINATION"*
> ***
>
> 1752: Freeswitch sends BYE to 2002 with reason "INCOMPATIBLE DESTINATION"*
> ***
>
> ** **
>
> Thanks Much,****
>
> Jerry****
>
> ** **
>
> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Friday, July 01, 2011 9:40 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Call Dropped When Answered If Bridged To
> Internal Extension And PSTN Number Via****
>
> ** **
>
> Wait, can you explain that again? Step us through the call flow. Who
> connects to whom and in what order, etc.
> -MC****
>
> On Thu, Jun 30, 2011 at 11:42 AM, Jerry Richards <
> jerry.richards at teotech.com> wrote:****
>
> Hello,****
>
>  ****
>
> If I bridge a call to both an internal extension and a PSTN number (via
> FreeTDM), and then answer the call at the internal extension, the call is
> dropped.  I posted the log at ****
>
>  ****
>
> http://pastebin.freeswitch.org/16634****
>
>  ****
>
> The log shows many errors of the type:****
>
>  ****
>
> reading on a session with no media!****
>
>  ****
>
> I have bypass-media 'true' (not sure if this has anything to do with it).
> Any clue why this is happening?****
>
>  ****
>
> Thanks,****
>
> Jerry ****
>
>
> _______________________________________________
> 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****
>
> ** **
>
> _______________________________________________
> 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 msc at freeswitch.org  Sat Jul  2 00:57:32 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Jul 2011 13:57:32 -0700
Subject: [Freeswitch-users] REGISTER response sent to the wrong por
In-Reply-To: <A66F955C-EA7C-4355-83C1-38FF53011F44@dchorton.com>
References: <A66F955C-EA7C-4355-83C1-38FF53011F44@dchorton.com>
Message-ID: <BANLkTi=iMx3hce_6XazPRY6VVEq8nOSM1w@mail.gmail.com>

Is your freeswitch also behind it's own firewall? If not then you probably
want to use the external.xml file as the basis for your public-facing SIP
ports. Some folks have simply renamed external.xml to internal.xml and
vice-versa.
-MC

On Fri, Jul 1, 2011 at 12:37 PM, Dave Horton <dave at dchorton.com> wrote:

>
> I have a phone behind a firewall registering with the internal profile.
>  The address in the Via header is not reachable, and the FS (correctly, IMO)
> sends the response back to the actual sending IP address.  However, rather
> than sending it to the actual sending udp port, it incorrectly (again, IMO)
> sends it to the port advertised in the Via header.  Of course the message
> does not get through the firewall, and the phone keeps retransmitting the
> REGISTER.
>
> Is this a bug, or am I missing something?  And is there a setting where I
> can force the behavior that I want (which is: send responses to the sending
> address:port if the sending address:port is different than the Via header).
>
> Trace below..
>
>
> recv 481 bytes from udp/[66.37.97.53]:1024 at 19:28:12.445081:
>   ------------------------------------------------------------------------
>   REGISTER sip:65.162.239.204 SIP/2.0
>   Via: SIP/2.0/UDP 10.93.14.119:5064;branch=z9hG4bK-e73de4bf
>   From: "1002" <sip:1002 at 65.162.239.204>;tag=7718f2af7f3dcc73o2
>   To: "1002" <sip:1002 at 65.162.239.204>
>   Call-ID: 930c7e31-648ae3db at 10.93.14.119
>   CSeq: 53622 REGISTER
>   Max-Forwards: 70
>   Contact: "1002" <sip:1002 at 10.93.14.119:5064>;expires=3600
>   User-Agent: Linksys/SPA962-5.2.8(SC)
>   Content-Length: 0
>   Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>   Supported: replaces
>
>   ------------------------------------------------------------------------
> send 671 bytes to udp/[66.37.97.53]:5064 at 19:28:12.445259:
>   ------------------------------------------------------------------------
>   SIP/2.0 401 Unauthorized
>   Via: SIP/2.0/UDP 10.93.14.119:5064
> ;branch=z9hG4bK-e73de4bf;received=66.37.97.53
>   From: "1002" <sip:1002 at 65.162.239.204>;tag=7718f2af7f3dcc73o2
>   To: "1002" <sip:1002 at 65.162.239.204>;tag=t1m9emDKB4trF
>   Call-ID: 930c7e31-648ae3db at 10.93.14.119
>   CSeq: 53622 REGISTER
>   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-25c725c 2011-06-30 18-30-24
> -0500
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>   WWW-Authenticate: Digest realm="65.162.239.204",
> nonce="7e426985-2f7a-40ff-a817-6c167207853f", algorithm=MD5, qop="auth"
>   Content-Length: 0
>
>
>
> _______________________________________________
> 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 max.asterisk at gmail.com  Sat Jul  2 08:02:24 2011
From: max.asterisk at gmail.com (Max Alex)
Date: Sat, 2 Jul 2011 09:32:24 +0530
Subject: [Freeswitch-users] Answer issue on inbound call
In-Reply-To: <BANLkTinvomsqyR3JMn0Z58Hoq4XJx=oRPw@mail.gmail.com>
References: <BANLkTin+f=uDMiOouyLWAorPvNTd1Z6xaA@mail.gmail.com>
	<BANLkTim=TOQD1tUvmHjeRuhAR-eKkGYgsw@mail.gmail.com>
	<BANLkTim3pTPV804t_Usk59eG2C56HAyFdA@mail.gmail.com>
	<BANLkTi=Q3qRBCDRARDWp15=Guwyq-jtNLQ@mail.gmail.com>
	<BANLkTinMokGeJU+wVVA0U-oxB97z+c+ZGg@mail.gmail.com>
	<BANLkTinEp1Cu5mB_UHw-Y+mitGsNByR3sQ@mail.gmail.com>
	<BANLkTikcWDbCkTPRow-TMqPN=D3FV+LjWw@mail.gmail.com>
	<BANLkTinE1292_p_VZd+D4N6vvEEot8LXRw@mail.gmail.com>
	<BANLkTin5zjPmdaV6Qa9ZM8dYtBGBUmcKOg@mail.gmail.com>
	<BANLkTim5jOswqjM9DwpH99UpjCm5MmVXjw@mail.gmail.com>
	<BANLkTinvomsqyR3JMn0Z58Hoq4XJx=oRPw@mail.gmail.com>
Message-ID: <BANLkTikaZHrsahdPXCDoE7t-g1wVTP2OGw@mail.gmail.com>

Hello,
Thanks for reply.
The current issue is when the call is ringing,
The sound file of voicemail comes up when i am rejecting phone, and that we
dont have to check for now.
We have to check for the preanswer issue of call when it is ringing to
phone.

Waiting for your reply.

Thanks,
Max Alex
Voip Developer



On Fri, Jul 1, 2011 at 9:51 PM, Michael Collins <msc at freeswitch.org> wrote:

> What's with the errors on the voicemail sound files?
> -MC
>
>
> On Thu, Jun 30, 2011 at 9:12 PM, Max Alex <max.asterisk at gmail.com> wrote:
>
>> Hi,
>> Can any one provide their suggestions and help for this issue,
>> I really need to resolve and get this working.
>>
>> Thanks,
>> Max Alex
>> Voip Developer
>>
>>
>>
>> On Tue, Jun 28, 2011 at 6:01 PM, Max Alex <max.asterisk at gmail.com> wrote:
>>
>>> Hi,
>>> Thanks for your reply.
>>> I have enabled logger as per your help.
>>> I have given completed log on following link of pastebin.
>>> http://pastebin.freeswitch.org/16616
>>>
>>> You can see this line as it is showing answered and this call is answered
>>> on my cell phone but on softphone it is ringing and i have rejected from
>>> there.
>>> 2011-06-28 17:47:52.909657 [NOTICE] mod_freetdm.c:1953 Channel [FreeTDM/
>>> 2:1/01234567890] has been answered
>>>
>>> It is pre answering the cell phone when it is ringing on phone
>>> Waiting for your help.
>>>
>>> Thanks,
>>> Max Alex
>>> Voip Developer
>>>
>>>
>>>
>>> On Mon, Jun 27, 2011 at 9:15 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Get a complete, unedited, unabridged console debug log w/ siptrace and
>>>> put it on pastebin w/ "FreeSWITCH Log" for the syntax highlighting. Use
>>>> "sofia global siptrace on" to make sure you can all SIP traffic.
>>>>
>>>> -MC
>>>>
>>>>
>>>> On Mon, Jun 27, 2011 at 6:05 AM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>
>>>>> Hi,
>>>>> Thanks for reply,
>>>>> I have tried the same way and reloaded freeswitch, but still it is
>>>>> answered on first ring of the call.
>>>>> When it is ringing the call on 1001 and the same time it is answered on
>>>>> cell phone, so something is done when it is ringing on 1001.
>>>>>
>>>>> Here is the dialplan for the same
>>>>>
>>>>>     <extension name="Local_Extension">
>>>>>       <condition field="destination_number"
>>>>> expression="^(10[01][0-9])$">
>>>>>         <action application="set" data="dialed_extension=$1"/>
>>>>>         <action application="export" data="dialed_extension=$1"/>
>>>>>         <action application="bind_meta_app" data="2 b s
>>>>> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>>>>>         <action application="bind_meta_app" data="3 b s
>>>>> execute_extension::cf XML features"/>-->
>>>>>         <action application="set" data="ringback=${us-ring}"/>
>>>>>         <action application="set"
>>>>> data="transfer_ringback=$${hold_music}"/>
>>>>>         <action application="set" data="call_timeout=30"/>
>>>>>           <action application="set" data="hangup_after_bridge=true"/>
>>>>>          <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="set"
>>>>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>>>>> var callgroup)}"/>
>>>>>         <action application="hash"
>>>>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>>>>>
>>>>>        <action application="bridge"
>>>>> data="{ignore_early_media=true}user/${dialed_extension}@
>>>>> ${domain_name}"/>
>>>>>         <action application="sleep" data="1000"/>
>>>>>         <action application="voicemail" data="default ${domain_name}
>>>>> ${dialed_extension}"/>
>>>>>       </condition>
>>>>>     </extension>
>>>>>
>>>>>
>>>>> Please help me for this issue.
>>>>>
>>>>>
>>>>> Thanks,
>>>>> Max Alex
>>>>> Voip Developer
>>>>>
>>>>>
>>>>>
>>>>> On Fri, Jun 24, 2011 at 11:38 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> Are you using the default dialplan? I think you might just need to
>>>>>> ignore early media on your bridge app. If you are using the default.xml file
>>>>>> then locate "Local_Extension" and the bridge line:
>>>>>>
>>>>>>     <action application="bridge" data="user/${dialed_extension}@
>>>>>> ${domain_name}"/>
>>>>>>
>>>>>> Change it to this, then try again:
>>>>>>
>>>>>>     <action application="bridge"
>>>>>> data="{ignore_early_media=true}user/${dialed_extension}@
>>>>>> ${domain_name}"/>
>>>>>>
>>>>>> If I understand correctly, the "symptom" you are experiencing is the
>>>>>> normal operation of the bridge app (and it's cousin, the originate API
>>>>>> command). When the b-leg sends back media, including ringing, the bridge (or
>>>>>> the originate) will be considered "successful," and in the case of bridge,
>>>>>> the b-leg's audio (the early media) will be connected to the a-leg. If you
>>>>>> set ignore_early_media=true then you are explicitly telling the bridge app
>>>>>> that you only want to connect the b-leg to the a-leg if the b-leg actually
>>>>>> answers.
>>>>>>
>>>>>> I hope that made sense...
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, Jun 23, 2011 at 9:32 PM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>> Thanks for reply.
>>>>>>> Current scenario is below.
>>>>>>>
>>>>>>> PSTN call -> sangoma device -> freeswitch incoming context -> routed
>>>>>>> to 1001 -> ringing (Answered on cellphone)
>>>>>>> Here when it is routed to 1001 the call it is started ringing, but on
>>>>>>> phone that call is answered and hearding sound of ringing.
>>>>>>>
>>>>>>> Required flow:
>>>>>>> PSTN call -> sangoma device -> freeswitch incoming context -> routed
>>>>>>> to 1001 -> ringing (Ringing on cellphone)
>>>>>>>
>>>>>>> I hope it is understable, the call should not be answered until 1001
>>>>>>> answer it, right not when it is started ring it is answered on cell phone.
>>>>>>> that should not be happen as it is not answered yet.
>>>>>>>
>>>>>>> Waiting for your reply.
>>>>>>>
>>>>>>>
>>>>>>> Thanks,
>>>>>>> Max Alex
>>>>>>> Voip Developer
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Fri, Jun 24, 2011 at 5:48 AM, Michael Collins <msc at freeswitch.org
>>>>>>> > wrote:
>>>>>>>
>>>>>>>> I'm not sure I understand the problem. What is happening vs. what
>>>>>>>> you believe should be happening?
>>>>>>>> -MC
>>>>>>>>
>>>>>>>>
>>>>>>>> On Thu, Jun 23, 2011 at 3:31 AM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>>>>>
>>>>>>>>> Hi,
>>>>>>>>> Thanks for your reply.
>>>>>>>>> Here is my configuration and log
>>>>>>>>> http://pastebin.freeswitch.org/16571
>>>>>>>>>
>>>>>>>>> I am using A200 analog sangoma device with freeswitch, it is
>>>>>>>>> working fine but when it is routing call to 1001 then it is answered.
>>>>>>>>> Please provider your suggestions.
>>>>>>>>>
>>>>>>>>> Thanks,
>>>>>>>>> Max Alex
>>>>>>>>> Voip Developer
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Wed, Jun 22, 2011 at 8:51 PM, Michael Collins <
>>>>>>>>> msc at freeswitch.org> wrote:
>>>>>>>>>
>>>>>>>>>> I thought the A200 was an analog card? Maybe I have my numbers
>>>>>>>>>> mixed up...
>>>>>>>>>>
>>>>>>>>>> Go ahead and collect a debug log of this call. It might help to
>>>>>>>>>> have your configs posted as well. Use pastebin.freeswitch.org.
>>>>>>>>>> See this wiki article for tips on how to collect information:
>>>>>>>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>>>>>>>>
>>>>>>>>>> -MC
>>>>>>>>>>
>>>>>>>>>> On Wed, Jun 22, 2011 at 3:23 AM, Max Alex <max.asterisk at gmail.com
>>>>>>>>>> > wrote:
>>>>>>>>>>
>>>>>>>>>>> Hi,
>>>>>>>>>>> I have installed freeswitch latest version with sangoma card A200
>>>>>>>>>>> as well,
>>>>>>>>>>> I have installed and configured freetdm module with wanpipe
>>>>>>>>>>> drivers for freeswitch,
>>>>>>>>>>> We are properly receiving the inbound calls in public context and
>>>>>>>>>>> then we are routing that call to 1001 extension,
>>>>>>>>>>> it is properly routing to 1001 as well, but we have one issue
>>>>>>>>>>> while routing on 1001.
>>>>>>>>>>>
>>>>>>>>>>> Here is the issue description.
>>>>>>>>>>> I am calling from my cell phone to that DID number of pri line,
>>>>>>>>>>> and then it will start ringing on 1001 extension,
>>>>>>>>>>> When 1001 extension start ringing the call is answered on my cell
>>>>>>>>>>> phone,
>>>>>>>>>>> it is something like freeswitch preanswer or autoanswer the call,
>>>>>>>>>>> how can i stop this answer call when it is ringing on 1001 extension,
>>>>>>>>>>> Waiting for good reply.
>>>>>>>>>>>
>>>>>>>>>>> Thanks,
>>>>>>>>>>> Max Alex
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> 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
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> 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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> 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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> 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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> 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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 steveu at coppice.org  Sat Jul  2 08:24:23 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 02 Jul 2011 12:24:23 +0800
Subject: [Freeswitch-users] change sounds file format of voicemail
In-Reply-To: <436BD03F7958410D97EFB7D79002E3FE@stor1>
References: <89F3619D8E6F437F82F3B1C00FA69725@e1705><EAC9775B572D499CA7869A2FA228C517@stor1>	<4E0DD7E4.9050102@coppice.org>
	<436BD03F7958410D97EFB7D79002E3FE@stor1>
Message-ID: <4E0E9D77.6030304@coppice.org>

On 07/02/2011 03:59 AM, Kris wrote:
> There are a lot of Dialogics out there and I just needed compativility with
> it's 8 bit wav and 4bit vox and I will submit it to the GIT so others who
> need that compatibility can use it. When I first started with 2 phone lines,
> I had a 286 with 40mb disk. When you have a few thousand files you can copy,
> convert and not worry about it. When you have a million, not so fast... I
> want to have both Dialogic and FS running side by side and sharing messages.
> I am just not a fan of converting gigabytes of files, then find that I am up
> the creek without a paddle-something goes wrong..I wanted to rewrite my
> program in a modern language like C#, instead of continuing with ancient
> C,that  I started with 20 years ago. 16 bit wav, do sound better at 4x the
> storage.Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
What on Earth do you have that uses 8 bit wav files?

Steve



From vetali100 at gmail.com  Sat Jul  2 09:34:40 2011
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Sat, 2 Jul 2011 08:34:40 +0300
Subject: [Freeswitch-users] REGISTER response sent to the wrong por
In-Reply-To: <A66F955C-EA7C-4355-83C1-38FF53011F44@dchorton.com>
References: <A66F955C-EA7C-4355-83C1-38FF53011F44@dchorton.com>
Message-ID: <CACc3D0Ugad026W_ztpNpLvw_8x8furwB=2J6_GZ05715gsXncg@mail.gmail.com>

You should set the following parameter in your internal.xml profile:

<param name="NDLB-force-rport" value="true"/>

In this case FS will send responses directly to the port from which it
received REGISTER and other SIP messages.

Regards,
Vitalie

2011/7/1 Dave Horton <dave at dchorton.com>

>
> I have a phone behind a firewall registering with the internal profile.
>  The address in the Via header is not reachable, and the FS (correctly, IMO)
> sends the response back to the actual sending IP address.  However, rather
> than sending it to the actual sending udp port, it incorrectly (again, IMO)
> sends it to the port advertised in the Via header.  Of course the message
> does not get through the firewall, and the phone keeps retransmitting the
> REGISTER.
>
> Is this a bug, or am I missing something?  And is there a setting where I
> can force the behavior that I want (which is: send responses to the sending
> address:port if the sending address:port is different than the Via header).
>
> Trace below..
>
>
> recv 481 bytes from udp/[66.37.97.53]:1024 at 19:28:12.445081:
>   ------------------------------------------------------------------------
>   REGISTER sip:65.162.239.204 SIP/2.0
>   Via: SIP/2.0/UDP 10.93.14.119:5064;branch=z9hG4bK-e73de4bf
>   From: "1002" <sip:1002 at 65.162.239.204>;tag=7718f2af7f3dcc73o2
>   To: "1002" <sip:1002 at 65.162.239.204>
>   Call-ID: 930c7e31-648ae3db at 10.93.14.119
>   CSeq: 53622 REGISTER
>   Max-Forwards: 70
>   Contact: "1002" <sip:1002 at 10.93.14.119:5064>;expires=3600
>   User-Agent: Linksys/SPA962-5.2.8(SC)
>   Content-Length: 0
>   Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>   Supported: replaces
>
>   ------------------------------------------------------------------------
> send 671 bytes to udp/[66.37.97.53]:5064 at 19:28:12.445259:
>   ------------------------------------------------------------------------
>   SIP/2.0 401 Unauthorized
>   Via: SIP/2.0/UDP 10.93.14.119:5064
> ;branch=z9hG4bK-e73de4bf;received=66.37.97.53
>   From: "1002" <sip:1002 at 65.162.239.204>;tag=7718f2af7f3dcc73o2
>   To: "1002" <sip:1002 at 65.162.239.204>;tag=t1m9emDKB4trF
>   Call-ID: 930c7e31-648ae3db at 10.93.14.119
>   CSeq: 53622 REGISTER
>   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-25c725c 2011-06-30 18-30-24
> -0500
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>   WWW-Authenticate: Digest realm="65.162.239.204",
> nonce="7e426985-2f7a-40ff-a817-6c167207853f", algorithm=MD5, qop="auth"
>   Content-Length: 0
>
>
>
> _______________________________________________
> 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 ankitwalia4u at gmail.com  Sat Jul  2 14:33:54 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Sat, 2 Jul 2011 16:03:54 +0530
Subject: [Freeswitch-users] LUA Script: IGNORE INFO DTMF
Message-ID: <CAOcuKuepq8bukbYceaQ2Vg1B=GMyPyeEMQQY-tdf-sk=DXJMWQ@mail.gmail.com>

Hi,

I am trying to get the digits pressed using statement in my LUA script
"digits = session:playAndGetDigits(3, 5, 3, 7000, "#", prompt, invalid,
"\\d+")"

The FS is not able to take the digits and giving the invalid response
always.

2011-07-02 15:51:30.405443 [WARNING] sofia.c:6715 IGNORE INFO DTMF(7) (This
channel was not configured to use INFO DTMF!)

I am trying establish a channel between default 1002 ext and custom ext.

Am I missing something?

Thanks
Ankit

-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From david.ponzone at ipeva.fr  Sat Jul  2 14:47:05 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sat, 2 Jul 2011 12:47:05 +0200
Subject: [Freeswitch-users] REGISTER response sent to the wrong por
In-Reply-To: <A66F955C-EA7C-4355-83C1-38FF53011F44@dchorton.com>
References: <A66F955C-EA7C-4355-83C1-38FF53011F44@dchorton.com>
Message-ID: <B4C403B8-FC09-43C5-982B-665004DE2380@ipeva.fr>

Dave,

Vitalie already gave you the fix on the FS side, but you can probably fix that on the phone.
I don't know the SPA962, but if the software is anywhere close to the SPA2102's, you may be able to configure the phone to send the rport paramater in the Via field.

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

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

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




Le 01/07/2011 ? 21:37, Dave Horton a ?crit :

> 
> I have a phone behind a firewall registering with the internal profile.  The address in the Via header is not reachable, and the FS (correctly, IMO) sends the response back to the actual sending IP address.  However, rather than sending it to the actual sending udp port, it incorrectly (again, IMO) sends it to the port advertised in the Via header.  Of course the message does not get through the firewall, and the phone keeps retransmitting the REGISTER.
> 
> Is this a bug, or am I missing something?  And is there a setting where I can force the behavior that I want (which is: send responses to the sending address:port if the sending address:port is different than the Via header).
> 
> Trace below..
> 
> 
> recv 481 bytes from udp/[66.37.97.53]:1024 at 19:28:12.445081:
>   ------------------------------------------------------------------------
>   REGISTER sip:65.162.239.204 SIP/2.0
>   Via: SIP/2.0/UDP 10.93.14.119:5064;branch=z9hG4bK-e73de4bf
>   From: "1002" <sip:1002 at 65.162.239.204>;tag=7718f2af7f3dcc73o2
>   To: "1002" <sip:1002 at 65.162.239.204>
>   Call-ID: 930c7e31-648ae3db at 10.93.14.119
>   CSeq: 53622 REGISTER
>   Max-Forwards: 70
>   Contact: "1002" <sip:1002 at 10.93.14.119:5064>;expires=3600
>   User-Agent: Linksys/SPA962-5.2.8(SC)
>   Content-Length: 0
>   Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>   Supported: replaces
> 
>   ------------------------------------------------------------------------
> send 671 bytes to udp/[66.37.97.53]:5064 at 19:28:12.445259:
>   ------------------------------------------------------------------------
>   SIP/2.0 401 Unauthorized
>   Via: SIP/2.0/UDP 10.93.14.119:5064;branch=z9hG4bK-e73de4bf;received=66.37.97.53
>   From: "1002" <sip:1002 at 65.162.239.204>;tag=7718f2af7f3dcc73o2
>   To: "1002" <sip:1002 at 65.162.239.204>;tag=t1m9emDKB4trF
>   Call-ID: 930c7e31-648ae3db at 10.93.14.119
>   CSeq: 53622 REGISTER
>   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-25c725c 2011-06-30 18-30-24 -0500
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>   WWW-Authenticate: Digest realm="65.162.239.204", nonce="7e426985-2f7a-40ff-a817-6c167207853f", algorithm=MD5, qop="auth"
>   Content-Length: 0
> 
> 
> 
> _______________________________________________
> 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 ankitwalia4u at gmail.com  Sat Jul  2 16:18:33 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Sat, 2 Jul 2011 17:48:33 +0530
Subject: [Freeswitch-users] LUA Script: IGNORE INFO DTMF
In-Reply-To: <CAOcuKuepq8bukbYceaQ2Vg1B=GMyPyeEMQQY-tdf-sk=DXJMWQ@mail.gmail.com>
References: <CAOcuKuepq8bukbYceaQ2Vg1B=GMyPyeEMQQY-tdf-sk=DXJMWQ@mail.gmail.com>
Message-ID: <CAOcuKue5HY-vA=oEH2bK-s=rn1uewxSUGVBp2-H06yMEeKTZeg@mail.gmail.com>

When I called from other system on LAN with windows and X-Lite. It worked
but when I was trying from SFLPhone on FS Server itself. It was not working.


On Sat, Jul 2, 2011 at 4:03 PM, ankIT WALiA <ankitwalia4u at gmail.com> wrote:

> Hi,
>
> I am trying to get the digits pressed using statement in my LUA script
> "digits = session:playAndGetDigits(3, 5, 3, 7000, "#", prompt, invalid,
> "\\d+")"
>
> The FS is not able to take the digits and giving the invalid response
> always.
>
> 2011-07-02 15:51:30.405443 [WARNING] sofia.c:6715 IGNORE INFO DTMF(7) (This
> channel was not configured to use INFO DTMF!)
>
> I am trying establish a channel between default 1002 ext and custom ext.
>
> Am I missing something?
>
> Thanks
> Ankit
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>



-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From monemran at gmail.com  Sat Jul  2 19:51:44 2011
From: monemran at gmail.com (Mohammad Emran)
Date: Sat, 2 Jul 2011 21:51:44 +0600
Subject: [Freeswitch-users] Linksys PAP2 tls with freeswitch
Message-ID: <DBBCBFC9-BB1C-4412-B40C-070BC8BD11C9@gmail.com>

Hi all,
Please help me to configure linksys pap2 tls with freeswitch.i have configured freeswitch as per wiki.

Thank you.

Sent from my iPad

From gmaruzz at gmail.com  Sat Jul  2 21:09:02 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 2 Jul 2011 19:09:02 +0200
Subject: [Freeswitch-users] LUA Script: IGNORE INFO DTMF
In-Reply-To: <CAOcuKuepq8bukbYceaQ2Vg1B=GMyPyeEMQQY-tdf-sk=DXJMWQ@mail.gmail.com>
References: <CAOcuKuepq8bukbYceaQ2Vg1B=GMyPyeEMQQY-tdf-sk=DXJMWQ@mail.gmail.com>
Message-ID: <CALXCt0oGLbL82-8fKU+Mv4Bta7Adr1Z1xEycxH6k9LEC2Q8zFw@mail.gmail.com>

You have the client sending dtmf via "info" and the server is not
listening to it (is probably listening to 2833 or inband).
So, setup the client to send dtmf via 2833.
For all dtmf questions, search for it in the wiki.
-giovanni

On 7/2/11, ankIT WALiA <ankitwalia4u at gmail.com> wrote:
> Hi,
>
> I am trying to get the digits pressed using statement in my LUA script
> "digits = session:playAndGetDigits(3, 5, 3, 7000, "#", prompt, invalid,
> "\\d+")"
>
> The FS is not able to take the digits and giving the invalid response
> always.
>
> 2011-07-02 15:51:30.405443 [WARNING] sofia.c:6715 IGNORE INFO DTMF(7) (This
> channel was not configured to use INFO DTMF!)
>
> I am trying establish a channel between default 1002 ext and custom ext.
>
> Am I missing something?
>
> Thanks
> Ankit
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From avi at avimarcus.net  Sat Jul  2 21:56:15 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 2 Jul 2011 20:56:15 +0300
Subject: [Freeswitch-users] Inband DTMF set in Event Socket
In-Reply-To: <BANLkTikkNSr9wf-SfhNQBW6ERH+sj=z=8w@mail.gmail.com>
References: <1309524076282-6537620.post@n2.nabble.com>
	<BANLkTinHT_o-_ytPr7gjL8hAoSzjzh3-7A@mail.gmail.com>
	<1309531579711-6538036.post@n2.nabble.com>
	<BANLkTinmbb5dwN64EjFe+vP50nxR5-C5rw@mail.gmail.com>
	<1309535289627-6538248.post@n2.nabble.com>
	<BANLkTikkNSr9wf-SfhNQBW6ERH+sj=z=8w@mail.gmail.com>
Message-ID: <CALNZuwk+g5SkTT76FbNxYkP5iWmfKMUx7rJmJ6+ddYtBwJjoaw@mail.gmail.com>

Oh woops. I saw:


<action application="start_dtmf_generate" data="true"/>

The "true" and my mind just jumped to variable. So I presume you don't need
the data=true, and that should be removed from the
wiki<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate>
?
-Avi


On Fri, Jul 1, 2011 at 7:07 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Jul 1, 2011 at 8:48 AM, rex.alex <rajesh.npnr at yahoo.com> wrote:
>
>> similar way I tried both
>>
>> api originate {origination_caller_id_name='XXX',
>> start_dtmf_generate=true}user/1000
>> &bridge({accountcode='XXX'}sofia/gateway/gw_name/phno)
>>
>> and
>>
>> api originate {origination_caller_id_name='XXX'}user/1000
>> &bridge({accountcode='XXX',
>> start_dtmf_generate=true}sofia/gateway/gw_name/phno)
>>
>>
> it's not working, but the inband type is working when i use it in dialplan.
>>
>> Please assist.
>>
>> Which leg needs to have inband DTMF - the calling or the called?
>
> You can use these variables to do a pre-execute on either leg:
> http://wiki.freeswitch.org/wiki/Variable_bridge_pre_execute_aleg_app
> http://wiki.freeswitch.org/wiki/Variable_bridge_pre_execute_aleg_data
> http://wiki.freeswitch.org/wiki/Variable_bridge_pre_execute_bleg_app
> http://wiki.freeswitch.org/wiki/Variable_bridge_pre_execute_bleg_data
>
> start_dtmf_generate is a dialplan app, not a chan var, so it needs to be
> executed, not "set".
>
> -MC
>
> _______________________________________________
> 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 rzhang at gosilverplus.com  Sun Jul  3 03:02:40 2011
From: rzhang at gosilverplus.com (ran zhang)
Date: Sat, 02 Jul 2011 16:02:40 -0700
Subject: [Freeswitch-users] how to change conference's caller-id-number at
	run-time?
Message-ID: <4E0FA390.40002@gosilverplus.com>

I want when the bridging conference is inviting someone to start the 
conference, the caller is shown as the real caller id number rather than 
whats stored in 'caller-id-number' in conference.conf.xml.






From zhongxiang721 at gmail.com  Sun Jul  3 05:06:31 2011
From: zhongxiang721 at gmail.com (salzh)
Date: Sun, 3 Jul 2011 09:06:31 +0800
Subject: [Freeswitch-users] get answertime dialstatus after bridge
Message-ID: <CAM3SFgDg+rinmVT7-geLxNPxvVGzNF=CzgfafUuGfB+OGPEp=w@mail.gmail.com>

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


From jan.berger at video24.no  Sun Jul  3 13:58:32 2011
From: jan.berger at video24.no (Jan Berger)
Date: Sun, 3 Jul 2011 11:58:32 +0200
Subject: [Freeswitch-users] MSVC 2010 Compile broken
Message-ID: <B719E25702AB40F0977C2624C12A8E36@dell9400>

Hi,

 

I just downloaded the latest FreeSWITCH and I have not been able to compile
it with Visual Studio 2010 so far.

 

Firstly - opening FreeSWITCH.2010.sln gives an error - a missing project.
Ignoring that and switching off spandsp I still end up with compiler errors
in Sofia.

 

Opening and running FreeSWITCH.2010.express.sln or converting
FreeSWITCH.2008.sln gives the same result - Sofia is the first that don't
compile.

 

The actual error I get is that http_host_class is undefined and checking
with the original Sofia I notice that http_protos.h that should have
contained this is empty ???

 

I assume it's a build step that don't work or have dropped out - any ideas?

 

Jan

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From member at linkedin.com  Sun Jul  3 14:00:20 2011
From: member at linkedin.com (Jacob Maldonado via LinkedIn)
Date: Sun, 3 Jul 2011 10:00:20 +0000 (UTC)
Subject: [Freeswitch-users] Invitation to connect on LinkedIn
Message-ID: <1246747267.6647999.1309687220243.JavaMail.app@ela4-bed77.prod>

LinkedIn
------------




    Jacob Maldonado requested to add you as a connection on LinkedIn:
  
------------------------------------------

Zohair,

I'd like to add you to my professional network on LinkedIn.

- Jacob

Accept invitation from Jacob Maldonado
http://www.linkedin.com/e/xbphn8-gpntyeq8-3q/vPtmrrfmcGvVxWv84eLqdl0FlSkzdWO84S6qfAKHOSUhfgvFvWjMOLQ/blk/I1484429223_3/1BpC5vrmRLoRZcjkkZt5YCpnlOt3RApnhMpmdzgmhxrSNBszYPnPcOczAOd3gUd359bTpKlnoTckFUbPoSd34VejgNd3cLrCBxbOYWrSlI/EML_comm_afe/

View invitation from Jacob Maldonado
http://www.linkedin.com/e/xbphn8-gpntyeq8-3q/vPtmrrfmcGvVxWv84eLqdl0FlSkzdWO84S6qfAKHOSUhfgvFvWjMOLQ/blk/I1484429223_3/3dvcP8Oej8Qd3wQckALqnpPbOYWrSlI/svi/ 
------------------------------------------

DID YOU KNOW LinkedIn can help you find the right service providers using recommendations from your trusted network? Using LinkedIn Services, you can take the risky guesswork out of selecting service providers by reading the recommendations of credible, trustworthy members of your network. 
http://www.linkedin.com/e/xbphn8-gpntyeq8-3q/svp/inv-25/

 
-- 
(c) 2011, LinkedIn Corporation
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From peter.olsson at visionutveckling.se  Sun Jul  3 15:43:39 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 3 Jul 2011 13:43:39 +0200
Subject: [Freeswitch-users] MSVC 2010 Compile broken
In-Reply-To: <B719E25702AB40F0977C2624C12A8E36@dell9400>
References: <B719E25702AB40F0977C2624C12A8E36@dell9400>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F64@cooper>

Make sure to configure git NOT to use autocrlf (autocrlf = false).

Follow the instructions here: http://wiki.freeswitch.org/wiki/Installation_for_Windows

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Jan Berger [jan.berger at video24.no]
Skickat: den 3 juli 2011 11:58
Till: 'FreeSWITCH Users Help'
?mne: [Freeswitch-users] MSVC 2010 Compile broken

Hi,

I just downloaded the latest FreeSWITCH and I have not been able to compile it with Visual Studio 2010 so far.

Firstly ? opening FreeSWITCH.2010.sln gives an error ? a missing project. Ignoring that and switching off spandsp I still end up with compiler errors in Sofia.

Opening and running FreeSWITCH.2010.express.sln or converting FreeSWITCH.2008.sln gives the same result ? Sofia is the first that don?t compile.

The actual error I get is that http_host_class is undefined and checking with the original Sofia I notice that http_protos.h that should have contained this is empty ???

I assume it?s a build step that don?t work or have dropped out ? any ideas?

Jan
!DSPAM:4e103e8e32761387656113!


From jan.berger at video24.no  Sun Jul  3 16:39:27 2011
From: jan.berger at video24.no (Jan Berger)
Date: Sun, 3 Jul 2011 14:39:27 +0200
Subject: [Freeswitch-users] MSVC 2010 Compile broken
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F64@cooper>
References: <B719E25702AB40F0977C2624C12A8E36@dell9400>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F64@cooper>
Message-ID: <56E60EA286E147528E1A89D1ABC5799C@dell9400>

Thanks - this was on a new machine so autocrlf was not switched off and this
fixed it. Incredible how much confusion EOL sequences create :(

It still complains about a win32 setup project that is in the wrong format -
but just ignoring this and hitting F5 a few times worked - some of the
download processes fails sometimes.

Jan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter
Olsson
Sent: 3. juli 2011 13:44
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] MSVC 2010 Compile broken

Make sure to configure git NOT to use autocrlf (autocrlf = false).

Follow the instructions here:
http://wiki.freeswitch.org/wiki/Installation_for_Windows

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Jan Berger
[jan.berger at video24.no]
Skickat: den 3 juli 2011 11:58
Till: 'FreeSWITCH Users Help'
?mne: [Freeswitch-users] MSVC 2010 Compile broken

Hi,

I just downloaded the latest FreeSWITCH and I have not been able to compile
it with Visual Studio 2010 so far.

Firstly ? opening FreeSWITCH.2010.sln gives an error ? a missing project.
Ignoring that and switching off spandsp I still end up with compiler errors
in Sofia.

Opening and running FreeSWITCH.2010.express.sln or converting
FreeSWITCH.2008.sln gives the same result ? Sofia is the first that don?t
compile.

The actual error I get is that http_host_class is undefined and checking
with the original Sofia I notice that http_protos.h that should have
contained this is empty ???

I assume it?s a build step that don?t work or have dropped out ? any ideas?

Jan
!DSPAM:4e103e8e32761387656113!

_______________________________________________
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 david.villasmil.work at gmail.com  Mon Jul  4 17:12:34 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 4 Jul 2011 15:12:34 +0200
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
Message-ID: <CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>

Hello All,

I think I'm as ready as i can be to publish this...
Can someone guide me into publishing via GIT?

thanks

David

On Thu, Feb 24, 2011 at 12:39 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello Guys,
>
> I'm finishing a "complete" wholesale application created on freeswitch and
> I was wondering whether it would be a good idea to put it up on the wiki. I
> just don't know how.
>
> Features include all the following parameters configurable via web
> interface:
>
> - Profile creation based on server IP where traffic is received. You can
> have multiple IPs, system will create multiple profiles/diaplans so it can
> differentiate.
>      - i.e. offer to the same customer a "gold" routing on IP1 and cheap
> routing on IP2
>
> - Customer add/modify/delete
>      - IP source
>      - Rates for client routes based on areacode
>      - Prepaid or postpaid.
>           - When cutomer balance is 0, no more calls are allowed.
>      - limit max channels
>      - Media by-pass
>           - When by-passed, customer and provider will exchange RTPs
> directly. Else, server will be in the middle.
>
> - Provider add/modify/delete
>      - costs for provider routes based on areacode
>      - limit max channels
>
> - Routing based on areacode, gives great granularity.
>
> - Routes can be assigned multiple gateways/providers which can in turn be
> distributed based on weigth. Includes overflow to next configured GW.
>
> - Basic financial report generation (totals) by customer/provider
>
> - Basic traffic ASR/ACD report (totals) by cutomer/provider
>
> - Basic user administration. (No access level, only total access)
>
> - CDR export to csv file.
>
>
>
>
> I also have a prepaid card app... no web interface on that one though...
>
> Thanks all
>
>
> David
>
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From monemran at gmail.com  Mon Jul  4 21:10:52 2011
From: monemran at gmail.com (Mohammad Emran)
Date: Mon, 4 Jul 2011 23:10:52 +0600
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
Message-ID: <163DDA0D-0B14-4623-B48B-5E9EEE907AED@gmail.com>

We can put on google code or git hub.

Sent from my iPad

On 4 Jul 2011, at 19:12, David Villasmil <david.villasmil.work at gmail.com> wrote:

> Hello All,
> 
> I think I'm as ready as i can be to publish this...
> Can someone guide me into publishing via GIT?
> 
> thanks
> 
> David
> 
> On Thu, Feb 24, 2011 at 12:39 PM, David Villasmil <david.villasmil.work at gmail.com> wrote:
> Hello Guys,
> 
> I'm finishing a "complete" wholesale application created on freeswitch and I was wondering whether it would be a good idea to put it up on the wiki. I just don't know how.
> 
> Features include all the following parameters configurable via web interface:
> 
> - Profile creation based on server IP where traffic is received. You can have multiple IPs, system will create multiple profiles/diaplans so it can differentiate. 
>      - i.e. offer to the same customer a "gold" routing on IP1 and cheap routing on IP2
> 
> - Customer add/modify/delete
>      - IP source
>      - Rates for client routes based on areacode
>      - Prepaid or postpaid.
>           - When cutomer balance is 0, no more calls are allowed.
>      - limit max channels
>      - Media by-pass
>           - When by-passed, customer and provider will exchange RTPs directly. Else, server will be in the middle.
> 
> - Provider add/modify/delete
>      - costs for provider routes based on areacode
>      - limit max channels
> 
> - Routing based on areacode, gives great granularity.
> 
> - Routes can be assigned multiple gateways/providers which can in turn be distributed based on weigth. Includes overflow to next configured GW.
> 
> - Basic financial report generation (totals) by customer/provider 
> 
> - Basic traffic ASR/ACD report (totals) by cutomer/provider
> 
> - Basic user administration. (No access level, only total access)
> 
> - CDR export to csv file.
> 
> 
> 
> 
> I also have a prepaid card app... no web interface on that one though...
> 
> Thanks all
> 
> 
> David
> 
> _______________________________________________
> 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 hmkias at gmail.com  Mon Jul  4 21:48:49 2011
From: hmkias at gmail.com (hmkias at gmail.com)
Date: Mon, 4 Jul 2011 17:48:49 +0000
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <163DDA0D-0B14-4623-B48B-5E9EEE907AED@gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com><CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com><163DDA0D-0B14-4623-B48B-5E9EEE907AED@gmail.com>
Message-ID: <376546190-1309801622-cardhu_decombobulator_blackberry.rim.net-724271591-@b13.c4.bise7.blackberry>

Great news for the community.  Thanks david.
Sent from BSNL with my BlackBerry? smartphone

-----Original Message-----
From: Mohammad Emran <monemran at gmail.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Mon, 4 Jul 2011 23:10:52 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Cc: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Complete wholesale app in freeswitch

_______________________________________________
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 freeswitch-list at puzzled.xs4all.nl  Mon Jul  4 22:53:26 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Mon, 04 Jul 2011 20:53:26 +0200
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
Message-ID: <4E120C26.70001@puzzled.xs4all.nl>

On 07/04/2011 03:12 PM, David Villasmil wrote:
> Hello All,
>
> I think I'm as ready as i can be to publish this...
> Can someone guide me into publishing via GIT?

You could ask the FreeSWITCH developers to allow you access to your own 
repo in the FreeSWITCH contrib area:

http://fisheye.freeswitch.org/browse/freeswitch-contrib

Regards,
Patrick


From gcd at i.ph  Tue Jul  5 02:07:59 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 5 Jul 2011 06:07:59 +0800
Subject: [Freeswitch-users] Fidelio
Message-ID: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>

hi everybody,

anyone working on interfacing FS with Fidelio Hotel PMS?  i can't find the
FIAS protocol/specs online.  is this freely available?

tks,
nandy
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From bryansmart at bryansmart.com  Tue Jul  5 03:56:06 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Mon, 4 Jul 2011 19:56:06 -0400
Subject: [Freeswitch-users] Skeleton Configuration?
Message-ID: <33B36741-B6BF-4D1B-A6BA-5F54E0C758E8@bryansmart.com>

Is a truly skeletal configuration available? As far as I can tell, the choice right now is to start with no configuration files at all, or else to install a sample set that is bursting with stuff.

So far, I've installed the samples, and tried to strip out the parts that are obviously not needed by me:

The dialplan is the best example that I can think of: all the exts that replace functions on Snom/Linksys phones, record/play video, demo substitute ring tone, call groups, add/del groups, dynamically create numerous types of conferences, call features like redial/hold/etc, the exts for the demo IVR, and so on. I know enough to know that I don't want those on a live system, but I don't know enough to know if it is safe to remove parts. Will I break refer or deflect if I get rid of the "refer" ext? Is "unloop" only necessary as a safety feature if I'm connecting sessions to SIP URIs outside of the local FS? ? Can I drop "external-sip-uri" if users won't be calling external URIs, or will that completely block FS from connecting to external URIs in all cases? I think I can dump features.xml altogether.

There are an internal and external SIP profile. If I'm only hosting a voice application, I think I can dump the internal profile and just run the default external profile on 5060, but there are a huge amount of options in there that look as if they're setup to cover cases where the external profile is used to connect to ITSPs, which I won't.

There are so many configuration files that I'm not comfortable with blowing the whole directory tree away and recreating a raw basics version of each of them. While I don't feel paranoid about a specific part of the security of the full samples install, my rule is always that I try to not run anything that I don't at least somewhat understand. There seem to be so many little demos and optional features that I know that I by no means understand the implications of what they could be used to do in combination with what I'll add later. It seems safer to start from the bare basics, and only add what is needed, but, for now, I must start with the samples, and try to strip away as much as I can without breaking FS.

For me, I'm using FS as a platform for running a voice app, so I need little, if any, PBX type features. Some people that might wish to use FS as a PBX might not want all of the demo bits, though. An absolute minimum set of configuration files, as well as a set containing the essentials for a app platform or PBX would be useful, if they don't already exist somewhere.

Bryan



From msc at freeswitch.org  Tue Jul  5 05:19:07 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Jul 2011 18:19:07 -0700
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <4E120C26.70001@puzzled.xs4all.nl>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
	<4E120C26.70001@puzzled.xs4all.nl>
Message-ID: <CAKzWOxWQUd9Pw9cQOhRotitWvm8Xjn57tVU1tPwsoJEFApadEQ@mail.gmail.com>

Raymond can get you set up with an account on the freeswitch-contrib repo.
We're all back in the office tomorrow so we'll check in then.
-MC

On Mon, Jul 4, 2011 at 11:53 AM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 07/04/2011 03:12 PM, David Villasmil wrote:
> > Hello All,
> >
> > I think I'm as ready as i can be to publish this...
> > Can someone guide me into publishing via GIT?
>
> You could ask the FreeSWITCH developers to allow you access to your own
> repo in the FreeSWITCH contrib area:
>
> http://fisheye.freeswitch.org/browse/freeswitch-contrib
>
> Regards,
> Patrick
>
> _______________________________________________
> 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
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From msc at freeswitch.org  Tue Jul  5 05:29:14 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Jul 2011 18:29:14 -0700
Subject: [Freeswitch-users] Skeleton Configuration?
In-Reply-To: <33B36741-B6BF-4D1B-A6BA-5F54E0C758E8@bryansmart.com>
References: <33B36741-B6BF-4D1B-A6BA-5F54E0C758E8@bryansmart.com>
Message-ID: <CAKzWOxV+usqYJ1L4WEfriF2qnSsWdwJm1E-cZ1RepcXV4Fo2wQ@mail.gmail.com>

I think the best "skeleton" config is Brian West's soft-phone configuration.
It looks like we haven't migrated the download link over to git yet so I
will see if I can't track it down. In the meantime if anyone has a copy of
that config please post it here.

What's nice about it is that it's basically just one file with an option to
read in gateway configs from a subdirectory. It's only a few hundred lines
of XML and it shows what you can do with just a single config file.

-MC

On Mon, Jul 4, 2011 at 4:56 PM, Bryan Smart <bryansmart at bryansmart.com>wrote:

> Is a truly skeletal configuration available? As far as I can tell, the
> choice right now is to start with no configuration files at all, or else to
> install a sample set that is bursting with stuff.
>
> So far, I've installed the samples, and tried to strip out the parts that
> are obviously not needed by me:
>
> The dialplan is the best example that I can think of: all the exts that
> replace functions on Snom/Linksys phones, record/play video, demo substitute
> ring tone, call groups, add/del groups, dynamically create numerous types of
> conferences, call features like redial/hold/etc, the exts for the demo IVR,
> and so on. I know enough to know that I don't want those on a live system,
> but I don't know enough to know if it is safe to remove parts. Will I break
> refer or deflect if I get rid of the "refer" ext? Is "unloop" only necessary
> as a safety feature if I'm connecting sessions to SIP URIs outside of the
> local FS? ? Can I drop "external-sip-uri" if users won't be calling external
> URIs, or will that completely block FS from connecting to external URIs in
> all cases? I think I can dump features.xml altogether.
>
> There are an internal and external SIP profile. If I'm only hosting a voice
> application, I think I can dump the internal profile and just run the
> default external profile on 5060, but there are a huge amount of options in
> there that look as if they're setup to cover cases where the external
> profile is used to connect to ITSPs, which I won't.
>
> There are so many configuration files that I'm not comfortable with blowing
> the whole directory tree away and recreating a raw basics version of each of
> them. While I don't feel paranoid about a specific part of the security of
> the full samples install, my rule is always that I try to not run anything
> that I don't at least somewhat understand. There seem to be so many little
> demos and optional features that I know that I by no means understand the
> implications of what they could be used to do in combination with what I'll
> add later. It seems safer to start from the bare basics, and only add what
> is needed, but, for now, I must start with the samples, and try to strip
> away as much as I can without breaking FS.
>
> For me, I'm using FS as a platform for running a voice app, so I need
> little, if any, PBX type features. Some people that might wish to use FS as
> a PBX might not want all of the demo bits, though. An absolute minimum set
> of configuration files, as well as a set containing the essentials for a app
> platform or PBX would be useful, if they don't already exist somewhere.
>
> Bryan
>
>
> _______________________________________________
> 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 bryansmart at bryansmart.com  Tue Jul  5 05:39:34 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Mon, 4 Jul 2011 21:39:34 -0400
Subject: [Freeswitch-users] Skeleton Configuration?
In-Reply-To: <CAKzWOxV+usqYJ1L4WEfriF2qnSsWdwJm1E-cZ1RepcXV4Fo2wQ@mail.gmail.com>
References: <33B36741-B6BF-4D1B-A6BA-5F54E0C758E8@bryansmart.com>
	<CAKzWOxV+usqYJ1L4WEfriF2qnSsWdwJm1E-cZ1RepcXV4Fo2wQ@mail.gmail.com>
Message-ID: <6C3107DD-ADD0-4A45-8F29-5CCF570EA248@bryansmart.com>

Does it still include conf files for the modules, but with all possible disabled?

For those of you with large deployed voice apps or switches, what do you do? Do you blow away the conf dir and totally start from scratch?

Bryan


On Jul 4, 2011, at 9:29 PM, Michael Collins wrote:

I think the best "skeleton" config is Brian West's soft-phone configuration. It looks like we haven't migrated the download link over to git yet so I will see if I can't track it down. In the meantime if anyone has a copy of that config please post it here.

What's nice about it is that it's basically just one file with an option to read in gateway configs from a subdirectory. It's only a few hundred lines of XML and it shows what you can do with just a single config file.

-MC

On Mon, Jul 4, 2011 at 4:56 PM, Bryan Smart <bryansmart at bryansmart.com<mailto:bryansmart at bryansmart.com>> wrote:
Is a truly skeletal configuration available? As far as I can tell, the choice right now is to start with no configuration files at all, or else to install a sample set that is bursting with stuff.

So far, I've installed the samples, and tried to strip out the parts that are obviously not needed by me:

The dialplan is the best example that I can think of: all the exts that replace functions on Snom/Linksys phones, record/play video, demo substitute ring tone, call groups, add/del groups, dynamically create numerous types of conferences, call features like redial/hold/etc, the exts for the demo IVR, and so on. I know enough to know that I don't want those on a live system, but I don't know enough to know if it is safe to remove parts. Will I break refer or deflect if I get rid of the "refer" ext? Is "unloop" only necessary as a safety feature if I'm connecting sessions to SIP URIs outside of the local FS? ? Can I drop "external-sip-uri" if users won't be calling external URIs, or will that completely block FS from connecting to external URIs in all cases? I think I can dump features.xml altogether.

There are an internal and external SIP profile. If I'm only hosting a voice application, I think I can dump the internal profile and just run the default external profile on 5060, but there are a huge amount of options in there that look as if they're setup to cover cases where the external profile is used to connect to ITSPs, which I won't.

There are so many configuration files that I'm not comfortable with blowing the whole directory tree away and recreating a raw basics version of each of them. While I don't feel paranoid about a specific part of the security of the full samples install, my rule is always that I try to not run anything that I don't at least somewhat understand. There seem to be so many little demos and optional features that I know that I by no means understand the implications of what they could be used to do in combination with what I'll add later. It seems safer to start from the bare basics, and only add what is needed, but, for now, I must start with the samples, and try to strip away as much as I can without breaking FS.

For me, I'm using FS as a platform for running a voice app, so I need little, if any, PBX type features. Some people that might wish to use FS as a PBX might not want all of the demo bits, though. An absolute minimum set of configuration files, as well as a set containing the essentials for a app platform or PBX would be useful, if they don't already exist somewhere.

Bryan


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

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/>

_______________________________________________
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<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 sc_zhangming at sina.com  Tue Jul  5 07:40:35 2011
From: sc_zhangming at sina.com (sc_zhangming)
Date: Tue, 5 Jul 2011 11:40:35 +0800
Subject: [Freeswitch-users] Video is not display
References: <201107041852227187719@sina.com>, <201107051053418751576@sina.com>,
	<201107051131039684717@sina.com>
Message-ID: <201107051140303904616@sina.com>


hi:
if use  fs_cli  originate user/70001  700002  agent is not vedio

if i use x-lite 4 input number the call  have vedio

my dialplan  configuration information:
 <condition field="destination_number" expression="^(7[0-4][0-9][0-9][0-9][0-9])$">
                  <action application="set" data="bypass_media=true"/> 
<action application="bridge" data="user/$1@${domain_name}"/>
                <action application="answer"/>
        </condition>
    </extension>

log:

2011-07-07 19:04:31.427458 [DEBUG] sofia.c:4669 Remote SDP:
v=0
o=- 12954249923203125 3 IN IP4 10.108.226.238
s=CounterPath X-Lite 4.0
c=IN IP4 10.108.226.238
t=0 0
a=ice-ufrag:2e4e41
a=ice-pwd:1af7090a62cdba6f63b4b6bd9fd01472
m=audio 64370 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=candidate:1 1 UDP 659136 10.108.226.238 64370 typ host
a=candidate:1 2 UDP 659134 10.108.226.238 64371 typ host
m=video 63720 RTP/AVP 34
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;VGA=2
a=candidate:1 1 UDP 659136 10.108.226.238 63720 typ host
a=candidate:1 2 UDP 659134 10.108.226.238 63721 typ host

2011-07-07 19:04:31.427458 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-07-07 19:04:31.427458 [DEBUG] sofia_glue.c:2690 Already using PCMU
2011-07-07 19:04:31.427458 [DEBUG] sofia_glue.c:4565 Set 2833 dtmf send payload to 101
2011-07-07 19:04:31.427458 [DEBUG] sofia_glue.c:4626 Video Codec Compare [H263:34]/[H263:34]
2011-07-07 19:04:31.427458 [DEBUG] sofia.c:5121 Processing updated SDP
2011-07-07 19:04:36.870143 [DEBUG] switch_channel.c:2535 (sofia/internal/sip:700001 at 10.108.226.222:39206) Callstate Change ACTIVE -> HANGUP
2011-07-07 19:04:36.870143 [NOTICE] sofia.c:537 Hangup sofia/internal/sip:700001 at 10.108.226.222:39206 [CS_HIBERNATE] [NORMAL_CLEARING]
2011-07-07 19:04:36.870143 [DEBUG] switch_channel.c:2551 Send signal sofia/internal/sip:700001 at 10.108.226.222:39206 [KILL]


sc_zhangming at sina.com.cn
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From david.ponzone at ipeva.fr  Tue Jul  5 11:30:20 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 5 Jul 2011 09:30:20 +0200
Subject: [Freeswitch-users] Skeleton Configuration?
In-Reply-To: <6C3107DD-ADD0-4A45-8F29-5CCF570EA248@bryansmart.com>
References: <33B36741-B6BF-4D1B-A6BA-5F54E0C758E8@bryansmart.com>
	<CAKzWOxV+usqYJ1L4WEfriF2qnSsWdwJm1E-cZ1RepcXV4Fo2wQ@mail.gmail.com>
	<6C3107DD-ADD0-4A45-8F29-5CCF570EA248@bryansmart.com>
Message-ID: <9DA2BEC9-6FDC-4E11-82FA-98CA595EBBB1@ipeva.fr>

Personally, I patched the default conf, one step at a time. The good thing with that is that it forces you to understand most parameters, which is always good thing (who would want to fly a plane, not knowing what use is the little yellow button, just there).
In a specific file, if you are not sure about a parameter, just remove it, and see if your app still works, but most default parameters do no harm.
For modules, the safe path is to disable all the useless ones, so you don't really care about the default conf.

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

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

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




Le 05/07/2011 ? 03:39, Bryan Smart a ?crit :

> Does it still include conf files for the modules, but with all possible disabled?
> 
> For those of you with large deployed voice apps or switches, what do you do? Do you blow away the conf dir and totally start from scratch?
> 
> Bryan
> 
> 
> On Jul 4, 2011, at 9:29 PM, Michael Collins wrote:
> 
>> I think the best "skeleton" config is Brian West's soft-phone configuration. It looks like we haven't migrated the download link over to git yet so I will see if I can't track it down. In the meantime if anyone has a copy of that config please post it here.
>> 
>> What's nice about it is that it's basically just one file with an option to read in gateway configs from a subdirectory. It's only a few hundred lines of XML and it shows what you can do with just a single config file.
>> 
>> -MC
>> 
>> On Mon, Jul 4, 2011 at 4:56 PM, Bryan Smart <bryansmart at bryansmart.com> wrote:
>> Is a truly skeletal configuration available? As far as I can tell, the choice right now is to start with no configuration files at all, or else to install a sample set that is bursting with stuff.
>> 
>> So far, I've installed the samples, and tried to strip out the parts that are obviously not needed by me:
>> 
>> The dialplan is the best example that I can think of: all the exts that replace functions on Snom/Linksys phones, record/play video, demo substitute ring tone, call groups, add/del groups, dynamically create numerous types of conferences, call features like redial/hold/etc, the exts for the demo IVR, and so on. I know enough to know that I don't want those on a live system, but I don't know enough to know if it is safe to remove parts. Will I break refer or deflect if I get rid of the "refer" ext? Is "unloop" only necessary as a safety feature if I'm connecting sessions to SIP URIs outside of the local FS? ? Can I drop "external-sip-uri" if users won't be calling external URIs, or will that completely block FS from connecting to external URIs in all cases? I think I can dump features.xml altogether.
>> 
>> There are an internal and external SIP profile. If I'm only hosting a voice application, I think I can dump the internal profile and just run the default external profile on 5060, but there are a huge amount of options in there that look as if they're setup to cover cases where the external profile is used to connect to ITSPs, which I won't.
>> 
>> There are so many configuration files that I'm not comfortable with blowing the whole directory tree away and recreating a raw basics version of each of them. While I don't feel paranoid about a specific part of the security of the full samples install, my rule is always that I try to not run anything that I don't at least somewhat understand. There seem to be so many little demos and optional features that I know that I by no means understand the implications of what they could be used to do in combination with what I'll add later. It seems safer to start from the bare basics, and only add what is needed, but, for now, I must start with the samples, and try to strip away as much as I can without breaking FS.
>> 
>> For me, I'm using FS as a platform for running a voice app, so I need little, if any, PBX type features. Some people that might wish to use FS as a PBX might not want all of the demo bits, though. An absolute minimum set of configuration files, as well as a set containing the essentials for a app platform or PBX would be useful, if they don't already exist somewhere.
>> 
>> Bryan
>> 
>> 
>> _______________________________________________
>> 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
>> 
>> _______________________________________________
>> 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
> 
> _______________________________________________
> 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
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From david.villasmil.work at gmail.com  Tue Jul  5 13:11:17 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 5 Jul 2011 11:11:17 +0200
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <CAKzWOxWQUd9Pw9cQOhRotitWvm8Xjn57tVU1tPwsoJEFApadEQ@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
	<4E120C26.70001@puzzled.xs4all.nl>
	<CAKzWOxWQUd9Pw9cQOhRotitWvm8Xjn57tVU1tPwsoJEFApadEQ@mail.gmail.com>
Message-ID: <CAFGRPVozi2qN5tn0xpb5=uZaaGybmNT1NEJ3XNoQ8zLP38u5jQ@mail.gmail.com>

Hello Michael,

Cool, I'll be waiting, then.

Thanks

David

On Tue, Jul 5, 2011 at 3:19 AM, Michael Collins <msc at freeswitch.org> wrote:

> Raymond can get you set up with an account on the freeswitch-contrib repo.
> We're all back in the office tomorrow so we'll check in then.
> -MC
>
>
> On Mon, Jul 4, 2011 at 11:53 AM, Patrick Lists <
> freeswitch-list at puzzled.xs4all.nl> wrote:
>
>> On 07/04/2011 03:12 PM, David Villasmil wrote:
>> > Hello All,
>> >
>> > I think I'm as ready as i can be to publish this...
>> > Can someone guide me into publishing via GIT?
>>
>> You could ask the FreeSWITCH developers to allow you access to your own
>> repo in the FreeSWITCH contrib area:
>>
>> http://fisheye.freeswitch.org/browse/freeswitch-contrib
>>
>> Regards,
>> Patrick
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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 sanjay.k.arora at gmail.com  Tue Jul  5 15:00:41 2011
From: sanjay.k.arora at gmail.com (Sanjay Arora)
Date: Tue, 5 Jul 2011 16:30:41 +0530
Subject: [Freeswitch-users] FreeSwitch/voip on EC2 or other cloud/virtual
	services
Message-ID: <CAAKp7utZkH+wq3DU7Uej60SKWEXoieRRVeYHRHrr1ONnCoPDqg@mail.gmail.com>

Hello all

I am looking to find out from users about voip on cloud, especially FS on
Amazon EC2. Google gave very few results. There were very few positive
posts/articles, among them Eric Wickstrom's post on this list around
2009<http://www.spinics.net/lists/freeswitch/msg03870.html>but many
seemed to be having clock/echo related problems.

I would like to ask if any of you are successfully using Amazon as a voip
hosting provider & what are the issues involved/stumbling blocks.

I am looking to install a sme multi-tenant production system and will be
selling voip minutes to a small focus target segment. I am mostly a
marketing man & a small-time hobby sys-admin, so I'd also like to know how
difficult it is.

On a side note, I would like to solicit your opinion of Sipwise free & pro
edition <http://www.sipwise.com/products/comparison/> especially with
respect to deployment on Amazon cloud.

With best regards & thanks.
Sanjay.
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From steveayre at gmail.com  Tue Jul  5 16:23:19 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 5 Jul 2011 13:23:19 +0100
Subject: [Freeswitch-users] FreeSwitch/voip on EC2 or other
	cloud/virtual services
In-Reply-To: <CAAKp7utZkH+wq3DU7Uej60SKWEXoieRRVeYHRHrr1ONnCoPDqg@mail.gmail.com>
References: <CAAKp7utZkH+wq3DU7Uej60SKWEXoieRRVeYHRHrr1ONnCoPDqg@mail.gmail.com>
Message-ID: <CAFiqYunfGZtz7HStPJweMEP3ZCaiYmmWku6oYqCJg4up2_92gw@mail.gmail.com>

I haven't personally, but I have seen some people on the list using EC2 for
FS. AFAIK it will work, essentially, for some value of work.

However, when running a realtime server such as FS on a virtual machine you
have to bear in mind that the guest OS has little control of timing. The
host can take away processing time unpredictibly for other tasks, especially
if the server is quite loaded. That means that keeping a constant timer
running can be tricky for FS to do, which is why you saw many people have
had clock and audio quality problems. Jitter buffers will cover it up, but
can only do so much.

How well it performs is probably best determined by setting it up yourself
and doing some testing.

-Steve




On 5 July 2011 12:00, Sanjay Arora <sanjay.k.arora at gmail.com> wrote:

> Hello all
>
> I am looking to find out from users about voip on cloud, especially FS on
> Amazon EC2. Google gave very few results. There were very few positive
> posts/articles, among them Eric Wickstrom's post on this list around 2009<http://www.spinics.net/lists/freeswitch/msg03870.html>but many seemed to be having clock/echo related problems.
>
> I would like to ask if any of you are successfully using Amazon as a voip
> hosting provider & what are the issues involved/stumbling blocks.
>
> I am looking to install a sme multi-tenant production system and will be
> selling voip minutes to a small focus target segment. I am mostly a
> marketing man & a small-time hobby sys-admin, so I'd also like to know how
> difficult it is.
>
> On a side note, I would like to solicit your opinion of Sipwise free & pro
> edition <http://www.sipwise.com/products/comparison/> especially with
> respect to deployment on Amazon cloud.
>
> With best regards & thanks.
> Sanjay.
>
>
> _______________________________________________
> 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 avi at avimarcus.net  Tue Jul  5 16:31:46 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 5 Jul 2011 15:31:46 +0300
Subject: [Freeswitch-users] FreeSwitch/voip on EC2 or other
	cloud/virtual services
In-Reply-To: <CAFiqYunfGZtz7HStPJweMEP3ZCaiYmmWku6oYqCJg4up2_92gw@mail.gmail.com>
References: <CAAKp7utZkH+wq3DU7Uej60SKWEXoieRRVeYHRHrr1ONnCoPDqg@mail.gmail.com>
	<CAFiqYunfGZtz7HStPJweMEP3ZCaiYmmWku6oYqCJg4up2_92gw@mail.gmail.com>
Message-ID: <CALNZuw=YrYjH78tW8OiwAVi4xT4n1jpyYjxH=cvr5fOKhXRs4g@mail.gmail.com>

I keep hearing that amazon has unpredictable timing - I've been using
Linode - xen - without any issues for FS for low volume. I tested 200
MOH channels on the lowest plan with only a few dropped packets. If
you're running that volume, you can afford more than $20/month for
their cheapest box with up to 39 other people on the same machine.

Linode says their CPU is the most under-utilized resource. The clock
tests in FS have always been good, but you can experiment with the
1000hz centos or new tickless + timerfd or something. But I've had
very good experience with Linode for my virtualization. (just don't go
California datacenter, HE has been having trouble...)

-Avi Marcus




On Tue, Jul 5, 2011 at 3:23 PM, Steven Ayre <steveayre at gmail.com> wrote:
> I haven't personally, but I have seen some people on the list using EC2 for
> FS. AFAIK it will work, essentially, for some value of work.
>
> However, when running a realtime server such as FS on a virtual machine you
> have to bear in mind that the guest OS has little control of timing. The
> host can take away processing time unpredictibly for other tasks, especially
> if the server is quite loaded. That means that keeping a constant timer
> running can be tricky for FS to do, which is why you saw many people have
> had clock and audio quality problems. Jitter buffers will cover it up, but
> can only do so much.
>
> How well it performs is probably best determined by setting it up yourself
> and doing some testing.
>
> -Steve
>
>
>
>
> On 5 July 2011 12:00, Sanjay Arora <sanjay.k.arora at gmail.com> wrote:
>>
>> Hello all
>>
>> I am looking to find out from users about voip on cloud, especially FS on
>> Amazon EC2. Google gave very few results. There were very few positive
>> posts/articles, among them Eric Wickstrom's post on this list around 2009
>> but many seemed to be having clock/echo related problems.
>>
>> I would like to ask if any of you are successfully using Amazon as a voip
>> hosting provider & what are the issues involved/stumbling blocks.
>>
>> I am looking to install a sme multi-tenant production system and will be
>> selling voip minutes to a small focus target segment. I am mostly a
>> marketing man & a small-time hobby sys-admin, so I'd also like to know how
>> difficult it is.
>>
>> On a side note, I would like to solicit your opinion of Sipwise free & pro
>> edition especially with respect to deployment on Amazon cloud.
>>
>> With best regards & thanks.
>> Sanjay.
>>
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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 gmaruzz at gmail.com  Tue Jul  5 17:22:13 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 5 Jul 2011 15:22:13 +0200
Subject: [Freeswitch-users] FreeSwitch/voip on EC2 or other
	cloud/virtual services
In-Reply-To: <CALNZuw=YrYjH78tW8OiwAVi4xT4n1jpyYjxH=cvr5fOKhXRs4g@mail.gmail.com>
References: <CAAKp7utZkH+wq3DU7Uej60SKWEXoieRRVeYHRHrr1ONnCoPDqg@mail.gmail.com>
	<CAFiqYunfGZtz7HStPJweMEP3ZCaiYmmWku6oYqCJg4up2_92gw@mail.gmail.com>
	<CALNZuw=YrYjH78tW8OiwAVi4xT4n1jpyYjxH=cvr5fOKhXRs4g@mail.gmail.com>
Message-ID: <CALXCt0pyaYpry-JtMyC=BnqCnnHQ_Wrq_v_jyLNKg=CQSCUfFQ@mail.gmail.com>

Honest: if you are not very, very, very into knowledge about kernels,
how kernels keep timing, xen, FreeSWITCH, etc (but very into
knowledge) *don't* put a production system in the cloud.
Ask some consultant to do it for you (if you want to be in the cloud),
or rent a real hardware server from some good server provider.
-giovanni

On 7/5/11, Avi Marcus <avi at avimarcus.net> wrote:
> I keep hearing that amazon has unpredictable timing - I've been using
> Linode - xen - without any issues for FS for low volume. I tested 200
> MOH channels on the lowest plan with only a few dropped packets. If
> you're running that volume, you can afford more than $20/month for
> their cheapest box with up to 39 other people on the same machine.
>
> Linode says their CPU is the most under-utilized resource. The clock
> tests in FS have always been good, but you can experiment with the
> 1000hz centos or new tickless + timerfd or something. But I've had
> very good experience with Linode for my virtualization. (just don't go
> California datacenter, HE has been having trouble...)
>
> -Avi Marcus
>
>
>
>
> On Tue, Jul 5, 2011 at 3:23 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> I haven't personally, but I have seen some people on the list using EC2
>> for
>> FS. AFAIK it will work, essentially, for some value of work.
>>
>> However, when running a realtime server such as FS on a virtual machine
>> you
>> have to bear in mind that the guest OS has little control of timing. The
>> host can take away processing time unpredictibly for other tasks,
>> especially
>> if the server is quite loaded. That means that keeping a constant timer
>> running can be tricky for FS to do, which is why you saw many people have
>> had clock and audio quality problems. Jitter buffers will cover it up, but
>> can only do so much.
>>
>> How well it performs is probably best determined by setting it up yourself
>> and doing some testing.
>>
>> -Steve
>>
>>
>>
>>
>> On 5 July 2011 12:00, Sanjay Arora <sanjay.k.arora at gmail.com> wrote:
>>>
>>> Hello all
>>>
>>> I am looking to find out from users about voip on cloud, especially FS on
>>> Amazon EC2. Google gave very few results. There were very few positive
>>> posts/articles, among them Eric Wickstrom's post on this list around 2009
>>> but many seemed to be having clock/echo related problems.
>>>
>>> I would like to ask if any of you are successfully using Amazon as a voip
>>> hosting provider & what are the issues involved/stumbling blocks.
>>>
>>> I am looking to install a sme multi-tenant production system and will be
>>> selling voip minutes to a small focus target segment. I am mostly a
>>> marketing man & a small-time hobby sys-admin, so I'd also like to know
>>> how
>>> difficult it is.
>>>
>>> On a side note, I would like to solicit your opinion of Sipwise free &
>>> pro
>>> edition especially with respect to deployment on Amazon cloud.
>>>
>>> With best regards & thanks.
>>> Sanjay.
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From sanjay.k.arora at gmail.com  Tue Jul  5 17:46:43 2011
From: sanjay.k.arora at gmail.com (Sanjay Arora)
Date: Tue, 5 Jul 2011 19:16:43 +0530
Subject: [Freeswitch-users] FreeSwitch/voip on EC2 or other
	cloud/virtual services
In-Reply-To: <CALXCt0pyaYpry-JtMyC=BnqCnnHQ_Wrq_v_jyLNKg=CQSCUfFQ@mail.gmail.com>
References: <CAAKp7utZkH+wq3DU7Uej60SKWEXoieRRVeYHRHrr1ONnCoPDqg@mail.gmail.com>
	<CAFiqYunfGZtz7HStPJweMEP3ZCaiYmmWku6oYqCJg4up2_92gw@mail.gmail.com>
	<CALNZuw=YrYjH78tW8OiwAVi4xT4n1jpyYjxH=cvr5fOKhXRs4g@mail.gmail.com>
	<CALXCt0pyaYpry-JtMyC=BnqCnnHQ_Wrq_v_jyLNKg=CQSCUfFQ@mail.gmail.com>
Message-ID: <CAAKp7uuKwg9pKY6D1uE8vdwa7dzjzkjD7SphOqMOvW6Hb38zqA@mail.gmail.com>

On Tue, Jul 5, 2011 at 6:52 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> Honest: if you are not very, very, very into knowledge about kernels,
> how kernels keep timing, xen, FreeSWITCH, etc (but very into
> knowledge) *don't* put a production system in the cloud.
> Ask some consultant to do it for you (if you want to be in the cloud),
> or rent a real hardware server from some good server provider.
> -giovanni
>

Actually I had thought of that...hiring a consultant to do the tuning of the
cloud install...but some people are telling me that such tuning is going to
be an ongoing job and I will have to keep on doing it thereby adding enough
consultant cost to my itty-bitty project, so as to make cloud cost gains
redundant.

Is the issue faced one-time at install or am I going to face it again and
again?

And I assume I would face those issues on all virtual computers, wether at
Amazone or linode?

Rgds.
Sanjay.
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From netcentrica at gmail.com  Tue Jul  5 18:59:38 2011
From: netcentrica at gmail.com (Mateusz Bartczak)
Date: Tue, 5 Jul 2011 16:59:38 +0200
Subject: [Freeswitch-users] Outbound socket,
	bridge command - how to specify leg_timeout?
Message-ID: <CAJWEF2auuZHKY0WaThXze=8tb=TM8ffPSbGOPo04TDjsM-5Djg@mail.gmail.com>

Hi all

I use outbound socket to handle call, I'm bridging call to external voip
provider. This works very nice but I also need to specify call timeout (once
answered).

I've tried following but with no success:

sendmsg
call-command: execute
execute-app-name: bridge
execute-app-arg:
{ignore_early_media=true,leg_timeout=10}sofia/gateway/trunk/600
event-lock: true

sendmsg
call-command: execute
execute-app-name: bridge
execute-app-arg:
{ignore_early_media=true}[leg_timeout=10]sofia/gateway/trunk/600
event-lock: true

sendmsg
call-command: execute
execute-app-name: bridge
execute-app-arg: [leg_timeout=10]sofia/gateway/trunk/600
event-lock: true


Leg_timeout seems to be ignored and call continues after 10 seconds. What's
the correct form of setting it for calls bridged via outbound socket
handler?

Is the leg_timeout a good choice to limit maximum answered call length?
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From david.ponzone at ipeva.fr  Tue Jul  5 19:01:38 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 5 Jul 2011 17:01:38 +0200
Subject: [Freeswitch-users] FreeSwitch/voip on EC2 or other
	cloud/virtual services
In-Reply-To: <CAAKp7uuKwg9pKY6D1uE8vdwa7dzjzkjD7SphOqMOvW6Hb38zqA@mail.gmail.com>
References: <CAAKp7utZkH+wq3DU7Uej60SKWEXoieRRVeYHRHrr1ONnCoPDqg@mail.gmail.com>
	<CAFiqYunfGZtz7HStPJweMEP3ZCaiYmmWku6oYqCJg4up2_92gw@mail.gmail.com>
	<CALNZuw=YrYjH78tW8OiwAVi4xT4n1jpyYjxH=cvr5fOKhXRs4g@mail.gmail.com>
	<CALXCt0pyaYpry-JtMyC=BnqCnnHQ_Wrq_v_jyLNKg=CQSCUfFQ@mail.gmail.com>
	<CAAKp7uuKwg9pKY6D1uE8vdwa7dzjzkjD7SphOqMOvW6Hb38zqA@mail.gmail.com>
Message-ID: <E6016CC2-0E1E-443F-A0DD-0BD7C87670D8@ipeva.fr>

I am not sure I understand the reason for a cloud solution.
Most of the time, it costs quite exactly the same and even more than a dedicated server.

There are probably good reasons to go for the cloud, but I think I am missing most of them;

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

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

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




Le 05/07/2011 ? 15:46, Sanjay Arora a ?crit :

> 
> On Tue, Jul 5, 2011 at 6:52 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> Honest: if you are not very, very, very into knowledge about kernels,
> how kernels keep timing, xen, FreeSWITCH, etc (but very into
> knowledge) *don't* put a production system in the cloud.
> Ask some consultant to do it for you (if you want to be in the cloud),
> or rent a real hardware server from some good server provider.
> -giovanni
> 
> Actually I had thought of that...hiring a consultant to do the tuning of the cloud install...but some people are telling me that such tuning is going to be an ongoing job and I will have to keep on doing it thereby adding enough consultant cost to my itty-bitty project, so as to make cloud cost gains redundant.
> 
> Is the issue faced one-time at install or am I going to face it again and again?
> 
> And I assume I would face those issues on all virtual computers, wether at Amazone or linode?
> 
> Rgds.
> Sanjay.
> _______________________________________________
> 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 avi at avimarcus.net  Tue Jul  5 19:04:17 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 5 Jul 2011 18:04:17 +0300
Subject: [Freeswitch-users] Outbound socket,
 bridge command - how to specify leg_timeout?
In-Reply-To: <CAJWEF2auuZHKY0WaThXze=8tb=TM8ffPSbGOPo04TDjsM-5Djg@mail.gmail.com>
References: <CAJWEF2auuZHKY0WaThXze=8tb=TM8ffPSbGOPo04TDjsM-5Djg@mail.gmail.com>
Message-ID: <CALNZuwnfzobQiMP=fV9Ezg2u0gmjjjY0j-_RNsqQtcs4E6WPpQ@mail.gmail.com>

leg_timeout  - http://wiki.freeswitch.org/wiki/Variable_leg_timeout - this
limits how long to TRY to connect the leg.
Once the call connects, this timer is no longer applied.

If you want to enforce a maximum answered time, then use:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_hangup
or http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_transfer
 <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_transfer>
-Avi Marcus


On Tue, Jul 5, 2011 at 5:59 PM, Mateusz Bartczak <netcentrica at gmail.com>
wrote:
> Hi all
>
> I use outbound socket to handle call, I'm bridging call to external voip
> provider. This works very nice but I also need to specify call timeout
(once
> answered).
>
> I've tried following but with no success:
>
> sendmsg
> call-command: execute
> execute-app-name: bridge
> execute-app-arg:
> {ignore_early_media=true,leg_timeout=10}sofia/gateway/trunk/600
> event-lock: true
>
> sendmsg
> call-command: execute
> execute-app-name: bridge
> execute-app-arg:
> {ignore_early_media=true}[leg_timeout=10]sofia/gateway/trunk/600
> event-lock: true
>
> sendmsg
> call-command: execute
> execute-app-name: bridge
> execute-app-arg: [leg_timeout=10]sofia/gateway/trunk/600
> event-lock: true
>
>
> Leg_timeout seems to be ignored and call continues after 10 seconds.
What's
> the correct form of setting it for calls bridged via outbound socket
> handler?
>
> Is the leg_timeout a good choice to limit maximum answered call length?
>
>
>
> _______________________________________________
> 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 sanjay.k.arora at gmail.com  Tue Jul  5 19:18:58 2011
From: sanjay.k.arora at gmail.com (Sanjay Arora)
Date: Tue, 5 Jul 2011 20:48:58 +0530
Subject: [Freeswitch-users] FreeSwitch/voip on EC2 or other
	cloud/virtual services
In-Reply-To: <E6016CC2-0E1E-443F-A0DD-0BD7C87670D8@ipeva.fr>
References: <CAAKp7utZkH+wq3DU7Uej60SKWEXoieRRVeYHRHrr1ONnCoPDqg@mail.gmail.com>
	<CAFiqYunfGZtz7HStPJweMEP3ZCaiYmmWku6oYqCJg4up2_92gw@mail.gmail.com>
	<CALNZuw=YrYjH78tW8OiwAVi4xT4n1jpyYjxH=cvr5fOKhXRs4g@mail.gmail.com>
	<CALXCt0pyaYpry-JtMyC=BnqCnnHQ_Wrq_v_jyLNKg=CQSCUfFQ@mail.gmail.com>
	<CAAKp7uuKwg9pKY6D1uE8vdwa7dzjzkjD7SphOqMOvW6Hb38zqA@mail.gmail.com>
	<E6016CC2-0E1E-443F-A0DD-0BD7C87670D8@ipeva.fr>
Message-ID: <CAAKp7ut3so_2DWJzrODLEU=CbeqvFMGOBMN+ntY-9i0_Y6a3Zw@mail.gmail.com>

On Tue, Jul 5, 2011 at 8:31 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> I am not sure I understand the reason for a cloud solution.
> Most of the time, it costs quite exactly the same and even more than a
> dedicated server.
>
> There are probably good reasons to go for the cloud, but I think I am
> missing most of them;
>

For me its scaling up, without much work. Starting up I start up with a
small EC2 instance, about 35$/month plus few dollars in bandwidth & I/O
charges, similar to an entry level linode or synapseglobal vps.

But migrating to a higher instance, medium instance around 175$ & so on.
Snapshop backups are so easy in amazon, that I would not have setup &
installation headaches again....simply save the instance to EBS & boot up in
a medium or large instance......or so I think ;-)

Many times I do tend to go on without understanding....so seeking third
party confirmation before I embark ;-)

Sanjay.
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From abid_freeswitch at live.com  Tue Jul  5 19:21:49 2011
From: abid_freeswitch at live.com (Abid Saleem)
Date: Tue, 5 Jul 2011 21:21:49 +0600
Subject: [Freeswitch-users] Load Balance Trunks
In-Reply-To: <BANLkTineHeUHCMpKNJS7msb6xCZzBqmkQg@mail.gmail.com>
References: <col109-w22D1131E50A2495C1521C9F4590@phx.gbl>,
	<C71A4485-E533-48AE-AC3F-C59D7A1E79CA@ipeva.fr>,
	<COL109-W612603C4C7BCC734CBD974F45B0@phx.gbl>,
	<BANLkTinT9O5+hzT1YczsmQV5ouL9EQ_2TQ@mail.gmail.com>,
	<233898962-1309536985-cardhu_decombobulator_blackberry.rim.net-2106826141-@b13.c4.bise7.blackberry>,
	<BANLkTikjYw=0UDyGdipBzadj0_kSoOVS2Q@mail.gmail.com>,
	<1637192224-1309538673-cardhu_decombobulator_blackberry.rim.net-1258156530-@b13.c4.bise7.blackberry>,
	<BANLkTineHeUHCMpKNJS7msb6xCZzBqmkQg@mail.gmail.com>
Message-ID: <COL109-W4456F57D6EF8C3739998F5F45F0@phx.gbl>


Hi Micheal, Avi and All,
Sorry for a little late response as I was away. I have seen quite a few questions from you guys, so I am answering them in one email as below. 
1- How does the provider notify you that each trunk has used its allotted time?Abid -> They have some counter in their IMS network to count on mins per trunk per day and they inform us by email.2- Are these trunks inbound only?Abid -> No. All these are Outbound Trunks. We just use them to send outgoing calls to our provider.
3- What happens when a call extends more than 120 mins on a trunk, would the call be disconnected?Abid -> The call is not disconnected right away but they send us a notification the next day. There is no real-time disconnection.
4- And then no more calls that day on that trunk?Abid -> Calls do not stop connecting immediately but they keep going. Currently their notification process is manual not automatic blocking.
Please help me if you can. Thanks.
Regards-----------------Abid SaleemTechnical Manager NGNTerminus Technologies
Date: Fri, 1 Jul 2011 09:49:02 -0700 
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Load Balance Trunks

On Fri, Jul 1, 2011 at 9:46 AM,  <hmkias at gmail.com> wrote:

Lcr entry could be activated on the the trunks. A cron job could monitor the usage and add or remove the trunk for the day.

Are these trunks inbound only?
-MC


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

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From krice at freeswitch.org  Tue Jul  5 19:44:09 2011
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 05 Jul 2011 10:44:09 -0500
Subject: [Freeswitch-users] FreeSwitch/voip on EC2 or other
 cloud/virtual services
In-Reply-To: <CAAKp7ut3so_2DWJzrODLEU=CbeqvFMGOBMN+ntY-9i0_Y6a3Zw@mail.gmail.com>
Message-ID: <CA389B79.18931%krice@freeswitch.org>

For $175 you can easily rent a full Dedicated Server w/ money left Over...

Heck for $200 I?ll rent you 2 dedicated servers


On 7/5/11 10:18 AM, "Sanjay Arora" <sanjay.k.arora at gmail.com> wrote:

> On Tue, Jul 5, 2011 at 8:31 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
>> I am not sure I understand the reason for a cloud solution.
>> Most of the time, it costs quite exactly the same and even more than a
>> dedicated server.
>> 
>> There are probably good reasons to go for the cloud, but I think I am missing
>> most of them;
> 
> For me its scaling up, without much work. Starting up I start up with a small
> EC2 instance, about 35$/month plus few dollars in bandwidth & I/O charges,
> similar to an entry level linode or synapseglobal vps.
> 
> But migrating to a higher instance, medium instance around 175$ & so on.
> Snapshop backups are so easy in amazon, that I would not have setup &
> installation headaches again....simply save the instance to EBS & boot up in a
> medium or large instance......or so I think ;-)
> 
> Many times I do tend to go on without understanding....so seeking third party
> confirmation before I embark ;-)
> 
> Sanjay.? 
> 
> 
> _______________________________________________
> 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
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From sanjay.k.arora at gmail.com  Tue Jul  5 19:52:16 2011
From: sanjay.k.arora at gmail.com (Sanjay Arora)
Date: Tue, 5 Jul 2011 21:22:16 +0530
Subject: [Freeswitch-users] FreeSwitch/voip on EC2 or other
	cloud/virtual services
In-Reply-To: <CA389B79.18931%krice@freeswitch.org>
References: <CAAKp7ut3so_2DWJzrODLEU=CbeqvFMGOBMN+ntY-9i0_Y6a3Zw@mail.gmail.com>
	<CA389B79.18931%krice@freeswitch.org>
Message-ID: <CAAKp7uv-C6t9p2ALZ9mVx3AbE9r7Z+zO4M7F5p-BUddFX8-FYQ@mail.gmail.com>

On Tue, Jul 5, 2011 at 9:14 PM, Ken Rice <krice at freeswitch.org> wrote:

>  For $175 you can easily rent a full Dedicated Server w/ money left
> Over...
>
> Heck for $200 I?ll rent you 2 dedicated servers
>
> Thanks, I'll keep that in mind ;-)

But is scaling to a higher server as easy as it would be in Amazon...just
boot up in a higher instance....or am I deluding myself?

Sanjay.
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From krice at freeswitch.org  Tue Jul  5 20:03:54 2011
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 05 Jul 2011 11:03:54 -0500
Subject: [Freeswitch-users] FreeSwitch/voip on EC2 or other
 cloud/virtual services
In-Reply-To: <CAAKp7uv-C6t9p2ALZ9mVx3AbE9r7Z+zO4M7F5p-BUddFX8-FYQ@mail.gmail.com>
Message-ID: <CA38A01A.18935%krice@freeswitch.org>

Its not specifically that easy, however keep in mind, that a $100 Dedicated
server, on reasonable bandwidth will probably handle more traffic then you
can imagine...



On 7/5/11 10:52 AM, "Sanjay Arora" <sanjay.k.arora at gmail.com> wrote:

> On Tue, Jul 5, 2011 at 9:14 PM, Ken Rice <krice at freeswitch.org> wrote:
>> For $175 you can easily rent a full Dedicated Server w/ money left Over...
>> 
>> Heck for $200 I?ll rent you 2 dedicated servers
>> 
> Thanks, I'll keep that in mind ;-)
> 
> But is scaling to a higher server as easy as it would be in Amazon...just boot
> up in a higher instance....or am I deluding myself?
> 
> Sanjay. 
> 
> 
> _______________________________________________
> 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
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From chris at cheeky.org  Tue Jul  5 20:17:11 2011
From: chris at cheeky.org (Chris Hemmings)
Date: Tue, 5 Jul 2011 17:17:11 +0100
Subject: [Freeswitch-users] FreeSwitch/voip on EC2 or other
	cloud/virtual services
In-Reply-To: <CA38A01A.18935%krice@freeswitch.org>
References: <CAAKp7uv-C6t9p2ALZ9mVx3AbE9r7Z+zO4M7F5p-BUddFX8-FYQ@mail.gmail.com>
	<CA38A01A.18935%krice@freeswitch.org>
Message-ID: <CAJW763kFu86BO6x=2dxZHhTpuPYbB4Fvfwkh5S7d1yrnOAvK8w@mail.gmail.com>

I must admit I'm a bit of a Cloud/VM fan.  The thought of not having to
think about the actual hardware you are running on is quite liberating.

Not having to worry about 'what if my server dies' is quite nice.

Creating a quick instance based on the one I'm currently running to test
something and not be tied into a X month contract.

Adding/Removing capacity to my VM as and when I want to.

Chris

On 5 July 2011 17:03, Ken Rice <krice at freeswitch.org> wrote:

>  Its not specifically that easy, however keep in mind, that a $100
> Dedicated server, on reasonable bandwidth will probably handle more traffic
> then you can imagine...
>
>
>
>
> On 7/5/11 10:52 AM, "Sanjay Arora" <sanjay.k.arora at gmail.com> wrote:
>
> On Tue, Jul 5, 2011 at 9:14 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> For $175 you can easily rent a full Dedicated Server w/ money left Over...
>
> Heck for $200 I?ll rent you 2 dedicated servers
>
> Thanks, I'll keep that in mind ;-)
>
> But is scaling to a higher server as easy as it would be in Amazon...just
> boot up in a higher instance....or am I deluding myself?
>
> Sanjay.
>
> ------------------------------
> _______________________________________________
> 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
>
>
> _______________________________________________
> 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
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From steveayre at gmail.com  Tue Jul  5 20:29:45 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 5 Jul 2011 17:29:45 +0100
Subject: [Freeswitch-users] FreeSwitch/voip on EC2 or other
	cloud/virtual services
In-Reply-To: <CAJW763kFu86BO6x=2dxZHhTpuPYbB4Fvfwkh5S7d1yrnOAvK8w@mail.gmail.com>
References: <CAAKp7uv-C6t9p2ALZ9mVx3AbE9r7Z+zO4M7F5p-BUddFX8-FYQ@mail.gmail.com>
	<CA38A01A.18935%krice@freeswitch.org>
	<CAJW763kFu86BO6x=2dxZHhTpuPYbB4Fvfwkh5S7d1yrnOAvK8w@mail.gmail.com>
Message-ID: <CAFiqYu=1oALrJzHgkgQefr4DkmyQLG03BfeLWOGH-w-Z22GeJQ@mail.gmail.com>

>
> Not having to worry about 'what if my server dies' is quite nice.
>

You still have to worry about that though. What if process X crashes? What
if the data centre fails. What if the data centre has an outage and loses
all your data (
http://www.datacenterknowledge.com/archives/2007/10/02/amazon-ec2-outage-wipes-out-data/
)

It just encourages people to be complacent because it becomes someone elses
problem and because since they're paying for it to be managed for them
they're assuming nothing can go wrong.

-Steve



On 5 July 2011 17:17, Chris Hemmings <chris at cheeky.org> wrote:

> I must admit I'm a bit of a Cloud/VM fan.  The thought of not having to
> think about the actual hardware you are running on is quite liberating.
>
> Not having to worry about 'what if my server dies' is quite nice.
>
> Creating a quick instance based on the one I'm currently running to test
> something and not be tied into a X month contract.
>
> Adding/Removing capacity to my VM as and when I want to.
>
> Chris
>
> On 5 July 2011 17:03, Ken Rice <krice at freeswitch.org> wrote:
>
>>  Its not specifically that easy, however keep in mind, that a $100
>> Dedicated server, on reasonable bandwidth will probably handle more traffic
>> then you can imagine...
>>
>>
>>
>>
>> On 7/5/11 10:52 AM, "Sanjay Arora" <sanjay.k.arora at gmail.com> wrote:
>>
>> On Tue, Jul 5, 2011 at 9:14 PM, Ken Rice <krice at freeswitch.org> wrote:
>>
>> For $175 you can easily rent a full Dedicated Server w/ money left Over...
>>
>> Heck for $200 I?ll rent you 2 dedicated servers
>>
>> Thanks, I'll keep that in mind ;-)
>>
>> But is scaling to a higher server as easy as it would be in Amazon...just
>> boot up in a higher instance....or am I deluding myself?
>>
>> Sanjay.
>>
>> ------------------------------
>> _______________________________________________
>> 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
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 lfurrea at gmail.com  Tue Jul  5 20:30:45 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Tue, 5 Jul 2011 10:30:45 -0600
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
Message-ID: <CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>

Hello Nandy,

A couple of months ago I started some research on the subject and concluded
I had to write my own interface to FS, however I haven't had the time to get
the project off ground yet.

I do have a copy of FIAS specification version 1.5 from 2001 which is
publicly available I am sure it's not the latest but it should cover the
basics.

Please contact me off list if you have a hard time getting it online.

Regards

On Mon, Jul 4, 2011 at 4:07 PM, Nandy Dagondon <gcd at i.ph> wrote:

> hi everybody,
>
> anyone working on interfacing FS with Fidelio Hotel PMS?  i can't find the
> FIAS protocol/specs online.  is this freely available?
>
> tks,
> nandy
>
>
> _______________________________________________
> 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
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>
>
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From gmaruzz at gmail.com  Tue Jul  5 20:38:19 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 5 Jul 2011 18:38:19 +0200
Subject: [Freeswitch-users] FreeSwitch/voip on EC2 or other
	cloud/virtual services
In-Reply-To: <CAFiqYu=1oALrJzHgkgQefr4DkmyQLG03BfeLWOGH-w-Z22GeJQ@mail.gmail.com>
References: <CAAKp7uv-C6t9p2ALZ9mVx3AbE9r7Z+zO4M7F5p-BUddFX8-FYQ@mail.gmail.com>
	<CA38A01A.18935%krice@freeswitch.org>
	<CAJW763kFu86BO6x=2dxZHhTpuPYbB4Fvfwkh5S7d1yrnOAvK8w@mail.gmail.com>
	<CAFiqYu=1oALrJzHgkgQefr4DkmyQLG03BfeLWOGH-w-Z22GeJQ@mail.gmail.com>
Message-ID: <CALXCt0pN-55-YSxHdCKiXTdZ03-rYS5kkdBjX8M8FdLDyi=R+w@mail.gmail.com>

No way you can run a production voice server on a small instance.
You need at least, at lest, a medium instance.
That's because of how irq are handled.
Reason to be in the cloud is automatica geo load bancing, automatic
start of instances on traffic spikes.
That begins to be cost effective when you run at least some virtual machines.
In your case, go for dedicated hardware ;)
-giovanni

On 7/5/11, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> Not having to worry about 'what if my server dies' is quite nice.
>>
>
> You still have to worry about that though. What if process X crashes? What
> if the data centre fails. What if the data centre has an outage and loses
> all your data (
> http://www.datacenterknowledge.com/archives/2007/10/02/amazon-ec2-outage-wipes-out-data/
> )
>
> It just encourages people to be complacent because it becomes someone elses
> problem and because since they're paying for it to be managed for them
> they're assuming nothing can go wrong.
>
> -Steve
>
>
>
> On 5 July 2011 17:17, Chris Hemmings <chris at cheeky.org> wrote:
>
>> I must admit I'm a bit of a Cloud/VM fan.  The thought of not having to
>> think about the actual hardware you are running on is quite liberating.
>>
>> Not having to worry about 'what if my server dies' is quite nice.
>>
>> Creating a quick instance based on the one I'm currently running to test
>> something and not be tied into a X month contract.
>>
>> Adding/Removing capacity to my VM as and when I want to.
>>
>> Chris
>>
>> On 5 July 2011 17:03, Ken Rice <krice at freeswitch.org> wrote:
>>
>>>  Its not specifically that easy, however keep in mind, that a $100
>>> Dedicated server, on reasonable bandwidth will probably handle more
>>> traffic
>>> then you can imagine...
>>>
>>>
>>>
>>>
>>> On 7/5/11 10:52 AM, "Sanjay Arora" <sanjay.k.arora at gmail.com> wrote:
>>>
>>> On Tue, Jul 5, 2011 at 9:14 PM, Ken Rice <krice at freeswitch.org> wrote:
>>>
>>> For $175 you can easily rent a full Dedicated Server w/ money left
>>> Over...
>>>
>>> Heck for $200 I?ll rent you 2 dedicated servers
>>>
>>> Thanks, I'll keep that in mind ;-)
>>>
>>> But is scaling to a higher server as easy as it would be in Amazon...just
>>> boot up in a higher instance....or am I deluding myself?
>>>
>>> Sanjay.
>>>
>>> ------------------------------
>>> _______________________________________________
>>> 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
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From krice at freeswitch.org  Tue Jul  5 20:41:59 2011
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 05 Jul 2011 11:41:59 -0500
Subject: [Freeswitch-users] FreeSwitch/voip on EC2 or other
 cloud/virtual services
In-Reply-To: <CAJW763kFu86BO6x=2dxZHhTpuPYbB4Fvfwkh5S7d1yrnOAvK8w@mail.gmail.com>
Message-ID: <CA38A907.18941%krice@freeswitch.org>

Who said anything about a X month contract? Not to mention what happens when
instance X goes batty and takes out Instance Y or at the minimal impacts CPU
and I/O performance and leaves you with crappy sounding calls...

Sure ?cloud? resource have some things that could be nice, like on demand
adding a webserver or the like, but that?s all store/forward services, not
real time services... I?m not willing to impact the call quality of a
customer generating $1000s in monthly billables to save $50 by using a $50
EC2 instance when I can get a dedicated server for $100 to $150 and be
assured that someone elses crazy process isnt going to affect mine...

Sure this doesn?t address issues such as bandwidth but you have that to deal
with regardless of EC2 or Dedicated Server...

Maybe I?ve just spent to much time dealing in the carrier arena where things
like capacity and quality matter...

K


On 7/5/11 11:17 AM, "Chris Hemmings" <chris at cheeky.org> wrote:

> I must admit I'm a bit of a Cloud/VM fan. ?The thought of not having to think
> about the actual hardware you are running on is quite liberating.
> 
> Not having to worry about 'what if my server dies' is quite nice.
> 
> Creating a quick instance based on the one I'm currently running to test
> something and not be tied into a X month contract.
> 
> Adding/Removing capacity to my VM as and when I want to.
> 
> Chris
> 
> On 5 July 2011 17:03, Ken Rice <krice at freeswitch.org> wrote:
>> Its not specifically that easy, however keep in mind, that a $100 Dedicated
>> server, on reasonable bandwidth will probably handle more traffic then you
>> can imagine...
>> 
>> 
>> 
>> 
>> On 7/5/11 10:52 AM, "Sanjay Arora" <sanjay.k.arora at gmail.com
>> <http://sanjay.k.arora at gmail.com> > wrote:
>> 
>>> On Tue, Jul 5, 2011 at 9:14 PM, Ken Rice <krice at freeswitch.org
>>> <http://krice at freeswitch.org> > wrote:
>>>> For $175 you can easily rent a full Dedicated Server w/ money left Over...
>>>> 
>>>> Heck for $200 I?ll rent you 2 dedicated servers
>>>> 
>>> Thanks, I'll keep that in mind ;-)
>>> 
>>> But is scaling to a higher server as easy as it would be in Amazon...just
>>> boot up in a higher instance....or am I deluding myself?
>>> 
>>> Sanjay. 
>>> 
>>> 
>>> _______________________________________________
>>> 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://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> 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
>> 
> 
> 
> 
> _______________________________________________
> 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 steveayre at gmail.com  Tue Jul  5 20:44:18 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 5 Jul 2011 17:44:18 +0100
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
Message-ID: <CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>

I'm assuming it's this document:
ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf
Quite easy to google once I had the version number.

You may find the nicest approach is to write a FOSS libfias, then write an
endpoint module to tie FS and libfias together. Plenty of existing endpoint
modules (mod_sofia mod_skinny mod_opal mod_h323 etc) can show you examples
to get you started. Don't forget to read the FS API documentation too:
http://docs.freeswitch.org/

I'm assuming there are no license/patent restrictions to using FIAS?

Good luck!

-Steve



On 5 July 2011 17:30, Luis F Urrea <lfurrea at gmail.com> wrote:

> Hello Nandy,
>
> A couple of months ago I started some research on the subject and concluded
> I had to write my own interface to FS, however I haven't had the time to get
> the project off ground yet.
>
> I do have a copy of FIAS specification version 1.5 from 2001 which is
> publicly available I am sure it's not the latest but it should cover the
> basics.
>
> Please contact me off list if you have a hard time getting it online.
>
> Regards
>
> On Mon, Jul 4, 2011 at 4:07 PM, Nandy Dagondon <gcd at i.ph> wrote:
>
>> hi everybody,
>>
>> anyone working on interfacing FS with Fidelio Hotel PMS?  i can't find the
>> FIAS protocol/specs online.  is this freely available?
>>
>> tks,
>> nandy
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 lfurrea at gmail.com  Tue Jul  5 20:52:15 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Tue, 5 Jul 2011 10:52:15 -0600
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
Message-ID: <CAB2KLycjzFtLc8jTBM9377CDi488K9nDmVD=Zz-13MZ7_w-cxA@mail.gmail.com>

Awesome! great suggestions to get started,

There is also a FIAS simulator floating around.

That one may be a little harder to find? :)

On Tue, Jul 5, 2011 at 10:44 AM, Steven Ayre <steveayre at gmail.com> wrote:

> I'm assuming it's this document:
> ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf
> Quite easy to google once I had the version number.
>
> You may find the nicest approach is to write a FOSS libfias, then write an
> endpoint module to tie FS and libfias together. Plenty of existing
> endpoint modules (mod_sofia mod_skinny mod_opal mod_h323 etc) can show you
> examples to get you started. Don't forget to read the FS API documentation
> too: http://docs.freeswitch.org/
>
> I'm assuming there are no license/patent restrictions to using FIAS?
>
> Good luck!
>
> -Steve
>
>
>
>
> On 5 July 2011 17:30, Luis F Urrea <lfurrea at gmail.com> wrote:
>
>> Hello Nandy,
>>
>> A couple of months ago I started some research on the subject and
>> concluded I had to write my own interface to FS, however I haven't had the
>> time to get the project off ground yet.
>>
>> I do have a copy of FIAS specification version 1.5 from 2001 which is
>> publicly available I am sure it's not the latest but it should cover the
>> basics.
>>
>> Please contact me off list if you have a hard time getting it online.
>>
>> Regards
>>
>> On Mon, Jul 4, 2011 at 4:07 PM, Nandy Dagondon <gcd at i.ph> wrote:
>>
>>> hi everybody,
>>>
>>> anyone working on interfacing FS with Fidelio Hotel PMS?  i can't find
>>> the FIAS protocol/specs online.  is this freely available?
>>>
>>> tks,
>>> nandy
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 jmesquita at freeswitch.org  Tue Jul  5 21:45:31 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 5 Jul 2011 14:45:31 -0300
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <CAB2KLycjzFtLc8jTBM9377CDi488K9nDmVD=Zz-13MZ7_w-cxA@mail.gmail.com>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
	<CAB2KLycjzFtLc8jTBM9377CDi488K9nDmVD=Zz-13MZ7_w-cxA@mail.gmail.com>
Message-ID: <CABH8QX2y1mu1L=9-KZk2HH-Xh9MFOS98p4ctvvQrF7AW_RJV_A@mail.gmail.com>

Guys, be careful because I think this document as well as the protocol are
confidential. I had to sign an NDA with Fidelio to get my hands on it and
pay a fee for it as well. You might as well confirm it since you all seem to
be in the US where this type of information might be easier to get.

There are LOTS of companies selling their connectors to Fidelio...

One other point is that you need to have the certification with them to be
considered compatible, otherwise, no consultant will install the connector
on the fidelio side.

Regards,
Jo?o Mesquita



On Tue, Jul 5, 2011 at 1:52 PM, Luis F Urrea <lfurrea at gmail.com> wrote:

> Awesome! great suggestions to get started,
>
> There is also a FIAS simulator floating around.
>
> That one may be a little harder to find? :)
>
> On Tue, Jul 5, 2011 at 10:44 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> I'm assuming it's this document:
>> ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf
>> Quite easy to google once I had the version number.
>>
>> You may find the nicest approach is to write a FOSS libfias, then write an
>> endpoint module to tie FS and libfias together. Plenty of existing
>> endpoint modules (mod_sofia mod_skinny mod_opal mod_h323 etc) can show
>> you examples to get you started. Don't forget to read the FS API
>> documentation too: http://docs.freeswitch.org/
>>
>> I'm assuming there are no license/patent restrictions to using FIAS?
>>
>> Good luck!
>>
>> -Steve
>>
>>
>>
>>
>> On 5 July 2011 17:30, Luis F Urrea <lfurrea at gmail.com> wrote:
>>
>>> Hello Nandy,
>>>
>>> A couple of months ago I started some research on the subject and
>>> concluded I had to write my own interface to FS, however I haven't had the
>>> time to get the project off ground yet.
>>>
>>> I do have a copy of FIAS specification version 1.5 from 2001 which is
>>> publicly available I am sure it's not the latest but it should cover the
>>> basics.
>>>
>>> Please contact me off list if you have a hard time getting it online.
>>>
>>> Regards
>>>
>>> On Mon, Jul 4, 2011 at 4:07 PM, Nandy Dagondon <gcd at i.ph> wrote:
>>>
>>>> hi everybody,
>>>>
>>>> anyone working on interfacing FS with Fidelio Hotel PMS?  i can't find
>>>> the FIAS protocol/specs online.  is this freely available?
>>>>
>>>> tks,
>>>> nandy
>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 yungwei at resolvity.com  Tue Jul  5 22:08:48 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Tue, 5 Jul 2011 14:08:48 -0400
Subject: [Freeswitch-users] originate a call to an extension and then
 connect to another extension
In-Reply-To: <BANLkTiki1Cd6Z05UfmF+JQA1hdx754ngMw@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950AC643195@VMBX102.ihostexchange.net>
	<BANLkTinnUwoN3nrMy0gioUmpyzh1LNg_tg@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC6431DC@VMBX102.ihostexchange.net>
	<BANLkTiki1Cd6Z05UfmF+JQA1hdx754ngMw@mail.gmail.com>
Message-ID: <33095823FD21DF429B481B5163264B7950CBAF7C03@VMBX102.ihostexchange.net>

Thanks for your feedback. Using originate along with loopback works.
However, DTMF input in /usr/local/freeswitch/sounds/answer.wav is always ignored.
Note /usr/local/freeswitch/sounds/answer.wav is an empty recording file with several DTMF input.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Monday, June 13, 2011 3:59 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] originate a call to an extension and then connect to another extension

originate loopback/question answer
On Mon, Jun 13, 2011 at 12:08 PM, Yungwei Chen wrote:
I don't mean registered users. I mean extensions defined in conf/dialplan/default/my_exten.xml as shown below.
In this case, I can do: originate sofia/internal/answer question?

  <extension name="question">
      <condition field="destination_number" expression="^questiony$">
        <action application="answer" />
        <action application="playback" data="/usr/local/freeswitch/sounds/question.wav" />
      </condition>
  </extension>

  <extension name="answer">
      <condition field="destination_number" expression="^answer$">
        <action application="answer" />
        <action application="playback" data="/usr/local/freeswitch/sounds/answer.wav" />
      </condition>
  </extension>

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:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Monday, June 13, 2011 1:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] originate a call to an extension and then connect to another extension

If you mean you want to call x1000 and then when it answers, immediately dial x1001? If so it's just this, assuming these are registered users:

originate user/1000 1001

If you want to call 1001 first then do this:

originate user/1001 1000

-MC
On Mon, Jun 13, 2011 at 10:49 AM, Yungwei Chen wrote:
Hi,

I am wondering if it's possible to originate a call to an extension and then connect to another extension. If so, how does the the dialstring look like? Thanks.

_______________________________________________
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 avi at avimarcus.net  Tue Jul  5 22:12:28 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 5 Jul 2011 21:12:28 +0300
Subject: [Freeswitch-users] originate a call to an extension and then
 connect to another extension
In-Reply-To: <33095823FD21DF429B481B5163264B7950CBAF7C03@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950AC643195@VMBX102.ihostexchange.net>
	<BANLkTinnUwoN3nrMy0gioUmpyzh1LNg_tg@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950AC6431DC@VMBX102.ihostexchange.net>
	<BANLkTiki1Cd6Z05UfmF+JQA1hdx754ngMw@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950CBAF7C03@VMBX102.ihostexchange.net>
Message-ID: <CALNZuw=VuyXezFSbL17-3WVtFf_6seg17YY_yFrKrMcvFf6=HQ@mail.gmail.com>

You probably need to run application start_dtmf
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
 <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf>to detect
the inband sounds from the recording.

-Avi Marcus


On Tue, Jul 5, 2011 at 9:08 PM, Yungwei Chen <yungwei at resolvity.com> wrote:

> Thanks for your feedback. Using originate along with loopback works.****
>
> However, DTMF input in /usr/local/freeswitch/sounds/answer.wav is always
> ignored.****
>
> Note /usr/local/freeswitch/sounds/answer.wav is an empty recording file
> with several DTMF input.****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, June 13, 2011 3:59 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] originate a call to an extension and
> then connect to another extension****
>
> ** **
>
> originate loopback/question answer****
>
> On Mon, Jun 13, 2011 at 12:08 PM, Yungwei Chen wrote:****
>
> I don't mean registered users. I mean extensions defined in
> conf/dialplan/default/my_exten.xml as shown below.  ****
>
> In this case, I can do: originate sofia/internal/answer question?****
>
>  ****
>
>   <extension name="question">****
>
>       <condition field="destination_number" expression="^questiony$">****
>
>         <action application="answer" />****
>
>         <action application="playback"
> data="/usr/local/freeswitch/sounds/question.wav" />****
>
>       </condition>****
>
>   </extension>****
>
>  ****
>
>   <extension name="answer">****
>
>       <condition field="destination_number" expression="^answer$">****
>
>         <action application="answer" />****
>
>         <action application="playback"
> data="/usr/local/freeswitch/sounds/answer.wav" />****
>
>       </condition>****
>
>   </extension>****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, June 13, 2011 1:37 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] originate a call to an extension and
> then connect to another extension****
>
>  ****
>
> If you mean you want to call x1000 and then when it answers, immediately
> dial x1001? If so it's just this, assuming these are registered users:****
>
>  ****
>
> originate user/1000 1001****
>
>  ****
>
> If you want to call 1001 first then do this:****
>
>  ****
>
> originate user/1001 1000****
>
>  ****
>
> -MC****
>
> On Mon, Jun 13, 2011 at 10:49 AM, Yungwei Chen wrote:****
>
> Hi,
>
> I am wondering if it's possible to originate a call to an extension and
> then connect to another extension. If so, how does the the dialstring look
> like? 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****
>
> ** **
>
> _______________________________________________
> 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 sebastien.peterson at gmail.com  Tue Jul  5 01:43:30 2011
From: sebastien.peterson at gmail.com (Sebastien Peterson)
Date: Mon, 4 Jul 2011 23:43:30 +0200
Subject: [Freeswitch-users] GTalk mod_dingaling read and send text messages
Message-ID: <CAHfJOUw45yDPYk5eTbL+F_4bh8BGRO=ZvpFErarNnxKxKRke1w@mail.gmail.com>

Hi freeswitch-users,
I am a new FreeSwitch user.

I am trying to send and read Gtalk text messages from the dialplan.
Is that something I can do and how? Could not find examples from the wiki.

Once I read the message, I need to place it into a variable.

In advance thanks !
Seb
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From michel.daggelinckx at gmail.com  Tue Jul  5 02:59:23 2011
From: michel.daggelinckx at gmail.com (Michel Daggelinckx)
Date: Tue, 5 Jul 2011 00:59:23 +0200
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <163DDA0D-0B14-4623-B48B-5E9EEE907AED@gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
	<163DDA0D-0B14-4623-B48B-5E9EEE907AED@gmail.com>
Message-ID: <CAFYZzbu9Nh3Ojb26G_pAiFn9MVmm59qeO8BL02_WLX5Fj0mfUg@mail.gmail.com>

Ask intralanman for acces to the FS contrib

On Mon, Jul 4, 2011 at 7:10 PM, Mohammad Emran <monemran at gmail.com> wrote:

> We can put on google code or git hub.
>
> Sent from my iPad
>
> On 4 Jul 2011, at 19:12, David Villasmil <david.villasmil.work at gmail.com>
> wrote:
>
> Hello All,
>
> I think I'm as ready as i can be to publish this...
> Can someone guide me into publishing via GIT?
>
> thanks
>
> David
>
> On Thu, Feb 24, 2011 at 12:39 PM, David Villasmil <<david.villasmil.work at gmail.com>
> david.villasmil.work at gmail.com> wrote:
>
>> Hello Guys,
>>
>> I'm finishing a "complete" wholesale application created on freeswitch and
>> I was wondering whether it would be a good idea to put it up on the wiki. I
>> just don't know how.
>>
>> Features include all the following parameters configurable via web
>> interface:
>>
>> - Profile creation based on server IP where traffic is received. You can
>> have multiple IPs, system will create multiple profiles/diaplans so it can
>> differentiate.
>>      - i.e. offer to the same customer a "gold" routing on IP1 and cheap
>> routing on IP2
>>
>> - Customer add/modify/delete
>>      - IP source
>>      - Rates for client routes based on areacode
>>      - Prepaid or postpaid.
>>           - When cutomer balance is 0, no more calls are allowed.
>>      - limit max channels
>>      - Media by-pass
>>           - When by-passed, customer and provider will exchange RTPs
>> directly. Else, server will be in the middle.
>>
>> - Provider add/modify/delete
>>      - costs for provider routes based on areacode
>>      - limit max channels
>>
>> - Routing based on areacode, gives great granularity.
>>
>> - Routes can be assigned multiple gateways/providers which can in turn be
>> distributed based on weigth. Includes overflow to next configured GW.
>>
>> - Basic financial report generation (totals) by customer/provider
>>
>> - Basic traffic ASR/ACD report (totals) by cutomer/provider
>>
>> - Basic user administration. (No access level, only total access)
>>
>> - CDR export to csv file.
>>
>>
>>
>>
>> I also have a prepaid card app... no web interface on that one though...
>>
>> Thanks all
>>
>>
>> David
>>
>
> _______________________________________________
> 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
>
>
> _______________________________________________
> 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 freeswitch-list at puzzled.xs4all.nl  Tue Jul  5 22:19:38 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Tue, 05 Jul 2011 20:19:38 +0200
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
Message-ID: <4E1355BA.6060304@puzzled.xs4all.nl>

On 07/05/2011 06:44 PM, Steven Ayre wrote:
> I'm assuming it's this document:
> ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf
> Quite easy to google once I had the version number.

If you strip the FIAS150.pdf from that link you can see a lot more docs 
and apps.

Regards,
Patrick


From rzhang at gosilverplus.com  Tue Jul  5 22:21:32 2011
From: rzhang at gosilverplus.com (ran zhang)
Date: Tue, 05 Jul 2011 11:21:32 -0700
Subject: [Freeswitch-users] pl HELP! how to change conference's
	caller-id-number at run-time?
Message-ID: <4E13562C.1040506@gosilverplus.com>

Hi All:

I want when the bridging conference is inviting someone to start the
conference, the caller is shown as the real caller id number rather than
whats stored in 'caller-id-number' in conference.conf.xml.

I have tried to change the 'effective_caller_id_number', it only affects when its a bridging call.



From msc at freeswitch.org  Tue Jul  5 22:21:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jul 2011 11:21:49 -0700
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <CAFYZzbu9Nh3Ojb26G_pAiFn9MVmm59qeO8BL02_WLX5Fj0mfUg@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
	<163DDA0D-0B14-4623-B48B-5E9EEE907AED@gmail.com>
	<CAFYZzbu9Nh3Ojb26G_pAiFn9MVmm59qeO8BL02_WLX5Fj0mfUg@mail.gmail.com>
Message-ID: <CAKzWOxX2CrDESPMOWMtCgSDxt1R2S3=70aDsTrRM2_p6isfkvA@mail.gmail.com>

FYI, Ray is out this week, so for now you might want to throw it up on
github for now.
-MC

On Mon, Jul 4, 2011 at 3:59 PM, Michel Daggelinckx <
michel.daggelinckx at gmail.com> wrote:

> Ask intralanman for acces to the FS contrib
>
> On Mon, Jul 4, 2011 at 7:10 PM, Mohammad Emran <monemran at gmail.com> wrote:
>
>> We can put on google code or git hub.
>>
>> Sent from my iPad
>>
>> On 4 Jul 2011, at 19:12, David Villasmil <david.villasmil.work at gmail.com>
>> wrote:
>>
>> Hello All,
>>
>> I think I'm as ready as i can be to publish this...
>> Can someone guide me into publishing via GIT?
>>
>> thanks
>>
>> David
>>
>> On Thu, Feb 24, 2011 at 12:39 PM, David Villasmil <<david.villasmil.work at gmail.com>
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Hello Guys,
>>>
>>> I'm finishing a "complete" wholesale application created on freeswitch
>>> and I was wondering whether it would be a good idea to put it up on the
>>> wiki. I just don't know how.
>>>
>>> Features include all the following parameters configurable via web
>>> interface:
>>>
>>> - Profile creation based on server IP where traffic is received. You can
>>> have multiple IPs, system will create multiple profiles/diaplans so it can
>>> differentiate.
>>>      - i.e. offer to the same customer a "gold" routing on IP1 and cheap
>>> routing on IP2
>>>
>>> - Customer add/modify/delete
>>>      - IP source
>>>      - Rates for client routes based on areacode
>>>      - Prepaid or postpaid.
>>>           - When cutomer balance is 0, no more calls are allowed.
>>>      - limit max channels
>>>      - Media by-pass
>>>           - When by-passed, customer and provider will exchange RTPs
>>> directly. Else, server will be in the middle.
>>>
>>> - Provider add/modify/delete
>>>      - costs for provider routes based on areacode
>>>      - limit max channels
>>>
>>> - Routing based on areacode, gives great granularity.
>>>
>>> - Routes can be assigned multiple gateways/providers which can in turn be
>>> distributed based on weigth. Includes overflow to next configured GW.
>>>
>>> - Basic financial report generation (totals) by customer/provider
>>>
>>> - Basic traffic ASR/ACD report (totals) by cutomer/provider
>>>
>>> - Basic user administration. (No access level, only total access)
>>>
>>> - CDR export to csv file.
>>>
>>>
>>>
>>>
>>> I also have a prepaid card app... no web interface on that one though...
>>>
>>> Thanks all
>>>
>>>
>>> David
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 steveayre at gmail.com  Tue Jul  5 22:22:30 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 5 Jul 2011 19:22:30 +0100
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <4E1355BA.6060304@puzzled.xs4all.nl>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
	<4E1355BA.6060304@puzzled.xs4all.nl>
Message-ID: <CAFiqYumRQrbcjT86GCBOO3H2u4B3oHqq3+wwON3S3-O-vCnMDg@mail.gmail.com>

Sure, but Jo?o is right - they want to clarify the legal position before
investing any time in doing any development work on it.

-Steve




On 5 July 2011 19:19, Patrick Lists <freeswitch-list at puzzled.xs4all.nl>wrote:

> On 07/05/2011 06:44 PM, Steven Ayre wrote:
> > I'm assuming it's this document:
> >
> ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf
> > Quite easy to google once I had the version number.
>
> If you strip the FIAS150.pdf from that link you can see a lot more docs
> and apps.
>
> Regards,
> Patrick
>
> _______________________________________________
> 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 fischetti at digiunit.it  Tue Jul  5 22:53:42 2011
From: fischetti at digiunit.it (Carmelo Fischetti - Digi Unit)
Date: Tue, 5 Jul 2011 20:53:42 +0200
Subject: [Freeswitch-users] NAT Traversal
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAE80uuUA8mNNh/+IJ5oVSWzCgAAAEAAAAOAW6hhZqshDpydrHqOqtDYBAAAAAA==@digiunit.it>

Hi all,

 

I'm experiencing a NAT traversal problem connecting some softphones to a
FreeSWITCH server.

At office the softphone registers with TLS-NAT, while at home it registers
with TLS and actually it's unreacheable (it can place calls but it cannot
receive them), even if the registration parameters (sofia status profile
internal reg) contain both the public (IP: x.y.z.w) and the private
(Contact: "user"
<sip:1010 at 192.168.1.100:6010;transport=TLS;line=3c4e8d673906c5d>) IP
addresses. In both cases the softphone is behind a router, the server is
behind a firewall and runs the ddclient service (there is a DynDNS name
assigned to it).

 

Am I missing some configuration detail at server side?

At the moment I've modified the default configuration files in the following
way.

 

The following lines in vars.xml:

<X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>

<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>

<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>

were substituted with:

<X-PRE-PROCESS cmd="set" data="domain=myhost.dyndns-server.com"/>

<X-PRE-PROCESS cmd="set"
data="external_rtp_ip=host:myhost.dyndns-server.com"/>

<X-PRE-PROCESS cmd="set"
data="external_sip_ip=host:myhost.dyndns-server.com"/>

 

The following lines in sip_profiles/internal.xml:

<param name="ext-rtp-ip" value="auto-nat"/>

<param name="ext-sip-ip" value="auto-nat"/>

were substituted with:

<param name="ext-rtp-ip" value="host:myhost.dyndns-server.com"/>

<param name="ext-sip-ip" value="host:myhost.dyndns-server.com"/>

and I've also added the line:

<param name="sip-force-contact" value="NDLB-connectile-dysfunction"/>

but I actually don't understand what it means (I've found it in some posts
about NAT traversal in FreeSWITCH).

 

Thanks in advance.

Carmelo 

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From lfurrea at gmail.com  Tue Jul  5 23:19:34 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Tue, 5 Jul 2011 13:19:34 -0600
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <CAFiqYumRQrbcjT86GCBOO3H2u4B3oHqq3+wwON3S3-O-vCnMDg@mail.gmail.com>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
	<4E1355BA.6060304@puzzled.xs4all.nl>
	<CAFiqYumRQrbcjT86GCBOO3H2u4B3oHqq3+wwON3S3-O-vCnMDg@mail.gmail.com>
Message-ID: <CAB2KLyfWZmKhn4ZGY9cG4FSOSXqrsxa4WTGwra5NweKGLEOSkg@mail.gmail.com>

HHmm that is correct,

There's work to be done on the legal side as well then.

I just assumed that since the spec was available through a single google
click the protocol was public domain.

thanks guys for your input, as always is highly appreciated

On Tue, Jul 5, 2011 at 12:22 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Sure, but Jo?o is right - they want to clarify the legal position before
> investing any time in doing any development work on it.
>
> -Steve
>
>
>
>
>
> On 5 July 2011 19:19, Patrick Lists <freeswitch-list at puzzled.xs4all.nl>wrote:
>
>> On 07/05/2011 06:44 PM, Steven Ayre wrote:
>> > I'm assuming it's this document:
>> >
>> ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf
>> > Quite easy to google once I had the version number.
>>
>> If you strip the FIAS150.pdf from that link you can see a lot more docs
>> and apps.
>>
>> Regards,
>> Patrick
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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 steveayre at gmail.com  Wed Jul  6 00:07:11 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 5 Jul 2011 21:07:11 +0100
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <CAB2KLyfWZmKhn4ZGY9cG4FSOSXqrsxa4WTGwra5NweKGLEOSkg@mail.gmail.com>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
	<4E1355BA.6060304@puzzled.xs4all.nl>
	<CAFiqYumRQrbcjT86GCBOO3H2u4B3oHqq3+wwON3S3-O-vCnMDg@mail.gmail.com>
	<CAB2KLyfWZmKhn4ZGY9cG4FSOSXqrsxa4WTGwra5NweKGLEOSkg@mail.gmail.com>
Message-ID: <CAFiqYumrwPhvNSKzxKmfNpoLbO7T_cEW2mRERHM0kXZ551v-JA@mail.gmail.com>

That doesn't look like Fidelio hosting it, so don't assume they're aware of
it, or that that company knows it's public.

You can always try talking to Fidelio for permission, but if they insist on
a NDA good luck getting it into a FOSS project. ;)

-Steve




On 5 July 2011 20:19, Luis F Urrea <lfurrea at gmail.com> wrote:

> HHmm that is correct,
>
> There's work to be done on the legal side as well then.
>
> I just assumed that since the spec was available through a single google
> click the protocol was public domain.
>
> thanks guys for your input, as always is highly appreciated
>
> On Tue, Jul 5, 2011 at 12:22 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Sure, but Jo?o is right - they want to clarify the legal position before
>> investing any time in doing any development work on it.
>>
>> -Steve
>>
>>
>>
>>
>>
>> On 5 July 2011 19:19, Patrick Lists <freeswitch-list at puzzled.xs4all.nl>wrote:
>>
>>> On 07/05/2011 06:44 PM, Steven Ayre wrote:
>>> > I'm assuming it's this document:
>>> >
>>> ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf
>>> > Quite easy to google once I had the version number.
>>>
>>> If you strip the FIAS150.pdf from that link you can see a lot more docs
>>> and apps.
>>>
>>> Regards,
>>> Patrick
>>>
>>> _______________________________________________
>>> 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
>>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 rzhang at gosilverplus.com  Wed Jul  6 00:16:10 2011
From: rzhang at gosilverplus.com (ran zhang)
Date: Tue, 05 Jul 2011 13:16:10 -0700
Subject: [Freeswitch-users] conference caller id number&name
Message-ID: <4E13710A.5060108@gosilverplus.com>

I want when the bridging conference is inviting someone to start the
conference, the caller is shown as the real caller id number&name rather than
whats stored in 'caller-id-number' and 'caller-id-name' in conference.conf.xml.



From sebastien.peterson at gmail.com  Wed Jul  6 00:17:57 2011
From: sebastien.peterson at gmail.com (Sebastien Peterson)
Date: Tue, 5 Jul 2011 22:17:57 +0200
Subject: [Freeswitch-users] GTalk mod_dingaling read and send text
	messages
In-Reply-To: <CAHfJOUw45yDPYk5eTbL+F_4bh8BGRO=ZvpFErarNnxKxKRke1w@mail.gmail.com>
References: <CAHfJOUw45yDPYk5eTbL+F_4bh8BGRO=ZvpFErarNnxKxKRke1w@mail.gmail.com>
Message-ID: <CAHfJOUwb9kch6Zq+Y3jihGdG1gJUXD3_=caok0OXYUS0ZQNkZg@mail.gmail.com>

More generally,
how do you send/receive xmpp messages with FreeSwitch ?

Cheers,
Seb

On Mon, Jul 4, 2011 at 11:43 PM, Sebastien Peterson <
sebastien.peterson at gmail.com> wrote:

> Hi freeswitch-users,
> I am a new FreeSwitch user.
>
> I am trying to send and read Gtalk text messages from the dialplan.
> Is that something I can do and how? Could not find examples from the wiki.
>
> Once I read the message, I need to place it into a variable.
>
> In advance thanks !
> Seb
>
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From gcd at i.ph  Wed Jul  6 00:36:49 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Wed, 6 Jul 2011 04:36:49 +0800
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <CAFiqYumrwPhvNSKzxKmfNpoLbO7T_cEW2mRERHM0kXZ551v-JA@mail.gmail.com>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
	<4E1355BA.6060304@puzzled.xs4all.nl>
	<CAFiqYumRQrbcjT86GCBOO3H2u4B3oHqq3+wwON3S3-O-vCnMDg@mail.gmail.com>
	<CAB2KLyfWZmKhn4ZGY9cG4FSOSXqrsxa4WTGwra5NweKGLEOSkg@mail.gmail.com>
	<CAFiqYumrwPhvNSKzxKmfNpoLbO7T_cEW2mRERHM0kXZ551v-JA@mail.gmail.com>
Message-ID: <CAHKqMb7=UrFfNqD7XcXDsaBNxoUND8GOR6xZid7cYKAzw8yong@mail.gmail.com>

tks to all contributors!  just thinking - fidelio could allow us, FS group,
to develop it but for certification, we have to secure it from Fidelio w/
corresponding fee.  i would be willing to pay for FIAS module and the
certification cuz it would be cheaper to get another connector.

On Wed, Jul 6, 2011 at 4:07 AM, Steven Ayre <steveayre at gmail.com> wrote:

> That doesn't look like Fidelio hosting it, so don't assume they're aware of
> it, or that that company knows it's public.
>
> You can always try talking to Fidelio for permission, but if they insist on
> a NDA good luck getting it into a FOSS project. ;)
>
> -Steve
>
>
>
>
>
> On 5 July 2011 20:19, Luis F Urrea <lfurrea at gmail.com> wrote:
>
>> HHmm that is correct,
>>
>> There's work to be done on the legal side as well then.
>>
>> I just assumed that since the spec was available through a single google
>> click the protocol was public domain.
>>
>> thanks guys for your input, as always is highly appreciated
>>
>> On Tue, Jul 5, 2011 at 12:22 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> Sure, but Jo?o is right - they want to clarify the legal position before
>>> investing any time in doing any development work on it.
>>>
>>> -Steve
>>>
>>>
>>>
>>>
>>>
>>> On 5 July 2011 19:19, Patrick Lists <freeswitch-list at puzzled.xs4all.nl>wrote:
>>>
>>>> On 07/05/2011 06:44 PM, Steven Ayre wrote:
>>>> > I'm assuming it's this document:
>>>> >
>>>> ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf<https://docs.google.com/viewer?url=ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%2520-%2520PMS%2520info/FIAS150.pdf&embedded=true&chrome=true>
>>>> > Quite easy to google once I had the version number.
>>>>
>>>> If you strip the FIAS150.pdf from that link you can see a lot more docs
>>>> and apps.
>>>>
>>>> Regards,
>>>> Patrick
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 sebastien.peterson at gmail.com  Wed Jul  6 00:45:58 2011
From: sebastien.peterson at gmail.com (Sebastien Peterson)
Date: Tue, 5 Jul 2011 22:45:58 +0200
Subject: [Freeswitch-users] GTalk mod_dingaling read and send text
	messages
In-Reply-To: <CAHfJOUwb9kch6Zq+Y3jihGdG1gJUXD3_=caok0OXYUS0ZQNkZg@mail.gmail.com>
References: <CAHfJOUw45yDPYk5eTbL+F_4bh8BGRO=ZvpFErarNnxKxKRke1w@mail.gmail.com>
	<CAHfJOUwb9kch6Zq+Y3jihGdG1gJUXD3_=caok0OXYUS0ZQNkZg@mail.gmail.com>
Message-ID: <CAHfJOUzTVpTkwM-=C7=Vdqc1raMWtSFJ1_UO7mAF8cL3=jrD1A@mail.gmail.com>

So sending works like this:

<action application="set" data="api_result=${uuid_chat(${uuid} test
message)}"/>

...Just still need to find how to receive now !!

Seb


On Tue, Jul 5, 2011 at 10:17 PM, Sebastien Peterson <
sebastien.peterson at gmail.com> wrote:

> More generally,
> how do you send/receive xmpp messages with FreeSwitch ?
>
> Cheers,
> Seb
>
>
> On Mon, Jul 4, 2011 at 11:43 PM, Sebastien Peterson <
> sebastien.peterson at gmail.com> wrote:
>
>> Hi freeswitch-users,
>> I am a new FreeSwitch user.
>>
>> I am trying to send and read Gtalk text messages from the dialplan.
>> Is that something I can do and how? Could not find examples from the wiki.
>>
>> Once I read the message, I need to place it into a variable.
>>
>> In advance thanks !
>> Seb
>>
>
>
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From sebastien.peterson at gmail.com  Wed Jul  6 01:06:27 2011
From: sebastien.peterson at gmail.com (Sebastien Peterson)
Date: Tue, 5 Jul 2011 23:06:27 +0200
Subject: [Freeswitch-users] GTalk mod_dingaling read and send text
	messages
In-Reply-To: <CAHfJOUzTVpTkwM-=C7=Vdqc1raMWtSFJ1_UO7mAF8cL3=jrD1A@mail.gmail.com>
References: <CAHfJOUw45yDPYk5eTbL+F_4bh8BGRO=ZvpFErarNnxKxKRke1w@mail.gmail.com>
	<CAHfJOUwb9kch6Zq+Y3jihGdG1gJUXD3_=caok0OXYUS0ZQNkZg@mail.gmail.com>
	<CAHfJOUzTVpTkwM-=C7=Vdqc1raMWtSFJ1_UO7mAF8cL3=jrD1A@mail.gmail.com>
Message-ID: <CAHfJOUzTyWR1VX=wXHt-b4ku4sZHipP_a9JZHjCbTpZ+UVJRzQ@mail.gmail.com>

In the logs, I can read:

2011-07-05 22:57:13.677155 [DEBUG] mod_dingaling.c:3143 SESSION MSG
[12345678]

"12345678" is the message I send from my GTalk client. So dingaling is
capturing that. I just need to assign it to a dialplan variable.

Cheers,
Seb


On Tue, Jul 5, 2011 at 10:45 PM, Sebastien Peterson <
sebastien.peterson at gmail.com> wrote:

> So sending works like this:
>
> <action application="set" data="api_result=${uuid_chat(${uuid} test
> message)}"/>
>
> ...Just still need to find how to receive now !!
>
> Seb
>
>
>
> On Tue, Jul 5, 2011 at 10:17 PM, Sebastien Peterson <
> sebastien.peterson at gmail.com> wrote:
>
>> More generally,
>> how do you send/receive xmpp messages with FreeSwitch ?
>>
>> Cheers,
>> Seb
>>
>>
>> On Mon, Jul 4, 2011 at 11:43 PM, Sebastien Peterson <
>> sebastien.peterson at gmail.com> wrote:
>>
>>> Hi freeswitch-users,
>>> I am a new FreeSwitch user.
>>>
>>> I am trying to send and read Gtalk text messages from the dialplan.
>>> Is that something I can do and how? Could not find examples from the
>>> wiki.
>>>
>>> Once I read the message, I need to place it into a variable.
>>>
>>> In advance thanks !
>>> Seb
>>>
>>
>>
>
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From msc at freeswitch.org  Wed Jul  6 01:08:51 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jul 2011 14:08:51 -0700
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <CAHKqMb7=UrFfNqD7XcXDsaBNxoUND8GOR6xZid7cYKAzw8yong@mail.gmail.com>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
	<4E1355BA.6060304@puzzled.xs4all.nl>
	<CAFiqYumRQrbcjT86GCBOO3H2u4B3oHqq3+wwON3S3-O-vCnMDg@mail.gmail.com>
	<CAB2KLyfWZmKhn4ZGY9cG4FSOSXqrsxa4WTGwra5NweKGLEOSkg@mail.gmail.com>
	<CAFiqYumrwPhvNSKzxKmfNpoLbO7T_cEW2mRERHM0kXZ551v-JA@mail.gmail.com>
	<CAHKqMb7=UrFfNqD7XcXDsaBNxoUND8GOR6xZid7cYKAzw8yong@mail.gmail.com>
Message-ID: <CAKzWOxUGB6ZXy=-F=q-apGX5QM+s0OFrX8zX=G1qDiDUiMkRfA@mail.gmail.com>

Do you know any hotel owners/operators who run a Fidelio PMS and who would
be willing to let you tinker? That would be the ultimate test. :)

I'm actually very interested in this type of project. Please email me off
list.

-MC

On Tue, Jul 5, 2011 at 1:36 PM, Nandy Dagondon <gcd at i.ph> wrote:

> tks to all contributors!  just thinking - fidelio could allow us, FS
> group,  to develop it but for certification, we have to secure it from
> Fidelio w/ corresponding fee.  i would be willing to pay for FIAS module and
> the certification cuz it would be cheaper to get another connector.
>
> On Wed, Jul 6, 2011 at 4:07 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> That doesn't look like Fidelio hosting it, so don't assume they're aware
>> of it, or that that company knows it's public.
>>
>> You can always try talking to Fidelio for permission, but if they insist
>> on a NDA good luck getting it into a FOSS project. ;)
>>
>> -Steve
>>
>>
>>
>>
>>
>> On 5 July 2011 20:19, Luis F Urrea <lfurrea at gmail.com> wrote:
>>
>>> HHmm that is correct,
>>>
>>> There's work to be done on the legal side as well then.
>>>
>>> I just assumed that since the spec was available through a single google
>>> click the protocol was public domain.
>>>
>>> thanks guys for your input, as always is highly appreciated
>>>
>>> On Tue, Jul 5, 2011 at 12:22 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>>> Sure, but Jo?o is right - they want to clarify the legal position before
>>>> investing any time in doing any development work on it.
>>>>
>>>> -Steve
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On 5 July 2011 19:19, Patrick Lists <freeswitch-list at puzzled.xs4all.nl>wrote:
>>>>
>>>>> On 07/05/2011 06:44 PM, Steven Ayre wrote:
>>>>> > I'm assuming it's this document:
>>>>> >
>>>>> ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf<https://docs.google.com/viewer?url=ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%2520-%2520PMS%2520info/FIAS150.pdf&embedded=true&chrome=true>
>>>>> > Quite easy to google once I had the version number.
>>>>>
>>>>> If you strip the FIAS150.pdf from that link you can see a lot more docs
>>>>> and apps.
>>>>>
>>>>> Regards,
>>>>> Patrick
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 yungwei at resolvity.com  Wed Jul  6 03:15:27 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Tue, 5 Jul 2011 19:15:27 -0400
Subject: [Freeswitch-users] Calling an extension on a remote freeswitch box
Message-ID: <33095823FD21DF429B481B5163264B7950CBAF7CA5@VMBX102.ihostexchange.net>

Hi,

I'm trying to place a call to an extension on a remote freeswitch box. 
I tried the following command, but it doesn't work. 
originate {ignore_early_media=true}sofia/internal/77777 at 192.168.1.20 test

Here's the error message.
2011-07-05 18:06:46.670942 [ERR] sofia_reg.c:1912 Cannot locate any authentication credentials to complete an authentication request for realm '"192.168.1.18"'

Am I using the right dial string? If so, how can I fix the error? Thanks.


From gcd at i.ph  Wed Jul  6 05:33:40 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Wed, 6 Jul 2011 09:33:40 +0800
Subject: [Freeswitch-users] Calling an extension on a remote freeswitch
	box
In-Reply-To: <33095823FD21DF429B481B5163264B7950CBAF7CA5@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950CBAF7CA5@VMBX102.ihostexchange.net>
Message-ID: <CAHKqMb5cOdCtyDrkRxgPBCLN7uy2MCjaMvpKt8mO=w0he_-8Ew@mail.gmail.com>

hi chen,

your local FS must register to the remote FS using the external sip_profile.
 configure the remote fs as a gateway. so your CLI comman should be:
originate sofia/gateway/remotefs/xxxx

-nandy
On Wed, Jul 6, 2011 at 7:15 AM, Yungwei Chen <yungwei at resolvity.com> wrote:

> Hi,
>
> I'm trying to place a call to an extension on a remote freeswitch box.
> I tried the following command, but it doesn't work.
> originate {ignore_early_media=true}sofia/internal/77777 at 192.168.1.20 test
>
> Here's the error message.
> 2011-07-05 18:06:46.670942 [ERR] sofia_reg.c:1912 Cannot locate any
> authentication credentials to complete an authentication request for realm
> '"192.168.1.18"'
>
> Am I using the right dial string? If so, how can I fix the error? Thanks.
>
> _______________________________________________
> 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 gcd at i.ph  Wed Jul  6 08:57:00 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Wed, 6 Jul 2011 12:57:00 +0800
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <4E1355BA.6060304@puzzled.xs4all.nl>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
	<4E1355BA.6060304@puzzled.xs4all.nl>
Message-ID: <CAHKqMb443_DtVH0Dy-Spt52zZt+vJVQHTgoEZe9uX2zPYqhZNA@mail.gmail.com>

not just more docs but i also found the DOS-based PMS emulator.


On Wed, Jul 6, 2011 at 2:19 AM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 07/05/2011 06:44 PM, Steven Ayre wrote:
> > I'm assuming it's this document:
> >
> ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf
> > Quite easy to google once I had the version number.
>
> If you strip the FIAS150.pdf from that link you can see a lot more docs
> and apps.
>
> Regards,
> Patrick
>
> _______________________________________________
> 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 brokendash at gmail.com  Wed Jul  6 10:05:10 2011
From: brokendash at gmail.com (broken dash)
Date: Wed, 6 Jul 2011 01:05:10 -0500
Subject: [Freeswitch-users] Playing multiple files simultaneously
In-Reply-To: <EB5BFD1EA1F444079A8BFBAC95976FE1@dell9400>
References: <2DB72AF2-2394-419A-ADC5-0B76FE594662@bryansmart.com>
	<EB5BFD1EA1F444079A8BFBAC95976FE1@dell9400>
Message-ID: <CABNzcD-8JCOgSwUrVp=V8mix5n7Rie6989GkTDWDqCYiaJMteg@mail.gmail.com>

I have been wanting these features and I found this...liquidsoap,
check out the transcoding section.  I was trying to play multiple
shoutcast audio streams and hopefully mux them together nicely, etc..
haven't gotten around to setting freeswitch to utilize it but I'm sure
it wouldn't be hard.

http://savonet.sourceforge.net/doc-1.0.0-beta2/cookbook.html


Cheers,
Brian

On Fri, Jul 1, 2011 at 1:49 AM, Jan Berger <jan.berger at video24.no> wrote:
> If you play several files FS will queue them if it is the same stream - you
> need to somehow set up a conference with multiple streams (callers).
>
> I would check options around music on hold.
>
> But, another way is to use FS to play several outgoing SIP streams and loop
> them back into a conference.
>
>
> Jan
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bryan
> Smart
> Sent: 1. juli 2011 07:13
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Playing multiple files simultaneously
>
> Is it possible for Freeswitch to play more than one file to a channel at a
> time? What I've seen and tried from the dialplan and scripts either queues
> files to play, or will stop a currently playing file so that the newly
> requested file will play. This also seems to be the case in conferences.
> When I send multiple play commands to conferences, the files are queued.
>
> As for how this might be used, think of an IVR that plays queued prompts,
> yet continuously plays looping music or a Shoutcast stream in the
> background. I also want to be able to play short cue tones that start at the
> same time as a prompt (don't want to pre-mix them in to a single file,
> though).
>
> Is this currently possible through any means? Perhaps with the event socket?
>
> Bryan
>
>
> _______________________________________________
> 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
>
>
> _______________________________________________
> 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 msc at freeswitch.org  Wed Jul  6 10:25:16 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Jul 2011 23:25:16 -0700
Subject: [Freeswitch-users] Trouble with bind_digit_action and
	exec:execute_extension
Message-ID: <CAKzWOxV0UT=ReDs2Z=xuOsywjHQ3AR2o7ct2viMCWPyE8yY8Aw@mail.gmail.com>

Hello,

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

-MC

Dialplan:
  <extension name="BDA Test">


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


      <action application="answer"/>


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

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

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

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

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

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

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


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


    </condition>


  </extension>





  <extension name="BDA Target">


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


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


    </condition>


  </extension>

Just dial 77xxxx and then press *1, *2, etc. You will hear the number
voiced, i.e. *1 will say "one", *2 will say "two" and so forth.
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From dujinfang at gmail.com  Wed Jul  6 10:34:59 2011
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 6 Jul 2011 14:34:59 +0800
Subject: [Freeswitch-users] Chinese sounds
Message-ID: <40057F672A4E43A7971543ECA4C3B97B@gmail.com>

Hi,

We got someone in the Chinese Community would like to record a Chinese version of sounds, here's some questions for the community -


1) Where can I find the original English scripts ? It would be hard to listen all sound files

2) What's base256/*.wav usage?

3) We'd like to share to world, so, how to upload to files.freeswitch.org when we are done?

4) I guess the raw recording should be 48K, 16bit, single channel, any other params should we set when recording?


Thanks.

-- 
Seven Du
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 slickqt at gmail.com  Wed Jul  6 11:41:23 2011
From: slickqt at gmail.com (jun yang)
Date: Wed, 6 Jul 2011 15:41:23 +0800
Subject: [Freeswitch-users] Chinese sounds
In-Reply-To: <40057F672A4E43A7971543ECA4C3B97B@gmail.com>
References: <40057F672A4E43A7971543ECA4C3B97B@gmail.com>
Message-ID: <CALkN2oyubCu6YEbbA_X8mhtAtDO44=rMmWKVFnTY=F5CVJwPug@mail.gmail.com>

good news,look forward to having default chinese sounds in freeswitch.

2011/7/6 Seven Du <dujinfang at gmail.com>:
> Hi,
> We got someone in the Chinese Community would like to record a Chinese
> version of sounds, here's some questions for the community -
>
> 1) Where can I find the original English scripts ? It would be hard to
> listen all sound files
> 2) What's base256/*.wav usage?
> 3) We'd like to share to world, so, how to upload to files.freeswitch.org
> when we are done?
> 4) I guess the raw recording should be 48K, 16bit, single channel, any other
> params should we set when recording?
>
> Thanks.
>
> --
> Seven Du
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj: ?http://www.freeswitch.org.cn
> Sent with Sparrow
>
> _______________________________________________
> 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 daniel at danielknoll.de  Wed Jul  6 15:36:47 2011
From: daniel at danielknoll.de (Daniel Knoll)
Date: Wed, 6 Jul 2011 13:36:47 +0200
Subject: [Freeswitch-users] curl.run data field post file
Message-ID: <CACREEeapWMX1nxrmeSJX4eeUH=fTHPX9m2+AeZnXXTxJckqqXA@mail.gmail.com>

Hi Group,

I figured the right syntax of curl.run out. and now I have a question.
How can I post the Content of a File into the "data" Field of curl.run()
If i set the Option to "@/tmp/filename.flac" or
"filename=@/tmp/filename.flac" the Content-Length is allways the
number of the Characters in the "data" Field.

my code line is:

curl.run("POST",
"http://www.google.com/speech-api/v1/recognize?xjerr=1&client=chromium&lang=de-DEc",
"filename=@/tmp/90.flac", my_callback, "my arg\n","","","audio/x-flac;
rate=16000");

The sniffing of HTTP Traffic look like this:


interface: eth0 (xxx.xxx.xxx.xxx/255.255.255.192)
filter: (ip or ip6) and ( port 80 )
####
T xxx.xxx.xxx.xxx:45159 -> 209.85.149.105:80 [AP]
  POST /speech-api/v1/recognize?xjerr=1&client=chromium&lang=de-DEc
HTTP/1.1..User-Agent:
   freeswitch-spidermonkey-curl/1.0..Host: www.google.com..Accept:
*/*..Content-Type: aud
  io/x-flac; rate=16000..Content-Length: 22....filename=@/tmp/90.flac
##
T 209.85.149.105:80 -> xxx.xxx.xxx.xxx:45159 [AP]
  HTTP/1.1 200 OK..Content-Type: application/json;
charset=utf-8..Content-Disposition: at
  tachment..Date: Wed, 06 Jul 2011 11:23:35 GMT..Expires: Wed, 06 Jul
2011 11:23:35 GMT..
  Cache-Control: private, max-age=0..X-Content-Type-Options:
nosniff..X-Frame-Options: SA
  MEORIGIN..X-XSS-Protection: 1; mode=block..Server:
GSE..Transfer-Encoding: chunked....4
  7..{"status":5,"id":"ec8d60cb80800296297afe166dee826c-1","hypotheses":[]}...0....
####exit
11 received, 0 dropped



any ideas?

Thanks for helping me.
Daniel


From freeswitch at peely.com  Wed Jul  6 16:33:23 2011
From: freeswitch at peely.com (peely)
Date: Wed, 6 Jul 2011 05:33:23 -0700 (PDT)
Subject: [Freeswitch-users] RTMP buffer size?
Message-ID: <1309955603001-6554200.post@n2.nabble.com>

Hi,

I've been having a good old play with mod_rtmp, which is a very exciting
addition, thanks!

The main feedback I get when testing with colleagues is the latency, which
seems to be somewhere in the order of 1.5 seconds each way.

My testbed is an RTMP client > FS server > PSTN so I could three points
where jitter buffers and encoders would be maintained. Ping time between
servers is in the order of 50-60ms.

Is there a way of controlling the size of the jitter buffer in mod_rtmp to
reduce it? jitter_buffer_msec doesn't seem to change anything in this
scenario.

On another note, it seems the option is rtmp.conf.xml to allow calls without
rgistration doesn't seem to enable this option, I always seem to have to get
the client to log in.


Cheers,



Neil.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/RTMP-buffer-size-tp6554200p6554200.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From shouldbeq931 at gmail.com  Wed Jul  6 16:53:57 2011
From: shouldbeq931 at gmail.com (shouldbe q931)
Date: Wed, 6 Jul 2011 13:53:57 +0100
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
 Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
	<4DF10C83.6020803@earthspike.net>
	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>
	<4DF1F07A.5000200@earthspike.net>
	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>
	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>
	<4DFB9831.4040709@earthspike.net>
	<BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>
Message-ID: <CAP1_MOCq5qRjsgd7XGag9ukjHiG2Se-pXvrghQ7hbB5ciwJK+A@mail.gmail.com>

On Fri, Jun 17, 2011 at 7:43 PM, shouldbe q931 <shouldbeq931 at gmail.com> wrote:
> On Fri, Jun 17, 2011 at 7:08 PM, John <freeswitch at earthspike.net> wrote:
>> Mike, Shouldbe,
>>
>> We now have CLIP on the line for sure, and I ordered COLP at the same time
>> and am assured that that has been turned on as well, but I still cannot get
>> outgoing CLI to work properly.? Mike, you mention setting the TON to
>> 'national'; where is that setting?
>>
>> At the moment, my dialplan looks like http://pastebin.freeswitch.org/16548
>> with .? An outgoing call gives the log 7 at
>> http://pastebin.freeswitch.org/16549 (numbers have been changed to protect
>> the innocent!) but the number that 01234 567890 sees on their Caller ID is
>> not 876543 but the main number of the line (876540, say).? I have tried
>> setting the outgoing_caller_id_number to 6, 10 and 11 digits, restarting FS
>> after each change, but with no success.
>>
>> The service provider has only one clue to offer: "With regards to the
>> configuration at the exchange, the line is set to 6 digits to switch." which
>> makes eminent sense, and suggests that I should be presenting 6 digits.
>> Incoming calls have a 6 digit called number and 10 digit calling number.
>>
>> Thanks for the help so far.? Any other ideas?
>>
>> John
>>
>>
>
> I don't have any experience with FS connecting over ISDN so can't help
> you much further on the FS configuration:-(
>
> However on the lines, I would be very tempted to see if you can borrow
> a BRI tester, or what I've frequently used in the past is an
> Eicon/Dialogic card as the test functions (under windows) are nearly
> as good, that might help you see if it _is_ a BT problem, or a FS
> configuration issue. It would need a 2k/XP machine, but something like
> http://cgi.ebay.co.uk/EICON-DIVA-2-01-PCI-GRAPHICS-CARD-/270761571564
> is what I have used in the past. I think I might have one of the
> PCMCIA ones somewhere, but I'd need to test it still works
>
> I'm fairly sure that the "6 digits to switch" is what they are sending
> to you, not what they are expecting from you. I can't access out
> remaining BRI (Avaya) PBX from here, but on our PRI (again Avaya)
> switches, we are sending 10 digits.
>
> Cheers
>

Apologies for the very long delay...

On our PRI systems we are sending 10 digits (2071231234), and on the
BRI system we are sending 11 digits (02071231234).

Cheers


From ankitwalia4u at gmail.com  Wed Jul  6 17:25:16 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Wed, 6 Jul 2011 18:55:16 +0530
Subject: [Freeswitch-users] Module_Exists for SpiderMonkey and PocketSpinx
 failing the dialplan at Execution state
Message-ID: <CAOcuKucJ4eVn4UhbwmurLjQ2_CxhFJ9=_byO0FOXPwsA7OHiPg@mail.gmail.com>

Hi all,

I tried to execute the pizza example for PocketSphinx for speech
recognition.

I added the grammar,enabled mod_pocketsphinx and spidermonkey, added sound
files and js files as per the wiki.
http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Install_.26_Configure

When, I call to the extension, though it passes the Regex, showing true for
the conditions. But could not execute the javascript application and
subsequently the js file.

The dialplan looks like *
<extension name="pizza_demo">
    <condition field="destination_number" expression="^(74992)$"/>
    <condition field="${module_exists(mod_spidermonkey)}"
expression="true"/>
    <condition field="${module_exists(mod_pocketsphinx)}"
expression="true">
    <action application="javascript" data="ps_pizza.js"/>
    </condition>
  </extension>*

If I remove the condition of module exist, the js file gets executed and the
call goes on. But, It does not work with Module exists condition.

I have pasted the logs http://pastebin.freeswitch.org/16680. I could not
even find anything concrete on the logs.

Thanks
Ankit

-- 
Life is like a rose its upto u feel it as its fragrance or thorns
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From roger.castaldo at gmail.com  Wed Jul  6 18:09:49 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Wed, 6 Jul 2011 10:09:49 -0400
Subject: [Freeswitch-users] Configuration Server
Message-ID: <CAAYf2Lvv1f8QBmDL9scJm45KcDA7CoVbCmGOcSwBvAH8bnjMVA@mail.gmail.com>

Hi everyone, I have been following freeswitch for quite some time now and
working on an ever evolving configuration server for it, which has now taken
yet another branch, that instead of writing xml configuration files now is
attempting to control the call flow itself using the outbound socket.
Currently it is a closed source project, originally because of some of my
closed source libraries I was using to develop it, but since they have
become open source I was wondering if there is any interest in working with
me on the development/testing of the product.  It is written in C# and is
entirely web based, using ajax calls and web services for everything.  I am
using a Relational Mapping library of my own design to map the data to the
database (currently a firebirdsql database).
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From a.luppi at seletech.com  Wed Jul  6 18:56:21 2011
From: a.luppi at seletech.com (Alessandro)
Date: Wed, 06 Jul 2011 16:56:21 +0200
Subject: [Freeswitch-users] Custom GUI for FS
Message-ID: <4E147795.3090700@seletech.com>

Hi,

I'm going to develop a custom GUI web-based for freeswitch that allow 
only to add/remove extension and show the call active and the status of 
extensions configured.
Where can I find in FS the information about actives call and extensions 
status?

Thanks

Alessandro Luppi

-- 
Ing. Alessandro Luppi
Software development
Seletech srl
Via Collodi 8, 20052 Monza (MI) - Italy
Tel: +39.039.5962000 - Fax: +39.039.9716905
email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu



From nagalenoj at gmail.com  Wed Jul  6 19:05:27 2011
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 6 Jul 2011 20:35:27 +0530
Subject: [Freeswitch-users] Custom GUI for FS
In-Reply-To: <4E147795.3090700@seletech.com>
References: <4E147795.3090700@seletech.com>
Message-ID: <CAA1Qw2ua8_Rsx-f6HZpg_g+k0w=RM-JuT0mrAxJfgce7xXCbDQ@mail.gmail.com>

Hi,
Hope, show(show calls, show channels) command will help you.

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


On Wed, Jul 6, 2011 at 8:26 PM, Alessandro <a.luppi at seletech.com> wrote:

> Hi,
>
> I'm going to develop a custom GUI web-based for freeswitch that allow
> only to add/remove extension and show the call active and the status of
> extensions configured.
> Where can I find in FS the information about actives call and extensions
> status?
>
> Thanks
>
> Alessandro Luppi
>
> --
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>
>
> _______________________________________________
> 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
>



-- 
Regards,
Nagalenoj H.
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From mitch.capper at gmail.com  Wed Jul  6 19:08:01 2011
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 6 Jul 2011 08:08:01 -0700
Subject: [Freeswitch-users] FSClient FreesSWITCH Softphone Updated
Message-ID: <CALn233r0VQn59N=V0sQthnyMWXoxfM=3r5sep9UX=ooHk3Gr5w@mail.gmail.com>

A new binary build of FSClient is out it has several changes from the
original build published and should fix any language issues people had
with the original installer.  It also adds support for call recording,
TLS, and a few other improvements.   In addition it has been
documented in the wiki at: http://wiki.freeswitch.com/wiki/FSClient
with features, usage, information to compile it yourself,
developer/plugin information etc.  Binary installer can be found at:
http://files.freeswitch.org/windows/installer/x86/FSClient.zip  it
will upgrade your settings automatically and you need not uninstall
the old version.   It also includes a patched fs_cli per patch
http://jira.freeswitch.org/browse/FS-3188 which can be used with
fs_logger.pl (freeswitch log collector more information can be found
at: http://fisheye.freeswitch.org/browse/~raw,r=HEAD/freeswitch-contrib/mitchcapper/fs_logger.txt
 or the perl app itself:
http://fisheye.freeswitch.org/browse/~raw,r=HEAD/freeswitch-contrib/mitchcapper/fs_logger.pl
  or a compiled binary at http://mitchcapper.com/fs_logger.exe ).
Feedback on both always welcome!

~Mitch


From avi at avimarcus.net  Wed Jul  6 19:08:29 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 6 Jul 2011 18:08:29 +0300
Subject: [Freeswitch-users] Custom GUI for FS
In-Reply-To: <4E147795.3090700@seletech.com>
References: <4E147795.3090700@seletech.com>
Message-ID: <CALNZuwm=RrrJuY6duQJEfw63NVTSUyEhVjahecXb6D-qsgPEUA@mail.gmail.com>

Hi - this, and much more, was already done in http://www.FusionPBX.com,
which is written in PHP.
It has flexible permissions and modules, so you can turn everything off but
the things you want and create new themes for it.

Or, you can use ESL or just "fs_cli -x " to get info from FreeSWITCH. You
can use those to run the "show calls/channels" and "sofia status profile
internal" and the like.
-Avi Marcus




On Wed, Jul 6, 2011 at 5:56 PM, Alessandro <a.luppi at seletech.com> wrote:

> Hi,
>
> I'm going to develop a custom GUI web-based for freeswitch that allow
> only to add/remove extension and show the call active and the status of
> extensions configured.
> Where can I find in FS the information about actives call and extensions
> status?
>
> Thanks
>
> Alessandro Luppi
>
> --
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu
>
>
> _______________________________________________
> 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 steveayre at gmail.com  Wed Jul  6 19:17:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 6 Jul 2011 16:17:44 +0100
Subject: [Freeswitch-users] Custom GUI for FS
In-Reply-To: <CAA1Qw2ua8_Rsx-f6HZpg_g+k0w=RM-JuT0mrAxJfgce7xXCbDQ@mail.gmail.com>
References: <4E147795.3090700@seletech.com>
	<CAA1Qw2ua8_Rsx-f6HZpg_g+k0w=RM-JuT0mrAxJfgce7xXCbDQ@mail.gmail.com>
Message-ID: <CAFiqYumz+HQwkrDEy187TYjkshvtArveb3g+anLq6j04PiSU9w@mail.gmail.com>

'show calls as xml' and 'show channels as xml' are much better for this sort
of thing, as the XML is easier to parse and will be more reliable (if the
status output formatting changes in a later update then programs might break
unless they're using the xml output).

-Steve



On 6 July 2011 16:05, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Hi,
> Hope, show(show calls, show channels) command will help you.
>
> http://wiki.freeswitch.org/wiki/Mod_commands#show
>
>
>
> On Wed, Jul 6, 2011 at 8:26 PM, Alessandro <a.luppi at seletech.com> wrote:
>
>> Hi,
>>
>> I'm going to develop a custom GUI web-based for freeswitch that allow
>> only to add/remove extension and show the call active and the status of
>> extensions configured.
>> Where can I find in FS the information about actives call and extensions
>> status?
>>
>> Thanks
>>
>> Alessandro Luppi
>>
>> --
>> Ing. Alessandro Luppi
>> Software development
>> Seletech srl
>> Via Collodi 8, 20052 Monza (MI) - Italy
>> Tel: +39.039.5962000 - Fax: +39.039.9716905
>> email: a.luppi at seletech.com - Web: www.seletech.com   or
>> www.seletech.eu
>>
>>
>> _______________________________________________
>> 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
>>
>
>
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> 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 a.luppi at seletech.com  Wed Jul  6 19:18:52 2011
From: a.luppi at seletech.com (Alessandro)
Date: Wed, 06 Jul 2011 17:18:52 +0200
Subject: [Freeswitch-users] Custom GUI for FS
In-Reply-To: <CALNZuwm=RrrJuY6duQJEfw63NVTSUyEhVjahecXb6D-qsgPEUA@mail.gmail.com>
References: <4E147795.3090700@seletech.com>
	<CALNZuwm=RrrJuY6duQJEfw63NVTSUyEhVjahecXb6D-qsgPEUA@mail.gmail.com>
Message-ID: <4E147CDC.3000903@seletech.com>

Ok,

thanks. I'll  try.

Best regards

Alessandro Luppi

Il 06/07/2011 17:08, Avi Marcus ha scritto:
> Hi - this, and much more, was already done in 
> http://www.FusionPBX.com, which is written in PHP.
> It has flexible permissions and modules, so you can turn everything 
> off but the things you want and create new themes for it.
>
> Or, you can use ESL or just "fs_cli -x " to get info from FreeSWITCH. 
> You can use those to run the "show calls/channels" and 
> "sofia status profile internal" and the like.
> -Avi Marcus
>
>
>
>
> On Wed, Jul 6, 2011 at 5:56 PM, Alessandro <a.luppi at seletech.com 
> <mailto:a.luppi at seletech.com>> wrote:
>
>     Hi,
>
>     I'm going to develop a custom GUI web-based for freeswitch that allow
>     only to add/remove extension and show the call active and the
>     status of
>     extensions configured.
>     Where can I find in FS the information about actives call and
>     extensions
>     status?
>
>     Thanks
>
>     Alessandro Luppi
>
>     --
>     Ing. Alessandro Luppi
>     Software development
>     Seletech srl
>     Via Collodi 8, 20052 Monza (MI) - Italy
>     Tel: +39.039.5962000 - Fax: +39.039.9716905
>     email: a.luppi at seletech.com <mailto:a.luppi at seletech.com> - Web:
>     www.seletech.com <http://www.seletech.com>   or www.seletech.eu
>     <http://www.seletech.eu>
>
>
>     _______________________________________________
>     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
>     <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
>
>
>
> _______________________________________________
> 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


-- 
Ing. Alessandro Luppi
Software development
Seletech srl
Via Collodi 8, 20052 Monza (MI) - Italy
Tel: +39.039.5962000 - Fax: +39.039.9716905
email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu

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From steveayre at gmail.com  Wed Jul  6 19:19:53 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 6 Jul 2011 16:19:53 +0100
Subject: [Freeswitch-users] Module_Exists for SpiderMonkey and
 PocketSpinx failing the dialplan at Execution state
In-Reply-To: <CAOcuKucJ4eVn4UhbwmurLjQ2_CxhFJ9=_byO0FOXPwsA7OHiPg@mail.gmail.com>
References: <CAOcuKucJ4eVn4UhbwmurLjQ2_CxhFJ9=_byO0FOXPwsA7OHiPg@mail.gmail.com>
Message-ID: <CAFiqYu=W3jZurDTght5Jb9f18vPPOjr6jhuTuRkQg36ykgje3w@mail.gmail.com>

Silly question, but are you sure they're loaded? Are they any error messages
when you try loading them?

What do the commands 'module_exists mod_spidermonkey' and 'module_exists
mod_pocketsphinx' give you from the fs_cli?

-Steve


On 6 July 2011 14:25, ankIT WALiA <ankitwalia4u at gmail.com> wrote:

> Hi all,
>
> I tried to execute the pizza example for PocketSphinx for speech
> recognition.
>
> I added the grammar,enabled mod_pocketsphinx and spidermonkey, added sound
> files and js files as per the wiki.
> http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Install_.26_Configure
>
> When, I call to the extension, though it passes the Regex, showing true for
> the conditions. But could not execute the javascript application and
> subsequently the js file.
>
> The dialplan looks like *
> <extension name="pizza_demo">
>     <condition field="destination_number" expression="^(74992)$"/>
>     <condition field="${module_exists(mod_spidermonkey)}"
> expression="true"/>
>     <condition field="${module_exists(mod_pocketsphinx)}"
> expression="true">
>     <action application="javascript" data="ps_pizza.js"/>
>     </condition>
>   </extension>*
>
> If I remove the condition of module exist, the js file gets executed and
> the call goes on. But, It does not work with Module exists condition.
>
> I have pasted the logs http://pastebin.freeswitch.org/16680. I could not
> even find anything concrete on the logs.
>
> Thanks
> Ankitx
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>
> _______________________________________________
> 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 steveu at coppice.org  Wed Jul  6 19:44:47 2011
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 06 Jul 2011 23:44:47 +0800
Subject: [Freeswitch-users] Chinese sounds
In-Reply-To: <40057F672A4E43A7971543ECA4C3B97B@gmail.com>
References: <40057F672A4E43A7971543ECA4C3B97B@gmail.com>
Message-ID: <4E1482EF.9070107@coppice.org>

Hi,

Mandarin or a wider selection of Chinese dialects?

On 07/06/2011 02:34 PM, Seven Du wrote:
> Hi,
>
> We got someone in the Chinese Community would like to record a Chinese 
> version of sounds, here's some questions for the community -
>
>
> 1) Where can I find the original English scripts ? It would be hard to 
> listen all sound files
>
> 2) What's base256/*.wav usage?
>
> 3) We'd like to share to world, so, how to upload to 
> files.freeswitch.org when we are done?
>
> 4) I guess the raw recording should be 48K, 16bit, single channel, any 
> other params should we set when recording?
>
>
> Thanks.
>
> -- 
> Seven Du
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
> Sent with Sparrow <http://www.sparrowmailapp.com>
>
>
> _______________________________________________
> 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 a.luppi at seletech.com  Wed Jul  6 19:56:09 2011
From: a.luppi at seletech.com (Alessandro)
Date: Wed, 06 Jul 2011 17:56:09 +0200
Subject: [Freeswitch-users] Custom GUI for FS
In-Reply-To: <CAFiqYumz+HQwkrDEy187TYjkshvtArveb3g+anLq6j04PiSU9w@mail.gmail.com>
References: <4E147795.3090700@seletech.com>	<CAA1Qw2ua8_Rsx-f6HZpg_g+k0w=RM-JuT0mrAxJfgce7xXCbDQ@mail.gmail.com>
	<CAFiqYumz+HQwkrDEy187TYjkshvtArveb3g+anLq6j04PiSU9w@mail.gmail.com>
Message-ID: <4E148599.1080700@seletech.com>

Yes,

I'll use  the command 'show calls as xml' it's better.

Thanks

Best Regards

Alessandro Luppi


Il 06/07/2011 17:17, Steven Ayre ha scritto:
> 'show calls as xml' and 'show channels as xml' are much better for 
> this sort of thing, as the XML is easier to parse and will be more 
> reliable (if the status output formatting changes in a later update 
> then programs might break unless they're using the xml output).
>
> -Steve
>
>
>
> On 6 July 2011 16:05, Nagalenoj H. <nagalenoj at gmail.com 
> <mailto:nagalenoj at gmail.com>> wrote:
>
>     Hi,
>     Hope, show(show calls, show channels) command will help you.
>
>     http://wiki.freeswitch.org/wiki/Mod_commands#show
>
>
>
>     On Wed, Jul 6, 2011 at 8:26 PM, Alessandro <a.luppi at seletech.com
>     <mailto:a.luppi at seletech.com>> wrote:
>
>         Hi,
>
>         I'm going to develop a custom GUI web-based for freeswitch
>         that allow
>         only to add/remove extension and show the call active and the
>         status of
>         extensions configured.
>         Where can I find in FS the information about actives call and
>         extensions
>         status?
>
>         Thanks
>
>         Alessandro Luppi
>
>         --
>         Ing. Alessandro Luppi
>         Software development
>         Seletech srl
>         Via Collodi 8, 20052 Monza (MI) - Italy
>         Tel: +39.039.5962000 - Fax: +39.039.9716905
>         email: a.luppi at seletech.com <mailto:a.luppi at seletech.com> -
>         Web: www.seletech.com <http://www.seletech.com>   or
>         www.seletech.eu <http://www.seletech.eu>
>
>
>         _______________________________________________
>         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
>         <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
>
>
>
>
>     -- 
>     Regards,
>     Nagalenoj H.
>
>     _______________________________________________
>     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
>     <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
>
>
>
> _______________________________________________
> 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


-- 
Ing. Alessandro Luppi
Software development
Seletech srl
Via Collodi 8, 20052 Monza (MI) - Italy
Tel: +39.039.5962000 - Fax: +39.039.9716905
email: a.luppi at seletech.com - Web: www.seletech.com   or   www.seletech.eu

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From umairbari at gmail.com  Wed Jul  6 17:18:06 2011
From: umairbari at gmail.com (Umair Bari)
Date: Wed, 6 Jul 2011 18:18:06 +0500
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <CAKzWOxX2CrDESPMOWMtCgSDxt1R2S3=70aDsTrRM2_p6isfkvA@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
	<163DDA0D-0B14-4623-B48B-5E9EEE907AED@gmail.com>
	<CAFYZzbu9Nh3Ojb26G_pAiFn9MVmm59qeO8BL02_WLX5Fj0mfUg@mail.gmail.com>
	<CAKzWOxX2CrDESPMOWMtCgSDxt1R2S3=70aDsTrRM2_p6isfkvA@mail.gmail.com>
Message-ID: <CAGKOgbK=JzEpWmR8=nFq_x47mFFy96nWRRkMZrogY0hxOMnr2g@mail.gmail.com>

Signup at: https://github.com/signup/free

Create a new repository: https://github.com/repositories/new
Help: http://help.github.com/create-a-repo/

A nice help with images:
Windows: http://help.github.com/win-set-up-git/
Mac: http://help.github.com/mac-set-up-git/
Linux: http://help.github.com/linux-set-up-git/

On Tue, Jul 5, 2011 at 11:21 PM, Michael Collins <msc at freeswitch.org> wrote:

> FYI, Ray is out this week, so for now you might want to throw it up on
> github for now.
> -MC
>
>
> On Mon, Jul 4, 2011 at 3:59 PM, Michel Daggelinckx <
> michel.daggelinckx at gmail.com> wrote:
>
>> Ask intralanman for acces to the FS contrib
>>
>> On Mon, Jul 4, 2011 at 7:10 PM, Mohammad Emran <monemran at gmail.com>wrote:
>>
>>> We can put on google code or git hub.
>>>
>>> Sent from my iPad
>>>
>>> On 4 Jul 2011, at 19:12, David Villasmil <david.villasmil.work at gmail.com>
>>> wrote:
>>>
>>> Hello All,
>>>
>>> I think I'm as ready as i can be to publish this...
>>> Can someone guide me into publishing via GIT?
>>>
>>> thanks
>>>
>>> David
>>>
>>> On Thu, Feb 24, 2011 at 12:39 PM, David Villasmil <<david.villasmil.work at gmail.com>
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>> Hello Guys,
>>>>
>>>> I'm finishing a "complete" wholesale application created on freeswitch
>>>> and I was wondering whether it would be a good idea to put it up on the
>>>> wiki. I just don't know how.
>>>>
>>>> Features include all the following parameters configurable via web
>>>> interface:
>>>>
>>>> - Profile creation based on server IP where traffic is received. You can
>>>> have multiple IPs, system will create multiple profiles/diaplans so it can
>>>> differentiate.
>>>>      - i.e. offer to the same customer a "gold" routing on IP1 and cheap
>>>> routing on IP2
>>>>
>>>> - Customer add/modify/delete
>>>>      - IP source
>>>>      - Rates for client routes based on areacode
>>>>      - Prepaid or postpaid.
>>>>           - When cutomer balance is 0, no more calls are allowed.
>>>>      - limit max channels
>>>>      - Media by-pass
>>>>           - When by-passed, customer and provider will exchange RTPs
>>>> directly. Else, server will be in the middle.
>>>>
>>>> - Provider add/modify/delete
>>>>      - costs for provider routes based on areacode
>>>>      - limit max channels
>>>>
>>>> - Routing based on areacode, gives great granularity.
>>>>
>>>> - Routes can be assigned multiple gateways/providers which can in turn
>>>> be distributed based on weigth. Includes overflow to next configured GW.
>>>>
>>>> - Basic financial report generation (totals) by customer/provider
>>>>
>>>> - Basic traffic ASR/ACD report (totals) by cutomer/provider
>>>>
>>>> - Basic user administration. (No access level, only total access)
>>>>
>>>> - CDR export to csv file.
>>>>
>>>>
>>>>
>>>>
>>>> I also have a prepaid card app... no web interface on that one though...
>>>>
>>>> Thanks all
>>>>
>>>>
>>>> David
>>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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
>
>


-- 

Thanks & Regards,

Umair Bari
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From lautram.mathieu at gmail.com  Wed Jul  6 20:52:14 2011
From: lautram.mathieu at gmail.com (Mathieu Lautram)
Date: Wed, 6 Jul 2011 18:52:14 +0200
Subject: [Freeswitch-users] No ringback with originate
Message-ID: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>

Hi all

I'm facing an issue with originate. Here is the case:

I do an originate to bridge A leg to B leg using bridge application.
The A leg works perfectly but, when the B leg is called, I can't hear
ringtone in the A leg.
I set ignore_early_media to true.

I would like to hear a ringtone without setting the ringback variable with
the tone of my country.

Is it possible to do that?

Thank you in advance.

-- 
Mathieu LAUTRAM
Application developer

BJT Partners - FRANCE
+33 1 79 75 99 60
+33 6 61 59 07 25
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From msc at freeswitch.org  Wed Jul  6 21:13:29 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Jul 2011 10:13:29 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxUrZtrXvJBWPWHX6vv-PPBQ=SCgS2CSwF60GU4_4TEU8Q@mail.gmail.com>

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

Lots of new stuff to talk about!
-MC
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From marketing at cluecon.com  Wed Jul  6 21:14:40 2011
From: marketing at cluecon.com (marketing at cluecon.com)
Date: Wed, 6 Jul 2011 17:14:40 +0000
Subject: [Freeswitch-users] ClueCon News: Free OpenSIPS Training,
	Sponsor Logo - Last Chance, Party RSVP
Message-ID: <000001310071c149-b57bc84c-034d-4e91-bb29-7aacce9f9ef9-000000@email.amazonses.com>

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

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

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

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

-Michael
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From bryansmart at bryansmart.com  Wed Jul  6 21:35:09 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Wed, 6 Jul 2011 13:35:09 -0400
Subject: [Freeswitch-users] Playing multiple files simultaneously
In-Reply-To: <CABNzcD-8JCOgSwUrVp=V8mix5n7Rie6989GkTDWDqCYiaJMteg@mail.gmail.com>
References: <2DB72AF2-2394-419A-ADC5-0B76FE594662@bryansmart.com>
	<EB5BFD1EA1F444079A8BFBAC95976FE1@dell9400>
	<CABNzcD-8JCOgSwUrVp=V8mix5n7Rie6989GkTDWDqCYiaJMteg@mail.gmail.com>
Message-ID: <E467B7C3-E540-45EB-8EFF-1E6707CE0984@bryansmart.com>

When I brought this up originally, I was considering possibilities for voice apps built on FS.

The long-standing user interface model for phone apps is like a voice version of a commandline. The app plays a linear stream of output, and then waits for you to respond. With rare exception, the only interface contexts that are widely used are a menu and a data entry prompt. If speech is involved, that increases the type of data that can be entered, but speech apps are usually the same: menus or data prompts.

With the capability of FS to globally bind digits and detect speech (so to know when to listen with a speech recognizer), it should be possible to create voice apps where digits and speech perform actions by raising events in the app, rather than being exclusively accepted by an input mechanism like a menu or data prompt. So far, I haven't run across voice apps that are designed using an event-driven model, at least in the way that desktop apps are designed. FS has got to be the place to prototype such apps, though. The whole system is event-based.

I know that it takes a stretch of the imagination to think of voice apps working in a new way. The phone has been stuck in a voice version of DOS for 25 or so years (since voicemail got a start as the first big voice app). Speech recognition has expanded possibilities for input, but has not changed the model of interface design.

I'm experimenting with what is possible with FS today. However, a more complex sound playback capability is needed to make a new type of audio interface possible.

An interface context, like a graphical desktop, can contain multiple objects. If a graphical object (icon) needs to notify the user of something, it generates a visual cue (flashes, changes color, adds a badge, and so on). Objects in an audio context need to notify, also, and they would do that by playing cues, speaking, etc. Those cues need to be able to partially overlap when played. New cues certainly shouldn't preempt older cues that haven't had a chance to play yet. Further, certain status cues should be heard, but must overlap currently playing audio, as it wouldn't be desirable for a status cue to interrupt the reading of an important message, for example.

So far, the only mixing environment in FS that I've been able to use is mod_conference, but that isn't ideal. To play multiple audio streams to a conference, it is necessary to connect multiple sessions to it, and treat them each like a sound source. One issue is that conference is meant to relay audio to everyone, and I can't help but think that these sound source sessions drain far too many resources. I've been setting them to deaf, so they should only relay audio in, but they might still be drawing more resources than necessary with detecting digits/speech, and other overhead that is part of every channel (I don't know the full list of what is involved). Plus, it just seems like a mistake to think of a voice app server where every incoming caller has their own conference, plus many sessions that exist to stream files to their conference. Maybe it wouldn't be as bad as it sounds.

I'm still trying to learn about FS internals, so I may have a good bit of the following wrong. Perhaps we could meet our needs by creating a module that exposed a new API command for playing sounds asynchronously on a channel. That way, the existing sound playback would continue to be lightweight. A voice app that worked through the event socket could use this module to handle playback and control of its sounds. Here are basic capabilities:

play: Should start playback of a sound, overlapping any existing sound previously requested through this command that has not finished playing. When playback starts, should return a unique ID for the sound instance (an event?). Optional playback parameters: volume/gain, start off set, loop count.

stop: Stops an existing sound instance from playing (by unique ID), or all sound instances.

change: Change sound instance parameters of an active sound instance: volume/gain, file position/time, loop count.

An event should be raised when any sound instance finishes playing. Should include info about cause: normal stop, manual stop, loop re-trigger, etc.

The module could handle the mixing, but it occurs to me that, looking ahead, it might be better to push a lot of that work out to another tech like OpenAL. After all, at some point, some clients might be connecting with 48Khz stereo audio. OpenAL supports positional audio and effects, in addition to raw mixing. OpenAL is available for several platforms, but not everywhere. Perhaps it would be more important to be widely available, at the cost of features.

Thoughts? Is a module the right way to go? Is using OpenAL a good idea, or should the module handle the mixing?

Bryan

On Jul 6, 2011, at 2:05 AM, broken dash wrote:

> I have been wanting these features and I found this...liquidsoap,
> check out the transcoding section.  I was trying to play multiple
> shoutcast audio streams and hopefully mux them together nicely, etc..
> haven't gotten around to setting freeswitch to utilize it but I'm sure
> it wouldn't be hard.
> 
> http://savonet.sourceforge.net/doc-1.0.0-beta2/cookbook.html
> 
> 
> Cheers,
> Brian
> 
> On Fri, Jul 1, 2011 at 1:49 AM, Jan Berger <jan.berger at video24.no> wrote:
>> If you play several files FS will queue them if it is the same stream - you
>> need to somehow set up a conference with multiple streams (callers).
>> 
>> I would check options around music on hold.
>> 
>> But, another way is to use FS to play several outgoing SIP streams and loop
>> them back into a conference.
>> 
>> 
>> Jan
>> 
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bryan
>> Smart
>> Sent: 1. juli 2011 07:13
>> To: FreeSWITCH Users Help
>> Subject: [Freeswitch-users] Playing multiple files simultaneously
>> 
>> Is it possible for Freeswitch to play more than one file to a channel at a
>> time? What I've seen and tried from the dialplan and scripts either queues
>> files to play, or will stop a currently playing file so that the newly
>> requested file will play. This also seems to be the case in conferences.
>> When I send multiple play commands to conferences, the files are queued.
>> 
>> As for how this might be used, think of an IVR that plays queued prompts,
>> yet continuously plays looping music or a Shoutcast stream in the
>> background. I also want to be able to play short cue tones that start at the
>> same time as a prompt (don't want to pre-mix them in to a single file,
>> though).
>> 
>> Is this currently possible through any means? Perhaps with the event socket?
>> 
>> Bryan
>> 
>> 
>> _______________________________________________
>> 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
>> 
>> 
>> _______________________________________________
>> 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
>> 
> 
> _______________________________________________
> 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 michel.daggelinckx at gmail.com  Wed Jul  6 21:37:57 2011
From: michel.daggelinckx at gmail.com (Michel Daggelinckx)
Date: Wed, 6 Jul 2011 19:37:57 +0200
Subject: [Freeswitch-users] Configuration Server
In-Reply-To: <CAAYf2Lvv1f8QBmDL9scJm45KcDA7CoVbCmGOcSwBvAH8bnjMVA@mail.gmail.com>
References: <CAAYf2Lvv1f8QBmDL9scJm45KcDA7CoVbCmGOcSwBvAH8bnjMVA@mail.gmail.com>
Message-ID: <CAFYZzbvVzfRt60neJC+DyDQMpsH+iWt7Ubv6YWALjN1fdf4fbw@mail.gmail.com>

hi,
I can do testing

On Wed, Jul 6, 2011 at 4:09 PM, Roger Castaldo <roger.castaldo at gmail.com>wrote:

> Hi everyone, I have been following freeswitch for quite some time now and
> working on an ever evolving configuration server for it, which has now taken
> yet another branch, that instead of writing xml configuration files now is
> attempting to control the call flow itself using the outbound socket.
> Currently it is a closed source project, originally because of some of my
> closed source libraries I was using to develop it, but since they have
> become open source I was wondering if there is any interest in working with
> me on the development/testing of the product.  It is written in C# and is
> entirely web based, using ajax calls and web services for everything.  I am
> using a Relational Mapping library of my own design to map the data to the
> database (currently a firebirdsql database).
>
> _______________________________________________
> 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 jan.berger at video24.no  Wed Jul  6 23:15:40 2011
From: jan.berger at video24.no (Jan Berger)
Date: Wed, 6 Jul 2011 21:15:40 +0200
Subject: [Freeswitch-users] Playing multiple files simultaneously
In-Reply-To: <E467B7C3-E540-45EB-8EFF-1E6707CE0984@bryansmart.com>
References: <2DB72AF2-2394-419A-ADC5-0B76FE594662@bryansmart.com>
	<EB5BFD1EA1F444079A8BFBAC95976FE1@dell9400>
	<CABNzcD-8JCOgSwUrVp=V8mix5n7Rie6989GkTDWDqCYiaJMteg@mail.gmail.com>
	<E467B7C3-E540-45EB-8EFF-1E6707CE0984@bryansmart.com>
Message-ID: <324D439C494A474BA40EF7C284238D21@dell9400>

Brian,

An asynchronous IVR is not exactly news. But, many of the IVR engines
implement blocking schemes on top of an asynchronous engine to simplify IVR
scripting. FS support both - I am not sure if it support all the required
events on speech, but that can be added if not. I need them as well, so your
not alone.

Sockets and ESL might be what you are looking for.

Jan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bryan
Smart
Sent: 6. juli 2011 19:35
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Playing multiple files simultaneously

When I brought this up originally, I was considering possibilities for voice
apps built on FS.

The long-standing user interface model for phone apps is like a voice
version of a commandline. The app plays a linear stream of output, and then
waits for you to respond. With rare exception, the only interface contexts
that are widely used are a menu and a data entry prompt. If speech is
involved, that increases the type of data that can be entered, but speech
apps are usually the same: menus or data prompts.

With the capability of FS to globally bind digits and detect speech (so to
know when to listen with a speech recognizer), it should be possible to
create voice apps where digits and speech perform actions by raising events
in the app, rather than being exclusively accepted by an input mechanism
like a menu or data prompt. So far, I haven't run across voice apps that are
designed using an event-driven model, at least in the way that desktop apps
are designed. FS has got to be the place to prototype such apps, though. The
whole system is event-based.

I know that it takes a stretch of the imagination to think of voice apps
working in a new way. The phone has been stuck in a voice version of DOS for
25 or so years (since voicemail got a start as the first big voice app).
Speech recognition has expanded possibilities for input, but has not changed
the model of interface design.

I'm experimenting with what is possible with FS today. However, a more
complex sound playback capability is needed to make a new type of audio
interface possible.

An interface context, like a graphical desktop, can contain multiple
objects. If a graphical object (icon) needs to notify the user of something,
it generates a visual cue (flashes, changes color, adds a badge, and so on).
Objects in an audio context need to notify, also, and they would do that by
playing cues, speaking, etc. Those cues need to be able to partially overlap
when played. New cues certainly shouldn't preempt older cues that haven't
had a chance to play yet. Further, certain status cues should be heard, but
must overlap currently playing audio, as it wouldn't be desirable for a
status cue to interrupt the reading of an important message, for example.

So far, the only mixing environment in FS that I've been able to use is
mod_conference, but that isn't ideal. To play multiple audio streams to a
conference, it is necessary to connect multiple sessions to it, and treat
them each like a sound source. One issue is that conference is meant to
relay audio to everyone, and I can't help but think that these sound source
sessions drain far too many resources. I've been setting them to deaf, so
they should only relay audio in, but they might still be drawing more
resources than necessary with detecting digits/speech, and other overhead
that is part of every channel (I don't know the full list of what is
involved). Plus, it just seems like a mistake to think of a voice app server
where every incoming caller has their own conference, plus many sessions
that exist to stream files to their conference. Maybe it wouldn't be as bad
as it sounds.

I'm still trying to learn about FS internals, so I may have a good bit of
the following wrong. Perhaps we could meet our needs by creating a module
that exposed a new API command for playing sounds asynchronously on a
channel. That way, the existing sound playback would continue to be
lightweight. A voice app that worked through the event socket could use this
module to handle playback and control of its sounds. Here are basic
capabilities:

play: Should start playback of a sound, overlapping any existing sound
previously requested through this command that has not finished playing.
When playback starts, should return a unique ID for the sound instance (an
event?). Optional playback parameters: volume/gain, start off set, loop
count.

stop: Stops an existing sound instance from playing (by unique ID), or all
sound instances.

change: Change sound instance parameters of an active sound instance:
volume/gain, file position/time, loop count.

An event should be raised when any sound instance finishes playing. Should
include info about cause: normal stop, manual stop, loop re-trigger, etc.

The module could handle the mixing, but it occurs to me that, looking ahead,
it might be better to push a lot of that work out to another tech like
OpenAL. After all, at some point, some clients might be connecting with
48Khz stereo audio. OpenAL supports positional audio and effects, in
addition to raw mixing. OpenAL is available for several platforms, but not
everywhere. Perhaps it would be more important to be widely available, at
the cost of features.

Thoughts? Is a module the right way to go? Is using OpenAL a good idea, or
should the module handle the mixing?

Bryan

On Jul 6, 2011, at 2:05 AM, broken dash wrote:

> I have been wanting these features and I found this...liquidsoap,
> check out the transcoding section.  I was trying to play multiple
> shoutcast audio streams and hopefully mux them together nicely, etc..
> haven't gotten around to setting freeswitch to utilize it but I'm sure
> it wouldn't be hard.
> 
> http://savonet.sourceforge.net/doc-1.0.0-beta2/cookbook.html
> 
> 
> Cheers,
> Brian
> 
> On Fri, Jul 1, 2011 at 1:49 AM, Jan Berger <jan.berger at video24.no> wrote:
>> If you play several files FS will queue them if it is the same stream -
you
>> need to somehow set up a conference with multiple streams (callers).
>> 
>> I would check options around music on hold.
>> 
>> But, another way is to use FS to play several outgoing SIP streams and
loop
>> them back into a conference.
>> 
>> 
>> Jan
>> 
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bryan
>> Smart
>> Sent: 1. juli 2011 07:13
>> To: FreeSWITCH Users Help
>> Subject: [Freeswitch-users] Playing multiple files simultaneously
>> 
>> Is it possible for Freeswitch to play more than one file to a channel at
a
>> time? What I've seen and tried from the dialplan and scripts either
queues
>> files to play, or will stop a currently playing file so that the newly
>> requested file will play. This also seems to be the case in conferences.
>> When I send multiple play commands to conferences, the files are queued.
>> 
>> As for how this might be used, think of an IVR that plays queued prompts,
>> yet continuously plays looping music or a Shoutcast stream in the
>> background. I also want to be able to play short cue tones that start at
the
>> same time as a prompt (don't want to pre-mix them in to a single file,
>> though).
>> 
>> Is this currently possible through any means? Perhaps with the event
socket?
>> 
>> Bryan
>> 
>> 
>> _______________________________________________
>> 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
>> 
>> 
>> _______________________________________________
>> 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
>> 
> 
> _______________________________________________
> 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


_______________________________________________
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 fs-list at communicatefreely.net  Wed Jul  6 23:18:33 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Wed, 06 Jul 2011 15:18:33 -0400
Subject: [Freeswitch-users] NOTIFY event not sent to user
Message-ID: <4E14B509.6000108@communicatefreely.net>

Hello,

I'm trying to send notify events to Aastra phones for screen pops and 
that sort of thing.  I'm triggering the event using ESL from a PHP 
script.  If I subscribe to the NOTIFY events on fs_cli, I can see my 
event coming in quite nicely, but nothing gets sent to the phone.  I am 
doing a packet capture between the phone and Freeswitch, and although I 
can see all the regular call setup and MWI packets, I don't get my 
notify packets coming through.

Here's the event as it is seen on fs_cli:

RECV EVENT
Core-UUID: c8db9457-8e51-e011-8f26-001517ac7a54
FreeSWITCH-Hostname: stefan.151front.communicatefreely.net
FreeSWITCH-IPv4: 66.207.200.21
FreeSWITCH-IPv6: ::1
Event-Date-Local: 2011-07-06 15:08:28
Event-Date-GMT: Wed, 06 Jul 2011 19:08:28 GMT
Event-Date-Timestamp: 1309979308360046
Event-Calling-File: mod_event_socket.c
Event-Calling-Function: read_packet
Event-Calling-Line-Number: 1131
Command: sendevent NOTIFY
Event-Name: NOTIFY
profile: internal
event-string: aastra-xml
user: 5101 at communicatefreely.net
host: stefan.151front.communicatefreely.net
content-type: application/xml
Content-Length: 223
Content-Length: 223

<AastraIPPhoneStatus Beep="yes" triggerDestroyOnExit="no">
        <Session>1</Session>
        <Message Index="test" Type="alert" Timeout="15">
                Hello, anyone there?
        </Message>
</AastraIPPhoneStatus>

freeswitch at default>

Just for background, if I do sofia status profile internal user 
5101 at communicatefreely.net I get all the details about the phone, and 
the username and hostname's match what's in the packet above.

Any suggestions?



From krice at freeswitch.org  Wed Jul  6 23:36:10 2011
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 06 Jul 2011 14:36:10 -0500
Subject: [Freeswitch-users] Ok I gotta share this...
Message-ID: <CA3A235A.18A71%krice@freeswitch.org>

Looks like Lanman and some other guys at Cuda had an Office Space minute

http://www.youtube.com/watch?v=CGL43sk16XE

K 
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From infos at madovsky.org  Wed Jul  6 23:42:21 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 6 Jul 2011 15:42:21 -0400
Subject: [Freeswitch-users] Ok I gotta share this...
References: <CA3A235A.18A71%krice@freeswitch.org>
Message-ID: <184D9C64DA61467F9F47582AA4345F97@e1705>

Ok I gotta share this...finally a great decision ! :)
  ----- Original Message ----- 
  From: Ken Rice 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, July 06, 2011 3:36 PM
  Subject: [Freeswitch-users] Ok I gotta share this...


  Looks like Lanman and some other guys at Cuda had an Office Space minute

  http://www.youtube.com/watch?v=CGL43sk16XE

  K 


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


  _______________________________________________
  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
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From bryansmart at bryansmart.com  Wed Jul  6 23:33:39 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Wed, 6 Jul 2011 15:33:39 -0400
Subject: [Freeswitch-users] Playing multiple files simultaneously
In-Reply-To: <324D439C494A474BA40EF7C284238D21@dell9400>
References: <2DB72AF2-2394-419A-ADC5-0B76FE594662@bryansmart.com>
	<EB5BFD1EA1F444079A8BFBAC95976FE1@dell9400>
	<CABNzcD-8JCOgSwUrVp=V8mix5n7Rie6989GkTDWDqCYiaJMteg@mail.gmail.com>
	<E467B7C3-E540-45EB-8EFF-1E6707CE0984@bryansmart.com>
	<324D439C494A474BA40EF7C284238D21@dell9400>
Message-ID: <B28516A2-2000-4344-9B45-A115429627A0@bryansmart.com>

Do you mean non-blocking IVR in the sense that speech/digits interrupt the prompts, or that the IVR app runs in a loop, handling other tasks  and sending audio output on the channel, until the user raises an event with digits or speech?

If the second, then it seems like layered audio would be a necessary capability. You wouldn't have as much control of when information is presented, so, with only being able to play a single sound/prompt at a time, there is potential for sounds interrupting each other, or else a long queue of them accumulating.

Bryan

On Jul 6, 2011, at 3:15 PM, Jan Berger wrote:

> Brian,
> 
> An asynchronous IVR is not exactly news. But, many of the IVR engines
> implement blocking schemes on top of an asynchronous engine to simplify IVR
> scripting. FS support both - I am not sure if it support all the required
> events on speech, but that can be added if not. I need them as well, so your
> not alone.
> 
> Sockets and ESL might be what you are looking for.
> 
> Jan
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bryan
> Smart
> Sent: 6. juli 2011 19:35
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Playing multiple files simultaneously
> 
> When I brought this up originally, I was considering possibilities for voice
> apps built on FS.
> 
> The long-standing user interface model for phone apps is like a voice
> version of a commandline. The app plays a linear stream of output, and then
> waits for you to respond. With rare exception, the only interface contexts
> that are widely used are a menu and a data entry prompt. If speech is
> involved, that increases the type of data that can be entered, but speech
> apps are usually the same: menus or data prompts.
> 
> With the capability of FS to globally bind digits and detect speech (so to
> know when to listen with a speech recognizer), it should be possible to
> create voice apps where digits and speech perform actions by raising events
> in the app, rather than being exclusively accepted by an input mechanism
> like a menu or data prompt. So far, I haven't run across voice apps that are
> designed using an event-driven model, at least in the way that desktop apps
> are designed. FS has got to be the place to prototype such apps, though. The
> whole system is event-based.
> 
> I know that it takes a stretch of the imagination to think of voice apps
> working in a new way. The phone has been stuck in a voice version of DOS for
> 25 or so years (since voicemail got a start as the first big voice app).
> Speech recognition has expanded possibilities for input, but has not changed
> the model of interface design.
> 
> I'm experimenting with what is possible with FS today. However, a more
> complex sound playback capability is needed to make a new type of audio
> interface possible.
> 
> An interface context, like a graphical desktop, can contain multiple
> objects. If a graphical object (icon) needs to notify the user of something,
> it generates a visual cue (flashes, changes color, adds a badge, and so on).
> Objects in an audio context need to notify, also, and they would do that by
> playing cues, speaking, etc. Those cues need to be able to partially overlap
> when played. New cues certainly shouldn't preempt older cues that haven't
> had a chance to play yet. Further, certain status cues should be heard, but
> must overlap currently playing audio, as it wouldn't be desirable for a
> status cue to interrupt the reading of an important message, for example.
> 
> So far, the only mixing environment in FS that I've been able to use is
> mod_conference, but that isn't ideal. To play multiple audio streams to a
> conference, it is necessary to connect multiple sessions to it, and treat
> them each like a sound source. One issue is that conference is meant to
> relay audio to everyone, and I can't help but think that these sound source
> sessions drain far too many resources. I've been setting them to deaf, so
> they should only relay audio in, but they might still be drawing more
> resources than necessary with detecting digits/speech, and other overhead
> that is part of every channel (I don't know the full list of what is
> involved). Plus, it just seems like a mistake to think of a voice app server
> where every incoming caller has their own conference, plus many sessions
> that exist to stream files to their conference. Maybe it wouldn't be as bad
> as it sounds.
> 
> I'm still trying to learn about FS internals, so I may have a good bit of
> the following wrong. Perhaps we could meet our needs by creating a module
> that exposed a new API command for playing sounds asynchronously on a
> channel. That way, the existing sound playback would continue to be
> lightweight. A voice app that worked through the event socket could use this
> module to handle playback and control of its sounds. Here are basic
> capabilities:
> 
> play: Should start playback of a sound, overlapping any existing sound
> previously requested through this command that has not finished playing.
> When playback starts, should return a unique ID for the sound instance (an
> event?). Optional playback parameters: volume/gain, start off set, loop
> count.
> 
> stop: Stops an existing sound instance from playing (by unique ID), or all
> sound instances.
> 
> change: Change sound instance parameters of an active sound instance:
> volume/gain, file position/time, loop count.
> 
> An event should be raised when any sound instance finishes playing. Should
> include info about cause: normal stop, manual stop, loop re-trigger, etc.
> 
> The module could handle the mixing, but it occurs to me that, looking ahead,
> it might be better to push a lot of that work out to another tech like
> OpenAL. After all, at some point, some clients might be connecting with
> 48Khz stereo audio. OpenAL supports positional audio and effects, in
> addition to raw mixing. OpenAL is available for several platforms, but not
> everywhere. Perhaps it would be more important to be widely available, at
> the cost of features.
> 
> Thoughts? Is a module the right way to go? Is using OpenAL a good idea, or
> should the module handle the mixing?
> 
> Bryan
> 
> On Jul 6, 2011, at 2:05 AM, broken dash wrote:
> 
>> I have been wanting these features and I found this...liquidsoap,
>> check out the transcoding section.  I was trying to play multiple
>> shoutcast audio streams and hopefully mux them together nicely, etc..
>> haven't gotten around to setting freeswitch to utilize it but I'm sure
>> it wouldn't be hard.
>> 
>> http://savonet.sourceforge.net/doc-1.0.0-beta2/cookbook.html
>> 
>> 
>> Cheers,
>> Brian
>> 
>> On Fri, Jul 1, 2011 at 1:49 AM, Jan Berger <jan.berger at video24.no> wrote:
>>> If you play several files FS will queue them if it is the same stream -
> you
>>> need to somehow set up a conference with multiple streams (callers).
>>> 
>>> I would check options around music on hold.
>>> 
>>> But, another way is to use FS to play several outgoing SIP streams and
> loop
>>> them back into a conference.
>>> 
>>> 
>>> Jan
>>> 
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bryan
>>> Smart
>>> Sent: 1. juli 2011 07:13
>>> To: FreeSWITCH Users Help
>>> Subject: [Freeswitch-users] Playing multiple files simultaneously
>>> 
>>> Is it possible for Freeswitch to play more than one file to a channel at
> a
>>> time? What I've seen and tried from the dialplan and scripts either
> queues
>>> files to play, or will stop a currently playing file so that the newly
>>> requested file will play. This also seems to be the case in conferences.
>>> When I send multiple play commands to conferences, the files are queued.
>>> 
>>> As for how this might be used, think of an IVR that plays queued prompts,
>>> yet continuously plays looping music or a Shoutcast stream in the
>>> background. I also want to be able to play short cue tones that start at
> the
>>> same time as a prompt (don't want to pre-mix them in to a single file,
>>> though).
>>> 
>>> Is this currently possible through any means? Perhaps with the event
> socket?
>>> 
>>> Bryan
>>> 
>>> 
>>> _______________________________________________
>>> 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
>>> 
>>> 
>>> _______________________________________________
>>> 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
>>> 
>> 
>> _______________________________________________
>> 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
> 
> 
> _______________________________________________
> 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
> 
> 
> _______________________________________________
> 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 infos at madovsky.org  Wed Jul  6 23:44:26 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 6 Jul 2011 15:44:26 -0400
Subject: [Freeswitch-users] Ok I gotta share this...
References: <CA3A235A.18A71%krice@freeswitch.org>
Message-ID: <95E60B128121421BA5D7213E7455E9E2@e1705>

Ok I gotta share this...MMhmm no cellphones neither i-phone in the videoclip :(
  ----- Original Message ----- 
  From: Ken Rice 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, July 06, 2011 3:36 PM
  Subject: [Freeswitch-users] Ok I gotta share this...


  Looks like Lanman and some other guys at Cuda had an Office Space minute

  http://www.youtube.com/watch?v=CGL43sk16XE

  K 


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


  _______________________________________________
  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
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From fs-list at communicatefreely.net  Thu Jul  7 00:58:07 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Wed, 06 Jul 2011 16:58:07 -0400
Subject: [Freeswitch-users] NOTIFY event not sent to user
In-Reply-To: <4E14B509.6000108@communicatefreely.net>
References: <4E14B509.6000108@communicatefreely.net>
Message-ID: <4E14CC5F.60205@communicatefreely.net>

I may have made some progress here, but I'm coming up on additional issues.

If I add a Call-ID header to the event that matches the subscription, it 
works.  This is a bit of a pain.  Any way to make it send to all the 
registrations for that user?

Also, in the resulting notify, I get this extra line:

Subscription-State:terminated;reason=timeout

Any way to disable that?  The phone comes back with "Bad Event", and I 
wonder if that header has something to do with it.

Thanks!

-Tim

Tim St. Pierre wrote:
> Hello,
>
> I'm trying to send notify events to Aastra phones for screen pops and 
> that sort of thing.  I'm triggering the event using ESL from a PHP 
> script.  If I subscribe to the NOTIFY events on fs_cli, I can see my 
> event coming in quite nicely, but nothing gets sent to the phone.  I am 
> doing a packet capture between the phone and Freeswitch, and although I 
> can see all the regular call setup and MWI packets, I don't get my 
> notify packets coming through.
>
> Here's the event as it is seen on fs_cli:
>
> RECV EVENT
> Core-UUID: c8db9457-8e51-e011-8f26-001517ac7a54
> FreeSWITCH-Hostname: stefan.151front.communicatefreely.net
> FreeSWITCH-IPv4: 66.207.200.21
> FreeSWITCH-IPv6: ::1
> Event-Date-Local: 2011-07-06 15:08:28
> Event-Date-GMT: Wed, 06 Jul 2011 19:08:28 GMT
> Event-Date-Timestamp: 1309979308360046
> Event-Calling-File: mod_event_socket.c
> Event-Calling-Function: read_packet
> Event-Calling-Line-Number: 1131
> Command: sendevent NOTIFY
> Event-Name: NOTIFY
> profile: internal
> event-string: aastra-xml
> user: 5101 at communicatefreely.net
> host: stefan.151front.communicatefreely.net
> content-type: application/xml
> Content-Length: 223
> Content-Length: 223
>
> <AastraIPPhoneStatus Beep="yes" triggerDestroyOnExit="no">
>         <Session>1</Session>
>         <Message Index="test" Type="alert" Timeout="15">
>                 Hello, anyone there?
>         </Message>
> </AastraIPPhoneStatus>
>
> freeswitch at default>
>
> Just for background, if I do sofia status profile internal user 
> 5101 at communicatefreely.net I get all the details about the phone, and 
> the username and hostname's match what's in the packet above.
>
> Any suggestions?
>
>
> _______________________________________________
> 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 jan.berger at video24.no  Thu Jul  7 01:06:42 2011
From: jan.berger at video24.no (Jan Berger)
Date: Wed, 6 Jul 2011 23:06:42 +0200
Subject: [Freeswitch-users] Playing multiple files simultaneously
In-Reply-To: <B28516A2-2000-4344-9B45-A115429627A0@bryansmart.com>
References: <2DB72AF2-2394-419A-ADC5-0B76FE594662@bryansmart.com>
	<EB5BFD1EA1F444079A8BFBAC95976FE1@dell9400>
	<CABNzcD-8JCOgSwUrVp=V8mix5n7Rie6989GkTDWDqCYiaJMteg@mail.gmail.com>
	<E467B7C3-E540-45EB-8EFF-1E6707CE0984@bryansmart.com>
	<324D439C494A474BA40EF7C284238D21@dell9400>
	<B28516A2-2000-4344-9B45-A115429627A0@bryansmart.com>
Message-ID: <67CD5F9974DF42FBB7E8BF917E21451B@dell9400>

The later, the first is probably what I call "barge-in" unless I
misunderstand you? You can do that as well.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bryan
Smart
Sent: 6. juli 2011 21:34
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Playing multiple files simultaneously

Do you mean non-blocking IVR in the sense that speech/digits interrupt the
prompts, or that the IVR app runs in a loop, handling other tasks  and
sending audio output on the channel, until the user raises an event with
digits or speech?

If the second, then it seems like layered audio would be a necessary
capability. You wouldn't have as much control of when information is
presented, so, with only being able to play a single sound/prompt at a time,
there is potential for sounds interrupting each other, or else a long queue
of them accumulating.

Bryan

On Jul 6, 2011, at 3:15 PM, Jan Berger wrote:

> Brian,
> 
> An asynchronous IVR is not exactly news. But, many of the IVR engines
> implement blocking schemes on top of an asynchronous engine to simplify
IVR
> scripting. FS support both - I am not sure if it support all the required
> events on speech, but that can be added if not. I need them as well, so
your
> not alone.
> 
> Sockets and ESL might be what you are looking for.
> 
> Jan
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bryan
> Smart
> Sent: 6. juli 2011 19:35
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Playing multiple files simultaneously
> 
> When I brought this up originally, I was considering possibilities for
voice
> apps built on FS.
> 
> The long-standing user interface model for phone apps is like a voice
> version of a commandline. The app plays a linear stream of output, and
then
> waits for you to respond. With rare exception, the only interface contexts
> that are widely used are a menu and a data entry prompt. If speech is
> involved, that increases the type of data that can be entered, but speech
> apps are usually the same: menus or data prompts.
> 
> With the capability of FS to globally bind digits and detect speech (so to
> know when to listen with a speech recognizer), it should be possible to
> create voice apps where digits and speech perform actions by raising
events
> in the app, rather than being exclusively accepted by an input mechanism
> like a menu or data prompt. So far, I haven't run across voice apps that
are
> designed using an event-driven model, at least in the way that desktop
apps
> are designed. FS has got to be the place to prototype such apps, though.
The
> whole system is event-based.
> 
> I know that it takes a stretch of the imagination to think of voice apps
> working in a new way. The phone has been stuck in a voice version of DOS
for
> 25 or so years (since voicemail got a start as the first big voice app).
> Speech recognition has expanded possibilities for input, but has not
changed
> the model of interface design.
> 
> I'm experimenting with what is possible with FS today. However, a more
> complex sound playback capability is needed to make a new type of audio
> interface possible.
> 
> An interface context, like a graphical desktop, can contain multiple
> objects. If a graphical object (icon) needs to notify the user of
something,
> it generates a visual cue (flashes, changes color, adds a badge, and so
on).
> Objects in an audio context need to notify, also, and they would do that
by
> playing cues, speaking, etc. Those cues need to be able to partially
overlap
> when played. New cues certainly shouldn't preempt older cues that haven't
> had a chance to play yet. Further, certain status cues should be heard,
but
> must overlap currently playing audio, as it wouldn't be desirable for a
> status cue to interrupt the reading of an important message, for example.
> 
> So far, the only mixing environment in FS that I've been able to use is
> mod_conference, but that isn't ideal. To play multiple audio streams to a
> conference, it is necessary to connect multiple sessions to it, and treat
> them each like a sound source. One issue is that conference is meant to
> relay audio to everyone, and I can't help but think that these sound
source
> sessions drain far too many resources. I've been setting them to deaf, so
> they should only relay audio in, but they might still be drawing more
> resources than necessary with detecting digits/speech, and other overhead
> that is part of every channel (I don't know the full list of what is
> involved). Plus, it just seems like a mistake to think of a voice app
server
> where every incoming caller has their own conference, plus many sessions
> that exist to stream files to their conference. Maybe it wouldn't be as
bad
> as it sounds.
> 
> I'm still trying to learn about FS internals, so I may have a good bit of
> the following wrong. Perhaps we could meet our needs by creating a module
> that exposed a new API command for playing sounds asynchronously on a
> channel. That way, the existing sound playback would continue to be
> lightweight. A voice app that worked through the event socket could use
this
> module to handle playback and control of its sounds. Here are basic
> capabilities:
> 
> play: Should start playback of a sound, overlapping any existing sound
> previously requested through this command that has not finished playing.
> When playback starts, should return a unique ID for the sound instance (an
> event?). Optional playback parameters: volume/gain, start off set, loop
> count.
> 
> stop: Stops an existing sound instance from playing (by unique ID), or all
> sound instances.
> 
> change: Change sound instance parameters of an active sound instance:
> volume/gain, file position/time, loop count.
> 
> An event should be raised when any sound instance finishes playing. Should
> include info about cause: normal stop, manual stop, loop re-trigger, etc.
> 
> The module could handle the mixing, but it occurs to me that, looking
ahead,
> it might be better to push a lot of that work out to another tech like
> OpenAL. After all, at some point, some clients might be connecting with
> 48Khz stereo audio. OpenAL supports positional audio and effects, in
> addition to raw mixing. OpenAL is available for several platforms, but not
> everywhere. Perhaps it would be more important to be widely available, at
> the cost of features.
> 
> Thoughts? Is a module the right way to go? Is using OpenAL a good idea, or
> should the module handle the mixing?
> 
> Bryan
> 
> On Jul 6, 2011, at 2:05 AM, broken dash wrote:
> 
>> I have been wanting these features and I found this...liquidsoap,
>> check out the transcoding section.  I was trying to play multiple
>> shoutcast audio streams and hopefully mux them together nicely, etc..
>> haven't gotten around to setting freeswitch to utilize it but I'm sure
>> it wouldn't be hard.
>> 
>> http://savonet.sourceforge.net/doc-1.0.0-beta2/cookbook.html
>> 
>> 
>> Cheers,
>> Brian
>> 
>> On Fri, Jul 1, 2011 at 1:49 AM, Jan Berger <jan.berger at video24.no> wrote:
>>> If you play several files FS will queue them if it is the same stream -
> you
>>> need to somehow set up a conference with multiple streams (callers).
>>> 
>>> I would check options around music on hold.
>>> 
>>> But, another way is to use FS to play several outgoing SIP streams and
> loop
>>> them back into a conference.
>>> 
>>> 
>>> Jan
>>> 
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Bryan
>>> Smart
>>> Sent: 1. juli 2011 07:13
>>> To: FreeSWITCH Users Help
>>> Subject: [Freeswitch-users] Playing multiple files simultaneously
>>> 
>>> Is it possible for Freeswitch to play more than one file to a channel at
> a
>>> time? What I've seen and tried from the dialplan and scripts either
> queues
>>> files to play, or will stop a currently playing file so that the newly
>>> requested file will play. This also seems to be the case in conferences.
>>> When I send multiple play commands to conferences, the files are queued.
>>> 
>>> As for how this might be used, think of an IVR that plays queued
prompts,
>>> yet continuously plays looping music or a Shoutcast stream in the
>>> background. I also want to be able to play short cue tones that start at
> the
>>> same time as a prompt (don't want to pre-mix them in to a single file,
>>> though).
>>> 
>>> Is this currently possible through any means? Perhaps with the event
> socket?
>>> 
>>> Bryan
>>> 
>>> 
>>> _______________________________________________
>>> 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
>>> 
>>> 
>>> _______________________________________________
>>> 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
>>> 
>> 
>> _______________________________________________
>> 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
> 
> 
> _______________________________________________
> 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
> 
> 
> _______________________________________________
> 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


_______________________________________________
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 curriegrad2004 at gmail.com  Thu Jul  7 01:47:52 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 6 Jul 2011 14:47:52 -0700
Subject: [Freeswitch-users] Ok I gotta share this...
In-Reply-To: <95E60B128121421BA5D7213E7455E9E2@e1705>
References: <CA3A235A.18A71%krice@freeswitch.org>
	<95E60B128121421BA5D7213E7455E9E2@e1705>
Message-ID: <CAMWiMy8E3Q1H1RFH8ih8CWHo6QNyYUjehpKLKqYwpobLuRkjFQ@mail.gmail.com>

Can't somebody re-write the firmware so... We don't have to destroy 'em all?

On Wed, Jul 6, 2011 at 12:44 PM, Madovsky <infos at madovsky.org> wrote:
> MMhmm no cellphones neither i-phone in the videoclip :(
>
> ----- Original Message -----
> From: Ken Rice
> To: FreeSWITCH Users Help
> Sent: Wednesday, July 06, 2011 3:36 PM
> Subject: [Freeswitch-users] Ok I gotta share this...
> Looks like Lanman and some other guys at Cuda had an Office Space minute
>
> http://www.youtube.com/watch?v=CGL43sk16XE
>
> K
>
> ________________________________
>
> _______________________________________________
> 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
>
> _______________________________________________
> 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 jerry.richards at teotech.com  Thu Jul  7 02:43:37 2011
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 6 Jul 2011 15:43:37 -0700
Subject: [Freeswitch-users] Call Dropped When Answered If Bridged To
 Internal Extension And PSTN Number Via
In-Reply-To: <BANLkTimkYK-FBC4PVVfsAOkCYJ1TcWVaOg@mail.gmail.com>
References: <2BF7FB90DF25EA4485949F3AF2B9D69624F481C617@VA3DIAXVS351.RED001.local>
	<BANLkTikMpW_C3YuJ7LBzyfEsFt5fdZGX4Q@mail.gmail.com>
	<2BF7FB90DF25EA4485949F3AF2B9D69624F481C95A@VA3DIAXVS351.RED001.local>
	<BANLkTimkYK-FBC4PVVfsAOkCYJ1TcWVaOg@mail.gmail.com>
Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D69624F48EA75A@VA3DIAXVS351.RED001.local>

MC,

Sorry for the late reply.  I simplified the scenario so extension 2002 calls just one internal endpoint 2001 at 192.168.72.133<mailto:2001 at 192.168.72.133> and one PSTN destination 4253491028.  I posted two traces: 1) Where the internal extension is answered and the call immediately disconnects (http://pastebin.freeswitch.org/16688), and 2) where the PSTN number is answered and the call is successfully connected (http://pastebin.freeswitch.org/16687).

Any guidance or ideas you may have are appreciated.

Thanks,
Jerry


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, July 01, 2011 1:54 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call Dropped When Answered If Bridged To Internal Extension And PSTN Number Via

So it looks like 2001 has multiple registrations, is that correct? It looks like it is calling 3 endpoints:
2001 at 192.168.73.120<mailto:2001 at 192.168.73.120>
2001 at 192.168.72.87<mailto:2001 at 192.168.72.87>
2001 at 192.168.73.125<mailto:2001 at 192.168.73.125>

I'm just curious - what happens if either of the other SIP endpoints answers this call?
-MC
On Fri, Jul 1, 2011 at 10:50 AM, Jerry Richards <jerry.richards at teotech.com<mailto:jerry.richards at teotech.com>> wrote:
MC,

I have two internal extensions (2001 and 2002) and a PRI connected to the PSTN.  Additionally, the dialplan is configured so when 2001 is called, it will bridge to two destinations (internal extension 2001 and PSTN number 4253491028).

Here is the detailed scenario of http://pastebin.freeswitch.org/16634:

Line: Description
0005: At internal 2002, call internal 2001
0490: Execute this bridge statement:
bridge({ignore_early_media=ring_ready}{presence_id=2001 at 192.168.72.79<mailto:2001 at 192.168.72.79>}sofia/internal/sip:2001 at 192.168.73.120:5060;transport=udp,sofia/internal/sip:2001 at 192.168.72.87:5060;transport=udp,sofia/internal/sip:2001 at 192.168.73.125:5060;transport=udp,[lcr_carrier=Carrier / Location 1/INTERNAL PRI,lcr_rate=0.00200,origination_caller_id_number=2002]freetdm/grp1/a/4253491028)
1098: At internal 2001, answer call
1290: A long series of "reading on a session with no media" logs starts
1720: Freeswitch sends BYE to 2001 with reason "INCOMPATIBLE DESTINATION"
1752: Freeswitch sends BYE to 2002 with reason "INCOMPATIBLE DESTINATION"

Thanks Much,
Jerry



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:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Friday, July 01, 2011 9:40 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call Dropped When Answered If Bridged To Internal Extension And PSTN Number Via

Wait, can you explain that again? Step us through the call flow. Who connects to whom and in what order, etc.
-MC
On Thu, Jun 30, 2011 at 11:42 AM, Jerry Richards <jerry.richards at teotech.com<mailto:jerry.richards at teotech.com>> wrote:
Hello,

If I bridge a call to both an internal extension and a PSTN number (via FreeTDM), and then answer the call at the internal extension, the call is dropped.  I posted the log at

http://pastebin.freeswitch.org/16634

The log shows many errors of the type:

reading on a session with no media!

I have bypass-media 'true' (not sure if this has anything to do with it).  Any clue why this is happening?

Thanks,
Jerry

_______________________________________________
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http://www.cluecon.com 877-7-4ACLUE

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


_______________________________________________
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<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From msc at freeswitch.org  Thu Jul  7 02:51:03 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Jul 2011 15:51:03 -0700
Subject: [Freeswitch-users] Chinese sounds
In-Reply-To: <40057F672A4E43A7971543ECA4C3B97B@gmail.com>
References: <40057F672A4E43A7971543ECA4C3B97B@gmail.com>
Message-ID: <CAKzWOxV7UXPtQH9w-xpVPW_okfrpNd361gSEG8nNQO=cgUfKwQ@mail.gmail.com>

On Tue, Jul 5, 2011 at 11:34 PM, Seven Du <dujinfang at gmail.com> wrote:

>  Hi,
>
> We got someone in the Chinese Community would like to record a Chinese
> version of sounds, here's some questions for the community -
>
>
> 1) Where can I find the original English scripts ? It would be hard to
> listen all sound files
>
In the source directory: docs/phrase/phrase_en.xml

>
> 2) What's base256/*.wav usage?
>
Those are for ZRTP


>
> 3) We'd like to share to world, so, how to upload to files.freeswitch.orgwhen we are done?
>
Raymond and I will make sure that the files get up there. Email me off list.


>
>
4) I guess the raw recording should be 48K, 16bit, single channel, any other
> params should we set when recording?
>
That sounds good.

-MC
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From msc at freeswitch.org  Thu Jul  7 02:52:42 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Jul 2011 15:52:42 -0700
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
Message-ID: <CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>

What is the originate string you are using? Also, if you are ignoring early
media then how is the A leg supposed to hear ringing unless you specify the
ringback tone to use?
-MC

On Wed, Jul 6, 2011 at 9:52 AM, Mathieu Lautram
<lautram.mathieu at gmail.com>wrote:

> Hi all
>
> I'm facing an issue with originate. Here is the case:
>
> I do an originate to bridge A leg to B leg using bridge application.
> The A leg works perfectly but, when the B leg is called, I can't hear
> ringtone in the A leg.
> I set ignore_early_media to true.
>
> I would like to hear a ringtone without setting the ringback variable with
> the tone of my country.
>
> Is it possible to do that?
>
> Thank you in advance.
>
> --
> Mathieu LAUTRAM
> Application developer
>
> BJT Partners - FRANCE
> +33 1 79 75 99 60
> +33 6 61 59 07 25
>
> _______________________________________________
> 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 brian at freeswitch.org  Thu Jul  7 03:20:30 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 6 Jul 2011 18:20:30 -0500
Subject: [Freeswitch-users] Ok I gotta share this...
In-Reply-To: <CAMWiMy8E3Q1H1RFH8ih8CWHo6QNyYUjehpKLKqYwpobLuRkjFQ@mail.gmail.com>
References: <CA3A235A.18A71%krice@freeswitch.org>
	<95E60B128121421BA5D7213E7455E9E2@e1705>
	<CAMWiMy8E3Q1H1RFH8ih8CWHo6QNyYUjehpKLKqYwpobLuRkjFQ@mail.gmail.com>
Message-ID: <B25BBEDC-68CD-46EA-BC66-4430A6C61FE3@freeswitch.org>

They had to be destroyed.   EVIL CAN NOT PERSIST!

/b

On Jul 6, 2011, at 4:47 PM, curriegrad2004 wrote:

> Can't somebody re-write the firmware so... We don't have to destroy 'em all?



From jan.berger at video24.no  Thu Jul  7 03:28:56 2011
From: jan.berger at video24.no (Jan Berger)
Date: Thu, 7 Jul 2011 01:28:56 +0200
Subject: [Freeswitch-users] Ok I gotta share this...
In-Reply-To: <B25BBEDC-68CD-46EA-BC66-4430A6C61FE3@freeswitch.org>
References: <CA3A235A.18A71%krice@freeswitch.org>
	<95E60B128121421BA5D7213E7455E9E2@e1705>
	<CAMWiMy8E3Q1H1RFH8ih8CWHo6QNyYUjehpKLKqYwpobLuRkjFQ@mail.gmail.com>
	<B25BBEDC-68CD-46EA-BC66-4430A6C61FE3@freeswitch.org>
Message-ID: <27FD6CCDBFBD4F25BDCF81C46BAC0009@dell9400>

How did I know that you would love this?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 7. juli 2011 01:21
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Ok I gotta share this...

They had to be destroyed.   EVIL CAN NOT PERSIST!

/b

On Jul 6, 2011, at 4:47 PM, curriegrad2004 wrote:

> Can't somebody re-write the firmware so... We don't have to destroy 'em
all?


_______________________________________________
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
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http://www.freeswitch.org



From lautram.mathieu at gmail.com  Thu Jul  7 03:44:07 2011
From: lautram.mathieu at gmail.com (Mathieu Lautram)
Date: Thu, 7 Jul 2011 01:44:07 +0200
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
Message-ID: <CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>

I know that when I'm using ignore early media the A leg can't hear ringing.
But the fact is when I set ignore early media to false, I don't know if the
call is successful for the attempting or for the connecting. What I would
like to have is the state "successful" but only for the connecting (that's
why I set ignore early media to true).

The originate string I am using is:

bgapi originate
{ignore_early_media=true}sofia/external/12345 at XXXXXX'&bridge(
{ignore_early_media=true}sofia/external/2468 at XXXXXXX
'

Thank you for your answer


2011/7/7 Michael Collins <msc at freeswitch.org>

> What is the originate string you are using? Also, if you are ignoring early
> media then how is the A leg supposed to hear ringing unless you specify the
> ringback tone to use?
> -MC
>
> On Wed, Jul 6, 2011 at 9:52 AM, Mathieu Lautram <lautram.mathieu at gmail.com
> > wrote:
>
>> Hi all
>>
>> I'm facing an issue with originate. Here is the case:
>>
>> I do an originate to bridge A leg to B leg using bridge application.
>> The A leg works perfectly but, when the B leg is called, I can't hear
>> ringtone in the A leg.
>> I set ignore_early_media to true.
>>
>> I would like to hear a ringtone without setting the ringback variable with
>> the tone of my country.
>>
>> Is it possible to do that?
>>
>> Thank you in advance.
>>
>> --
>> Mathieu LAUTRAM
>> Application developer
>>
>> BJT Partners - FRANCE
>> +33 1 79 75 99 60
>> +33 6 61 59 07 25
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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
>
>


-- 
Mathieu LAUTRAM
Application developer

BJT Partners - FRANCE
+33 1 79 75 99 60
+33 6 61 59 07 25
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From anthony.minessale at gmail.com  Thu Jul  7 04:09:22 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Jul 2011 19:09:22 -0500
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
Message-ID: <CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>

try ignore_early_media=ring_ready or set ringback on a to ${us-ring}
and ignore_early_media=true

On Wed, Jul 6, 2011 at 6:44 PM, Mathieu Lautram
<lautram.mathieu at gmail.com> wrote:
> I know that when I'm using ignore early media the A leg can't hear ringing.
> But the fact is when I set ignore early media to false, I don't know if the
> call is successful for the attempting or for the connecting. What I would
> like to have is the state "successful" but only for the connecting (that's
> why I set ignore early media to true).
> The originate string I am using is:
> bgapi originate {ignore_early_media=true}sofia/external/12345 at XXXXXX
> '&bridge( {ignore_early_media=true}sofia/external/2468 at XXXXXXX'
> Thank you for your answer
>
> 2011/7/7 Michael Collins <msc at freeswitch.org>
>>
>> What is the originate string you are using? Also, if you are ignoring
>> early media then how is the A leg supposed to hear ringing unless you
>> specify the ringback tone to use?
>> -MC
>>
>> On Wed, Jul 6, 2011 at 9:52 AM, Mathieu Lautram
>> <lautram.mathieu at gmail.com> wrote:
>>>
>>> Hi all
>>> I'm facing an issue with originate. Here is the case:
>>> I do an originate to bridge A leg to B leg using bridge application.
>>> The A leg works perfectly but, when the B leg is called, I can't hear
>>> ringtone in the A leg.
>>> I set ignore_early_media to true.
>>> I would like to hear a ringtone without setting the ringback variable
>>> with the tone of my country.
>>> Is it possible to do that?
>>> Thank you in advance.
>>>
>>> --
>>> Mathieu LAUTRAM
>>> Application developer
>>>
>>> BJT Partners - FRANCE
>>> +33 1 79 75 99 60
>>> +33 6 61 59 07 25
>>>
>>> _______________________________________________
>>> 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
>>>
>>
>>
>> _______________________________________________
>> 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
>>
>
>
>
> --
> Mathieu LAUTRAM
> Application developer
>
> BJT Partners - FRANCE
> +33 1 79 75 99 60
> +33 6 61 59 07 25
>
> _______________________________________________
> 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
>
>



-- 
Anthony Minessale II

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

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

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


From steveu at coppice.org  Thu Jul  7 05:11:14 2011
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 07 Jul 2011 09:11:14 +0800
Subject: [Freeswitch-users] Playing multiple files simultaneously
In-Reply-To: <2DB72AF2-2394-419A-ADC5-0B76FE594662@bryansmart.com>
References: <2DB72AF2-2394-419A-ADC5-0B76FE594662@bryansmart.com>
Message-ID: <4E1507B2.3090803@coppice.org>

On 07/01/2011 01:13 PM, Bryan Smart wrote:
> Is it possible for Freeswitch to play more than one file to a channel at a time? What I've seen and tried from the dialplan and scripts either queues files to play, or will stop a currently playing file so that the newly requested file will play. This also seems to be the case in conferences. When I send multiple play commands to conferences, the files are queued.
>
> As for how this might be used, think of an IVR that plays queued prompts, yet continuously plays looping music or a Shoutcast stream in the background. I also want to be able to play short cue tones that start at the same time as a prompt (don't want to pre-mix them in to a single file, though).
>
> Is this currently possible through any means? Perhaps with the event socket?
>
Be really careful making plans like this. Multiple sounds will mix 
nicely with something like ALaw or uLaw. If you try it with a low bit 
rate codec, like G.729 or a cellphone codec, the result will sound 
awful. Modern telephony really is geared to one voice and one voice only 
at any moment.

Steve



From kees at mroffice.org  Thu Jul  7 10:01:35 2011
From: kees at mroffice.org (Kees Varekamp)
Date: Thu, 7 Jul 2011 18:01:35 +1200
Subject: [Freeswitch-users] Recording - one leg of the call goes faster than
	the other
Message-ID: <CAHbVE92=9Q5JsUZZk+CG0pWNwO8fC246XdgQrQFdBrOj7XRV9A@mail.gmail.com>

Hi all,

We are having a weird issue with call recordings: one leg of the call goes
around 1 second per minute faster than the other. At the end of a 10 minute
recording this means that answers come around 10 seconds ahead of the
corresponding questions. This is on FreeSWITCH Version 1.0.head (git-38f06a3
2011-05-19 15-39-43 -0500)

This happens pretty much on all the recordings we do.

Found this similar sounding problem on the bug tracker:
http://jira.freeswitch.org/browse/FS-2532.
Has anyone experienced this same problem? Is there a solution?

Thanks,

Kees
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From boris at tagnet.ru  Thu Jul  7 10:07:21 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 07 Jul 2011 12:07:21 +0600
Subject: [Freeswitch-users] SIP 100 response and media
Message-ID: <4E154D19.1080100@tagnet.ru>

Hello!

     This thread again. There are two logs. First between Cisco<->Cisco 
call and second between Cisco<->FreeSwitch. Cisco<->Cisco works fine. As 
I can see the difference is that
Cisco has a=rtpmap:8 PCMA/8000 and FreeSwitch not. May be this is a 
problem? Is this field mandatory?


Jul  6 18:20:39.341: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:28032 at 192.168.35.15:5060 SIP/2.0
Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F
Remote-Party-ID: "test" 
<sip:30030 at 10.240.14.15>;party=calling;screen=no;privacy=off
From: "test" <sip:30030 at 10.240.14.15>;tag=3F5B1E64-2263
To: <sip:28032 at 192.168.35.15>
Date: Wed, 06 Jul 2011 18:20:39 GMT
Call-ID: 7C686709-A73311E0-95798064-7F05C5CA at 10.240.14.15
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2076058060-2805141984-2507440228-2131084746
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1309976439
Contact: <sip:30030 at 10.240.14.15:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 187

v=0
o=CiscoSystemsSIP-GW-UserAgent 0 703 IN IP4 10.240.14.15
s=SIP Call
c=IN IP4 10.240.14.15
t=0 0
m=audio 18720 RTP/AVP 8
c=IN IP4 10.240.14.15
a=rtpmap:8 PCMA/8000
a=ptime:20

Jul  6 18:20:39.361: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F
From: "test" <sip:30030 at 10.240.14.15>;tag=3F5B1E64-2263
To: <sip:28032 at 192.168.35.15>
Date: Wed, 06 Jul 2011 18:20:39 GMT
Call-ID: 7C686709-A73311E0-95798064-7F05C5CA at 10.240.14.15
Timestamp: 1309976439
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


Jul  6 18:20:39.369: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F
From: "test" <sip:30030 at 10.240.14.15>;tag=3F5B1E64-2263
To: <sip:28032 at 192.168.35.15>;tag=CBEFC0-16E4
Date: Wed, 06 Jul 2011 18:20:39 GMT
Call-ID: 7C686709-A73311E0-95798064-7F05C5CA at 10.240.14.15
Timestamp: 1309976439
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:28032 at 192.168.35.15:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 194


When call is between Cisco<->Freeswitch messages are
Jul  6 18:21:11.966: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:73435230030 at 192.168.3.6:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp
Max-Forwards: 67
From: "anonymous" <sip:0000000000 at 192.168.1.1>;tag=QFKH52Z9NaN2c
To: <sip:73435230030 at 192.168.3.6:5060>
Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac
CSeq: 14652171 INVITE
Contact: <sip:mod_sofia at 192.168.1.1:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1c95ad9 2011-01-20 
22-43-50 -0300
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE
Supported: 100rel, 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: 329
X-FS-Support: update_display
Remote-Party-ID: "anonymous" 
<sip:0000000000 at 192.168.1.1>;party=calling;screen=yes;privacy=off


v=0
o=FreeSWITCH 1309956111 1309956112 IN IP4 192.168.1.1
s=FreeSWITCH
c=IN IP4 192.168.1.1
t=0 0

m=audio 20360 RTP/AVP 8 0 18 101 13
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
m=video 19030 RTP/AVP 31 34 98
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:98 H264/90000


Jul  6 18:21:11.986: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp
From: "anonymous" <sip:0000000000 at 192.168.1.1>;tag=QFKH52Z9NaN2c
To: <sip:73435230030 at 192.168.3.6:5060>
Date: Wed, 06 Jul 2011 18:21:11 GMT
Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac
CSeq: 14652171 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0





-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005



From kees at mroffice.org  Thu Jul  7 10:53:41 2011
From: kees at mroffice.org (Kees Varekamp)
Date: Thu, 7 Jul 2011 18:53:41 +1200
Subject: [Freeswitch-users] Flex RTMP Client Added To Git Tree!
In-Reply-To: <CAKzWOxXgDVPWdQH90y9OQz0kJP8UMT_MeKfhWWki78cui-WHrA@mail.gmail.com>
References: <CAKzWOxXgDVPWdQH90y9OQz0kJP8UMT_MeKfhWWki78cui-WHrA@mail.gmail.com>
Message-ID: <CAHbVE916P6DcbX9LD6UOikbjMQF+3rp45h1rs47uDMq9dKk+2w@mail.gmail.com>

Awesome! tnx guys!

Kees

On Thu, Jul 7, 2011 at 05:27, Michael Collins <msc at freeswitch.org> wrote:

> I thought many of you would be interested in this story about the RTMP Flex
> client:
>
> http://www.freeswitch.org/node/332
>
> The sample Flex client has been added to tree, so feel free to start
> experimenting!
>
> -Michael
>
> _______________________________________________
> 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 freeswitch at earthspike.net  Thu Jul  7 11:37:54 2011
From: freeswitch at earthspike.net (John)
Date: Thu, 07 Jul 2011 08:37:54 +0100
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
 Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <CAP1_MOCq5qRjsgd7XGag9ukjHiG2Se-pXvrghQ7hbB5ciwJK+A@mail.gmail.com>
References: <4DED4854.6010203@earthspike.net>	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>	<4DF10C83.6020803@earthspike.net>	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>	<4DF1F07A.5000200@earthspike.net>	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>	<4DFB9831.4040709@earthspike.net>	<BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>
	<CAP1_MOCq5qRjsgd7XGag9ukjHiG2Se-pXvrghQ7hbB5ciwJK+A@mail.gmail.com>
Message-ID: <4E156252.2030308@earthspike.net>

On 06/07/11 13:53, shouldbe q931 wrote:
> On Fri, Jun 17, 2011 at 7:43 PM, shouldbe q931<shouldbeq931 at gmail.com>  wrote:
>> On Fri, Jun 17, 2011 at 7:08 PM, John<freeswitch at earthspike.net>  wrote:
>>> Mike, Shouldbe,
>>>
>>> We now have CLIP on the line for sure, and I ordered COLP at the same time
>>> and am assured that that has been turned on as well, but I still cannot get
>>> outgoing CLI to work properly.  Mike, you mention setting the TON to
>>> 'national'; where is that setting?
>>>
>>> At the moment, my dialplan looks like http://pastebin.freeswitch.org/16548
>>> with .  An outgoing call gives the log 7 at
>>> http://pastebin.freeswitch.org/16549 (numbers have been changed to protect
>>> the innocent!) but the number that 01234 567890 sees on their Caller ID is
>>> not 876543 but the main number of the line (876540, say).  I have tried
>>> setting the outgoing_caller_id_number to 6, 10 and 11 digits, restarting FS
>>> after each change, but with no success.
>>>
>>> The service provider has only one clue to offer: "With regards to the
>>> configuration at the exchange, the line is set to 6 digits to switch." which
>>> makes eminent sense, and suggests that I should be presenting 6 digits.
>>> Incoming calls have a 6 digit called number and 10 digit calling number.
>>>
>>> Thanks for the help so far.  Any other ideas?
>>>
>>> John
>>>
>>>
>> I don't have any experience with FS connecting over ISDN so can't help
>> you much further on the FS configuration:-(
>>
>> However on the lines, I would be very tempted to see if you can borrow
>> a BRI tester, or what I've frequently used in the past is an
>> Eicon/Dialogic card as the test functions (under windows) are nearly
>> as good, that might help you see if it _is_ a BT problem, or a FS
>> configuration issue. It would need a 2k/XP machine, but something like
>> http://cgi.ebay.co.uk/EICON-DIVA-2-01-PCI-GRAPHICS-CARD-/270761571564
>> is what I have used in the past. I think I might have one of the
>> PCMCIA ones somewhere, but I'd need to test it still works
>>
>> I'm fairly sure that the "6 digits to switch" is what they are sending
>> to you, not what they are expecting from you. I can't access out
>> remaining BRI (Avaya) PBX from here, but on our PRI (again Avaya)
>> switches, we are sending 10 digits.
>>
>> Cheers
>>
> Apologies for the very long delay...
>
> On our PRI systems we are sending 10 digits (2071231234), and on the
> BRI system we are sending 11 digits (02071231234).
>
> Cheers
Thanks.  I have tried all of these digit options, along with setting the 
TON and NPI, so I am now working through every possible option and tweak 
with the supplier (including including codes to remove/reapply CPS).  I 
am trying to avoid booking this as a fault, until I am sure I won't be 
charged for a 'no fault found' but it is beginning to look unavoidable.  
I am certainly building enough information for a good wiki page on this 
when I finally get it cracked!  Incidentally, the Sangoma 'wanpipemon' 
provides a useful PCAP mode which delivers the entire D-channel as a 
wireshark PCAP file, so I am able to debug the Q.931 signalling down to 
bit level.

John


From boris at tagnet.ru  Thu Jul  7 12:46:16 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 07 Jul 2011 14:46:16 +0600
Subject: [Freeswitch-users] question about continue_on_fail
Message-ID: <4E157258.1050702@tagnet.ru>

Hello!

Here is an extension:
<extension name="ext_domestic_e164">
<condition field="destination_number" expression="^(7[0-9]{10})$">
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=true"/>
<!-- Least Cost Routing scheme -->
<action application="lcr" data="$1 lcr_tagnet"/>
<action application="set" data="call_timeout=60"/>
<action application="set" data="bridge_answer_timeout=60"/>
<action application="set" data="disable-transcoding=true"/>
<action application="bridge" data="${lcr_auto_route}"/>
</condition>
</extension>

lcr_auto_route returns 2 bridges to a same destination.
[lcr_carrier=naukanet.ru,lcr_rate=1.45140,v_ats_dstport=50002]sofia/epbx/89979049898ZZZ at X.X.X.42:5061|[lcr_carrier=mtt.ru,lcr_rate=1.60480,v_ats_dstport=50006]sofia/epbx/79049898ZZZ at Y.Y.Y.132:5060

Here is a problem: when the reject from the remote is [CS_CONSUME_MEDIA] 
[USER_BUSY] the call is routed to the next bridge. When the reject is 
[CS_EXCHANGE_MEDIA] [USER_BUSY] the bridge hangs up. So, what is the 
right way to stop processing when USER_BUSY arrived? I tried 
continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION without 
success.



-- 
Regards,
Boris




From dujinfang at gmail.com  Thu Jul  7 14:37:32 2011
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 7 Jul 2011 18:37:32 +0800
Subject: [Freeswitch-users] Chinese sounds
In-Reply-To: <4E1482EF.9070107@coppice.org>
References: <40057F672A4E43A7971543ECA4C3B97B@gmail.com>
	<4E1482EF.9070107@coppice.org>
Message-ID: <A5720BAB93BA484DA06D7D0397953B52@gmail.com>



On Wednesday, July 6, 2011 at 11:44 PM, Steve Underwood wrote:

> Hi,
> 
> Mandarin or a wider selection of Chinese dialects?
> 
Mandarin only for now.

> On 07/06/2011 02:34 PM, Seven Du wrote:
> > Hi,
> > 
> > We got someone in the Chinese Community would like to record a Chinese 
> > version of sounds, here's some questions for the community -
> > 
> > 
> > 1) Where can I find the original English scripts ? It would be hard to 
> > listen all sound files
> > 
> > 2) What's base256/*.wav usage?
> > 
> > 3) We'd like to share to world, so, how to upload to 
> > files.freeswitch.org (http://files.freeswitch.org) when we are done?
> > 
> > 4) I guess the raw recording should be 48K, 16bit, single channel, any 
> > other params should we set when recording?
> > 
> > 
> > Thanks.
> > 
> > -- 
> > Seven Du
> > About: http://about.me/dujinfang
> > Blog: http://www.dujinfang.com
> > Proj: http://www.freeswitch.org.cn
> > Sent with Sparrow <http://www.sparrowmailapp.com>
> > 
> > 
> > _______________________________________________
> > 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 (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
> 
> 
> _______________________________________________
> 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 (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 shouldbeq931 at gmail.com  Thu Jul  7 15:21:48 2011
From: shouldbeq931 at gmail.com (shouldbe q931)
Date: Thu, 7 Jul 2011 12:21:48 +0100
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
 Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <4E156252.2030308@earthspike.net>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
	<4DF10C83.6020803@earthspike.net>
	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>
	<4DF1F07A.5000200@earthspike.net>
	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>
	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>
	<4DFB9831.4040709@earthspike.net>
	<BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>
	<CAP1_MOCq5qRjsgd7XGag9ukjHiG2Se-pXvrghQ7hbB5ciwJK+A@mail.gmail.com>
	<4E156252.2030308@earthspike.net>
Message-ID: <CAP1_MOBAFpYqJNKhgWzv3aYJeKbj+RJDQJCaQByVcrMVp_s8RA@mail.gmail.com>

On Thu, Jul 7, 2011 at 8:37 AM, John <freeswitch at earthspike.net> wrote:
> On 06/07/11 13:53, shouldbe q931 wrote:
>> On Fri, Jun 17, 2011 at 7:43 PM, shouldbe q931<shouldbeq931 at gmail.com> ?wrote:
>>> On Fri, Jun 17, 2011 at 7:08 PM, John<freeswitch at earthspike.net> ?wrote:
>>>> Mike, Shouldbe,
>>>>
>>>> We now have CLIP on the line for sure, and I ordered COLP at the same time
>>>> and am assured that that has been turned on as well, but I still cannot get
>>>> outgoing CLI to work properly. ?Mike, you mention setting the TON to
>>>> 'national'; where is that setting?
>>>>
>>>> At the moment, my dialplan looks like http://pastebin.freeswitch.org/16548
>>>> with . ?An outgoing call gives the log 7 at
>>>> http://pastebin.freeswitch.org/16549 (numbers have been changed to protect
>>>> the innocent!) but the number that 01234 567890 sees on their Caller ID is
>>>> not 876543 but the main number of the line (876540, say). ?I have tried
>>>> setting the outgoing_caller_id_number to 6, 10 and 11 digits, restarting FS
>>>> after each change, but with no success.
>>>>
>>>> The service provider has only one clue to offer: "With regards to the
>>>> configuration at the exchange, the line is set to 6 digits to switch." which
>>>> makes eminent sense, and suggests that I should be presenting 6 digits.
>>>> Incoming calls have a 6 digit called number and 10 digit calling number.
>>>>
>>>> Thanks for the help so far. ?Any other ideas?
>>>>
>>>> John
>>>>
>>>>
>>> I don't have any experience with FS connecting over ISDN so can't help
>>> you much further on the FS configuration:-(
>>>
>>> However on the lines, I would be very tempted to see if you can borrow
>>> a BRI tester, or what I've frequently used in the past is an
>>> Eicon/Dialogic card as the test functions (under windows) are nearly
>>> as good, that might help you see if it _is_ a BT problem, or a FS
>>> configuration issue. It would need a 2k/XP machine, but something like
>>> http://cgi.ebay.co.uk/EICON-DIVA-2-01-PCI-GRAPHICS-CARD-/270761571564
>>> is what I have used in the past. I think I might have one of the
>>> PCMCIA ones somewhere, but I'd need to test it still works
>>>
>>> I'm fairly sure that the "6 digits to switch" is what they are sending
>>> to you, not what they are expecting from you. I can't access out
>>> remaining BRI (Avaya) PBX from here, but on our PRI (again Avaya)
>>> switches, we are sending 10 digits.
>>>
>>> Cheers
>>>
>> Apologies for the very long delay...
>>
>> On our PRI systems we are sending 10 digits (2071231234), and on the
>> BRI system we are sending 11 digits (02071231234).
>>
>> Cheers
> Thanks. ?I have tried all of these digit options, along with setting the
> TON and NPI, so I am now working through every possible option and tweak
> with the supplier (including including codes to remove/reapply CPS). ?I
> am trying to avoid booking this as a fault, until I am sure I won't be
> charged for a 'no fault found' but it is beginning to look unavoidable.
> I am certainly building enough information for a good wiki page on this
> when I finally get it cracked! ?Incidentally, the Sangoma 'wanpipemon'
> provides a useful PCAP mode which delivers the entire D-channel as a
> wireshark PCAP file, so I am able to debug the Q.931 signalling down to
> bit level.
>
> John
>

If you can see in the capture that you are sending the digits, and the
CLI is not being set, then open a fault with BT.

Alternatively, find a friendly "legacy" PBX maintainer that has a
trend/harrier, or take your FS box to somewhere that has some BRI
lines that are known to work with CLIP and COLP and try your FS box on
them...


From lautram.mathieu at gmail.com  Thu Jul  7 15:27:23 2011
From: lautram.mathieu at gmail.com (Mathieu Lautram)
Date: Thu, 7 Jul 2011 13:27:23 +0200
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
Message-ID: <CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>

I've already tried those parameters and it doesn't work.
I can use ignore_early_medi=true and ringback=${us-ring} but what if my B
leg is in France? Or in England?
The schema is like that:

France:

A leg have to hear the ringtone of France

US:

A leg have to hear the ringtone of the US

England:

A leg have to hear the ringtone of England

Other countries:

same thing

Have I to check the destination to apply the correct ringtone or is there an
other solution to let A leg hear the ringtone of the correct country?
Is Freeswitch allow me to know if the call to the B leg is successful for a
"connecting" state (and not "attempting" state) with ignore_early_media set
to false?

Best regards

2011/7/7 Anthony Minessale <anthony.minessale at gmail.com>

> try ignore_early_media=ring_ready or set ringback on a to ${us-ring}
> and ignore_early_media=true
>
> On Wed, Jul 6, 2011 at 6:44 PM, Mathieu Lautram
> <lautram.mathieu at gmail.com> wrote:
> > I know that when I'm using ignore early media the A leg can't hear
> ringing.
> > But the fact is when I set ignore early media to false, I don't know if
> the
> > call is successful for the attempting or for the connecting. What I would
> > like to have is the state "successful" but only for the connecting
> (that's
> > why I set ignore early media to true).
> > The originate string I am using is:
> > bgapi originate {ignore_early_media=true}sofia/external/12345 at XXXXXX
> > '&bridge( {ignore_early_media=true}sofia/external/2468 at XXXXXXX'
> > Thank you for your answer
> >
> > 2011/7/7 Michael Collins <msc at freeswitch.org>
> >>
> >> What is the originate string you are using? Also, if you are ignoring
> >> early media then how is the A leg supposed to hear ringing unless you
> >> specify the ringback tone to use?
> >> -MC
> >>
> >> On Wed, Jul 6, 2011 at 9:52 AM, Mathieu Lautram
> >> <lautram.mathieu at gmail.com> wrote:
> >>>
> >>> Hi all
> >>> I'm facing an issue with originate. Here is the case:
> >>> I do an originate to bridge A leg to B leg using bridge application.
> >>> The A leg works perfectly but, when the B leg is called, I can't hear
> >>> ringtone in the A leg.
> >>> I set ignore_early_media to true.
> >>> I would like to hear a ringtone without setting the ringback variable
> >>> with the tone of my country.
> >>> Is it possible to do that?
> >>> Thank you in advance.
> >>>
> >>> --
> >>> Mathieu LAUTRAM
> >>> Application developer
> >>>
> >>> BJT Partners - FRANCE
> >>> +33 1 79 75 99 60
> >>> +33 6 61 59 07 25
> >>>
> >>> _______________________________________________
> >>> 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
> >>>
> >>
> >>
> >> _______________________________________________
> >> 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
> >>
> >
> >
> >
> > --
> > Mathieu LAUTRAM
> > Application developer
> >
> > BJT Partners - FRANCE
> > +33 1 79 75 99 60
> > +33 6 61 59 07 25
> >
> > _______________________________________________
> > 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
> >
> >
>
>
>
> --
> 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
>



-- 
Mathieu LAUTRAM
Application developer

BJT Partners - FRANCE
+33 1 79 75 99 60
+33 6 61 59 07 25
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From davidwaf at gmail.com  Thu Jul  7 15:52:08 2011
From: davidwaf at gmail.com (David Wafula)
Date: Thu, 7 Jul 2011 13:52:08 +0200
Subject: [Freeswitch-users] Flex RTMP Client Added To Git Tree!
In-Reply-To: <CAKzWOxXgDVPWdQH90y9OQz0kJP8UMT_MeKfhWWki78cui-WHrA@mail.gmail.com>
References: <CAKzWOxXgDVPWdQH90y9OQz0kJP8UMT_MeKfhWWki78cui-WHrA@mail.gmail.com>
Message-ID: <CAK9Ws2cjOBAqt_vUzawk9+cqEyqxRag7MD=nbj6Xva+97TxaSw@mail.gmail.com>

On Wed, Jul 6, 2011 at 7:27 PM, Michael Collins <msc at freeswitch.org> wrote:

> I thought many of you would be interested in this story about the RTMP Flex
> client:
>
> http://www.freeswitch.org/node/332
>
> The sample Flex client has been added to tree, so feel free to start
> experimenting!
>
>
Thank you!!!!!!!!!


-- 
David Wafula
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From sid.kshatriya at gmail.com  Thu Jul  7 16:22:32 2011
From: sid.kshatriya at gmail.com (Sidharth Kshatriya)
Date: Thu, 7 Jul 2011 17:52:32 +0530
Subject: [Freeswitch-users] Flex RTMP Client Added To Git Tree!
In-Reply-To: <CAK9Ws2cjOBAqt_vUzawk9+cqEyqxRag7MD=nbj6Xva+97TxaSw@mail.gmail.com>
References: <CAKzWOxXgDVPWdQH90y9OQz0kJP8UMT_MeKfhWWki78cui-WHrA@mail.gmail.com>
	<CAK9Ws2cjOBAqt_vUzawk9+cqEyqxRag7MD=nbj6Xva+97TxaSw@mail.gmail.com>
Message-ID: <CAHZjNtaN3CHdjvLA1b9QpZPSqGQCA-uayYCF7CSjsoyNUvCEbQ@mail.gmail.com>

Very clean and easy to understand code. Thanks!

Please note that its almost trivial to covert this to a flash application
(the flex application would be easily become a ~1MB download because it your
browser will download the whole flex framework). Flash is going to be very
very light. Maybe even ~50KB!

On Thu, Jul 7, 2011 at 5:22 PM, David Wafula <davidwaf at gmail.com> wrote:

>
>
> On Wed, Jul 6, 2011 at 7:27 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> I thought many of you would be interested in this story about the RTMP
>> Flex client:
>>
>> http://www.freeswitch.org/node/332
>>
>> The sample Flex client has been added to tree, so feel free to start
>> experimenting!
>>
>>
> Thank you!!!!!!!!!
>
>
> --
> David Wafula
>
> _______________________________________________
> 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
>
>


-- 
Sidharth Kshatriya
www.sidk.info
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From dujinfang at gmail.com  Thu Jul  7 16:55:06 2011
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 7 Jul 2011 20:55:06 +0800
Subject: [Freeswitch-users] Custom GUI for FS
In-Reply-To: <4E148599.1080700@seletech.com>
References: <4E147795.3090700@seletech.com>
	<CAA1Qw2ua8_Rsx-f6HZpg_g+k0w=RM-JuT0mrAxJfgce7xXCbDQ@mail.gmail.com>
	<CAFiqYumz+HQwkrDEy187TYjkshvtArveb3g+anLq6j04PiSU9w@mail.gmail.com>
	<4E148599.1080700@seletech.com>
Message-ID: <ADE97DCBF30C48D5A43D8DA3EDF0AC5F@gmail.com>

I use ODBC and select * directly from aatabase. Just another option.

-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn
Sent with Sparrow (http://www.sparrowmailapp.com)

On Wednesday, July 6, 2011 at 11:56 PM, Alessandro wrote:

>  Yes,
> 
>  I'll use  the command 'show calls as xml' it's better.
> 
>  Thanks
> 
>  Best Regards
> 
>  Alessandro Luppi 
> 
> 
>  Il 06/07/2011 17:17, Steven Ayre ha scritto: 
> > 'show calls as xml' and 'show channels as xml' are much better for this sort of thing, as the XML is easier to parse and will be more reliable (if the status output formatting changes in a later update then programs might break unless they're using the xml output).
> > 
> >  -Steve
> > 
> > 
> > 
> > On 6 July 2011 16:05, Nagalenoj H. <nagalenoj at gmail.com (mailto:nagalenoj at gmail.com)> wrote:
> > >  Hi,
> > >  Hope, show(show calls, show channels) command will help you.
> > > 
> > > http://wiki.freeswitch.org/wiki/Mod_commands#show 
> > > 
> > > 
> > > 
> > >  On Wed, Jul 6, 2011 at 8:26 PM, Alessandro <a.luppi at seletech.com (mailto:a.luppi at seletech.com)> wrote:
> > > >  Hi,
> > > > 
> > > >  I'm going to develop a custom GUI web-based for freeswitch that allow
> > > >  only to add/remove extension and show the call active and the status of
> > > >  extensions configured.
> > > >  Where can I find in FS the information about actives call and extensions
> > > >  status?
> > > > 
> > > >  Thanks
> > > > 
> > > >  Alessandro Luppi
> > > > 
> > > >  --
> > > >  Ing. Alessandro Luppi
> > > >  Software development
> > > >  Seletech srl
> > > >  Via Collodi 8, 20052 Monza (MI) - Italy
> > > >  Tel: +39.039.5962000 - Fax: +39.039.9716905
> > > >  email: a.luppi at seletech.com (mailto:a.luppi at seletech.com) - Web: www.seletech.com (http://www.seletech.com)  or www.seletech.eu (http://www.seletech.eu)
> > > > 
> > > > 
> > > >  _______________________________________________
> > > >  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 (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
> > > 
> > > 
> > > 
> > >  -- 
> > >  Regards,
> > > Nagalenoj H.
> > > 
> > >  _______________________________________________
> > >  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 (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
> > > 
> > 
> >  _______________________________________________ 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 (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 
> 
> -- Ing. Alessandro Luppi Software development Seletech srl Via Collodi 8, 20052 Monza (MI) - Italy Tel: +39.039.5962000 - Fax: +39.039.9716905 email: a.luppi at seletech.com (mailto:a.luppi at seletech.com) - Web: www.seletech.com (http://www.seletech.com) or www.seletech.eu (http://www.seletech.eu) 
> _______________________________________________
> 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 (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 kris at kriskinc.com  Thu Jul  7 17:06:07 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 7 Jul 2011 09:06:07 -0400
Subject: [Freeswitch-users] SIP 100 response and media
In-Reply-To: <4E154D19.1080100@tagnet.ru>
References: <4E154D19.1080100@tagnet.ru>
Message-ID: <CAKDfjgdK+BrunGC0UY=greJw_DpUftermo_Dzp9k9kyKhesbew@mail.gmail.com>

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

Set this to true in your dialplan before calling bridge when placing a
call to the Cisco.

On Thu, Jul 7, 2011 at 2:07 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ? ? This thread again. There are two logs. First between Cisco<->Cisco
> call and second between Cisco<->FreeSwitch. Cisco<->Cisco works fine. As
> I can see the difference is that
> Cisco has a=rtpmap:8 PCMA/8000 and FreeSwitch not. May be this is a
> problem? Is this field mandatory?
>
>
> Jul ?6 18:20:39.341: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Received:
> INVITE sip:28032 at 192.168.35.15:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F
> Remote-Party-ID: "test"
> <sip:30030 at 10.240.14.15>;party=calling;screen=no;privacy=off
> From: "test" <sip:30030 at 10.240.14.15>;tag=3F5B1E64-2263
> To: <sip:28032 at 192.168.35.15>
> Date: Wed, 06 Jul 2011 18:20:39 GMT
> Call-ID: 7C686709-A73311E0-95798064-7F05C5CA at 10.240.14.15
> Supported: 100rel,timer,resource-priority,replaces,sdp-anat
> Min-SE: ?1800
> Cisco-Guid: 2076058060-2805141984-2507440228-2131084746
> User-Agent: Cisco-SIPGateway/IOS-12.x
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY, INFO, REGISTER
> CSeq: 101 INVITE
> Max-Forwards: 70
> Timestamp: 1309976439
> Contact: <sip:30030 at 10.240.14.15:5060>
> Expires: 180
> Allow-Events: telephone-event
> Content-Type: application/sdp
> Content-Disposition: session;handling=required
> Content-Length: 187
>
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 0 703 IN IP4 10.240.14.15
> s=SIP Call
> c=IN IP4 10.240.14.15
> t=0 0
> m=audio 18720 RTP/AVP 8
> c=IN IP4 10.240.14.15
> a=rtpmap:8 PCMA/8000
> a=ptime:20
>
> Jul ?6 18:20:39.361: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Sent:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F
> From: "test" <sip:30030 at 10.240.14.15>;tag=3F5B1E64-2263
> To: <sip:28032 at 192.168.35.15>
> Date: Wed, 06 Jul 2011 18:20:39 GMT
> Call-ID: 7C686709-A73311E0-95798064-7F05C5CA at 10.240.14.15
> Timestamp: 1309976439
> CSeq: 101 INVITE
> Allow-Events: telephone-event
> Server: Cisco-SIPGateway/IOS-12.x
> Content-Length: 0
>
>
> Jul ?6 18:20:39.369: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Sent:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F
> From: "test" <sip:30030 at 10.240.14.15>;tag=3F5B1E64-2263
> To: <sip:28032 at 192.168.35.15>;tag=CBEFC0-16E4
> Date: Wed, 06 Jul 2011 18:20:39 GMT
> Call-ID: 7C686709-A73311E0-95798064-7F05C5CA at 10.240.14.15
> Timestamp: 1309976439
> CSeq: 101 INVITE
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY, INFO, REGISTER
> Allow-Events: telephone-event
> Contact: <sip:28032 at 192.168.35.15:5060>
> Supported: replaces
> Supported: sdp-anat
> Server: Cisco-SIPGateway/IOS-12.x
> Content-Type: application/sdp
> Content-Disposition: session;handling=required
> Content-Length: 194
>
>
> When call is between Cisco<->Freeswitch messages are
> Jul ?6 18:21:11.966: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Received:
> INVITE sip:73435230030 at 192.168.3.6:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp
> Max-Forwards: 67
> From: "anonymous" <sip:0000000000 at 192.168.1.1>;tag=QFKH52Z9NaN2c
> To: <sip:73435230030 at 192.168.3.6:5060>
> Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac
> CSeq: 14652171 INVITE
> Contact: <sip:mod_sofia at 192.168.1.1:5060>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1c95ad9 2011-01-20
> 22-43-50 -0300
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: 100rel, 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: 329
> X-FS-Support: update_display
> Remote-Party-ID: "anonymous"
> <sip:0000000000 at 192.168.1.1>;party=calling;screen=yes;privacy=off
>
>
> v=0
> o=FreeSWITCH 1309956111 1309956112 IN IP4 192.168.1.1
> s=FreeSWITCH
> c=IN IP4 192.168.1.1
> t=0 0
>
> m=audio 20360 RTP/AVP 8 0 18 101 13
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> m=video 19030 RTP/AVP 31 34 98
> a=rtpmap:31 H261/90000
> a=rtpmap:34 H263/90000
> a=rtpmap:98 H264/90000
>
>
> Jul ?6 18:21:11.986: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Sent:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp
> From: "anonymous" <sip:0000000000 at 192.168.1.1>;tag=QFKH52Z9NaN2c
> To: <sip:73435230030 at 192.168.3.6:5060>
> Date: Wed, 06 Jul 2011 18:21:11 GMT
> Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac
> CSeq: 14652171 INVITE
> Allow-Events: telephone-event
> Server: Cisco-SIPGateway/IOS-12.x
> Content-Length: 0
>
>
>
>
>
> --
> ? ?????????,
> ? ????? ?????????
> ? ??? "??????"
> ? ???. +7 (3435) 230001
> ? ???? +7 (3435) 230005
>
>
> _______________________________________________
> 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
>



-- 
Kristian Kielhofner


From steveayre at gmail.com  Thu Jul  7 17:33:07 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 7 Jul 2011 14:33:07 +0100
Subject: [Freeswitch-users] New BSD timer module
Message-ID: <CAFiqYumUFpv0SjGJmdoZf2rAy2LGiFz7HndUuiRTRS-bGbpErQ@mail.gmail.com>

Hi everyone,

I've submitted a Jira with a proposed new mod_kqueue timer module. This is a
BSD equivalent of mod_timerfd.
http://jira.freeswitch.org/browse/FS-3398

I'd appreciate it if anyone that uses FreeSWITCH on BSD that feels like it
would give it a quick test to see how well it works.

To install it:
- Create a new directory src/mod/timers/mod_kqueue
- Place the mod_kqueue.c file from the jira in that directory
- Add timers/mod_kqueue to modules.conf
- Build and install
- Load it from modules.conf.xml or fs_cli.

Regards,
Steven Ayre
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From fs-list at communicatefreely.net  Thu Jul  7 17:50:56 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Thu, 07 Jul 2011 09:50:56 -0400
Subject: [Freeswitch-users] New BSD timer module
In-Reply-To: <CAFiqYumUFpv0SjGJmdoZf2rAy2LGiFz7HndUuiRTRS-bGbpErQ@mail.gmail.com>
References: <CAFiqYumUFpv0SjGJmdoZf2rAy2LGiFz7HndUuiRTRS-bGbpErQ@mail.gmail.com>
Message-ID: <4E15B9C0.7070608@communicatefreely.net>

Hi Steven,

Yes, we are running our FreeSwitch deployment on FreeBSD 8.1 and 8.2  I 
would be glad to try it out, although it might be a few days before I 
get to it.

What is it's purpose, and how would I know if it's working well or not?

Thanks!

-Tim

Steven Ayre wrote:
> Hi everyone,
>
> I've submitted a Jira with a proposed new mod_kqueue timer module. 
> This is a BSD equivalent of mod_timerfd.
> http://jira.freeswitch.org/browse/FS-3398
>
> I'd appreciate it if anyone that uses FreeSWITCH on BSD that feels 
> like it would give it a quick test to see how well it works.
>
> To install it:
> - Create a new directory src/mod/timers/mod_kqueue
> - Place the mod_kqueue.c file from the jira in that directory
> - Add timers/mod_kqueue to modules.conf
> - Build and install
> - Load it from modules.conf.xml or fs_cli.
>
> Regards,
> Steven Ayre
> ------------------------------------------------------------------------
>
> _______________________________________________
> 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 steveayre at gmail.com  Thu Jul  7 18:24:20 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 7 Jul 2011 15:24:20 +0100
Subject: [Freeswitch-users] New BSD timer module
In-Reply-To: <4E15B9C0.7070608@communicatefreely.net>
References: <CAFiqYumUFpv0SjGJmdoZf2rAy2LGiFz7HndUuiRTRS-bGbpErQ@mail.gmail.com>
	<4E15B9C0.7070608@communicatefreely.net>
Message-ID: <CAFiqYu=W2AoPZ=8hwuB1ub3ix_a1N9a3Mf5PE9DbG3_Z8v9M3w@mail.gmail.com>

It's an alternative method of providing timing. When FS wants to wait for,
say, 10ms it uses a timer module to wait that period of time. The software
one uses delays and attempts to calibrate itself for clock drift. mod_kqueue
uses an interval timer that BSD provides which is hopefully also accurate, a
similar idea to mod_timerfd. The testing is more to see whether that's
actually the case or whether the software one turns out to be more reliable.

You can do a quick test using the commands "timer_test 10 200 soft" and
"timer_test 10 200 kqueue". They'll each try sleeping for 10ms 200 times but
using first the software timer and then the kqueue one. You'll see that
nothing every sleeps exactly 10ms. It'll tell you how long it took to do,
ideally it'll be pretty close to 10ms*200 = 2s, but if it's not working well
it might take longer (or less time). It'll also tell you the average time it
slept for, which should be approximately 10ms. The closer the better.
There are also several other intervals that would need to be tried, e.g.
10,20,40,60,120.

The second test is updating your configuration to see whether your audio
quality is better, worse or about the same. For mod_sofia calls that's done
by setting <param name="timer-name" value="kqueue"/> on the SIP profiles.

It's highly experimental, so I'd suggest if you are going to try it out you
don't use it for production traffic just yet.

-Steve




On 7 July 2011 14:50, Tim St. Pierre <fs-list at communicatefreely.net> wrote:

> Hi Steven,
>
> Yes, we are running our FreeSwitch deployment on FreeBSD 8.1 and 8.2  I
> would be glad to try it out, although it might be a few days before I
> get to it.
>
> What is it's purpose, and how would I know if it's working well or not?
>
> Thanks!
>
> -Tim
>
> Steven Ayre wrote:
> > Hi everyone,
> >
> > I've submitted a Jira with a proposed new mod_kqueue timer module.
> > This is a BSD equivalent of mod_timerfd.
> > http://jira.freeswitch.org/browse/FS-3398
> >
> > I'd appreciate it if anyone that uses FreeSWITCH on BSD that feels
> > like it would give it a quick test to see how well it works.
> >
> > To install it:
> > - Create a new directory src/mod/timers/mod_kqueue
> > - Place the mod_kqueue.c file from the jira in that directory
> > - Add timers/mod_kqueue to modules.conf
> > - Build and install
> > - Load it from modules.conf.xml or fs_cli.
> >
> > Regards,
> > Steven Ayre
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > 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
> >
>
>
> _______________________________________________
> 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 boris at tagnet.ru  Thu Jul  7 18:36:20 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 07 Jul 2011 20:36:20 +0600
Subject: [Freeswitch-users] SIP 100 response and media
In-Reply-To: <CAKDfjgdK+BrunGC0UY=greJw_DpUftermo_Dzp9k9kyKhesbew@mail.gmail.com>
References: <4E154D19.1080100@tagnet.ru>
	<CAKDfjgdK+BrunGC0UY=greJw_DpUftermo_Dzp9k9kyKhesbew@mail.gmail.com>
Message-ID: <4E15C464.8020706@tagnet.ru>

Hello!

     I tried but.... it works only with <action application="export" 
data="verbose_sdp=true"/>, it this right? I thinked simple "set" is enough.
> http://wiki.freeswitch.org/wiki/Variable_verbose_sdp
>
> Set this to true in your dialplan before calling bridge when placing a
> call to the Cisco.
>
> On Thu, Jul 7, 2011 at 2:07 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> Hello!
>>
>>      This thread again. There are two logs. First between Cisco<->Cisco
>> call and second between Cisco<->FreeSwitch. Cisco<->Cisco works fine. As
>> I can see the difference is that
>> Cisco has a=rtpmap:8 PCMA/8000 and FreeSwitch not. May be this is a
>> problem? Is this field mandatory?
>>
>>
>> Jul  6 18:20:39.341: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>> Received:
>> INVITE sip:28032 at 192.168.35.15:5060 SIP/2.0
>> Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F
>> Remote-Party-ID: "test"
>> <sip:30030 at 10.240.14.15>;party=calling;screen=no;privacy=off
>> From: "test"<sip:30030 at 10.240.14.15>;tag=3F5B1E64-2263
>> To:<sip:28032 at 192.168.35.15>
>> Date: Wed, 06 Jul 2011 18:20:39 GMT
>> Call-ID: 7C686709-A73311E0-95798064-7F05C5CA at 10.240.14.15
>> Supported: 100rel,timer,resource-priority,replaces,sdp-anat
>> Min-SE:  1800
>> Cisco-Guid: 2076058060-2805141984-2507440228-2131084746
>> User-Agent: Cisco-SIPGateway/IOS-12.x
>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>> CSeq: 101 INVITE
>> Max-Forwards: 70
>> Timestamp: 1309976439
>> Contact:<sip:30030 at 10.240.14.15:5060>
>> Expires: 180
>> Allow-Events: telephone-event
>> Content-Type: application/sdp
>> Content-Disposition: session;handling=required
>> Content-Length: 187
>>
>> v=0
>> o=CiscoSystemsSIP-GW-UserAgent 0 703 IN IP4 10.240.14.15
>> s=SIP Call
>> c=IN IP4 10.240.14.15
>> t=0 0
>> m=audio 18720 RTP/AVP 8
>> c=IN IP4 10.240.14.15
>> a=rtpmap:8 PCMA/8000
>> a=ptime:20
>>
>> Jul  6 18:20:39.361: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>> Sent:
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F
>> From: "test"<sip:30030 at 10.240.14.15>;tag=3F5B1E64-2263
>> To:<sip:28032 at 192.168.35.15>
>> Date: Wed, 06 Jul 2011 18:20:39 GMT
>> Call-ID: 7C686709-A73311E0-95798064-7F05C5CA at 10.240.14.15
>> Timestamp: 1309976439
>> CSeq: 101 INVITE
>> Allow-Events: telephone-event
>> Server: Cisco-SIPGateway/IOS-12.x
>> Content-Length: 0
>>
>>
>> Jul  6 18:20:39.369: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>> Sent:
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F
>> From: "test"<sip:30030 at 10.240.14.15>;tag=3F5B1E64-2263
>> To:<sip:28032 at 192.168.35.15>;tag=CBEFC0-16E4
>> Date: Wed, 06 Jul 2011 18:20:39 GMT
>> Call-ID: 7C686709-A73311E0-95798064-7F05C5CA at 10.240.14.15
>> Timestamp: 1309976439
>> CSeq: 101 INVITE
>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>> Allow-Events: telephone-event
>> Contact:<sip:28032 at 192.168.35.15:5060>
>> Supported: replaces
>> Supported: sdp-anat
>> Server: Cisco-SIPGateway/IOS-12.x
>> Content-Type: application/sdp
>> Content-Disposition: session;handling=required
>> Content-Length: 194
>>
>>
>> When call is between Cisco<->Freeswitch messages are
>> Jul  6 18:21:11.966: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>> Received:
>> INVITE sip:73435230030 at 192.168.3.6:5060 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp
>> Max-Forwards: 67
>> From: "anonymous"<sip:0000000000 at 192.168.1.1>;tag=QFKH52Z9NaN2c
>> To:<sip:73435230030 at 192.168.3.6:5060>
>> Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac
>> CSeq: 14652171 INVITE
>> Contact:<sip:mod_sofia at 192.168.1.1:5060>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1c95ad9 2011-01-20
>> 22-43-50 -0300
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE
>> Supported: 100rel, 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: 329
>> X-FS-Support: update_display
>> Remote-Party-ID: "anonymous"
>> <sip:0000000000 at 192.168.1.1>;party=calling;screen=yes;privacy=off
>>
>>
>> v=0
>> o=FreeSWITCH 1309956111 1309956112 IN IP4 192.168.1.1
>> s=FreeSWITCH
>> c=IN IP4 192.168.1.1
>> t=0 0
>>
>> m=audio 20360 RTP/AVP 8 0 18 101 13
>> a=fmtp:18 annexb=no
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> m=video 19030 RTP/AVP 31 34 98
>> a=rtpmap:31 H261/90000
>> a=rtpmap:34 H263/90000
>> a=rtpmap:98 H264/90000
>>
>>
>> Jul  6 18:21:11.986: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>> Sent:
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp
>> From: "anonymous"<sip:0000000000 at 192.168.1.1>;tag=QFKH52Z9NaN2c
>> To:<sip:73435230030 at 192.168.3.6:5060>
>> Date: Wed, 06 Jul 2011 18:21:11 GMT
>> Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac
>> CSeq: 14652171 INVITE
>> Allow-Events: telephone-event
>> Server: Cisco-SIPGateway/IOS-12.x
>> Content-Length: 0
>>
>>
>>
>>
>>
>> --
>> ? ?????????,
>>    ????? ?????????
>>    ??? "??????"
>>    ???. +7 (3435) 230001
>>    ???? +7 (3435) 230005
>>
>>
>> _______________________________________________
>> 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
>>
>
>


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005



From david.villasmil.work at gmail.com  Thu Jul  7 19:05:32 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 7 Jul 2011 17:05:32 +0200
Subject: [Freeswitch-users] absolute_codec_string
Message-ID: <CAFGRPVrekdj81XYWULO4kgccTGSQMFC5=tVEtJ+QgHHX2D3sug@mail.gmail.com>

Hello Guys,

I might be getting it wrong, but i'm trying to send out for the B-leg ONLY
g.729 with this in the dial  plan:

<variable name="absolute_codec_string" value="G729"/>

FS is still offering what it gets on the A-leg...

any ideas?

Thanks

David
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From david.villasmil.work at gmail.com  Thu Jul  7 19:10:19 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 7 Jul 2011 17:10:19 +0200
Subject: [Freeswitch-users] absolute_codec_string
In-Reply-To: <CAFGRPVrekdj81XYWULO4kgccTGSQMFC5=tVEtJ+QgHHX2D3sug@mail.gmail.com>
References: <CAFGRPVrekdj81XYWULO4kgccTGSQMFC5=tVEtJ+QgHHX2D3sug@mail.gmail.com>
Message-ID: <CAFGRPVqmnTH0LX_pcLZRb2F6q93QzbjV94BczGRDbxvyELjvVg@mail.gmail.com>

Never mind, I was using:

session:execute("set","proxy_media=true")

I took it out and it seems to be working now :)

Thanks!

David

On Thu, Jul 7, 2011 at 5:05 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello Guys,
>
> I might be getting it wrong, but i'm trying to send out for the B-leg ONLY
> g.729 with this in the dial  plan:
>
> <variable name="absolute_codec_string" value="G729"/>
>
> FS is still offering what it gets on the A-leg...
>
> any ideas?
>
> Thanks
>
> David
>
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From bryansmart at bryansmart.com  Thu Jul  7 20:14:07 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Thu, 7 Jul 2011 12:14:07 -0400
Subject: [Freeswitch-users] Playing multiple files simultaneously
In-Reply-To: <4E1507B2.3090803@coppice.org>
References: <2DB72AF2-2394-419A-ADC5-0B76FE594662@bryansmart.com>
	<4E1507B2.3090803@coppice.org>
Message-ID: <95BE2DAB-4EDC-4DE3-AE6F-A9119CF07992@bryansmart.com>

Thanks for the warning, Steve.

The cues are short, and I'm trying to keep them thin (simple waveforms based on multifrequency sine waves for tone, and short filtered white noise where clicks or percussive transients are required). I'm not an expert on low bit rate audio, but, from the bit that I know, I think that simple audio cues like this will mix better than sampled musical instruments playing riffs, real world sounds, and so on. Sounds alright on GSM, but I haven't tested with G729 yet. Do you think this will work for me?

I'm not actually planning an overlapping cacophony of sound. The cues are short, but I'd like them to be able to play in the background while text to speech or a recorded prompt is playing. I don't anticipate lots of them playing at once, but I wanted the mixing capability so that they could play under the speech, and so that, when two play close in a row, the second one wouldn't chop off the end of the first.

Do you know if it is better to try to extend existing commands, like playback, or if it would be better to have a separate module and commands for playing audio in this way? I realize that mixing would introduce at least a small overhead, even if all files and the channel were same format, and most people would want to avoid that little performance hit unless they specifically needed the mixing capability.

I feel cautious about poking my nose in to programming issues here. I've been programming to one degree or another for 25 years, and even had it as my main gig for several years, but all of my experience has either been personal projects, or closed source commercial development for an employer. I don't have my head around people and technical processes for contributing to an open source project, so even tweaking a single line in a source file seems to involve a lot of process in order to submit it. I know that the group uses git for version control, but, if I make a tweak, do I submit it to some sort of approval branch first? If their is a processes doc for the group, I'd be glad to read it.

Thanks
Bryan

On Jul 6, 2011, at 9:11 PM, Steve Underwood wrote:

> On 07/01/2011 01:13 PM, Bryan Smart wrote:
>> Is it possible for Freeswitch to play more than one file to a channel at a time? What I've seen and tried from the dialplan and scripts either queues files to play, or will stop a currently playing file so that the newly requested file will play. This also seems to be the case in conferences. When I send multiple play commands to conferences, the files are queued.
>> 
>> As for how this might be used, think of an IVR that plays queued prompts, yet continuously plays looping music or a Shoutcast stream in the background. I also want to be able to play short cue tones that start at the same time as a prompt (don't want to pre-mix them in to a single file, though).
>> 
>> Is this currently possible through any means? Perhaps with the event socket?
>> 
> Be really careful making plans like this. Multiple sounds will mix 
> nicely with something like ALaw or uLaw. If you try it with a low bit 
> rate codec, like G.729 or a cellphone codec, the result will sound 
> awful. Modern telephony really is geared to one voice and one voice only 
> at any moment.
> 
> Steve
> 
> 
> _______________________________________________
> 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 msc at freeswitch.org  Thu Jul  7 20:22:54 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 7 Jul 2011 09:22:54 -0700
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
Message-ID: <CAKzWOxU4VJQZuhL=DbmC+keKAm58=+TDCJ0AXLWNSmgsfLRAsQ@mail.gmail.com>

On Thu, Jul 7, 2011 at 4:27 AM, Mathieu Lautram
<lautram.mathieu at gmail.com>wrote:

> I've already tried those parameters and it doesn't work.
> I can use ignore_early_medi=true and ringback=${us-ring} but what if my B
> leg is in France? Or in England?
>
You can set the ringback to whatever you'd like. The trick will be doing
some sort of lookup on the dialed number to decide what the country is and
then use the appropriate ringback tone. If you look in vars.xml we have
quite a few countries in there. If any are missing you can always look them
up and add them to vars.xml. There are plenty of people who can assist with
the TGML if you can track down the frequencies and cadences for each country
you need. From there a simple Lua script or even mod_lcr could be employed
to handle the creation of the dialstring.

I suppose I should have asked this question earlier: why, exactly do you
need to ignore early media? The reason I ask is that there might be an
alternative way to address the problem you are trying to solve by ignoring
early media.

Thanks,
MC


> The schema is like that:
>
> France:
>
> A leg have to hear the ringtone of France
>
> US:
>
> A leg have to hear the ringtone of the US
>
> England:
>
> A leg have to hear the ringtone of England
>
> Other countries:
>
> same thing
>
> Have I to check the destination to apply the correct ringtone or is there
> an other solution to let A leg hear the ringtone of the correct country?
> Is Freeswitch allow me to know if the call to the B leg is successful for a
> "connecting" state (and not "attempting" state) with ignore_early_media set
> to false?
>
> Best regards
>
> 2011/7/7 Anthony Minessale <anthony.minessale at gmail.com>
>
>> try ignore_early_media=ring_ready or set ringback on a to ${us-ring}
>> and ignore_early_media=true
>>
>>
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From kris at kriskinc.com  Thu Jul  7 20:52:15 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 7 Jul 2011 12:52:15 -0400
Subject: [Freeswitch-users] SIP 100 response and media
In-Reply-To: <4E15C464.8020706@tagnet.ru>
References: <4E154D19.1080100@tagnet.ru>
	<CAKDfjgdK+BrunGC0UY=greJw_DpUftermo_Dzp9k9kyKhesbew@mail.gmail.com>
	<4E15C464.8020706@tagnet.ru>
Message-ID: <CAKDfjgc5QA8RjzbcnDbXZ5_McZnH93A45Uz6qFNDXX4Wy5mFcw@mail.gmail.com>

Export would set it on the a and b legs (unless you use nolocal).
That is what you want.

On Thu, Jul 7, 2011 at 10:36 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ? ? I tried but.... it works only with <action application="export"
> data="verbose_sdp=true"/>, it this right? I thinked simple "set" is enough.
>> http://wiki.freeswitch.org/wiki/Variable_verbose_sdp
>>
>> Set this to true in your dialplan before calling bridge when placing a
>> call to the Cisco.
>>
>> On Thu, Jul 7, 2011 at 2:07 AM, Boris Kovalenko<boris at tagnet.ru> ?wrote:
>>> Hello!
>>>
>>> ? ? ?This thread again. There are two logs. First between Cisco<->Cisco
>>> call and second between Cisco<->FreeSwitch. Cisco<->Cisco works fine. As
>>> I can see the difference is that
>>> Cisco has a=rtpmap:8 PCMA/8000 and FreeSwitch not. May be this is a
>>> problem? Is this field mandatory?
>>>
>>>
>>> Jul ?6 18:20:39.341: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>> Received:
>>> INVITE sip:28032 at 192.168.35.15:5060 SIP/2.0
>>> Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F
>>> Remote-Party-ID: "test"
>>> <sip:30030 at 10.240.14.15>;party=calling;screen=no;privacy=off
>>> From: "test"<sip:30030 at 10.240.14.15>;tag=3F5B1E64-2263
>>> To:<sip:28032 at 192.168.35.15>
>>> Date: Wed, 06 Jul 2011 18:20:39 GMT
>>> Call-ID: 7C686709-A73311E0-95798064-7F05C5CA at 10.240.14.15
>>> Supported: 100rel,timer,resource-priority,replaces,sdp-anat
>>> Min-SE: ?1800
>>> Cisco-Guid: 2076058060-2805141984-2507440228-2131084746
>>> User-Agent: Cisco-SIPGateway/IOS-12.x
>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>> CSeq: 101 INVITE
>>> Max-Forwards: 70
>>> Timestamp: 1309976439
>>> Contact:<sip:30030 at 10.240.14.15:5060>
>>> Expires: 180
>>> Allow-Events: telephone-event
>>> Content-Type: application/sdp
>>> Content-Disposition: session;handling=required
>>> Content-Length: 187
>>>
>>> v=0
>>> o=CiscoSystemsSIP-GW-UserAgent 0 703 IN IP4 10.240.14.15
>>> s=SIP Call
>>> c=IN IP4 10.240.14.15
>>> t=0 0
>>> m=audio 18720 RTP/AVP 8
>>> c=IN IP4 10.240.14.15
>>> a=rtpmap:8 PCMA/8000
>>> a=ptime:20
>>>
>>> Jul ?6 18:20:39.361: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>> Sent:
>>> SIP/2.0 100 Trying
>>> Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F
>>> From: "test"<sip:30030 at 10.240.14.15>;tag=3F5B1E64-2263
>>> To:<sip:28032 at 192.168.35.15>
>>> Date: Wed, 06 Jul 2011 18:20:39 GMT
>>> Call-ID: 7C686709-A73311E0-95798064-7F05C5CA at 10.240.14.15
>>> Timestamp: 1309976439
>>> CSeq: 101 INVITE
>>> Allow-Events: telephone-event
>>> Server: Cisco-SIPGateway/IOS-12.x
>>> Content-Length: 0
>>>
>>>
>>> Jul ?6 18:20:39.369: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>> Sent:
>>> SIP/2.0 200 OK
>>> Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F
>>> From: "test"<sip:30030 at 10.240.14.15>;tag=3F5B1E64-2263
>>> To:<sip:28032 at 192.168.35.15>;tag=CBEFC0-16E4
>>> Date: Wed, 06 Jul 2011 18:20:39 GMT
>>> Call-ID: 7C686709-A73311E0-95798064-7F05C5CA at 10.240.14.15
>>> Timestamp: 1309976439
>>> CSeq: 101 INVITE
>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>> Allow-Events: telephone-event
>>> Contact:<sip:28032 at 192.168.35.15:5060>
>>> Supported: replaces
>>> Supported: sdp-anat
>>> Server: Cisco-SIPGateway/IOS-12.x
>>> Content-Type: application/sdp
>>> Content-Disposition: session;handling=required
>>> Content-Length: 194
>>>
>>>
>>> When call is between Cisco<->Freeswitch messages are
>>> Jul ?6 18:21:11.966: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>> Received:
>>> INVITE sip:73435230030 at 192.168.3.6:5060 SIP/2.0
>>> Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp
>>> Max-Forwards: 67
>>> From: "anonymous"<sip:0000000000 at 192.168.1.1>;tag=QFKH52Z9NaN2c
>>> To:<sip:73435230030 at 192.168.3.6:5060>
>>> Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac
>>> CSeq: 14652171 INVITE
>>> Contact:<sip:mod_sofia at 192.168.1.1:5060>
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1c95ad9 2011-01-20
>>> 22-43-50 -0300
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>>> REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE
>>> Supported: 100rel, 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: 329
>>> X-FS-Support: update_display
>>> Remote-Party-ID: "anonymous"
>>> <sip:0000000000 at 192.168.1.1>;party=calling;screen=yes;privacy=off
>>>
>>>
>>> v=0
>>> o=FreeSWITCH 1309956111 1309956112 IN IP4 192.168.1.1
>>> s=FreeSWITCH
>>> c=IN IP4 192.168.1.1
>>> t=0 0
>>>
>>> m=audio 20360 RTP/AVP 8 0 18 101 13
>>> a=fmtp:18 annexb=no
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>> m=video 19030 RTP/AVP 31 34 98
>>> a=rtpmap:31 H261/90000
>>> a=rtpmap:34 H263/90000
>>> a=rtpmap:98 H264/90000
>>>
>>>
>>> Jul ?6 18:21:11.986: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>> Sent:
>>> SIP/2.0 100 Trying
>>> Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp
>>> From: "anonymous"<sip:0000000000 at 192.168.1.1>;tag=QFKH52Z9NaN2c
>>> To:<sip:73435230030 at 192.168.3.6:5060>
>>> Date: Wed, 06 Jul 2011 18:21:11 GMT
>>> Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac
>>> CSeq: 14652171 INVITE
>>> Allow-Events: telephone-event
>>> Server: Cisco-SIPGateway/IOS-12.x
>>> Content-Length: 0
>>>
>>>
>>>
>>>
>>>
>>> --
>>> ? ?????????,
>>> ? ?????? ?????????
>>> ? ???? "??????"
>>> ? ????. +7 (3435) 230001
>>> ? ????? +7 (3435) 230005
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>
>>
>
>
> --
> ? ?????????,
> ? ????? ?????????
> ? ??? "??????"
> ? ???. +7 (3435) 230001
> ? ???? +7 (3435) 230005
>
>
> _______________________________________________
> 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
>



-- 
Kristian Kielhofner


From david.ponzone at ipeva.fr  Thu Jul  7 20:54:08 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 7 Jul 2011 18:54:08 +0200
Subject: [Freeswitch-users] absolute_codec_string
In-Reply-To: <CAFGRPVrekdj81XYWULO4kgccTGSQMFC5=tVEtJ+QgHHX2D3sug@mail.gmail.com>
References: <CAFGRPVrekdj81XYWULO4kgccTGSQMFC5=tVEtJ+QgHHX2D3sug@mail.gmail.com>
Message-ID: <B252EFB5-B3BC-4C59-A522-6D523423991B@ipeva.fr>

David,

in the dialplan, you are supposed to set a variable with

<action application="export" data="absolute_codec_string=G729" />

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

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

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




Le 07/07/2011 ? 17:05, David Villasmil a ?crit :

> Hello Guys,
> 
> I might be getting it wrong, but i'm trying to send out for the B-leg ONLY g.729 with this in the dial  plan:
> 
> <variable name="absolute_codec_string" value="G729"/>
> 
> FS is still offering what it gets on the A-leg...
> 
> any ideas?
> 
> Thanks
> 
> David
> _______________________________________________
> 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 david.ponzone at ipeva.fr  Thu Jul  7 20:56:56 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 7 Jul 2011 18:56:56 +0200
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAKzWOxU4VJQZuhL=DbmC+keKAm58=+TDCJ0AXLWNSmgsfLRAsQ@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
	<CAKzWOxU4VJQZuhL=DbmC+keKAm58=+TDCJ0AXLWNSmgsfLRAsQ@mail.gmail.com>
Message-ID: <E8F1B32B-017A-442A-86EB-118F30AB9E9D@ipeva.fr>

Because he wants to detect when the call is really answered.
But what I don't understand is that he should have that info with ESL.

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

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

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




Le 07/07/2011 ? 18:22, Michael Collins a ?crit :

> 
> 
> On Thu, Jul 7, 2011 at 4:27 AM, Mathieu Lautram <lautram.mathieu at gmail.com> wrote:
> I've already tried those parameters and it doesn't work.
> I can use ignore_early_medi=true and ringback=${us-ring} but what if my B leg is in France? Or in England?
> You can set the ringback to whatever you'd like. The trick will be doing some sort of lookup on the dialed number to decide what the country is and then use the appropriate ringback tone. If you look in vars.xml we have quite a few countries in there. If any are missing you can always look them up and add them to vars.xml. There are plenty of people who can assist with the TGML if you can track down the frequencies and cadences for each country you need. From there a simple Lua script or even mod_lcr could be employed to handle the creation of the dialstring.
> 
> I suppose I should have asked this question earlier: why, exactly do you need to ignore early media? The reason I ask is that there might be an alternative way to address the problem you are trying to solve by ignoring early media.
> 
> Thanks,
> MC
>  
> The schema is like that:
> 
> France:
> A leg have to hear the ringtone of France
> US:
> A leg have to hear the ringtone of the US
> England:
> A leg have to hear the ringtone of England
> Other countries:
> same thing
> 
> Have I to check the destination to apply the correct ringtone or is there an other solution to let A leg hear the ringtone of the correct country?
> Is Freeswitch allow me to know if the call to the B leg is successful for a "connecting" state (and not "attempting" state) with ignore_early_media set to false?
> 
> Best regards
> 
> 2011/7/7 Anthony Minessale <anthony.minessale at gmail.com>
> try ignore_early_media=ring_ready or set ringback on a to ${us-ring}
> and ignore_early_media=true
> 
> _______________________________________________
> 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 vaad.fabi at gmail.com  Thu Jul  7 21:09:06 2011
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Thu, 07 Jul 2011 20:09:06 +0300
Subject: [Freeswitch-users] Flex RTMP Client Added To Git Tree!
In-Reply-To: <CAKzWOxXgDVPWdQH90y9OQz0kJP8UMT_MeKfhWWki78cui-WHrA@mail.gmail.com>
References: <CAKzWOxXgDVPWdQH90y9OQz0kJP8UMT_MeKfhWWki78cui-WHrA@mail.gmail.com>
Message-ID: <4E15E832.3030803@gmail.com>

  Nice work!
Tnx guys! ;)



On 07/06/2011 08:27 PM, Michael Collins wrote:
> I thought many of you would be interested in this story about the RTMP 
> Flex client:
>
> http://www.freeswitch.org/node/332
>
> The sample Flex client has been added to tree, so feel free to start 
> experimenting!
>
> -Michael
>
>
> _______________________________________________
> 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


-- 
Vadim F.

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From lautram.mathieu at gmail.com  Thu Jul  7 21:19:27 2011
From: lautram.mathieu at gmail.com (Mathieu Lautram)
Date: Thu, 7 Jul 2011 19:19:27 +0200
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <E8F1B32B-017A-442A-86EB-118F30AB9E9D@ipeva.fr>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
	<CAKzWOxU4VJQZuhL=DbmC+keKAm58=+TDCJ0AXLWNSmgsfLRAsQ@mail.gmail.com>
	<E8F1B32B-017A-442A-86EB-118F30AB9E9D@ipeva.fr>
Message-ID: <CAMNz+btY4OVNot6VH-E0Lgsheo9O7p0HFUMb3RHgGMYyCd_Ruw@mail.gmail.com>

No, David, It's not exactly the point.
I don't really want to use ignore_early_media, but this is the only way to
know if the call is correctly connected (and not answered yet).
I can't use the several ringtones located in vars.xml because the B leg
could be everywhere in the world, so there are too much possibilities. Also,
sometimes B leg will have to hear additionnal informations ; not only a
ringtone.
So with ESL, is there a way to know if the B leg is really ringing? Or
perhaps without ESL? Maybe I'm in the wrong way...

Thanks for all your answers

Best Regards

2011/7/7 David Ponzone <david.ponzone at ipeva.fr>

> Because he wants to detect when the call is really answered.
> But what I don't understand is that he should have that info with ESL.
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 07/07/2011 ? 18:22, Michael Collins a ?crit :
>
>
>
> On Thu, Jul 7, 2011 at 4:27 AM, Mathieu Lautram <lautram.mathieu at gmail.com
> > wrote:
>
>> I've already tried those parameters and it doesn't work.
>> I can use ignore_early_medi=true and ringback=${us-ring} but what if my B
>> leg is in France? Or in England?
>>
> You can set the ringback to whatever you'd like. The trick will be doing
> some sort of lookup on the dialed number to decide what the country is and
> then use the appropriate ringback tone. If you look in vars.xml we have
> quite a few countries in there. If any are missing you can always look them
> up and add them to vars.xml. There are plenty of people who can assist with
> the TGML if you can track down the frequencies and cadences for each country
> you need. From there a simple Lua script or even mod_lcr could be employed
> to handle the creation of the dialstring.
>
> I suppose I should have asked this question earlier: why, exactly do you
> need to ignore early media? The reason I ask is that there might be an
> alternative way to address the problem you are trying to solve by ignoring
> early media.
>
> Thanks,
> MC
>
>
>> The schema is like that:
>>
>> France:
>>
>> A leg have to hear the ringtone of France
>>
>> US:
>>
>> A leg have to hear the ringtone of the US
>>
>> England:
>>
>> A leg have to hear the ringtone of England
>>
>> Other countries:
>>
>> same thing
>>
>> Have I to check the destination to apply the correct ringtone or is there
>> an other solution to let A leg hear the ringtone of the correct country?
>> Is Freeswitch allow me to know if the call to the B leg is successful for
>> a "connecting" state (and not "attempting" state) with ignore_early_media
>> set to false?
>>
>> Best regards
>>
>> 2011/7/7 Anthony Minessale <anthony.minessale at gmail.com>
>>
>>> try ignore_early_media=ring_ready or set ringback on a to ${us-ring}
>>> and ignore_early_media=true
>>>
>>> _______________________________________________
> 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
>
>
>
> _______________________________________________
> 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
>
>


-- 
Mathieu LAUTRAM
Application developer

BJT Partners - FRANCE
+33 1 79 75 99 60
+33 6 61 59 07 25
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From freeswitch at peely.com  Thu Jul  7 22:16:16 2011
From: freeswitch at peely.com (peely)
Date: Thu, 7 Jul 2011 11:16:16 -0700 (PDT)
Subject: [Freeswitch-users] RTMP buffer size?
In-Reply-To: <1309955603001-6554200.post@n2.nabble.com>
References: <1309955603001-6554200.post@n2.nabble.com>
Message-ID: <1310062576455-6559405.post@n2.nabble.com>

Doing some more analysis, the noticable latency only seems to be on the
receive side of the Flash control i.e. through the speakers of the PC using
the Flash control. From the Flash control out to the PSTN the delay seems
reasonably low.

I can't compile the flash control from the source provided as I don't have
Flex. The only thing noticable I can see if the incomingNetStream.bufferTime
which seems to be 0.2 seconds in freeswitch.mxml.

Is there any chance of having this configuable?


Thanks,


Neil.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/RTMP-buffer-size-tp6554200p6559405.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From periferral at gmail.com  Thu Jul  7 10:27:00 2011
From: periferral at gmail.com (Avi)
Date: Wed, 06 Jul 2011 23:27:00 -0700
Subject: [Freeswitch-users] mod dingaling, gv incoming calls dialplan
Message-ID: <4E1551B4.7020204@gmail.com>

Hello all. 

I apologize for the newbie question but after much reading, I can't seem to figure out what the incoming dialplan for GV/extension needs to look like. Currently I have what is listed in the twiki page for dingaling and google voice. However there is a line in the twiki page that says
"Put the following in your dialplan (same context as the receiving extension)". I'm not sure what the receiving extension should look like since I can't seem to find an example of that.

Here is what I'm seeing in the logs

http://pastebin.com/7j5e7qTm

Since I have nothing for the extension 2001, I presume the call hangs up on the FS end. The extension never rings. Any help is appreciated and I sure hope I get some.

FYI, outgoing calls seems to be working just fine.

Thanks much.


From msc at freeswitch.org  Thu Jul  7 22:26:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 7 Jul 2011 11:26:36 -0700
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAMNz+btY4OVNot6VH-E0Lgsheo9O7p0HFUMb3RHgGMYyCd_Ruw@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
	<CAKzWOxU4VJQZuhL=DbmC+keKAm58=+TDCJ0AXLWNSmgsfLRAsQ@mail.gmail.com>
	<E8F1B32B-017A-442A-86EB-118F30AB9E9D@ipeva.fr>
	<CAMNz+btY4OVNot6VH-E0Lgsheo9O7p0HFUMb3RHgGMYyCd_Ruw@mail.gmail.com>
Message-ID: <CAKzWOxUQdve-a+PH91aAADGBGpqZhUBLUitbThuboCwejHhLmw@mail.gmail.com>

Can you lay out all the possibilities for the B leg? What are all the
scenarios that you have to account for? In other words, under what
circumstances would *not* ignoring early media fail for you? If you pass the
early media from the B leg to the A leg, what are the problems you would
face?
-MC

On Thu, Jul 7, 2011 at 10:19 AM, Mathieu Lautram
<lautram.mathieu at gmail.com>wrote:

> No, David, It's not exactly the point.
> I don't really want to use ignore_early_media, but this is the only way to
> know if the call is correctly connected (and not answered yet).
> I can't use the several ringtones located in vars.xml because the B leg
> could be everywhere in the world, so there are too much possibilities. Also,
> sometimes B leg will have to hear additionnal informations ; not only a
> ringtone.
> So with ESL, is there a way to know if the B leg is really ringing? Or
> perhaps without ESL? Maybe I'm in the wrong way...
>
> Thanks for all your answers
>
> Best Regards
>
> 2011/7/7 David Ponzone <david.ponzone at ipeva.fr>
>
>> Because he wants to detect when the call is really answered.
>> But what I don't understand is that he should have that info with ESL.
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 07/07/2011 ? 18:22, Michael Collins a ?crit :
>>
>>
>>
>> On Thu, Jul 7, 2011 at 4:27 AM, Mathieu Lautram <
>> lautram.mathieu at gmail.com> wrote:
>>
>>> I've already tried those parameters and it doesn't work.
>>> I can use ignore_early_medi=true and ringback=${us-ring} but what if my B
>>> leg is in France? Or in England?
>>>
>> You can set the ringback to whatever you'd like. The trick will be doing
>> some sort of lookup on the dialed number to decide what the country is and
>> then use the appropriate ringback tone. If you look in vars.xml we have
>> quite a few countries in there. If any are missing you can always look them
>> up and add them to vars.xml. There are plenty of people who can assist with
>> the TGML if you can track down the frequencies and cadences for each country
>> you need. From there a simple Lua script or even mod_lcr could be employed
>> to handle the creation of the dialstring.
>>
>> I suppose I should have asked this question earlier: why, exactly do you
>> need to ignore early media? The reason I ask is that there might be an
>> alternative way to address the problem you are trying to solve by ignoring
>> early media.
>>
>> Thanks,
>> MC
>>
>>
>>> The schema is like that:
>>>
>>> France:
>>>
>>> A leg have to hear the ringtone of France
>>>
>>> US:
>>>
>>> A leg have to hear the ringtone of the US
>>>
>>> England:
>>>
>>> A leg have to hear the ringtone of England
>>>
>>> Other countries:
>>>
>>> same thing
>>>
>>> Have I to check the destination to apply the correct ringtone or is there
>>> an other solution to let A leg hear the ringtone of the correct country?
>>> Is Freeswitch allow me to know if the call to the B leg is successful for
>>> a "connecting" state (and not "attempting" state) with ignore_early_media
>>> set to false?
>>>
>>> Best regards
>>>
>>> 2011/7/7 Anthony Minessale <anthony.minessale at gmail.com>
>>>
>>>> try ignore_early_media=ring_ready or set ringback on a to ${us-ring}
>>>> and ignore_early_media=true
>>>>
>>>> _______________________________________________
>> 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
>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
>
> --
> Mathieu LAUTRAM
> Application developer
>
> BJT Partners - FRANCE
> +33 1 79 75 99 60
> +33 6 61 59 07 25
>
> _______________________________________________
> 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 justlikeef at gmail.com  Thu Jul  7 23:39:26 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Thu, 7 Jul 2011 15:39:26 -0400
Subject: [Freeswitch-users] Having problem completing calls between two
	phones behind double NAT...
Message-ID: <201107071539.27015.justlikeef@gmail.com>

The network layout it something like:

FS and Ext 3 <- Local Net ->Inet Rtr(Static NAT) <== INET ==>Inet Rtr(Dynamic NAT)<- Local Net -> Ext 1 and Ext 2

where:

The public IP of the Freeswitch box is static
	There are 2 sip profiles, one for the internal clients on 5060 and one for the NATd clients on 5070 with the external IP hard set and Freeswitch running with -nonat
Ext 3 is on the same subnet as the FS server and is using sipinterface_1 to connect
Ext 1 and Ext2 are on the same subnet on the remote side of the double nat and using sipinterfacec_2

I can call from Ext 1 to Ext 3 with no problem
I can call from Ext 2 to Ext 3 with no problem

When I try to call from Ext 1 to Ext 2 or Ext 2 to Ext 1, I get a 606 USER_NOT_REGISTERED

show registrations seem to indicate that they are, and I can't find any difference in the call processing to the extensions that it works to, and those that it doesn't.

Pastebin with registrations and failed call log is: http://pastebin.freeswitch.org/16705

Thanks in advance for any help...


From nazim.aghabayov at gmail.com  Fri Jul  8 00:27:39 2011
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Fri, 08 Jul 2011 01:27:39 +0500
Subject: [Freeswitch-users] Freeswitch dynamic routing of all calls
In-Reply-To: <BANLkTi=UJ0+GKwd3T2uom=MuS4aLqudDmA@mail.gmail.com>
References: <BANLkTimuCjRcVccyrPpgQeY6N+UamyNQ=A@mail.gmail.com>	<518E6CA8-4198-4484-BF03-2DEFFA6493EA@gmail.com>
	<BANLkTi=UJ0+GKwd3T2uom=MuS4aLqudDmA@mail.gmail.com>
Message-ID: <4E1616BB.6000405@gmail.com>

Hello Mateusz,

I'm finding  ESL powerful yet complicated and using mod_xml_curl for 
dynamic routing.
mod_xml_curl with  with bindings = "dialplan" handles routing just fine.
http://wiki.freeswitch.org/wiki/Mod_xml_curl

Another approach is to a the Lua scripts for setting the variables 
"inside" the dialplan.
FreeSWITH is a flexible system, so there are a lot of ways to achieve 
the same result.

Best Regards,
Nazim


On 07/01/2011 05:55 PM, Mateusz Bartczak wrote:
> OK but how can I respond for received events?
>
> I subscribe to receive events using following event socket command:
>
> event plain CHANNEL_CREATE
>
> then I got all events of that type and that's great but the question is how
> can I handle those events, what's the syntax and where to put it?
>
> I will simply explain what I'm trying to achieve:
>
> 1. User dialed number
> 2. CHANNEL_CREATE event is created
> 3. I got this event using socket
> 4. What to do now? How to respond for that event? For example I would like
> to respond with dialstring to use, user called number 123, I would like to
> return something like sofia/gateway123/00123
>
>
>
> 2011/6/30 Steven Ayre<steveayre at gmail.com>
>
>> Yes it's possible and I suggest you look at esl
>>
>> Steve on iPhone
>>
>> On 30 Jun 2011, at 09:54, Mateusz Bartczak<netcentrica at gmail.com>  wrote:
>>
>>> Hi all
>>>
>>> I'm new to FS and I would like to know is it possible to implement
>> following scenario:
>>> 1. User dials number
>>> 2. Routing script detects outgoing call event. Every call needs to be
>> handled by routing script.
>>> 3. Routing script takes in input: user name, domain, dialed number. Than
>> it query database to find best SIP trunk to route the call, it also checks
>> destination price per minute and calculates maximum call time for prepaid
>> user.
>>> 4. Routing script output is: SIP trunk to use, SIP call parameters (ie.
>> callerid), maximum call duration
>>> 5. FS read output from routing script and make call using returned
>> parameters
>>> Preferred routing implementation technology: background running unix
>> deamon written in Java or PHP. Connection with FS via socket.
>>> Event routing script will be multi-threaded, must be able to deal with a
>> lot of calls in parallel and processing of one call should not block
>> processing of other calls (I have this problem with Yate voip server, and
>> that's really big problem)
>>> Is it possible to do this using FS?
>>> Any advices where to search for additional info? I know that there is
>> event handler but can it return "dialstring" for outgoing call events?
>>> Some code examples?
>>>
>>> I will really appreciate your help
>>>



From lloydie.t at gmail.com  Fri Jul  8 00:52:38 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Thu, 7 Jul 2011 21:52:38 +0100
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
 Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <CAP1_MOBAFpYqJNKhgWzv3aYJeKbj+RJDQJCaQByVcrMVp_s8RA@mail.gmail.com>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
	<4DF10C83.6020803@earthspike.net>
	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>
	<4DF1F07A.5000200@earthspike.net>
	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>
	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>
	<4DFB9831.4040709@earthspike.net>
	<BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>
	<CAP1_MOCq5qRjsgd7XGag9ukjHiG2Se-pXvrghQ7hbB5ciwJK+A@mail.gmail.com>
	<4E156252.2030308@earthspike.net>
	<CAP1_MOBAFpYqJNKhgWzv3aYJeKbj+RJDQJCaQByVcrMVp_s8RA@mail.gmail.com>
Message-ID: <CAE=3qOZaqoa1zt1_q_gaYzME=DOJYqhW4i7afhrBYc4PxViq5g@mail.gmail.com>

Might be able to help with a Trend, depending on where you are in UK

On 7 July 2011 12:21, shouldbe q931 <shouldbeq931 at gmail.com> wrote:

> On Thu, Jul 7, 2011 at 8:37 AM, John <freeswitch at earthspike.net> wrote:
> > On 06/07/11 13:53, shouldbe q931 wrote:
> >> On Fri, Jun 17, 2011 at 7:43 PM, shouldbe q931<shouldbeq931 at gmail.com>
>  wrote:
> >>> On Fri, Jun 17, 2011 at 7:08 PM, John<freeswitch at earthspike.net>
>  wrote:
> >>>> Mike, Shouldbe,
> >>>>
> >>>> We now have CLIP on the line for sure, and I ordered COLP at the same
> time
> >>>> and am assured that that has been turned on as well, but I still
> cannot get
> >>>> outgoing CLI to work properly.  Mike, you mention setting the TON to
> >>>> 'national'; where is that setting?
> >>>>
> >>>> At the moment, my dialplan looks like
> http://pastebin.freeswitch.org/16548
> >>>> with .  An outgoing call gives the log 7 at
> >>>> http://pastebin.freeswitch.org/16549 (numbers have been changed to
> protect
> >>>> the innocent!) but the number that 01234 567890 sees on their Caller
> ID is
> >>>> not 876543 but the main number of the line (876540, say).  I have
> tried
> >>>> setting the outgoing_caller_id_number to 6, 10 and 11 digits,
> restarting FS
> >>>> after each change, but with no success.
> >>>>
> >>>> The service provider has only one clue to offer: "With regards to the
> >>>> configuration at the exchange, the line is set to 6 digits to switch."
> which
> >>>> makes eminent sense, and suggests that I should be presenting 6
> digits.
> >>>> Incoming calls have a 6 digit called number and 10 digit calling
> number.
> >>>>
> >>>> Thanks for the help so far.  Any other ideas?
> >>>>
> >>>> John
> >>>>
> >>>>
> >>> I don't have any experience with FS connecting over ISDN so can't help
> >>> you much further on the FS configuration:-(
> >>>
> >>> However on the lines, I would be very tempted to see if you can borrow
> >>> a BRI tester, or what I've frequently used in the past is an
> >>> Eicon/Dialogic card as the test functions (under windows) are nearly
> >>> as good, that might help you see if it _is_ a BT problem, or a FS
> >>> configuration issue. It would need a 2k/XP machine, but something like
> >>> http://cgi.ebay.co.uk/EICON-DIVA-2-01-PCI-GRAPHICS-CARD-/270761571564
> >>> is what I have used in the past. I think I might have one of the
> >>> PCMCIA ones somewhere, but I'd need to test it still works
> >>>
> >>> I'm fairly sure that the "6 digits to switch" is what they are sending
> >>> to you, not what they are expecting from you. I can't access out
> >>> remaining BRI (Avaya) PBX from here, but on our PRI (again Avaya)
> >>> switches, we are sending 10 digits.
> >>>
> >>> Cheers
> >>>
> >> Apologies for the very long delay...
> >>
> >> On our PRI systems we are sending 10 digits (2071231234), and on the
> >> BRI system we are sending 11 digits (02071231234).
> >>
> >> Cheers
> > Thanks.  I have tried all of these digit options, along with setting the
> > TON and NPI, so I am now working through every possible option and tweak
> > with the supplier (including including codes to remove/reapply CPS).  I
> > am trying to avoid booking this as a fault, until I am sure I won't be
> > charged for a 'no fault found' but it is beginning to look unavoidable.
> > I am certainly building enough information for a good wiki page on this
> > when I finally get it cracked!  Incidentally, the Sangoma 'wanpipemon'
> > provides a useful PCAP mode which delivers the entire D-channel as a
> > wireshark PCAP file, so I am able to debug the Q.931 signalling down to
> > bit level.
> >
> > John
> >
>
> If you can see in the capture that you are sending the digits, and the
> CLI is not being set, then open a fault with BT.
>
> Alternatively, find a friendly "legacy" PBX maintainer that has a
> trend/harrier, or take your FS box to somewhere that has some BRI
> lines that are known to work with CLIP and COLP and try your FS box on
> them...
>
> _______________________________________________
> 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 anthony.minessale at gmail.com  Fri Jul  8 01:05:19 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Jul 2011 16:05:19 -0500
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
Message-ID: <CAKbxfG-h2_seFcYhu-5zwDbhD-z12xGuw7GmmKPPgSLGux3_zg@mail.gmail.com>

yes,

bridge_early_media=true

this is assuming that your provider is actually providing these intl ringtones

On Thu, Jul 7, 2011 at 6:27 AM, Mathieu Lautram
<lautram.mathieu at gmail.com> wrote:
> I've already tried those parameters and it doesn't work.
> I can use ignore_early_medi=true and ringback=${us-ring} but what if my B
> leg is in France? Or in England?
> The schema is like that:
> France:
>
> A leg have to hear the ringtone of France
>
> US:
>
> A leg have to hear the ringtone of the US
>
> England:
>
> A leg have to hear the ringtone of England
>
> Other countries:
>
> same thing
>
> Have I to check the destination to apply the correct ringtone or is there an
> other solution to let A leg hear the ringtone of the correct country?
> Is Freeswitch allow me to know if the call to the B leg is successful for a
> "connecting" state (and not "attempting" state) with ignore_early_media set
> to false?
> Best regards
>
> 2011/7/7 Anthony Minessale <anthony.minessale at gmail.com>
>>
>> try ignore_early_media=ring_ready or set ringback on a to ${us-ring}
>> and ignore_early_media=true
>>
>> On Wed, Jul 6, 2011 at 6:44 PM, Mathieu Lautram
>> <lautram.mathieu at gmail.com> wrote:
>> > I know that when I'm using ignore early media the A leg can't hear
>> > ringing.
>> > But the fact is when I set ignore early media to false, I don't know if
>> > the
>> > call is successful for the attempting or for the connecting. What I
>> > would
>> > like to have is the state "successful" but only for the connecting
>> > (that's
>> > why I set ignore early media to true).
>> > The originate string I am using is:
>> > bgapi originate {ignore_early_media=true}sofia/external/12345 at XXXXXX
>> > '&bridge( {ignore_early_media=true}sofia/external/2468 at XXXXXXX'
>> > Thank you for your answer
>> >
>> > 2011/7/7 Michael Collins <msc at freeswitch.org>
>> >>
>> >> What is the originate string you are using? Also, if you are ignoring
>> >> early media then how is the A leg supposed to hear ringing unless you
>> >> specify the ringback tone to use?
>> >> -MC
>> >>
>> >> On Wed, Jul 6, 2011 at 9:52 AM, Mathieu Lautram
>> >> <lautram.mathieu at gmail.com> wrote:
>> >>>
>> >>> Hi all
>> >>> I'm facing an issue with originate. Here is the case:
>> >>> I do an originate to bridge A leg to B leg using bridge application.
>> >>> The A leg works perfectly but, when the B leg is called, I can't hear
>> >>> ringtone in the A leg.
>> >>> I set ignore_early_media to true.
>> >>> I would like to hear a ringtone without setting the ringback variable
>> >>> with the tone of my country.
>> >>> Is it possible to do that?
>> >>> Thank you in advance.
>> >>>
>> >>> --
>> >>> Mathieu LAUTRAM
>> >>> Application developer
>> >>>
>> >>> BJT Partners - FRANCE
>> >>> +33 1 79 75 99 60
>> >>> +33 6 61 59 07 25
>> >>>
>> >>> _______________________________________________
>> >>> 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
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> 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
>> >>
>> >
>> >
>> >
>> > --
>> > Mathieu LAUTRAM
>> > Application developer
>> >
>> > BJT Partners - FRANCE
>> > +33 1 79 75 99 60
>> > +33 6 61 59 07 25
>> >
>> > _______________________________________________
>> > 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
>> >
>> >
>>
>>
>>
>> --
>> 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
>
>
>
> --
> Mathieu LAUTRAM
> Application developer
>
> BJT Partners - FRANCE
> +33 1 79 75 99 60
> +33 6 61 59 07 25
>
> _______________________________________________
> 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
>
>



-- 
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  Fri Jul  8 01:10:43 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Jul 2011 16:10:43 -0500
Subject: [Freeswitch-users] RTMP buffer size?
In-Reply-To: <1310062576455-6559405.post@n2.nabble.com>
References: <1309955603001-6554200.post@n2.nabble.com>
	<1310062576455-6559405.post@n2.nabble.com>
Message-ID: <CAKbxfG_xJPFkuntcGiqArq+vMJsO7mfYP53RyvxRgDuy2SWPbg@mail.gmail.com>

we have to come up with a way to have the server send pings to the
client and when the round trip is too large send another message to
the client to make it restart the audio layer by closing and
re-opening the port.
there is not much as far as options with this because they do not
expose much on the client side.


On Thu, Jul 7, 2011 at 1:16 PM, peely <freeswitch at peely.com> wrote:
> Doing some more analysis, the noticable latency only seems to be on the
> receive side of the Flash control i.e. through the speakers of the PC using
> the Flash control. From the Flash control out to the PSTN the delay seems
> reasonably low.
>
> I can't compile the flash control from the source provided as I don't have
> Flex. The only thing noticable I can see if the incomingNetStream.bufferTime
> which seems to be 0.2 seconds in freeswitch.mxml.
>
> Is there any chance of having this configuable?
>
>
> Thanks,
>
>
> Neil.
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/RTMP-buffer-size-tp6554200p6559405.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> 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
>



-- 
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 vetali100 at gmail.com  Fri Jul  8 01:24:46 2011
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Fri, 8 Jul 2011 00:24:46 +0300
Subject: [Freeswitch-users] Freeswitch dynamic routing of all calls
In-Reply-To: <4E1616BB.6000405@gmail.com>
References: <BANLkTimuCjRcVccyrPpgQeY6N+UamyNQ=A@mail.gmail.com>
	<518E6CA8-4198-4484-BF03-2DEFFA6493EA@gmail.com>
	<BANLkTi=UJ0+GKwd3T2uom=MuS4aLqudDmA@mail.gmail.com>
	<4E1616BB.6000405@gmail.com>
Message-ID: <CACc3D0XOUkM0Qt30ruLJ11goqmL2H46MOD90Y4TdWUcgMvUz=g@mail.gmail.com>

Hi,

It is easy to achieve using powerful Lua scripting and Core ODBC (
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core)

If you did not have any experience with Lua, just copy paste script below,
and fix syntax errors if any :)
(Or you can achieve similar using perl, javascript, etc...)

Steps:

1. Add the following block inside your default dialplan (for example you
want to handle international calls, prefix "00" or "+")
-----------------------------------------------------------------------------
<extension name="international_with_some_routing">
      <condition field="destination_number" expression="^(00|\+)(\d+)">

          <action application="lua" data="call.lua $2"/>
      </condition>
</extension>
-----------------------------------------------------------------------------


2. Create "call.lua" with the following lines and put it into "scripts"
folder under freeswitch main folder:
-----------------------------------------------------------------------------
--get db handler
local dbh = assert(freeswitch.Dbh("my_routing_db","my_user","my_pass"));

subscriber_number = session:getVariable("accountcode"); --or
session:getVariable("caller_id_number"); if you wish
called_number = argv[1];

--let's check about money
balance = 0;
dbh:query("select balance from subscriber where
subscriber_number='"..subscriber_number.."'",
function(row)
    balance = tonumber(row.balance);
end);

if (balance <= 0) then
    freeswitch.consoleLog("WARNING", "Call denied for
"..subscriber_number.." to "..called_number.." Balance: "..balance.."\n");
    return;
end

--select price plan and gateway (you can implement more complex logic here,
to select the plan with lowest cost, etc)
--assuming you use mysql
dbh:query("select price_buy, price_sell, provider from dialplan where
instr('"..called_number.."',prefix)=1 order by prefix desc limit 1",
function(row)
    price_buy=tonumber(row.price_buy);
    price_sell=tonumber(row.price_sell);
    called_gateway=row.provider;
end);

--add some checks if needed
call_duration_min = balance / price_sell;
call_duration_sec = call_duration_min * 60;

--release db handler
dbh:release();

--prepare outgoing session
called_parameters =
"{ignore_early_media=true,originate_timeout=90,price_sell="..price_sell..",price_buy="..price_buy..",subscriber_number="..subscriber_number..",hangup_after_bridge=true}";
called_string =
called_parameters.."sofia/gateway/"..called_gateway.."/"..called_number;

session:setVariable("ringback", "%(2000,4000,440,480)");
called_session = freeswitch.Session(called_string, session);

--sometimes it is needed :)
session:sleep(200);

if (called_session:ready()) then
    -- this will disconnect the call after allowed seconds
    session:execute("sched_hangup","+"..call_duration_sec);
    --finally, bridge
    freeswitch.bridge(session, called_session);
end

--that's it
-----------------------------------------------------------------------------

Make sure that you have valid registered gateway as returned by
"called_gateway" in your conf/sip_profiles.

Any questions, I will be glad to assist.

Regards,
Vitalie




2011/7/7 Nazim Aghabayov <nazim.aghabayov at gmail.com>

> Hello Mateusz,
>
> I'm finding  ESL powerful yet complicated and using mod_xml_curl for
> dynamic routing.
> mod_xml_curl with  with bindings = "dialplan" handles routing just fine.
> http://wiki.freeswitch.org/wiki/Mod_xml_curl
>
> Another approach is to a the Lua scripts for setting the variables
> "inside" the dialplan.
> FreeSWITH is a flexible system, so there are a lot of ways to achieve
> the same result.
>
> Best Regards,
> Nazim
>
>
> On 07/01/2011 05:55 PM, Mateusz Bartczak wrote:
> > OK but how can I respond for received events?
> >
> > I subscribe to receive events using following event socket command:
> >
> > event plain CHANNEL_CREATE
> >
> > then I got all events of that type and that's great but the question is
> how
> > can I handle those events, what's the syntax and where to put it?
> >
> > I will simply explain what I'm trying to achieve:
> >
> > 1. User dialed number
> > 2. CHANNEL_CREATE event is created
> > 3. I got this event using socket
> > 4. What to do now? How to respond for that event? For example I would
> like
> > to respond with dialstring to use, user called number 123, I would like
> to
> > return something like sofia/gateway123/00123
> >
> >
> >
> > 2011/6/30 Steven Ayre<steveayre at gmail.com>
> >
> >> Yes it's possible and I suggest you look at esl
> >>
> >> Steve on iPhone
> >>
> >> On 30 Jun 2011, at 09:54, Mateusz Bartczak<netcentrica at gmail.com>
>  wrote:
> >>
> >>> Hi all
> >>>
> >>> I'm new to FS and I would like to know is it possible to implement
> >> following scenario:
> >>> 1. User dials number
> >>> 2. Routing script detects outgoing call event. Every call needs to be
> >> handled by routing script.
> >>> 3. Routing script takes in input: user name, domain, dialed number.
> Than
> >> it query database to find best SIP trunk to route the call, it also
> checks
> >> destination price per minute and calculates maximum call time for
> prepaid
> >> user.
> >>> 4. Routing script output is: SIP trunk to use, SIP call parameters (ie.
> >> callerid), maximum call duration
> >>> 5. FS read output from routing script and make call using returned
> >> parameters
> >>> Preferred routing implementation technology: background running unix
> >> deamon written in Java or PHP. Connection with FS via socket.
> >>> Event routing script will be multi-threaded, must be able to deal with
> a
> >> lot of calls in parallel and processing of one call should not block
> >> processing of other calls (I have this problem with Yate voip server,
> and
> >> that's really big problem)
> >>> Is it possible to do this using FS?
> >>> Any advices where to search for additional info? I know that there is
> >> event handler but can it return "dialstring" for outgoing call events?
> >>> Some code examples?
> >>>
> >>> I will really appreciate your help
> >>>
>
>
> _______________________________________________
> 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 jan.berger at video24.no  Fri Jul  8 01:31:25 2011
From: jan.berger at video24.no (Jan Berger)
Date: Thu, 7 Jul 2011 23:31:25 +0200
Subject: [Freeswitch-users] Playing multiple files simultaneously
In-Reply-To: <95BE2DAB-4EDC-4DE3-AE6F-A9119CF07992@bryansmart.com>
References: <2DB72AF2-2394-419A-ADC5-0B76FE594662@bryansmart.com>
	<4E1507B2.3090803@coppice.org>
	<95BE2DAB-4EDC-4DE3-AE6F-A9119CF07992@bryansmart.com>
Message-ID: <516272DB8645493D999830F86B737B0A@dell9400>

Get on to the dev list and make suggestions - maybe make shorter mails :) -
my own experience is that if I post long mails no-one answers, but if I
spilt them up in smaller I get plenty with response.

/Jan

I feel cautious about poking my nose in to programming issues here. I've
been programming to one degree or another for 25 years, and even had it as
my main gig for several years, but all of my experience has either been
personal projects, or closed source commercial development for an employer.
I don't have my head around people and technical processes for contributing
to an open source project, so even tweaking a single line in a source file
seems to involve a lot of process in order to submit it. I know that the
group uses git for version control, but, if I make a tweak, do I submit it
to some sort of approval branch first? If their is a processes doc for the
group, I'd be glad to read it.

Thanks
Bryan




From jan.berger at video24.no  Fri Jul  8 02:09:32 2011
From: jan.berger at video24.no (Jan Berger)
Date: Fri, 8 Jul 2011 00:09:32 +0200
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
	Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <CAE=3qOZaqoa1zt1_q_gaYzME=DOJYqhW4i7afhrBYc4PxViq5g@mail.gmail.com>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
	<4DF10C83.6020803@earthspike.net>
	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>
	<4DF1F07A.5000200@earthspike.net>
	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>
	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>
	<4DFB9831.4040709@earthspike.net>
	<BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>
	<CAP1_MOCq5qRjsgd7XGag9ukjHiG2Se-pXvrghQ7hbB5ciwJK+A@mail.gmail.com>
	<4E156252.2030308@earthspike.net>
	<CAP1_MOBAFpYqJNKhgWzv3aYJeKbj+RJDQJCaQByVcrMVp_s8RA@mail.gmail.com>
	<CAE=3qOZaqoa1zt1_q_gaYzME=DOJYqhW4i7afhrBYc4PxViq5g@mail.gmail.com>
Message-ID: <C06E47096A4E434AB7466C6309AC9FE5@dell9400>

I must admit I don't remember what CLIP and CLOP is - and my Q.931
experience is getting a bit rusty.

Can you get a snoop of L3 out and in so I can see what you send and what the
switch responds back?

I prefer to use Called and Calling - The numbers contain a few bits that
tell what number this is. The most common are unknown, national and
international. These bits must match the actual number you send. The 10
digit number is a unknown, the 11 digit is a national. International start
with 00 nn

On top of that you will face that modern switches are quite capable and able
to configure whatever behaviour they want per line with regards to
called/calling - so you need to find out exactly what the switch expect on
both called and caller for outgoing to behave as you want. 

/Jan

>>
>> On our PRI systems we are sending 10 digits (2071231234), and on the
>> BRI system we are sending 11 digits (02071231234).
>>



 

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From freeswitch at earthspike.net  Fri Jul  8 02:32:49 2011
From: freeswitch at earthspike.net (John)
Date: Thu, 07 Jul 2011 23:32:49 +0100
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM
 -	Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <C06E47096A4E434AB7466C6309AC9FE5@dell9400>
References: <4DED4854.6010203@earthspike.net>	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>	<4DF10C83.6020803@earthspike.net>	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>	<4DF1F07A.5000200@earthspike.net>	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>	<4DFB9831.4040709@earthspike.net>	<BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>	<CAP1_MOCq5qRjsgd7XGag9ukjHiG2Se-pXvrghQ7hbB5ciwJK+A@mail.gmail.com>	<4E156252.2030308@earthspike.net>	<CAP1_MOBAFpYqJNKhgWzv3aYJeKbj+RJDQJCaQByVcrMVp_s8RA@mail.gmail.com>	<CAE=3qOZaqoa1zt1_q_gaYzME=DOJYqhW4i7afhrBYc4PxViq5g@mail.gmail.com>
	<C06E47096A4E434AB7466C6309AC9FE5@dell9400>
Message-ID: <4E163411.50507@earthspike.net>

Jan,

I have tried every combination of unknown, national and international 
with every combination of number length.  I have also confirmed that by 
using wanpipemon to capture the D-channel messages.  When I went through 
them with Wireshark and compared them with Q.931, everything was being 
set correctly by the Sangoma card/drivers.  I am at the point now where 
I need someone who has worked with BT ISDN switches to tell me what they 
can accept.  It seems that this is a national secret over here, or 
instead that they can accept almost anything and my lines have been 
configured to override the outgoing calling number with the base 
number.  Either way, the only way in which I think I am now going to get 
a result is by booking a fault.  Your helpful comment about modern 
switches being able to work this out also confirms that I am probably 
not looking at a Q.931 formatting error on my part, but on a blockage in 
the exchange.  I don't have a problem with the called number, for example.

John

On 07/07/11 23:09, Jan Berger wrote:
>
> I must admit I don't remember what CLIP and CLOP is -- and my Q.931 
> experience is getting a bit rusty...
>
> Can you get a snoop of L3 out and in so I can see what you send and 
> what the switch responds back?
>
> I prefer to use Called and Calling -- The numbers contain a few bits 
> that tell what number this is. The most common are unknown, national 
> and international. These bits must match the actual number you send. 
> The 10 digit number is a unknown, the 11 digit is a national. 
> International start with 00 nn
>
> On top of that you will face that modern switches are quite capable 
> and able to configure whatever behaviour they want per line with 
> regards to called/calling -- so you need to find out exactly what the 
> switch expect on both called and caller for outgoing to behave as you 
> want.
>
> /Jan
>
> >>
> >> On our PRI systems we are sending 10 digits (2071231234), and on the
> >> BRI system we are sending 11 digits (02071231234).
> >>
>
>
> _______________________________________________
> 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 jan.berger at video24.no  Fri Jul  8 02:56:57 2011
From: jan.berger at video24.no (Jan Berger)
Date: Fri, 8 Jul 2011 00:56:57 +0200
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
	Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <4E163411.50507@earthspike.net>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
	<4DF10C83.6020803@earthspike.net>
	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>
	<4DF1F07A.5000200@earthspike.net>
	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>
	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>
	<4DFB9831.4040709@earthspike.net>
	<BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>
	<CAP1_MOCq5qRjsgd7XGag9ukjHiG2Se-pXvrghQ7hbB5ciwJK+A@mail.gmail.com>
	<4E156252.2030308@earthspike.net>
	<CAP1_MOBAFpYqJNKhgWzv3aYJeKbj+RJDQJCaQByVcrMVp_s8RA@mail.gmail.com>
	<CAE=3qOZaqoa1zt1_q_gaYzME=DOJYqhW4i7afhrBYc4PxViq5g@mail.gmail.com>
	<C06E47096A4E434AB7466C6309AC9FE5@dell9400>
	<4E163411.50507@earthspike.net>
Message-ID: <8188E54D4856433D88DE13F1DE95224C@dell9400>

I assume CLIP and COLP are Q.932, or more exactly Q.951 -and not some BT
scheme using Q.931 only?

 

I need to dig into the spec a little on this one .

 

/Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of John
Sent: 8. juli 2011 00:33
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
Sangoma B700 - ISDN connection questions - UK)

 

Jan,

I have tried every combination of unknown, national and international with
every combination of number length.  I have also confirmed that by using
wanpipemon to capture the D-channel messages.  When I went through them with
Wireshark and compared them with Q.931, everything was being set correctly
by the Sangoma card/drivers.  I am at the point now where I need someone who
has worked with BT ISDN switches to tell me what they can accept.  It seems
that this is a national secret over here, or instead that they can accept
almost anything and my lines have been configured to override the outgoing
calling number with the base number.  Either way, the only way in which I
think I am now going to get a result is by booking a fault.  Your helpful
comment about modern switches being able to work this out also confirms that
I am probably not looking at a Q.931 formatting error on my part, but on a
blockage in the exchange.  I don't have a problem with the called number,
for example.

John

On 07/07/11 23:09, Jan Berger wrote: 

I must admit I don't remember what CLIP and CLOP is - and my Q.931
experience is getting a bit rusty.

Can you get a snoop of L3 out and in so I can see what you send and what the
switch responds back?

I prefer to use Called and Calling - The numbers contain a few bits that
tell what number this is. The most common are unknown, national and
international. These bits must match the actual number you send. The 10
digit number is a unknown, the 11 digit is a national. International start
with 00 nn

On top of that you will face that modern switches are quite capable and able
to configure whatever behaviour they want per line with regards to
called/calling - so you need to find out exactly what the switch expect on
both called and caller for outgoing to behave as you want. 

/Jan

>>
>> On our PRI systems we are sending 10 digits (2071231234), and on the
>> BRI system we are sending 11 digits (02071231234).
>>




 

 
 
_______________________________________________
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 wstephen80 at gmail.com  Fri Jul  8 03:00:37 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Fri, 8 Jul 2011 01:00:37 +0200
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
 Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <4E163411.50507@earthspike.net>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
	<4DF10C83.6020803@earthspike.net>
	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>
	<4DF1F07A.5000200@earthspike.net>
	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>
	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>
	<4DFB9831.4040709@earthspike.net>
	<BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>
	<CAP1_MOCq5qRjsgd7XGag9ukjHiG2Se-pXvrghQ7hbB5ciwJK+A@mail.gmail.com>
	<4E156252.2030308@earthspike.net>
	<CAP1_MOBAFpYqJNKhgWzv3aYJeKbj+RJDQJCaQByVcrMVp_s8RA@mail.gmail.com>
	<CAE=3qOZaqoa1zt1_q_gaYzME=DOJYqhW4i7afhrBYc4PxViq5g@mail.gmail.com>
	<C06E47096A4E434AB7466C6309AC9FE5@dell9400>
	<4E163411.50507@earthspike.net>
Message-ID: <CAO4LeU17OczaeyUNzdJGWPvu=RFRz6zcDWBaPhJe36sEXvz+Zw@mail.gmail.com>

With some providers I have to set the "screening indicator" of calling party
number to "network provided" ... you can add this try...

(freetdm_screening_ind = network-provided)

Stephen

On Fri, Jul 8, 2011 at 12:32 AM, John <freeswitch at earthspike.net> wrote:

> **
> Jan,
>
> I have tried every combination of unknown, national and international with
> every combination of number length.  I have also confirmed that by using
> wanpipemon to capture the D-channel messages.  When I went through them with
> Wireshark and compared them with Q.931, everything was being set correctly
> by the Sangoma card/drivers.  I am at the point now where I need someone who
> has worked with BT ISDN switches to tell me what they can accept.  It seems
> that this is a national secret over here, or instead that they can accept
> almost anything and my lines have been configured to override the outgoing
> calling number with the base number.  Either way, the only way in which I
> think I am now going to get a result is by booking a fault.  Your helpful
> comment about modern switches being able to work this out also confirms that
> I am probably not looking at a Q.931 formatting error on my part, but on a
> blockage in the exchange.  I don't have a problem with the called number,
> for example.
>
> John
>
>
> On 07/07/11 23:09, Jan Berger wrote:
>
>   I must admit I don?t remember what CLIP and CLOP is ? and my Q.931
> experience is getting a bit rusty?****
>
> Can you get a snoop of L3 out and in so I can see what you send and what
> the switch responds back?****
>
> I prefer to use Called and Calling ? The numbers contain a few bits that
> tell what number this is. The most common are unknown, national and
> international. These bits must match the actual number you send. The 10
> digit number is a unknown, the 11 digit is a national. International start
> with 00 nn****
>
> On top of that you will face that modern switches are quite capable and
> able to configure whatever behaviour they want per line with regards to
> called/calling ? so you need to find out exactly what the switch expect on
> both called and caller for outgoing to behave as you want. ****
>
> /Jan****
>
> >>
> >> On our PRI systems we are sending 10 digits (2071231234), and on the
> >> BRI system we are sending 11 digits (02071231234).
> >>
>
> ****
>
> ** **
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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
>
>
>
> _______________________________________________
> 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 jan.berger at video24.no  Fri Jul  8 03:37:30 2011
From: jan.berger at video24.no (Jan Berger)
Date: Fri, 8 Jul 2011 01:37:30 +0200
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
	Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <CAO4LeU17OczaeyUNzdJGWPvu=RFRz6zcDWBaPhJe36sEXvz+Zw@mail.gmail.com>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
	<4DF10C83.6020803@earthspike.net>
	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>
	<4DF1F07A.5000200@earthspike.net>
	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>
	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>
	<4DFB9831.4040709@earthspike.net>
	<BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>
	<CAP1_MOCq5qRjsgd7XGag9ukjHiG2Se-pXvrghQ7hbB5ciwJK+A@mail.gmail.com>
	<4E156252.2030308@earthspike.net>
	<CAP1_MOBAFpYqJNKhgWzv3aYJeKbj+RJDQJCaQByVcrMVp_s8RA@mail.gmail.com>
	<CAE=3qOZaqoa1zt1_q_gaYzME=DOJYqhW4i7afhrBYc4PxViq5g@mail.gmail.com>
	<C06E47096A4E434AB7466C6309AC9FE5@dell9400>
	<4E163411.50507@earthspike.net>
	<CAO4LeU17OczaeyUNzdJGWPvu=RFRz6zcDWBaPhJe36sEXvz+Zw@mail.gmail.com>
Message-ID: <50DDFDB0971A4EE7AB29FA97A8247E8E@dell9400>

You should read Q.951 and make sure SETUP is correct it's a few words, but
it's very straight forward - if your convinced it is correct are call BT.

 

Q.951 defines how to set up the numbers, so unless BT have given you a
specific procedure you go by the spec and claim a fault. Send me the
wireshark trace if you want me to look at it - but this should be rather
straight forward.

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stephen
Wilde
Sent: 8. juli 2011 01:01
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
Sangoma B700 - ISDN connection questions - UK)

 

With some providers I have to set the "screening indicator" of calling party
number to "network provided" ... you can add this try...

 

(freetdm_screening_ind = network-provided)

 

Stephen

 

On Fri, Jul 8, 2011 at 12:32 AM, John <freeswitch at earthspike.net> wrote:

Jan,

I have tried every combination of unknown, national and international with
every combination of number length.  I have also confirmed that by using
wanpipemon to capture the D-channel messages.  When I went through them with
Wireshark and compared them with Q.931, everything was being set correctly
by the Sangoma card/drivers.  I am at the point now where I need someone who
has worked with BT ISDN switches to tell me what they can accept.  It seems
that this is a national secret over here, or instead that they can accept
almost anything and my lines have been configured to override the outgoing
calling number with the base number.  Either way, the only way in which I
think I am now going to get a result is by booking a fault.  Your helpful
comment about modern switches being able to work this out also confirms that
I am probably not looking at a Q.931 formatting error on my part, but on a
blockage in the exchange.  I don't have a problem with the called number,
for example.

John



On 07/07/11 23:09, Jan Berger wrote: 

I must admit I don't remember what CLIP and CLOP is - and my Q.931
experience is getting a bit rusty.

Can you get a snoop of L3 out and in so I can see what you send and what the
switch responds back?

I prefer to use Called and Calling - The numbers contain a few bits that
tell what number this is. The most common are unknown, national and
international. These bits must match the actual number you send. The 10
digit number is a unknown, the 11 digit is a national. International start
with 00 nn

On top of that you will face that modern switches are quite capable and able
to configure whatever behaviour they want per line with regards to
called/calling - so you need to find out exactly what the switch expect on
both called and caller for outgoing to behave as you want. 

/Jan

>>
>> On our PRI systems we are sending 10 digits (2071231234), and on the
>> BRI system we are sending 11 digits (02071231234).
>>

 

 
_______________________________________________
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

 


_______________________________________________
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 lloydie.t at gmail.com  Fri Jul  8 05:22:26 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Fri, 8 Jul 2011 02:22:26 +0100
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
 Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <50DDFDB0971A4EE7AB29FA97A8247E8E@dell9400>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
	<4DF10C83.6020803@earthspike.net>
	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>
	<4DF1F07A.5000200@earthspike.net>
	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>
	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>
	<4DFB9831.4040709@earthspike.net>
	<BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>
	<CAP1_MOCq5qRjsgd7XGag9ukjHiG2Se-pXvrghQ7hbB5ciwJK+A@mail.gmail.com>
	<4E156252.2030308@earthspike.net>
	<CAP1_MOBAFpYqJNKhgWzv3aYJeKbj+RJDQJCaQByVcrMVp_s8RA@mail.gmail.com>
	<CAE=3qOZaqoa1zt1_q_gaYzME=DOJYqhW4i7afhrBYc4PxViq5g@mail.gmail.com>
	<C06E47096A4E434AB7466C6309AC9FE5@dell9400>
	<4E163411.50507@earthspike.net>
	<CAO4LeU17OczaeyUNzdJGWPvu=RFRz6zcDWBaPhJe36sEXvz+Zw@mail.gmail.com>
	<50DDFDB0971A4EE7AB29FA97A8247E8E@dell9400>
Message-ID: <CAE=3qOYawAJfO6QUskwUwpTy0fYkay66d6pgHQZmdfZnfQWaNg@mail.gmail.com>

John,
        I am in Oxford. A little way, but I do get over to the west now and
again.

On 8 July 2011 00:37, Jan Berger <jan.berger at video24.no> wrote:

> ** **
>
> You should read Q.951 and make sure SETUP is correct it?s a few words, but
> it?s very straight forward ? if your convinced it is correct are call BT.*
> ***
>
> ** **
>
> Q.951 defines how to set up the numbers, so unless BT have given you a
> specific procedure you go by the spec and claim a fault. Send me the
> wireshark trace if you want me to look at it ? but this should be rather
> straight forward.****
>
> ** **
>
> Jan****
>
> ** **
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Stephen
> Wilde
> *Sent:* 8. juli 2011 01:01
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
> Sangoma B700 - ISDN connection questions - ****UK****)
> ****
>
>  ** **
>
> With some providers I have to set the "screening indicator" of calling
> party number to "network provided" ... you can add this try...****
>
> ** **
>
> (freetdm_screening_ind = network-provided)****
>
> ** **
>
> Stephen****
>
> ** **
>
> On Fri, Jul 8, 2011 at 12:32 AM, John <freeswitch at earthspike.net> wrote:**
> **
>
> Jan,
>
> I have tried every combination of unknown, national and international with
> every combination of number length.  I have also confirmed that by using
> wanpipemon to capture the D-channel messages.  When I went through them with
> Wireshark and compared them with Q.931, everything was being set correctly
> by the Sangoma card/drivers.  I am at the point now where I need someone who
> has worked with BT ISDN switches to tell me what they can accept.  It seems
> that this is a national secret over here, or instead that they can accept
> almost anything and my lines have been configured to override the outgoing
> calling number with the base number.  Either way, the only way in which I
> think I am now going to get a result is by booking a fault.  Your helpful
> comment about modern switches being able to work this out also confirms that
> I am probably not looking at a Q.931 formatting error on my part, but on a
> blockage in the exchange.  I don't have a problem with the called number,
> for example.
>
> John****
>
>
>
> On 07/07/11 23:09, Jan Berger wrote: ****
>
>    I must admit I don?t remember what CLIP and CLOP is ? and my Q.931
> experience is getting a bit rusty?****
>
> Can you get a snoop of L3 out and in so I can see what you send and what
> the switch responds back?****
>
> I prefer to use Called and Calling ? The numbers contain a few bits that
> tell what number this is. The most common are unknown, national and
> international. These bits must match the actual number you send. The 10
> digit number is a unknown, the 11 digit is a national. International start
> with 00 nn****
>
> On top of that you will face that modern switches are quite capable and
> able to configure whatever behaviour they want per line with regards to
> called/calling ? so you need to find out exactly what the switch expect on
> both called and caller for outgoing to behave as you want. ****
>
> /Jan****
>
> >>
> >> On our PRI systems we are sending 10 digits (2071231234), and on the
> >> BRI system we are sending 11 digits (02071231234).
> >>****
>
>  ****
>
> ** **
>
> _______________________________________________****
>
> 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****
>
>  ** **
>
>
> _______________________________________________
> 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****
>
> ** **
>
> _______________________________________________
> 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 dave at dchorton.com  Fri Jul  8 08:40:47 2011
From: dave at dchorton.com (Dave Horton)
Date: Thu, 7 Jul 2011 21:40:47 -0700
Subject: [Freeswitch-users] From header rewrite on B2BUA scenario
Message-ID: <3F31FEDC-E580-455A-86EB-65017543AF72@dchorton.com>

I have a scenario where I have FS configured so as to take an incoming call and send it out through a configured gateway.  Now this gateway does not require authentication, so in my gateway configuration I have the register param set to false:

    <gateway name="gwInternal">
      <param name="username" value="foo"/>
      <param name="password" value="bar"/>
      <param name="realm" value="204.215.65.202"/>
      <param name="proxy" value="204.215.65.202"/>
      <param name="register" value="false"/>
      <param name="context" value="public"/>
    </gateway>

Now, even though I am not registering to this gateway, FS requires me to provide a username and password or else I get an error when starting/loading this configuration.  So you can see above that I have just put in a username of "foo", just to have something there.

Now I come to the problem.  The result of the above configuration is that the From header coming in looks like this:

 From:  <sip:+15083084809 at 204.215.65.201:5060>;tag=3519088051-756100

and the one that FS creates to send out through this gateway looks like this:

  From: "+15083084809" <sip:foo at 204.215.65.202;transport=udp>;tag=Sr6taB8rFgt1H

You can see that it has inserted the username of 'foo' in the user part of the uri of the From header.

The reason that this is a problem is that this INVITE is going out to another remote FS, as it turns out.  And when the dialed remote extension off that FS PBX does not pick up it then goes into voicemail, and the username of 'foo' is what is stored as the calling number.  So when the user on that remote PBX collects that voicemail and tries to do call return, the FS ends up trying to transfer a call to 'foo', which fails.  So call return is not working in the remote FS due to this rewriting of the From header.

Is there a way I can configure my FS to send calls out through a gateway and maintain the user part of the From URI that was received on the incoming call?  Or, alternatively, can I configure the voicemail application to look for the caller id field from somewhere else than the user part of the From URI ?

Dave

From peter.olsson at visionutveckling.se  Fri Jul  8 09:54:00 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 8 Jul 2011 07:54:00 +0200
Subject: [Freeswitch-users] From header rewrite on B2BUA scenario
Message-ID: <42B35970-204B-40BF-A09B-748DE27AB657@visionutveckling.se>

Make sure to set caller-id-in-from to true in the gateway config.

/Peter

----- Reply message -----
Fr?n: "Dave Horton" <dave at dchorton.com>
Datum: fre, jul 8, 2011 06:53
Rubrik: [Freeswitch-users] From header rewrite on B2BUA scenario
Till: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

I have a scenario where I have FS configured so as to take an incoming call and send it out through a configured gateway.  Now this gateway does not require authentication, so in my gateway configuration I have the register param set to false:

    <gateway name="gwInternal">
      <param name="username" value="foo"/>
      <param name="password" value="bar"/>
      <param name="realm" value="204.215.65.202"/>
      <param name="proxy" value="204.215.65.202"/>
      <param name="register" value="false"/>
      <param name="context" value="public"/>
    </gateway>

Now, even though I am not registering to this gateway, FS requires me to provide a username and password or else I get an error when starting/loading this configuration.  So you can see above that I have just put in a username of "foo", just to have something there.

Now I come to the problem.  The result of the above configuration is that the From header coming in looks like this:

 From:  <sip:+15083084809 at 204.215.65.201:5060>;tag=3519088051-756100

and the one that FS creates to send out through this gateway looks like this:

  From: "+15083084809" <sip:foo at 204.215.65.202;transport=udp>;tag=Sr6taB8rFgt1H

You can see that it has inserted the username of 'foo' in the user part of the uri of the From header.

The reason that this is a problem is that this INVITE is going out to another remote FS, as it turns out.  And when the dialed remote extension off that FS PBX does not pick up it then goes into voicemail, and the username of 'foo' is what is stored as the calling number.  So when the user on that remote PBX collects that voicemail and tries to do call return, the FS ends up trying to transfer a call to 'foo', which fails.  So call return is not working in the remote FS due to this rewriting of the From header.

Is there a way I can configure my FS to send calls out through a gateway and maintain the user part of the From URI that was received on the incoming call?  Or, alternatively, can I configure the voicemail application to look for the caller id field from somewhere else than the user part of the From URI ?

Dave
_______________________________________________
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
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http://www.freeswitch.org

!DSPAM:4e168bec32761053165832!



From gcd at i.ph  Fri Jul  8 10:20:57 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Fri, 8 Jul 2011 14:20:57 +0800
Subject: [Freeswitch-users] Transcoder cards & g.729 licenses
Message-ID: <CAHKqMb6WuJ6pMKwLoRoTOq5g=O91kvv9xSsZQqGxJgYeBkEnFQ@mail.gmail.com>

hello guys,

if i use a transcoder card e.g. sangoma D150 for TDM-to-VoIP setup, do i
still need to purchase 30  G.729 licenses for every E1 trunk? if yes, will
the licenses be bound to the transcoder card?  tks.

-nandy
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From krice at freeswitch.org  Fri Jul  8 11:12:56 2011
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 08 Jul 2011 02:12:56 -0500
Subject: [Freeswitch-users] Transcoder cards & g.729 licenses
In-Reply-To: <CAHKqMb6WuJ6pMKwLoRoTOq5g=O91kvv9xSsZQqGxJgYeBkEnFQ@mail.gmail.com>
Message-ID: <CA3C1828.18B1B%krice@freeswitch.org>

Sangoma Hardware transcoders include the required licensing... See
http://sangoma.com/products/hardware_products/transcoding/d150.html


On 7/8/11 1:20 AM, "Nandy Dagondon" <gcd at i.ph> wrote:

> hello guys,
> 
> if i use a transcoder card e.g. sangoma D150 for TDM-to-VoIP setup, do i still
> need to purchase 30? G.729 licenses for every E1 trunk? if yes, will the
> licenses be bound to the transcoder card? ?tks.
> 
> -nandy
> 
> 
> _______________________________________________
> 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 avi at avimarcus.net  Fri Jul  8 15:18:10 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 8 Jul 2011 14:18:10 +0300
Subject: [Freeswitch-users] Lua to use mod_lcr's "as xml"
Message-ID: <CALNZuw=ubgY1YLf_YHq8boemHOgBDOv0XAK7cFKqs2qTtKcgig@mail.gmail.com>

How can I run mod_lcr's "as xml" and then use lua to deal with the result as
an array or table?

Thanks,
Avi
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From lautram.mathieu at gmail.com  Fri Jul  8 15:29:12 2011
From: lautram.mathieu at gmail.com (Mathieu Lautram)
Date: Fri, 8 Jul 2011 13:29:12 +0200
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAKbxfG-h2_seFcYhu-5zwDbhD-z12xGuw7GmmKPPgSLGux3_zg@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
	<CAKbxfG-h2_seFcYhu-5zwDbhD-z12xGuw7GmmKPPgSLGux3_zg@mail.gmail.com>
Message-ID: <CAMNz+bsev6pqGc4u8YxhPvsFtzcpBC8R0W6frW52BtaGbJidJQ@mail.gmail.com>

bridge_early_media is not really appropriate for my issue and I have already
tried this parameter.
In fact, I would like to let A leg hear a ringtone(and sometimes vocal
informations) when the phone of the B leg is *effectively* ringing.
Do you think it is possible? Maybe ignoring early media is not the solution
to my problem?

Thanks

Best regards,

-- 
Mathieu LAUTRAM
Application developer

BJT Partners - FRANCE
+33 1 79 75 99 60
+33 6 61 59 07 25
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From erik.dekkers at wvds.nl  Fri Jul  8 16:00:19 2011
From: erik.dekkers at wvds.nl (Erik Dekkers)
Date: Fri, 8 Jul 2011 14:00:19 +0200
Subject: [Freeswitch-users] New BSD timer module
In-Reply-To: <CAFiqYumUFpv0SjGJmdoZf2rAy2LGiFz7HndUuiRTRS-bGbpErQ@mail.gmail.com>
References: <CAFiqYumUFpv0SjGJmdoZf2rAy2LGiFz7HndUuiRTRS-bGbpErQ@mail.gmail.com>
Message-ID: <CDF5E00590C9BA4AB96BE7CDD28C86AE0797AEC9AB@FS1.wvds.local>

Hi Steve,

Im using BSD as operating system for Freeswitch. Can you give a description why I would like to use kqueue cause I'm not having troubles with Freeswitch whatsoever.
Btw, I'm willing to test :)

Regards,

Erik (wvds-nl)

Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Steven Ayre
Verzonden: donderdag 7 juli 2011 15:33
Aan: FreeSWITCH Users Help
Onderwerp: [Freeswitch-users] New BSD timer module

Hi everyone,

I've submitted a Jira with a proposed new mod_kqueue timer module. This is a BSD equivalent of mod_timerfd.
http://jira.freeswitch.org/browse/FS-3398

I'd appreciate it if anyone that uses FreeSWITCH on BSD that feels like it would give it a quick test to see how well it works.

To install it:
- Create a new directory src/mod/timers/mod_kqueue
- Place the mod_kqueue.c file from the jira in that directory
- Add timers/mod_kqueue to modules.conf
- Build and install
- Load it from modules.conf.xml or fs_cli.

Regards,
Steven Ayre
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From freeswitch at peely.com  Fri Jul  8 18:33:06 2011
From: freeswitch at peely.com (peely)
Date: Fri, 8 Jul 2011 07:33:06 -0700 (PDT)
Subject: [Freeswitch-users] RTMP buffer size?
In-Reply-To: <CAKbxfG_xJPFkuntcGiqArq+vMJsO7mfYP53RyvxRgDuy2SWPbg@mail.gmail.com>
References: <1309955603001-6554200.post@n2.nabble.com>
	<1310062576455-6559405.post@n2.nabble.com>
	<CAKbxfG_xJPFkuntcGiqArq+vMJsO7mfYP53RyvxRgDuy2SWPbg@mail.gmail.com>
Message-ID: <1310135586267-6562695.post@n2.nabble.com>

Thanks for your response.

In the end I dug around and found that Flex has an open source compiler.

I had a poke around the NetStreamInfo object which you can create from the
NetStream, it does have some useful data and showed up to 300ms of buffered
data. Seems that up to twice the specified NetStream.bufferTime can be built
up according to Adobe documentation.

Whilst I managed to get the latency down a decent amount by capping the
buffer with a  NetStream.bufferTimeMax, it appears that the lion's share of
the latency was flash onwards in my browser!

I solved that by executing "export PULSE_LATENCY_MSEC=20" prior to
/usr/bin/chromium-browser, this has a considerable effect, certainly on
Ubuntu which uses PulseAudio.

N.




--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/RTMP-buffer-size-tp6554200p6562695.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Fri Jul  8 18:43:15 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Jul 2011 09:43:15 -0500
Subject: [Freeswitch-users] RTMP buffer size?
In-Reply-To: <1310135586267-6562695.post@n2.nabble.com>
References: <1309955603001-6554200.post@n2.nabble.com>
	<1310062576455-6559405.post@n2.nabble.com>
	<CAKbxfG_xJPFkuntcGiqArq+vMJsO7mfYP53RyvxRgDuy2SWPbg@mail.gmail.com>
	<1310135586267-6562695.post@n2.nabble.com>
Message-ID: <CAKbxfG8Hb6rFd5XUpnaorRtHsD2xXMK0aVZmxvj_8kZw2cG1jA@mail.gmail.com>

do you have a patch for the change you made?

On Fri, Jul 8, 2011 at 9:33 AM, peely <freeswitch at peely.com> wrote:
> Thanks for your response.
>
> In the end I dug around and found that Flex has an open source compiler.
>
> I had a poke around the NetStreamInfo object which you can create from the
> NetStream, it does have some useful data and showed up to 300ms of buffered
> data. Seems that up to twice the specified NetStream.bufferTime can be built
> up according to Adobe documentation.
>
> Whilst I managed to get the latency down a decent amount by capping the
> buffer with a ?NetStream.bufferTimeMax, it appears that the lion's share of
> the latency was flash onwards in my browser!
>
> I solved that by executing "export PULSE_LATENCY_MSEC=20" prior to
> /usr/bin/chromium-browser, this has a considerable effect, certainly on
> Ubuntu which uses PulseAudio.
>
> N.
>
>
>
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/RTMP-buffer-size-tp6554200p6562695.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> 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
>



-- 
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 freeswitch at peely.com  Fri Jul  8 18:44:52 2011
From: freeswitch at peely.com (peely)
Date: Fri, 8 Jul 2011 07:44:52 -0700 (PDT)
Subject: [Freeswitch-users] Flex RTMP Client Added To Git Tree!
In-Reply-To: <CAKzWOxXgDVPWdQH90y9OQz0kJP8UMT_MeKfhWWki78cui-WHrA@mail.gmail.com>
References: <CAKzWOxXgDVPWdQH90y9OQz0kJP8UMT_MeKfhWWki78cui-WHrA@mail.gmail.com>
Message-ID: <1310136292311-6562743.post@n2.nabble.com>

You should have insisted in the headers that nobody be allowed to compile it
without the original ringing sound.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Flex-RTMP-Client-Added-To-Git-Tree-tp6555307p6562743.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From freeswitch at peely.com  Fri Jul  8 18:51:58 2011
From: freeswitch at peely.com (peely)
Date: Fri, 8 Jul 2011 07:51:58 -0700 (PDT)
Subject: [Freeswitch-users] RTMP buffer size?
In-Reply-To: <CAKbxfG8Hb6rFd5XUpnaorRtHsD2xXMK0aVZmxvj_8kZw2cG1jA@mail.gmail.com>
References: <1309955603001-6554200.post@n2.nabble.com>
	<1310062576455-6559405.post@n2.nabble.com>
	<CAKbxfG_xJPFkuntcGiqArq+vMJsO7mfYP53RyvxRgDuy2SWPbg@mail.gmail.com>
	<1310135586267-6562695.post@n2.nabble.com>
	<CAKbxfG8Hb6rFd5XUpnaorRtHsD2xXMK0aVZmxvj_8kZw2cG1jA@mail.gmail.com>
Message-ID: <1310136718924-6562769.post@n2.nabble.com>

Sure, I'll generate one when I get back home.

You want it throwing on pastebin?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/RTMP-buffer-size-tp6554200p6562769.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Fri Jul  8 18:57:00 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Jul 2011 09:57:00 -0500
Subject: [Freeswitch-users] RTMP buffer size?
In-Reply-To: <1310136718924-6562769.post@n2.nabble.com>
References: <1309955603001-6554200.post@n2.nabble.com>
	<1310062576455-6559405.post@n2.nabble.com>
	<CAKbxfG_xJPFkuntcGiqArq+vMJsO7mfYP53RyvxRgDuy2SWPbg@mail.gmail.com>
	<1310135586267-6562695.post@n2.nabble.com>
	<CAKbxfG8Hb6rFd5XUpnaorRtHsD2xXMK0aVZmxvj_8kZw2cG1jA@mail.gmail.com>
	<1310136718924-6562769.post@n2.nabble.com>
Message-ID: <CAKbxfG9+ASonV-RUL-QQRHhQE=aqZXqKe6AgqUBEyadQLo7Jiw@mail.gmail.com>

you could attach it to here

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


On Fri, Jul 8, 2011 at 9:51 AM, peely <freeswitch at peely.com> wrote:
> Sure, I'll generate one when I get back home.
>
> You want it throwing on pastebin?
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/RTMP-buffer-size-tp6554200p6562769.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> 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
>



-- 
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  Fri Jul  8 19:05:32 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 8 Jul 2011 18:05:32 +0300
Subject: [Freeswitch-users] Flex RTMP Client Added To Git Tree!
In-Reply-To: <1310136292311-6562743.post@n2.nabble.com>
References: <CAKzWOxXgDVPWdQH90y9OQz0kJP8UMT_MeKfhWWki78cui-WHrA@mail.gmail.com>
	<1310136292311-6562743.post@n2.nabble.com>
Message-ID: <CALNZuwmsBHpxuByfFo4Y8S8BpDfksMEOBEUy2M+rX1xbHo7P-Q@mail.gmail.com>

That audio is hilarious!
Hmm. I wonder how much flack I'll get for setting that as the ringback on
April fools day.
-Avi

On Fri, Jul 8, 2011 at 5:44 PM, peely <freeswitch at peely.com> wrote:

> You should have insisted in the headers that nobody be allowed to compile
> it
> without the original ringing sound.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Flex-RTMP-Client-Added-To-Git-Tree-tp6555307p6562743.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> 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 msc at freeswitch.org  Fri Jul  8 19:11:48 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 8 Jul 2011 08:11:48 -0700
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAMNz+bsev6pqGc4u8YxhPvsFtzcpBC8R0W6frW52BtaGbJidJQ@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
	<CAKbxfG-h2_seFcYhu-5zwDbhD-z12xGuw7GmmKPPgSLGux3_zg@mail.gmail.com>
	<CAMNz+bsev6pqGc4u8YxhPvsFtzcpBC8R0W6frW52BtaGbJidJQ@mail.gmail.com>
Message-ID: <CAKzWOxWiLpP6j=JXQE=oM+t6QF_fy0e30DCC5EO=9qGf+OFmgg@mail.gmail.com>

On Fri, Jul 8, 2011 at 4:29 AM, Mathieu Lautram
<lautram.mathieu at gmail.com>wrote:

> bridge_early_media is not really appropriate for my issue and I have
> already tried this parameter.
> In fact, I would like to let A leg hear a ringtone(and sometimes vocal
> informations) when the phone of the B leg is *effectively* ringing.
> Do you think it is possible? Maybe ignoring early media is not the solution
> to my problem?
>

Honestly, I've not yet heard a valid reason for ignoring early media at all
in your scenario. Under what conditions do you *not* want early media from
the B leg sent to the A leg?

-MC
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From msc at freeswitch.org  Fri Jul  8 19:18:22 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 8 Jul 2011 08:18:22 -0700
Subject: [Freeswitch-users] Load Balance Trunks
In-Reply-To: <COL109-W4456F57D6EF8C3739998F5F45F0@phx.gbl>
References: <col109-w22D1131E50A2495C1521C9F4590@phx.gbl>
	<C71A4485-E533-48AE-AC3F-C59D7A1E79CA@ipeva.fr>
	<COL109-W612603C4C7BCC734CBD974F45B0@phx.gbl>
	<BANLkTinT9O5+hzT1YczsmQV5ouL9EQ_2TQ@mail.gmail.com>
	<233898962-1309536985-cardhu_decombobulator_blackberry.rim.net-2106826141-@b13.c4.bise7.blackberry>
	<BANLkTikjYw=0UDyGdipBzadj0_kSoOVS2Q@mail.gmail.com>
	<1637192224-1309538673-cardhu_decombobulator_blackberry.rim.net-1258156530-@b13.c4.bise7.blackberry>
	<BANLkTineHeUHCMpKNJS7msb6xCZzBqmkQg@mail.gmail.com>
	<COL109-W4456F57D6EF8C3739998F5F45F0@phx.gbl>
Message-ID: <CAKzWOxWuC-VVgS+sbJ1HtF=FXmxoDRpb7NCrHCsV+EfVeRhvtg@mail.gmail.com>

I think this might be a bit too involved for mailing list. At this point I
would recommend consulting at freeswitch.org to see if there is a professional
who can assist you.
-MC


On Tue, Jul 5, 2011 at 8:21 AM, Abid Saleem <abid_freeswitch at live.com>wrote:

>  Hi Micheal, Avi and All,
>
> Sorry for a little late response as I was away. I have seen quite a few
> questions from you guys, so I am answering them in one email as below.
>
> 1- How does the provider notify you that each trunk has used its allotted
> time?
> Abid -> They have some counter in their IMS network to count on mins per
> trunk per day and they inform us by email.
> 2- Are these trunks inbound only?
> Abid -> No. All these are Outbound Trunks. We just use them to send
> outgoing calls to our provider.
>
> 3- What happens when a call extends more than 120 mins on a trunk, would
> the call be disconnected?
> Abid -> The call is not disconnected right away but they send us a
> notification the next day. There is no real-time disconnection.
>
> 4- And then no more calls that day on that trunk?
> Abid -> Calls do not stop connecting immediately but they keep going.
> Currently their notification process is manual not automatic blocking.
>
> Please help me if you can. Thanks.
>
> Regards
> -----------------
> Abid Saleem
> Technical Manager NGN
> Terminus Technologies
>
> ------------------------------
> Date: Fri, 1 Jul 2011 09:49:02 -0700
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
>
> Subject: Re: [Freeswitch-users] Load Balance Trunks
>
> On Fri, Jul 1, 2011 at 9:46 AM, <hmkias at gmail.com> wrote:
>
> Lcr entry could be activated on the the trunks. A cron job could monitor
> the usage and add or remove the trunk for the day.
>
>
> Are these trunks inbound only?
> -MC
>
> _______________________________________________ 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
>
> _______________________________________________
> 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
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>
>
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From jmoran at secureachsystems.com  Fri Jul  8 19:35:29 2011
From: jmoran at secureachsystems.com (Jason Moran)
Date: Fri, 8 Jul 2011 11:35:29 -0400
Subject: [Freeswitch-users] how-to catch errors in session streamFile
Message-ID: <361E98F99D3CC3439EED59BC1924ED69508307@SERVER2003.SecuReachSystems.local>

Once in a while I pass a bad file name or some other bad thing happens
with session.streamFile. If I have the console up I can see an error
such as 

[ERR] mod_sndfile.c:194 Error Opening File [the_file_path_and_name]
[System error : No such file or directory.]

 

However, session.streamFile returns false whether it gets an error or
successfully plays the file. Is there any other way to capture success
or capture that an error happened?

 

-Jason Moran

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From gavin.henry at gmail.com  Fri Jul  8 22:44:36 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 8 Jul 2011 19:44:36 +0100
Subject: [Freeswitch-users] Load Balance Trunks
In-Reply-To: <COL109-W4456F57D6EF8C3739998F5F45F0@phx.gbl>
References: <col109-w22D1131E50A2495C1521C9F4590@phx.gbl>
	<C71A4485-E533-48AE-AC3F-C59D7A1E79CA@ipeva.fr>
	<COL109-W612603C4C7BCC734CBD974F45B0@phx.gbl>
	<BANLkTinT9O5+hzT1YczsmQV5ouL9EQ_2TQ@mail.gmail.com>
	<233898962-1309536985-cardhu_decombobulator_blackberry.rim.net-2106826141-@b13.c4.bise7.blackberry>
	<BANLkTikjYw=0UDyGdipBzadj0_kSoOVS2Q@mail.gmail.com>
	<1637192224-1309538673-cardhu_decombobulator_blackberry.rim.net-1258156530-@b13.c4.bise7.blackberry>
	<BANLkTineHeUHCMpKNJS7msb6xCZzBqmkQg@mail.gmail.com>
	<COL109-W4456F57D6EF8C3739998F5F45F0@phx.gbl>
Message-ID: <CAA8_NKANOR7+mq1mOD=r1ch2S+dknf-FEcn++L0=z4hY0egMoQ@mail.gmail.com>

Hi Abid,

What country are you based in? Do you have a good connection into the UK?

Thanks.

On 7/5/11, Abid Saleem <abid_freeswitch at live.com> wrote:
>
> Hi Micheal, Avi and All,
> Sorry for a little late response as I was away. I have seen quite a few
> questions from you guys, so I am answering them in one email as below.
> 1- How does the provider notify you that each trunk has used its allotted
> time?Abid -> They have some counter in their IMS network to count on mins
> per trunk per day and they inform us by email.2- Are these trunks inbound
> only?Abid -> No. All these are Outbound Trunks. We just use them to send
> outgoing calls to our provider.
> 3- What happens when a call extends more than 120 mins on a trunk, would the
> call be disconnected?Abid -> The call is not disconnected right away but
> they send us a notification the next day. There is no real-time
> disconnection.
> 4- And then no more calls that day on that trunk?Abid -> Calls do not stop
> connecting immediately but they keep going. Currently their notification
> process is manual not automatic blocking.
> Please help me if you can. Thanks.
> Regards-----------------Abid SaleemTechnical Manager NGNTerminus
> Technologies
> Date: Fri, 1 Jul 2011 09:49:02 -0700
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Load Balance Trunks
>
> On Fri, Jul 1, 2011 at 9:46 AM,  <hmkias at gmail.com> wrote:
>
> Lcr entry could be activated on the the trunks. A cron job could monitor the
> usage and add or remove the trunk for the day.
>
> Are these trunks inbound only?
> -MC
>
>
> _______________________________________________
> 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 		 	   		

-- 
Sent from my mobile device

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http://www.surevoip.co.uk


From gavin.henry at gmail.com  Fri Jul  8 22:51:55 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 8 Jul 2011 19:51:55 +0100
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
 Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <4E163411.50507@earthspike.net>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
	<4DF10C83.6020803@earthspike.net>
	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>
	<4DF1F07A.5000200@earthspike.net>
	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>
	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>
	<4DFB9831.4040709@earthspike.net>
	<BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>
	<CAP1_MOCq5qRjsgd7XGag9ukjHiG2Se-pXvrghQ7hbB5ciwJK+A@mail.gmail.com>
	<4E156252.2030308@earthspike.net>
	<CAP1_MOBAFpYqJNKhgWzv3aYJeKbj+RJDQJCaQByVcrMVp_s8RA@mail.gmail.com>
	<CAE=3qOZaqoa1zt1_q_gaYzME=DOJYqhW4i7afhrBYc4PxViq5g@mail.gmail.com>
	<C06E47096A4E434AB7466C6309AC9FE5@dell9400>
	<4E163411.50507@earthspike.net>
Message-ID: <CAA8_NKCs_CJKEtHG9CpDZvK9vXevGtjEA18tNoEZ0yzKc2kmYg@mail.gmail.com>

Hi John,

We set this up for a customer using a Sangoma card and FusionPBX not
long ago. I can dig out the config next week when back in the office.
It's for ISDN30e but should help.

Thanks.

On 7/7/11, John <freeswitch at earthspike.net> wrote:
> Jan,
>
> I have tried every combination of unknown, national and international
> with every combination of number length.  I have also confirmed that by
> using wanpipemon to capture the D-channel messages.  When I went through
> them with Wireshark and compared them with Q.931, everything was being
> set correctly by the Sangoma card/drivers.  I am at the point now where
> I need someone who has worked with BT ISDN switches to tell me what they
> can accept.  It seems that this is a national secret over here, or
> instead that they can accept almost anything and my lines have been
> configured to override the outgoing calling number with the base
> number.  Either way, the only way in which I think I am now going to get
> a result is by booking a fault.  Your helpful comment about modern
> switches being able to work this out also confirms that I am probably
> not looking at a Q.931 formatting error on my part, but on a blockage in
> the exchange.  I don't have a problem with the called number, for example.
>
> John
>
> On 07/07/11 23:09, Jan Berger wrote:
>>
>> I must admit I don't remember what CLIP and CLOP is -- and my Q.931
>> experience is getting a bit rusty...
>>
>> Can you get a snoop of L3 out and in so I can see what you send and
>> what the switch responds back?
>>
>> I prefer to use Called and Calling -- The numbers contain a few bits
>> that tell what number this is. The most common are unknown, national
>> and international. These bits must match the actual number you send.
>> The 10 digit number is a unknown, the 11 digit is a national.
>> International start with 00 nn
>>
>> On top of that you will face that modern switches are quite capable
>> and able to configure whatever behaviour they want per line with
>> regards to called/calling -- so you need to find out exactly what the
>> switch expect on both called and caller for outgoing to behave as you
>> want.
>>
>> /Jan
>>
>> >>
>> >> On our PRI systems we are sending 10 digits (2071231234), and on the
>> >> BRI system we are sending 11 digits (02071231234).
>> >>
>>
>>
>> _______________________________________________
>> 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
>
>

-- 
Sent from my mobile device

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


From kris at livecall.com  Sat Jul  9 03:16:43 2011
From: kris at livecall.com (Kris)
Date: Fri, 8 Jul 2011 16:16:43 -0700
Subject: [Freeswitch-users] how-to catch errors in session streamFile
References: <361E98F99D3CC3439EED59BC1924ED69508307@SERVER2003.SecuReachSystems.local>
Message-ID: <E18854675C214127B0DD64FA8EBD8D0F@stor1>

You can experiment with 
Convert.ToInt32(Session.GetVariable("playback_samples"));

string PlayedMilliseconds = Session.GetVariable("playback_ms");

It might give an indication if any of the file was played. 

----- Original Message ----- 
From: "Jason Moran" <jmoran at secureachsystems.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, July 08, 2011 8:35 AM
Subject: [Freeswitch-users] how-to catch errors in session streamFile


Once in a while I pass a bad file name or some other bad thing happens
with session.streamFile. If I have the console up I can see an error
such as 

[ERR] mod_sndfile.c:194 Error Opening File [the_file_path_and_name]
[System error : No such file or directory.]

 

However, session.streamFile returns false whether it gets an error or
successfully plays the file. Is there any other way to capture success
or capture that an error happened?

 

-Jason Moran




From fs-list at communicatefreely.net  Sat Jul  9 04:22:57 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 08 Jul 2011 20:22:57 -0400
Subject: [Freeswitch-users] NOTIFY event not sent to user
In-Reply-To: <4E14CC5F.60205@communicatefreely.net>
References: <4E14B509.6000108@communicatefreely.net>
	<4E14CC5F.60205@communicatefreely.net>
Message-ID: <4E179F61.5090106@communicatefreely.net>

I guess call ID really matters.

Once I set the call ID to the subscription I wanted, it worked.

I had to add a little extra code that looked up the call IDs from the 
sip_registrations table first, then issued a NOTIFY to each one 
individually.



Tim St. Pierre wrote:
> I may have made some progress here, but I'm coming up on additional issues.
>
> If I add a Call-ID header to the event that matches the subscription, it 
> works.  This is a bit of a pain.  Any way to make it send to all the 
> registrations for that user?
>
> Also, in the resulting notify, I get this extra line:
>
> Subscription-State:terminated;reason=timeout
>
> Any way to disable that?  The phone comes back with "Bad Event", and I 
> wonder if that header has something to do with it.
>
> Thanks!
>
> -Tim
>
> Tim St. Pierre wrote:
>   
>> Hello,
>>
>> I'm trying to send notify events to Aastra phones for screen pops and 
>> that sort of thing.  I'm triggering the event using ESL from a PHP 
>> script.  If I subscribe to the NOTIFY events on fs_cli, I can see my 
>> event coming in quite nicely, but nothing gets sent to the phone.  I am 
>> doing a packet capture between the phone and Freeswitch, and although I 
>> can see all the regular call setup and MWI packets, I don't get my 
>> notify packets coming through.
>>
>> Here's the event as it is seen on fs_cli:
>>
>> RECV EVENT
>> Core-UUID: c8db9457-8e51-e011-8f26-001517ac7a54
>> FreeSWITCH-Hostname: stefan.151front.communicatefreely.net
>> FreeSWITCH-IPv4: 66.207.200.21
>> FreeSWITCH-IPv6: ::1
>> Event-Date-Local: 2011-07-06 15:08:28
>> Event-Date-GMT: Wed, 06 Jul 2011 19:08:28 GMT
>> Event-Date-Timestamp: 1309979308360046
>> Event-Calling-File: mod_event_socket.c
>> Event-Calling-Function: read_packet
>> Event-Calling-Line-Number: 1131
>> Command: sendevent NOTIFY
>> Event-Name: NOTIFY
>> profile: internal
>> event-string: aastra-xml
>> user: 5101 at communicatefreely.net
>> host: stefan.151front.communicatefreely.net
>> content-type: application/xml
>> Content-Length: 223
>> Content-Length: 223
>>
>> <AastraIPPhoneStatus Beep="yes" triggerDestroyOnExit="no">
>>         <Session>1</Session>
>>         <Message Index="test" Type="alert" Timeout="15">
>>                 Hello, anyone there?
>>         </Message>
>> </AastraIPPhoneStatus>
>>
>> freeswitch at default>
>>
>> Just for background, if I do sofia status profile internal user 
>> 5101 at communicatefreely.net I get all the details about the phone, and 
>> the username and hostname's match what's in the packet above.
>>
>> Any suggestions?
>>
>>
>> _______________________________________________
>> 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
>>   
>>     
>
>
> _______________________________________________
> 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 kerem.erciyes at gmail.com  Sat Jul  9 16:02:25 2011
From: kerem.erciyes at gmail.com (Kerem Erciyes)
Date: Sat, 9 Jul 2011 15:02:25 +0300
Subject: [Freeswitch-users] FreeSwitch/voip on EC2 or other
	cloud/virtual services
In-Reply-To: <CA38A907.18941%krice@freeswitch.org>
References: <CAJW763kFu86BO6x=2dxZHhTpuPYbB4Fvfwkh5S7d1yrnOAvK8w@mail.gmail.com>
	<CA38A907.18941%krice@freeswitch.org>
Message-ID: <CAJ6wjQFvdgAh=CzQ1NV_n0pRqmo2oNjXQ5z_x8O7A4Q+-dKiug@mail.gmail.com>

Hi,

Just wanted to plug that Linode is satisfactorily like AWS Clouds. You can
get routine snapshots (called backups), stack-scripts to deploy more servers
of the same nature and geographical distribution.

I've been using Linodes for 2 years now and very happy with it.

I also have a t1.micro AWS instance running a test setup of FreeSwitch so if
anybody want to try some test via Skype or SIP just let me know.

Ciao,
Kerem

Kerem Erciyes - Sistem Danismani
http://keremerciyes.com
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From bwibowo at gmail.com  Sat Jul  9 16:44:07 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Sat, 9 Jul 2011 19:44:07 +0700
Subject: [Freeswitch-users] mod_dingaling error
Message-ID: <CAKnRR0ZQ-qTqx+jX8NwQvQ3_tLuEgBvkGCX3rW2hgxbP8atkrw@mail.gmail.com>

hi
i have 2 freeswitch running, 1 has mod_dingaling installed and normal. but
the other when i dial i got the call ringing then got ivr "please try again"
from freeswitch.
configuration between 2 FS almost same.
any idea please share


thx

budi
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From michel.daggelinckx at gmail.com  Sat Jul  9 19:18:29 2011
From: michel.daggelinckx at gmail.com (Michel Daggelinckx)
Date: Sat, 9 Jul 2011 17:18:29 +0200
Subject: [Freeswitch-users] mod_dingaling error
In-Reply-To: <CAKnRR0ZQ-qTqx+jX8NwQvQ3_tLuEgBvkGCX3rW2hgxbP8atkrw@mail.gmail.com>
References: <CAKnRR0ZQ-qTqx+jX8NwQvQ3_tLuEgBvkGCX3rW2hgxbP8atkrw@mail.gmail.com>
Message-ID: <CAFYZzbsM9hqSMYJmBsCwnj4S=p7YVepyVCSedUzDX6COgUB9Vw@mail.gmail.com>

I almost have an idea, now if you can share your configs, (passwords and
adresses remover) we can realy check this out.


Michel


On Sat, Jul 9, 2011 at 2:44 PM, budi wibowo <bwibowo at gmail.com> wrote:

> hi
> i have 2 freeswitch running, 1 has mod_dingaling installed and normal. but
> the other when i dial i got the call ringing then got ivr "please try again"
> from freeswitch.
> configuration between 2 FS almost same.
> any idea please share
>
>
> thx
>
> budi
>
> _______________________________________________
> 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 viraptor at gmail.com  Sun Jul 10 01:11:13 2011
From: viraptor at gmail.com (=?ISO-8859-2?Q?Stanis=B3aw_Pitucha?=)
Date: Sat, 9 Jul 2011 22:11:13 +0100
Subject: [Freeswitch-users] Call with failover
Message-ID: <CAFrYPTNOT0WjJfMNO-S-FtE+q2RJ3z-vS2K6FcFDq=d2oeAHeQ@mail.gmail.com>

Hi all,
I've got a simple problem to solve that doesn't seem to be so simple
to execute. For some incoming call, I'd like to call one destination,
then if it fails (user not registered / busy / any other failure) call
another destination. Both destinations should be arbitrary, but for
now I'm ok with the first being set to a `user/...`. Second needs to
be sent through the dialplan (might need setting billing details,
etc.), or might be another user.

At the moment, I'm sending the call to "user/...|loopback/.../..." -
which often results in the call being simply hungup right after
loopback answers.
>From #freeswitch I learned that loopback can do that (still don't
understand it, didn't find any related bug either) and I should use
xml_curl or event sockets - but no real examples. So after doing a bit
of research, I run into the following problems:

- If I use dialplan via xml curl, I would either need to pass the
control back to some other context (using loopback seems to be the
only option), or compute all parameters in one go and do a transfer to
the final set of destinations (not possible in every case)

- If I use eventsocket, how can I react to the connection failure? It
seems that I will only get an event about the hangup and the channel
will disappear on its own. Is there even a way to do a "on hangup, do
this before anything else happens"?

Are there any examples of a similar scenario on the web? Any links /
longer explanations would be appreciated.

PS. is it defined somewhere why/when does loopback destroy the call on
accepted connection?

-- 
KTHXBYE,

Stanis?aw Pitucha


From avi at avimarcus.net  Sun Jul 10 01:31:00 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 10 Jul 2011 00:31:00 +0300
Subject: [Freeswitch-users] Call with failover
In-Reply-To: <CAFrYPTNOT0WjJfMNO-S-FtE+q2RJ3z-vS2K6FcFDq=d2oeAHeQ@mail.gmail.com>
References: <CAFrYPTNOT0WjJfMNO-S-FtE+q2RJ3z-vS2K6FcFDq=d2oeAHeQ@mail.gmail.com>
Message-ID: <CALNZuwnE2X4jmZojZj=25z-NEoi6xELqDQBADjBP5WaXkQPWvw@mail.gmail.com>

You can try using a transfer after the bridge.

-Avi

2011/7/10 Stanis?aw Pitucha <viraptor at gmail.com>

> Hi all,
> I've got a simple problem to solve that doesn't seem to be so simple
> to execute. For some incoming call, I'd like to call one destination,
> then if it fails (user not registered / busy / any other failure) call
> another destination. Both destinations should be arbitrary, but for
> now I'm ok with the first being set to a `user/...`. Second needs to
> be sent through the dialplan (might need setting billing details,
> etc.), or might be another user.
>
> At the moment, I'm sending the call to "user/...|loopback/.../..." -
> which often results in the call being simply hungup right after
> loopback answers.
> From #freeswitch I learned that loopback can do that (still don't
> understand it, didn't find any related bug either) and I should use
> xml_curl or event sockets - but no real examples. So after doing a bit
> of research, I run into the following problems:
>
> - If I use dialplan via xml curl, I would either need to pass the
> control back to some other context (using loopback seems to be the
> only option), or compute all parameters in one go and do a transfer to
> the final set of destinations (not possible in every case)
>
> - If I use eventsocket, how can I react to the connection failure? It
> seems that I will only get an event about the hangup and the channel
> will disappear on its own. Is there even a way to do a "on hangup, do
> this before anything else happens"?
>
> Are there any examples of a similar scenario on the web? Any links /
> longer explanations would be appreciated.
>
> PS. is it defined somewhere why/when does loopback destroy the call on
> accepted connection?
>
> --
> KTHXBYE,
>
> Stanis?aw Pitucha
>
> _______________________________________________
> 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 bwibowo at gmail.com  Sun Jul 10 04:16:42 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Sun, 10 Jul 2011 07:16:42 +0700
Subject: [Freeswitch-users] mod_dingaling error
In-Reply-To: <CAFYZzbsM9hqSMYJmBsCwnj4S=p7YVepyVCSedUzDX6COgUB9Vw@mail.gmail.com>
References: <CAKnRR0ZQ-qTqx+jX8NwQvQ3_tLuEgBvkGCX3rW2hgxbP8atkrw@mail.gmail.com>
	<CAFYZzbsM9hqSMYJmBsCwnj4S=p7YVepyVCSedUzDX6COgUB9Vw@mail.gmail.com>
Message-ID: <CAKnRR0bg+CNYVMERXWDYEqOXFQjKLrV8XryJHYokMKDbSckwMg@mail.gmail.com>

hi
FS version FreeSWITCH Version 1.0.head (git-8105f79 2011-07-09 00-06-03
-0400) on Centos 4.6, other server is centos 5.5

client.xml

<include>
 <profile type="client">
   <param name="name" value="gtalk"/>
   <param name="login" value="x009 at gmail.com/gtalk"/>
   <param name="password" value="123"/>
   <param name="server" value="talk.google.com"/>
   <param name="dialplan" value="XML"/>
   <param name="message" value="Gtalk Integration"/>
   <param name="rtp-ip" value="auto"/>
   <!-- if you are behind NAT, you should probably use stun -->
   <param name="ext-rtp-ip" value="202.147"/>
   <param name="auto-login" value="true"/>
   <param name="auto-reply" value="join my conference"/>
   <param name="sasl" value="plain"/>
   <param name="tls" value="true"/>
   <param name="use-rtp-timer" value="true"/>
   <param name="candidate-acl" value="wan.auto"/>
   <param name="exten" value="1000"/> <!-- The extension you want a call in
to connect to -->
 </profile>

</include>

and this is the 00_gtalk.xml in /usr/local/freeswitch/conf/dialplan/default
<include>
    <extension name="gvoice_out">
        <condition field="destination_number" expression="^(1\d{10})$">
          <action application="set" data="hangup_after_bridge=true"/>
          <action application="bridge" data="dingaling/gtalk/+$
1 at voice.google.com"/>
        </condition>
    </extension>
</include>


br

budi wibowo

On Sat, Jul 9, 2011 at 10:18 PM, Michel Daggelinckx <
michel.daggelinckx at gmail.com> wrote:

> I almost have an idea, now if you can share your configs, (passwords and
> adresses remover) we can realy check this out.
>
>
> Michel
>
>
> On Sat, Jul 9, 2011 at 2:44 PM, budi wibowo <bwibowo at gmail.com> wrote:
>
>> hi
>> i have 2 freeswitch running, 1 has mod_dingaling installed and normal. but
>> the other when i dial i got the call ringing then got ivr "please try again"
>> from freeswitch.
>> configuration between 2 FS almost same.
>> any idea please share
>>
>>
>> thx
>>
>> budi
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 admin at blindi.net  Sun Jul 10 04:52:23 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Sun, 10 Jul 2011 02:52:23 +0200 (CEST)
Subject: [Freeswitch-users] Uptimecommand on cli?
In-Reply-To: <CAKnRR0bg+CNYVMERXWDYEqOXFQjKLrV8XryJHYokMKDbSckwMg@mail.gmail.com>
References: <CAKnRR0ZQ-qTqx+jX8NwQvQ3_tLuEgBvkGCX3rW2hgxbP8atkrw@mail.gmail.com>
	<CAFYZzbsM9hqSMYJmBsCwnj4S=p7YVepyVCSedUzDX6COgUB9Vw@mail.gmail.com>
	<CAKnRR0bg+CNYVMERXWDYEqOXFQjKLrV8XryJHYokMKDbSckwMg@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1107100246080.19434@tomback.antagus.de>

Hi guys,
is there a cli command with which you can find out how long fs is
  already running?
thanks.


---------------
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http://www.blindi.net/mailman/listinfo/blinde-misc

From krice at freeswitch.org  Sun Jul 10 05:03:30 2011
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 09 Jul 2011 20:03:30 -0500
Subject: [Freeswitch-users] Uptimecommand on cli?
In-Reply-To: <Pine.LNX.4.64.1107100246080.19434@tomback.antagus.de>
Message-ID: <CA3E6492.18B90%krice@freeswitch.org>

status


On 7/9/11 7:52 PM, "Thomas Hoellriegel" <admin at blindi.net> wrote:

> Hi guys,
> is there a cli command with which you can find out how long fs is
>   already running?
> 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
> _______________________________________________
> 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 mitch.johnson7 at gmail.com  Sun Jul 10 06:58:52 2011
From: mitch.johnson7 at gmail.com (mitch Johnson)
Date: Sat, 9 Jul 2011 22:58:52 -0400
Subject: [Freeswitch-users] One way calling
Message-ID: <CADMSHVFon7idZCvddOdERpfN7nZ+FHsWqLkBHpE3y+eY0OCJVw@mail.gmail.com>

I have a phone hanging off a cisco callmanager, the DN is 8000.  I have
phones on the FS, DN 1000.  When I dial from the Cisco phone to the
Freeswitch phone (8000 to 1000) the phone rings.  However, when I dial from
Freeswitch to Cisco Callmanager, the call drops immediately, nothing rings
out.

>From the FS_CLI, I see this output:

2011-07-09 22:43:59.209255 [NOTICE] switch_channel.c:812 New Channel
sofia/internal/1000 at 172.16.200.60 [2657e237-f43d-4574-a7a3-7ef1fab749b8]
2011-07-09 22:43:59.215279 [INFO] mod_dialplan_xml.c:331 Processing Test
user <1000>->8000 in context public
2011-07-09 22:43:59.219276 [NOTICE] switch_ivr.c:1606 Transfer
sofia/internal/1000 at 172.16.200.60 to XML[8000 at default]
2011-07-09 22:43:59.221278 [INFO] mod_dialplan_xml.c:331 Processing Test
user <1000>->8000 in context default
2011-07-09 22:43:59.311287 [NOTICE] mod_dptools.c:929 Channel
[sofia/internal/1000 at 172.16.200.60] has been answered
2011-07-09 22:43:59.625253 [INFO] switch_rtp.c:2972 Auto Changing port from
68.225.42.67:21349 to 192.168.60.142:4000
2011-07-09 22:44:00.365253 [NOTICE] switch_core_state_machine.c:189
sofia/internal/1000 at 172.16.200.60 has executed the last dialplan
instruction, hanging up.
2011-07-09 22:44:00.365253 [NOTICE] switch_core_state_machine.c:191 Hangup
sofia/internal/1000 at 172.16.200.60 [CS_EXECUTE] [NORMAL_CLEARING]
2011-07-09 22:44:00.365253 [NOTICE] switch_core_session.c:1306 Session 7
(sofia/internal/1000 at 172.16.200.60) Ended
2011-07-09 22:44:00.365253 [NOTICE] switch_core_session.c:1308 Close Channel
sofia/internal/1000 at 172.16.200.60 [CS_DESTROY]


I've noticed that it looks as if it's trying to send the call to the
Internet, all these calls are internal.  My trunk to the Cisco Callmanager
is below:

 <extension name="Dial to CM8">
      <condition field="destination_number" expression="^(800\d)$">
        <action application="bridge" data="sofia/internal/$1 at 172.16.200.100
"/>
      </condition>
     </extension>



Below is a successful call from the Callmanager to the Freeswitch:

2011-07-09 22:45:56.543784 [NOTICE] switch_channel.c:812 New Channel
sofia/internal/8000 at 172.16.200.100 [d06d7795-0c35-4bfe-85bf-9f01498bfafc]
2011-07-09 22:45:56.810823 [NOTICE] sofia.c:5285 Channel [sofia/internal/
8000 at 172.16.200.100] has been answered
2011-07-09 22:45:56.812617 [INFO] mod_dialplan_xml.c:331 Processing 8000
<8000>->1000 in context public
2011-07-09 22:45:56.812617 [NOTICE] switch_ivr.c:1606 Transfer
sofia/internal/8000 at 172.16.200.100 to XML[1000 at default]
2011-07-09 22:45:56.812617 [INFO] mod_dialplan_xml.c:331 Processing 8000
<8000>->1000 in context default
2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *1
execute_extension::dx XML features
2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *2
record_session::/usr/local/freeswitch/recordings/8000.2011-07-09-22-45-56.wav
2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *3
execute_extension::cf XML features
2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *4
execute_extension::att_xfer XML features
2011-07-09 22:45:56.828985 [NOTICE] switch_channel.c:812 New Channel
sofia/internal/sip:1000 at 192.168.60.142:60936[846acd6a-3188-4569-9b81-2da9b3e88a87]
2011-07-09 22:45:57.010788 [INFO] sofia.c:729 sofia/internal/
sip:1000 at 192.168.60.142:60936 Update Callee ID to "Outbound Call" <1000>
2011-07-09 22:45:57.010788 [NOTICE] sofia.c:4779 Ring-Ready sofia/internal/
sip:1000 at 192.168.60.142:60936!
2011-07-09 22:46:03.091789 [NOTICE] sofia.c:5285 Channel [sofia/internal/
sip:1000 at 192.168.60.142:60936] has been answered
2011-07-09 22:46:03.924748 [INFO] switch_rtp.c:2972 Auto Changing port from
68.225.42.67:22147 to 192.168.60.142:4002
2011-07-09 22:46:04.920744 [NOTICE] sofia.c:537 Hangup sofia/internal/
sip:1000 at 192.168.60.142:60936 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2011-07-09 22:46:04.924746 [NOTICE] switch_ivr_bridge.c:1328 Hangup
sofia/internal/8000 at 172.16.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1306 Session 9
(sofia/internal/sip:1000 at 192.168.60.142:60936) Ended
2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1308 Close Channel
sofia/internal/sip:1000 at 192.168.60.142:60936 [CS_DESTROY]
2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1306 Session 8
(sofia/internal/8000 at 172.16.200.100) Ended
2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1308 Close Channel
sofia/internal/8000 at 172.16.200.100 [CS_DESTROY]


I have done more extensive debugs and nothing really jumps out at me.

Any help would be greatly appreciated,

Thanks,

Mitch
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From david.ponzone at ipeva.fr  Sun Jul 10 15:01:55 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 10 Jul 2011 13:01:55 +0200
Subject: [Freeswitch-users] One way calling
In-Reply-To: <CADMSHVFon7idZCvddOdERpfN7nZ+FHsWqLkBHpE3y+eY0OCJVw@mail.gmail.com>
References: <CADMSHVFon7idZCvddOdERpfN7nZ+FHsWqLkBHpE3y+eY0OCJVw@mail.gmail.com>
Message-ID: <7A42121C-B7AD-4BF6-B7E8-D65522D8EB33@ipeva.fr>

Mitch

> 
>  <extension name="Dial to CM8">
>       <condition field="destination_number" expression="^(800\d)$">
>         <action application="bridge" data="sofia/internal/$1 at 172.16.200.100"/>
>       </condition>
>      </extension>
> 

Is this extension in the default dialplan ?

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From mitch.johnson7 at gmail.com  Sun Jul 10 19:18:52 2011
From: mitch.johnson7 at gmail.com (mitch Johnson)
Date: Sun, 10 Jul 2011 11:18:52 -0400
Subject: [Freeswitch-users] One way calling (David Ponzone)
Message-ID: <CADMSHVHKpJ_JAvdLZB0tq+YWpEU1PDLaNL4hp2oVsu++tNFOwg@mail.gmail.com>

David,

Yes it is, it's the last entry in the default dialplan.

Thankyou,

Mitch

---------- Forwarded message ----------
> From: mitch Johnson <mitch.johnson7 at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Sat, 9 Jul 2011 22:58:52 -0400
> Subject: [Freeswitch-users] One way calling
> I have a phone hanging off a cisco callmanager, the DN is 8000.  I have
> phones on the FS, DN 1000.  When I dial from the Cisco phone to the
> Freeswitch phone (8000 to 1000) the phone rings.  However, when I dial from
> Freeswitch to Cisco Callmanager, the call drops immediately, nothing rings
> out.
>
> From the FS_CLI, I see this output:
>
> 2011-07-09 22:43:59.209255 [NOTICE] switch_channel.c:812 New Channel
> sofia/internal/1000 at 172.16.200.60 [2657e237-f43d-4574-a7a3-7ef1fab749b8]
> 2011-07-09 22:43:59.215279 [INFO] mod_dialplan_xml.c:331 Processing Test
> user <1000>->8000 in context public
> 2011-07-09 22:43:59.219276 [NOTICE] switch_ivr.c:1606 Transfer
> sofia/internal/1000 at 172.16.200.60 to XML[8000 at default]
> 2011-07-09 22:43:59.221278 [INFO] mod_dialplan_xml.c:331 Processing Test
> user <1000>->8000 in context default
> 2011-07-09 22:43:59.311287 [NOTICE] mod_dptools.c:929 Channel
> [sofia/internal/1000 at 172.16.200.60] has been answered
> 2011-07-09 22:43:59.625253 [INFO] switch_rtp.c:2972 Auto Changing port from
> 68.225.42.67:21349 to 192.168.60.142:4000
> 2011-07-09 22:44:00.365253 [NOTICE] switch_core_state_machine.c:189
> sofia/internal/1000 at 172.16.200.60 has executed the last dialplan
> instruction, hanging up.
> 2011-07-09 22:44:00.365253 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia/internal/1000 at 172.16.200.60 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-07-09 22:44:00.365253 [NOTICE] switch_core_session.c:1306 Session 7
> (sofia/internal/1000 at 172.16.200.60) Ended
> 2011-07-09 22:44:00.365253 [NOTICE] switch_core_session.c:1308 Close
> Channel sofia/internal/1000 at 172.16.200.60 [CS_DESTROY]
>
>
> I've noticed that it looks as if it's trying to send the call to the
> Internet, all these calls are internal.  My trunk to the Cisco Callmanager
> is below:
>
>  <extension name="Dial to CM8">
>       <condition field="destination_number" expression="^(800\d)$">
>         <action application="bridge" data="sofia/internal/$
> 1 at 172.16.200.100"/>
>       </condition>
>      </extension>
>
>
>
> Below is a successful call from the Callmanager to the Freeswitch:
>
> 2011-07-09 22:45:56.543784 [NOTICE] switch_channel.c:812 New Channel
> sofia/internal/8000 at 172.16.200.100 [d06d7795-0c35-4bfe-85bf-9f01498bfafc]
> 2011-07-09 22:45:56.810823 [NOTICE] sofia.c:5285 Channel [sofia/internal/
> 8000 at 172.16.200.100] has been answered
> 2011-07-09 22:45:56.812617 [INFO] mod_dialplan_xml.c:331 Processing 8000
> <8000>->1000 in context public
> 2011-07-09 22:45:56.812617 [NOTICE] switch_ivr.c:1606 Transfer
> sofia/internal/8000 at 172.16.200.100 to XML[1000 at default]
> 2011-07-09 22:45:56.812617 [INFO] mod_dialplan_xml.c:331 Processing 8000
> <8000>->1000 in context default
> 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *1
> execute_extension::dx XML features
> 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *2
> record_session::/usr/local/freeswitch/recordings/8000.2011-07-09-22-45-56.wav
> 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *3
> execute_extension::cf XML features
> 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *4
> execute_extension::att_xfer XML features
> 2011-07-09 22:45:56.828985 [NOTICE] switch_channel.c:812 New Channel
> sofia/internal/sip:1000 at 192.168.60.142:60936[846acd6a-3188-4569-9b81-2da9b3e88a87]
> 2011-07-09 22:45:57.010788 [INFO] sofia.c:729 sofia/internal/
> sip:1000 at 192.168.60.142:60936 Update Callee ID to "Outbound Call" <1000>
> 2011-07-09 22:45:57.010788 [NOTICE] sofia.c:4779 Ring-Ready sofia/internal/
> sip:1000 at 192.168.60.142:60936!
> 2011-07-09 22:46:03.091789 [NOTICE] sofia.c:5285 Channel [sofia/internal/
> sip:1000 at 192.168.60.142:60936] has been answered
> 2011-07-09 22:46:03.924748 [INFO] switch_rtp.c:2972 Auto Changing port from
> 68.225.42.67:22147 to 192.168.60.142:4002
> 2011-07-09 22:46:04.920744 [NOTICE] sofia.c:537 Hangup sofia/internal/
> sip:1000 at 192.168.60.142:60936 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2011-07-09 22:46:04.924746 [NOTICE] switch_ivr_bridge.c:1328 Hangup
> sofia/internal/8000 at 172.16.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1306 Session 9
> (sofia/internal/sip:1000 at 192.168.60.142:60936) Ended
> 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1308 Close
> Channel sofia/internal/sip:1000 at 192.168.60.142:60936 [CS_DESTROY]
> 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1306 Session 8
> (sofia/internal/8000 at 172.16.200.100) Ended
> 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1308 Close
> Channel sofia/internal/8000 at 172.16.200.100 [CS_DESTROY]
>
>
> I have done more extensive debugs and nothing really jumps out at me.
>
> Any help would be greatly appreciated,
>
> Thanks,
>
> Mitch
>
>
> ---------- Forwarded message ----------
> From: David Ponzone <david.ponzone at ipeva.fr>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Sun, 10 Jul 2011 13:01:55 +0200
> Subject: Re: [Freeswitch-users] One way calling
> Mitch
>
>
>  <extension name="Dial to CM8">
>       <condition field="destination_number" expression="^(800\d)$">
>         <action application="bridge" data="sofia/internal/$
> 1 at 172.16.200.100"/>
>       </condition>
>      </extension>
>
>
> Is this extension in the default dialplan ?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rgelfand2 at gmail.com  Sun Jul 10 20:21:21 2011
From: rgelfand2 at gmail.com (Roman Gelfand)
Date: Sun, 10 Jul 2011 12:21:21 -0400
Subject: [Freeswitch-users] Speech Recognition
Message-ID: <CAJbW+rkEuGhEBKxZQyM0z3AZ0zVSA0yneeTB8uZHVicB052fKA@mail.gmail.com>

I am looking to script a speech recognition ivr system.  Has anyone used
cmusphinx to accomplish?  If not has anyone use a good open source speech
recognition engine with freeswitch?

Thanks in advance
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From curriegrad2004 at gmail.com  Sun Jul 10 20:27:42 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 10 Jul 2011 09:27:42 -0700
Subject: [Freeswitch-users] Speech Recognition
In-Reply-To: <CAJbW+rkEuGhEBKxZQyM0z3AZ0zVSA0yneeTB8uZHVicB052fKA@mail.gmail.com>
References: <CAJbW+rkEuGhEBKxZQyM0z3AZ0zVSA0yneeTB8uZHVicB052fKA@mail.gmail.com>
Message-ID: <CAMWiMy-fQZSP7VRBdBDumvg8zoW7cDTfmvKpt=vDJqkSdJKGPA@mail.gmail.com>

There's pocketsphinx in FreeSwitch.

On Sun, Jul 10, 2011 at 9:21 AM, Roman Gelfand <rgelfand2 at gmail.com> wrote:
> I am looking to script a speech recognition ivr system.? Has anyone used
> cmusphinx to accomplish?? If not has anyone use a good open source speech
> recognition engine with freeswitch?
>
> Thanks in advance
> _______________________________________________
> 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 jan.berger at video24.no  Sun Jul 10 21:20:51 2011
From: jan.berger at video24.no (Jan Berger)
Date: Sun, 10 Jul 2011 19:20:51 +0200
Subject: [Freeswitch-users] Speech Recognition
In-Reply-To: <CAMWiMy-fQZSP7VRBdBDumvg8zoW7cDTfmvKpt=vDJqkSdJKGPA@mail.gmail.com>
References: <CAJbW+rkEuGhEBKxZQyM0z3AZ0zVSA0yneeTB8uZHVicB052fKA@mail.gmail.com>
	<CAMWiMy-fQZSP7VRBdBDumvg8zoW7cDTfmvKpt=vDJqkSdJKGPA@mail.gmail.com>
Message-ID: <1E59A19717CD4D3DB2895DABDF8D7485@dell9400>

You can also use pocketsphinx through MRCP.

I was about to say that the challenge is language models - but, to be honest
this page changed my mind:
http://sourceforge.net/projects/cmusphinx/files/Acoustic%20and%20Language%20
Models/

This is actually far better coverage than a majority of the commercial
actors that I have queried - and I checked this list for 1 year ago so it's
quite a change.

Just watch out for license conflicts. 

/Jan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
curriegrad2004
Sent: 10. juli 2011 18:28
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Speech Recognition

There's pocketsphinx in FreeSwitch.

On Sun, Jul 10, 2011 at 9:21 AM, Roman Gelfand <rgelfand2 at gmail.com> wrote:
> I am looking to script a speech recognition ivr system.? Has anyone used
> cmusphinx to accomplish?? If not has anyone use a good open source speech
> recognition engine with freeswitch?
>
> Thanks in advance
> _______________________________________________
> 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
>
>

_______________________________________________
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 david.ponzone at ipeva.fr  Sun Jul 10 22:02:13 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 10 Jul 2011 20:02:13 +0200
Subject: [Freeswitch-users] One way calling (David Ponzone)
In-Reply-To: <CADMSHVHKpJ_JAvdLZB0tq+YWpEU1PDLaNL4hp2oVsu++tNFOwg@mail.gmail.com>
References: <CADMSHVHKpJ_JAvdLZB0tq+YWpEU1PDLaNL4hp2oVsu++tNFOwg@mail.gmail.com>
Message-ID: <E97C831D-B431-445B-A469-CA0D1BEE8D3B@ipeva.fr>

So, sorry for the obvious question, but did you reloadxml ?

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

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

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




Le 10/07/2011 ? 17:18, mitch Johnson a ?crit :

> David,
> 
> Yes it is, it's the last entry in the default dialplan.
> 
> Thankyou,
> 
> Mitch
> 
> ---------- Forwarded message ----------
> From: mitch Johnson <mitch.johnson7 at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Sat, 9 Jul 2011 22:58:52 -0400
> Subject: [Freeswitch-users] One way calling
> I have a phone hanging off a cisco callmanager, the DN is 8000.  I have phones on the FS, DN 1000.  When I dial from the Cisco phone to the Freeswitch phone (8000 to 1000) the phone rings.  However, when I dial from Freeswitch to Cisco Callmanager, the call drops immediately, nothing rings out.
> 
> From the FS_CLI, I see this output:
> 
> 2011-07-09 22:43:59.209255 [NOTICE] switch_channel.c:812 New Channel sofia/internal/1000 at 172.16.200.60 [2657e237-f43d-4574-a7a3-7ef1fab749b8]
> 2011-07-09 22:43:59.215279 [INFO] mod_dialplan_xml.c:331 Processing Test user <1000>->8000 in context public
> 2011-07-09 22:43:59.219276 [NOTICE] switch_ivr.c:1606 Transfer sofia/internal/1000 at 172.16.200.60 to XML[8000 at default]
> 2011-07-09 22:43:59.221278 [INFO] mod_dialplan_xml.c:331 Processing Test user <1000>->8000 in context default
> 2011-07-09 22:43:59.311287 [NOTICE] mod_dptools.c:929 Channel [sofia/internal/1000 at 172.16.200.60] has been answered
> 2011-07-09 22:43:59.625253 [INFO] switch_rtp.c:2972 Auto Changing port from 68.225.42.67:21349 to 192.168.60.142:4000
> 2011-07-09 22:44:00.365253 [NOTICE] switch_core_state_machine.c:189 sofia/internal/1000 at 172.16.200.60 has executed the last dialplan instruction, hanging up.
> 2011-07-09 22:44:00.365253 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/internal/1000 at 172.16.200.60 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-07-09 22:44:00.365253 [NOTICE] switch_core_session.c:1306 Session 7 (sofia/internal/1000 at 172.16.200.60) Ended
> 2011-07-09 22:44:00.365253 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/1000 at 172.16.200.60 [CS_DESTROY]
> 
> 
> I've noticed that it looks as if it's trying to send the call to the Internet, all these calls are internal.  My trunk to the Cisco Callmanager is below:
> 
>  <extension name="Dial to CM8">
>       <condition field="destination_number" expression="^(800\d)$">
>         <action application="bridge" data="sofia/internal/$1 at 172.16.200.100"/>
>       </condition>
>      </extension>
> 
> 
> 
> Below is a successful call from the Callmanager to the Freeswitch:
> 
> 2011-07-09 22:45:56.543784 [NOTICE] switch_channel.c:812 New Channel sofia/internal/8000 at 172.16.200.100 [d06d7795-0c35-4bfe-85bf-9f01498bfafc]
> 2011-07-09 22:45:56.810823 [NOTICE] sofia.c:5285 Channel [sofia/internal/8000 at 172.16.200.100] has been answered
> 2011-07-09 22:45:56.812617 [INFO] mod_dialplan_xml.c:331 Processing 8000 <8000>->1000 in context public
> 2011-07-09 22:45:56.812617 [NOTICE] switch_ivr.c:1606 Transfer sofia/internal/8000 at 172.16.200.100 to XML[1000 at default]
> 2011-07-09 22:45:56.812617 [INFO] mod_dialplan_xml.c:331 Processing 8000 <8000>->1000 in context default
> 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *1 execute_extension::dx XML features
> 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/8000.2011-07-09-22-45-56.wav
> 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *3 execute_extension::cf XML features
> 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *4 execute_extension::att_xfer XML features
> 2011-07-09 22:45:56.828985 [NOTICE] switch_channel.c:812 New Channel sofia/internal/sip:1000 at 192.168.60.142:60936 [846acd6a-3188-4569-9b81-2da9b3e88a87]
> 2011-07-09 22:45:57.010788 [INFO] sofia.c:729 sofia/internal/sip:1000 at 192.168.60.142:60936 Update Callee ID to "Outbound Call" <1000>
> 2011-07-09 22:45:57.010788 [NOTICE] sofia.c:4779 Ring-Ready sofia/internal/sip:1000 at 192.168.60.142:60936!
> 2011-07-09 22:46:03.091789 [NOTICE] sofia.c:5285 Channel [sofia/internal/sip:1000 at 192.168.60.142:60936] has been answered
> 2011-07-09 22:46:03.924748 [INFO] switch_rtp.c:2972 Auto Changing port from 68.225.42.67:22147 to 192.168.60.142:4002
> 2011-07-09 22:46:04.920744 [NOTICE] sofia.c:537 Hangup sofia/internal/sip:1000 at 192.168.60.142:60936 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2011-07-09 22:46:04.924746 [NOTICE] switch_ivr_bridge.c:1328 Hangup sofia/internal/8000 at 172.16.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1306 Session 9 (sofia/internal/sip:1000 at 192.168.60.142:60936) Ended
> 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/sip:1000 at 192.168.60.142:60936 [CS_DESTROY]
> 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1306 Session 8 (sofia/internal/8000 at 172.16.200.100) Ended
> 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/8000 at 172.16.200.100 [CS_DESTROY]
> 
> 
> I have done more extensive debugs and nothing really jumps out at me.
> 
> Any help would be greatly appreciated,
> 
> Thanks,
> 
> Mitch
> 
> 
> ---------- Forwarded message ----------
> From: David Ponzone <david.ponzone at ipeva.fr>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Sun, 10 Jul 2011 13:01:55 +0200
> Subject: Re: [Freeswitch-users] One way calling
> Mitch
> 
>> 
>>  <extension name="Dial to CM8">
>>       <condition field="destination_number" expression="^(800\d)$">
>>         <action application="bridge" data="sofia/internal/$1 at 172.16.200.100"/>
>>       </condition>
>>      </extension>
>> 
> 
> Is this extension in the default dialplan ?
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> 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 cmcureau at gmail.com  Sun Jul 10 22:04:18 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Sun, 10 Jul 2011 13:04:18 -0500
Subject: [Freeswitch-users] segmentation fault with mod_spandsp
Message-ID: <CAL0fNgWbvFn8trf8L0eG2cZ9=xx4XM0oP461iBcdstFOcV-rTA@mail.gmail.com>

Hi there!

This morning, I did a fresh git checkout and attempted to build freeswitch
with mod_freetdm and mod_flite defined in the modules.conf.  I am now
getting a segfault when compiling:

making all mod_spandsp
Making all in src
/bin/bash: line 1: 22453 Segmentation fault      ./make_modem_filter -m V.17
-r > v17_v32bis_rx_floating_rrc.h
make[7]: *** [v17_v32bis_rx_floating_rrc.h] Error 139
make[6]: *** [all] Error 2
make[5]: *** [all-recursive] Error 1
make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la]
Error 2
make[3]: *** [mod_spandsp-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2

running make_modem_filter manually gets the same result.  Twiddling with the
commandline options doesn't help either.

OS: Debian 6.0.1a amd64, also a fresh install from last night.

Any idea what is happening?

Thanks in advance!
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From mitch.johnson7 at gmail.com  Sun Jul 10 22:15:10 2011
From: mitch.johnson7 at gmail.com (Mitch Johnson)
Date: Sun, 10 Jul 2011 14:15:10 -0400
Subject: [Freeswitch-users] One way calling (David Ponzone)
In-Reply-To: <mailman.12536.1310320968.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.12536.1310320968.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <C6408740-C207-49E9-B099-731D94ACCC6E@gmail.com>

Yes, I did a reloadxml, and I even restarted FreeSWITCH.

Thanks

From: David Ponzone <david.ponzone at ipeva.fr>
Date: July 10, 2011 2:02:13 PM EDT
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] One way calling (David Ponzone)
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>


So, sorry for the obvious question, but did you reloadxml ?

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

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

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




Le 10/07/2011 ? 17:18, mitch Johnson a ?crit :

> David,
> 
> Yes it is, it's the last entry in the default dialplan.
> 
> Thankyou,
> 
> Mitch
> 
> ---------- Forwarded message ----------
> From: mitch Johnson <mitch.johnson7 at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Sat, 9 Jul 2011 22:58:52 -0400
> Subject: [Freeswitch-users] One way calling
> I have a phone hanging off a cisco callmanager, the DN is 8000.  I have phones on the FS, DN 1000.  When I dial from the Cisco phone to the Freeswitch phone (8000 to 1000) the phone rings.  However, when I dial from Freeswitch to Cisco Callmanager, the call drops immediately, nothing rings out.
> 
> From the FS_CLI, I see this output:
> 
> 2011-07-09 22:43:59.209255 [NOTICE] switch_channel.c:812 New Channel sofia/internal/1000 at 172.16.200.60 [2657e237-f43d-4574-a7a3-7ef1fab749b8]
> 2011-07-09 22:43:59.215279 [INFO] mod_dialplan_xml.c:331 Processing Test user <1000>->8000 in context public
> 2011-07-09 22:43:59.219276 [NOTICE] switch_ivr.c:1606 Transfer sofia/internal/1000 at 172.16.200.60 to XML[8000 at default]
> 2011-07-09 22:43:59.221278 [INFO] mod_dialplan_xml.c:331 Processing Test user <1000>->8000 in context default
> 2011-07-09 22:43:59.311287 [NOTICE] mod_dptools.c:929 Channel [sofia/internal/1000 at 172.16.200.60] has been answered
> 2011-07-09 22:43:59.625253 [INFO] switch_rtp.c:2972 Auto Changing port from 68.225.42.67:21349 to 192.168.60.142:4000
> 2011-07-09 22:44:00.365253 [NOTICE] switch_core_state_machine.c:189 sofia/internal/1000 at 172.16.200.60 has executed the last dialplan instruction, hanging up.
> 2011-07-09 22:44:00.365253 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/internal/1000 at 172.16.200.60 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-07-09 22:44:00.365253 [NOTICE] switch_core_session.c:1306 Session 7 (sofia/internal/1000 at 172.16.200.60) Ended
> 2011-07-09 22:44:00.365253 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/1000 at 172.16.200.60 [CS_DESTROY]
> 
> 
> I've noticed that it looks as if it's trying to send the call to the Internet, all these calls are internal.  My trunk to the Cisco Callmanager is below:
> 
>  <extension name="Dial to CM8">
>       <condition field="destination_number" expression="^(800\d)$">
>         <action application="bridge" data="sofia/internal/$1 at 172.16.200.100"/>
>       </condition>
>      </extension>
> 
> 
> 
> Below is a successful call from the Callmanager to the Freeswitch:
> 
> 2011-07-09 22:45:56.543784 [NOTICE] switch_channel.c:812 New Channel sofia/internal/8000 at 172.16.200.100 [d06d7795-0c35-4bfe-85bf-9f01498bfafc]
> 2011-07-09 22:45:56.810823 [NOTICE] sofia.c:5285 Channel [sofia/internal/8000 at 172.16.200.100] has been answered
> 2011-07-09 22:45:56.812617 [INFO] mod_dialplan_xml.c:331 Processing 8000 <8000>->1000 in context public
> 2011-07-09 22:45:56.812617 [NOTICE] switch_ivr.c:1606 Transfer sofia/internal/8000 at 172.16.200.100 to XML[1000 at default]
> 2011-07-09 22:45:56.812617 [INFO] mod_dialplan_xml.c:331 Processing 8000 <8000>->1000 in context default
> 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *1 execute_extension::dx XML features
> 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/8000.2011-07-09-22-45-56.wav
> 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *3 execute_extension::cf XML features
> 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *4 execute_extension::att_xfer XML features
> 2011-07-09 22:45:56.828985 [NOTICE] switch_channel.c:812 New Channel sofia/internal/sip:1000 at 192.168.60.142:60936 [846acd6a-3188-4569-9b81-2da9b3e88a87]
> 2011-07-09 22:45:57.010788 [INFO] sofia.c:729 sofia/internal/sip:1000 at 192.168.60.142:60936 Update Callee ID to "Outbound Call" <1000>
> 2011-07-09 22:45:57.010788 [NOTICE] sofia.c:4779 Ring-Ready sofia/internal/sip:1000 at 192.168.60.142:60936!
> 2011-07-09 22:46:03.091789 [NOTICE] sofia.c:5285 Channel [sofia/internal/sip:1000 at 192.168.60.142:60936] has been answered
> 2011-07-09 22:46:03.924748 [INFO] switch_rtp.c:2972 Auto Changing port from 68.225.42.67:22147 to 192.168.60.142:4002
> 2011-07-09 22:46:04.920744 [NOTICE] sofia.c:537 Hangup sofia/internal/sip:1000 at 192.168.60.142:60936 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2011-07-09 22:46:04.924746 [NOTICE] switch_ivr_bridge.c:1328 Hangup sofia/internal/8000 at 172.16.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1306 Session 9 (sofia/internal/sip:1000 at 192.168.60.142:60936) Ended
> 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/sip:1000 at 192.168.60.142:60936 [CS_DESTROY]
> 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1306 Session 8 (sofia/internal/8000 at 172.16.200.100) Ended
> 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/8000 at 172.16.200.100 [CS_DESTROY]
> 
> 
> I have done more extensive debugs and nothing really jumps out at me.
> 
> Any help would be greatly appreciated,
> 
> Thanks,
> 
> Mitch
> 
> 
> ---------- Forwarded message ----------
> From: David Ponzone <david.ponzone at ipeva.fr>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Sun, 10 Jul 2011 13:01:55 +0200
> Subject: Re: [Freeswitch-users] One way calling
> Mitch
> 
>> 
>>  <extension name="Dial to CM8">
>>       <condition field="destination_number" expression="^(800\d)$">
>>         <action application="bridge" data="sofia/internal/$1 at 172.16.200.100"/>
>>       </condition>
>>      </extension>
>> 
> 
> Is this extension in the default dialplan ?
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Jul 10 23:58:19 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 10 Jul 2011 20:58:19 +0100
Subject: [Freeswitch-users] segmentation fault with mod_spandsp
In-Reply-To: <CAL0fNgWbvFn8trf8L0eG2cZ9=xx4XM0oP461iBcdstFOcV-rTA@mail.gmail.com>
References: <CAL0fNgWbvFn8trf8L0eG2cZ9=xx4XM0oP461iBcdstFOcV-rTA@mail.gmail.com>
Message-ID: <CAFiqYunTKgES1CPdkyDQGJsia6WBE_C3eDaOn9UQv5Wfh=nq8A@mail.gmail.com>

Did you get a coredump generated?

-Steve


On 10 July 2011 19:04, Chris Cureau <cmcureau at gmail.com> wrote:

> Hi there!
>
> This morning, I did a fresh git checkout and attempted to build freeswitch
> with mod_freetdm and mod_flite defined in the modules.conf.  I am now
> getting a segfault when compiling:
>
> making all mod_spandsp
> Making all in src
> /bin/bash: line 1: 22453 Segmentation fault      ./make_modem_filter -m
> V.17 -r > v17_v32bis_rx_floating_rrc.h
> make[7]: *** [v17_v32bis_rx_floating_rrc.h] Error 139
> make[6]: *** [all] Error 2
> make[5]: *** [all-recursive] Error 1
> make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la]
> Error 2
> make[3]: *** [mod_spandsp-all] Error 1
> make[2]: *** [all-recursive] Error 1
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
>
> running make_modem_filter manually gets the same result.  Twiddling with
> the commandline options doesn't help either.
>
> OS: Debian 6.0.1a amd64, also a fresh install from last night.
>
> Any idea what is happening?
>
> Thanks in advance!
>
>
> _______________________________________________
> 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 henk at oegema.com  Mon Jul 11 00:22:07 2011
From: henk at oegema.com (Henk Oegema)
Date: Sun, 10 Jul 2011 22:22:07 +0200
Subject: [Freeswitch-users] Fwd: No voice channels to mobile.
In-Reply-To: <1309537772.2523.124.camel@DELL>
References: <1309537772.2523.124.camel@DELL>
Message-ID: <CAJb=zMqutSwqWE-68ASnB3G1JMURFQmGUb5i3FL9xAdQCx5NsQ@mail.gmail.com>

My problem is even more serious than what I was facing earlier (see previous
message)

Due to a defect router (too much rain inside  :-(  I changed to a new one.
but now there is no voice on incoming calls. I can't hear them. but they can
hear me.
The other way around, when I call to a mobile (or landline) eveything is ok.
I was not facing this problem with the old modem.

My port forwarding is:
SIP      5060-5091    TCP&UDP
RTP    16384-32768
STUN  3478-3479

What should I do ??

---------- Forwarded message ----------
From: Henk Oegema <henk at oegema.com>
Date: Fri, Jul 1, 2011 at 6:29 PM
Subject: No voice channels to mobile.
To: freeswitch-users at lists.freeswitch.org


**
I'm facing the following problem:

External calls come in via an ITSP (Localphone) and then routed to internal
extension 1000.
That works OK.

If there's no answer, then the call is routed outbound via ITSP (Powervoip)
to my mobile +316xxxxxxxxx)
In this case there is NO voice communication at all.
(a direct call from 1000 to a mobile is OK)


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

..........................................................................................

..........................................................................................
      <action application="set" data="continue_on_fail=true"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="bridge"
data="{originate_timeout=20}sofia/internal/1000%$${domain}"/>
<----------audio OK
      <action application="bridge"
data="{originate_timeout=20}sofia/gateway/powervoip/003163086xxxx"/>
<------------NO audio both ways
      <action application="voicemail" data="default $${domain} 1000"/>
    </condition>
  </extension>

How do I approach this problem?

Rgds,
Henk
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From cmcureau at gmail.com  Mon Jul 11 00:27:43 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Sun, 10 Jul 2011 15:27:43 -0500
Subject: [Freeswitch-users] segmentation fault with mod_spandsp
In-Reply-To: <CAFiqYunTKgES1CPdkyDQGJsia6WBE_C3eDaOn9UQv5Wfh=nq8A@mail.gmail.com>
References: <CAL0fNgWbvFn8trf8L0eG2cZ9=xx4XM0oP461iBcdstFOcV-rTA@mail.gmail.com>
	<CAFiqYunTKgES1CPdkyDQGJsia6WBE_C3eDaOn9UQv5Wfh=nq8A@mail.gmail.com>
Message-ID: <CAL0fNgUHXMeeWjvKjCFX3JiFhbxooRi9Lo0gth+w0uHVNPiCEQ@mail.gmail.com>

Hi, Steve...thanks for replying!

No core dump...but I did pass it through gdb (without compiling with
debugging symbols)...

GNU gdb (GDB) 7.0.1-debian
Copyright (C) 2009 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html
>
This is free software: you are free to change and redistribute it.
There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
and "show warranty" for details.
This GDB was configured as "x86_64-linux-gnu".
For bug reporting instructions, please see:
<http://www.gnu.org/software/gdb/bugs/>...
Reading symbols from
/usr/local/src/freeswitch/libs/spandsp/src/make_modem_filter...(no debugging
symbols found)...done.
(gdb) set args -m V.17
(gdb) run
Starting program:
/usr/local/src/freeswitch/libs/spandsp/src/make_modem_filter -m V.17

Program received signal SIGSEGV, Segmentation fault.
0x0000000000400d1e in make_rx_filter ()


On Sun, Jul 10, 2011 at 2:58 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Did you get a coredump generated?
>
> -Steve
>
>
> On 10 July 2011 19:04, Chris Cureau <cmcureau at gmail.com> wrote:
>
>> Hi there!
>>
>> This morning, I did a fresh git checkout and attempted to build freeswitch
>> with mod_freetdm and mod_flite defined in the modules.conf.  I am now
>> getting a segfault when compiling:
>>
>> making all mod_spandsp
>> Making all in src
>> /bin/bash: line 1: 22453 Segmentation fault      ./make_modem_filter -m
>> V.17 -r > v17_v32bis_rx_floating_rrc.h
>> make[7]: *** [v17_v32bis_rx_floating_rrc.h] Error 139
>> make[6]: *** [all] Error 2
>> make[5]: *** [all-recursive] Error 1
>> make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la]
>> Error 2
>> make[3]: *** [mod_spandsp-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>> make[1]: *** [all-recursive] Error 1
>> make: *** [all] Error 2
>>
>> running make_modem_filter manually gets the same result.  Twiddling with
>> the commandline options doesn't help either.
>>
>> OS: Debian 6.0.1a amd64, also a fresh install from last night.
>>
>> Any idea what is happening?
>>
>> Thanks in advance!
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 rgelfand2 at gmail.com  Mon Jul 11 00:39:26 2011
From: rgelfand2 at gmail.com (Roman Gelfand)
Date: Sun, 10 Jul 2011 16:39:26 -0400
Subject: [Freeswitch-users] Speech Recognition
In-Reply-To: <1E59A19717CD4D3DB2895DABDF8D7485@dell9400>
References: <CAJbW+rkEuGhEBKxZQyM0z3AZ0zVSA0yneeTB8uZHVicB052fKA@mail.gmail.com>
	<CAMWiMy-fQZSP7VRBdBDumvg8zoW7cDTfmvKpt=vDJqkSdJKGPA@mail.gmail.com>
	<1E59A19717CD4D3DB2895DABDF8D7485@dell9400>
Message-ID: <CAJbW+rkJjORLaM2+zkELLCT9tL0ecF6Ownd0h7k4y5WWpSJ_XQ@mail.gmail.com>

thanks a lot for your help.   I found mod_pocketsphinx.so and mod_unimrcp.so
modules in my freeswitch installation.  Correct me if I am wrong, this is
all I need, in terms of server infrastructure, to write and execute speech
recognition scrips.  Would you know where I could find sample code and steps
to deploy.

Thanks again
On Sun, Jul 10, 2011 at 1:20 PM, Jan Berger <jan.berger at video24.no> wrote:

> You can also use pocketsphinx through MRCP.
>
> I was about to say that the challenge is language models - but, to be
> honest
> this page changed my mind:
>
> http://sourceforge.net/projects/cmusphinx/files/Acoustic%20and%20Language%20
> Models/<http://sourceforge.net/projects/cmusphinx/files/Acoustic%20and%20Language%20Models/>
>
> This is actually far better coverage than a majority of the commercial
> actors that I have queried - and I checked this list for 1 year ago so it's
> quite a change.
>
> Just watch out for license conflicts.
>
> /Jan
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> curriegrad2004
> Sent: 10. juli 2011 18:28
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Speech Recognition
>
> There's pocketsphinx in FreeSwitch.
>
> On Sun, Jul 10, 2011 at 9:21 AM, Roman Gelfand <rgelfand2 at gmail.com>
> wrote:
> > I am looking to script a speech recognition ivr system.  Has anyone used
> > cmusphinx to accomplish?  If not has anyone use a good open source speech
> > recognition engine with freeswitch?
> >
> > Thanks in advance
> > _______________________________________________
> > 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
> >
> >
>
> _______________________________________________
> 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
>
>
> _______________________________________________
> 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 curriegrad2004 at gmail.com  Mon Jul 11 01:06:40 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 10 Jul 2011 14:06:40 -0700
Subject: [Freeswitch-users] Speech Recognition
In-Reply-To: <CAJbW+rkJjORLaM2+zkELLCT9tL0ecF6Ownd0h7k4y5WWpSJ_XQ@mail.gmail.com>
References: <CAJbW+rkEuGhEBKxZQyM0z3AZ0zVSA0yneeTB8uZHVicB052fKA@mail.gmail.com>
	<CAMWiMy-fQZSP7VRBdBDumvg8zoW7cDTfmvKpt=vDJqkSdJKGPA@mail.gmail.com>
	<1E59A19717CD4D3DB2895DABDF8D7485@dell9400>
	<CAJbW+rkJjORLaM2+zkELLCT9tL0ecF6Ownd0h7k4y5WWpSJ_XQ@mail.gmail.com>
Message-ID: <CAMWiMy8tZiGuYga2L5pz2MYwQj2mJALkvDJU_HU+yDiKUw1Rug@mail.gmail.com>

Hopefully the wiki should have examples on this. The Pizza extension
demo actually shows the usage of pocketsphinx pretty well, you should
go check that our first.

On Sun, Jul 10, 2011 at 1:39 PM, Roman Gelfand <rgelfand2 at gmail.com> wrote:
> thanks a lot for your help.?? I found mod_pocketsphinx.so and mod_unimrcp.so
> modules?in my freeswitch installation.? Correct me if I am wrong, this is
> all I need, in terms of server infrastructure,?to write and execute?speech
> recognition?scrips.? Would you know where I could find sample code and steps
> to deploy.
>
> Thanks?again
> On Sun, Jul 10, 2011 at 1:20 PM, Jan Berger <jan.berger at video24.no> wrote:
>>
>> You can also use pocketsphinx through MRCP.
>>
>> I was about to say that the challenge is language models - but, to be
>> honest
>> this page changed my mind:
>>
>> http://sourceforge.net/projects/cmusphinx/files/Acoustic%20and%20Language%20
>> Models/
>>
>> This is actually far better coverage than a majority of the commercial
>> actors that I have queried - and I checked this list for 1 year ago so
>> it's
>> quite a change.
>>
>> Just watch out for license conflicts.
>>
>> /Jan
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> curriegrad2004
>> Sent: 10. juli 2011 18:28
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Speech Recognition
>>
>> There's pocketsphinx in FreeSwitch.
>>
>> On Sun, Jul 10, 2011 at 9:21 AM, Roman Gelfand <rgelfand2 at gmail.com>
>> wrote:
>> > I am looking to script a speech recognition ivr system.? Has anyone used
>> > cmusphinx to accomplish?? If not has anyone use a good open source
>> > speech
>> > recognition engine with freeswitch?
>> >
>> > Thanks in advance
>> > _______________________________________________
>> > 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
>> >
>> >
>>
>> _______________________________________________
>> 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
>>
>>
>> _______________________________________________
>> 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
>
>
> _______________________________________________
> 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 david.ponzone at ipeva.fr  Mon Jul 11 01:42:37 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 10 Jul 2011 23:42:37 +0200
Subject: [Freeswitch-users] One way calling (David Ponzone)
In-Reply-To: <C6408740-C207-49E9-B099-731D94ACCC6E@gmail.com>
References: <mailman.12536.1310320968.24555.freeswitch-users@lists.freeswitch.org>
	<C6408740-C207-49E9-B099-731D94ACCC6E@gmail.com>
Message-ID: <0E33D7A0-0B54-4321-99E7-965ED5F983F1@ipeva.fr>

So your next step is to paste the relevant complete config files to pastebin!

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

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

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




Le 10/07/2011 ? 20:15, Mitch Johnson a ?crit :

> Yes, I did a reloadxml, and I even restarted FreeSWITCH.
> 
> Thanks
> 
> From: David Ponzone <david.ponzone at ipeva.fr>
> Date: July 10, 2011 2:02:13 PM EDT
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] One way calling (David Ponzone)
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> So, sorry for the obvious question, but did you reloadxml ?
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 10/07/2011 ? 17:18, mitch Johnson a ?crit :
> 
>> David,
>> 
>> Yes it is, it's the last entry in the default dialplan.
>> 
>> Thankyou,
>> 
>> Mitch
>> 
>> ---------- Forwarded message ----------
>> From: mitch Johnson <mitch.johnson7 at gmail.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Sat, 9 Jul 2011 22:58:52 -0400
>> Subject: [Freeswitch-users] One way calling
>> I have a phone hanging off a cisco callmanager, the DN is 8000.  I have phones on the FS, DN 1000.  When I dial from the Cisco phone to the Freeswitch phone (8000 to 1000) the phone rings.  However, when I dial from Freeswitch to Cisco Callmanager, the call drops immediately, nothing rings out.
>> 
>> From the FS_CLI, I see this output:
>> 
>> 2011-07-09 22:43:59.209255 [NOTICE] switch_channel.c:812 New Channel sofia/internal/1000 at 172.16.200.60 [2657e237-f43d-4574-a7a3-7ef1fab749b8]
>> 2011-07-09 22:43:59.215279 [INFO] mod_dialplan_xml.c:331 Processing Test user <1000>->8000 in context public
>> 2011-07-09 22:43:59.219276 [NOTICE] switch_ivr.c:1606 Transfer sofia/internal/1000 at 172.16.200.60 to XML[8000 at default]
>> 2011-07-09 22:43:59.221278 [INFO] mod_dialplan_xml.c:331 Processing Test user <1000>->8000 in context default
>> 2011-07-09 22:43:59.311287 [NOTICE] mod_dptools.c:929 Channel [sofia/internal/1000 at 172.16.200.60] has been answered
>> 2011-07-09 22:43:59.625253 [INFO] switch_rtp.c:2972 Auto Changing port from 68.225.42.67:21349 to 192.168.60.142:4000
>> 2011-07-09 22:44:00.365253 [NOTICE] switch_core_state_machine.c:189 sofia/internal/1000 at 172.16.200.60 has executed the last dialplan instruction, hanging up.
>> 2011-07-09 22:44:00.365253 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/internal/1000 at 172.16.200.60 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2011-07-09 22:44:00.365253 [NOTICE] switch_core_session.c:1306 Session 7 (sofia/internal/1000 at 172.16.200.60) Ended
>> 2011-07-09 22:44:00.365253 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/1000 at 172.16.200.60 [CS_DESTROY]
>> 
>> 
>> I've noticed that it looks as if it's trying to send the call to the Internet, all these calls are internal.  My trunk to the Cisco Callmanager is below:
>> 
>>  <extension name="Dial to CM8">
>>       <condition field="destination_number" expression="^(800\d)$">
>>         <action application="bridge" data="sofia/internal/$1 at 172.16.200.100"/>
>>       </condition>
>>      </extension>
>> 
>> 
>> 
>> Below is a successful call from the Callmanager to the Freeswitch:
>> 
>> 2011-07-09 22:45:56.543784 [NOTICE] switch_channel.c:812 New Channel sofia/internal/8000 at 172.16.200.100 [d06d7795-0c35-4bfe-85bf-9f01498bfafc]
>> 2011-07-09 22:45:56.810823 [NOTICE] sofia.c:5285 Channel [sofia/internal/8000 at 172.16.200.100] has been answered
>> 2011-07-09 22:45:56.812617 [INFO] mod_dialplan_xml.c:331 Processing 8000 <8000>->1000 in context public
>> 2011-07-09 22:45:56.812617 [NOTICE] switch_ivr.c:1606 Transfer sofia/internal/8000 at 172.16.200.100 to XML[1000 at default]
>> 2011-07-09 22:45:56.812617 [INFO] mod_dialplan_xml.c:331 Processing 8000 <8000>->1000 in context default
>> 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *1 execute_extension::dx XML features
>> 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/8000.2011-07-09-22-45-56.wav
>> 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *3 execute_extension::cf XML features
>> 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *4 execute_extension::att_xfer XML features
>> 2011-07-09 22:45:56.828985 [NOTICE] switch_channel.c:812 New Channel sofia/internal/sip:1000 at 192.168.60.142:60936 [846acd6a-3188-4569-9b81-2da9b3e88a87]
>> 2011-07-09 22:45:57.010788 [INFO] sofia.c:729 sofia/internal/sip:1000 at 192.168.60.142:60936 Update Callee ID to "Outbound Call" <1000>
>> 2011-07-09 22:45:57.010788 [NOTICE] sofia.c:4779 Ring-Ready sofia/internal/sip:1000 at 192.168.60.142:60936!
>> 2011-07-09 22:46:03.091789 [NOTICE] sofia.c:5285 Channel [sofia/internal/sip:1000 at 192.168.60.142:60936] has been answered
>> 2011-07-09 22:46:03.924748 [INFO] switch_rtp.c:2972 Auto Changing port from 68.225.42.67:22147 to 192.168.60.142:4002
>> 2011-07-09 22:46:04.920744 [NOTICE] sofia.c:537 Hangup sofia/internal/sip:1000 at 192.168.60.142:60936 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>> 2011-07-09 22:46:04.924746 [NOTICE] switch_ivr_bridge.c:1328 Hangup sofia/internal/8000 at 172.16.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1306 Session 9 (sofia/internal/sip:1000 at 192.168.60.142:60936) Ended
>> 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/sip:1000 at 192.168.60.142:60936 [CS_DESTROY]
>> 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1306 Session 8 (sofia/internal/8000 at 172.16.200.100) Ended
>> 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/8000 at 172.16.200.100 [CS_DESTROY]
>> 
>> 
>> I have done more extensive debugs and nothing really jumps out at me.
>> 
>> Any help would be greatly appreciated,
>> 
>> Thanks,
>> 
>> Mitch
>> 
>> 
>> ---------- Forwarded message ----------
>> From: David Ponzone <david.ponzone at ipeva.fr>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Date: Sun, 10 Jul 2011 13:01:55 +0200
>> Subject: Re: [Freeswitch-users] One way calling
>> Mitch
>> 
>>> 
>>>  <extension name="Dial to CM8">
>>>       <condition field="destination_number" expression="^(800\d)$">
>>>         <action application="bridge" data="sofia/internal/$1 at 172.16.200.100"/>
>>>       </condition>
>>>      </extension>
>>> 
>> 
>> Is this extension in the default dialplan ?
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> _______________________________________________
> 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 steveayre at gmail.com  Mon Jul 11 03:05:53 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 11 Jul 2011 00:05:53 +0100
Subject: [Freeswitch-users] Speech Recognition
In-Reply-To: <CAJbW+rkJjORLaM2+zkELLCT9tL0ecF6Ownd0h7k4y5WWpSJ_XQ@mail.gmail.com>
References: <CAJbW+rkEuGhEBKxZQyM0z3AZ0zVSA0yneeTB8uZHVicB052fKA@mail.gmail.com>
	<CAMWiMy-fQZSP7VRBdBDumvg8zoW7cDTfmvKpt=vDJqkSdJKGPA@mail.gmail.com>
	<1E59A19717CD4D3DB2895DABDF8D7485@dell9400>
	<CAJbW+rkJjORLaM2+zkELLCT9tL0ecF6Ownd0h7k4y5WWpSJ_XQ@mail.gmail.com>
Message-ID: <9749A262-279C-40BA-B5C2-65F01FBD50B5@gmail.com>

mod_pocketsphinx lets you use pocketsphinx within the same server.

mod_unimrcp communicates with an external server via the mrcp protocol. That can be running on the same hardware or another.

Steve on iPhone

On 10 Jul 2011, at 21:39, Roman Gelfand <rgelfand2 at gmail.com> wrote:

> thanks a lot for your help.   I found mod_pocketsphinx.so and mod_unimrcp.so modules in my freeswitch installation.  Correct me if I am wrong, this is all I need, in terms of server infrastructure, to write and execute speech recognition scrips.  Would you know where I could find sample code and steps to deploy.
>  
> Thanks again 
> On Sun, Jul 10, 2011 at 1:20 PM, Jan Berger <jan.berger at video24.no> wrote:
> You can also use pocketsphinx through MRCP.
> 
> I was about to say that the challenge is language models - but, to be honest
> this page changed my mind:
> http://sourceforge.net/projects/cmusphinx/files/Acoustic%20and%20Language%20
> Models/
> 
> This is actually far better coverage than a majority of the commercial
> actors that I have queried - and I checked this list for 1 year ago so it's
> quite a change.
> 
> Just watch out for license conflicts.
> 
> /Jan
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> curriegrad2004
> Sent: 10. juli 2011 18:28
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Speech Recognition
> 
> There's pocketsphinx in FreeSwitch.
> 
> On Sun, Jul 10, 2011 at 9:21 AM, Roman Gelfand <rgelfand2 at gmail.com> wrote:
> > I am looking to script a speech recognition ivr system.  Has anyone used
> > cmusphinx to accomplish?  If not has anyone use a good open source speech
> > recognition engine with freeswitch?
> >
> > Thanks in advance
> > _______________________________________________
> > 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
> >
> >
> 
> _______________________________________________
> 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
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> http://www.freeswitch.org
> 
> 
> _______________________________________________
> 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
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From steveayre at gmail.com  Mon Jul 11 03:07:20 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 11 Jul 2011 00:07:20 +0100
Subject: [Freeswitch-users] segmentation fault with mod_spandsp
In-Reply-To: <CAL0fNgUHXMeeWjvKjCFX3JiFhbxooRi9Lo0gth+w0uHVNPiCEQ@mail.gmail.com>
References: <CAL0fNgWbvFn8trf8L0eG2cZ9=xx4XM0oP461iBcdstFOcV-rTA@mail.gmail.com>
	<CAFiqYunTKgES1CPdkyDQGJsia6WBE_C3eDaOn9UQv5Wfh=nq8A@mail.gmail.com>
	<CAL0fNgUHXMeeWjvKjCFX3JiFhbxooRi9Lo0gth+w0uHVNPiCEQ@mail.gmail.com>
Message-ID: <BB0E5DD3-0C85-4874-AA05-A9F1DEB12029@gmail.com>

I run on Debian too. If Steve Underwood doesn't beat me to it I'll try to reproduce it in the morning.

Steve on iPhone

On 10 Jul 2011, at 21:27, Chris Cureau <cmcureau at gmail.com> wrote:

> Hi, Steve...thanks for replying!
> 
> No core dump...but I did pass it through gdb (without compiling with debugging symbols)...
> 
> GNU gdb (GDB) 7.0.1-debian
> Copyright (C) 2009 Free Software Foundation, Inc.
> License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html>
> This is free software: you are free to change and redistribute it.
> There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
> and "show warranty" for details.
> This GDB was configured as "x86_64-linux-gnu".
> For bug reporting instructions, please see:
> <http://www.gnu.org/software/gdb/bugs/>...
> Reading symbols from /usr/local/src/freeswitch/libs/spandsp/src/make_modem_filter...(no debugging symbols found)...done.
> (gdb) set args -m V.17
> (gdb) run
> Starting program: /usr/local/src/freeswitch/libs/spandsp/src/make_modem_filter -m V.17
> 
> Program received signal SIGSEGV, Segmentation fault.
> 0x0000000000400d1e in make_rx_filter ()
> 
> 
> On Sun, Jul 10, 2011 at 2:58 PM, Steven Ayre <steveayre at gmail.com> wrote:
> Did you get a coredump generated?
> 
> -Steve
> 
> 
> On 10 July 2011 19:04, Chris Cureau <cmcureau at gmail.com> wrote:
> Hi there!
> 
> This morning, I did a fresh git checkout and attempted to build freeswitch with mod_freetdm and mod_flite defined in the modules.conf.  I am now getting a segfault when compiling:
> 
> making all mod_spandsp
> Making all in src
> /bin/bash: line 1: 22453 Segmentation fault      ./make_modem_filter -m V.17 -r > v17_v32bis_rx_floating_rrc.h
> make[7]: *** [v17_v32bis_rx_floating_rrc.h] Error 139
> make[6]: *** [all] Error 2
> make[5]: *** [all-recursive] Error 1
> make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la] Error 2
> make[3]: *** [mod_spandsp-all] Error 1
> make[2]: *** [all-recursive] Error 1
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
> 
> running make_modem_filter manually gets the same result.  Twiddling with the commandline options doesn't help either.
> 
> OS: Debian 6.0.1a amd64, also a fresh install from last night.
> 
> Any idea what is happening?
> 
> Thanks in advance!
> 
> 
> _______________________________________________
> 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
> 
> 
> 
> _______________________________________________
> 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
> 
> 
> _______________________________________________
> 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
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From steveayre at gmail.com  Mon Jul 11 03:09:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 11 Jul 2011 00:09:33 +0100
Subject: [Freeswitch-users] Fwd: No voice channels to mobile.
In-Reply-To: <CAJb=zMqutSwqWE-68ASnB3G1JMURFQmGUb5i3FL9xAdQCx5NsQ@mail.gmail.com>
References: <1309537772.2523.124.camel@DELL>
	<CAJb=zMqutSwqWE-68ASnB3G1JMURFQmGUb5i3FL9xAdQCx5NsQ@mail.gmail.com>
Message-ID: <513AD72A-FA7C-4278-896B-1AAE9346816C@gmail.com>

Is there any sip alg feature on the router? Try disabling it if there is.

Also does the router support upnp? If it does then FS can setup the port mapping that way.

Is it FS or the client that's behind nat?

Steve on iPhone

On 10 Jul 2011, at 21:22, Henk Oegema <henk at oegema.com> wrote:

> My problem is even more serious than what I was facing earlier (see previous message)
> 
> Due to a defect router (too much rain inside  :-(  I changed to a new one. but now there is no voice on incoming calls. I can't hear them. but they can hear me.
> The other way around, when I call to a mobile (or landline) eveything is ok.
> I was not facing this problem with the old modem.
> 
> My port forwarding is:
> SIP      5060-5091    TCP&UDP
> RTP    16384-32768
> STUN  3478-3479
> 
> What should I do ??
> 
> ---------- Forwarded message ----------
> From: Henk Oegema <henk at oegema.com>
> Date: Fri, Jul 1, 2011 at 6:29 PM
> Subject: No voice channels to mobile.
> To: freeswitch-users at lists.freeswitch.org
> 
> 
> I'm facing the following problem:
> 
> External calls come in via an ITSP (Localphone) and then routed to internal extension 1000.
> That works OK.
> 
> If there's no answer, then the call is routed outbound via ITSP (Powervoip) to my mobile +316xxxxxxxxx)
> In this case there is NO voice communication at all. 
> (a direct call from 1000 to a mobile is OK)
> 
> 
> <extension name="1000">
>     <condition field="destination_number" expression="^1000$">
>       ..........................................................................................
>       ..........................................................................................
>       <action application="set" data="continue_on_fail=true"/>
>       <action application="set" data="hangup_after_bridge=true"/>
>       <action application="bridge" data="{originate_timeout=20}sofia/internal/1000%$${domain}"/>     <----------audio OK
>       <action application="bridge" data="{originate_timeout=20}sofia/gateway/powervoip/003163086xxxx"/>  <------------NO audio both ways
>       <action application="voicemail" data="default $${domain} 1000"/>
>     </condition>
>   </extension>
> 
> How do I approach this problem?
> 
> Rgds,
> Henk
> 
> _______________________________________________
> 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
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From cmcureau at gmail.com  Mon Jul 11 04:48:24 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Sun, 10 Jul 2011 19:48:24 -0500
Subject: [Freeswitch-users] segmentation fault with mod_spandsp
In-Reply-To: <BB0E5DD3-0C85-4874-AA05-A9F1DEB12029@gmail.com>
References: <CAL0fNgWbvFn8trf8L0eG2cZ9=xx4XM0oP461iBcdstFOcV-rTA@mail.gmail.com>
	<CAFiqYunTKgES1CPdkyDQGJsia6WBE_C3eDaOn9UQv5Wfh=nq8A@mail.gmail.com>
	<CAL0fNgUHXMeeWjvKjCFX3JiFhbxooRi9Lo0gth+w0uHVNPiCEQ@mail.gmail.com>
	<BB0E5DD3-0C85-4874-AA05-A9F1DEB12029@gmail.com>
Message-ID: <CAL0fNgVFm6G3ko2QLLpj59827Rs6cCn9YKBHrUbnL65tncOTrA@mail.gmail.com>

I tried to send a core dump earlier on the list, but I can't see it in the
archive for some reason...if anyone wants the core, let me know.

Otherwise, this is the text of the previous message:

I don't claim to be an expert at gdb at all, but after compiling with
debugging symbols here's what I got:

GNU gdb (GDB) 7.0.1-debian
Copyright (C) 2009 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html
>
This is free software: you are free to change and redistribute it.
There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
and "show warranty" for details.
This GDB was configured as "x86_64-linux-gnu".
For bug reporting instructions, please see:
<http://www.gnu.org/software/gdb/bugs/>...
Reading symbols from
/usr/local/src/freeswitch/libs/spandsp/src/make_modem_filter...done.
(gdb) set args -m V.17 -i -r
(gdb) break 470
Breakpoint 1 at 0x4016cc: file ../src/make_modem_filter.c, line 470.
(gdb) run
Starting program:
/usr/local/src/freeswitch/libs/spandsp/src/make_modem_filter -m V.17 -i -r

Breakpoint 1, main (argc=5, argv=0x7fffffffebf8) at
../src/make_modem_filter.c:470
470         make_rx_filter(rx_coeff_sets,
(gdb) print rx_coeff_sets
$1 = 192
(gdb) print rx_coeffs_per_filter
$2 = 27
(gdb) print carrier
$3 = 1800
(gdb) print baud_rate
$4 = 2400
(gdb) print rx_excess_bandwidth
$5 = 0.5
(gdb) print fixed_point
$6 = 1
(gdb) print rx_tag
$7 = 0x4026bf ""
(gdb) step

Program received signal SIGSEGV, Segmentation fault.
0x0000000000400d1e in make_rx_filter (coeff_sets=Cannot access memory at
address 0x7ffffff9e62c
) at ../src/make_modem_filter.c:163
163 {
(gdb)

On Sun, Jul 10, 2011 at 6:07 PM, Steven Ayre <steveayre at gmail.com> wrote:

> I run on Debian too. If Steve Underwood doesn't beat me to it I'll try to
> reproduce it in the morning.
>
> Steve on iPhone
>
> On 10 Jul 2011, at 21:27, Chris Cureau <cmcureau at gmail.com> wrote:
>
> Hi, Steve...thanks for replying!
>
> No core dump...but I did pass it through gdb (without compiling with
> debugging symbols)...
>
> GNU gdb (GDB) 7.0.1-debian
> Copyright (C) 2009 Free Software Foundation, Inc.
> License GPLv3+: GNU GPL version 3 or later <<http://gnu.org/licenses/gpl.html>
> http://gnu.org/licenses/gpl.html>
> This is free software: you are free to change and redistribute it.
> There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
> and "show warranty" for details.
> This GDB was configured as "x86_64-linux-gnu".
> For bug reporting instructions, please see:
> < <http://www.gnu.org/software/gdb/bugs/>
> http://www.gnu.org/software/gdb/bugs/>...
> Reading symbols from
> /usr/local/src/freeswitch/libs/spandsp/src/make_modem_filter...(no debugging
> symbols found)...done.
> (gdb) set args -m V.17
> (gdb) run
> Starting program:
> /usr/local/src/freeswitch/libs/spandsp/src/make_modem_filter -m V.17
>
> Program received signal SIGSEGV, Segmentation fault.
> 0x0000000000400d1e in make_rx_filter ()
>
>
> On Sun, Jul 10, 2011 at 2:58 PM, Steven Ayre < <steveayre at gmail.com>
> steveayre at gmail.com> wrote:
>
>> Did you get a coredump generated?
>>
>> -Steve
>>
>>
>> On 10 July 2011 19:04, Chris Cureau < <cmcureau at gmail.com>
>> cmcureau at gmail.com> wrote:
>>
>>> Hi there!
>>>
>>> This morning, I did a fresh git checkout and attempted to build
>>> freeswitch with mod_freetdm and mod_flite defined in the modules.conf.  I am
>>> now getting a segfault when compiling:
>>>
>>> making all mod_spandsp
>>> Making all in src
>>> /bin/bash: line 1: 22453 Segmentation fault      ./make_modem_filter -m
>>> V.17 -r > v17_v32bis_rx_floating_rrc.h
>>> make[7]: *** [v17_v32bis_rx_floating_rrc.h] Error 139
>>> make[6]: *** [all] Error 2
>>> make[5]: *** [all-recursive] Error 1
>>> make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la]
>>> Error 2
>>> make[3]: *** [mod_spandsp-all] Error 1
>>> make[2]: *** [all-recursive] Error 1
>>> make[1]: *** [all-recursive] Error 1
>>> make: *** [all] Error 2
>>>
>>> running make_modem_filter manually gets the same result.  Twiddling with
>>> the commandline options doesn't help either.
>>>
>>> OS: Debian 6.0.1a amd64, also a fresh install from last night.
>>>
>>> Any idea what is happening?
>>>
>>> Thanks in advance!
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>  <http://www.cluecon.com>http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>  <http://www.cluecon.com>http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>>
> _______________________________________________
> 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
>
>
> _______________________________________________
> 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
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>
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From fieldpeak at gmail.com  Mon Jul 11 06:25:38 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Mon, 11 Jul 2011 10:25:38 +0800
Subject: [Freeswitch-users] How to configure for FS playing voice prompt in
 case the called extension is in a call.
Message-ID: <CAOTmbMzMVtRjUvTj5UgHH7MG=XyHSWRvw7QT6G-tiLFd57H1Vw@mail.gmail.com>

Hi Gurus,

Could anyone advise how to realize have FS to play a voice prompt e.g. 'the
extension you dialed is busy now, please dial the other exsention' to
replacing busy tone when the called extension is in a call...

the details is below,

When calling to FS, the FS will play IVR "welcome to call us, please input
the extension number, for operator please press 0",
then the caller input the extension number, in case the extension is in a
call, the caller will hear busy tone...
it needs the system play "the extension you dailed is busy, please dial the
other extension", currently my dial plan as following, could anyone advise
how to change based on below dial plan or any other new dail plan can
realize it...

Thanks a lot!

    <extension name="Local_Extension">
      <condition field="destination_number" expression="^(1\d{3})$">
    <action application="set" data="dialed_extension=$1"/>
    <action application="export" data="dialed_extension=$1"/>
    <action application="set" data="ringback=${us-ring}"/>
    <action application="set" data="transfer_ringback=$${hold_music}"/>
    <action application="set" data="call_timeout=30"/>
    <action application="set" data="hangup_after_bridge=true"/>

    <!-- to enable voice mail, set continue_on_fail=true below -->
    <action application="set" data="continue_on_fail=false"/>
    <action application="hash"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
    <action application="hash"
data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
    <action application="bridge"
data="sofia/internal/${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>


    <extension name="IVR" >
       <condition field="destination_number" expression="^888$">
           <action application="answer"/>
           <action application="sleep" data="2000"/>
           <action application="ivr" data="IVR"/>
       </condition>
    </extension>

        <!--  -->
    <menu name="IVR"
        greet-long="C:/VSWITCH/recordings/greeting_tts.wav"
        greet-short="C:/VSWITCH/recordings/greeting_tts.wav"
        invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
        exit-sound="voicemail/vm-goodbye.wav"
        confirm-macro=""
        confirm-key=""
        tts-engine="flite"
        tts-voice="rms"
        confirm-attempts="3"
        timeout="10000"
        inter-digit-timeout="2000"
        max-failures="3"
        max-timeouts="3"
        digit-len="5">
        <entry action="menu-exec-app" digits="/(^\*\d{3,5}$|^\d{3,5}$)/"
param="transfer $1 XML default"/>
    </menu>

Regards,
Charles
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From cmcureau at gmail.com  Mon Jul 11 07:49:33 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Sun, 10 Jul 2011 22:49:33 -0500
Subject: [Freeswitch-users] segmentation fault with mod_spandsp
In-Reply-To: <CAL0fNgWbvFn8trf8L0eG2cZ9=xx4XM0oP461iBcdstFOcV-rTA@mail.gmail.com>
References: <CAL0fNgWbvFn8trf8L0eG2cZ9=xx4XM0oP461iBcdstFOcV-rTA@mail.gmail.com>
Message-ID: <CAL0fNgXbRyQyPcG+fz4KYDjacECGfZXqwxk5-Jev_aZOPN7ycg@mail.gmail.com>

Okay...I think I found the problem...quite by accident, though.

I was compiling in a terminal that I had earlier executed 'ulimit -s 240'
in.  Apparently there wasn't enough stack to run the program!  Running with
a different terminal works fine.  Sorry to generate useless chatter!

Cheers,
Chris

On Sun, Jul 10, 2011 at 1:04 PM, Chris Cureau <cmcureau at gmail.com> wrote:

> Hi there!
>
> This morning, I did a fresh git checkout and attempted to build freeswitch
> with mod_freetdm and mod_flite defined in the modules.conf.  I am now
> getting a segfault when compiling:
>
> making all mod_spandsp
> Making all in src
> /bin/bash: line 1: 22453 Segmentation fault      ./make_modem_filter -m
> V.17 -r > v17_v32bis_rx_floating_rrc.h
> make[7]: *** [v17_v32bis_rx_floating_rrc.h] Error 139
> make[6]: *** [all] Error 2
> make[5]: *** [all-recursive] Error 1
> make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la]
> Error 2
> make[3]: *** [mod_spandsp-all] Error 1
> make[2]: *** [all-recursive] Error 1
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
>
> running make_modem_filter manually gets the same result.  Twiddling with
> the commandline options doesn't help either.
>
> OS: Debian 6.0.1a amd64, also a fresh install from last night.
>
> Any idea what is happening?
>
> Thanks in advance!
>
>
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From infos at madovsky.org  Mon Jul 11 07:52:30 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 10 Jul 2011 23:52:30 -0400
Subject: [Freeswitch-users] live messenger RevEng
Message-ID: <6AAB0FDA6D3C4237AFEE86D3F53E9843@e1705>

for the geeks who d like to create a mod_live_messenger

http://msnwcrec.arrozcru.org/vc_1_1.pdf

http://www.codeproject.com/KB/macros/wlmplugin.aspx

http://msnwcrec.arrozcru.org/

http://imfreedom.org/wiki/MSN
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From jaybinks at gmail.com  Mon Jul 11 08:15:37 2011
From: jaybinks at gmail.com (jay binks)
Date: Mon, 11 Jul 2011 14:15:37 +1000
Subject: [Freeswitch-users] segmentation fault with mod_spandsp
In-Reply-To: <CAL0fNgVFm6G3ko2QLLpj59827Rs6cCn9YKBHrUbnL65tncOTrA@mail.gmail.com>
References: <CAL0fNgWbvFn8trf8L0eG2cZ9=xx4XM0oP461iBcdstFOcV-rTA@mail.gmail.com>
	<CAFiqYunTKgES1CPdkyDQGJsia6WBE_C3eDaOn9UQv5Wfh=nq8A@mail.gmail.com>
	<CAL0fNgUHXMeeWjvKjCFX3JiFhbxooRi9Lo0gth+w0uHVNPiCEQ@mail.gmail.com>
	<BB0E5DD3-0C85-4874-AA05-A9F1DEB12029@gmail.com>
	<CAL0fNgVFm6G3ko2QLLpj59827Rs6cCn9YKBHrUbnL65tncOTrA@mail.gmail.com>
Message-ID: <CAO47HpfwG-2v+gGvB95unmGJxMZ+exno-=ACFTnkSJa-A9GsZw@mail.gmail.com>

the list is ALWAYS the wrong place to report bugs / send core dumps..

if this is a bug that you can reproduce.
create a JIRA for it, and upload the core there..
Also be sure to include steps to reproduce etc.

then if it needs further discussion, discuss it on the list with the link to
the jira
for others to look at..




On Mon, Jul 11, 2011 at 10:48 AM, Chris Cureau <cmcureau at gmail.com> wrote:

> I tried to send a core dump earlier on the list, but I can't see it in the
> archive for some reason...if anyone wants the core, let me know.
>
> Otherwise, this is the text of the previous message:
>
> I don't claim to be an expert at gdb at all, but after compiling with
> debugging symbols here's what I got:
>
> GNU gdb (GDB) 7.0.1-debian
> Copyright (C) 2009 Free Software Foundation, Inc.
> License GPLv3+: GNU GPL version 3 or later <
> http://gnu.org/licenses/gpl.html>
> This is free software: you are free to change and redistribute it.
> There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
> and "show warranty" for details.
> This GDB was configured as "x86_64-linux-gnu".
> For bug reporting instructions, please see:
> <http://www.gnu.org/software/gdb/bugs/>...
> Reading symbols from
> /usr/local/src/freeswitch/libs/spandsp/src/make_modem_filter...done.
> (gdb) set args -m V.17 -i -r
> (gdb) break 470
> Breakpoint 1 at 0x4016cc: file ../src/make_modem_filter.c, line 470.
> (gdb) run
> Starting program:
> /usr/local/src/freeswitch/libs/spandsp/src/make_modem_filter -m V.17 -i -r
>
> Breakpoint 1, main (argc=5, argv=0x7fffffffebf8) at
> ../src/make_modem_filter.c:470
> 470         make_rx_filter(rx_coeff_sets,
> (gdb) print rx_coeff_sets
> $1 = 192
> (gdb) print rx_coeffs_per_filter
> $2 = 27
> (gdb) print carrier
> $3 = 1800
> (gdb) print baud_rate
> $4 = 2400
> (gdb) print rx_excess_bandwidth
> $5 = 0.5
> (gdb) print fixed_point
> $6 = 1
> (gdb) print rx_tag
> $7 = 0x4026bf ""
> (gdb) step
>
> Program received signal SIGSEGV, Segmentation fault.
> 0x0000000000400d1e in make_rx_filter (coeff_sets=Cannot access memory at
> address 0x7ffffff9e62c
> ) at ../src/make_modem_filter.c:163
> 163 {
> (gdb)
>
> On Sun, Jul 10, 2011 at 6:07 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> I run on Debian too. If Steve Underwood doesn't beat me to it I'll try to
>> reproduce it in the morning.
>>
>> Steve on iPhone
>>
>> On 10 Jul 2011, at 21:27, Chris Cureau <cmcureau at gmail.com> wrote:
>>
>> Hi, Steve...thanks for replying!
>>
>> No core dump...but I did pass it through gdb (without compiling with
>> debugging symbols)...
>>
>> GNU gdb (GDB) 7.0.1-debian
>> Copyright (C) 2009 Free Software Foundation, Inc.
>> License GPLv3+: GNU GPL version 3 or later <<http://gnu.org/licenses/gpl.html>
>> http://gnu.org/licenses/gpl.html>
>> This is free software: you are free to change and redistribute it.
>> There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
>> and "show warranty" for details.
>> This GDB was configured as "x86_64-linux-gnu".
>> For bug reporting instructions, please see:
>> < <http://www.gnu.org/software/gdb/bugs/>
>> http://www.gnu.org/software/gdb/bugs/>...
>> Reading symbols from
>> /usr/local/src/freeswitch/libs/spandsp/src/make_modem_filter...(no debugging
>> symbols found)...done.
>> (gdb) set args -m V.17
>> (gdb) run
>> Starting program:
>> /usr/local/src/freeswitch/libs/spandsp/src/make_modem_filter -m V.17
>>
>> Program received signal SIGSEGV, Segmentation fault.
>> 0x0000000000400d1e in make_rx_filter ()
>>
>>
>> On Sun, Jul 10, 2011 at 2:58 PM, Steven Ayre < <steveayre at gmail.com>
>> steveayre at gmail.com> wrote:
>>
>>> Did you get a coredump generated?
>>>
>>> -Steve
>>>
>>>
>>> On 10 July 2011 19:04, Chris Cureau < <cmcureau at gmail.com>
>>> cmcureau at gmail.com> wrote:
>>>
>>>> Hi there!
>>>>
>>>> This morning, I did a fresh git checkout and attempted to build
>>>> freeswitch with mod_freetdm and mod_flite defined in the modules.conf.  I am
>>>> now getting a segfault when compiling:
>>>>
>>>> making all mod_spandsp
>>>> Making all in src
>>>> /bin/bash: line 1: 22453 Segmentation fault      ./make_modem_filter -m
>>>> V.17 -r > v17_v32bis_rx_floating_rrc.h
>>>> make[7]: *** [v17_v32bis_rx_floating_rrc.h] Error 139
>>>> make[6]: *** [all] Error 2
>>>> make[5]: *** [all-recursive] Error 1
>>>> make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la]
>>>> Error 2
>>>> make[3]: *** [mod_spandsp-all] Error 1
>>>> make[2]: *** [all-recursive] Error 1
>>>> make[1]: *** [all-recursive] Error 1
>>>> make: *** [all] Error 2
>>>>
>>>> running make_modem_filter manually gets the same result.  Twiddling with
>>>> the commandline options doesn't help either.
>>>>
>>>> OS: Debian 6.0.1a amd64, also a fresh install from last night.
>>>>
>>>> Any idea what is happening?
>>>>
>>>> Thanks in advance!
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>  <http://www.cluecon.com>http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>  <http://www.cluecon.com>http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> 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
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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
>
>


-- 
Sincerely

Jay
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From steveayre at gmail.com  Mon Jul 11 12:13:10 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 11 Jul 2011 09:13:10 +0100
Subject: [Freeswitch-users] How to configure for FS playing voice prompt
 in case the called extension is in a call.
In-Reply-To: <CAOTmbMzMVtRjUvTj5UgHH7MG=XyHSWRvw7QT6G-tiLFd57H1Vw@mail.gmail.com>
References: <CAOTmbMzMVtRjUvTj5UgHH7MG=XyHSWRvw7QT6G-tiLFd57H1Vw@mail.gmail.com>
Message-ID: <CAFiqYuncFULQn=BnWDh2s0m6VhAX3aqN-zPbFaMD19cowfOERA@mail.gmail.com>

There are two ways....

Most phones will return a USER_BUSY release code if you bridge to a phone
that's busy. If the phone does then you can continue_on_fail so that you
(pre-)answer the call and play that message after the bridge. Something
like:

<extension name="Local_Extension">
    <condition field="destination_number" expression="^(1\d{3})$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="set" data="call_timeout=30"/>
        <action application="set" data="hangup_after_bridge=true"/> <!--
Hangup on successful call -->
        <action application="set" data="continue_on_fail=true"/> <!-- Don't
hangup on failed call, continue to IVR prompt -->
        <action application="hash"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
        <action application="bridge"
data="user/${dialed_extension}%${domain_name}"/>
        <!-- Put the IVR prompt here, see
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits -->
    </condition>
</extension>

The 2nd option is that you use Limit to determine if another person is
already on a call to the phone. That only really works if you run the the
only servers that call the phone since it requires tracking all calls to the
phone. It will sometimes work better though - especially if the phone has
multiple lines. http://wiki.freeswitch.org/wiki/Limit

-Steve


On 11 July 2011 03:25, fieldpeak <fieldpeak at gmail.com> wrote:

> Hi Gurus,
>
> Could anyone advise how to realize have FS to play a voice prompt e.g. 'the
> extension you dialed is busy now, please dial the other exsention' to
> replacing busy tone when the called extension is in a call...
>
> the details is below,
>
> When calling to FS, the FS will play IVR "welcome to call us, please input
> the extension number, for operator please press 0",
> then the caller input the extension number, in case the extension is in a
> call, the caller will hear busy tone...
> it needs the system play "the extension you dailed is busy, please dial the
> other extension", currently my dial plan as following, could anyone advise
> how to change based on below dial plan or any other new dail plan can
> realize it...
>
> Thanks a lot!
>
>     <extension name="Local_Extension">
>       <condition field="destination_number" expression="^(1\d{3})$">
>     <action application="set" data="dialed_extension=$1"/>
>     <action application="export" data="dialed_extension=$1"/>
>     <action application="set" data="ringback=${us-ring}"/>
>     <action application="set" data="transfer_ringback=$${hold_music}"/>
>     <action application="set" data="call_timeout=30"/>
>     <action application="set" data="hangup_after_bridge=true"/>
>
>     <!-- to enable voice mail, set continue_on_fail=true below -->
>     <action application="set" data="continue_on_fail=false"/>
>     <action application="hash"
> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>     <action application="hash"
> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>     <action application="bridge"
> data="sofia/internal/${dialed_extension}%${domain_name}"/>
>     <action application="answer"/>
>
>     <action application="sleep" data="1000"/>
>     <action application="bridge" data="loopback/app=voicemail:default
> ${domain_name} ${dialed_extension}"/>
>       </condition>
>     </extension>
>
>
>     <extension name="IVR" >
>        <condition field="destination_number" expression="^888$">
>            <action application="answer"/>
>            <action application="sleep" data="2000"/>
>            <action application="ivr" data="IVR"/>
>        </condition>
>     </extension>
>
>         <!--  -->
>     <menu name="IVR"
>         greet-long="C:/VSWITCH/recordings/greeting_tts.wav"
>         greet-short="C:/VSWITCH/recordings/greeting_tts.wav"
>         invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>         exit-sound="voicemail/vm-goodbye.wav"
>         confirm-macro=""
>         confirm-key=""
>         tts-engine="flite"
>         tts-voice="rms"
>         confirm-attempts="3"
>         timeout="10000"
>         inter-digit-timeout="2000"
>         max-failures="3"
>         max-timeouts="3"
>         digit-len="5">
>         <entry action="menu-exec-app" digits="/(^\*\d{3,5}$|^\d{3,5}$)/"
> param="transfer $1 XML default"/>
>     </menu>
>
> Regards,
> Charles
>
> _______________________________________________
> 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 tomasz at kopacki.eu  Mon Jul 11 12:30:10 2011
From: tomasz at kopacki.eu (Tomasz Kopacki)
Date: Mon, 11 Jul 2011 08:30:10 +0000
Subject: [Freeswitch-users] domain_name variable
Message-ID: <443EC226AAEABB48B58CAF9D56D80AB4A89C88@hektor.dom.local>

Hi,
I've got a problem with device variable - user_context.
In my scenario, context name equals domain name, so i want to use variable to set user_context.
I've tried 'user_context=${domain_name}' but it doesnt work, every call goes to public context.
How to do this properly ?

From steveayre at gmail.com  Mon Jul 11 13:07:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 11 Jul 2011 10:07:01 +0100
Subject: [Freeswitch-users] domain_name variable
In-Reply-To: <443EC226AAEABB48B58CAF9D56D80AB4A89C88@hektor.dom.local>
References: <443EC226AAEABB48B58CAF9D56D80AB4A89C88@hektor.dom.local>
Message-ID: <CAFiqYunYA-OrX4V=r_N8cMBp_h1Ef_C5M5sVCBUeBoAAQoQkLw@mail.gmail.com>

Where are you setting the user_context? In the user directory?
Are you sure the calls are being authenticated properly?

-Steve


On 11 July 2011 09:30, Tomasz Kopacki <tomasz at kopacki.eu> wrote:

> Hi,
> I've got a problem with device variable - user_context.
> In my scenario, context name equals domain name, so i want to use variable
> to set user_context.
> I've tried 'user_context=${domain_name}' but it doesnt work, every call
> goes to public context.
> How to do this properly ?
> _______________________________________________
> 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 steveayre at gmail.com  Mon Jul 11 13:07:40 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 11 Jul 2011 10:07:40 +0100
Subject: [Freeswitch-users] segmentation fault with mod_spandsp
In-Reply-To: <CAL0fNgXbRyQyPcG+fz4KYDjacECGfZXqwxk5-Jev_aZOPN7ycg@mail.gmail.com>
References: <CAL0fNgWbvFn8trf8L0eG2cZ9=xx4XM0oP461iBcdstFOcV-rTA@mail.gmail.com>
	<CAL0fNgXbRyQyPcG+fz4KYDjacECGfZXqwxk5-Jev_aZOPN7ycg@mail.gmail.com>
Message-ID: <CAFiqYu=qenxmYLy-w9K2bzQMT6sYTivcGcT9Pw-K_yCoFmmzfQ@mail.gmail.com>

Thanks for that observation - useful to know.

-Steve



On 11 July 2011 04:49, Chris Cureau <cmcureau at gmail.com> wrote:

> Okay...I think I found the problem...quite by accident, though.
>
> I was compiling in a terminal that I had earlier executed 'ulimit -s 240'
> in.  Apparently there wasn't enough stack to run the program!  Running with
> a different terminal works fine.  Sorry to generate useless chatter!
>
> Cheers,
> Chris
>
>
> On Sun, Jul 10, 2011 at 1:04 PM, Chris Cureau <cmcureau at gmail.com> wrote:
>
>> Hi there!
>>
>> This morning, I did a fresh git checkout and attempted to build freeswitch
>> with mod_freetdm and mod_flite defined in the modules.conf.  I am now
>> getting a segfault when compiling:
>>
>> making all mod_spandsp
>> Making all in src
>> /bin/bash: line 1: 22453 Segmentation fault      ./make_modem_filter -m
>> V.17 -r > v17_v32bis_rx_floating_rrc.h
>> make[7]: *** [v17_v32bis_rx_floating_rrc.h] Error 139
>> make[6]: *** [all] Error 2
>> make[5]: *** [all-recursive] Error 1
>> make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la]
>> Error 2
>> make[3]: *** [mod_spandsp-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>> make[1]: *** [all-recursive] Error 1
>> make: *** [all] Error 2
>>
>> running make_modem_filter manually gets the same result.  Twiddling with
>> the commandline options doesn't help either.
>>
>> OS: Debian 6.0.1a amd64, also a fresh install from last night.
>>
>> Any idea what is happening?
>>
>> Thanks in advance!
>>
>>
>
> _______________________________________________
> 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 boris at tagnet.ru  Mon Jul 11 13:12:07 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Mon, 11 Jul 2011 15:12:07 +0600
Subject: [Freeswitch-users] domain_name variable
In-Reply-To: <443EC226AAEABB48B58CAF9D56D80AB4A89C88@hektor.dom.local>
References: <443EC226AAEABB48B58CAF9D56D80AB4A89C88@hektor.dom.local>
Message-ID: <4E1ABE67.5010808@tagnet.ru>

Hello!

     Where have You set it? This variable must be defined in user 
directory or You should do something like:
<action application='transfer' data='${destination_number} XML 
${domain_name}"/>

> Hi,
> I've got a problem with device variable - user_context.
> In my scenario, context name equals domain name, so i want to use variable to set user_context.
> I've tried 'user_context=${domain_name}' but it doesnt work, every call goes to public context.
> How to do this properly ?
> _______________________________________________
> 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


-- 
Regards,
Boris




From tomasz at kopacki.eu  Mon Jul 11 13:19:42 2011
From: tomasz at kopacki.eu (Tomasz Kopacki)
Date: Mon, 11 Jul 2011 09:19:42 +0000
Subject: [Freeswitch-users] ODP:  domain_name variable
In-Reply-To: <CAFiqYunYA-OrX4V=r_N8cMBp_h1Ef_C5M5sVCBUeBoAAQoQkLw@mail.gmail.com>
References: <443EC226AAEABB48B58CAF9D56D80AB4A89C88@hektor.dom.local>,
	<CAFiqYunYA-OrX4V=r_N8cMBp_h1Ef_C5M5sVCBUeBoAAQoQkLw@mail.gmail.com>
Message-ID: <443EC226AAEABB48B58CAF9D56D80AB4A89CD7@hektor.dom.local>

Yes, I set this in directory file. It works when i write down the domain name, but i want to use var.
With this config, all calls are going through 'public' context

<variables>
	<variable name="toll_allow" value="domestic,international,local" />
	<variable name="accountcode" value="1001" />
	<variable name="callgroup" value="default" />
	<variable name="outbound_caller_id_name" value="..." />
	<variable name="effective_caller_id_name" value="..." />
	<variable name="effective_caller_id_numer" value="..." />
	<variable name="outbound_caller_id_numer" value="..." />
        ...
	<variable name="user_context" value="${domain_name}" />
</variables>


________________________________________
Od: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] w imieniu Steven Ayre [steveayre at gmail.com]
Wys?ano: 11 lipca 2011 11:07
Do: FreeSWITCH Users Help
Temat: Re: [Freeswitch-users] domain_name variable

Where are you setting the user_context? In the user directory?
Are you sure the calls are being authenticated properly?

-Steve


On 11 July 2011 09:30, Tomasz Kopacki <tomasz at kopacki.eu<mailto:tomasz at kopacki.eu>> wrote:
Hi,
I've got a problem with device variable - user_context.
In my scenario, context name equals domain name, so i want to use variable to set user_context.
I've tried 'user_context=${domain_name}' but it doesnt work, every call goes to public context.
How to do this properly ?
_______________________________________________
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<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



From steveayre at gmail.com  Mon Jul 11 14:10:21 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 11 Jul 2011 11:10:21 +0100
Subject: [Freeswitch-users] ODP: domain_name variable
In-Reply-To: <443EC226AAEABB48B58CAF9D56D80AB4A89CD7@hektor.dom.local>
References: <443EC226AAEABB48B58CAF9D56D80AB4A89C88@hektor.dom.local>
	<CAFiqYunYA-OrX4V=r_N8cMBp_h1Ef_C5M5sVCBUeBoAAQoQkLw@mail.gmail.com>
	<443EC226AAEABB48B58CAF9D56D80AB4A89CD7@hektor.dom.local>
Message-ID: <CAFiqYu=gKCVVjCMLhVsnejvpG03W8b1t8hgikATmeXQ63geBog@mail.gmail.com>

That domain name variable won't be set there yet. You'll have to hardcode it
into the directory entry (or use mod_xml_curl where your program can
automatically put the same domain as that of the request).

-Steve



2011/7/11 Tomasz Kopacki <tomasz at kopacki.eu>

> Yes, I set this in directory file. It works when i write down the domain
> name, but i want to use var.
> With this config, all calls are going through 'public' context
>
> <variables>
>        <variable name="toll_allow" value="domestic,international,local" />
>        <variable name="accountcode" value="1001" />
>        <variable name="callgroup" value="default" />
>        <variable name="outbound_caller_id_name" value="..." />
>        <variable name="effective_caller_id_name" value="..." />
>        <variable name="effective_caller_id_numer" value="..." />
>        <variable name="outbound_caller_id_numer" value="..." />
>        ...
>        <variable name="user_context" value="${domain_name}" />
> </variables>
>
>
> ________________________________________
> Od: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] w imieniu Steven Ayre [
> steveayre at gmail.com]
> Wys?ano: 11 lipca 2011 11:07
> Do: FreeSWITCH Users Help
> Temat: Re: [Freeswitch-users] domain_name variable
>
> Where are you setting the user_context? In the user directory?
> Are you sure the calls are being authenticated properly?
>
> -Steve
>
>
> On 11 July 2011 09:30, Tomasz Kopacki <tomasz at kopacki.eu<mailto:
> tomasz at kopacki.eu>> wrote:
> Hi,
> I've got a problem with device variable - user_context.
> In my scenario, context name equals domain name, so i want to use variable
> to set user_context.
> I've tried 'user_context=${domain_name}' but it doesnt work, every call
> goes to public context.
> How to do this properly ?
> _______________________________________________
> 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<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
>
>
> _______________________________________________
> 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 lautram.mathieu at gmail.com  Mon Jul 11 16:08:26 2011
From: lautram.mathieu at gmail.com (Mathieu Lautram)
Date: Mon, 11 Jul 2011 14:08:26 +0200
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAKzWOxWiLpP6j=JXQE=oM+t6QF_fy0e30DCC5EO=9qGf+OFmgg@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
	<CAKbxfG-h2_seFcYhu-5zwDbhD-z12xGuw7GmmKPPgSLGux3_zg@mail.gmail.com>
	<CAMNz+bsev6pqGc4u8YxhPvsFtzcpBC8R0W6frW52BtaGbJidJQ@mail.gmail.com>
	<CAKzWOxWiLpP6j=JXQE=oM+t6QF_fy0e30DCC5EO=9qGf+OFmgg@mail.gmail.com>
Message-ID: <CAMNz+bvn+L-87UjFLYn=HX5S_FJ0cLWrwfKVrEpa-UDXk04Etg@mail.gmail.com>

In fact, I want to know when the phone of B leg is effectively ringing. If I
don't ignore early media, FreeSWITCH will consider the bridge successful
even if the phone of B leg is not ringing. So I have to ignore early media
to know that. But, if I set ignore_early_media to true, A leg will not hear
any ringtone.

2011/7/8 Michael Collins <msc at freeswitch.org>

>
>
> On Fri, Jul 8, 2011 at 4:29 AM, Mathieu Lautram <lautram.mathieu at gmail.com
> > wrote:
>
>> bridge_early_media is not really appropriate for my issue and I have
>> already tried this parameter.
>> In fact, I would like to let A leg hear a ringtone(and sometimes vocal
>> informations) when the phone of the B leg is *effectively* ringing.
>> Do you think it is possible? Maybe ignoring early media is not the
>> solution to my problem?
>>
>
> Honestly, I've not yet heard a valid reason for ignoring early media at all
> in your scenario. Under what conditions do you *not* want early media from
> the B leg sent to the A leg?
>
> -MC
>
> _______________________________________________
> 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
>
>


-- 
Mathieu LAUTRAM
Application developer

BJT Partners - FRANCE
+33 1 79 75 99 60
+33 6 61 59 07 25
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From ian at medicalit.com.au  Sun Jul 10 16:17:22 2011
From: ian at medicalit.com.au (Ian Yates)
Date: Sun, 10 Jul 2011 12:17:22 +0000
Subject: [Freeswitch-users] First few seconds of call are silent
Message-ID: <A917871C958FCE419C05B1A59FAB2BEC36F144F0@Exchange2010.medit.local>

Hi,

We have a Microsoft Lync environment for our internal phone system and are using FreeSwitch to let Lync talk with our VOIP provider.  Whilst the VOIP provider properly support Lync (one of the few to do so) our NAT firewall ruins the TCP-based SIP packets as it doesn't forward through the SIP OPTIONS packet and Lync eventually thinks the VOIP provider isn't there.  Thus we're using FreeSwitch to support out NAT firewall (I've give it our static public IP address) and have disabled the firewall device's SIP ALG so that it doesn't ruin the SIP packets.

Anyway, I have this working very well except that most of the time (not all the time - maybe 90%) we just have silence for the first 3-6 seconds after each call connects.  This usually means that if we call someone we miss the "Hi, thanks for calling company XYZ.  You're speaking with ABC" and confusion results.  The call counter in Lync starts counting as soon as the call connects (we've tested with fixed landlines in one hand and our Lync-based phone in the other) but there's just silence.  Lync to Lync calls don't have this problem.

I've tried various options such as forcing codecs (as best as I could make it anyway) to G711a  since that's all Lync supports anyway (our at least it's happy to use this).  I've also tried
<param name="rtp-autoflush-during-bridge" value="false"/> in both legs of the FreeSwitch connections (ie to our SIP trunk provider and to Lync) without any success.  I was hoping that if it was some sort of lag then at least I would see it lag a lot in the call (and then have a definite problem to solve) but that wasn't the case as nothing changed.

Should I do something with the proxy media settings?  Reading the Wiki page at http://wiki.freeswitch.org/wiki/Proxy_Media seems to indicate that it really won't make much of a difference (our CPU isn't all that busy in the virtual machine running FreeSwitch and we only have a handful of phones anyway).  Also, I think I'd like to take advantage of some of FreeSwitch's very powerful features in the future to take care of some of what seems trickier to do in Lync such as having our support hunt group fall back to mobile phones when we're out of the office (Lync does simultaneous ringing of a user's mobile phone for direct calls but not, as far as I can tell, for hunt groups)

I previously had the 3CX software SIP phone talking to FreeSwitch when I was first testing out the system.  I've since gutted the FreeSwitch XML config (the "internal" / "external" SIP profiles are gone) so I can't try the software phone at the moment but hope to get it going again soon.  This way I can better see if it's Lync or FreeSwitch causing the problem.  Before I fumble around getting that going I was hoping someone might just know the obvious thing I'm overlooking!   :)

Thanks very much for your help.

Ian

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From cmcureau at gmail.com  Mon Jul 11 01:54:02 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Sun, 10 Jul 2011 16:54:02 -0500
Subject: [Freeswitch-users] segmentation fault with mod_spandsp
In-Reply-To: <CAL0fNgUHXMeeWjvKjCFX3JiFhbxooRi9Lo0gth+w0uHVNPiCEQ@mail.gmail.com>
References: <CAL0fNgWbvFn8trf8L0eG2cZ9=xx4XM0oP461iBcdstFOcV-rTA@mail.gmail.com>
	<CAFiqYunTKgES1CPdkyDQGJsia6WBE_C3eDaOn9UQv5Wfh=nq8A@mail.gmail.com>
	<CAL0fNgUHXMeeWjvKjCFX3JiFhbxooRi9Lo0gth+w0uHVNPiCEQ@mail.gmail.com>
Message-ID: <CAL0fNgWcG7Tq9ooKUAF_Loy1464j_8sTrOKV2=5RaX-K4YkNFQ@mail.gmail.com>

I don't claim to be an expert at gdb at all, but after compiling with
debugging symbols here's what I got:

GNU gdb (GDB) 7.0.1-debian
Copyright (C) 2009 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html
>
This is free software: you are free to change and redistribute it.
There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
and "show warranty" for details.
This GDB was configured as "x86_64-linux-gnu".
For bug reporting instructions, please see:
<http://www.gnu.org/software/gdb/bugs/>...
Reading symbols from
/usr/local/src/freeswitch/libs/spandsp/src/make_modem_filter...done.
(gdb) set args -m V.17 -i -r
(gdb) break 470
Breakpoint 1 at 0x4016cc: file ../src/make_modem_filter.c, line 470.
(gdb) run
Starting program:
/usr/local/src/freeswitch/libs/spandsp/src/make_modem_filter -m V.17 -i -r

Breakpoint 1, main (argc=5, argv=0x7fffffffebf8) at
../src/make_modem_filter.c:470
470        make_rx_filter(rx_coeff_sets,
(gdb) print rx_coeff_sets
$1 = 192
(gdb) print rx_coeffs_per_filter
$2 = 27
(gdb) print carrier
$3 = 1800
(gdb) print baud_rate
$4 = 2400
(gdb) print rx_excess_bandwidth
$5 = 0.5
(gdb) print fixed_point
$6 = 1
(gdb) print rx_tag
$7 = 0x4026bf ""
(gdb) step

Program received signal SIGSEGV, Segmentation fault.
0x0000000000400d1e in make_rx_filter (coeff_sets=Cannot access memory at
address 0x7ffffff9e62c
) at ../src/make_modem_filter.c:163
163 {
(gdb)

Also attached a core file for your viewing pleasure :)


On Sun, Jul 10, 2011 at 3:27 PM, Chris Cureau <cmcureau at gmail.com> wrote:

> Hi, Steve...thanks for replying!
>
> No core dump...but I did pass it through gdb (without compiling with
> debugging symbols)...
>
> GNU gdb (GDB) 7.0.1-debian
> Copyright (C) 2009 Free Software Foundation, Inc.
> License GPLv3+: GNU GPL version 3 or later <
> http://gnu.org/licenses/gpl.html>
> This is free software: you are free to change and redistribute it.
> There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
> and "show warranty" for details.
> This GDB was configured as "x86_64-linux-gnu".
> For bug reporting instructions, please see:
> <http://www.gnu.org/software/gdb/bugs/>...
> Reading symbols from
> /usr/local/src/freeswitch/libs/spandsp/src/make_modem_filter...(no debugging
> symbols found)...done.
> (gdb) set args -m V.17
> (gdb) run
> Starting program:
> /usr/local/src/freeswitch/libs/spandsp/src/make_modem_filter -m V.17
>
> Program received signal SIGSEGV, Segmentation fault.
> 0x0000000000400d1e in make_rx_filter ()
>
>
> On Sun, Jul 10, 2011 at 2:58 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Did you get a coredump generated?
>>
>> -Steve
>>
>>
>> On 10 July 2011 19:04, Chris Cureau <cmcureau at gmail.com> wrote:
>>
>>> Hi there!
>>>
>>> This morning, I did a fresh git checkout and attempted to build
>>> freeswitch with mod_freetdm and mod_flite defined in the modules.conf.  I am
>>> now getting a segfault when compiling:
>>>
>>> making all mod_spandsp
>>> Making all in src
>>> /bin/bash: line 1: 22453 Segmentation fault      ./make_modem_filter -m
>>> V.17 -r > v17_v32bis_rx_floating_rrc.h
>>> make[7]: *** [v17_v32bis_rx_floating_rrc.h] Error 139
>>> make[6]: *** [all] Error 2
>>> make[5]: *** [all-recursive] Error 1
>>> make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la]
>>> Error 2
>>> make[3]: *** [mod_spandsp-all] Error 1
>>> make[2]: *** [all-recursive] Error 1
>>> make[1]: *** [all-recursive] Error 1
>>> make: *** [all] Error 2
>>>
>>> running make_modem_filter manually gets the same result.  Twiddling with
>>> the commandline options doesn't help either.
>>>
>>> OS: Debian 6.0.1a amd64, also a fresh install from last night.
>>>
>>> Any idea what is happening?
>>>
>>> Thanks in advance!
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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 va_mclean at yahoo.com  Mon Jul 11 02:19:17 2011
From: va_mclean at yahoo.com (Mclean Va)
Date: Sun, 10 Jul 2011 15:19:17 -0700 (PDT)
Subject: [Freeswitch-users] FS start up error messages
Message-ID: <1310336357.20804.YahooMailNeo@web121610.mail.ne1.yahoo.com>

I got the following error messages from the most recent version of FS from binaries or built from source:

2011-07-10 18:11:41.281250 [ERR] switch_nat.c:200 Error checking for PMP [init failed]
2011-07-10 18:11:46.109375 [ERR] switch_nat.c:257 Bind Error
2011-07-10 18:11:46.109375 [ERR] switch_nat.c:364 Unable to initialize NAT thread


How critical are those errors and how to fix them?

Thanks
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From josh at cozecom.com  Mon Jul 11 05:00:35 2011
From: josh at cozecom.com (Josh Bryant)
Date: Sun, 10 Jul 2011 21:00:35 -0400
Subject: [Freeswitch-users] Freeswitch hangup in Bria
Message-ID: <CACw2DOP510P7LBvMvfNcxEsficJeWmRiF_tmo82X2pukxtCkLQ@mail.gmail.com>

Hello,

I recently purchased Bria 3 for the Mac to connect to Freeswitch. It
registers fine and I'm able to originate a call from Freeswitch to Bria.
However, after playing a file using the playback command, Bria hangs up on
the call after about 30 seconds. Is this a problem with Freeswitch or Bria?
I have not had this problem with X-Lite or any other client.

I originate the call successfully:

api originate sofia/192.168.1.7/1000 &park()

----------

I playback a wave file successfully:

SendMsg 181e1211-3f4d-425a-b6dc-b5438494b156
call-command: execute
execute-app-name: playback
execute-app-arg: /Users/josh/coze/donuts.wav

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

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

Then the call drops after about 30 seconds. From the FS console log:

2011-07-10 20:56:20.928915 [DEBUG] switch_ivr.c:576 sofia/internal/bakery
Command Execute playback(/Users/josh/coze/donuts.wav)
EXECUTE sofia/internal/bakery playback(/Users/josh/coze/donuts.wav)
2011-07-10 20:56:20.969209 [DEBUG] switch_core_file.c:176 File
/Users/josh/coze/donuts.wav sample rate 22050 doesn't match requested rate
8000
2011-07-10 20:56:20.969209 [DEBUG] switch_ivr_play_say.c:1279 Codec
Activated L16 at 8000hz 1 channels 20ms
2011-07-10 20:56:24.276635 [DEBUG] switch_ivr_play_say.c:1649 done playing
file
2011-07-10 20:56:56.220375 [DEBUG] switch_core_session.c:855 Send signal
sofia/internal/bakery [BREAK]
2011-07-10 20:56:56.240485 [DEBUG] switch_channel.c:2729
(sofia/internal/bakery) Callstate Change ACTIVE -> HANGUP
2011-07-10 20:56:56.240485 [NOTICE] sofia.c:558 Hangup sofia/internal/bakery
[CS_EXECUTE] [NORMAL_CLEARING]
2011-07-10 20:56:56.240485 [DEBUG] switch_channel.c:2745 Send signal
sofia/internal/bakery [KILL]
2011-07-10 20:56:56.240485 [DEBUG] switch_core_session.c:1154 Send signal
sofia/internal/bakery [BREAK]
2011-07-10 20:56:56.240485 [DEBUG] switch_core_session.c:2127
sofia/internal/bakery skip receive message [APPLICATION_EXEC_COMPLETE]
(channel is hungup already)
2011-07-10 20:56:56.240485 [DEBUG] switch_core_state_machine.c:380
(sofia/internal/bakery) State EXECUTE going to sleep
2011-07-10 20:56:56.240485 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/bakery) Running State Change CS_HANGUP
2011-07-10 20:56:56.240485 [DEBUG] switch_core_state_machine.c:575
(sofia/internal/bakery) State HANGUP
2011-07-10 20:56:56.240485 [DEBUG] mod_sofia.c:452 sofia/internal/bakery
Overriding SIP cause 480 with 200 from the other leg
2011-07-10 20:56:56.240485 [DEBUG] mod_sofia.c:458 Channel
sofia/internal/bakery hanging up, cause: NORMAL_CLEARING
2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:46
sofia/internal/bakery Standard HANGUP, cause: NORMAL_CLEARING
2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:575
(sofia/internal/bakery) State HANGUP going to sleep
2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:356
(sofia/internal/bakery) State Change CS_HANGUP -> CS_REPORTING
2011-07-10 20:56:56.280671 [DEBUG] switch_core_session.c:1154 Send signal
sofia/internal/bakery [BREAK]
2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/bakery) Running State Change CS_REPORTING
2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:635
(sofia/internal/bakery) State REPORTING
2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:53
sofia/internal/bakery Standard REPORTING, cause: NORMAL_CLEARING
2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:635
(sofia/internal/bakery) State REPORTING going to sleep
2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:350
(sofia/internal/bakery) State Change CS_REPORTING -> CS_DESTROY
2011-07-10 20:56:56.280671 [DEBUG] switch_core_session.c:1154 Send signal
sofia/internal/bakery [BREAK]
2011-07-10 20:56:56.280671 [DEBUG] switch_core_session.c:1326 Session 31
(sofia/internal/bakery) Locked, Waiting on external entities
2011-07-10 20:56:56.280671 [NOTICE] switch_core_session.c:1344 Session 31
(sofia/internal/bakery) Ended
2011-07-10 20:56:56.280671 [NOTICE] switch_core_session.c:1346 Close Channel
sofia/internal/bakery [CS_DESTROY]
2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:464
(sofia/internal/bakery) Callstate Change HANGUP -> DOWN
2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:467
(sofia/internal/bakery) Running State Change CS_DESTROY
2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:477
(sofia/internal/bakery) State DESTROY
2011-07-10 20:56:56.280671 [DEBUG] mod_sofia.c:363 sofia/internal/bakery
SOFIA DESTROY
2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:60
sofia/internal/bakery Standard DESTROY
2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:477
(sofia/internal/bakery) State DESTROY going to sleep
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From michaelbishop1984 at gmail.com  Mon Jul 11 11:43:56 2011
From: michaelbishop1984 at gmail.com (Michael B)
Date: Mon, 11 Jul 2011 00:43:56 -0700 (PDT)
Subject: [Freeswitch-users] SIP OPTIONS
Message-ID: <1310370236230-6570046.post@n2.nabble.com>

Hello, Our carrier by default sends us a SIP OPTIONS message at certain
intervals to ensure we are alive. Is there anyway I can configure Freeswitch
to respond to this with 200OK or similar? I have read that FS is able to do
this itself via a SIP PING, but can't find anything that talks about the
other way, i.e. FS being the one to receive the OPTIONS. Any advice would be
greatly appreciated.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SIP-OPTIONS-tp6570046p6570046.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Mon Jul 11 19:05:03 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jul 2011 08:05:03 -0700
Subject: [Freeswitch-users] FS start up error messages
In-Reply-To: <1310336357.20804.YahooMailNeo@web121610.mail.ne1.yahoo.com>
References: <1310336357.20804.YahooMailNeo@web121610.mail.ne1.yahoo.com>
Message-ID: <CAKzWOxWfWU8VksFvgSsR=Y7uZvYhne8n3_wXTo8q2NeSU1DsjA@mail.gmail.com>

nothing to worry about. it's just the autonat stuff looking for the external
ip addr. if you know your external ip address you can set it manually in the
sip profile. Also, you can start fs with the "-nonat" flag and it will skip
this test.

-MC

On Sun, Jul 10, 2011 at 3:19 PM, Mclean Va <va_mclean at yahoo.com> wrote:

> I got the following error messages from the most recent version of FS from
> binaries or built from source:
>
> 2011-07-10 18:11:41.281250 [ERR] switch_nat.c:200 Error checking for PMP
> [init failed]
> 2011-07-10 18:11:46.109375 [ERR] switch_nat.c:257 Bind Error
> 2011-07-10 18:11:46.109375 [ERR] switch_nat.c:364 Unable to initialize NAT
> thread
>
> How critical are those errors and how to fix them?
>
> Thanks
>
> _______________________________________________
> 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 msc at freeswitch.org  Mon Jul 11 19:07:20 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jul 2011 08:07:20 -0700
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAMNz+bvn+L-87UjFLYn=HX5S_FJ0cLWrwfKVrEpa-UDXk04Etg@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
	<CAKbxfG-h2_seFcYhu-5zwDbhD-z12xGuw7GmmKPPgSLGux3_zg@mail.gmail.com>
	<CAMNz+bsev6pqGc4u8YxhPvsFtzcpBC8R0W6frW52BtaGbJidJQ@mail.gmail.com>
	<CAKzWOxWiLpP6j=JXQE=oM+t6QF_fy0e30DCC5EO=9qGf+OFmgg@mail.gmail.com>
	<CAMNz+bvn+L-87UjFLYn=HX5S_FJ0cLWrwfKVrEpa-UDXk04Etg@mail.gmail.com>
Message-ID: <CAKzWOxWgoiaEE58Ku9qbdmMAgUBvv8=gfxo364Emr=F7Nmx+UA@mail.gmail.com>

On Mon, Jul 11, 2011 at 5:08 AM, Mathieu Lautram
<lautram.mathieu at gmail.com>wrote:

> In fact, I want to know when the phone of B leg is effectively ringing. If
> I don't ignore early media, FreeSWITCH will consider the bridge successful
> even if the phone of B leg is not ringing. So I have to ignore early media
> to know that. But, if I set ignore_early_media to true, A leg will not hear
> any ringtone.
>

In what scenarios are you dealing with "the bridge successful even if the
phone of B leg is not ringing"? Until you identify those specifically
there's not much we can do to assist you.

-MC
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From steveayre at gmail.com  Mon Jul 11 19:32:40 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 11 Jul 2011 16:32:40 +0100
Subject: [Freeswitch-users] SIP OPTIONS
In-Reply-To: <1310370236230-6570046.post@n2.nabble.com>
References: <1310370236230-6570046.post@n2.nabble.com>
Message-ID: <CAFiqYu=q9NeK5reYxQ-m4kxcXwzhEHfjt5rqpNbiaXBhFB89Ag@mail.gmail.com>

Yes... just load mod_sofia as normal.
It will repond to these OPTIONS requests with 200 OK by default, as long as
there is a sip profile configured for that IP:port.

-Steve


On 11 July 2011 08:43, Michael B <michaelbishop1984 at gmail.com> wrote:

> Hello, Our carrier by default sends us a SIP OPTIONS message at certain
> intervals to ensure we are alive. Is there anyway I can configure
> Freeswitch
> to respond to this with 200OK or similar? I have read that FS is able to do
> this itself via a SIP PING, but can't find anything that talks about the
> other way, i.e. FS being the one to receive the OPTIONS. Any advice would
> be
> greatly appreciated.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/SIP-OPTIONS-tp6570046p6570046.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> 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 steveayre at gmail.com  Mon Jul 11 19:38:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 11 Jul 2011 16:38:33 +0100
Subject: [Freeswitch-users] SIP OPTIONS
In-Reply-To: <CAFiqYu=q9NeK5reYxQ-m4kxcXwzhEHfjt5rqpNbiaXBhFB89Ag@mail.gmail.com>
References: <1310370236230-6570046.post@n2.nabble.com>
	<CAFiqYu=q9NeK5reYxQ-m4kxcXwzhEHfjt5rqpNbiaXBhFB89Ag@mail.gmail.com>
Message-ID: <CAFiqYumKLN9_W_bJHcu=LJKrVtaU78bYV+7fq1A80UMQaSfDNA@mail.gmail.com>

By the way, ANY standards-compliant SIP server should respond to the OPTIONS
with a usable response.

>From RFC3261, Section 11 says: "All UAs MUST support the OPTIONS method."
And Section 12 says "The response code chosen MUST be the same that would
have been chosen had the request been an INVITE."

The RFC suggests that can mean it would return user busy etc (FS treats most
replies to the ping as success). In practice I usually only see servers
replying with 200 OK or 403 Forbidden (not logged in).

In the case of FS it would return 200 OK because any more precise response
code requires going to the dialplan to process the call to determine the
user busy etc result.

-Steve


On 11 July 2011 16:32, Steven Ayre <steveayre at gmail.com> wrote:

> Yes... just load mod_sofia as normal.
> It will repond to these OPTIONS requests with 200 OK by default, as long as
> there is a sip profile configured for that IP:port.
>
> -Steve
>
>
>
> On 11 July 2011 08:43, Michael B <michaelbishop1984 at gmail.com> wrote:
>
>> Hello, Our carrier by default sends us a SIP OPTIONS message at certain
>> intervals to ensure we are alive. Is there anyway I can configure
>> Freeswitch
>> to respond to this with 200OK or similar? I have read that FS is able to
>> do
>> this itself via a SIP PING, but can't find anything that talks about the
>> other way, i.e. FS being the one to receive the OPTIONS. Any advice would
>> be
>> greatly appreciated.
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/SIP-OPTIONS-tp6570046p6570046.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> 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 steveayre at gmail.com  Mon Jul 11 19:40:14 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 11 Jul 2011 16:40:14 +0100
Subject: [Freeswitch-users] FS start up error messages
In-Reply-To: <CAKzWOxWfWU8VksFvgSsR=Y7uZvYhne8n3_wXTo8q2NeSU1DsjA@mail.gmail.com>
References: <1310336357.20804.YahooMailNeo@web121610.mail.ne1.yahoo.com>
	<CAKzWOxWfWU8VksFvgSsR=Y7uZvYhne8n3_wXTo8q2NeSU1DsjA@mail.gmail.com>
Message-ID: <CAFiqYu=h4RfYfPe8+4FL_j7-fEV7BmbHUqRvaXuqOAc6LpngXA@mail.gmail.com>

>
> Also, you can start fs with the "-nonat" flag and it will skip this test.
>

This also has the advantage if your FS server is on a public IP address that
FS will start up *much* faster.

-Steve



On 11 July 2011 16:05, Michael Collins <msc at freeswitch.org> wrote:

> nothing to worry about. it's just the autonat stuff looking for the
> external ip addr. if you know your external ip address you can set it
> manually in the sip profile. Also, you can start fs with the "-nonat" flag
> and it will skip this test.
>
> -MC
>
> On Sun, Jul 10, 2011 at 3:19 PM, Mclean Va <va_mclean at yahoo.com> wrote:
>
>> I got the following error messages from the most recent version of FS from
>> binaries or built from source:
>>
>> 2011-07-10 18:11:41.281250 [ERR] switch_nat.c:200 Error checking for PMP
>> [init failed]
>> 2011-07-10 18:11:46.109375 [ERR] switch_nat.c:257 Bind Error
>> 2011-07-10 18:11:46.109375 [ERR] switch_nat.c:364 Unable to initialize NAT
>> thread
>>
>> How critical are those errors and how to fix them?
>>
>> Thanks
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 wes-fs at 499x.com  Mon Jul 11 20:33:31 2011
From: wes-fs at 499x.com (Wes)
Date: Mon, 11 Jul 2011 11:33:31 -0500
Subject: [Freeswitch-users] play wav files, or speak prompts, into recording?
Message-ID: <4E1B25DB.1000002@499x.com>

Not sure how to phrase this correctly, which is why I'm having trouble 
searching for help on the internet.

I'd like to "speak" some audio into the recording that I am making of 
the user's voice.

For example, say I have the caller press some digits and then detect the 
digits that have been pressed, then, I can easily speak those digits 
back to them for confirmation, but what I'd really like to do is begin 
recording the call, and then speak those digits, or any other 
introductory message into the recording before they start speaking their 
message.

Say they type in the digits 123, then I'd read into the recording:  "The 
following is the dictation for user number 123, taken on July 3, 2011", 
and then the person would start speaking and the recording would continue.

Perhaps this would have to be done by some sort of post processing, 
where I append the introductory message onto the user's recording using 
sox, but I'd still like to use the voice system in freeswitch because it 
is good at reading numbers, speaking time, etc.

Thanks.


From msc at freeswitch.org  Mon Jul 11 20:44:05 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jul 2011 09:44:05 -0700
Subject: [Freeswitch-users] play wav files, or speak prompts,
	into recording?
In-Reply-To: <4E1B25DB.1000002@499x.com>
References: <4E1B25DB.1000002@499x.com>
Message-ID: <CAKzWOxXgNuA0aCChzN5YisgEbL34wmrio11uUxf2ZxN7SVxG9Q@mail.gmail.com>

You can post-process this. Just record the stuff you want into a specific
file name, put that file name into a chan var, and then use a hangup-hook or
cron job to do the post-processing.
-MC

On Mon, Jul 11, 2011 at 9:33 AM, Wes <wes-fs at 499x.com> wrote:

> Not sure how to phrase this correctly, which is why I'm having trouble
> searching for help on the internet.
>
> I'd like to "speak" some audio into the recording that I am making of
> the user's voice.
>
> For example, say I have the caller press some digits and then detect the
> digits that have been pressed, then, I can easily speak those digits
> back to them for confirmation, but what I'd really like to do is begin
> recording the call, and then speak those digits, or any other
> introductory message into the recording before they start speaking their
> message.
>
> Say they type in the digits 123, then I'd read into the recording:  "The
> following is the dictation for user number 123, taken on July 3, 2011",
> and then the person would start speaking and the recording would continue.
>
> Perhaps this would have to be done by some sort of post processing,
> where I append the introductory message onto the user's recording using
> sox, but I'd still like to use the voice system in freeswitch because it
> is good at reading numbers, speaking time, etc.
>
> Thanks.
>
> _______________________________________________
> 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 lautram.mathieu at gmail.com  Mon Jul 11 20:48:29 2011
From: lautram.mathieu at gmail.com (Mathieu Lautram)
Date: Mon, 11 Jul 2011 18:48:29 +0200
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAKzWOxWgoiaEE58Ku9qbdmMAgUBvv8=gfxo364Emr=F7Nmx+UA@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
	<CAKbxfG-h2_seFcYhu-5zwDbhD-z12xGuw7GmmKPPgSLGux3_zg@mail.gmail.com>
	<CAMNz+bsev6pqGc4u8YxhPvsFtzcpBC8R0W6frW52BtaGbJidJQ@mail.gmail.com>
	<CAKzWOxWiLpP6j=JXQE=oM+t6QF_fy0e30DCC5EO=9qGf+OFmgg@mail.gmail.com>
	<CAMNz+bvn+L-87UjFLYn=HX5S_FJ0cLWrwfKVrEpa-UDXk04Etg@mail.gmail.com>
	<CAKzWOxWgoiaEE58Ku9qbdmMAgUBvv8=gfxo364Emr=F7Nmx+UA@mail.gmail.com>
Message-ID: <CAMNz+btwO-qwz-RhBNe8=+BFZ1ucbWTgiVTYSkO0JcnP9ZW0xQ@mail.gmail.com>

Well, I think I was not very clear beacause of lack of informations about
what I wanted to do.
So now, here is the scenario:

I have a big originate string that I send through a socket:

bgapi originate
{var=value}[var=value...]sofia/route1/num|[var=value...]sofia/route2/num...
'&bridge(
{var=value...}[var=value...]sofia/route1/num|[var=value...]sofia/route2/num...)'

So, as you can see, I have pipe between the differents routes. Actually, if
I set ignore_early_media to TRUE and if the first route (route1) fails,
Freeswitch will go to the next route (route2) because it waits a state
"answered" to continue. BUT, the A leg won't hear any ringtone.
On the contrary, if I set ignore_early_media to FALSE and if the first route
(route1) fails, Freeswitch will NOT go to the next route (route2) because it
considers that the call is successful with the state "ringing".

So what I need is that Freeswitch considers a call successful ONLY when it
is "answered" and also that the A leg can hear a ringtone.

I apologize for the wrong way that I took since the beginning.

Thanks a lot

2011/7/11 Michael Collins <msc at freeswitch.org>

>
>
> On Mon, Jul 11, 2011 at 5:08 AM, Mathieu Lautram <
> lautram.mathieu at gmail.com> wrote:
>
>> In fact, I want to know when the phone of B leg is effectively ringing. If
>> I don't ignore early media, FreeSWITCH will consider the bridge successful
>> even if the phone of B leg is not ringing. So I have to ignore early media
>> to know that. But, if I set ignore_early_media to true, A leg will not hear
>> any ringtone.
>>
>
> In what scenarios are you dealing with "the bridge successful even if the
> phone of B leg is not ringing"? Until you identify those specifically
> there's not much we can do to assist you.
>
> -MC
>
>
> _______________________________________________
> 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
>
>


-- 
Mathieu LAUTRAM
Application developer

BJT Partners - FRANCE
+33 1 79 75 99 60
+33 6 61 59 07 25
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From spencer at 5ninesolutions.com  Mon Jul 11 20:52:54 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 11 Jul 2011 09:52:54 -0700
Subject: [Freeswitch-users] 482 Merged Request
Message-ID: <FD8607B2-145E-47FC-B33D-782E866FF4D1@5ninesolutions.com>

Hello all,
Sorry to dig up an old thread but we are having the same issue and I couldn't seem to find a solution.  Our scenario is similar to this:

http://lists.freeswitch.org/pipermail/freeswitch-users/2009-September/046709.html

Basically we are using FreeSWITCH as a basic B2BUA for topology hiding, CNAM lookups and number normalizing in our SBC setup.  A call looks like this:

Client UA -> Kamailio -> FS --back to--> Kamailo -> ISTPs

I have three profiles configured:
Outbound
Inbound
Media

If we respond with a 486 or 404 to our providers gateways, they simply drop the call (which is totally fine).  This doesn't give the person calling any real indication because the just hear the call drop.  I have another profile setup to provide progress tones as early media and then issue the appropriate final response (404, 486, etc.) The problem is this.  If a call passes thru a FS profile, back to the proxy and generates an error code that we redirect that call to the FS media profile to generate the tones, we get a 482 Merged Request back from FS because the Call-ID is the same and the call that just went thru, event though the transaction has ended.  Is there any way to do this without using another server just to generate the tones?

Thanks,
Spencer




From spencer at 5ninesolutions.com  Mon Jul 11 21:18:05 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 11 Jul 2011 10:18:05 -0700
Subject: [Freeswitch-users] txfax and rxfax X headers
Message-ID: <F39F0E46-D0AF-44DF-A17C-2F2E3E7CD12B@5ninesolutions.com>

Hello all,
Is there a way to add the fax result text as an X- header to the BYE after a fax attempt has been made.  It seems that this would be useful in debugging a fax call.  

Just a thought :-)

Spencer




From david.ponzone at ipeva.fr  Mon Jul 11 21:33:38 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 11 Jul 2011 19:33:38 +0200
Subject: [Freeswitch-users] First few seconds of call are silent
In-Reply-To: <A917871C958FCE419C05B1A59FAB2BEC36F144F0@Exchange2010.medit.local>
References: <A917871C958FCE419C05B1A59FAB2BEC36F144F0@Exchange2010.medit.local>
Message-ID: <5AC6863F-3EDF-43BB-BDA8-12EDF82D77C0@ipeva.fr>

Ian,

at least, you managed to get it working :)
During my trials with Lync, I had incoming calls (FS->Lync) disconnecting after 30 seconds.
Never solved that.

You should take a RTP trace on FS to see if FS sends the RTP flow to Lync as soon as expected (around 400ms after it received the incoming flow from Lync).

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

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

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




Le 10/07/2011 ? 14:17, Ian Yates a ?crit :

> Hi,
>  
> We have a Microsoft Lync environment for our internal phone system and are using FreeSwitch to let Lync talk with our VOIP provider.  Whilst the VOIP provider properly support Lync (one of the few to do so) our NAT firewall ruins the TCP-based SIP packets as it doesn?t forward through the SIP OPTIONS packet and Lync eventually thinks the VOIP provider isn?t there.  Thus we?re using FreeSwitch to support out NAT firewall (I?ve give it our static public IP address) and have disabled the firewall device?s SIP ALG so that it doesn?t ruin the SIP packets.
>  
> Anyway, I have this working very well except that most of the time (not all the time ? maybe 90%) we just have silence for the first 3-6 seconds after each call connects.  This usually means that if we call someone we miss the ?Hi, thanks for calling company XYZ.  You?re speaking with ABC? and confusion results.  The call counter in Lync starts counting as soon as the call connects (we?ve tested with fixed landlines in one hand and our Lync-based phone in the other) but there?s just silence.  Lync to Lync calls don?t have this problem.
>  
> I?ve tried various options such as forcing codecs (as best as I could make it anyway) to G711a  since that?s all Lync supports anyway (our at least it?s happy to use this).  I?ve also tried
> <param name="rtp-autoflush-during-bridge" value="false"/> in both legs of the FreeSwitch connections (ie to our SIP trunk provider and to Lync) without any success.  I was hoping that if it was some sort of lag then at least I would see it lag a lot in the call (and then have a definite problem to solve) but that wasn?t the case as nothing changed.
>  
> Should I do something with the proxy media settings?  Reading the Wiki page athttp://wiki.freeswitch.org/wiki/Proxy_Media seems to indicate that it really won?t make much of a difference (our CPU isn?t all that busy in the virtual machine running FreeSwitch and we only have a handful of phones anyway).  Also, I think I?d like to take advantage of some of FreeSwitch?s very powerful features in the future to take care of some of what seems trickier to do in Lync such as having our support hunt group fall back to mobile phones when we?re out of the office (Lync does simultaneous ringing of a user?s mobile phone for direct calls but not, as far as I can tell, for hunt groups)
>  
> I previously had the 3CX software SIP phone talking to FreeSwitch when I was first testing out the system.  I?ve since gutted the FreeSwitch XML config (the ?internal? / ?external? SIP profiles are gone) so I can?t try the software phone at the moment but hope to get it going again soon.  This way I can better see if it?s Lync or FreeSwitch causing the problem.  Before I fumble around getting that going I was hoping someone might just know the obvious thing I?m overlooking!   J
>  
> Thanks very much for your help.
>  
> Ian
>  
> _______________________________________________
> 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 wes-fs at 499x.com  Mon Jul 11 21:43:44 2011
From: wes-fs at 499x.com (Wes)
Date: Mon, 11 Jul 2011 12:43:44 -0500
Subject: [Freeswitch-users] play wav files, or speak prompts,
	into recording?
In-Reply-To: <CAKzWOxXgNuA0aCChzN5YisgEbL34wmrio11uUxf2ZxN7SVxG9Q@mail.gmail.com>
References: <4E1B25DB.1000002@499x.com>
	<CAKzWOxXgNuA0aCChzN5YisgEbL34wmrio11uUxf2ZxN7SVxG9Q@mail.gmail.com>
Message-ID: <4E1B3650.1060208@499x.com>

I'm having trouble figuring out how to record stuff into a file.  I know 
how to say stuff to the caller, and I know how to record the callers 
voice, but how would I say stuff directly into a file, instead of to the 
caller on the line?  And then, now that I think about it, this can't 
possibly happen in real time while interacting with the caller, because 
it takes time to read some digits, and if I want the read digits to go 
into a file, and not be spoken to the caller on the line, this probably 
has to happen in another thread....

I'm writing this interaction with lua scripting.

On 7/11/2011 11:44 AM, Michael Collins wrote:
> You can post-process this. Just record the stuff you want into a 
> specific file name, put that file name into a chan var, and then use a 
> hangup-hook or cron job to do the post-processing.
> -MC
>
> On Mon, Jul 11, 2011 at 9:33 AM, Wes <wes-fs at 499x.com 
> <mailto:wes-fs at 499x.com>> wrote:
>
>     Not sure how to phrase this correctly, which is why I'm having trouble
>     searching for help on the internet.
>
>     I'd like to "speak" some audio into the recording that I am making of
>     the user's voice.
>
>     For example, say I have the caller press some digits and then
>     detect the
>     digits that have been pressed, then, I can easily speak those digits
>     back to them for confirmation, but what I'd really like to do is begin
>     recording the call, and then speak those digits, or any other
>     introductory message into the recording before they start speaking
>     their
>     message.
>
>     Say they type in the digits 123, then I'd read into the recording:
>      "The
>     following is the dictation for user number 123, taken on July 3,
>     2011",
>     and then the person would start speaking and the recording would
>     continue.
>
>     Perhaps this would have to be done by some sort of post processing,
>     where I append the introductory message onto the user's recording
>     using
>     sox, but I'd still like to use the voice system in freeswitch
>     because it
>     is good at reading numbers, speaking time, etc.
>
>     Thanks.
>
>     _______________________________________________
>     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
>     <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
>
>
>
> _______________________________________________
> 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 jalsot at gmail.com  Mon Jul 11 21:45:15 2011
From: jalsot at gmail.com (Tamas Jalsovszky)
Date: Mon, 11 Jul 2011 19:45:15 +0200
Subject: [Freeswitch-users] txfax and rxfax X headers
In-Reply-To: <F39F0E46-D0AF-44DF-A17C-2F2E3E7CD12B@5ninesolutions.com>
References: <F39F0E46-D0AF-44DF-A17C-2F2E3E7CD12B@5ninesolutions.com>
Message-ID: <CAHCLHGuh5TWe5Azo0S6B6Yweq0u5KpD4XA-7PBngJgZUdE-mCQ@mail.gmail.com>

Hello,

You can send whatever header in BYE message with sip_bye_h channel
variables.
http://wiki.freeswitch.org/wiki/Sofia#Adding_Response_Headers

Br,
 T.

On Mon, Jul 11, 2011 at 7:18 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Hello all,
> Is there a way to add the fax result text as an X- header to the BYE after
> a fax attempt has been made.  It seems that this would be useful in
> debugging a fax call.
>
> Just a thought :-)
>
> Spencer
>
>
>
> _______________________________________________
> 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 msc at freeswitch.org  Mon Jul 11 21:55:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jul 2011 10:55:49 -0700
Subject: [Freeswitch-users] play wav files, or speak prompts,
	into recording?
In-Reply-To: <4E1B3650.1060208@499x.com>
References: <4E1B25DB.1000002@499x.com>
	<CAKzWOxXgNuA0aCChzN5YisgEbL34wmrio11uUxf2ZxN7SVxG9Q@mail.gmail.com>
	<4E1B3650.1060208@499x.com>
Message-ID: <CAKzWOxXcq8ZdPdGDGuNrcQsp8ixUnvuGRG0bV00=05i5fi2UCQ@mail.gmail.com>

Well, if you don't need the "stuff" to be voiced to the caller then yes, you
can just do it in another thread. I suppose there are several ways of doing
it, but if you just want FS to do all the work for you then I would create a
simple dialplan extension that you can call via originate and then just play
silence_stream://10000 on the b leg while doing the recording, say app, etc.
on the a leg.

-MC

On Mon, Jul 11, 2011 at 10:43 AM, Wes <wes-fs at 499x.com> wrote:

> **
> I'm having trouble figuring out how to record stuff into a file.  I know
> how to say stuff to the caller, and I know how to record the callers voice,
> but how would I say stuff directly into a file, instead of to the caller on
> the line?  And then, now that I think about it, this can't possibly happen
> in real time while interacting with the caller, because it takes time to
> read some digits, and if I want the read digits to go into a file, and not
> be spoken to the caller on the line, this probably has to happen in another
> thread....
>
> I'm writing this interaction with lua scripting.
>
>
> On 7/11/2011 11:44 AM, Michael Collins wrote:
>
> You can post-process this. Just record the stuff you want into a specific
> file name, put that file name into a chan var, and then use a hangup-hook or
> cron job to do the post-processing.
> -MC
>
>  On Mon, Jul 11, 2011 at 9:33 AM, Wes <wes-fs at 499x.com> wrote:
>
>> Not sure how to phrase this correctly, which is why I'm having trouble
>> searching for help on the internet.
>>
>> I'd like to "speak" some audio into the recording that I am making of
>> the user's voice.
>>
>> For example, say I have the caller press some digits and then detect the
>> digits that have been pressed, then, I can easily speak those digits
>> back to them for confirmation, but what I'd really like to do is begin
>> recording the call, and then speak those digits, or any other
>> introductory message into the recording before they start speaking their
>> message.
>>
>> Say they type in the digits 123, then I'd read into the recording:  "The
>> following is the dictation for user number 123, taken on July 3, 2011",
>> and then the person would start speaking and the recording would continue.
>>
>> Perhaps this would have to be done by some sort of post processing,
>> where I append the introductory message onto the user's recording using
>> sox, but I'd still like to use the voice system in freeswitch because it
>> is good at reading numbers, speaking time, etc.
>>
>> Thanks.
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> 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
>
>
> _______________________________________________
> 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 frank at telonium.com  Mon Jul 11 22:14:35 2011
From: frank at telonium.com (Frank Park)
Date: Mon, 11 Jul 2011 14:14:35 -0400
Subject: [Freeswitch-users] Uptimecommand on cli?
In-Reply-To: <CA3E6492.18B90%krice@freeswitch.org>
References: <Pine.LNX.4.64.1107100246080.19434@tomback.antagus.de>
	<CA3E6492.18B90%krice@freeswitch.org>
Message-ID: <CAAp+uUF7+KgKf-XD5MVgY=SchTqu6Xx-TPKJPBH6FNtn5az5jA@mail.gmail.com>

F2 button works well too.

Frank

On Sat, Jul 9, 2011 at 9:03 PM, Ken Rice <krice at freeswitch.org> wrote:

> status
>
>
> On 7/9/11 7:52 PM, "Thomas Hoellriegel" <admin at blindi.net> wrote:
>
> > Hi guys,
> > is there a cli command with which you can find out how long fs is
> >   already running?
> > 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
> > _______________________________________________
> > 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
>
>
>
> _______________________________________________
> 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
>



-- 

----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----
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From spencer at 5ninesolutions.com  Mon Jul 11 22:18:07 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 11 Jul 2011 11:18:07 -0700
Subject: [Freeswitch-users] txfax and rxfax X headers
In-Reply-To: <CAHCLHGuh5TWe5Azo0S6B6Yweq0u5KpD4XA-7PBngJgZUdE-mCQ@mail.gmail.com>
References: <F39F0E46-D0AF-44DF-A17C-2F2E3E7CD12B@5ninesolutions.com>
	<CAHCLHGuh5TWe5Azo0S6B6Yweq0u5KpD4XA-7PBngJgZUdE-mCQ@mail.gmail.com>
Message-ID: <1F486366-2130-48BD-9D1D-29059854F03E@5ninesolutions.com>

Thanks, sorry for asking without looking first.  I thought it must be more difficult.  :-)

FYI..
After a call to rxfax I did:
<action application="set" data="sip_bye_h_X-FS-Fax-Result=${fax_result_t
ext}"/>

and if we generate the BYE I get the following example header added:
X-FS-Fax-Result: Disconnected after permitted retries

Thanks!



On Jul 11, 2011, at 10:45 AM, Tamas Jalsovszky wrote:

> Hello,
> 
> You can send whatever header in BYE message with sip_bye_h channel variables.
> http://wiki.freeswitch.org/wiki/Sofia#Adding_Response_Headers
> 
> Br,
>  T.
> 
> On Mon, Jul 11, 2011 at 7:18 PM, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
> Hello all,
> Is there a way to add the fax result text as an X- header to the BYE after a fax attempt has been made.  It seems that this would be useful in debugging a fax call.
> 
> Just a thought :-)
> 
> Spencer
> 
> 
> 
> _______________________________________________
> 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
> 
> _______________________________________________
> 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 msc at freeswitch.org  Tue Jul 12 00:05:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jul 2011 13:05:37 -0700
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAMNz+btwO-qwz-RhBNe8=+BFZ1ucbWTgiVTYSkO0JcnP9ZW0xQ@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
	<CAKbxfG-h2_seFcYhu-5zwDbhD-z12xGuw7GmmKPPgSLGux3_zg@mail.gmail.com>
	<CAMNz+bsev6pqGc4u8YxhPvsFtzcpBC8R0W6frW52BtaGbJidJQ@mail.gmail.com>
	<CAKzWOxWiLpP6j=JXQE=oM+t6QF_fy0e30DCC5EO=9qGf+OFmgg@mail.gmail.com>
	<CAMNz+bvn+L-87UjFLYn=HX5S_FJ0cLWrwfKVrEpa-UDXk04Etg@mail.gmail.com>
	<CAKzWOxWgoiaEE58Ku9qbdmMAgUBvv8=gfxo364Emr=F7Nmx+UA@mail.gmail.com>
	<CAMNz+btwO-qwz-RhBNe8=+BFZ1ucbWTgiVTYSkO0JcnP9ZW0xQ@mail.gmail.com>
Message-ID: <CAKzWOxU_hsyNnJVP057WS0KzJzoeNhoZUBO4AMgVg+moTdobhQ@mail.gmail.com>

On Mon, Jul 11, 2011 at 9:48 AM, Mathieu Lautram
<lautram.mathieu at gmail.com>wrote:

> Well, I think I was not very clear beacause of lack of informations about
> what I wanted to do.
> So now, here is the scenario:
>
> I have a big originate string that I send through a socket:
>
> bgapi originate
> {var=value}[var=value...]sofia/route1/num|[var=value...]sofia/route2/num...
> '&bridge(
> {var=value...}[var=value...]sofia/route1/num|[var=value...]sofia/route2/num...)'
>
> So, as you can see, I have pipe between the differents routes. Actually, if
> I set ignore_early_media to TRUE and if the first route (route1) fails,
> Freeswitch will go to the next route (route2) because it waits a state
> "answered" to continue. BUT, the A leg won't hear any ringtone.
> On the contrary, if I set ignore_early_media to FALSE and if the first
> route (route1) fails, Freeswitch will NOT go to the next route (route2)
> because it considers that the call is successful with the state "ringing".
>
> So what I need is that Freeswitch considers a call successful ONLY when it
> is "answered" and also that the A leg can hear a ringtone.
>

Yes, you have a bit of a conundrum here. Since you have multiple possible
ring targets there simply is no way to use their early media for ringing. So
it looks like you're all the way back to one of the original suggestions
which is to supply your own ringback. To do this you will need to look up
the tones for each target country you will be calling and then set the
ringback variable accordingly.

Anybody else have thoughts on how else this might be accomplished with the
given constraints?
-MC
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From david.ponzone at ipeva.fr  Tue Jul 12 00:32:33 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 11 Jul 2011 22:32:33 +0200
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAKzWOxU_hsyNnJVP057WS0KzJzoeNhoZUBO4AMgVg+moTdobhQ@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
	<CAKbxfG-h2_seFcYhu-5zwDbhD-z12xGuw7GmmKPPgSLGux3_zg@mail.gmail.com>
	<CAMNz+bsev6pqGc4u8YxhPvsFtzcpBC8R0W6frW52BtaGbJidJQ@mail.gmail.com>
	<CAKzWOxWiLpP6j=JXQE=oM+t6QF_fy0e30DCC5EO=9qGf+OFmgg@mail.gmail.com>
	<CAMNz+bvn+L-87UjFLYn=HX5S_FJ0cLWrwfKVrEpa-UDXk04Etg@mail.gmail.com>
	<CAKzWOxWgoiaEE58Ku9qbdmMAgUBvv8=gfxo364Emr=F7Nmx+UA@mail.gmail.com>
	<CAMNz+btwO-qwz-RhBNe8=+BFZ1ucbWTgiVTYSkO0JcnP9ZW0xQ@mail.gmail.com>
	<CAKzWOxU_hsyNnJVP057WS0KzJzoeNhoZUBO4AMgVg+moTdobhQ@mail.gmail.com>
Message-ID: <54B1FE87-B82D-4699-BC4A-A1F571ADF6E0@ipeva.fr>

Michael,

I can't believe there is no way to tell FreeSWITCH to ignore the early media (so 180/183 SDP) and only to trust the OK/SDP before deciding if a call was connected.
I am pretty sure I saw that some months ago.

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

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

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




Le 11/07/2011 ? 22:05, Michael Collins a ?crit :

> 
> 
> On Mon, Jul 11, 2011 at 9:48 AM, Mathieu Lautram <lautram.mathieu at gmail.com> wrote:
> Well, I think I was not very clear beacause of lack of informations about what I wanted to do.
> So now, here is the scenario:
> 
> I have a big originate string that I send through a socket:
> 
> bgapi originate {var=value}[var=value...]sofia/route1/num|[var=value...]sofia/route2/num... '&bridge( {var=value...}[var=value...]sofia/route1/num|[var=value...]sofia/route2/num...)'
> 
> So, as you can see, I have pipe between the differents routes. Actually, if I set ignore_early_media to TRUE and if the first route (route1) fails, Freeswitch will go to the next route (route2) because it waits a state "answered" to continue. BUT, the A leg won't hear any ringtone.
> On the contrary, if I set ignore_early_media to FALSE and if the first route (route1) fails, Freeswitch will NOT go to the next route (route2) because it considers that the call is successful with the state "ringing".
> 
> So what I need is that Freeswitch considers a call successful ONLY when it is "answered" and also that the A leg can hear a ringtone.
> 
> Yes, you have a bit of a conundrum here. Since you have multiple possible ring targets there simply is no way to use their early media for ringing. So it looks like you're all the way back to one of the original suggestions which is to supply your own ringback. To do this you will need to look up the tones for each target country you will be calling and then set the ringback variable accordingly. 
> 
> Anybody else have thoughts on how else this might be accomplished with the given constraints?
> -MC
> _______________________________________________
> 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 msc at freeswitch.org  Tue Jul 12 00:59:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jul 2011 13:59:13 -0700
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <54B1FE87-B82D-4699-BC4A-A1F571ADF6E0@ipeva.fr>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
	<CAKbxfG-h2_seFcYhu-5zwDbhD-z12xGuw7GmmKPPgSLGux3_zg@mail.gmail.com>
	<CAMNz+bsev6pqGc4u8YxhPvsFtzcpBC8R0W6frW52BtaGbJidJQ@mail.gmail.com>
	<CAKzWOxWiLpP6j=JXQE=oM+t6QF_fy0e30DCC5EO=9qGf+OFmgg@mail.gmail.com>
	<CAMNz+bvn+L-87UjFLYn=HX5S_FJ0cLWrwfKVrEpa-UDXk04Etg@mail.gmail.com>
	<CAKzWOxWgoiaEE58Ku9qbdmMAgUBvv8=gfxo364Emr=F7Nmx+UA@mail.gmail.com>
	<CAMNz+btwO-qwz-RhBNe8=+BFZ1ucbWTgiVTYSkO0JcnP9ZW0xQ@mail.gmail.com>
	<CAKzWOxU_hsyNnJVP057WS0KzJzoeNhoZUBO4AMgVg+moTdobhQ@mail.gmail.com>
	<54B1FE87-B82D-4699-BC4A-A1F571ADF6E0@ipeva.fr>
Message-ID: <CAKzWOxVe70WLdGm3Zszgnm09Wx2Ft_NPEsTrEUaAY-HtCNQ=BA@mail.gmail.com>

On Mon, Jul 11, 2011 at 1:32 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Michael,
>
> I can't believe there is no way to tell FreeSWITCH to ignore the early
> media (so 180/183 SDP) and only to trust the OK/SDP before deciding if a
> call was connected.
>

I don't believe that this part is what is challenging. The hard part is
sending some sort of audio to the A leg while the B leg is ringing but has
not answered yet.

-MC
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From steveayre at gmail.com  Tue Jul 12 01:01:11 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 11 Jul 2011 22:01:11 +0100
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <54B1FE87-B82D-4699-BC4A-A1F571ADF6E0@ipeva.fr>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
	<CAKbxfG-h2_seFcYhu-5zwDbhD-z12xGuw7GmmKPPgSLGux3_zg@mail.gmail.com>
	<CAMNz+bsev6pqGc4u8YxhPvsFtzcpBC8R0W6frW52BtaGbJidJQ@mail.gmail.com>
	<CAKzWOxWiLpP6j=JXQE=oM+t6QF_fy0e30DCC5EO=9qGf+OFmgg@mail.gmail.com>
	<CAMNz+bvn+L-87UjFLYn=HX5S_FJ0cLWrwfKVrEpa-UDXk04Etg@mail.gmail.com>
	<CAKzWOxWgoiaEE58Ku9qbdmMAgUBvv8=gfxo364Emr=F7Nmx+UA@mail.gmail.com>
	<CAMNz+btwO-qwz-RhBNe8=+BFZ1ucbWTgiVTYSkO0JcnP9ZW0xQ@mail.gmail.com>
	<CAKzWOxU_hsyNnJVP057WS0KzJzoeNhoZUBO4AMgVg+moTdobhQ@mail.gmail.com>
	<54B1FE87-B82D-4699-BC4A-A1F571ADF6E0@ipeva.fr>
Message-ID: <CAFiqYukHus_3B3fbML4NarrPPbvwb=GZxrFf5UZT_2YD1+CE7g@mail.gmail.com>

The issue isn't that you can't do that, you can.

The issue is that you can't do that at the same time as passing back the
ringback from a bleg to an aleg, unless you generate the ringback yourself
(you can't both handle the 18x and ignore it at the same time).

-Steve



On 11 July 2011 21:32, David Ponzone <david.ponzone at ipeva.fr> wrote:

> Michael,
>
> I can't believe there is no way to tell FreeSWITCH to ignore the early
> media (so 180/183 SDP) and only to trust the OK/SDP before deciding if a
> call was connected.
> I am pretty sure I saw that some months ago.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 11/07/2011 ? 22:05, Michael Collins a ?crit :
>
>
>
> On Mon, Jul 11, 2011 at 9:48 AM, Mathieu Lautram <
> lautram.mathieu at gmail.com> wrote:
>
>> Well, I think I was not very clear beacause of lack of informations about
>> what I wanted to do.
>> So now, here is the scenario:
>>
>> I have a big originate string that I send through a socket:
>>
>> bgapi originate
>> {var=value}[var=value...]sofia/route1/num|[var=value...]sofia/route2/num...
>> '&bridge(
>> {var=value...}[var=value...]sofia/route1/num|[var=value...]sofia/route2/num...)'
>>
>> So, as you can see, I have pipe between the differents routes. Actually,
>> if I set ignore_early_media to TRUE and if the first route (route1) fails,
>> Freeswitch will go to the next route (route2) because it waits a state
>> "answered" to continue. BUT, the A leg won't hear any ringtone.
>> On the contrary, if I set ignore_early_media to FALSE and if the first
>> route (route1) fails, Freeswitch will NOT go to the next route (route2)
>> because it considers that the call is successful with the state "ringing".
>>
>> So what I need is that Freeswitch considers a call successful ONLY when it
>> is "answered" and also that the A leg can hear a ringtone.
>>
>
> Yes, you have a bit of a conundrum here. Since you have multiple possible
> ring targets there simply is no way to use their early media for ringing. So
> it looks like you're all the way back to one of the original suggestions
> which is to supply your own ringback. To do this you will need to look up
> the tones for each target country you will be calling and then set the
> ringback variable accordingly.
>
> Anybody else have thoughts on how else this might be accomplished with the
> given constraints?
> -MC
> _______________________________________________
> 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
>
>
>
> _______________________________________________
> 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 yungwei at resolvity.com  Tue Jul 12 01:07:39 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Mon, 11 Jul 2011 17:07:39 -0400
Subject: [Freeswitch-users] Invalid ASR module [mod_unimrcp]
Message-ID: <33095823FD21DF429B481B5163264B7950CBAF84CE@VMBX102.ihostexchange.net>

Hi,

I'm testing the pizza demo shipped with freeswitch. I tested it with pocketsphinx module, and that works.
Now I'm trying to do the same with a different ASR via unimrcp.

In ps_pizza.js, I replaced pocketsphinx in the following statement in ps_pizza.js with mod_unimrcp and unimrcp respectively, none of which works. 
What am I missing here? Thanks.
var asr = new SpeechDetect(session, "pocketsphinx");



From va_mclean at yahoo.com  Tue Jul 12 01:14:49 2011
From: va_mclean at yahoo.com (yahoo2003)
Date: Mon, 11 Jul 2011 14:14:49 -0700 (PDT)
Subject: [Freeswitch-users] FS start up error messages
In-Reply-To: <CAKzWOxWfWU8VksFvgSsR=Y7uZvYhne8n3_wXTo8q2NeSU1DsjA@mail.gmail.com>
References: <1310336357.20804.YahooMailNeo@web121610.mail.ne1.yahoo.com>
	<CAKzWOxWfWU8VksFvgSsR=Y7uZvYhne8n3_wXTo8q2NeSU1DsjA@mail.gmail.com>
Message-ID: <1310418889640-6572764.post@n2.nabble.com>

I did not set external ip address manually, 2 providers worked but Google
Voice doesn't work, I'm wandering if it's "external ip address" related, if
it's, which file I should modify to include "external ip address".

Thanks



mercutioviz wrote:
> 
> nothing to worry about. it's just the autonat stuff looking for the
> external
> ip addr. if you know your external ip address you can set it manually in
> the
> sip profile. Also, you can start fs with the "-nonat" flag and it will
> skip
> this test.
> 
> -MC
> 


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-start-up-error-messages-tp6571254p6572764.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From cmrienzo at gmail.com  Tue Jul 12 01:15:25 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Mon, 11 Jul 2011 17:15:25 -0400
Subject: [Freeswitch-users] Invalid ASR module [mod_unimrcp]
In-Reply-To: <33095823FD21DF429B481B5163264B7950CBAF84CE@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950CBAF84CE@VMBX102.ihostexchange.net>
Message-ID: <CAD-qRkgM06Zqtqcwn4K8hiHk8ND86wZ_AJfBN+6rqb3jZpS67Q@mail.gmail.com>

"unimrcp" is the engine to use.  Make sure your mrcp profile is set up
correctly.  The wiki has examples for various servers.

On Jul 11, 2011 5:09 PM, "Yungwei Chen" <yungwei at resolvity.com> wrote:

Hi,

I'm testing the pizza demo shipped with freeswitch. I tested it with
pocketsphinx module, and that works.
Now I'm trying to do the same with a different ASR via unimrcp.

In ps_pizza.js, I replaced pocketsphinx in the following statement in
ps_pizza.js with mod_unimrcp and unimrcp respectively, none of which works.
What am I missing here? Thanks.
var asr = new SpeechDetect(session, "pocketsphinx");


_______________________________________________
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 msc at freeswitch.org  Tue Jul 12 01:17:01 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jul 2011 14:17:01 -0700
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAFiqYukHus_3B3fbML4NarrPPbvwb=GZxrFf5UZT_2YD1+CE7g@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
	<CAKbxfG-h2_seFcYhu-5zwDbhD-z12xGuw7GmmKPPgSLGux3_zg@mail.gmail.com>
	<CAMNz+bsev6pqGc4u8YxhPvsFtzcpBC8R0W6frW52BtaGbJidJQ@mail.gmail.com>
	<CAKzWOxWiLpP6j=JXQE=oM+t6QF_fy0e30DCC5EO=9qGf+OFmgg@mail.gmail.com>
	<CAMNz+bvn+L-87UjFLYn=HX5S_FJ0cLWrwfKVrEpa-UDXk04Etg@mail.gmail.com>
	<CAKzWOxWgoiaEE58Ku9qbdmMAgUBvv8=gfxo364Emr=F7Nmx+UA@mail.gmail.com>
	<CAMNz+btwO-qwz-RhBNe8=+BFZ1ucbWTgiVTYSkO0JcnP9ZW0xQ@mail.gmail.com>
	<CAKzWOxU_hsyNnJVP057WS0KzJzoeNhoZUBO4AMgVg+moTdobhQ@mail.gmail.com>
	<54B1FE87-B82D-4699-BC4A-A1F571ADF6E0@ipeva.fr>
	<CAFiqYukHus_3B3fbML4NarrPPbvwb=GZxrFf5UZT_2YD1+CE7g@mail.gmail.com>
Message-ID: <CAKzWOxWvTjrCYv6pdN183+gCH6y2=5jegT-HxbTC9GSdMkaXHA@mail.gmail.com>

On Mon, Jul 11, 2011 at 2:01 PM, Steven Ayre <steveayre at gmail.com> wrote:

> The issue isn't that you can't do that, you can.
>
> The issue is that you can't do that at the same time as passing back the
> ringback from a bleg to an aleg, unless you generate the ringback yourself
> (you can't both handle the 18x and ignore it at the same time).
>

Agreed, which is where we're currently stuck. The question is exactly *how*
to generate ringback to the A leg when his B leg could be call anywhere in
the world. The OP said he wanted the appropriate ringback. I suggested that
he look at the target phone number, parse out the country code, and then
supply the appropriate ringback value for that country.

If there are alternate solutions to this then by all means present them so
that Mathieu can test drive them.
-MC
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From yungwei at resolvity.com  Tue Jul 12 01:26:16 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Mon, 11 Jul 2011 17:26:16 -0400
Subject: [Freeswitch-users] Invalid ASR module [mod_unimrcp]
In-Reply-To: <CAD-qRkgM06Zqtqcwn4K8hiHk8ND86wZ_AJfBN+6rqb3jZpS67Q@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950CBAF84CE@VMBX102.ihostexchange.net>
	<CAD-qRkgM06Zqtqcwn4K8hiHk8ND86wZ_AJfBN+6rqb3jZpS67Q@mail.gmail.com>
Message-ID: <33095823FD21DF429B481B5163264B7950CBAF84DA@VMBX102.ihostexchange.net>

When reviewing the log file, I also noticed the following message:  EXECUTE sofia/internal/1000 at 192.168.216.98 detect_speech(unimrcp pizza_order  undefined)
Now I'm wondering which folder ps_pizza.js looks for grammars. Thanks.

Here's the statement in ps_pizza.js.
pizza.orderObtainer.setGrammar("pizza_order", "", "result.interpretation.input", dft_min, dft_confirm, true);

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Christopher Rienzo
Sent: Monday, July 11, 2011 4:15 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Invalid ASR module [mod_unimrcp]


"unimrcp" is the engine to use.  Make sure your mrcp profile is set up correctly.  The wiki has examples for various servers.
On Jul 11, 2011 5:09 PM, "Yungwei Chen" <yungwei at resolvity.com<mailto:yungwei at resolvity.com>> wrote:

Hi,

I'm testing the pizza demo shipped with freeswitch. I tested it with pocketsphinx module, and that works.
Now I'm trying to do the same with a different ASR via unimrcp.

In ps_pizza.js, I replaced pocketsphinx in the following statement in ps_pizza.js with mod_unimrcp and unimrcp respectively, none of which works.
What am I missing here? Thanks.
var asr = new SpeechDetect(session, "pocketsphinx");


_______________________________________________
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<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 yungwei at resolvity.com  Tue Jul 12 01:31:58 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Mon, 11 Jul 2011 17:31:58 -0400
Subject: [Freeswitch-users] Invalid ASR module [mod_unimrcp]
In-Reply-To: <33095823FD21DF429B481B5163264B7950CBAF84DA@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950CBAF84CE@VMBX102.ihostexchange.net>
	<CAD-qRkgM06Zqtqcwn4K8hiHk8ND86wZ_AJfBN+6rqb3jZpS67Q@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950CBAF84DA@VMBX102.ihostexchange.net>
Message-ID: <33095823FD21DF429B481B5163264B7950CBAF84DD@VMBX102.ihostexchange.net>

I found the definition of setGrammar(....)
        this.setGrammar = function (grammar_name, path, obj_path, min_score, confirm_score, halt) {
                this.asr.setGrammar(this.grammar_name);

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei Chen
Sent: Monday, July 11, 2011 4:26 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Invalid ASR module [mod_unimrcp]

When reviewing the log file, I also noticed the following message:  EXECUTE sofia/internal/1000 at 192.168.216.98 detect_speech(unimrcp pizza_order  undefined)
Now I'm wondering which folder ps_pizza.js looks for grammars. Thanks.

Here's the statement in ps_pizza.js.
pizza.orderObtainer.setGrammar("pizza_order", "", "result.interpretation.input", dft_min, dft_confirm, true);

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Christopher Rienzo
Sent: Monday, July 11, 2011 4:15 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Invalid ASR module [mod_unimrcp]


"unimrcp" is the engine to use.  Make sure your mrcp profile is set up correctly.  The wiki has examples for various servers.
On Jul 11, 2011 5:09 PM, "Yungwei Chen" <yungwei at resolvity.com<mailto:yungwei at resolvity.com>> wrote:

Hi,

I'm testing the pizza demo shipped with freeswitch. I tested it with pocketsphinx module, and that works.
Now I'm trying to do the same with a different ASR via unimrcp.

In ps_pizza.js, I replaced pocketsphinx in the following statement in ps_pizza.js with mod_unimrcp and unimrcp respectively, none of which works.
What am I missing here? Thanks.
var asr = new SpeechDetect(session, "pocketsphinx");


_______________________________________________
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<mailto:FreeSWITCH-users at lists.freeswitch.org>
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From yungwei at resolvity.com  Tue Jul 12 01:53:14 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Mon, 11 Jul 2011 17:53:14 -0400
Subject: [Freeswitch-users] Invalid ASR module [mod_unimrcp]
In-Reply-To: <33095823FD21DF429B481B5163264B7950CBAF84DD@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950CBAF84CE@VMBX102.ihostexchange.net>
	<CAD-qRkgM06Zqtqcwn4K8hiHk8ND86wZ_AJfBN+6rqb3jZpS67Q@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950CBAF84DA@VMBX102.ihostexchange.net>
	<33095823FD21DF429B481B5163264B7950CBAF84DD@VMBX102.ihostexchange.net>
Message-ID: <33095823FD21DF429B481B5163264B7950CBAF84EE@VMBX102.ihostexchange.net>

I tried the following, but I'm still getting the same error, detect_speech(unimrcp pizza_order file://usr/local/freeswitch/grammar/pizza_order.gram undefined).  What am I missing here? Thanks.
pizza.orderObtainer.setGrammar("pizza_order", "file://usr/local/freeswitch/grammar/pizza_order.gram", "result.interpretation.input", dft_min, dft_confirm, true);
pizza.orderObtainer.setGrammar("pizza_order", "/usr/local/freeswitch/grammar/pizza_order.gram", "result.interpretation.input", dft_min, dft_confirm, true);

Here's the content of pizza_order.gram:
#JSGF V1.0;

/**
  * JSGF Grammar for pizza_order
  */

grammar pizza_order;

public <delivery> = [ takeout | pickup | delivery ];


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei Chen
Sent: Monday, July 11, 2011 4:32 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Invalid ASR module [mod_unimrcp]

I found the definition of setGrammar(....)
        this.setGrammar = function (grammar_name, path, obj_path, min_score, confirm_score, halt) {
                this.asr.setGrammar(this.grammar_name);

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei Chen
Sent: Monday, July 11, 2011 4:26 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Invalid ASR module [mod_unimrcp]

When reviewing the log file, I also noticed the following message:  EXECUTE sofia/internal/1000 at 192.168.216.98 detect_speech(unimrcp pizza_order  undefined)
Now I'm wondering which folder ps_pizza.js looks for grammars. Thanks.

Here's the statement in ps_pizza.js.
pizza.orderObtainer.setGrammar("pizza_order", "", "result.interpretation.input", dft_min, dft_confirm, true);

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Christopher Rienzo
Sent: Monday, July 11, 2011 4:15 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Invalid ASR module [mod_unimrcp]


"unimrcp" is the engine to use.  Make sure your mrcp profile is set up correctly.  The wiki has examples for various servers.
On Jul 11, 2011 5:09 PM, "Yungwei Chen" <yungwei at resolvity.com<mailto:yungwei at resolvity.com>> wrote:

Hi,

I'm testing the pizza demo shipped with freeswitch. I tested it with pocketsphinx module, and that works.
Now I'm trying to do the same with a different ASR via unimrcp.

In ps_pizza.js, I replaced pocketsphinx in the following statement in ps_pizza.js with mod_unimrcp and unimrcp respectively, none of which works.
What am I missing here? Thanks.
var asr = new SpeechDetect(session, "pocketsphinx");


_______________________________________________
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<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From lautram.mathieu at gmail.com  Tue Jul 12 02:10:46 2011
From: lautram.mathieu at gmail.com (Mathieu Lautram)
Date: Tue, 12 Jul 2011 00:10:46 +0200
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAKzWOxWvTjrCYv6pdN183+gCH6y2=5jegT-HxbTC9GSdMkaXHA@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
	<CAKbxfG-h2_seFcYhu-5zwDbhD-z12xGuw7GmmKPPgSLGux3_zg@mail.gmail.com>
	<CAMNz+bsev6pqGc4u8YxhPvsFtzcpBC8R0W6frW52BtaGbJidJQ@mail.gmail.com>
	<CAKzWOxWiLpP6j=JXQE=oM+t6QF_fy0e30DCC5EO=9qGf+OFmgg@mail.gmail.com>
	<CAMNz+bvn+L-87UjFLYn=HX5S_FJ0cLWrwfKVrEpa-UDXk04Etg@mail.gmail.com>
	<CAKzWOxWgoiaEE58Ku9qbdmMAgUBvv8=gfxo364Emr=F7Nmx+UA@mail.gmail.com>
	<CAMNz+btwO-qwz-RhBNe8=+BFZ1ucbWTgiVTYSkO0JcnP9ZW0xQ@mail.gmail.com>
	<CAKzWOxU_hsyNnJVP057WS0KzJzoeNhoZUBO4AMgVg+moTdobhQ@mail.gmail.com>
	<54B1FE87-B82D-4699-BC4A-A1F571ADF6E0@ipeva.fr>
	<CAFiqYukHus_3B3fbML4NarrPPbvwb=GZxrFf5UZT_2YD1+CE7g@mail.gmail.com>
	<CAKzWOxWvTjrCYv6pdN183+gCH6y2=5jegT-HxbTC9GSdMkaXHA@mail.gmail.com>
Message-ID: <CAMNz+bsv_Eyr0yikiiDy7+LFt684aZk9FD7e8q6OxitSVkCo=w@mail.gmail.com>

Thank you all for your replies

So if I understand well, for now, I have two solutions:

1/ check when the state is "answered" with events thrown by FS
2/ check the country code of the bleg and apply the correct ringtone

So, as Michael said, if you have any ideas, I'll be glad to test them.

Thanks a lot Michael and all of you for your efficiency :-)

Best regards

2011/7/11 Michael Collins <msc at freeswitch.org>

>
>
> On Mon, Jul 11, 2011 at 2:01 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> The issue isn't that you can't do that, you can.
>>
>> The issue is that you can't do that at the same time as passing back the
>> ringback from a bleg to an aleg, unless you generate the ringback yourself
>> (you can't both handle the 18x and ignore it at the same time).
>>
>
> Agreed, which is where we're currently stuck. The question is exactly *how*
> to generate ringback to the A leg when his B leg could be call anywhere in
> the world. The OP said he wanted the appropriate ringback. I suggested that
> he look at the target phone number, parse out the country code, and then
> supply the appropriate ringback value for that country.
>
> If there are alternate solutions to this then by all means present them so
> that Mathieu can test drive them.
> -MC
>
>
> _______________________________________________
> 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
>
>


-- 
Mathieu LAUTRAM
Application developer

BJT Partners - FRANCE
+33 1 79 75 99 60
+33 6 61 59 07 25
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From yungwei at resolvity.com  Tue Jul 12 02:14:53 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Mon, 11 Jul 2011 18:14:53 -0400
Subject: [Freeswitch-users] Invalid ASR module [mod_unimrcp]
In-Reply-To: <33095823FD21DF429B481B5163264B7950CBAF84EE@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950CBAF84CE@VMBX102.ihostexchange.net>
	<CAD-qRkgM06Zqtqcwn4K8hiHk8ND86wZ_AJfBN+6rqb3jZpS67Q@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950CBAF84DA@VMBX102.ihostexchange.net>
	<33095823FD21DF429B481B5163264B7950CBAF84DD@VMBX102.ihostexchange.net>
	<33095823FD21DF429B481B5163264B7950CBAF84EE@VMBX102.ihostexchange.net>
Message-ID: <33095823FD21DF429B481B5163264B7950CBAF84F8@VMBX102.ihostexchange.net>

It looks like the following statements builds a grammar on the fly, which means that I don't need to specify the path to the grammar file.
pizza.orderObtainer = new SpeechObtainer(asr, 1, 5000);
pizza.orderObtainer.setGrammar("pizza_order", "", "result.interpretation.input", dft_min, dft_confirm, true);
pizza.orderObtainer.setTopSound("GP-DeliveryorTakeout");
pizza.orderObtainer.setBadSound("GP-NoDeliveryorTake-out");
pizza.orderObtainer.addItemAlias("Delivery", "Delivery");
pizza.orderObtainer.addItemAlias("Takeout,Pickup", "Pickup");

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei Chen
Sent: Monday, July 11, 2011 4:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Invalid ASR module [mod_unimrcp]

I tried the following, but I'm still getting the same error, detect_speech(unimrcp pizza_order file://usr/local/freeswitch/grammar/pizza_order.gram undefined).  What am I missing here? Thanks.
pizza.orderObtainer.setGrammar("pizza_order", "file://usr/local/freeswitch/grammar/pizza_order.gram", "result.interpretation.input", dft_min, dft_confirm, true);
pizza.orderObtainer.setGrammar("pizza_order", "/usr/local/freeswitch/grammar/pizza_order.gram", "result.interpretation.input", dft_min, dft_confirm, true);

Here's the content of pizza_order.gram:
#JSGF V1.0;

/**
  * JSGF Grammar for pizza_order
  */

grammar pizza_order;

public <delivery> = [ takeout | pickup | delivery ];


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei Chen
Sent: Monday, July 11, 2011 4:32 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Invalid ASR module [mod_unimrcp]

I found the definition of setGrammar(....)
        this.setGrammar = function (grammar_name, path, obj_path, min_score, confirm_score, halt) {
                this.asr.setGrammar(this.grammar_name);

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei Chen
Sent: Monday, July 11, 2011 4:26 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Invalid ASR module [mod_unimrcp]

When reviewing the log file, I also noticed the following message:  EXECUTE sofia/internal/1000 at 192.168.216.98 detect_speech(unimrcp pizza_order  undefined)
Now I'm wondering which folder ps_pizza.js looks for grammars. Thanks.

Here's the statement in ps_pizza.js.
pizza.orderObtainer.setGrammar("pizza_order", "", "result.interpretation.input", dft_min, dft_confirm, true);

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Christopher Rienzo
Sent: Monday, July 11, 2011 4:15 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Invalid ASR module [mod_unimrcp]


"unimrcp" is the engine to use.  Make sure your mrcp profile is set up correctly.  The wiki has examples for various servers.
On Jul 11, 2011 5:09 PM, "Yungwei Chen" <yungwei at resolvity.com<mailto:yungwei at resolvity.com>> wrote:

Hi,

I'm testing the pizza demo shipped with freeswitch. I tested it with pocketsphinx module, and that works.
Now I'm trying to do the same with a different ASR via unimrcp.

In ps_pizza.js, I replaced pocketsphinx in the following statement in ps_pizza.js with mod_unimrcp and unimrcp respectively, none of which works.
What am I missing here? Thanks.
var asr = new SpeechDetect(session, "pocketsphinx");


_______________________________________________
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<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From brian at freeswitch.org  Tue Jul 12 02:21:36 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 11 Jul 2011 17:21:36 -0500
Subject: [Freeswitch-users] Freeswitch hangup in Bria
In-Reply-To: <CACw2DOP510P7LBvMvfNcxEsficJeWmRiF_tmo82X2pukxtCkLQ@mail.gmail.com>
References: <CACw2DOP510P7LBvMvfNcxEsficJeWmRiF_tmo82X2pukxtCkLQ@mail.gmail.com>
Message-ID: <3E1CCE1F-7993-473B-A6EE-4CB3175891BF@freeswitch.org>

Without some sort of log I can't tell you is Bria on the same network as FS behind nat?  If so don't let Bria use your public IP in the communications with FreeSWITCH otherwise its going to hair pin thru your NAT and not many of the consumer NAT implementations do this correctly.

/b

On Jul 10, 2011, at 8:00 PM, Josh Bryant wrote:

> Hello,
> 
> I recently purchased Bria 3 for the Mac to connect to Freeswitch. It
> registers fine and I'm able to originate a call from Freeswitch to Bria.
> However, after playing a file using the playback command, Bria hangs up on
> the call after about 30 seconds. Is this a problem with Freeswitch or Bria?
> I have not had this problem with X-Lite or any other client.
> 
> I originate the call successfully:
> 
> api originate sofia/192.168.1.7/1000 &park()
> 
> ----------
> 
> I playback a wave file successfully:
> 
> SendMsg 181e1211-3f4d-425a-b6dc-b5438494b156
> call-command: execute
> execute-app-name: playback
> execute-app-arg: /Users/josh/coze/donuts.wav
> 
> Content-Type: command/reply
> Reply-Text: +OK
> 
> --------------------------
> 
> Then the call drops after about 30 seconds. From the FS console log:
> 
> 2011-07-10 20:56:20.928915 [DEBUG] switch_ivr.c:576 sofia/internal/bakery
> Command Execute playback(/Users/josh/coze/donuts.wav)
> EXECUTE sofia/internal/bakery playback(/Users/josh/coze/donuts.wav)
> 2011-07-10 20:56:20.969209 [DEBUG] switch_core_file.c:176 File
> /Users/josh/coze/donuts.wav sample rate 22050 doesn't match requested rate
> 8000
> 2011-07-10 20:56:20.969209 [DEBUG] switch_ivr_play_say.c:1279 Codec
> Activated L16 at 8000hz 1 channels 20ms
> 2011-07-10 20:56:24.276635 [DEBUG] switch_ivr_play_say.c:1649 done playing
> file
> 2011-07-10 20:56:56.220375 [DEBUG] switch_core_session.c:855 Send signal
> sofia/internal/bakery [BREAK]
> 2011-07-10 20:56:56.240485 [DEBUG] switch_channel.c:2729
> (sofia/internal/bakery) Callstate Change ACTIVE -> HANGUP
> 2011-07-10 20:56:56.240485 [NOTICE] sofia.c:558 Hangup sofia/internal/bakery
> [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-07-10 20:56:56.240485 [DEBUG] switch_channel.c:2745 Send signal
> sofia/internal/bakery [KILL]
> 2011-07-10 20:56:56.240485 [DEBUG] switch_core_session.c:1154 Send signal
> sofia/internal/bakery [BREAK]
> 2011-07-10 20:56:56.240485 [DEBUG] switch_core_session.c:2127
> sofia/internal/bakery skip receive message [APPLICATION_EXEC_COMPLETE]
> (channel is hungup already)
> 2011-07-10 20:56:56.240485 [DEBUG] switch_core_state_machine.c:380
> (sofia/internal/bakery) State EXECUTE going to sleep
> 2011-07-10 20:56:56.240485 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/bakery) Running State Change CS_HANGUP
> 2011-07-10 20:56:56.240485 [DEBUG] switch_core_state_machine.c:575
> (sofia/internal/bakery) State HANGUP
> 2011-07-10 20:56:56.240485 [DEBUG] mod_sofia.c:452 sofia/internal/bakery
> Overriding SIP cause 480 with 200 from the other leg
> 2011-07-10 20:56:56.240485 [DEBUG] mod_sofia.c:458 Channel
> sofia/internal/bakery hanging up, cause: NORMAL_CLEARING
> 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/bakery Standard HANGUP, cause: NORMAL_CLEARING
> 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:575
> (sofia/internal/bakery) State HANGUP going to sleep
> 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:356
> (sofia/internal/bakery) State Change CS_HANGUP -> CS_REPORTING
> 2011-07-10 20:56:56.280671 [DEBUG] switch_core_session.c:1154 Send signal
> sofia/internal/bakery [BREAK]
> 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/bakery) Running State Change CS_REPORTING
> 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:635
> (sofia/internal/bakery) State REPORTING
> 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/bakery Standard REPORTING, cause: NORMAL_CLEARING
> 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:635
> (sofia/internal/bakery) State REPORTING going to sleep
> 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:350
> (sofia/internal/bakery) State Change CS_REPORTING -> CS_DESTROY
> 2011-07-10 20:56:56.280671 [DEBUG] switch_core_session.c:1154 Send signal
> sofia/internal/bakery [BREAK]
> 2011-07-10 20:56:56.280671 [DEBUG] switch_core_session.c:1326 Session 31
> (sofia/internal/bakery) Locked, Waiting on external entities
> 2011-07-10 20:56:56.280671 [NOTICE] switch_core_session.c:1344 Session 31
> (sofia/internal/bakery) Ended
> 2011-07-10 20:56:56.280671 [NOTICE] switch_core_session.c:1346 Close Channel
> sofia/internal/bakery [CS_DESTROY]
> 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:464
> (sofia/internal/bakery) Callstate Change HANGUP -> DOWN
> 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:467
> (sofia/internal/bakery) Running State Change CS_DESTROY
> 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:477
> (sofia/internal/bakery) State DESTROY
> 2011-07-10 20:56:56.280671 [DEBUG] mod_sofia.c:363 sofia/internal/bakery
> SOFIA DESTROY
> 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/bakery Standard DESTROY
> 2011-07-10 20:56:56.280671 [DEBUG] switch_core_state_machine.c:477
> (sofia/internal/bakery) State DESTROY going to sleep
> _______________________________________________
> 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 david.ponzone at ipeva.fr  Tue Jul 12 02:26:13 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 12 Jul 2011 00:26:13 +0200
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <CAFiqYukHus_3B3fbML4NarrPPbvwb=GZxrFf5UZT_2YD1+CE7g@mail.gmail.com>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
	<CAKbxfG-h2_seFcYhu-5zwDbhD-z12xGuw7GmmKPPgSLGux3_zg@mail.gmail.com>
	<CAMNz+bsev6pqGc4u8YxhPvsFtzcpBC8R0W6frW52BtaGbJidJQ@mail.gmail.com>
	<CAKzWOxWiLpP6j=JXQE=oM+t6QF_fy0e30DCC5EO=9qGf+OFmgg@mail.gmail.com>
	<CAMNz+bvn+L-87UjFLYn=HX5S_FJ0cLWrwfKVrEpa-UDXk04Etg@mail.gmail.com>
	<CAKzWOxWgoiaEE58Ku9qbdmMAgUBvv8=gfxo364Emr=F7Nmx+UA@mail.gmail.com>
	<CAMNz+btwO-qwz-RhBNe8=+BFZ1ucbWTgiVTYSkO0JcnP9ZW0xQ@mail.gmail.com>
	<CAKzWOxU_hsyNnJVP057WS0KzJzoeNhoZUBO4AMgVg+moTdobhQ@mail.gmail.com>
	<54B1FE87-B82D-4699-BC4A-A1F571ADF6E0@ipeva.fr>
	<CAFiqYukHus_3B3fbML4NarrPPbvwb=GZxrFf5UZT_2YD1+CE7g@mail.gmail.com>
Message-ID: <11A85161-DFF9-4D61-A6D1-883EA06B92B7@ipeva.fr>

Well, I used to think that should be 2 different things.
Early media is a way to provide ringing, including custom ringing and custom message without answering the call.
So it was logical for me to think that it should be possible to pass it from B to A, without flagging leg-B as answered, as it is not.
Is it specific of FS, or is it a general rule ?

Perhaps Mathieu could solve its issue (going from first route to next one) by using leg_timeout/originate_timeout on each bridge ?

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

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

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




Le 11/07/2011 ? 23:01, Steven Ayre a ?crit :

> The issue isn't that you can't do that, you can.
> 
> The issue is that you can't do that at the same time as passing back the ringback from a bleg to an aleg, unless you generate the ringback yourself (you can't both handle the 18x and ignore it at the same time).
> 
> -Steve
> 
> 
> 
> On 11 July 2011 21:32, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Michael,
> 
> I can't believe there is no way to tell FreeSWITCH to ignore the early media (so 180/183 SDP) and only to trust the OK/SDP before deciding if a call was connected.
> I am pretty sure I saw that some months ago.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 11/07/2011 ? 22:05, Michael Collins a ?crit :
> 
>> 
>> 
>> On Mon, Jul 11, 2011 at 9:48 AM, Mathieu Lautram <lautram.mathieu at gmail.com> wrote:
>> Well, I think I was not very clear beacause of lack of informations about what I wanted to do.
>> So now, here is the scenario:
>> 
>> I have a big originate string that I send through a socket:
>> 
>> bgapi originate {var=value}[var=value...]sofia/route1/num|[var=value...]sofia/route2/num... '&bridge( {var=value...}[var=value...]sofia/route1/num|[var=value...]sofia/route2/num...)'
>> 
>> So, as you can see, I have pipe between the differents routes. Actually, if I set ignore_early_media to TRUE and if the first route (route1) fails, Freeswitch will go to the next route (route2) because it waits a state "answered" to continue. BUT, the A leg won't hear any ringtone.
>> On the contrary, if I set ignore_early_media to FALSE and if the first route (route1) fails, Freeswitch will NOT go to the next route (route2) because it considers that the call is successful with the state "ringing".
>> 
>> So what I need is that Freeswitch considers a call successful ONLY when it is "answered" and also that the A leg can hear a ringtone.
>> 
>> Yes, you have a bit of a conundrum here. Since you have multiple possible ring targets there simply is no way to use their early media for ringing. So it looks like you're all the way back to one of the original suggestions which is to supply your own ringback. To do this you will need to look up the tones for each target country you will be calling and then set the ringback variable accordingly. 
>> 
>> Anybody else have thoughts on how else this might be accomplished with the given constraints?
>> -MC
>> _______________________________________________
>> 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
> 
> 
> _______________________________________________
> 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
> 
> 
> _______________________________________________
> 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 daniel at danielknoll.de  Tue Jul 12 02:27:19 2011
From: daniel at danielknoll.de (Daniel Knoll)
Date: Tue, 12 Jul 2011 00:27:19 +0200
Subject: [Freeswitch-users] mod_xml_curl changing order static/dynamic
	dialplans
Message-ID: <621E1FDB-93A7-48F1-A0DB-8AF41D6C9A2F@danielknoll.de>

Hi Group,

i used for my dialplan a combination of  static and dynamic dialplan, to reduce the http requests on fixed plans. 

But I saw the first request looking for matching numbers is going to the dynamic dialplan (using mod_xml_curl) and after that FS using the Static.
How can I chang the order? its better looking first at the static and then in the dynamic dialplan.

As Static Config I use this in external.xml described Static Section from mod_xml_curl Wikipage

<param name="dialplan" value="XML,XML:/usr/local/freeswitch/conf/dialplan/dp.xml"/>

Thanks for your help
Daniel

From msc at freeswitch.org  Tue Jul 12 02:44:00 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Jul 2011 15:44:00 -0700
Subject: [Freeswitch-users] No ringback with originate
In-Reply-To: <11A85161-DFF9-4D61-A6D1-883EA06B92B7@ipeva.fr>
References: <CAMNz+bspJOh6oz8rQBEVQ-h5ZpXQZ4m0uA33=q5xZUTzWJq05w@mail.gmail.com>
	<CAKzWOxVYY1GiskOKPs055mBiAcDDkx1fxvATpcRLujQtknLWAg@mail.gmail.com>
	<CAMNz+btFUd6wbwoDczen_OrT=P9PkwBY9FXBme7E0a2vMTu3EA@mail.gmail.com>
	<CAKbxfG9fgfAzcmtUgL9hdXi=x9yXFBqQUJB634qDrxDiO2Y1Vw@mail.gmail.com>
	<CAMNz+buoC_R-6Q8e6YGbTVTy-KFbpHY1AvhcomyMV1Nk=DbF5A@mail.gmail.com>
	<CAKbxfG-h2_seFcYhu-5zwDbhD-z12xGuw7GmmKPPgSLGux3_zg@mail.gmail.com>
	<CAMNz+bsev6pqGc4u8YxhPvsFtzcpBC8R0W6frW52BtaGbJidJQ@mail.gmail.com>
	<CAKzWOxWiLpP6j=JXQE=oM+t6QF_fy0e30DCC5EO=9qGf+OFmgg@mail.gmail.com>
	<CAMNz+bvn+L-87UjFLYn=HX5S_FJ0cLWrwfKVrEpa-UDXk04Etg@mail.gmail.com>
	<CAKzWOxWgoiaEE58Ku9qbdmMAgUBvv8=gfxo364Emr=F7Nmx+UA@mail.gmail.com>
	<CAMNz+btwO-qwz-RhBNe8=+BFZ1ucbWTgiVTYSkO0JcnP9ZW0xQ@mail.gmail.com>
	<CAKzWOxU_hsyNnJVP057WS0KzJzoeNhoZUBO4AMgVg+moTdobhQ@mail.gmail.com>
	<54B1FE87-B82D-4699-BC4A-A1F571ADF6E0@ipeva.fr>
	<CAFiqYukHus_3B3fbML4NarrPPbvwb=GZxrFf5UZT_2YD1+CE7g@mail.gmail.com>
	<11A85161-DFF9-4D61-A6D1-883EA06B92B7@ipeva.fr>
Message-ID: <CAKzWOxWY50Y_krmbSa09v5uQ6kkYGaCEc65cH9KsGGj4cXTPZg@mail.gmail.com>

On Mon, Jul 11, 2011 at 3:26 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Well, I used to think that should be 2 different things.
> Early media is a way to provide ringing, including custom ringing and
> custom message without answering the call.
> So it was logical for me to think that it should be possible to pass it
> from B to A, without flagging leg-B as answered, as it is not.
> Is it specific of FS, or is it a general rule ?
>
> Perhaps Mathieu could solve its issue (going from first route to next one)
> by using leg_timeout/originate_timeout on each bridge ?
>
>
One of the dangers in this situation is that you don't necessarily know what
kind of early media the far end is sending. If you bridge the early media on
leg B to leg A then the call is bridged. Are you going to monitor the early
media on leg B and then unbridge if you get something you don't like? Not
likely.

If you cannot rely on early media then it's wise to ignore it and handle
ringback on your own. Unless, of course, there's a third option that someone
else has to offer.

-MC
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From fieldpeak at gmail.com  Tue Jul 12 12:31:31 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 12 Jul 2011 16:31:31 +0800
Subject: [Freeswitch-users] How to configure for FS playing voice prompt
 in case the called extension is in a call.
In-Reply-To: <CAFiqYuncFULQn=BnWDh2s0m6VhAX3aqN-zPbFaMD19cowfOERA@mail.gmail.com>
References: <CAOTmbMzMVtRjUvTj5UgHH7MG=XyHSWRvw7QT6G-tiLFd57H1Vw@mail.gmail.com>
	<CAFiqYuncFULQn=BnWDh2s0m6VhAX3aqN-zPbFaMD19cowfOERA@mail.gmail.com>
Message-ID: <CAOTmbMwtVZJd_44rs7QH1GATSPygyfMJq2Ao-kgxw1x-L_vkVg@mail.gmail.com>

Hi Steve,

Thanks for your kindly help, it works with below dial plan...
However, for both user busy or no answer, it played the voice prompt...
i would like it only prompts in case user busy but not no answer, although i
configured the "failure_causes=USER_BUSY" as below dialplan, it looks not
help...attached is a log...

    <extension name="Local_Extension">
      <condition field="destination_number" expression="^(1\d{3})$">
    <action application="set" data="dialed_extension=$1"/>
    <action application="export" data="dialed_extension=$1"/>
    <action application="set" data="ringback=${us-ring}"/>
    <action application="set" data="transfer_ringback=$${hold_music}"/>
    <action application="set" data="call_timeout=30"/>

    <action application="set" data="failure_causes=USER_BUSY"/>

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

    <!-- to enable voice mail, set continue_on_fail=true below -->
    <action application="set" data="continue_on_fail=true"/>
    <action application="hash"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
    <action application="hash"
data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
    <action application="bridge"
data="sofia/internal/${dialed_extension}%${domain_name}"/>
    <action application="answer"/>

    <action application="sleep" data="1000"/>
    <!-- play a voice prompt e.g. 'the extension you dialed is busy now,
please dial the other exsention' to replacing busy tone when the called
extension is in a call... -->
    <action application="play_and_get_digits" data="2 5 3 4000 #
$${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav
other_ext_var \d+"/>
    <action application="transfer" data="${other_ext_var} XML default"/>

      </condition>
    </extension>

Regards,
Charles


2011/7/11 Steven Ayre <steveayre at gmail.com>

> There are two ways....
>
> Most phones will return a USER_BUSY release code if you bridge to a phone
> that's busy. If the phone does then you can continue_on_fail so that you
> (pre-)answer the call and play that message after the bridge. Something
> like:
>
> <extension name="Local_Extension">
>     <condition field="destination_number" expression="^(1\d{3})$">
>         <action application="set" data="dialed_extension=$1"/>
>         <action application="export" data="dialed_extension=$1"/>
>         <action application="set" data="call_timeout=30"/>
>         <action application="set" data="hangup_after_bridge=true"/> <!--
> Hangup on successful call -->
>         <action application="set" data="continue_on_fail=true"/> <!-- Don't
> hangup on failed call, continue to IVR prompt -->
>
>         <action application="hash"
> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>         <action application="hash"
> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>         <action application="bridge"
> data="user/${dialed_extension}%${domain_name}"/>
>         <!-- Put the IVR prompt here, see
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits-->
>     </condition>
> </extension>
>
> The 2nd option is that you use Limit to determine if another person is
> already on a call to the phone. That only really works if you run the the
> only servers that call the phone since it requires tracking all calls to the
> phone. It will sometimes work better though - especially if the phone has
> multiple lines. http://wiki.freeswitch.org/wiki/Limit
>
> -Steve
>
>
> On 11 July 2011 03:25, fieldpeak <fieldpeak at gmail.com> wrote:
>
>> Hi Gurus,
>>
>> Could anyone advise how to realize have FS to play a voice prompt e.g.
>> 'the extension you dialed is busy now, please dial the other exsention' to
>> replacing busy tone when the called extension is in a call...
>>
>> the details is below,
>>
>> When calling to FS, the FS will play IVR "welcome to call us, please input
>> the extension number, for operator please press 0",
>> then the caller input the extension number, in case the extension is in a
>> call, the caller will hear busy tone...
>> it needs the system play "the extension you dailed is busy, please dial
>> the other extension", currently my dial plan as following, could anyone
>> advise how to change based on below dial plan or any other new dail plan can
>> realize it...
>>
>> Thanks a lot!
>>
>>     <extension name="Local_Extension">
>>       <condition field="destination_number" expression="^(1\d{3})$">
>>     <action application="set" data="dialed_extension=$1"/>
>>     <action application="export" data="dialed_extension=$1"/>
>>     <action application="set" data="ringback=${us-ring}"/>
>>     <action application="set" data="transfer_ringback=$${hold_music}"/>
>>     <action application="set" data="call_timeout=30"/>
>>     <action application="set" data="hangup_after_bridge=true"/>
>>
>>     <!-- to enable voice mail, set continue_on_fail=true below -->
>>     <action application="set" data="continue_on_fail=false"/>
>>     <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>     <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>     <action application="bridge"
>> data="sofia/internal/${dialed_extension}%${domain_name}"/>
>>     <action application="answer"/>
>>
>>     <action application="sleep" data="1000"/>
>>     <action application="bridge" data="loopback/app=voicemail:default
>> ${domain_name} ${dialed_extension}"/>
>>       </condition>
>>     </extension>
>>
>>
>>     <extension name="IVR" >
>>        <condition field="destination_number" expression="^888$">
>>            <action application="answer"/>
>>            <action application="sleep" data="2000"/>
>>            <action application="ivr" data="IVR"/>
>>        </condition>
>>     </extension>
>>
>>         <!--  -->
>>     <menu name="IVR"
>>         greet-long="C:/VSWITCH/recordings/greeting_tts.wav"
>>         greet-short="C:/VSWITCH/recordings/greeting_tts.wav"
>>         invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>         exit-sound="voicemail/vm-goodbye.wav"
>>         confirm-macro=""
>>         confirm-key=""
>>         tts-engine="flite"
>>         tts-voice="rms"
>>         confirm-attempts="3"
>>         timeout="10000"
>>         inter-digit-timeout="2000"
>>         max-failures="3"
>>         max-timeouts="3"
>>         digit-len="5">
>>         <entry action="menu-exec-app" digits="/(^\*\d{3,5}$|^\d{3,5}$)/"
>> param="transfer $1 XML default"/>
>>     </menu>
>>
>> Regards,
>> Charles
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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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vswitch at mypc> 2011-07-12 16:28:07.779375 [DEBUG] sofia.c:6539 IP 192.168.200.100 Rejected by acl "172.28.0.0/16". Falling back to Digest auth.
2011-07-12 16:28:07.782375 [WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia profile 'internal' for [1001 at 192.168.200.100] from ip 192.1
68.200.100
2011-07-12 16:28:07.853379 [DEBUG] sofia.c:6539 IP 192.168.200.100 Rejected by acl "172.28.0.0/16". Falling back to Digest auth.
2011-07-12 16:28:07.884381 [NOTICE] switch_channel.c:812 New Channel sofia/internal/1005 at 192.168.200.100 [4201fb5a-c6ee-451a-bf5b-18614ce9353b]
2011-07-12 16:28:07.885381 [DEBUG] sofia.c:4760 Channel sofia/internal/1005 at 192.168.200.100 entering state [received][100]
2011-07-12 16:28:07.885381 [DEBUG] sofia.c:4771 Remote SDP:
v=0
o=- 111347976 111347994 IN IP4 192.168.200.100
s=eyeBeam
c=IN IP4 192.168.200.100
t=0 0
m=audio 6200 RTP/AVP 100 6 0 8 3 18 98 97 5 102 101
a=rtpmap:100 speex/16000
a=rtpmap:98 ilbc/8000
a=rtpmap:97 speex/8000
a=rtpmap:102 l16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 1 : 0730C003 366B174B 192.168.200.100 6200

2011-07-12 16:28:07.886381 [DEBUG] sofia.c:4916 (sofia/internal/1005 at 192.168.200.100) State Change CS_NEW -> CS_INIT
2011-07-12 16:28:07.886381 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1005 at 192.168.200.100 [BREAK]
2011-07-12 16:28:07.886381 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1005 at 192.168.200.100) Running State Change CS_INIT
2011-07-12 16:28:07.886381 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/1005 at 192.168.200.100) State INIT
2011-07-12 16:28:07.886381 [DEBUG] mod_sofia.c:84 sofia/internal/1005 at 192.168.200.100 SOFIA INIT
2011-07-12 16:28:07.886381 [DEBUG] mod_sofia.c:124 (sofia/internal/1005 at 192.168.200.100) State Change CS_INIT -> CS_ROUTING
2011-07-12 16:28:07.886381 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1005 at 192.168.200.100 [BREAK]
2011-07-12 16:28:07.886381 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/1005 at 192.168.200.100) State INIT going to sleep
2011-07-12 16:28:07.886381 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1005 at 192.168.200.100) Running State Change CS_ROUTING
2011-07-12 16:28:07.886381 [DEBUG] switch_channel.c:1668 (sofia/internal/1005 at 192.168.200.100) Callstate Change DOWN -> RINGING
2011-07-12 16:28:07.886381 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1005 at 192.168.200.100) State ROUTING
2011-07-12 16:28:07.886381 [DEBUG] mod_sofia.c:147 sofia/internal/1005 at 192.168.200.100 SOFIA ROUTING
2011-07-12 16:28:07.886381 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1005 at 192.168.200.100 Standard ROUTING
2011-07-12 16:28:07.886381 [INFO] mod_dialplan_xml.c:331 Processing 1005 <1005>->1001 in context default
Dialplan: sofia/internal/1005 at 192.168.200.100 parsing [default->unloop] continue=false
Dialplan: sofia/internal/1005 at 192.168.200.100 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.200.100 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.200.100 parsing [default->Local_Extension] continue=false
Dialplan: sofia/internal/1005 at 192.168.200.100 Regex (PASS) [Local_Extension] destination_number(1001) =~ /^(1\d{3})$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.200.100 Action set(dialed_extension=1001)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action export(dialed_extension=1001)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action set(ringback=${us-ring})
Dialplan: sofia/internal/1005 at 192.168.200.100 Action set(transfer_ringback=local_stream://moh)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action set(call_timeout=30)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action set(failure_causes=USER_BUSY)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action set(hangup_after_bridge=true)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action set(continue_on_fail=true)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
Dialplan: sofia/internal/1005 at 192.168.200.100 Action hash(insert/${domain_name}-last_dial_ext/global/${uuid})
Dialplan: sofia/internal/1005 at 192.168.200.100 Action bridge(sofia/internal/${dialed_extension}%${domain_name})
Dialplan: sofia/internal/1005 at 192.168.200.100 Action answer()
Dialplan: sofia/internal/1005 at 192.168.200.100 Action sleep(1000)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action play_and_get_digits(2 5 3 4000 # C:\VSWITCH/sounds/en/us/callie/conference/8000/conf-pin.wav /inv
alid.wav other_ext_var \d+)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action transfer(${other_ext_var} XML default)
2011-07-12 16:28:07.890381 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1005 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE
2011-07-12 16:28:07.890381 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1005 at 192.168.200.100 [BREAK]
2011-07-12 16:28:07.890381 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1005 at 192.168.200.100) State ROUTING going to sleep
2011-07-12 16:28:07.890381 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1005 at 192.168.200.100) Running State Change CS_EXECUTE
2011-07-12 16:28:07.890381 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/1005 at 192.168.200.100) State EXECUTE
2011-07-12 16:28:07.890381 [DEBUG] mod_sofia.c:240 sofia/internal/1005 at 192.168.200.100 SOFIA EXECUTE
2011-07-12 16:28:07.890381 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1005 at 192.168.200.100 Standard EXECUTE
EXECUTE sofia/internal/1005 at 192.168.200.100 set(dialed_extension=1001)
2011-07-12 16:28:07.890381 [DEBUG] mod_dptools.c:1060 sofia/internal/1005 at 192.168.200.100 SET [dialed_extension]=[1001]
EXECUTE sofia/internal/1005 at 192.168.200.100 export(dialed_extension=1001)
2011-07-12 16:28:07.890381 [DEBUG] switch_channel.c:961 EXPORT (export_vars) [dialed_extension]=[1001]
EXECUTE sofia/internal/1005 at 192.168.200.100 set(ringback=%(2000, 4000, 440.0, 480.0))
2011-07-12 16:28:07.890381 [DEBUG] mod_dptools.c:1060 sofia/internal/1005 at 192.168.200.100 SET [ringback]=[%(2000, 4000, 440.0, 480.0)]
EXECUTE sofia/internal/1005 at 192.168.200.100 set(transfer_ringback=local_stream://moh)
2011-07-12 16:28:07.890381 [DEBUG] mod_dptools.c:1060 sofia/internal/1005 at 192.168.200.100 SET [transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/1005 at 192.168.200.100 set(call_timeout=30)
2011-07-12 16:28:07.890381 [DEBUG] mod_dptools.c:1060 sofia/internal/1005 at 192.168.200.100 SET [call_timeout]=[30]
EXECUTE sofia/internal/1005 at 192.168.200.100 set(failure_causes=USER_BUSY)
2011-07-12 16:28:07.890381 [DEBUG] mod_dptools.c:1060 sofia/internal/1005 at 192.168.200.100 SET [failure_causes]=[USER_BUSY]
EXECUTE sofia/internal/1005 at 192.168.200.100 set(hangup_after_bridge=true)
2011-07-12 16:28:07.890381 [DEBUG] mod_dptools.c:1060 sofia/internal/1005 at 192.168.200.100 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1005 at 192.168.200.100 set(continue_on_fail=true)
2011-07-12 16:28:07.899381 [DEBUG] mod_dptools.c:1060 sofia/internal/1005 at 192.168.200.100 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/1005 at 192.168.200.100 hash(insert/192.168.200.100-last_dial_ext/1001/4201fb5a-c6ee-451a-bf5b-18614ce9353b)
EXECUTE sofia/internal/1005 at 192.168.200.100 hash(insert/192.168.200.100-last_dial_ext/global/4201fb5a-c6ee-451a-bf5b-18614ce9353b)
EXECUTE sofia/internal/1005 at 192.168.200.100 bridge(sofia/internal/1001%192.168.200.100)
2011-07-12 16:28:07.901382 [DEBUG] switch_channel.c:918 sofia/internal/1005 at 192.168.200.100 EXPORTING[export_vars] [dialed_extension]=[1001] to event
2011-07-12 16:28:07.901382 [NOTICE] switch_channel.c:812 New Channel sofia/internal/1001 [ffbcec4e-6366-4a97-ac52-53266512e421]
2011-07-12 16:28:07.901382 [DEBUG] mod_sofia.c:4300 (sofia/internal/1001) State Change CS_NEW -> CS_INIT
2011-07-12 16:28:07.901382 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1001 [BREAK]
2011-07-12 16:28:07.905382 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1001) Running State Change CS_INIT
2011-07-12 16:28:07.905382 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/1001) State INIT
2011-07-12 16:28:07.905382 [DEBUG] mod_sofia.c:84 sofia/internal/1001 SOFIA INIT
2011-07-12 16:28:07.906382 [DEBUG] sofia_glue.c:1757 sofia/internal/1001 Patched SDP
---
v=0
o=- 111347976 111347994 IN IP4 192.168.200.100
s=eyeBeam
c=IN IP4 192.168.200.100
t=0 0
m=audio 6200 RTP/AVP 100 6 0 8 3 18 98 97 5 102 101
a=rtpmap:100 speex/16000
a=rtpmap:98 ilbc/8000
a=rtpmap:97 speex/8000
a=rtpmap:102 l16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 1 : 0730C003 366B174B 192.168.200.100 6200

+++
v=0
o=FreeSWITCH 0287508612 0287508613 IN IP4 192.168.200.100
s=FreeSWITCH
c=IN IP4 192.168.200.100
t=0 0
m=audio 18968 RTP/AVP 100 6 0 8 3 18 98 97 5 102 101
a=rtpmap:100 speex/16000
a=rtpmap:98 ilbc/8000
a=rtpmap:97 speex/8000
a=rtpmap:102 l16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 1 : 0730C003 366B174B 192.168.200.100 6200

2011-07-12 16:28:07.906382 [DEBUG] mod_sofia.c:124 (sofia/internal/1001) State Change CS_INIT -> CS_ROUTING
2011-07-12 16:28:07.906382 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1001 [BREAK]
2011-07-12 16:28:07.906382 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/1001) State INIT going to sleep
2011-07-12 16:28:07.906382 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1001) Running State Change CS_ROUTING
2011-07-12 16:28:07.906382 [DEBUG] switch_channel.c:1668 (sofia/internal/1001) Callstate Change DOWN -> RINGING
2011-07-12 16:28:07.906382 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1001) State ROUTING
2011-07-12 16:28:07.906382 [DEBUG] mod_sofia.c:147 sofia/internal/1001 SOFIA ROUTING
2011-07-12 16:28:07.906382 [DEBUG] sofia.c:4760 Channel sofia/internal/1001 entering state [calling][0]
2011-07-12 16:28:07.909382 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/1001) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-07-12 16:28:07.909382 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1001 [BREAK]
2011-07-12 16:28:07.909382 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1001) State ROUTING going to sleep
2011-07-12 16:28:07.909382 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1001) Running State Change CS_CONSUME_MEDIA
2011-07-12 16:28:07.909382 [DEBUG] switch_core_state_machine.c:383 (sofia/internal/1001) State CONSUME_MEDIA
2011-07-12 16:28:07.909382 [DEBUG] switch_core_state_machine.c:383 (sofia/internal/1001) State CONSUME_MEDIA going to sleep
2011-07-12 16:28:07.981386 [DEBUG] sofia.c:4760 Channel sofia/internal/1001 entering state [terminated][486]
2011-07-12 16:28:07.981386 [DEBUG] switch_channel.c:2563 (sofia/internal/1001) Callstate Change RINGING -> HANGUP
2011-07-12 16:28:07.982386 [NOTICE] sofia.c:5406 Hangup sofia/internal/1001 [CS_CONSUME_MEDIA] [USER_BUSY]
2011-07-12 16:28:07.982386 [DEBUG] switch_channel.c:2579 Send signal sofia/internal/1001 [KILL]
2011-07-12 16:28:07.982386 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1001 [BREAK]
2011-07-12 16:28:07.982386 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1001) Running State Change CS_HANGUP
2011-07-12 16:28:07.984386 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1001) State HANGUP
2011-07-12 16:28:07.984386 [DEBUG] mod_sofia.c:451 sofia/internal/1001 Overriding SIP cause 486 with 486 from the other leg
2011-07-12 16:28:07.984386 [DEBUG] mod_sofia.c:457 Channel sofia/internal/1001 hanging up, cause: USER_BUSY
2011-07-12 16:28:07.984386 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1001 Standard HANGUP, cause: USER_BUSY
2011-07-12 16:28:07.984386 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1001) State HANGUP going to sleep
2011-07-12 16:28:07.984386 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/1001) State Change CS_HANGUP -> CS_REPORTING
2011-07-12 16:28:07.984386 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1001 [BREAK]
2011-07-12 16:28:07.984386 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1001) Running State Change CS_REPORTING
2011-07-12 16:28:07.984386 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1001) State REPORTING
2011-07-12 16:28:07.984386 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1001 Standard REPORTING, cause: USER_BUSY
2011-07-12 16:28:07.984386 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1001) State REPORTING going to sleep
2011-07-12 16:28:07.984386 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/1001) State Change CS_REPORTING -> CS_DESTROY
2011-07-12 16:28:07.984386 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1001 [BREAK]
2011-07-12 16:28:07.984386 [DEBUG] switch_core_session.c:1288 Session 2 (sofia/internal/1001) Locked, Waiting on external entities
2011-07-12 16:28:07.992387 [DEBUG] switch_ivr_originate.c:3492 Originate Resulted in Error Cause: 17 [USER_BUSY]
2011-07-12 16:28:07.992387 [INFO] mod_dptools.c:2640 Originate Failed.  Cause: USER_BUSY
EXECUTE sofia/internal/1005 at 192.168.200.100 answer()
2011-07-12 16:28:07.993387 [DEBUG] mod_sofia.c:649 Disabling proxy mode due to call answer with no bridge
2011-07-12 16:28:07.993387 [DEBUG] sofia_glue.c:4637 Audio Codec Compare [speex:100:16000:20:0]/[PCMA:8:8000:20:64000]
2011-07-12 16:28:07.993387 [DEBUG] sofia_glue.c:4637 Audio Codec Compare [speex:100:16000:20:0]/[PCMU:0:8000:20:64000]
2011-07-12 16:28:07.993387 [DEBUG] sofia_glue.c:4637 Audio Codec Compare [speex:100:16000:20:0]/[G722:9:8000:20:64000]
2011-07-12 16:28:07.993387 [DEBUG] sofia_glue.c:4637 Audio Codec Compare [speex:100:16000:20:0]/[GSM:3:8000:20:13200]
2011-07-12 16:28:07.993387 [DEBUG] sofia_glue.c:4637 Audio Codec Compare [DVI4:6:16000:20:0]/[PCMA:8:8000:20:64000]
2011-07-12 16:28:07.993387 [DEBUG] sofia_glue.c:4637 Audio Codec Compare [DVI4:6:16000:20:0]/[PCMU:0:8000:20:64000]
2011-07-12 16:28:07.993387 [DEBUG] sofia_glue.c:4637 Audio Codec Compare [DVI4:6:16000:20:0]/[G722:9:8000:20:64000]
2011-07-12 16:28:07.993387 [DEBUG] sofia_glue.c:4637 Audio Codec Compare [DVI4:6:16000:20:0]/[GSM:3:8000:20:13200]
2011-07-12 16:28:07.993387 [DEBUG] sofia_glue.c:4637 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMA:8:8000:20:64000]
2011-07-12 16:28:07.993387 [DEBUG] sofia_glue.c:4637 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-07-12 16:28:07.993387 [DEBUG] sofia_glue.c:2760 Set Codec sofia/internal/1005 at 192.168.200.100 PCMU/8000 20 ms 160 samples 64000 bits
2011-07-12 16:28:07.995387 [DEBUG] sofia_glue.c:4751 Set 2833 dtmf send/recv payload to 101
2011-07-12 16:28:07.995387 [DEBUG] sofia_glue.c:3001 AUDIO RTP [sofia/internal/1005 at 192.168.200.100] 192.168.200.100 port 20404 -> 192.168.200.100 por
t 6200 codec: 0 ms: 20
2011-07-12 16:28:07.996387 [NOTICE] switch_core_session.c:1306 Session 2 (sofia/internal/1001) Ended
2011-07-12 16:28:07.996387 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/1001 [CS_DESTROY]
2011-07-12 16:28:07.996387 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/1001) Callstate Change HANGUP -> DOWN
2011-07-12 16:28:07.996387 [DEBUG] switch_core_state_machine.c:457 (sofia/internal/1001) Running State Change CS_DESTROY
2011-07-12 16:28:07.996387 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1001) State DESTROY
2011-07-12 16:28:07.996387 [DEBUG] mod_sofia.c:362 sofia/internal/1001 SOFIA DESTROY
2011-07-12 16:28:07.996387 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1001 Standard DESTROY
2011-07-12 16:28:07.996387 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1001) State DESTROY going to sleep
2011-07-12 16:28:07.996387 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
2011-07-12 16:28:07.997387 [DEBUG] sofia_glue.c:3263 Set 2833 dtmf send payload to 101
2011-07-12 16:28:07.997387 [DEBUG] sofia_glue.c:3268 Set 2833 dtmf receive payload to 101
2011-07-12 16:28:07.997387 [NOTICE] sofia_glue.c:3772 Pre-Answer sofia/internal/1005 at 192.168.200.100!
2011-07-12 16:28:07.997387 [DEBUG] switch_channel.c:2639 (sofia/internal/1005 at 192.168.200.100) Callstate Change RINGING -> EARLY
2011-07-12 16:28:07.999387 [DEBUG] mod_sofia.c:681 Local SDP sofia/internal/1005 at 192.168.200.100:
v=0
o=FreeSWITCH 1310438883 1310438884 IN IP4 192.168.200.100
s=FreeSWITCH
c=IN IP4 192.168.200.100
t=0 0
m=audio 20404 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

2011-07-12 16:28:07.999387 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/1005 at 192.168.200.100 [BREAK]
2011-07-12 16:28:07.999387 [DEBUG] switch_channel.c:2821 (sofia/internal/1005 at 192.168.200.100) Callstate Change EARLY -> ACTIVE
2011-07-12 16:28:07.999387 [NOTICE] mod_dptools.c:930 Channel [sofia/internal/1005 at 192.168.200.100] has been answered
2011-07-12 16:28:08.000387 [DEBUG] sofia.c:4760 Channel sofia/internal/1005 at 192.168.200.100 entering state [completed][200]
EXECUTE sofia/internal/1005 at 192.168.200.100 sleep(1000)
2011-07-12 16:28:08.111394 [DEBUG] switch_rtp.c:3082 Correct ip/port confirmed.
2011-07-12 16:28:08.111394 [DEBUG] sofia.c:4760 Channel sofia/internal/1005 at 192.168.200.100 entering state [ready][200]
EXECUTE sofia/internal/1005 at 192.168.200.100 play_and_get_digits(2 5 3 4000 # C:\VSWITCH/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav
other_ext_var \d+)
2011-07-12 16:28:09.027446 [DEBUG] switch_ivr_play_say.c:1278 Codec Activated L16 at 8000hz 1 channels 20ms
2011-07-12 16:28:10.751545 [DEBUG] switch_ivr_play_say.c:1648 done playing file
2011-07-12 16:28:12.871666 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 1:800
2011-07-12 16:28:13.611708 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 0:800
2011-07-12 16:28:13.811720 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 0:960
2011-07-12 16:28:14.431755 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 2:800
2011-07-12 16:28:15.231801 [DEBUG] switch_rtp.c:3280 RTP RECV DTMF #:800
2011-07-12 16:28:15.231801 [DEBUG] switch_ivr_play_say.c:1975 Test Regex [1002][\d+]
EXECUTE sofia/internal/1005 at 192.168.200.100 transfer(1002 XML default)
2011-07-12 16:28:15.239801 [DEBUG] switch_ivr.c:1600 (sofia/internal/1005 at 192.168.200.100) State Change CS_EXECUTE -> CS_ROUTING
2011-07-12 16:28:15.239801 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1005 at 192.168.200.100 [BREAK]
2011-07-12 16:28:15.239801 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/1005 at 192.168.200.100 [BREAK]
2011-07-12 16:28:15.239801 [NOTICE] switch_ivr.c:1606 Transfer sofia/internal/1005 at 192.168.200.100 to XML[1002 at default]
2011-07-12 16:28:15.239801 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/1005 at 192.168.200.100) State EXECUTE going to sleep
2011-07-12 16:28:15.239801 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1005 at 192.168.200.100) Running State Change CS_ROUTING
2011-07-12 16:28:15.239801 [DEBUG] switch_channel.c:1668 (sofia/internal/1005 at 192.168.200.100) Callstate Change ACTIVE -> RINGING
2011-07-12 16:28:15.239801 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1005 at 192.168.200.100) State ROUTING
2011-07-12 16:28:15.239801 [DEBUG] mod_sofia.c:147 sofia/internal/1005 at 192.168.200.100 SOFIA ROUTING
2011-07-12 16:28:15.239801 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1005 at 192.168.200.100 Standard ROUTING
2011-07-12 16:28:15.239801 [INFO] mod_dialplan_xml.c:331 Processing 1005 <1005>->1002 in context default
Dialplan: sofia/internal/1005 at 192.168.200.100 parsing [default->unloop] continue=false
Dialplan: sofia/internal/1005 at 192.168.200.100 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.200.100 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.200.100 parsing [default->Local_Extension] continue=false
Dialplan: sofia/internal/1005 at 192.168.200.100 Regex (PASS) [Local_Extension] destination_number(1002) =~ /^(1\d{3})$/ break=on-false
Dialplan: sofia/internal/1005 at 192.168.200.100 Action set(dialed_extension=1002)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action export(dialed_extension=1002)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action set(ringback=${us-ring})
Dialplan: sofia/internal/1005 at 192.168.200.100 Action set(transfer_ringback=local_stream://moh)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action set(call_timeout=30)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action set(failure_causes=USER_BUSY)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action set(hangup_after_bridge=true)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action set(continue_on_fail=true)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
Dialplan: sofia/internal/1005 at 192.168.200.100 Action hash(insert/${domain_name}-last_dial_ext/global/${uuid})
Dialplan: sofia/internal/1005 at 192.168.200.100 Action bridge(sofia/internal/${dialed_extension}%${domain_name})
Dialplan: sofia/internal/1005 at 192.168.200.100 Action answer()
Dialplan: sofia/internal/1005 at 192.168.200.100 Action sleep(1000)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action play_and_get_digits(2 5 3 4000 # C:\VSWITCH/sounds/en/us/callie/conference/8000/conf-pin.wav /inv
alid.wav other_ext_var \d+)
Dialplan: sofia/internal/1005 at 192.168.200.100 Action transfer(${other_ext_var} XML default)
2011-07-12 16:28:15.251802 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1005 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE
2011-07-12 16:28:15.251802 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1005 at 192.168.200.100 [BREAK]
2011-07-12 16:28:15.251802 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1005 at 192.168.200.100) State ROUTING going to sleep
2011-07-12 16:28:15.251802 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1005 at 192.168.200.100) Running State Change CS_EXECUTE
2011-07-12 16:28:15.251802 [DEBUG] switch_channel.c:1670 (sofia/internal/1005 at 192.168.200.100) Callstate Change RINGING -> ACTIVE
2011-07-12 16:28:15.251802 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/1005 at 192.168.200.100) State EXECUTE
2011-07-12 16:28:15.251802 [DEBUG] mod_sofia.c:240 sofia/internal/1005 at 192.168.200.100 SOFIA EXECUTE
2011-07-12 16:28:15.251802 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1005 at 192.168.200.100 Standard EXECUTE
EXECUTE sofia/internal/1005 at 192.168.200.100 set(dialed_extension=1002)
2011-07-12 16:28:15.252802 [DEBUG] mod_dptools.c:1060 sofia/internal/1005 at 192.168.200.100 SET [dialed_extension]=[1002]
EXECUTE sofia/internal/1005 at 192.168.200.100 export(dialed_extension=1002)
2011-07-12 16:28:15.252802 [DEBUG] switch_channel.c:961 EXPORT (export_vars) [dialed_extension]=[1002]
EXECUTE sofia/internal/1005 at 192.168.200.100 set(ringback=%(2000, 4000, 440.0, 480.0))
2011-07-12 16:28:15.253802 [DEBUG] mod_dptools.c:1060 sofia/internal/1005 at 192.168.200.100 SET [ringback]=[%(2000, 4000, 440.0, 480.0)]
EXECUTE sofia/internal/1005 at 192.168.200.100 set(transfer_ringback=local_stream://moh)
2011-07-12 16:28:15.254802 [DEBUG] mod_dptools.c:1060 sofia/internal/1005 at 192.168.200.100 SET [transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/1005 at 192.168.200.100 set(call_timeout=30)
2011-07-12 16:28:15.255802 [DEBUG] mod_dptools.c:1060 sofia/internal/1005 at 192.168.200.100 SET [call_timeout]=[30]
EXECUTE sofia/internal/1005 at 192.168.200.100 set(failure_causes=USER_BUSY)
2011-07-12 16:28:15.255802 [DEBUG] mod_dptools.c:1060 sofia/internal/1005 at 192.168.200.100 SET [failure_causes]=[USER_BUSY]
EXECUTE sofia/internal/1005 at 192.168.200.100 set(hangup_after_bridge=true)
2011-07-12 16:28:15.256802 [DEBUG] mod_dptools.c:1060 sofia/internal/1005 at 192.168.200.100 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1005 at 192.168.200.100 set(continue_on_fail=true)
2011-07-12 16:28:15.256802 [DEBUG] mod_dptools.c:1060 sofia/internal/1005 at 192.168.200.100 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/1005 at 192.168.200.100 hash(insert/192.168.200.100-last_dial_ext/1002/4201fb5a-c6ee-451a-bf5b-18614ce9353b)
EXECUTE sofia/internal/1005 at 192.168.200.100 hash(insert/192.168.200.100-last_dial_ext/global/4201fb5a-c6ee-451a-bf5b-18614ce9353b)
EXECUTE sofia/internal/1005 at 192.168.200.100 bridge(sofia/internal/1002%192.168.200.100)
2011-07-12 16:28:15.260802 [DEBUG] switch_channel.c:918 sofia/internal/1005 at 192.168.200.100 EXPORTING[export_vars] [dialed_extension]=[1002] to event
2011-07-12 16:28:15.260802 [DEBUG] switch_channel.c:918 sofia/internal/1005 at 192.168.200.100 EXPORTING[export_vars] [dialed_extension]=[1002] to event
2011-07-12 16:28:15.261803 [NOTICE] switch_channel.c:812 New Channel sofia/internal/1002 [84302f81-aba4-4c61-90da-45118757025e]
2011-07-12 16:28:15.261803 [DEBUG] mod_sofia.c:4300 (sofia/internal/1002) State Change CS_NEW -> CS_INIT
2011-07-12 16:28:15.261803 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 [BREAK]
2011-07-12 16:28:15.262803 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002) Running State Change CS_INIT
2011-07-12 16:28:15.262803 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/1002) State INIT
2011-07-12 16:28:15.262803 [DEBUG] mod_sofia.c:84 sofia/internal/1002 SOFIA INIT
2011-07-12 16:28:15.263803 [DEBUG] mod_sofia.c:124 (sofia/internal/1002) State Change CS_INIT -> CS_ROUTING
2011-07-12 16:28:15.263803 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 [BREAK]
2011-07-12 16:28:15.263803 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/1002) State INIT going to sleep
2011-07-12 16:28:15.263803 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002) Running State Change CS_ROUTING
2011-07-12 16:28:15.263803 [DEBUG] switch_channel.c:1668 (sofia/internal/1002) Callstate Change DOWN -> RINGING
2011-07-12 16:28:15.263803 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1002) State ROUTING
2011-07-12 16:28:15.263803 [DEBUG] mod_sofia.c:147 sofia/internal/1002 SOFIA ROUTING
2011-07-12 16:28:15.263803 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/1002) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-07-12 16:28:15.263803 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 [BREAK]
2011-07-12 16:28:15.263803 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1002) State ROUTING going to sleep
2011-07-12 16:28:15.263803 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002) Running State Change CS_CONSUME_MEDIA
2011-07-12 16:28:15.263803 [DEBUG] switch_core_state_machine.c:383 (sofia/internal/1002) State CONSUME_MEDIA
2011-07-12 16:28:15.263803 [DEBUG] sofia.c:4760 Channel sofia/internal/1002 entering state [calling][0]
2011-07-12 16:28:15.263803 [DEBUG] switch_core_state_machine.c:383 (sofia/internal/1002) State CONSUME_MEDIA going to sleep
2011-07-12 16:28:15.327806 [INFO] sofia.c:739 sofia/internal/1002 Update Callee ID to "Outbound Call" <1002>
2011-07-12 16:28:15.327806 [DEBUG] sofia.c:4760 Channel sofia/internal/1002 entering state [proceeding][180]
2011-07-12 16:28:15.327806 [NOTICE] sofia.c:4838 Ring-Ready sofia/internal/1002!
2011-07-12 16:28:15.351808 [DEBUG] switch_ivr_originate.c:1150 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
2011-07-12 16:28:15.351808 [DEBUG] switch_core_codec.c:116 sofia/internal/1005 at 192.168.200.100 Push codec L16:70
2011-07-12 16:28:15.351808 [DEBUG] switch_ivr_originate.c:1182 Play Ringback File [local_stream://moh]
2011-07-12 16:28:15.351808 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/8000] 8000hz
2011-07-12 16:28:37.390068 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1003 at company-a.org] from ip 192.1
68.200.201
2011-07-12 16:28:37.477073 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1002 at company-a.org] from ip 192.1
68.200.201
2011-07-12 16:28:37.577079 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1001 at company-a.org] from ip 192.1
68.200.201
2011-07-12 16:28:45.011504 [DEBUG] switch_core_codec.c:141 sofia/internal/1005 at 192.168.200.100 Restore previous codec PCMU:0.
2011-07-12 16:28:45.011504 [DEBUG] switch_channel.c:2563 (sofia/internal/1002) Callstate Change RINGING -> HANGUP
2011-07-12 16:28:45.011504 [NOTICE] switch_ivr_originate.c:3329 Hangup sofia/internal/1002 [CS_CONSUME_MEDIA] [NO_ANSWER]
2011-07-12 16:28:45.011504 [DEBUG] switch_channel.c:2579 Send signal sofia/internal/1002 [KILL]
2011-07-12 16:28:45.011504 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 [BREAK]
2011-07-12 16:28:45.011504 [INFO] mod_dptools.c:2640 Originate Failed.  Cause: NO_ANSWER
2011-07-12 16:28:45.011504 [DEBUG] mod_dptools.c:2671 Failure causes [USER_BUSY]:  Cause: NO_ANSWER
EXECUTE sofia/internal/1005 at 192.168.200.100 answer()
EXECUTE sofia/internal/1005 at 192.168.200.100 sleep(1000)
2011-07-12 16:28:45.011504 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002) Running State Change CS_HANGUP
2011-07-12 16:28:45.011504 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1002) State HANGUP
2011-07-12 16:28:45.011504 [DEBUG] mod_sofia.c:457 Channel sofia/internal/1002 hanging up, cause: NO_ANSWER
2011-07-12 16:28:45.011504 [DEBUG] mod_sofia.c:510 Sending CANCEL to sofia/internal/1002
2011-07-12 16:28:45.011504 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1002 Standard HANGUP, cause: NO_ANSWER
2011-07-12 16:28:45.011504 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1002) State HANGUP going to sleep
2011-07-12 16:28:45.011504 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/1002) State Change CS_HANGUP -> CS_REPORTING
2011-07-12 16:28:45.011504 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 [BREAK]
2011-07-12 16:28:45.011504 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002) Running State Change CS_REPORTING
2011-07-12 16:28:45.011504 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1002) State REPORTING
2011-07-12 16:28:45.011504 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1002 Standard REPORTING, cause: NO_ANSWER
2011-07-12 16:28:45.011504 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1002) State REPORTING going to sleep
2011-07-12 16:28:45.011504 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/1002) State Change CS_REPORTING -> CS_DESTROY
2011-07-12 16:28:45.011504 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 [BREAK]
2011-07-12 16:28:45.011504 [DEBUG] switch_core_session.c:1288 Session 3 (sofia/internal/1002) Locked, Waiting on external entities
2011-07-12 16:28:45.011504 [NOTICE] switch_core_session.c:1306 Session 3 (sofia/internal/1002) Ended
2011-07-12 16:28:45.011504 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/1002 [CS_DESTROY]
2011-07-12 16:28:45.011504 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/1002) Callstate Change HANGUP -> DOWN
2011-07-12 16:28:45.011504 [DEBUG] switch_core_state_machine.c:457 (sofia/internal/1002) Running State Change CS_DESTROY
2011-07-12 16:28:45.011504 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1002) State DESTROY
2011-07-12 16:28:45.011504 [DEBUG] mod_sofia.c:362 sofia/internal/1002 SOFIA DESTROY
2011-07-12 16:28:45.011504 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1002 Standard DESTROY
2011-07-12 16:28:45.011504 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1002) State DESTROY going to sleep
EXECUTE sofia/internal/1005 at 192.168.200.100 play_and_get_digits(2 5 3 4000 # C:\VSWITCH/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav
other_ext_var \d+)
2011-07-12 16:28:46.017562 [DEBUG] switch_ivr_play_say.c:1278 Codec Activated L16 at 8000hz 1 channels 20ms
2011-07-12 16:28:47.751661 [DEBUG] switch_ivr_play_say.c:1648 done playing file
2011-07-12 16:28:51.771891 [ERR] mod_sndfile.c:194 Error Opening File [/invalid.wav] [System error : ???????????
]
2011-07-12 16:28:51.771891 [DEBUG] switch_ivr_play_say.c:1278 Codec Activated L16 at 8000hz 1 channels 20ms
2011-07-12 16:28:53.511990 [DEBUG] switch_ivr_play_say.c:1648 done playing file
2011-07-12 16:28:57.531220 [ERR] mod_sndfile.c:194 Error Opening File [/invalid.wav] [System error : ???????????
]
2011-07-12 16:28:57.531220 [DEBUG] switch_ivr_play_say.c:1278 Codec Activated L16 at 8000hz 1 channels 20ms

From admin at blindi.net  Tue Jul 12 12:47:41 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Tue, 12 Jul 2011 10:47:41 +0200 (CEST)
Subject: [Freeswitch-users] Why does group_confirm_file 3 times from?
Message-ID: <Pine.LNX.4.64.1107121032350.8901@tomback.antagus.de>

Hi guys,
I Create a callscreening-extions:
<extension name="callscreen">
<condition field="destination_number" expression="^callscreen">
  <action application="set" data="group_confirm_file=playback
  $${base_dir}/sounds/anrufe/achtung-dringend-call.wav"/>
   <action application="set" data="group_confirm_key=exec"/>
<action application="bridge"  data="loopback/0/operator1"/>
</condition>
</extension>

The Problem:
Fs plays before connecting this file 3 times from.
Can I change this?
It would be nice if fs this file plays only once.
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 admin at blindi.net  Tue Jul 12 12:54:47 2011
From: admin at blindi.net (Thomas Hoellriegel)
Date: Tue, 12 Jul 2011 10:54:47 +0200 (CEST)
Subject: [Freeswitch-users] Uptimecommand on cli?
In-Reply-To: <CAAp+uUF7+KgKf-XD5MVgY=SchTqu6Xx-TPKJPBH6FNtn5az5jA@mail.gmail.com>
References: <Pine.LNX.4.64.1107100246080.19434@tomback.antagus.de>
	<CA3E6492.18B90%krice@freeswitch.org>
	<CAAp+uUF7+KgKf-XD5MVgY=SchTqu6Xx-TPKJPBH6FNtn5az5jA@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1107121053010.8901@tomback.antagus.de>

Am 11.07.11 um 14:14 schrieb Frank Park:

> F2 button works well too.
Thanks, but status is my friend;). I woring under linux text only.

---------------
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 david.ponzone at ipeva.fr  Tue Jul 12 13:08:02 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 12 Jul 2011 11:08:02 +0200
Subject: [Freeswitch-users] How to configure for FS playing voice prompt
	in case the called extension is in a call.
In-Reply-To: <CAOTmbMwtVZJd_44rs7QH1GATSPygyfMJq2Ao-kgxw1x-L_vkVg@mail.gmail.com>
References: <CAOTmbMzMVtRjUvTj5UgHH7MG=XyHSWRvw7QT6G-tiLFd57H1Vw@mail.gmail.com>
	<CAFiqYuncFULQn=BnWDh2s0m6VhAX3aqN-zPbFaMD19cowfOERA@mail.gmail.com>
	<CAOTmbMwtVZJd_44rs7QH1GATSPygyfMJq2Ao-kgxw1x-L_vkVg@mail.gmail.com>
Message-ID: <5F28A4A5-726E-44EC-97CA-E00920F95114@ipeva.fr>

Check what your phone sends back when busy.

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

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

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




Le 12/07/2011 ? 10:31, fieldpeak a ?crit :

> Hi Steve,
> 
> Thanks for your kindly help, it works with below dial plan...
> However, for both user busy or no answer, it played the voice prompt...
> i would like it only prompts in case user busy but not no answer, although i configured the "failure_causes=USER_BUSY" as below dialplan, it looks not help...attached is a log...
> 
>     <extension name="Local_Extension">
>       <condition field="destination_number" expression="^(1\d{3})$">
>     <action application="set" data="dialed_extension=$1"/>
>     <action application="export" data="dialed_extension=$1"/>
>     <action application="set" data="ringback=${us-ring}"/>
>     <action application="set" data="transfer_ringback=$${hold_music}"/>
>     <action application="set" data="call_timeout=30"/>
>     
>     <action application="set" data="failure_causes=USER_BUSY"/>
>     
>     <action application="set" data="hangup_after_bridge=true"/>
>     
>     <!-- to enable voice mail, set continue_on_fail=true below -->
>     <action application="set" data="continue_on_fail=true"/>
>     <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>     <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>     <action application="bridge" data="sofia/internal/${dialed_extension}%${domain_name}"/>
>     <action application="answer"/>
>     
>     <action application="sleep" data="1000"/>
>     <!-- play a voice prompt e.g. 'the extension you dialed is busy now, please dial the other exsention' to replacing busy tone when the called extension is in a call... -->    
>     <action application="play_and_get_digits" data="2 5 3 4000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav other_ext_var \d+"/>
>     <action application="transfer" data="${other_ext_var} XML default"/>
>         
>       </condition>
>     </extension>
> 
> Regards,
> Charles
> 
> 
> 2011/7/11 Steven Ayre <steveayre at gmail.com>
> There are two ways....
> 
> Most phones will return a USER_BUSY release code if you bridge to a phone that's busy. If the phone does then you can continue_on_fail so that you (pre-)answer the call and play that message after the bridge. Something like:
> <extension name="Local_Extension">
>     <condition field="destination_number" expression="^(1\d{3})$">
>         <action application="set" data="dialed_extension=$1"/>
>         <action application="export" data="dialed_extension=$1"/>
>         <action application="set" data="call_timeout=30"/>
>         <action application="set" data="hangup_after_bridge=true"/> <!-- Hangup on successful call -->
>         <action application="set" data="continue_on_fail=true"/> <!-- Don't hangup on failed call, continue to IVR prompt -->
> 
>         <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>         <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>         <action application="bridge" data="user/${dialed_extension}%${domain_name}"/>
>         <!-- Put the IVR prompt here, see http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits -->
>     </condition>
> </extension>
> The 2nd option is that you use Limit to determine if another person is already on a call to the phone. That only really works if you run the the only servers that call the phone since it requires tracking all calls to the phone. It will sometimes work better though - especially if the phone has multiple lines. http://wiki.freeswitch.org/wiki/Limit
> 
> -Steve
> 
> 
> On 11 July 2011 03:25, fieldpeak <fieldpeak at gmail.com> wrote:
> Hi Gurus,
>     
> Could anyone advise how to realize have FS to play a voice prompt e.g. 'the extension you dialed is busy now, please dial the other exsention' to replacing busy tone when the called extension is in a call...
> 
> the details is below,
> 
> When calling to FS, the FS will play IVR "welcome to call us, please input the extension number, for operator please press 0",
> then the caller input the extension number, in case the extension is in a call, the caller will hear busy tone...
> it needs the system play "the extension you dailed is busy, please dial the other extension", currently my dial plan as following, could anyone advise how to change based on below dial plan or any other new dail plan can realize it...
> 
> Thanks a lot!
> 
>     <extension name="Local_Extension">
>       <condition field="destination_number" expression="^(1\d{3})$">
>     <action application="set" data="dialed_extension=$1"/>
>     <action application="export" data="dialed_extension=$1"/>
>     <action application="set" data="ringback=${us-ring}"/>
>     <action application="set" data="transfer_ringback=$${hold_music}"/>
>     <action application="set" data="call_timeout=30"/>
>     <action application="set" data="hangup_after_bridge=true"/>
>     
>     <!-- to enable voice mail, set continue_on_fail=true below -->
>     <action application="set" data="continue_on_fail=false"/>
>     <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>     <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>     <action application="bridge" data="sofia/internal/${dialed_extension}%${domain_name}"/>
>     <action application="answer"/>
>     
>     <action application="sleep" data="1000"/>
>     <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
>       </condition>
>     </extension>
>     
>     
>     <extension name="IVR" >
>        <condition field="destination_number" expression="^888$">
>            <action application="answer"/>
>            <action application="sleep" data="2000"/>
>            <action application="ivr" data="IVR"/>
>        </condition>
>     </extension>
>     
>         <!--  -->
>     <menu name="IVR"
>         greet-long="C:/VSWITCH/recordings/greeting_tts.wav"
>         greet-short="C:/VSWITCH/recordings/greeting_tts.wav"
>         invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>         exit-sound="voicemail/vm-goodbye.wav"
>         confirm-macro=""
>         confirm-key=""
>         tts-engine="flite"
>         tts-voice="rms"
>         confirm-attempts="3"
>         timeout="10000"
>         inter-digit-timeout="2000"
>         max-failures="3"
>         max-timeouts="3"
>         digit-len="5">
>         <entry action="menu-exec-app" digits="/(^\*\d{3,5}$|^\d{3,5}$)/" param="transfer $1 XML default"/>
>     </menu>
>     
> Regards,
> Charles
> 
> _______________________________________________
> 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
> 
> 
> 
> _______________________________________________
> 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
> 
> 
> <log_no_answer.txt>_______________________________________________
> 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
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From va_mclean at yahoo.com  Tue Jul 12 15:04:11 2011
From: va_mclean at yahoo.com (Mclean Va)
Date: Tue, 12 Jul 2011 04:04:11 -0700 (PDT)
Subject: [Freeswitch-users] Google Voice for Windows version of FS
Message-ID: <1310468651.13286.YahooMailNeo@web121612.mail.ne1.yahoo.com>

Does anyone have Windows versions of the latest FS working for Google voice?

I have tried pre-built binaries and build from source with Dingaling but always receive "Unauthorized" errors.

Do I need to put my external IP address somewhere to use GV?

Thanks
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From boris at tagnet.ru  Tue Jul 12 16:46:10 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 12 Jul 2011 18:46:10 +0600
Subject: [Freeswitch-users] DISA question
Message-ID: <4E1C4212.5010409@tagnet.ru>

Hello!

     Original DISA example 
(http://wiki.freeswitch.org/wiki/Examples_disa.js) uses 'transfer' 
application after the number is collected. In this case log record for 
the A leg contains original caller_id_number and destination_number 
collected from DISA. Is this possible to use another application instead 
of transfer (may be originate?)? This application should do another call 
(possible originated from a dummy gateway) so I have 2 A leg records? I 
read about 'originate' application but can't understand if it may be 
usefull.

-- 
Regards,
Boris




From roger.castaldo at gmail.com  Tue Jul 12 17:22:03 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Tue, 12 Jul 2011 09:22:03 -0400
Subject: [Freeswitch-users] Configuration Server
In-Reply-To: <CAFYZzbvVzfRt60neJC+DyDQMpsH+iWt7Ubv6YWALjN1fdf4fbw@mail.gmail.com>
References: <CAAYf2Lvv1f8QBmDL9scJm45KcDA7CoVbCmGOcSwBvAH8bnjMVA@mail.gmail.com>
	<CAFYZzbvVzfRt60neJC+DyDQMpsH+iWt7Ubv6YWALjN1fdf4fbw@mail.gmail.com>
Message-ID: <CAAYf2Luggy0mghz2EFbZFpmOc1dW4M9-iSpjCaS3zpSb1Nj7Og@mail.gmail.com>

Thank you Michel for the offer of help but there seems to be a lack of
interest so I suppose I shall continue puttering away at it and see if some
developers get interested later on when I have a more viable stable release
that could peak more interest in it being open source.  I will, however,
possibly look at releasing my socket code as an open source library when
done which will allow C# apps to connect to freeswitch servers via the event
socket and control/monitor the servers through those connections.

On Wed, Jul 6, 2011 at 1:37 PM, Michel Daggelinckx <
michel.daggelinckx at gmail.com> wrote:

> hi,
> I can do testing
>
> On Wed, Jul 6, 2011 at 4:09 PM, Roger Castaldo <roger.castaldo at gmail.com>wrote:
>
>> Hi everyone, I have been following freeswitch for quite some time now and
>> working on an ever evolving configuration server for it, which has now taken
>> yet another branch, that instead of writing xml configuration files now is
>> attempting to control the call flow itself using the outbound socket.
>> Currently it is a closed source project, originally because of some of my
>> closed source libraries I was using to develop it, but since they have
>> become open source I was wondering if there is any interest in working with
>> me on the development/testing of the product.  It is written in C# and is
>> entirely web based, using ajax calls and web services for everything.  I am
>> using a Relational Mapping library of my own design to map the data to the
>> database (currently a firebirdsql database).
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 jeff at jefflenk.com  Tue Jul 12 18:05:18 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 12 Jul 2011 07:05:18 -0700 (PDT)
Subject: [Freeswitch-users] Google Voice for Windows version of FS
In-Reply-To: <1310468651.13286.YahooMailNeo@web121612.mail.ne1.yahoo.com>
References: <1310468651.13286.YahooMailNeo@web121612.mail.ne1.yahoo.com>
Message-ID: <1310479518002-6575189.post@n2.nabble.com>

You will have to build from source as the prebuilt binaries do not have
gnutls built-in. See
http://wiki.freeswitch.org/wiki/Dingaling#Windows_using_MSVC_2008 for some
notes on this.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Google-Voice-for-Windows-version-of-FS-tp6574607p6575189.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From va_mclean at yahoo.com  Tue Jul 12 19:00:54 2011
From: va_mclean at yahoo.com (Mclean Va)
Date: Tue, 12 Jul 2011 08:00:54 -0700 (PDT)
Subject: [Freeswitch-users] Google Voice for Windows version of FS
In-Reply-To: <1310479518002-6575189.post@n2.nabble.com>
References: <1310468651.13286.YahooMailNeo@web121612.mail.ne1.yahoo.com>
	<1310479518002-6575189.post@n2.nabble.com>
Message-ID: <1310482854.79304.YahooMailNeo@web121610.mail.ne1.yahoo.com>

I did read the link and thought it only applied to VS2008, I was using VS2010.

Thanks





________________________________
From: Jeff Lenk <jeff at jefflenk.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Tuesday, July 12, 2011 10:05 AM
Subject: Re: [Freeswitch-users] Google Voice for Windows version of FS

You will have to build from source as the prebuilt binaries do not have
gnutls built-in. See
http://wiki.freeswitch.org/wiki/Dingaling#Windows_using_MSVC_2008 for some
notes on this.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Google-Voice-for-Windows-version-of-FS-tp6574607p6575189.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
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
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From jeff at jefflenk.com  Tue Jul 12 19:18:05 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 12 Jul 2011 08:18:05 -0700 (PDT)
Subject: [Freeswitch-users] Google Voice for Windows version of FS
In-Reply-To: <1310482854.79304.YahooMailNeo@web121610.mail.ne1.yahoo.com>
References: <1310468651.13286.YahooMailNeo@web121612.mail.ne1.yahoo.com>
	<1310479518002-6575189.post@n2.nabble.com>
	<1310482854.79304.YahooMailNeo@web121610.mail.ne1.yahoo.com>
Message-ID: <1310483885842-6575490.post@n2.nabble.com>

yeah the note specifically mentions 2008 but the same principles apply to
2010.
Using precompiled external libs is not something that we support(all the
current code in the project is built from src) but if you wish to do that
yourself thats up to you.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Google-Voice-for-Windows-version-of-FS-tp6574607p6575490.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From va_mclean at yahoo.com  Tue Jul 12 19:33:42 2011
From: va_mclean at yahoo.com (Mclean Va)
Date: Tue, 12 Jul 2011 08:33:42 -0700 (PDT)
Subject: [Freeswitch-users] Google Voice for Windows version of FS
In-Reply-To: <1310483885842-6575490.post@n2.nabble.com>
References: <1310468651.13286.YahooMailNeo@web121612.mail.ne1.yahoo.com>
	<1310479518002-6575189.post@n2.nabble.com>
	<1310482854.79304.YahooMailNeo@web121610.mail.ne1.yahoo.com>
	<1310483885842-6575490.post@n2.nabble.com>
Message-ID: <1310484822.88080.YahooMailNeo@web121617.mail.ne1.yahoo.com>

Thanks for clarifications, I'll rebuild from source to include gnutls.



________________________________
From: Jeff Lenk <jeff at jefflenk.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Tuesday, July 12, 2011 11:18 AM
Subject: Re: [Freeswitch-users] Google Voice for Windows version of FS

yeah the note specifically mentions 2008 but the same principles apply to
2010.
Using precompiled external libs is not something that we support(all the
current code in the project is built from src) but if you wish to do that
yourself thats up to you.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Google-Voice-for-Windows-version-of-FS-tp6574607p6575490.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
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
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From mario_fs at mgtech.com  Tue Jul 12 19:35:46 2011
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 12 Jul 2011 08:35:46 -0700
Subject: [Freeswitch-users] Outgoing routing question
Message-ID: <38F8615D-6378-4BCC-A53B-9F15C5415C35@mgtech.com>

I currently have 5 lines and 5 xml files in the dialplan/default directory. An outgoing call requires one of 5 prefixes. What I want is to have Freeswitch automatically determine the correct gateway based on the called number and if it's not a known number use the dialed prefix or if none fail the call. I also would like to reduce the redundancy in the xml files, each of the 5 has the same thing except for the prefix and gateway. Can I do this:

1. Have an xml named 01_xxx that strips things like "*31#" and create a variable.
2. Auto extend the variable with local area code if only 7 digits.
3. Set another variable to a gateway name based on the number (will just have a list of numbers in the xml file)

The question is: when it ends will another xml execute (02_xxx) that calls the gateway or do I have the call the gateway in this one and only one xml file since I am trying to intercept everything there?

Am I on the right track? Finally, I saw mention of an enum script in the wiki but could not find it in the directory. Thanks,
Mario G

From boris at tagnet.ru  Tue Jul 12 19:42:31 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 12 Jul 2011 21:42:31 +0600
Subject: [Freeswitch-users] Outgoing routing question
In-Reply-To: <38F8615D-6378-4BCC-A53B-9F15C5415C35@mgtech.com>
References: <38F8615D-6378-4BCC-A53B-9F15C5415C35@mgtech.com>
Message-ID: <4E1C6B67.3080104@tagnet.ru>

Hello!

     Yes, You can with use of continue=true for the extension. Also 
please read about inline=true for application 'set'. It will be usefull 
if You want to use your variables in nexts tests.
> I currently have 5 lines and 5 xml files in the dialplan/default directory. An outgoing call requires one of 5 prefixes. What I want is to have Freeswitch automatically determine the correct gateway based on the called number and if it's not a known number use the dialed prefix or if none fail the call. I also would like to reduce the redundancy in the xml files, each of the 5 has the same thing except for the prefix and gateway. Can I do this:
>
> 1. Have an xml named 01_xxx that strips things like "*31#" and create a variable.
> 2. Auto extend the variable with local area code if only 7 digits.
> 3. Set another variable to a gateway name based on the number (will just have a list of numbers in the xml file)
>
> The question is: when it ends will another xml execute (02_xxx) that calls the gateway or do I have the call the gateway in this one and only one xml file since I am trying to intercept everything there?
>
> Am I on the right track? Finally, I saw mention of an enum script in the wiki but could not find it in the directory. Thanks,
> Mario G
> _______________________________________________
> 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


-- 
Regards,
Boris




From mario_fs at mgtech.com  Tue Jul 12 19:51:36 2011
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 12 Jul 2011 08:51:36 -0700
Subject: [Freeswitch-users] Outgoing routing question
In-Reply-To: <4E1C6B67.3080104@tagnet.ru>
References: <38F8615D-6378-4BCC-A53B-9F15C5415C35@mgtech.com>
	<4E1C6B67.3080104@tagnet.ru>
Message-ID: <2F23F323-56D5-4552-9EE0-D400FB018C46@mgtech.com>

Thank you!  I am reading about them right now. Looks like what I need.

On Jul 12, 2011, at 8:42 AM, Boris Kovalenko wrote:

> Hello!
> 
>     Yes, You can with use of continue=true for the extension. Also 
> please read about inline=true for application 'set'. It will be usefull 
> if You want to use your variables in nexts tests.
>> I currently have 5 lines and 5 xml files in the dialplan/default directory. An outgoing call requires one of 5 prefixes. What I want is to have Freeswitch automatically determine the correct gateway based on the called number and if it's not a known number use the dialed prefix or if none fail the call. I also would like to reduce the redundancy in the xml files, each of the 5 has the same thing except for the prefix and gateway. Can I do this:
>> 
>> 1. Have an xml named 01_xxx that strips things like "*31#" and create a variable.
>> 2. Auto extend the variable with local area code if only 7 digits.
>> 3. Set another variable to a gateway name based on the number (will just have a list of numbers in the xml file)
>> 
>> The question is: when it ends will another xml execute (02_xxx) that calls the gateway or do I have the call the gateway in this one and only one xml file since I am trying to intercept everything there?
>> 
>> Am I on the right track? Finally, I saw mention of an enum script in the wiki but could not find it in the directory. Thanks,
>> Mario G
>> _______________________________________________
>> 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
> 
> 
> -- 
> Regards,
> Boris
> 
> 
> 
> _______________________________________________
> 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 lists at ticm.com  Tue Jul 12 10:42:33 2011
From: lists at ticm.com (Bret Watson)
Date: Tue, 12 Jul 2011 14:42:33 +0800
Subject: [Freeswitch-users] weirdness.. low audio levels to provider..
Message-ID: <4e1bece4.43832a0a.1112.0019@mx.google.com>

Hi All,
I've got a weird one..

Calls between endpoints are fine, but calls outside suffer from my 
side having very low audio levels. Both for incoming calls and outgoing calls.

I tried the set_audio_level write =4 bit but no effect.

My provider is pennytel - previously I've gone direct from the 3102 
with no level issues

Any hints? what would be useful for diagnosis?

Thanks

Bret



From stviper at gmail.com  Tue Jul 12 11:24:59 2011
From: stviper at gmail.com (=?ISO-8859-2?B?qXRlZmFuIMh1ZGFp?=)
Date: Tue, 12 Jul 2011 09:24:59 +0200
Subject: [Freeswitch-users] Inter-digits time for dialling ISDN
Message-ID: <CAP3+0r3VVjL4Lh1OtViQpwg0FC_KoF3u92F+ovcoLjWTfm3+5w@mail.gmail.com>

Hi FreeSwitch community,

We are using FreeSwitch with FreeTDM on Windows 7 machine. In the machine is
installed Sangoma A102.

My question is how or where to configure "inter-digits time" for dialling of
the number?

Because when we dial some number FS suddenly decides that it has all digits
and it changes state from COLLECT to RING.
You can see it in log: It gets first 4 digits "0015" and when he finished
collecting it had 5 digits "00151".

14:44:51.785156 [INFO] ftmod_sangoma_isdn_stack_rcv.c:75 [s1c20][1:21]
Received SETUP (suId:1 suInstId:0 spInstId:1)
14:44:51.785156 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:57 [s1c20][1:21]
Processing SETUP (suId:1 suInstId:0 spInstId:1)
14:44:51.785156 [INFO] ftmod_sangoma_isdn_stack_hndl.c:142 [s1c20][1:21]
Incoming call: Called No:[0015] Calling No:[99994]
14:44:51.785156 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:197 [s1c20][1:21]
Changed state from DOWN to COLLECT
14:44:51.785156 [DEBUG] ftdm_state.c:511 [s1c20][1:21] Executing state
processor for COLLECT
14:44:51.785156 [DEBUG] ftmod_sangoma_isdn.c:623 [s1c20][1:21] processing
state change to COLLECT
14:44:51.785156 [INFO] ftmod_sangoma_isdn_stack_out.c:102 [s1c20][1:21]
Sending SETUP ACK (suId:1 suInstId:1 spInstId:1 dchan:1 ces:0)
14:44:51.785156 [DEBUG] ftmod_sangoma_isdn.c:884 [s1c20][1:21] Completed
state change from DOWN to COLLECT in 0ms
14:44:51.928710 [INFO] ftmod_sangoma_isdn_stack_rcv.c:172 [s1c20][1:21]
Received INFO (suId:1 suInstId:1 spInstId:1 ces:0)
14:44:51.928710 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:368 [s1c20][1:21]
Processing INFO (suId:1 suInstId:1 spInstId:1 ces:0)
14:44:51.928710 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:464 [s1c20][1:21]
Processing INFO (suId:1 suInstId:1 spInstId:1)
14:44:51.928710 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:477 [s1c20][1:21]
Changed state from COLLECT to RING
14:44:51.928710 [DEBUG] ftdm_state.c:511 [s1c20][1:21] Executing state
processor for RING
14:44:51.928710 [DEBUG] ftmod_sangoma_isdn.c:623 [s1c20][1:21] processing
state change to RING
14:44:51.928710 [DEBUG] ftmod_sangoma_isdn.c:647 [s1c20][1:21] Sending
incoming call from 99994 to 00151 to FTDM core
14:44:51.928710 [DEBUG] ftmod_sangoma_isdn.c:884 [s1c20][1:21] Completed
state change from COLLECT to RING in 0ms
14:44:51.928710 [DEBUG] mod_freetdm.c:2268 got clear channel sig [START]
14:44:51.928710 [DEBUG] ftdm_io.c:3061 [s1c20][1:21] Enabled software DTMF
detector
14:44:51.928710 [DEBUG] mod_freetdm.c:391 Set codec PCMA 20ms
14:44:51.928710 [DEBUG] mod_freetdm.c:1625 Connect inbound channel
FreeTDM/1:20/00151

My freetdm.conf.xml

<config_profiles>
<profile name="myProfile">
<param name="switchtype" value="euroisdn"/>
<param name="signalling" value="cpe"/>
<param name="overlap" value="yes"/>
<param name="min_digits" value="0"/>
</profile>
</config_profiles>
 <sangoma_pri_spans>
<span name="wanpipe1" cfgprofile="myProfile">
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
<span name="wanpipe2" cfgprofile="myProfile">
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
</sangoma_pri_spans>

Thanks for help !!!

Regards,
Stefan
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From rgelfand2 at gmail.com  Tue Jul 12 20:53:39 2011
From: rgelfand2 at gmail.com (Roman Gelfand)
Date: Tue, 12 Jul 2011 12:53:39 -0400
Subject: [Freeswitch-users] PocketSwitch Module
Message-ID: <CAJbW+rk1i4kmJAs7+-XP_R7cyaXYN1XuTNq359rviKprecsEgQ@mail.gmail.com>

Has anyone used this module for speech recognition prompts and database
interaction?  If so, what are your impressions?

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

Hello all!

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

Thanks,
Michael
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From philippe at ppmt.org  Tue Jul 12 22:37:53 2011
From: philippe at ppmt.org (Philippe Le Toquin)
Date: Tue, 12 Jul 2011 14:37:53 -0400
Subject: [Freeswitch-users] WIKI and command printout explanation
Message-ID: <CACUErJdMVha_3rPON6jmiVWRBVLJuoP+cCgUY=v1iOKVNhiRbw@mail.gmail.com>

Hello,

I am trying to get more familiar wit FS cli and found the page that list the
commands and their usage.

What I am looking for is an explanation of their printout. Some of them are
easy to understand but other not so (not to me at least)

Is there a wiki page that explains

For example the simple status command give me that:

status
UP 0 years, 29 days, 17 hours, 38 minutes, 3 seconds, 716 milliseconds, 375
microseconds
512 session(s) since startup
0 session(s) 0/30
1000 session(s) max
min idle cpu 0.00/100.00


I had 512 session since startup.....fair enough but then it says "1000
session(s) max"

What does it means? that FS will stop working after 10000 sessions or that I
can't get more than 1000 in parallel?

Regards

/Philippe
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From gmaruzz at gmail.com  Tue Jul 12 23:03:18 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 12 Jul 2011 21:03:18 +0200
Subject: [Freeswitch-users] WIKI and command printout explanation
In-Reply-To: <CACUErJdMVha_3rPON6jmiVWRBVLJuoP+cCgUY=v1iOKVNhiRbw@mail.gmail.com>
References: <CACUErJdMVha_3rPON6jmiVWRBVLJuoP+cCgUY=v1iOKVNhiRbw@mail.gmail.com>
Message-ID: <CALXCt0o5uH7tqV5e6=C4XWRHmAa_=mhVdZre5ExL0Uhvm3UAxQ@mail.gmail.com>

no more than 1000 in parallel.

Before you ask: min idle cpu gives you at what idle cpu will stop accepting
calls (in your case at 0 idle, so 100percent busy) and what was the average
idle in last 5 minutes (in your case 100 percent idle, 0 percent busy)

0 sessions are ongoing, with a max of 30 new sessions /second allowed to
spawn (eg: no more than 30 new session second ramp up)

all this is configurable

-giovanni



On Tue, Jul 12, 2011 at 8:37 PM, Philippe Le Toquin <philippe at ppmt.org>wrote:

> Hello,
>
> I am trying to get more familiar wit FS cli and found the page that list
> the commands and their usage.
>
> What I am looking for is an explanation of their printout. Some of them are
> easy to understand but other not so (not to me at least)
>
> Is there a wiki page that explains
>
> For example the simple status command give me that:
>
> status
> UP 0 years, 29 days, 17 hours, 38 minutes, 3 seconds, 716 milliseconds, 375
> microseconds
> 512 session(s) since startup
> 0 session(s) 0/30
> 1000 session(s) max
> min idle cpu 0.00/100.00
>
>
> I had 512 session since startup.....fair enough but then it says "1000
> session(s) max"
>
> What does it means? that FS will stop working after 10000 sessions or that
> I can't get more than 1000 in parallel?
>
> Regards
>
> /Philippe
>
> _______________________________________________
> 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
>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From djbinter at gmail.com  Tue Jul 12 23:06:11 2011
From: djbinter at gmail.com (DJB International)
Date: Tue, 12 Jul 2011 12:06:11 -0700
Subject: [Freeswitch-users] WIKI and command printout explanation
In-Reply-To: <CALXCt0o5uH7tqV5e6=C4XWRHmAa_=mhVdZre5ExL0Uhvm3UAxQ@mail.gmail.com>
References: <CACUErJdMVha_3rPON6jmiVWRBVLJuoP+cCgUY=v1iOKVNhiRbw@mail.gmail.com>
	<CALXCt0o5uH7tqV5e6=C4XWRHmAa_=mhVdZre5ExL0Uhvm3UAxQ@mail.gmail.com>
Message-ID: <CAAB3jWvOZbzqxzVz-jKYTVyFV49E3ZnOxtVTYdfGE2LyLZWh8A@mail.gmail.com>

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

-djbinter

On Tue, Jul 12, 2011 at 12:03 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> no more than 1000 in parallel.
>
> Before you ask: min idle cpu gives you at what idle cpu will stop accepting
> calls (in your case at 0 idle, so 100percent busy) and what was the average
> idle in last 5 minutes (in your case 100 percent idle, 0 percent busy)
>
> 0 sessions are ongoing, with a max of 30 new sessions /second allowed to
> spawn (eg: no more than 30 new session second ramp up)
>
> all this is configurable
>
> -giovanni
>
>
>
> On Tue, Jul 12, 2011 at 8:37 PM, Philippe Le Toquin <philippe at ppmt.org>wrote:
>
>> Hello,
>>
>> I am trying to get more familiar wit FS cli and found the page that list
>> the commands and their usage.
>>
>> What I am looking for is an explanation of their printout. Some of them
>> are easy to understand but other not so (not to me at least)
>>
>> Is there a wiki page that explains
>>
>> For example the simple status command give me that:
>>
>> status
>> UP 0 years, 29 days, 17 hours, 38 minutes, 3 seconds, 716 milliseconds,
>> 375 microseconds
>> 512 session(s) since startup
>> 0 session(s) 0/30
>> 1000 session(s) max
>> min idle cpu 0.00/100.00
>>
>>
>> I had 512 session since startup.....fair enough but then it says "1000
>> session(s) max"
>>
>> What does it means? that FS will stop working after 10000 sessions or that
>> I can't get more than 1000 in parallel?
>>
>> Regards
>>
>> /Philippe
>>
>> _______________________________________________
>> 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
>>
>>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
> _______________________________________________
> 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 steveayre at gmail.com  Tue Jul 12 23:25:10 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 12 Jul 2011 20:25:10 +0100
Subject: [Freeswitch-users] WIKI and command printout explanation
In-Reply-To: <CACUErJdMVha_3rPON6jmiVWRBVLJuoP+cCgUY=v1iOKVNhiRbw@mail.gmail.com>
References: <CACUErJdMVha_3rPON6jmiVWRBVLJuoP+cCgUY=v1iOKVNhiRbw@mail.gmail.com>
Message-ID: <18621059-CFD8-4849-A9A9-D7527964B0D6@gmail.com>

http://wiki.FreeSWITCH.org/

All API commands (the cli ones) are in modules you load.

Most of the 'standard' ones are in mod_commands - search for it in the wiki. The commands are documented there.

Other module specific commands will be documented on the module's page - eg the 'sofia' command is documented on the mod_sofia page.

Steve on iPhone

On 12 Jul 2011, at 19:37, Philippe Le Toquin <philippe at ppmt.org> wrote:

> Hello,
> 
> I am trying to get more familiar wit FS cli and found the page that list the commands and their usage.
> 
> What I am looking for is an explanation of their printout. Some of them are easy to understand but other not so (not to me at least)
> 
> Is there a wiki page that explains 
> 
> For example the simple status command give me that:
> 
> status
> UP 0 years, 29 days, 17 hours, 38 minutes, 3 seconds, 716 milliseconds, 375 microseconds
> 512 session(s) since startup
> 0 session(s) 0/30
> 1000 session(s) max
> min idle cpu 0.00/100.00
> 
> 
> I had 512 session since startup.....fair enough but then it says "1000 session(s) max"
> 
> What does it means? that FS will stop working after 10000 sessions or that I can't get more than 1000 in parallel?
> 
> Regards
> 
> /Philippe
> _______________________________________________
> 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 mario_fs at mgtech.com  Tue Jul 12 23:31:53 2011
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 12 Jul 2011 12:31:53 -0700
Subject: [Freeswitch-users] An easy way to test a list of numbers in
	condition?
Message-ID: <887F35B0-7832-4492-AB6A-56F782E6DCED@mgtech.com>

I am trying to find an easy way to test multiple numbers in a condition without putting all the numbers on one line. There can be 25 numbers or more. Preferably have comments. I looked at many web sites on XML, etc. but could not find a way to continue lines. Freeswitch treats everything in the expression including all spaces as part of the expression. Anyone know of a good way to do this? Thanks,
Mario G

  <extension name="out.group1">
    <condition field="${destination_number_w}" expression="^1111111111|             <!-- joe -->
                                                            2222222222|3333333333|  <!-- jon -->
                                                            4444444444|5555555555   <!-- pam -->
                                                                        $">
      <action application="set" data="destination_number_w=1$1"/>
      <action application="bridge" data="sofia/gateway/group1/1$1"/>
    </condition>
  </extension>

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From msc at freeswitch.org  Wed Jul 13 01:35:18 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Jul 2011 14:35:18 -0700
Subject: [Freeswitch-users] An easy way to test a list of numbers in
	condition?
In-Reply-To: <887F35B0-7832-4492-AB6A-56F782E6DCED@mgtech.com>
References: <887F35B0-7832-4492-AB6A-56F782E6DCED@mgtech.com>
Message-ID: <CAKzWOxV-AYChw5YfcCg=MrsRD-gZQ7BZhw9Y=QNbpWfqCH3h_Q@mail.gmail.com>

If you need to test this many numbers then you are better of with something
other than regular expressions. I would investigate mod_xml_curl or mod_lcr.
An alternative would be to write a small Lua script that checks the
destination_number against a list.

-MC

On Tue, Jul 12, 2011 at 12:31 PM, Mario G <mario_fs at mgtech.com> wrote:

> I am trying to find an easy way to test multiple numbers in a condition
> without putting all the numbers on one line. There can be 25 numbers or
> more. Preferably have comments. I looked at many web sites on XML, etc. but
> could not find a way to continue lines. Freeswitch treats everything in the
> expression including all spaces as part of the expression. Anyone know of a
> good way to do this? Thanks,
> Mario G
>
>   <extension name="out.group1">
>     <condition field="${destination_number_w}" expression="^1111111111|
>           <!-- joe -->
>
> 2222222222|3333333333|  <!-- jon -->
>
> 4444444444|5555555555   <!-- pam -->
>                                                                         $"
> >
>       <action application="set" data="destination_number_w=1$1"/>
>       <action application="bridge" data="sofia/gateway/group1/1$1"/>
>     </condition>
>   </extension>
>
>
> _______________________________________________
> 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 avi at avimarcus.net  Wed Jul 13 02:12:43 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 13 Jul 2011 01:12:43 +0300
Subject: [Freeswitch-users] An easy way to test a list of numbers in
	condition?
In-Reply-To: <CAKzWOxV-AYChw5YfcCg=MrsRD-gZQ7BZhw9Y=QNbpWfqCH3h_Q@mail.gmail.com>
References: <887F35B0-7832-4492-AB6A-56F782E6DCED@mgtech.com>
	<CAKzWOxV-AYChw5YfcCg=MrsRD-gZQ7BZhw9Y=QNbpWfqCH3h_Q@mail.gmail.com>
Message-ID: <CALNZuwkVRqFseBeWrS1vrvhuEb0Qz4aCBsrGy=xBgtkvAJVfCw@mail.gmail.com>

A simple array check with lua, or a sql lookup with lua or mod_odbc_query
from the git contrib would probably make this easy.
-Avi

On Wed, Jul 13, 2011 at 12:35 AM, Michael Collins <msc at freeswitch.org>wrote:

> If you need to test this many numbers then you are better of with something
> other than regular expressions. I would investigate mod_xml_curl or mod_lcr.
> An alternative would be to write a small Lua script that checks the
> destination_number against a list.
>
> -MC
>
> On Tue, Jul 12, 2011 at 12:31 PM, Mario G <mario_fs at mgtech.com> wrote:
>
>> I am trying to find an easy way to test multiple numbers in a condition
>> without putting all the numbers on one line. There can be 25 numbers or
>> more. Preferably have comments. I looked at many web sites on XML, etc. but
>> could not find a way to continue lines. Freeswitch treats everything in the
>> expression including all spaces as part of the expression. Anyone know of a
>> good way to do this? Thanks,
>> Mario G
>>
>>   <extension name="out.group1">
>>     <condition field="${destination_number_w}" expression="^1111111111|
>>           <!-- joe -->
>>
>> 2222222222|3333333333|  <!-- jon -->
>>
>> 4444444444|5555555555   <!-- pam -->
>>                                                                         $"
>> >
>>       <action application="set" data="destination_number_w=1$1"/>
>>       <action application="bridge" data="sofia/gateway/group1/1$1"/>
>>     </condition>
>>   </extension>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 mario_fs at mgtech.com  Wed Jul 13 02:40:11 2011
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 12 Jul 2011 15:40:11 -0700
Subject: [Freeswitch-users] An easy way to test a list of numbers in
	condition?
In-Reply-To: <CALNZuwkVRqFseBeWrS1vrvhuEb0Qz4aCBsrGy=xBgtkvAJVfCw@mail.gmail.com>
References: <887F35B0-7832-4492-AB6A-56F782E6DCED@mgtech.com>
	<CAKzWOxV-AYChw5YfcCg=MrsRD-gZQ7BZhw9Y=QNbpWfqCH3h_Q@mail.gmail.com>
	<CALNZuwkVRqFseBeWrS1vrvhuEb0Qz4aCBsrGy=xBgtkvAJVfCw@mail.gmail.com>
Message-ID: <A3E5055E-C01D-475A-8836-280464D1F323@mgtech.com>

I was trying to avoid adding more software to OS X and having to learn new stuff just for a couple of groups of numbers... I tried many things and the best I could do was put them all on on line with a comment below for each name. Thanks.

On Jul 12, 2011, at 3:12 PM, Avi Marcus wrote:

> A simple array check with lua, or a sql lookup with lua or mod_odbc_query from the git contrib would probably make this easy.
> -Avi
> 
> On Wed, Jul 13, 2011 at 12:35 AM, Michael Collins <msc at freeswitch.org> wrote:
> If you need to test this many numbers then you are better of with something other than regular expressions. I would investigate mod_xml_curl or mod_lcr. An alternative would be to write a small Lua script that checks the destination_number against a list.
> 
> -MC
> 
> On Tue, Jul 12, 2011 at 12:31 PM, Mario G <mario_fs at mgtech.com> wrote:
> I am trying to find an easy way to test multiple numbers in a condition without putting all the numbers on one line. There can be 25 numbers or more. Preferably have comments. I looked at many web sites on XML, etc. but could not find a way to continue lines. Freeswitch treats everything in the expression including all spaces as part of the expression. Anyone know of a good way to do this? Thanks,
> Mario G
> 
>   <extension name="out.group1">
>     <condition field="${destination_number_w}" expression="^1111111111|             <!-- joe -->
>                                                             2222222222|3333333333|  <!-- jon -->
>                                                             4444444444|5555555555   <!-- pam -->
>                                                                         $">
>       <action application="set" data="destination_number_w=1$1"/>
>       <action application="bridge" data="sofia/gateway/group1/1$1"/>
>     </condition>
>   </extension>
> 
> 
> _______________________________________________
> 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
> 
> 
> 
> _______________________________________________
> 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
> 
> 
> _______________________________________________
> 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 va_mclean at yahoo.com  Wed Jul 13 02:57:32 2011
From: va_mclean at yahoo.com (Mclean Va)
Date: Tue, 12 Jul 2011 15:57:32 -0700 (PDT)
Subject: [Freeswitch-users] Google Voice for Windows version of FS
In-Reply-To: <1310483885842-6575490.post@n2.nabble.com>
References: <1310468651.13286.YahooMailNeo@web121612.mail.ne1.yahoo.com>
	<1310479518002-6575189.post@n2.nabble.com>
	<1310482854.79304.YahooMailNeo@web121610.mail.ne1.yahoo.com>
	<1310483885842-6575490.post@n2.nabble.com>
Message-ID: <1310511452.58175.YahooMailNeo@web121618.mail.ne1.yahoo.com>

I re-build FS with gnutlc and copied dlls to release folder according to the link, I still got "not authorized" errors:

2011-07-12 18:41:52.953125 [NOTICE] libdingaling.c:1373 SEND:
-------------------------------------------------------------------------------
<iq xmlns="jabber:client" from="XXXXXXXX at gmail.com/gtalk" to="+1703xxxxxxx at vo
ice.google.com" type="set" id="301">
<ses:session xmlns:ses="http://www.google.com/session" type="transport-info" i
d="6126364944" initiator="XXXXXXX at gmail.com/gtalk">
<transport xmlns="http://www.google.com/transport/p2p">
?     <candidate name="rtp" address="(my extern IP)" port="16774" username="lFSpOF
kF3fncvj9j" password="lFSpOFkF3fncvj9j" preference="1.0" protocol="udp" type="st
un" network="0" generation="0"></candidate>
</transport>
</ses:session>
</iq>

2011-07-12 18:41:52.968750 [INFO] libdingaling.c:1371 RECV:
-------------------------------------------------------------------------------
<stream:error>
<not-authorized xmlns="urn:ietf:params:xml:ns:xmpp-streams"></not-authorized>
</stream:error>

I don't understand in SEND string, why there are? id="6126364944" ? username="lFSpOF
kF3fncvj9j" password="lFSpOFkF3fncvj9j". I did not set those values, why username and password are same. 


Thanks for any help.




________________________________
From: Jeff Lenk <jeff at jefflenk.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Tuesday, July 12, 2011 10:18 AM
Subject: Re: [Freeswitch-users] Google Voice for Windows version of FS

yeah the note specifically mentions 2008 but the same principles apply to
2010.
Using precompiled external libs is not something that we support(all the
current code in the project is built from src) but if you wish to do that
yourself thats up to you.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Google-Voice-for-Windows-version-of-FS-tp6574607p6575490.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

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

FreeSWITCH-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From msc at freeswitch.org  Wed Jul 13 03:39:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Jul 2011 16:39:37 -0700
Subject: [Freeswitch-users] An easy way to test a list of numbers in
	condition?
In-Reply-To: <A3E5055E-C01D-475A-8836-280464D1F323@mgtech.com>
References: <887F35B0-7832-4492-AB6A-56F782E6DCED@mgtech.com>
	<CAKzWOxV-AYChw5YfcCg=MrsRD-gZQ7BZhw9Y=QNbpWfqCH3h_Q@mail.gmail.com>
	<CALNZuwkVRqFseBeWrS1vrvhuEb0Qz4aCBsrGy=xBgtkvAJVfCw@mail.gmail.com>
	<A3E5055E-C01D-475A-8836-280464D1F323@mgtech.com>
Message-ID: <CAKzWOxXWHVLTZcbp0ViFqRkZ-ZeUn8j0STZJN_L5s=AgUE=rpQ@mail.gmail.com>

Well, the good news is that mod_lua comes pre-built, so you can always just
use it. :)
-MC

On Tue, Jul 12, 2011 at 3:40 PM, Mario G <mario_fs at mgtech.com> wrote:

> I was trying to avoid adding more software to OS X and having to learn new
> stuff just for a couple of groups of numbers... I tried many things and the
> best I could do was put them all on on line with a comment below for each
> name. Thanks.
>
> On Jul 12, 2011, at 3:12 PM, Avi Marcus wrote:
>
> A simple array check with lua, or a sql lookup with lua or mod_odbc_query
> from the git contrib would probably make this easy.
> -Avi
>
> On Wed, Jul 13, 2011 at 12:35 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> If you need to test this many numbers then you are better of with
>> something other than regular expressions. I would investigate mod_xml_curl
>> or mod_lcr. An alternative would be to write a small Lua script that checks
>> the destination_number against a list.
>>
>> -MC
>>
>> On Tue, Jul 12, 2011 at 12:31 PM, Mario G <mario_fs at mgtech.com> wrote:
>>
>>> I am trying to find an easy way to test multiple numbers in a condition
>>> without putting all the numbers on one line. There can be 25 numbers or
>>> more. Preferably have comments. I looked at many web sites on XML, etc. but
>>> could not find a way to continue lines. Freeswitch treats everything in the
>>> expression including all spaces as part of the expression. Anyone know of a
>>> good way to do this? Thanks,
>>> Mario G
>>>
>>>   <extension name="out.group1">
>>>     <condition field="${destination_number_w}" expression="^1111111111|
>>>             <!-- joe -->
>>>
>>> 2222222222|3333333333|  <!-- jon -->
>>>
>>> 4444444444|5555555555   <!-- pam -->
>>>
>>> $">
>>>       <action application="set" data="destination_number_w=1$1"/>
>>>       <action application="bridge" data="sofia/gateway/group1/1$1"/>
>>>     </condition>
>>>   </extension>
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
> _______________________________________________
> 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
>
>
>
> _______________________________________________
> 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
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>
>
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From viraptor at gmail.com  Wed Jul 13 03:49:57 2011
From: viraptor at gmail.com (viraptor at gmail.com)
Date: Wed, 13 Jul 2011 00:49:57 +0100
Subject: [Freeswitch-users] Call with failover
In-Reply-To: <CALNZuwnE2X4jmZojZj=25z-NEoi6xELqDQBADjBP5WaXkQPWvw@mail.gmail.com>
Message-ID: <4e1cdda5.cccce30a.3463.ffffe447@mx.google.com>

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From aram at connectto.com  Wed Jul 13 03:49:03 2011
From: aram at connectto.com (Aram Ter-Martirosyan)
Date: Tue, 12 Jul 2011 16:49:03 -0700
Subject: [Freeswitch-users] Does anybody implemented the FreeSWITHC as an
	SBC?
Message-ID: <028101cc40ee$496f0b30$dc4d2190$@com>

Does anybody implemented the FreeSWITHC as an SBC? I already have our SIP
switch and it does almost all the telephony functionalities, including
registration, call handling, and different call features like hunting, call
forwarding, etc.

 I want the SBC to translate the SIP registrations and the INVITEs to our
existing SIP registrar coming in from the remote VoIP users. 
 
 I also want to create redundant call and registration handling by means of
several SBCs, maybe having one Outbound server and several SBCs behind it,
so when one is down the others can handle the same traffic.

 

Is this possible with FreeSWITCH?

sbc's - can be instructed to rewrite headers by rules or scripts. they also
have different security functions and abilities for remote user NAT
traversal, some can also transcode.

 

            Thanks in advance,

 

Aram Ter-Martirosyan

ConnectTo Communications, Inc.

 <http://www.connectto.com/> http://www.ConnectTo.com

555 Riverdale, Suite A

Glendale, CA 91204

aram at ConnectTo.com

tel 818.546.4601

fax 818.546.4617

Turning Technology Into Business Solutions

 

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From mario_fs at mgtech.com  Wed Jul 13 04:21:23 2011
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 12 Jul 2011 17:21:23 -0700
Subject: [Freeswitch-users] An easy way to test a list of numbers in
	condition?
In-Reply-To: <CAKzWOxXWHVLTZcbp0ViFqRkZ-ZeUn8j0STZJN_L5s=AgUE=rpQ@mail.gmail.com>
References: <887F35B0-7832-4492-AB6A-56F782E6DCED@mgtech.com>
	<CAKzWOxV-AYChw5YfcCg=MrsRD-gZQ7BZhw9Y=QNbpWfqCH3h_Q@mail.gmail.com>
	<CALNZuwkVRqFseBeWrS1vrvhuEb0Qz4aCBsrGy=xBgtkvAJVfCw@mail.gmail.com>
	<A3E5055E-C01D-475A-8836-280464D1F323@mgtech.com>
	<CAKzWOxXWHVLTZcbp0ViFqRkZ-ZeUn8j0STZJN_L5s=AgUE=rpQ@mail.gmail.com>
Message-ID: <DC908EE5-89AC-4A00-8529-AEE9E9C825A1@mgtech.com>

I was also thinking of trying javascript since FS supports it and looks like OS X has it. Do you think LUA is better for this?

On Jul 12, 2011, at 4:39 PM, Michael Collins wrote:

> Well, the good news is that mod_lua comes pre-built, so you can always just use it. :)
> -MC
> 
> On Tue, Jul 12, 2011 at 3:40 PM, Mario G <mario_fs at mgtech.com> wrote:
> I was trying to avoid adding more software to OS X and having to learn new stuff just for a couple of groups of numbers... I tried many things and the best I could do was put them all on on line with a comment below for each name. Thanks.
> 
> On Jul 12, 2011, at 3:12 PM, Avi Marcus wrote:
> 
>> A simple array check with lua, or a sql lookup with lua or mod_odbc_query from the git contrib would probably make this easy.
>> -Avi
>> 
>> On Wed, Jul 13, 2011 at 12:35 AM, Michael Collins <msc at freeswitch.org> wrote:
>> If you need to test this many numbers then you are better of with something other than regular expressions. I would investigate mod_xml_curl or mod_lcr. An alternative would be to write a small Lua script that checks the destination_number against a list.
>> 
>> -MC
>> 
>> On Tue, Jul 12, 2011 at 12:31 PM, Mario G <mario_fs at mgtech.com> wrote:
>> I am trying to find an easy way to test multiple numbers in a condition without putting all the numbers on one line. There can be 25 numbers or more. Preferably have comments. I looked at many web sites on XML, etc. but could not find a way to continue lines. Freeswitch treats everything in the expression including all spaces as part of the expression. Anyone know of a good way to do this? Thanks,
>> Mario G
>> 
>>   <extension name="out.group1">
>>     <condition field="${destination_number_w}" expression="^1111111111|             <!-- joe -->
>>                                                             2222222222|3333333333|  <!-- jon -->
>>                                                             4444444444|5555555555   <!-- pam -->
>>                                                                         $">
>>       <action application="set" data="destination_number_w=1$1"/>
>>       <action application="bridge" data="sofia/gateway/group1/1$1"/>
>>     </condition>
>>   </extension>
>> 
>> 
>> _______________________________________________
>> 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
>> 
>> 
>> 
>> _______________________________________________
>> 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
>> 
>> 
>> _______________________________________________
>> 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
> 
> 
> _______________________________________________
> 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
> 
> 
> _______________________________________________
> 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 krice at freeswitch.org  Wed Jul 13 04:24:50 2011
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 12 Jul 2011 19:24:50 -0500
Subject: [Freeswitch-users] Does anybody implemented the FreeSWITHC as
 an SBC?
In-Reply-To: <028101cc40ee$496f0b30$dc4d2190$@com>
Message-ID: <CA425002.18F24%krice@freeswitch.org>

FreeSWITCH is a B2BUA it will not pass thru registration or notifies etc...

What it will do is load balance Calls across several proxies etc...

You are looking for something more along the lines of a SIP Proxy

K

On 7/12/11 6:49 PM, "Aram Ter-Martirosyan" <aram at connectto.com> wrote:

> Does anybody implemented the FreeSWITHC as an SBC? I already have our SIP
> switch and it does almost all the telephony functionalities, including
> registration, call handling, and different call features like hunting, call
> forwarding, etc.
>  
>  I want the SBC to translate the SIP registrations and the INVITEs to our
>  existing SIP registrar coming in from the remote VoIP users.
>  
>  I also want to create redundant call and registration handling by means of
>  several SBCs, maybe having one Outbound server and several SBCs behind it,
>  so when one is down the others can handle the same traffic.
>  
> Is this possible with FreeSWITCH?
> sbc's - can be instructed to rewrite headers by rules or scripts. they also
> have different security functions and abilities for remote user NAT traversal,
> some can also transcode.
> 
>  
> 
>             Thanks in advance,
> 
>  
> Aram Ter-Martirosyan
> ConnectTo Communications, Inc.
> http://www.ConnectTo.com <http://www.connectto.com/>
> 555 Riverdale, Suite A
> Glendale, CA 91204
> aram at ConnectTo.com
> tel 818.546.4601
> fax 818.546.4617
> Turning Technology Into Business Solutions
>  
> 
> 
> _______________________________________________
> 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 philippe at ppmt.org  Wed Jul 13 04:40:21 2011
From: philippe at ppmt.org (Philippe Le Toquin)
Date: Tue, 12 Jul 2011 20:40:21 -0400
Subject: [Freeswitch-users] WIKI and command printout explanation
In-Reply-To: <CALXCt0o5uH7tqV5e6=C4XWRHmAa_=mhVdZre5ExL0Uhvm3UAxQ@mail.gmail.com>
References: <CACUErJdMVha_3rPON6jmiVWRBVLJuoP+cCgUY=v1iOKVNhiRbw@mail.gmail.com>
	<CALXCt0o5uH7tqV5e6=C4XWRHmAa_=mhVdZre5ExL0Uhvm3UAxQ@mail.gmail.com>
Message-ID: <4E1CE975.40008@ppmt.org>

Hello,

thanks all for you very prompt answers. It shows that however much you
look and search some things
will always evade you .

I am now going to go through all the commands to try to understand better

/Philippe

On 11-07-12 03:03 PM, Giovanni Maruzzelli wrote:
> no more than 1000 in parallel.
>
> Before you ask: min idle cpu gives you at what idle cpu will stop
> accepting calls (in your case at 0 idle, so 100percent busy) and what
> was the average idle in last 5 minutes (in your case 100 percent idle,
> 0 percent busy)
>
> 0 sessions are ongoing, with a max of 30 new sessions /second allowed
> to spawn (eg: no more than 30 new session second ramp up)
>
> all this is configurable
>
> -giovanni
>
>
>
> On Tue, Jul 12, 2011 at 8:37 PM, Philippe Le Toquin <philippe at ppmt.org
> <mailto:philippe at ppmt.org>> wrote:
>
>     Hello,
>
>     I am trying to get more familiar wit FS cli and found the page
>     that list the commands and their usage.
>
>     What I am looking for is an explanation of their printout. Some of
>     them are easy to understand but other not so (not to me at least)
>
>     Is there a wiki page that explains
>
>     For example the simple status command give me that:
>
>     status
>     UP 0 years, 29 days, 17 hours, 38 minutes, 3 seconds, 716
>     milliseconds, 375 microseconds
>     512 session(s) since startup
>     0 session(s) 0/30
>     1000 session(s) max
>     min idle cpu 0.00/100.00
>
>
>     I had 512 session since startup.....fair enough but then it says
>     "1000 session(s) max"
>
>     What does it means? that FS will stop working after 10000 sessions
>     or that I can't get more than 1000 in parallel?
>
>     Regards
>
>     /Philippe
>
>     _______________________________________________
>     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
>     <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
>
>
>
>
> -- 
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
> _______________________________________________
> 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
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> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Wed Jul 13 07:44:10 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Jul 2011 22:44:10 -0500
Subject: [Freeswitch-users] Configuration Server
In-Reply-To: <CAAYf2Luggy0mghz2EFbZFpmOc1dW4M9-iSpjCaS3zpSb1Nj7Og@mail.gmail.com>
References: <CAAYf2Lvv1f8QBmDL9scJm45KcDA7CoVbCmGOcSwBvAH8bnjMVA@mail.gmail.com>
	<CAFYZzbvVzfRt60neJC+DyDQMpsH+iWt7Ubv6YWALjN1fdf4fbw@mail.gmail.com>
	<CAAYf2Luggy0mghz2EFbZFpmOc1dW4M9-iSpjCaS3zpSb1Nj7Og@mail.gmail.com>
Message-ID: <CAKbxfG9DzHz0HrBvKRpJvbezdaV-kAAKJwNH1iEG7D2+XHDVYg@mail.gmail.com>

You should attend the wednesday community call.  Most of the windows
people do not watch the lists but they tend to be in irc or on the
phones WED 12:00 central.

You can also get the code into contrib and work with the other windows
developers if you wish.


On Tue, Jul 12, 2011 at 8:22 AM, Roger Castaldo
<roger.castaldo at gmail.com> wrote:
> Thank you Michel for the offer of help but there seems to be a lack of
> interest so I suppose I shall continue puttering away at it and see if some
> developers get interested later on when I have a more viable stable release
> that could peak more interest in it being open source.? I will, however,
> possibly look at releasing my socket code as an open source library when
> done which will allow C# apps to connect to freeswitch servers via the event
> socket and control/monitor the servers through those connections.
>
> On Wed, Jul 6, 2011 at 1:37 PM, Michel Daggelinckx
> <michel.daggelinckx at gmail.com> wrote:
>>
>> hi,
>> I can do testing
>>
>> On Wed, Jul 6, 2011 at 4:09 PM, Roger Castaldo <roger.castaldo at gmail.com>
>> wrote:
>>>
>>> Hi everyone, I have been following freeswitch for quite some time now and
>>> working on an ever evolving configuration server for it, which has now taken
>>> yet another branch, that instead of writing xml configuration files now is
>>> attempting to control the call flow itself using the outbound socket.
>>> Currently it is a closed source project, originally because of some of my
>>> closed source libraries I was using to develop it, but since they have
>>> become open source I was wondering if there is any interest in working with
>>> me on the development/testing of the product.? It is written in C# and is
>>> entirely web based, using ajax calls and web services for everything.? I am
>>> using a Relational Mapping library of my own design to map the data to the
>>> database (currently a firebirdsql database).
>>>
>>> _______________________________________________
>>> 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
>>>
>>
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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
>
>



-- 
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 abubacker at bksystems.co.in  Wed Jul 13 10:37:32 2011
From: abubacker at bksystems.co.in (abubacker)
Date: Wed, 13 Jul 2011 12:07:32 +0530
Subject: [Freeswitch-users] How to set the member-flags in mod_conference
Message-ID: <4E1D3D2C.6070109@bksys.co.in>

Dear all,
     My requirement is to organise a conference where I want some of the 
users to be mute ,
and some of the members to be deaf and some of the member to be endconf

I was trying to set the variable "member-flags" as mute or deaf or 
endconf but its not working
whether I am doing the right thing !

Dialplan :

<extension name="Conference">
<condition field="destination_number" expression="^3001$">
<action application="answer" />
<action application="set" data="member-flags=mute"/>
<action application="conference" data="room1"/>
</condition>
</extension>

<extension name="Conference">
<condition field="destination_number" expression="^3002$">
<action application="answer" />
<action application="set" data="member-flags=endconf"/>
<action application="conference" data="room1"/>
</condition>
</extension>

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

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



From daniel at danielknoll.de  Wed Jul 13 13:01:42 2011
From: daniel at danielknoll.de (Daniel Knoll)
Date: Wed, 13 Jul 2011 11:01:42 +0200
Subject: [Freeswitch-users] session.recordFile create empy wav in while loop
Message-ID: <CACREEeaJ8On1Oo2K64PMisbJE+jASFC9tLTxXNZg71bS+3vVig@mail.gmail.com>

Hey Guys,

i have a strange Problem with session.recordFile in a while loop and i
don't know how to solve it.
The first run in the loop is absolutly ok, session.recordFile is
creating a good clean wavefile.
but in the second, third ..  run, every wav file has a size of 47724
byte and no audio content.

Please help me, because it is an urgent case.

Thankx for getting help.
Daniel


From steveayre at gmail.com  Wed Jul 13 13:07:21 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 13 Jul 2011 10:07:21 +0100
Subject: [Freeswitch-users] session.recordFile create empy wav in while
	loop
In-Reply-To: <CACREEeaJ8On1Oo2K64PMisbJE+jASFC9tLTxXNZg71bS+3vVig@mail.gmail.com>
References: <CACREEeaJ8On1Oo2K64PMisbJE+jASFC9tLTxXNZg71bS+3vVig@mail.gmail.com>
Message-ID: <CAFiqYumaqz-tP9=bjSemCgjaxThfki9DNDFKkDEH7seL7ADHxQ@mail.gmail.com>

Are you able to share the code?

Are you checking session.ready in the while loop's condition? Might the
channel be hung up when it reaches the 2nd recordFile?

-Steve


On 13 July 2011 10:01, Daniel Knoll <daniel at danielknoll.de> wrote:

> Hey Guys,
>
> i have a strange Problem with session.recordFile in a while loop and i
> don't know how to solve it.
> The first run in the loop is absolutly ok, session.recordFile is
> creating a good clean wavefile.
> but in the second, third ..  run, every wav file has a size of 47724
> byte and no audio content.
>
> Please help me, because it is an urgent case.
>
> Thankx for getting help.
> Daniel
>
> _______________________________________________
> 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 ankitwalia4u at gmail.com  Wed Jul 13 13:14:48 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Wed, 13 Jul 2011 14:44:48 +0530
Subject: [Freeswitch-users] V92 modem with freeswitch
Message-ID: <CAOcuKucME9wAXkUfVPRWXTzqTs0WhtYyEr=n1KNLUbw0L9PCxw@mail.gmail.com>

Hi all,

I found a V.92 56k PCI Modem in my old hardware.

I want to run some basic functionality test with my PSTN phone line and FS.

Can I connect my phone line with FS using this card. If yes how do we
configure to use this card.

I am new to telephony. I don't know if whatever I said above make sense.

Thanks

Ankit
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From boris at tagnet.ru  Wed Jul 13 13:20:54 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Wed, 13 Jul 2011 15:20:54 +0600
Subject: [Freeswitch-users] V92 modem with freeswitch
In-Reply-To: <CAOcuKucME9wAXkUfVPRWXTzqTs0WhtYyEr=n1KNLUbw0L9PCxw@mail.gmail.com>
References: <CAOcuKucME9wAXkUfVPRWXTzqTs0WhtYyEr=n1KNLUbw0L9PCxw@mail.gmail.com>
Message-ID: <4E1D6376.1060401@tagnet.ru>

Hello!

     No You can't. Modem has no ability to act as voice device.
> Hi all,
>
> I found a V.92 56k PCI Modem in my old hardware.
> I want to run some basic functionality test with my PSTN phone line and FS.
> Can I connect my phone line with FS using this card. If yes how do we configure to use this card.
> I am new to telephony. I don't know if whatever I said above make sense.
> Thanks
> Ankit
>
>
> _______________________________________________
> 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


-- 
Regards,
Boris


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From daniel at danielknoll.de  Wed Jul 13 13:24:24 2011
From: daniel at danielknoll.de (Daniel Knoll)
Date: Wed, 13 Jul 2011 11:24:24 +0200
Subject: [Freeswitch-users] session.recordFile create empy wav in while
	loop
In-Reply-To: <CAFiqYumaqz-tP9=bjSemCgjaxThfki9DNDFKkDEH7seL7ADHxQ@mail.gmail.com>
References: <CACREEeaJ8On1Oo2K64PMisbJE+jASFC9tLTxXNZg71bS+3vVig@mail.gmail.com>
	<CAFiqYumaqz-tP9=bjSemCgjaxThfki9DNDFKkDEH7seL7ADHxQ@mail.gmail.com>
Message-ID: <CACREEeZFOKXHDTu-5YEsP1UUN64Tj+tLYP=hj95F67=gjZV3vA@mail.gmail.com>

Hi Steve, thanks for fast answer. The Channel is't hung up, because I
stream in the same loop an Audio File

here is the code, the session.recordFile is executed in the second
run, because, the filesize is changing for the file at first run.

do {
	var rd = session.uuid;
	session.streamFile("/dev/shm/freeswitch/sounds/conf-asrintro.wav", "");
	session.recordFile("/dev/shm/"+rd+".wav", "", "", 10, 500, 3);
       	system("sox -v 1.7 /dev/shm/"+rd+".wav -r 16000 /dev/shm/"+rd+".flac");
	var fd = new File("|/usr/bin/googlespeech.sh /dev/shm/"+rd+".flac");
	fd.open("read");
	var dtmf = fd.read("8");
	dtmf = dtmf.replace(/[^0-9]/g, '');
	console_log("notice", "DTMF: "+ dtmf +"\n");
	roomExists = checkRoom(dtmf);
	if(!roomExists){
		retry_count++;
		session.streamFile("/dev/shm/freeswitch/sounds/conf-invalid.wav", "");
		var dtmf = "";
	}
} while(!roomExists && retry_count < 5);


I googled at this and found some guys he has also the same problem
http://asterisk.voicemeup.com/viewtopic.php?p=62294&sid=8360229099e219df50bbb016cd43cbfd

Can you help me ?

Thanx Daniel





2011/7/13, Steven Ayre <steveayre at gmail.com>:
> Are you able to share the code?
>
> Are you checking session.ready in the while loop's condition? Might the
> channel be hung up when it reaches the 2nd recordFile?
>
> -Steve
>
>
> On 13 July 2011 10:01, Daniel Knoll <daniel at danielknoll.de> wrote:
>
>> Hey Guys,
>>
>> i have a strange Problem with session.recordFile in a while loop and i
>> don't know how to solve it.
>> The first run in the loop is absolutly ok, session.recordFile is
>> creating a good clean wavefile.
>> but in the second, third ..  run, every wav file has a size of 47724
>> byte and no audio content.
>>
>> Please help me, because it is an urgent case.
>>
>> Thankx for getting help.
>> Daniel
>>
>> _______________________________________________
>> 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
>>
>


-- 
Daniel Knoll
Liberdastr.. 9
12047 Berlin

fon +49 (0)179 20 16 50 8
mail daniel at danielknoll.de
web www.danielknoll.de


From fieldpeak at gmail.com  Wed Jul 13 14:06:30 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Wed, 13 Jul 2011 18:06:30 +0800
Subject: [Freeswitch-users] How to configure for FS playing voice prompt
 in case the called extension is in a call.
In-Reply-To: <5F28A4A5-726E-44EC-97CA-E00920F95114@ipeva.fr>
References: <CAOTmbMzMVtRjUvTj5UgHH7MG=XyHSWRvw7QT6G-tiLFd57H1Vw@mail.gmail.com>
	<CAFiqYuncFULQn=BnWDh2s0m6VhAX3aqN-zPbFaMD19cowfOERA@mail.gmail.com>
	<CAOTmbMwtVZJd_44rs7QH1GATSPygyfMJq2Ao-kgxw1x-L_vkVg@mail.gmail.com>
	<5F28A4A5-726E-44EC-97CA-E00920F95114@ipeva.fr>
Message-ID: <CAOTmbMxJoQTkZ+UnCxtU_VHRHPUU0+sn2dFSa2Q2AMZgAJKXBw@mail.gmail.com>

My phone sends back '486 Busy Here'.

2011/7/12 David Ponzone <david.ponzone at ipeva.fr>

> Check what your phone sends back when busy.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 12/07/2011 ? 10:31, fieldpeak a ?crit :
>
> Hi Steve,
>
> Thanks for your kindly help, it works with below dial plan...
> However, for both user busy or no answer, it played the voice prompt...
> i would like it only prompts in case user busy but not no answer, although
> i configured the "failure_causes=USER_BUSY" as below dialplan, it looks not
> help...attached is a log...
>
>     <extension name="Local_Extension">
>       <condition field="destination_number" expression="^(1\d{3})$">
>     <action application="set" data="dialed_extension=$1"/>
>     <action application="export" data="dialed_extension=$1"/>
>     <action application="set" data="ringback=${us-ring}"/>
>     <action application="set" data="transfer_ringback=$${hold_music}"/>
>     <action application="set" data="call_timeout=30"/>
>
>     <action application="set" data="failure_causes=USER_BUSY"/>
>
>     <action application="set" data="hangup_after_bridge=true"/>
>
>     <!-- to enable voice mail, set continue_on_fail=true below -->
>     <action application="set" data="continue_on_fail=true"/>
>     <action application="hash"
> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>     <action application="hash"
> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>     <action application="bridge"
> data="sofia/internal/${dialed_extension}%${domain_name}"/>
>     <action application="answer"/>
>
>     <action application="sleep" data="1000"/>
>     <!-- play a voice prompt e.g. 'the extension you dialed is busy now,
> please dial the other exsention' to replacing busy tone when the called
> extension is in a call... -->
>     <action application="play_and_get_digits" data="2 5 3 4000 #
> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav
> other_ext_var \d+"/>
>     <action application="transfer" data="${other_ext_var} XML default"/>
>
>       </condition>
>     </extension>
>
> Regards,
> Charles
>
>
> 2011/7/11 Steven Ayre <steveayre at gmail.com>
>
>> There are two ways....
>>
>> Most phones will return a USER_BUSY release code if you bridge to a phone
>> that's busy. If the phone does then you can continue_on_fail so that you
>> (pre-)answer the call and play that message after the bridge. Something
>> like:
>>
>> <extension name="Local_Extension">
>>     <condition field="destination_number" expression="^(1\d{3})$">
>>         <action application="set" data="dialed_extension=$1"/>
>>         <action application="export" data="dialed_extension=$1"/>
>>         <action application="set" data="call_timeout=30"/>
>>         <action application="set" data="hangup_after_bridge=true"/> <!--
>> Hangup on successful call -->
>>         <action application="set" data="continue_on_fail=true"/> <!--
>> Don't hangup on failed call, continue to IVR prompt -->
>>
>>         <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>         <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>         <action application="bridge"
>> data="user/${dialed_extension}%${domain_name}"/>
>>         <!-- Put the IVR prompt here, see
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits-->
>>     </condition>
>> </extension>
>>
>> The 2nd option is that you use Limit to determine if another person is
>> already on a call to the phone. That only really works if you run the the
>> only servers that call the phone since it requires tracking all calls to the
>> phone. It will sometimes work better though - especially if the phone has
>> multiple lines. http://wiki.freeswitch.org/wiki/Limit
>>
>> -Steve
>>
>>
>> On 11 July 2011 03:25, fieldpeak <fieldpeak at gmail.com> wrote:
>>
>>> Hi Gurus,
>>>
>>> Could anyone advise how to realize have FS to play a voice prompt e.g.
>>> 'the extension you dialed is busy now, please dial the other exsention' to
>>> replacing busy tone when the called extension is in a call...
>>>
>>> the details is below,
>>>
>>> When calling to FS, the FS will play IVR "welcome to call us, please
>>> input the extension number, for operator please press 0",
>>> then the caller input the extension number, in case the extension is in a
>>> call, the caller will hear busy tone...
>>> it needs the system play "the extension you dailed is busy, please dial
>>> the other extension", currently my dial plan as following, could anyone
>>> advise how to change based on below dial plan or any other new dail plan can
>>> realize it...
>>>
>>> Thanks a lot!
>>>
>>>     <extension name="Local_Extension">
>>>       <condition field="destination_number" expression="^(1\d{3})$">
>>>     <action application="set" data="dialed_extension=$1"/>
>>>     <action application="export" data="dialed_extension=$1"/>
>>>     <action application="set" data="ringback=${us-ring}"/>
>>>     <action application="set" data="transfer_ringback=$${hold_music}"/>
>>>     <action application="set" data="call_timeout=30"/>
>>>     <action application="set" data="hangup_after_bridge=true"/>
>>>
>>>     <!-- to enable voice mail, set continue_on_fail=true below -->
>>>     <action application="set" data="continue_on_fail=false"/>
>>>     <action application="hash"
>>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>>     <action application="hash"
>>> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>>     <action application="bridge"
>>> data="sofia/internal/${dialed_extension}%${domain_name}"/>
>>>     <action application="answer"/>
>>>
>>>     <action application="sleep" data="1000"/>
>>>     <action application="bridge" data="loopback/app=voicemail:default
>>> ${domain_name} ${dialed_extension}"/>
>>>       </condition>
>>>     </extension>
>>>
>>>
>>>     <extension name="IVR" >
>>>        <condition field="destination_number" expression="^888$">
>>>            <action application="answer"/>
>>>            <action application="sleep" data="2000"/>
>>>            <action application="ivr" data="IVR"/>
>>>        </condition>
>>>     </extension>
>>>
>>>         <!--  -->
>>>     <menu name="IVR"
>>>         greet-long="C:/VSWITCH/recordings/greeting_tts.wav"
>>>         greet-short="C:/VSWITCH/recordings/greeting_tts.wav"
>>>         invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>>         exit-sound="voicemail/vm-goodbye.wav"
>>>         confirm-macro=""
>>>         confirm-key=""
>>>         tts-engine="flite"
>>>         tts-voice="rms"
>>>         confirm-attempts="3"
>>>         timeout="10000"
>>>         inter-digit-timeout="2000"
>>>         max-failures="3"
>>>         max-timeouts="3"
>>>         digit-len="5">
>>>         <entry action="menu-exec-app" digits="/(^\*\d{3,5}$|^\d{3,5}$)/"
>>> param="transfer $1 XML default"/>
>>>     </menu>
>>>
>>> Regards,
>>> Charles
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
> <log_no_answer.txt>_______________________________________________
>
> 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
>
>
>
> _______________________________________________
> 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 steveayre at gmail.com  Wed Jul 13 14:16:36 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 13 Jul 2011 11:16:36 +0100
Subject: [Freeswitch-users] V92 modem with freeswitch
In-Reply-To: <4E1D6376.1060401@tagnet.ru>
References: <CAOcuKucME9wAXkUfVPRWXTzqTs0WhtYyEr=n1KNLUbw0L9PCxw@mail.gmail.com>
	<4E1D6376.1060401@tagnet.ru>
Message-ID: <CAFiqYumyS6hYPpxEdCUyH-8eMtYjZqorg-4gngQVNn2w71frng@mail.gmail.com>

That's not quite true... voice modems do let you do this, but not all
modems are voice modems.

-Steve



On 13 July 2011 10:20, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ??? No You can't. Modem has no ability to act as voice device.
>
> Hi all,
> I found a V.92 56k PCI Modem in my old hardware.
>
> I want to run some basic functionality test with my PSTN phone line and FS.
>
> Can I connect my phone line with FS using this card. If yes how do we
> configure to use this card.
>
> I am new to telephony. I don't know if whatever I said above make sense.
>
> Thanks
>
> Ankit
>
> _______________________________________________
> 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
>
>
> --
> Regards,
> Boris
>
>
> _______________________________________________
> 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 hmkias at gmail.com  Wed Jul 13 14:22:00 2011
From: hmkias at gmail.com (hmkias at gmail.com)
Date: Wed, 13 Jul 2011 10:22:00 +0000
Subject: [Freeswitch-users] V92 modem with freeswitch
In-Reply-To: <CAFiqYumyS6hYPpxEdCUyH-8eMtYjZqorg-4gngQVNn2w71frng@mail.gmail.com>
References: <CAOcuKucME9wAXkUfVPRWXTzqTs0WhtYyEr=n1KNLUbw0L9PCxw@mail.gmail.com><4E1D6376.1060401@tagnet.ru><CAFiqYumyS6hYPpxEdCUyH-8eMtYjZqorg-4gngQVNn2w71frng@mail.gmail.com>
Message-ID: <2027362404-1310552396-cardhu_decombobulator_blackberry.rim.net-564069076-@b13.c4.bise7.blackberry>

I have used a pci modem as fxo with asterisk. 
Sent from BSNL with my BlackBerry? smartphone

-----Original Message-----
From: Steven Ayre <steveayre at gmail.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Wed, 13 Jul 2011 11:16:36 
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] V92 modem with freeswitch

That's not quite true... voice modems do let you do this, but not all
modems are voice modems.

-Steve



On 13 July 2011 10:20, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ??? No You can't. Modem has no ability to act as voice device.
>
> Hi all,
> I found a V.92 56k PCI Modem in my old hardware.
>
> I want to run some basic functionality test with my PSTN phone line and FS.
>
> Can I connect my phone line with FS using this card. If yes how do we
> configure to use this card.
>
> I am new to telephony. I don't know if whatever I said above make sense.
>
> Thanks
>
> Ankit
>
>_______________________________________________
> 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
>
>
> --
> Regards,
> Boris
>
>
>_______________________________________________
> 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
>
>

_______________________________________________
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 nandy1925 at gmail.com  Wed Jul 13 15:02:11 2011
From: nandy1925 at gmail.com (nandy1925 at gmail.com)
Date: Wed, 13 Jul 2011 11:02:11 +0000
Subject: [Freeswitch-users] V92 modem with freeswitch
In-Reply-To: <2027362404-1310552396-cardhu_decombobulator_blackberry.rim.net-564069076-@b13.c4.bise7.blackberry>
Message-ID: <bcaec51dd57924499704a7f15991@google.com>

hi ankit,

the pci modem that works w/ asterisk as FXO has a distinct chipset not  
found in all cards. the FXO card as an interrupt timer (req'd by asterisk)  
but not in FS. IMHO, you better off use an FXO gateway. it performs better  
and u hv many options available like T.38.
-nandy
On Jul 13, 2011 6:22pm, hmkias at gmail.com wrote:
> I have used a pci modem as fxo with asterisk.

> Sent from BSNL with my BlackBerry? smartphone



> -----Original Message-----

> From: Steven Ayre steveayre at gmail.com>

> Sender: freeswitch-users-bounces at lists.freeswitch.org

> Date: Wed, 13 Jul 2011 11:16:36

> To: FreeSWITCH Users Helpfreeswitch-users at lists.freeswitch.org>

> Reply-To: FreeSWITCH Users Help freeswitch-users at lists.freeswitch.org>

> Subject: Re: [Freeswitch-users] V92 modem with freeswitch



> That's not quite true... voice modems do let you do this, but not all

> modems are voice modems.



> -Steve







> On 13 July 2011 10:20, Boris Kovalenko boris at tagnet.ru> wrote:

> > Hello!

> >

> > No You can't. Modem has no ability to act as voice device.

> >

> > Hi all,

> > I found a V.92 56k PCI Modem in my old hardware.

> >

> > I want to run some basic functionality test with my PSTN phone line and  
> FS.

> >

> > Can I connect my phone line with FS using this card. If yes how do we

> > configure to use this card.

> >

> > I am new to telephony. I don't know if whatever I said above make sense.

> >

> > Thanks

> >

> > Ankit

> >

> >_______________________________________________

> > 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

> >

> >

> > --

> > Regards,

> > Boris

> >

> >

> >_______________________________________________

> > 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

> >

> >



> _______________________________________________

> 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

> _______________________________________________

> 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 roger.castaldo at gmail.com  Wed Jul 13 15:41:39 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Wed, 13 Jul 2011 07:41:39 -0400
Subject: [Freeswitch-users] Configuration Server
In-Reply-To: <CAKbxfG9DzHz0HrBvKRpJvbezdaV-kAAKJwNH1iEG7D2+XHDVYg@mail.gmail.com>
References: <CAAYf2Lvv1f8QBmDL9scJm45KcDA7CoVbCmGOcSwBvAH8bnjMVA@mail.gmail.com>
	<CAFYZzbvVzfRt60neJC+DyDQMpsH+iWt7Ubv6YWALjN1fdf4fbw@mail.gmail.com>
	<CAAYf2Luggy0mghz2EFbZFpmOc1dW4M9-iSpjCaS3zpSb1Nj7Og@mail.gmail.com>
	<CAKbxfG9DzHz0HrBvKRpJvbezdaV-kAAKJwNH1iEG7D2+XHDVYg@mail.gmail.com>
Message-ID: <CAAYf2LsFNwRCw7LJg5gsYDG4C5KLAbqmzdP5wD+8qyqDeNTH0A@mail.gmail.com>

Sadly I do not think I will be able to make the community call today.  That
being said, this system right now has issues running on a windows machine
because it uses a lot of security and backend features of linux, it is
actually being designed and built to run on a stripped down version of
mono.  Granted it should run on windows now, I am just having difficulties
finding a firewall equivalent to iptables in windows.

On Tue, Jul 12, 2011 at 11:44 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You should attend the wednesday community call.  Most of the windows
> people do not watch the lists but they tend to be in irc or on the
> phones WED 12:00 central.
>
> You can also get the code into contrib and work with the other windows
> developers if you wish.
>
>
> On Tue, Jul 12, 2011 at 8:22 AM, Roger Castaldo
> <roger.castaldo at gmail.com> wrote:
> > Thank you Michel for the offer of help but there seems to be a lack of
> > interest so I suppose I shall continue puttering away at it and see if
> some
> > developers get interested later on when I have a more viable stable
> release
> > that could peak more interest in it being open source.  I will, however,
> > possibly look at releasing my socket code as an open source library when
> > done which will allow C# apps to connect to freeswitch servers via the
> event
> > socket and control/monitor the servers through those connections.
> >
> > On Wed, Jul 6, 2011 at 1:37 PM, Michel Daggelinckx
> > <michel.daggelinckx at gmail.com> wrote:
> >>
> >> hi,
> >> I can do testing
> >>
> >> On Wed, Jul 6, 2011 at 4:09 PM, Roger Castaldo <
> roger.castaldo at gmail.com>
> >> wrote:
> >>>
> >>> Hi everyone, I have been following freeswitch for quite some time now
> and
> >>> working on an ever evolving configuration server for it, which has now
> taken
> >>> yet another branch, that instead of writing xml configuration files now
> is
> >>> attempting to control the call flow itself using the outbound socket.
> >>> Currently it is a closed source project, originally because of some of
> my
> >>> closed source libraries I was using to develop it, but since they have
> >>> become open source I was wondering if there is any interest in working
> with
> >>> me on the development/testing of the product.  It is written in C# and
> is
> >>> entirely web based, using ajax calls and web services for everything.
> I am
> >>> using a Relational Mapping library of my own design to map the data to
> the
> >>> database (currently a firebirdsql database).
> >>>
> >>> _______________________________________________
> >>> 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
> >>>
> >>
> >>
> >> _______________________________________________
> >> 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
> >>
> >
> >
> > _______________________________________________
> > 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
> >
> >
>
>
>
> --
> 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
>
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From abubacker at bksystems.co.in  Wed Jul 13 16:48:43 2011
From: abubacker at bksystems.co.in (abubacker)
Date: Wed, 13 Jul 2011 18:18:43 +0530
Subject: [Freeswitch-users] How to set the member-flags in mod_conference
In-Reply-To: <4E1D3D2C.6070109@bksys.co.in>
References: <4E1D3D2C.6070109@bksys.co.in>
Message-ID: <4E1D942B.9070809@bksys.co.in>

On Wednesday 13 July 2011 12:07 PM, abubacker wrote:
> Dear all,
>       My requirement is to organise a conference where I want some of the
> users to be mute ,
> and some of the members to be deaf and some of the member to be endconf
>
> I was trying to set the variable "member-flags" as mute or deaf or
> endconf but its not working
> whether I am doing the right thing !
>
> Dialplan :
>
> <extension name="Conference">
> <condition field="destination_number" expression="^3001$">
> <action application="answer" />
> <action application="set" data="member-flags=mute"/>
> <action application="conference" data="room1"/>
> </condition>
> </extension>
>
> <extension name="Conference">
> <condition field="destination_number" expression="^3002$">
> <action application="answer" />
> <action application="set" data="member-flags=endconf"/>
> <action application="conference" data="room1"/>
> </condition>
> </extension>
>
>    

That can be done using flags
ex :
<action application="conference" data="room1+flags{endconf}"/>

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

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



From anthony.minessale at gmail.com  Wed Jul 13 19:24:42 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Jul 2011 10:24:42 -0500
Subject: [Freeswitch-users] Configuration Server
In-Reply-To: <CAAYf2LsFNwRCw7LJg5gsYDG4C5KLAbqmzdP5wD+8qyqDeNTH0A@mail.gmail.com>
References: <CAAYf2Lvv1f8QBmDL9scJm45KcDA7CoVbCmGOcSwBvAH8bnjMVA@mail.gmail.com>
	<CAFYZzbvVzfRt60neJC+DyDQMpsH+iWt7Ubv6YWALjN1fdf4fbw@mail.gmail.com>
	<CAAYf2Luggy0mghz2EFbZFpmOc1dW4M9-iSpjCaS3zpSb1Nj7Og@mail.gmail.com>
	<CAKbxfG9DzHz0HrBvKRpJvbezdaV-kAAKJwNH1iEG7D2+XHDVYg@mail.gmail.com>
	<CAAYf2LsFNwRCw7LJg5gsYDG4C5KLAbqmzdP5wD+8qyqDeNTH0A@mail.gmail.com>
Message-ID: <CAKbxfG-ReDkv3RomyVyMQ2ZP2QmVngiGbzJ5LVffo9FcKz9JPg@mail.gmail.com>

So that's interesting.  I sometimes forget we have mono support since
most of the .NET people lean towards windows.  Perhaps you should
consider attending one of the upcoming calls and do a presentation on
using FS + mono and see if you can encourage anyone to try it.


On Wed, Jul 13, 2011 at 6:41 AM, Roger Castaldo
<roger.castaldo at gmail.com> wrote:
> Sadly I do not think I will be able to make the community call today.? That
> being said, this system right now has issues running on a windows machine
> because it uses a lot of security and backend features of linux, it is
> actually being designed and built to run on a stripped down version of
> mono.? Granted it should run on windows now, I am just having difficulties
> finding a firewall equivalent to iptables in windows.
>
> On Tue, Jul 12, 2011 at 11:44 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> You should attend the wednesday community call. ?Most of the windows
>> people do not watch the lists but they tend to be in irc or on the
>> phones WED 12:00 central.
>>
>> You can also get the code into contrib and work with the other windows
>> developers if you wish.
>>
>>
>> On Tue, Jul 12, 2011 at 8:22 AM, Roger Castaldo
>> <roger.castaldo at gmail.com> wrote:
>> > Thank you Michel for the offer of help but there seems to be a lack of
>> > interest so I suppose I shall continue puttering away at it and see if
>> > some
>> > developers get interested later on when I have a more viable stable
>> > release
>> > that could peak more interest in it being open source.? I will, however,
>> > possibly look at releasing my socket code as an open source library when
>> > done which will allow C# apps to connect to freeswitch servers via the
>> > event
>> > socket and control/monitor the servers through those connections.
>> >
>> > On Wed, Jul 6, 2011 at 1:37 PM, Michel Daggelinckx
>> > <michel.daggelinckx at gmail.com> wrote:
>> >>
>> >> hi,
>> >> I can do testing
>> >>
>> >> On Wed, Jul 6, 2011 at 4:09 PM, Roger Castaldo
>> >> <roger.castaldo at gmail.com>
>> >> wrote:
>> >>>
>> >>> Hi everyone, I have been following freeswitch for quite some time now
>> >>> and
>> >>> working on an ever evolving configuration server for it, which has now
>> >>> taken
>> >>> yet another branch, that instead of writing xml configuration files
>> >>> now is
>> >>> attempting to control the call flow itself using the outbound socket.
>> >>> Currently it is a closed source project, originally because of some of
>> >>> my
>> >>> closed source libraries I was using to develop it, but since they have
>> >>> become open source I was wondering if there is any interest in working
>> >>> with
>> >>> me on the development/testing of the product.? It is written in C# and
>> >>> is
>> >>> entirely web based, using ajax calls and web services for everything.
>> >>> I am
>> >>> using a Relational Mapping library of my own design to map the data to
>> >>> the
>> >>> database (currently a firebirdsql database).
>> >>>
>> >>> _______________________________________________
>> >>> 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
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> 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
>> >>
>> >
>> >
>> > _______________________________________________
>> > 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
>> >
>> >
>>
>>
>>
>> --
>> 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
>
>
> _______________________________________________
> 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
>
>



-- 
Anthony Minessale II

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

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

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


From tculjaga at gmail.com  Wed Jul 13 19:27:15 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 13 Jul 2011 17:27:15 +0200
Subject: [Freeswitch-users] memory leak
Message-ID: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>

hello,

i upgraded to the latest GIT compiled configured FS without any issues...
everything looks great... but when i put this into some traffic i noticed FS
uses more and more memory... until it finished into swap. I didn't have
issues like that previously but unfortunately i cannot recall what was the
version with no issues.

The server is 64bit CentOS 5.4 and the FS version is the latest (well 4 days
old tops). Im using just the standard modules (the list provided below) and
FS keeps leaking memory

    <load module="mod_console"/>
    <load module="mod_logfile"/>
    <load module="mod_enum"/>
    <load module="mod_cdr_csv"/>
    <load module="mod_event_socket"/>
    <load module="mod_sofia"/>
    <load module="mod_loopback"/>
    <load module="mod_commands"/>
    <load module="mod_conference"/>
    <load module="mod_db"/>
    <load module="mod_dptools"/>
    <load module="mod_expr"/>
    <load module="mod_fifo"/>
    <load module="mod_hash"/>
    <load module="mod_voicemail"/>
    <load module="mod_esf"/>
    <load module="mod_fsv"/>
    <load module="mod_cluechoo"/>
    <load module="mod_valet_parking"/>
    <load module="mod_dialplan_xml"/>
    <load module="mod_dialplan_asterisk"/>
    <load module="mod_spandsp"/>
    <load module="mod_g723_1"/>
    <load module="mod_g729"/>
    <load module="mod_amr"/>
    <load module="mod_ilbc"/>
    <load module="mod_speex"/>
    <load module="mod_h26x"/>
    <load module="mod_siren"/>
    <load module="mod_sndfile"/>
    <load module="mod_native_file"/>
    <load module="mod_local_stream"/>
    <load module="mod_tone_stream"/>
    <load module="mod_spidermonkey"/>
    <load module="mod_lua"/>
    <load module="mod_say_en"/>


This is the dialplan im triggering...

   <extension name="doRedirect">
      <condition field="destination_number" expression="^300030.*$">
         <action application="info"/>
         <action application="redirect"
data="<sip:0999901234567889 at 10.1.1.11:5060;user=phone>;q=0.99"/>
         <action application="hangup"/> <!-- well ... not really needed but
only Got knows for sure-->
      </condition>
   </extension>

I have i have tried a pure "hello world" dialplan as:

   <extension name="testDP">
      <condition field="destination_number" expression="^300030.*$">
         <action application="info"/>
         <action application="hangup"/>
      </condition>
   </extension>

But the situation is exactly the same.



This is the behavior i'm having .... memory just keep growing and growing
until it hits the swap where the server is doomed.

 tail -f smaps_log.log (this is from cat /proc/$PID/smaps | grep heap)
TIM,SIZE,RSS,SHARED_CLEAN,SHARED_DIRTY,PRIVATE_CLEAN,PRIVATE_DIRTY,SWAP
2011-07-13 13:50:34,19268,18916,0,0,0,18916,0
2011-07-13 13:51:34,19268,18972,0,0,0,18972,0
2011-07-13 13:52:34,20324,20088,0,0,0,20088,0
2011-07-13 13:53:34,21264,20920,0,0,0,20920,0
2011-07-13 13:54:34,21048,20856,0,0,0,20856,0
2011-07-13 13:55:35,22396,22076,0,0,0,22076,0
2011-07-13 13:56:35,22396,22164,0,0,0,22256,0
2011-07-13 13:57:35,23220,22936,0,0,0,22936,0
2011-07-13 13:58:35,23480,23288,0,0,0,23288,0

$ tail -f mem.log (this is from ps)
TIMESTAMP            PID  RSS    %MEM
2011-07-13 13:48:55  2405 65144 12.8
2011-07-13 13:49:55  2405 66932 13.2
2011-07-13 13:50:55  2405 69576 13.7
2011-07-13 13:51:55  2405 71552 14.1
2011-07-13 13:52:55  2405 73576 14.5
2011-07-13 13:53:55  2405 75804 14.9
2011-07-13 13:54:55  2405 78388 15.4
2011-07-13 13:55:55  2405 80544 15.9
2011-07-13 13:56:55  2405 81784 16.1
2011-07-13 13:57:55  2405 83720 16.5
2011-07-13 13:58:55  2405 86524 17.0




FS runs under root account and it is started as:

/usr/local/freeswitch/bin/freeswitch -nc -nonat -db /dev/shm -conf
/usr/local/freeswitch/conf/ -log /usr/local/freeswitch/log/ -run
/usr/local/freeswitch/run/


Also ulimit is like this;

        ulimit -c unlimited
        ulimit -d unlimited
        ulimit -f unlimited
        ulimit -i unlimited
        ulimit -n 999999
        ulimit -q unlimited
        ulimit -u unlimited
        ulimit -v unlimited
        ulimit -x unlimited
        ulimit -s 240
        ulimit -l unlimited


so, does anyone have a clue on why the memory goes up ?
... what I'm doing wrong ?

Regards,
Tihomir.
-------------- next part --------------
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From jeff at jefflenk.com  Wed Jul 13 19:41:27 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 13 Jul 2011 08:41:27 -0700 (PDT)
Subject: [Freeswitch-users] Configuration Server
In-Reply-To: <CAAYf2LsFNwRCw7LJg5gsYDG4C5KLAbqmzdP5wD+8qyqDeNTH0A@mail.gmail.com>
References: <CAAYf2Lvv1f8QBmDL9scJm45KcDA7CoVbCmGOcSwBvAH8bnjMVA@mail.gmail.com>
	<CAFYZzbvVzfRt60neJC+DyDQMpsH+iWt7Ubv6YWALjN1fdf4fbw@mail.gmail.com>
	<CAAYf2Luggy0mghz2EFbZFpmOc1dW4M9-iSpjCaS3zpSb1Nj7Og@mail.gmail.com>
	<CAKbxfG9DzHz0HrBvKRpJvbezdaV-kAAKJwNH1iEG7D2+XHDVYg@mail.gmail.com>
	<CAAYf2LsFNwRCw7LJg5gsYDG4C5KLAbqmzdP5wD+8qyqDeNTH0A@mail.gmail.com>
Message-ID: <1310571687415-6579526.post@n2.nabble.com>

I have been considering looking into FS related firewall issues on windows
but have not had the drive to do so. Please do join us for the weekly conf
sometime.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Configuration-Server-tp6554539p6579526.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From justlikeef at gmail.com  Wed Jul 13 19:41:58 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Wed, 13 Jul 2011 11:41:58 -0400
Subject: [Freeswitch-users] Still having problems making calls to phones
	behind double nat
Message-ID: <201107131141.59013.justlikeef@gmail.com>

I am still unable to figure this one out and would appreciate any help.

I have two phones at a remote location that can call out through the switch and have two way audio, but no one can call into them.  The switch returns a 606 User Not registered even though show registrations seems to indicate that they are. 

I don't see anything in the sip trace to indicate that it actually tried to contact the destination user, but INFO keepalives are going back and forth...

Configuration and sofia log are here: http://pastebin.freeswitch.org/16784


From jeff at jefflenk.com  Wed Jul 13 19:45:22 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 13 Jul 2011 08:45:22 -0700 (PDT)
Subject: [Freeswitch-users] Google Voice for Windows version of FS
In-Reply-To: <1310511452.58175.YahooMailNeo@web121618.mail.ne1.yahoo.com>
References: <1310468651.13286.YahooMailNeo@web121612.mail.ne1.yahoo.com>
	<1310479518002-6575189.post@n2.nabble.com>
	<1310482854.79304.YahooMailNeo@web121610.mail.ne1.yahoo.com>
	<1310483885842-6575490.post@n2.nabble.com>
	<1310511452.58175.YahooMailNeo@web121618.mail.ne1.yahoo.com>
Message-ID: <1310571922569-6579538.post@n2.nabble.com>

have you verified that your dingaling dll actually is referencing
gnutls(using depends or equiv)?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Google-Voice-for-Windows-version-of-FS-tp6574607p6579538.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Wed Jul 13 19:48:56 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Jul 2011 10:48:56 -0500
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
Message-ID: <CAKbxfG9CG0Q11guXN=ONM+Vg8oc1do_ABdn1=6Wh8DywO-0a4w@mail.gmail.com>

4 days is a long time.
At the moment you decide to post here you should try latest again to be safe.

Are you saying you are using sipp or something to call this extension?


On Wed, Jul 13, 2011 at 10:27 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
> hello,
>
> i upgraded to the latest GIT compiled configured FS without any issues...
> everything looks great... but when i put this into some traffic i noticed FS
> uses more and more memory... until it finished into swap. I didn't have
> issues like that previously but unfortunately i cannot recall what was the
> version with no issues.
>
> The server is 64bit CentOS 5.4 and the FS version is the latest (well 4 days
> old tops). Im using just the standard modules (the list provided below) and
> FS keeps leaking memory
>
> ??? <load module="mod_console"/>
> ??? <load module="mod_logfile"/>
> ??? <load module="mod_enum"/>
> ??? <load module="mod_cdr_csv"/>
> ??? <load module="mod_event_socket"/>
> ??? <load module="mod_sofia"/>
> ??? <load module="mod_loopback"/>
> ??? <load module="mod_commands"/>
> ??? <load module="mod_conference"/>
> ??? <load module="mod_db"/>
> ??? <load module="mod_dptools"/>
> ??? <load module="mod_expr"/>
> ??? <load module="mod_fifo"/>
> ??? <load module="mod_hash"/>
> ??? <load module="mod_voicemail"/>
> ??? <load module="mod_esf"/>
> ??? <load module="mod_fsv"/>
> ??? <load module="mod_cluechoo"/>
> ??? <load module="mod_valet_parking"/>
> ??? <load module="mod_dialplan_xml"/>
> ??? <load module="mod_dialplan_asterisk"/>
> ??? <load module="mod_spandsp"/>
> ??? <load module="mod_g723_1"/>
> ??? <load module="mod_g729"/>
> ??? <load module="mod_amr"/>
> ??? <load module="mod_ilbc"/>
> ??? <load module="mod_speex"/>
> ??? <load module="mod_h26x"/>
> ??? <load module="mod_siren"/>
> ??? <load module="mod_sndfile"/>
> ??? <load module="mod_native_file"/>
> ??? <load module="mod_local_stream"/>
> ??? <load module="mod_tone_stream"/>
> ??? <load module="mod_spidermonkey"/>
> ??? <load module="mod_lua"/>
> ??? <load module="mod_say_en"/>
>
>
> This is the dialplan im triggering...
>
> ?? <extension name="doRedirect">
> ????? <condition field="destination_number" expression="^300030.*$">
> ???????? <action application="info"/>
> ???????? <action application="redirect"
> data="<sip:0999901234567889 at 10.1.1.11:5060;user=phone>;q=0.99"/>
> ???????? <action application="hangup"/> <!-- well ... not really needed but
> only Got knows for sure-->
> ????? </condition>
> ?? </extension>
>
> I have i have tried a pure "hello world" dialplan as:
>
> ?? <extension name="testDP">
> ????? <condition field="destination_number" expression="^300030.*$">
> ???????? <action application="info"/>
> ???????? <action application="hangup"/>
> ????? </condition>
> ?? </extension>
>
> But the situation is exactly the same.
>
>
>
> This is the behavior i'm having .... memory just keep growing and growing
> until it hits the swap where the server is doomed.
>
> ?tail -f smaps_log.log (this is from cat /proc/$PID/smaps | grep heap)
> TIM,SIZE,RSS,SHARED_CLEAN,SHARED_DIRTY,PRIVATE_CLEAN,PRIVATE_DIRTY,SWAP
> 2011-07-13 13:50:34,19268,18916,0,0,0,18916,0
> 2011-07-13 13:51:34,19268,18972,0,0,0,18972,0
> 2011-07-13 13:52:34,20324,20088,0,0,0,20088,0
> 2011-07-13 13:53:34,21264,20920,0,0,0,20920,0
> 2011-07-13 13:54:34,21048,20856,0,0,0,20856,0
> 2011-07-13 13:55:35,22396,22076,0,0,0,22076,0
> 2011-07-13 13:56:35,22396,22164,0,0,0,22256,0
> 2011-07-13 13:57:35,23220,22936,0,0,0,22936,0
> 2011-07-13 13:58:35,23480,23288,0,0,0,23288,0
>
> $ tail -f mem.log (this is from ps)
> TIMESTAMP??????????? PID? RSS??? %MEM
> 2011-07-13 13:48:55? 2405 65144 12.8
> 2011-07-13 13:49:55? 2405 66932 13.2
> 2011-07-13 13:50:55? 2405 69576 13.7
> 2011-07-13 13:51:55? 2405 71552 14.1
> 2011-07-13 13:52:55? 2405 73576 14.5
> 2011-07-13 13:53:55? 2405 75804 14.9
> 2011-07-13 13:54:55? 2405 78388 15.4
> 2011-07-13 13:55:55? 2405 80544 15.9
> 2011-07-13 13:56:55? 2405 81784 16.1
> 2011-07-13 13:57:55? 2405 83720 16.5
> 2011-07-13 13:58:55? 2405 86524 17.0
>
>
>
>
> FS runs under root account and it is started as:
>
> /usr/local/freeswitch/bin/freeswitch -nc -nonat -db /dev/shm -conf
> /usr/local/freeswitch/conf/ -log /usr/local/freeswitch/log/ -run
> /usr/local/freeswitch/run/
>
>
> Also ulimit is like this;
>
> ??????? ulimit -c unlimited
> ??????? ulimit -d unlimited
> ??????? ulimit -f unlimited
> ??????? ulimit -i unlimited
> ??????? ulimit -n 999999
> ??????? ulimit -q unlimited
> ??????? ulimit -u unlimited
> ??????? ulimit -v unlimited
> ??????? ulimit -x unlimited
> ??????? ulimit -s 240
> ??????? ulimit -l unlimited
>
>
> so, does anyone have a clue on why the memory goes up ?
> ... what I'm doing wrong ?
>
> Regards,
> Tihomir.
>
>
>
>
>
>
> _______________________________________________
> 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
>
>



-- 
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  Wed Jul 13 19:52:38 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Jul 2011 10:52:38 -0500
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAKbxfG9CG0Q11guXN=ONM+Vg8oc1do_ABdn1=6Wh8DywO-0a4w@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
	<CAKbxfG9CG0Q11guXN=ONM+Vg8oc1do_ABdn1=6Wh8DywO-0a4w@mail.gmail.com>
Message-ID: <CAKbxfG-F-VKpe1xSWUYbyQS1-W+4HSG-+6PpWwRrEPF+a2XGpg@mail.gmail.com>

oh and btw

you should be filing a jira not posting it to here


On Wed, Jul 13, 2011 at 10:48 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> 4 days is a long time.
> At the moment you decide to post here you should try latest again to be safe.
>
> Are you saying you are using sipp or something to call this extension?
>
>
> On Wed, Jul 13, 2011 at 10:27 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>> hello,
>>
>> i upgraded to the latest GIT compiled configured FS without any issues...
>> everything looks great... but when i put this into some traffic i noticed FS
>> uses more and more memory... until it finished into swap. I didn't have
>> issues like that previously but unfortunately i cannot recall what was the
>> version with no issues.
>>
>> The server is 64bit CentOS 5.4 and the FS version is the latest (well 4 days
>> old tops). Im using just the standard modules (the list provided below) and
>> FS keeps leaking memory
>>
>> ??? <load module="mod_console"/>
>> ??? <load module="mod_logfile"/>
>> ??? <load module="mod_enum"/>
>> ??? <load module="mod_cdr_csv"/>
>> ??? <load module="mod_event_socket"/>
>> ??? <load module="mod_sofia"/>
>> ??? <load module="mod_loopback"/>
>> ??? <load module="mod_commands"/>
>> ??? <load module="mod_conference"/>
>> ??? <load module="mod_db"/>
>> ??? <load module="mod_dptools"/>
>> ??? <load module="mod_expr"/>
>> ??? <load module="mod_fifo"/>
>> ??? <load module="mod_hash"/>
>> ??? <load module="mod_voicemail"/>
>> ??? <load module="mod_esf"/>
>> ??? <load module="mod_fsv"/>
>> ??? <load module="mod_cluechoo"/>
>> ??? <load module="mod_valet_parking"/>
>> ??? <load module="mod_dialplan_xml"/>
>> ??? <load module="mod_dialplan_asterisk"/>
>> ??? <load module="mod_spandsp"/>
>> ??? <load module="mod_g723_1"/>
>> ??? <load module="mod_g729"/>
>> ??? <load module="mod_amr"/>
>> ??? <load module="mod_ilbc"/>
>> ??? <load module="mod_speex"/>
>> ??? <load module="mod_h26x"/>
>> ??? <load module="mod_siren"/>
>> ??? <load module="mod_sndfile"/>
>> ??? <load module="mod_native_file"/>
>> ??? <load module="mod_local_stream"/>
>> ??? <load module="mod_tone_stream"/>
>> ??? <load module="mod_spidermonkey"/>
>> ??? <load module="mod_lua"/>
>> ??? <load module="mod_say_en"/>
>>
>>
>> This is the dialplan im triggering...
>>
>> ?? <extension name="doRedirect">
>> ????? <condition field="destination_number" expression="^300030.*$">
>> ???????? <action application="info"/>
>> ???????? <action application="redirect"
>> data="<sip:0999901234567889 at 10.1.1.11:5060;user=phone>;q=0.99"/>
>> ???????? <action application="hangup"/> <!-- well ... not really needed but
>> only Got knows for sure-->
>> ????? </condition>
>> ?? </extension>
>>
>> I have i have tried a pure "hello world" dialplan as:
>>
>> ?? <extension name="testDP">
>> ????? <condition field="destination_number" expression="^300030.*$">
>> ???????? <action application="info"/>
>> ???????? <action application="hangup"/>
>> ????? </condition>
>> ?? </extension>
>>
>> But the situation is exactly the same.
>>
>>
>>
>> This is the behavior i'm having .... memory just keep growing and growing
>> until it hits the swap where the server is doomed.
>>
>> ?tail -f smaps_log.log (this is from cat /proc/$PID/smaps | grep heap)
>> TIM,SIZE,RSS,SHARED_CLEAN,SHARED_DIRTY,PRIVATE_CLEAN,PRIVATE_DIRTY,SWAP
>> 2011-07-13 13:50:34,19268,18916,0,0,0,18916,0
>> 2011-07-13 13:51:34,19268,18972,0,0,0,18972,0
>> 2011-07-13 13:52:34,20324,20088,0,0,0,20088,0
>> 2011-07-13 13:53:34,21264,20920,0,0,0,20920,0
>> 2011-07-13 13:54:34,21048,20856,0,0,0,20856,0
>> 2011-07-13 13:55:35,22396,22076,0,0,0,22076,0
>> 2011-07-13 13:56:35,22396,22164,0,0,0,22256,0
>> 2011-07-13 13:57:35,23220,22936,0,0,0,22936,0
>> 2011-07-13 13:58:35,23480,23288,0,0,0,23288,0
>>
>> $ tail -f mem.log (this is from ps)
>> TIMESTAMP??????????? PID? RSS??? %MEM
>> 2011-07-13 13:48:55? 2405 65144 12.8
>> 2011-07-13 13:49:55? 2405 66932 13.2
>> 2011-07-13 13:50:55? 2405 69576 13.7
>> 2011-07-13 13:51:55? 2405 71552 14.1
>> 2011-07-13 13:52:55? 2405 73576 14.5
>> 2011-07-13 13:53:55? 2405 75804 14.9
>> 2011-07-13 13:54:55? 2405 78388 15.4
>> 2011-07-13 13:55:55? 2405 80544 15.9
>> 2011-07-13 13:56:55? 2405 81784 16.1
>> 2011-07-13 13:57:55? 2405 83720 16.5
>> 2011-07-13 13:58:55? 2405 86524 17.0
>>
>>
>>
>>
>> FS runs under root account and it is started as:
>>
>> /usr/local/freeswitch/bin/freeswitch -nc -nonat -db /dev/shm -conf
>> /usr/local/freeswitch/conf/ -log /usr/local/freeswitch/log/ -run
>> /usr/local/freeswitch/run/
>>
>>
>> Also ulimit is like this;
>>
>> ??????? ulimit -c unlimited
>> ??????? ulimit -d unlimited
>> ??????? ulimit -f unlimited
>> ??????? ulimit -i unlimited
>> ??????? ulimit -n 999999
>> ??????? ulimit -q unlimited
>> ??????? ulimit -u unlimited
>> ??????? ulimit -v unlimited
>> ??????? ulimit -x unlimited
>> ??????? ulimit -s 240
>> ??????? ulimit -l unlimited
>>
>>
>> so, does anyone have a clue on why the memory goes up ?
>> ... what I'm doing wrong ?
>>
>> Regards,
>> Tihomir.
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
>
>
> --
> 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
>



-- 
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 roger.castaldo at gmail.com  Wed Jul 13 20:14:06 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Wed, 13 Jul 2011 12:14:06 -0400
Subject: [Freeswitch-users] Configuration Server
In-Reply-To: <1310571687415-6579526.post@n2.nabble.com>
References: <CAAYf2Lvv1f8QBmDL9scJm45KcDA7CoVbCmGOcSwBvAH8bnjMVA@mail.gmail.com>
	<CAFYZzbvVzfRt60neJC+DyDQMpsH+iWt7Ubv6YWALjN1fdf4fbw@mail.gmail.com>
	<CAAYf2Luggy0mghz2EFbZFpmOc1dW4M9-iSpjCaS3zpSb1Nj7Og@mail.gmail.com>
	<CAKbxfG9DzHz0HrBvKRpJvbezdaV-kAAKJwNH1iEG7D2+XHDVYg@mail.gmail.com>
	<CAAYf2LsFNwRCw7LJg5gsYDG4C5KLAbqmzdP5wD+8qyqDeNTH0A@mail.gmail.com>
	<1310571687415-6579526.post@n2.nabble.com>
Message-ID: <CAAYf2Ls+HmBkAA4V2sfP7cE3yQAjBJfcRpC4s-oJ6=xr9BdpyQ@mail.gmail.com>

I may have to do that, on a different note this configuration server
actually runs independent of freeswitch and if I have my way might be able
to handle controlling multiple servers.  It does everything through some
basic file system operations as well as the in and outbound event sockets.
At one point I was looking into an equivalent for iptables in windows to be
able to handle setting it up in windows but that was put on the back burner
because I was able to build a minmal live CD using Slitaz as a base that
boots, runs freeswitch as well as the config server and uses somewhere
around 100MB for the OS and has the config server sitting between 20MB and
60MB so i decided windows can go to the wayside for now, unless someone
knows of a good firewall utility that can be used through .Net to setup and
control the windows firewall to a refinement level that I can get from
iptables?

On Wed, Jul 13, 2011 at 11:41 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

> I have been considering looking into FS related firewall issues on windows
> but have not had the drive to do so. Please do join us for the weekly conf
> sometime.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Configuration-Server-tp6554539p6579526.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> 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 msc at freeswitch.org  Wed Jul 13 20:15:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Jul 2011 09:15:28 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxVE_zSqZKjYXkA50NNjTY7R6Ouv7Hd5-MkaDU5ghPpAWw@mail.gmail.com>

Hello all,

Today's agenda is here:

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

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

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

Talk to you soon!

-Michael
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From steveayre at gmail.com  Wed Jul 13 20:17:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 13 Jul 2011 17:17:01 +0100
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
Message-ID: <CAFiqYukgW1UOgxNUZs8rPvWNmXx5sDVmGS_D+r_xdLMsXg_i4Q@mail.gmail.com>

Does "fsctl reclaim_mem" help at all?

-Steve


On 13 July 2011 16:27, Tihomir Culjaga <tculjaga at gmail.com> wrote:
> hello,
>
> i upgraded to the latest GIT compiled configured FS without any issues...
> everything looks great... but when i put this into some traffic i noticed FS
> uses more and more memory... until it finished into swap. I didn't have
> issues like that previously but unfortunately i cannot recall what was the
> version with no issues.
>
> The server is 64bit CentOS 5.4 and the FS version is the latest (well 4 days
> old tops). Im using just the standard modules (the list provided below) and
> FS keeps leaking memory
>
> ??? <load module="mod_console"/>
> ??? <load module="mod_logfile"/>
> ??? <load module="mod_enum"/>
> ??? <load module="mod_cdr_csv"/>
> ??? <load module="mod_event_socket"/>
> ??? <load module="mod_sofia"/>
> ??? <load module="mod_loopback"/>
> ??? <load module="mod_commands"/>
> ??? <load module="mod_conference"/>
> ??? <load module="mod_db"/>
> ??? <load module="mod_dptools"/>
> ??? <load module="mod_expr"/>
> ??? <load module="mod_fifo"/>
> ??? <load module="mod_hash"/>
> ??? <load module="mod_voicemail"/>
> ??? <load module="mod_esf"/>
> ??? <load module="mod_fsv"/>
> ??? <load module="mod_cluechoo"/>
> ??? <load module="mod_valet_parking"/>
> ??? <load module="mod_dialplan_xml"/>
> ??? <load module="mod_dialplan_asterisk"/>
> ??? <load module="mod_spandsp"/>
> ??? <load module="mod_g723_1"/>
> ??? <load module="mod_g729"/>
> ??? <load module="mod_amr"/>
> ??? <load module="mod_ilbc"/>
> ??? <load module="mod_speex"/>
> ??? <load module="mod_h26x"/>
> ??? <load module="mod_siren"/>
> ??? <load module="mod_sndfile"/>
> ??? <load module="mod_native_file"/>
> ??? <load module="mod_local_stream"/>
> ??? <load module="mod_tone_stream"/>
> ??? <load module="mod_spidermonkey"/>
> ??? <load module="mod_lua"/>
> ??? <load module="mod_say_en"/>
>
>
> This is the dialplan im triggering...
>
> ?? <extension name="doRedirect">
> ????? <condition field="destination_number" expression="^300030.*$">
> ???????? <action application="info"/>
> ???????? <action application="redirect"
> data="<sip:0999901234567889 at 10.1.1.11:5060;user=phone>;q=0.99"/>
> ???????? <action application="hangup"/> <!-- well ... not really needed but
> only Got knows for sure-->
> ????? </condition>
> ?? </extension>
>
> I have i have tried a pure "hello world" dialplan as:
>
> ?? <extension name="testDP">
> ????? <condition field="destination_number" expression="^300030.*$">
> ???????? <action application="info"/>
> ???????? <action application="hangup"/>
> ????? </condition>
> ?? </extension>
>
> But the situation is exactly the same.
>
>
>
> This is the behavior i'm having .... memory just keep growing and growing
> until it hits the swap where the server is doomed.
>
> ?tail -f smaps_log.log (this is from cat /proc/$PID/smaps | grep heap)
> TIM,SIZE,RSS,SHARED_CLEAN,SHARED_DIRTY,PRIVATE_CLEAN,PRIVATE_DIRTY,SWAP
> 2011-07-13 13:50:34,19268,18916,0,0,0,18916,0
> 2011-07-13 13:51:34,19268,18972,0,0,0,18972,0
> 2011-07-13 13:52:34,20324,20088,0,0,0,20088,0
> 2011-07-13 13:53:34,21264,20920,0,0,0,20920,0
> 2011-07-13 13:54:34,21048,20856,0,0,0,20856,0
> 2011-07-13 13:55:35,22396,22076,0,0,0,22076,0
> 2011-07-13 13:56:35,22396,22164,0,0,0,22256,0
> 2011-07-13 13:57:35,23220,22936,0,0,0,22936,0
> 2011-07-13 13:58:35,23480,23288,0,0,0,23288,0
>
> $ tail -f mem.log (this is from ps)
> TIMESTAMP??????????? PID? RSS??? %MEM
> 2011-07-13 13:48:55? 2405 65144 12.8
> 2011-07-13 13:49:55? 2405 66932 13.2
> 2011-07-13 13:50:55? 2405 69576 13.7
> 2011-07-13 13:51:55? 2405 71552 14.1
> 2011-07-13 13:52:55? 2405 73576 14.5
> 2011-07-13 13:53:55? 2405 75804 14.9
> 2011-07-13 13:54:55? 2405 78388 15.4
> 2011-07-13 13:55:55? 2405 80544 15.9
> 2011-07-13 13:56:55? 2405 81784 16.1
> 2011-07-13 13:57:55? 2405 83720 16.5
> 2011-07-13 13:58:55? 2405 86524 17.0
>
>
>
>
> FS runs under root account and it is started as:
>
> /usr/local/freeswitch/bin/freeswitch -nc -nonat -db /dev/shm -conf
> /usr/local/freeswitch/conf/ -log /usr/local/freeswitch/log/ -run
> /usr/local/freeswitch/run/
>
>
> Also ulimit is like this;
>
> ??????? ulimit -c unlimited
> ??????? ulimit -d unlimited
> ??????? ulimit -f unlimited
> ??????? ulimit -i unlimited
> ??????? ulimit -n 999999
> ??????? ulimit -q unlimited
> ??????? ulimit -u unlimited
> ??????? ulimit -v unlimited
> ??????? ulimit -x unlimited
> ??????? ulimit -s 240
> ??????? ulimit -l unlimited
>
>
> so, does anyone have a clue on why the memory goes up ?
> ... what I'm doing wrong ?
>
> Regards,
> Tihomir.
>
>
>
>
>
>
> _______________________________________________
> 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 dgarcia at anew.com.ve  Wed Jul 13 16:41:26 2011
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Wed, 13 Jul 2011 08:11:26 -0430
Subject: [Freeswitch-users] V92 modem with freeswitch
In-Reply-To: <CAOcuKucME9wAXkUfVPRWXTzqTs0WhtYyEr=n1KNLUbw0L9PCxw@mail.gmail.com>
References: <CAOcuKucME9wAXkUfVPRWXTzqTs0WhtYyEr=n1KNLUbw0L9PCxw@mail.gmail.com>
Message-ID: <4E1D9276.6070400@anew.com.ve>

Hi Ankit,

You could use your V.92 56k PCI Modem old hardware but you should have 
to build a driver for Freeswitch, Freeswitch have a list of hardware 
that have been tested.

I recomend you to take a look to freeswitch web site. If you would iike 
to test telephony functions (in VoIP mode, SIP) you dont need telephony 
hardware. But if you would like interface your FS box with a PSTN line/ 
POTS buy hardware from digium or sangoma like TDM400P (digium) or 
(A200); they are quite cheap, around $400 or less.



On 7/13/2011 4:44 AM, ankIT WALiA wrote:
> Hi all,
>
> I found a V.92 56k PCI Modem in my old hardware.
> I want to run some basic functionality test with my PSTN phone line and FS.
> Can I connect my phone line with FS using this card. If yes how do we configure to use this card.
> I am new to telephony. I don't know if whatever I said above make sense.
> Thanks
> Ankit
>
>
> _______________________________________________
> 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
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 10.0.1390 / Virus Database: 1516/3761 - Release Date: 07/12/11
>


-- 
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 stviper at gmail.com  Wed Jul 13 20:07:00 2011
From: stviper at gmail.com (=?ISO-8859-2?B?qXRlZmFuIMh1ZGFp?=)
Date: Wed, 13 Jul 2011 18:07:00 +0200
Subject: [Freeswitch-users] Overlap dialing problem
Message-ID: <CAP3+0r3gy7CPPs3kAwOsNt6a_LLFQQZeRL9AhiTMJ81yoHO67A@mail.gmail.com>

Hi,

does FreeSwitch support for overlap dialing or did somebody try to use it?

I am using Sangoma card A102 under Windows 7 with FreeTDM. I have configured
my freetdm.conf.xml
with this two lines:

         <param name="overlap" value="yes"/>
         <param name="min_digits" value="0"/>

But when I'm dialing number (which has 14 digits) I hear disconnect tone
after dialing first 5 digits. I think that reason is because FreeTDM
module is  waiting number with minimum digits equals to "0" (see my
min_digits).
I have also try set min_digits to 20 and then dial same number with 14
digits but result was same. So, I looked to log file and found that
FreeTDM module got in thist case STATUS CONFIRM message but he was still
waiting for next digits and his time expired.

Can you help me please?

Thanks a lot.
Stefan
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From msc at freeswitch.org  Wed Jul 13 20:26:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Jul 2011 09:26:44 -0700
Subject: [Freeswitch-users] How to set the member-flags in mod_conference
In-Reply-To: <4E1D942B.9070809@bksys.co.in>
References: <4E1D3D2C.6070109@bksys.co.in>
	<4E1D942B.9070809@bksys.co.in>
Message-ID: <CAKzWOxXBEzOzTvX7Kxoxb07iiWDdvcX+27sFrypW9M9Tp_MUAg@mail.gmail.com>

>
>
> That can be done using flags
> ex :
> <action application="conference" data="room1+flags{endconf}"/>
>

Did you try this:
<action application="conference" data="room1+flags{${endconf}}"/>
Also, you might need to add inline="true" on your set app:
<action application="set" data="member-flags=mute" inline="true"/>

Let us know...

-MC
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From yungwei at resolvity.com  Wed Jul 13 20:28:55 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 13 Jul 2011 12:28:55 -0400
Subject: [Freeswitch-users] multiple active grammars supported?
Message-ID: <33095823FD21DF429B481B5163264B7950CBC034F1@VMBX102.ihostexchange.net>

Hi,

I tried the pizza demo shipped with freeswitch, and I am wondering if multiple active grammars are supported.

For example:
question: was this the pizza you had in mind?
response: yes/no (defined in grammar 1) Or 
          help   (defined in grammar 2; this case will be handled differently)

Thanks.


From steveu at coppice.org  Wed Jul 13 20:53:07 2011
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 14 Jul 2011 00:53:07 +0800
Subject: [Freeswitch-users] V92 modem with freeswitch
In-Reply-To: <CAOcuKucME9wAXkUfVPRWXTzqTs0WhtYyEr=n1KNLUbw0L9PCxw@mail.gmail.com>
References: <CAOcuKucME9wAXkUfVPRWXTzqTs0WhtYyEr=n1KNLUbw0L9PCxw@mail.gmail.com>
Message-ID: <4E1DCD73.3030304@coppice.org>

On 07/13/2011 05:14 PM, ankIT WALiA wrote:
> Hi all,
>
> I found a V.92 56k PCI Modem in my old hardware.
> I want to run some basic functionality test with my PSTN phone line and FS.
> Can I connect my phone line with FS using this card. If yes how do we configure to use this card.
> I am new to telephony. I don't know if whatever I said above make sense.
>
If it uses the chip which the Digium x100p card uses, their driver 
should work for you. If it uses any other chip you either give up or 
search for documentation and start developing your own driver.

Steve


From wstephen80 at gmail.com  Wed Jul 13 20:59:44 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Wed, 13 Jul 2011 18:59:44 +0200
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
Message-ID: <CAO4LeU0g_88Yti9A84NCrAkMh1vDF3=xhsmRAPQ1+OsUoLpwSg@mail.gmail.com>

I had a similar issue with commit 9cf44f3a5ee7cba9fc378447780499d41c4de671
of 6 July with a memory increase rate of 2Gb/day but now I have done a
revert to an old commit (2246f3ca75eee93a7e7f4409aa816513b99e657a) for a
reason not related to the memory leak.
For this reason I have not yet open a jira.
Due to a feature released recently, I'll do a git pull in next hours so I'll
share the info I have.

Stephen


On Wed, Jul 13, 2011 at 5:27 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hello,
>
> i upgraded to the latest GIT compiled configured FS without any issues...
> everything looks great... but when i put this into some traffic i noticed FS
> uses more and more memory... until it finished into swap. I didn't have
> issues like that previously but unfortunately i cannot recall what was the
> version with no issues.
>
> The server is 64bit CentOS 5.4 and the FS version is the latest (well 4
> days old tops). Im using just the standard modules (the list provided below)
> and FS keeps leaking memory
>
>     <load module="mod_console"/>
>     <load module="mod_logfile"/>
>     <load module="mod_enum"/>
>     <load module="mod_cdr_csv"/>
>     <load module="mod_event_socket"/>
>     <load module="mod_sofia"/>
>     <load module="mod_loopback"/>
>     <load module="mod_commands"/>
>     <load module="mod_conference"/>
>     <load module="mod_db"/>
>     <load module="mod_dptools"/>
>     <load module="mod_expr"/>
>     <load module="mod_fifo"/>
>     <load module="mod_hash"/>
>     <load module="mod_voicemail"/>
>     <load module="mod_esf"/>
>     <load module="mod_fsv"/>
>     <load module="mod_cluechoo"/>
>     <load module="mod_valet_parking"/>
>     <load module="mod_dialplan_xml"/>
>     <load module="mod_dialplan_asterisk"/>
>     <load module="mod_spandsp"/>
>     <load module="mod_g723_1"/>
>     <load module="mod_g729"/>
>     <load module="mod_amr"/>
>     <load module="mod_ilbc"/>
>     <load module="mod_speex"/>
>     <load module="mod_h26x"/>
>     <load module="mod_siren"/>
>     <load module="mod_sndfile"/>
>     <load module="mod_native_file"/>
>     <load module="mod_local_stream"/>
>     <load module="mod_tone_stream"/>
>     <load module="mod_spidermonkey"/>
>     <load module="mod_lua"/>
>     <load module="mod_say_en"/>
>
>
> This is the dialplan im triggering...
>
>    <extension name="doRedirect">
>       <condition field="destination_number" expression="^300030.*$">
>          <action application="info"/>
>          <action application="redirect"
> data="<sip:0999901234567889 at 10.1.1.11:5060;user=phone>;q=0.99"/>
>          <action application="hangup"/> <!-- well ... not really needed but
> only Got knows for sure-->
>       </condition>
>    </extension>
>
> I have i have tried a pure "hello world" dialplan as:
>
>    <extension name="testDP">
>       <condition field="destination_number" expression="^300030.*$">
>          <action application="info"/>
>          <action application="hangup"/>
>       </condition>
>    </extension>
>
> But the situation is exactly the same.
>
>
>
> This is the behavior i'm having .... memory just keep growing and growing
> until it hits the swap where the server is doomed.
>
>  tail -f smaps_log.log (this is from cat /proc/$PID/smaps | grep heap)
> TIM,SIZE,RSS,SHARED_CLEAN,SHARED_DIRTY,PRIVATE_CLEAN,PRIVATE_DIRTY,SWAP
> 2011-07-13 13:50:34,19268,18916,0,0,0,18916,0
> 2011-07-13 13:51:34,19268,18972,0,0,0,18972,0
> 2011-07-13 13:52:34,20324,20088,0,0,0,20088,0
> 2011-07-13 13:53:34,21264,20920,0,0,0,20920,0
> 2011-07-13 13:54:34,21048,20856,0,0,0,20856,0
> 2011-07-13 13:55:35,22396,22076,0,0,0,22076,0
> 2011-07-13 13:56:35,22396,22164,0,0,0,22256,0
> 2011-07-13 13:57:35,23220,22936,0,0,0,22936,0
> 2011-07-13 13:58:35,23480,23288,0,0,0,23288,0
>
> $ tail -f mem.log (this is from ps)
> TIMESTAMP            PID  RSS    %MEM
> 2011-07-13 13:48:55  2405 65144 12.8
> 2011-07-13 13:49:55  2405 66932 13.2
> 2011-07-13 13:50:55  2405 69576 13.7
> 2011-07-13 13:51:55  2405 71552 14.1
> 2011-07-13 13:52:55  2405 73576 14.5
> 2011-07-13 13:53:55  2405 75804 14.9
> 2011-07-13 13:54:55  2405 78388 15.4
> 2011-07-13 13:55:55  2405 80544 15.9
> 2011-07-13 13:56:55  2405 81784 16.1
> 2011-07-13 13:57:55  2405 83720 16.5
> 2011-07-13 13:58:55  2405 86524 17.0
>
>
>
>
> FS runs under root account and it is started as:
>
> /usr/local/freeswitch/bin/freeswitch -nc -nonat -db /dev/shm -conf
> /usr/local/freeswitch/conf/ -log /usr/local/freeswitch/log/ -run
> /usr/local/freeswitch/run/
>
>
> Also ulimit is like this;
>
>         ulimit -c unlimited
>         ulimit -d unlimited
>         ulimit -f unlimited
>         ulimit -i unlimited
>         ulimit -n 999999
>         ulimit -q unlimited
>         ulimit -u unlimited
>         ulimit -v unlimited
>         ulimit -x unlimited
>         ulimit -s 240
>         ulimit -l unlimited
>
>
> so, does anyone have a clue on why the memory goes up ?
> ... what I'm doing wrong ?
>
> Regards,
> Tihomir.
>
>
>
>
>
>
> _______________________________________________
> 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 anthony.minessale at gmail.com  Wed Jul 13 21:40:55 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Jul 2011 12:40:55 -0500
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAO4LeU0g_88Yti9A84NCrAkMh1vDF3=xhsmRAPQ1+OsUoLpwSg@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
	<CAO4LeU0g_88Yti9A84NCrAkMh1vDF3=xhsmRAPQ1+OsUoLpwSg@mail.gmail.com>
Message-ID: <CAKbxfG_K1LRUNZcdCzgpQSkR+wZcD=0ZdxkfvUtv17OSZaKT_A@mail.gmail.com>

Try this commit (or newer)  I think it fixes the problem

commit e339b549e014f30c5de256cf2b3ba94a91bb06e3
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Wed Jul 13 10:37:32 2011 -0500

    FS-3386 this is probably relevant, try this revision




On Wed, Jul 13, 2011 at 11:59 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> I had a similar issue with commit?9cf44f3a5ee7cba9fc378447780499d41c4de671
> of 6 July with a memory increase rate of 2Gb/day but now?I have done a
> revert to an old commit (2246f3ca75eee93a7e7f4409aa816513b99e657a) for a
> reason not related to the memory leak.
> For this reason I have not yet open a jira.
> Due to a feature released recently, I'll do a git pull in next hours so I'll
> share the info I have.
> Stephen
>
> On Wed, Jul 13, 2011 at 5:27 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>>
>> hello,
>>
>> i upgraded to the latest GIT compiled configured FS without any issues...
>> everything looks great... but when i put this into some traffic i noticed FS
>> uses more and more memory... until it finished into swap. I didn't have
>> issues like that previously but unfortunately i cannot recall what was the
>> version with no issues.
>>
>> The server is 64bit CentOS 5.4 and the FS version is the latest (well 4
>> days old tops). Im using just the standard modules (the list provided below)
>> and FS keeps leaking memory
>>
>> ??? <load module="mod_console"/>
>> ??? <load module="mod_logfile"/>
>> ??? <load module="mod_enum"/>
>> ??? <load module="mod_cdr_csv"/>
>> ??? <load module="mod_event_socket"/>
>> ??? <load module="mod_sofia"/>
>> ??? <load module="mod_loopback"/>
>> ??? <load module="mod_commands"/>
>> ??? <load module="mod_conference"/>
>> ??? <load module="mod_db"/>
>> ??? <load module="mod_dptools"/>
>> ??? <load module="mod_expr"/>
>> ??? <load module="mod_fifo"/>
>> ??? <load module="mod_hash"/>
>> ??? <load module="mod_voicemail"/>
>> ??? <load module="mod_esf"/>
>> ??? <load module="mod_fsv"/>
>> ??? <load module="mod_cluechoo"/>
>> ??? <load module="mod_valet_parking"/>
>> ??? <load module="mod_dialplan_xml"/>
>> ??? <load module="mod_dialplan_asterisk"/>
>> ??? <load module="mod_spandsp"/>
>> ??? <load module="mod_g723_1"/>
>> ??? <load module="mod_g729"/>
>> ??? <load module="mod_amr"/>
>> ??? <load module="mod_ilbc"/>
>> ??? <load module="mod_speex"/>
>> ??? <load module="mod_h26x"/>
>> ??? <load module="mod_siren"/>
>> ??? <load module="mod_sndfile"/>
>> ??? <load module="mod_native_file"/>
>> ??? <load module="mod_local_stream"/>
>> ??? <load module="mod_tone_stream"/>
>> ??? <load module="mod_spidermonkey"/>
>> ??? <load module="mod_lua"/>
>> ??? <load module="mod_say_en"/>
>>
>>
>> This is the dialplan im triggering...
>>
>> ?? <extension name="doRedirect">
>> ????? <condition field="destination_number" expression="^300030.*$">
>> ???????? <action application="info"/>
>> ???????? <action application="redirect"
>> data="<sip:0999901234567889 at 10.1.1.11:5060;user=phone>;q=0.99"/>
>> ???????? <action application="hangup"/> <!-- well ... not really needed
>> but only Got knows for sure-->
>> ????? </condition>
>> ?? </extension>
>>
>> I have i have tried a pure "hello world" dialplan as:
>>
>> ?? <extension name="testDP">
>> ????? <condition field="destination_number" expression="^300030.*$">
>> ???????? <action application="info"/>
>> ???????? <action application="hangup"/>
>> ????? </condition>
>> ?? </extension>
>>
>> But the situation is exactly the same.
>>
>>
>>
>> This is the behavior i'm having .... memory just keep growing and growing
>> until it hits the swap where the server is doomed.
>>
>> ?tail -f smaps_log.log (this is from cat /proc/$PID/smaps | grep heap)
>> TIM,SIZE,RSS,SHARED_CLEAN,SHARED_DIRTY,PRIVATE_CLEAN,PRIVATE_DIRTY,SWAP
>> 2011-07-13 13:50:34,19268,18916,0,0,0,18916,0
>> 2011-07-13 13:51:34,19268,18972,0,0,0,18972,0
>> 2011-07-13 13:52:34,20324,20088,0,0,0,20088,0
>> 2011-07-13 13:53:34,21264,20920,0,0,0,20920,0
>> 2011-07-13 13:54:34,21048,20856,0,0,0,20856,0
>> 2011-07-13 13:55:35,22396,22076,0,0,0,22076,0
>> 2011-07-13 13:56:35,22396,22164,0,0,0,22256,0
>> 2011-07-13 13:57:35,23220,22936,0,0,0,22936,0
>> 2011-07-13 13:58:35,23480,23288,0,0,0,23288,0
>>
>> $ tail -f mem.log (this is from ps)
>> TIMESTAMP??????????? PID? RSS??? %MEM
>> 2011-07-13 13:48:55? 2405 65144 12.8
>> 2011-07-13 13:49:55? 2405 66932 13.2
>> 2011-07-13 13:50:55? 2405 69576 13.7
>> 2011-07-13 13:51:55? 2405 71552 14.1
>> 2011-07-13 13:52:55? 2405 73576 14.5
>> 2011-07-13 13:53:55? 2405 75804 14.9
>> 2011-07-13 13:54:55? 2405 78388 15.4
>> 2011-07-13 13:55:55? 2405 80544 15.9
>> 2011-07-13 13:56:55? 2405 81784 16.1
>> 2011-07-13 13:57:55? 2405 83720 16.5
>> 2011-07-13 13:58:55? 2405 86524 17.0
>>
>>
>>
>>
>> FS runs under root account and it is started as:
>>
>> /usr/local/freeswitch/bin/freeswitch -nc -nonat -db /dev/shm -conf
>> /usr/local/freeswitch/conf/ -log /usr/local/freeswitch/log/ -run
>> /usr/local/freeswitch/run/
>>
>>
>> Also ulimit is like this;
>>
>> ??????? ulimit -c unlimited
>> ??????? ulimit -d unlimited
>> ??????? ulimit -f unlimited
>> ??????? ulimit -i unlimited
>> ??????? ulimit -n 999999
>> ??????? ulimit -q unlimited
>> ??????? ulimit -u unlimited
>> ??????? ulimit -v unlimited
>> ??????? ulimit -x unlimited
>> ??????? ulimit -s 240
>> ??????? ulimit -l unlimited
>>
>>
>> so, does anyone have a clue on why the memory goes up ?
>> ... what I'm doing wrong ?
>>
>> Regards,
>> Tihomir.
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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
>
>



-- 
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 freeswitch at earthspike.net  Thu Jul 14 00:16:17 2011
From: freeswitch at earthspike.net (John)
Date: Wed, 13 Jul 2011 21:16:17 +0100
Subject: [Freeswitch-users] UK ISDN outgoing CLI (was Re: FreeTDM -
 Sangoma B700 - ISDN connection questions - UK)
In-Reply-To: <CAA8_NKCs_CJKEtHG9CpDZvK9vXevGtjEA18tNoEZ0yzKc2kmYg@mail.gmail.com>
References: <4DED4854.6010203@earthspike.net>	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>	<4DF10C83.6020803@earthspike.net>	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>	<4DF1F07A.5000200@earthspike.net>	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>	<4DFB9831.4040709@earthspike.net>	<BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>	<CAP1_MOCq5qRjsgd7XGag9ukjHiG2Se-pXvrghQ7hbB5ciwJK+A@mail.gmail.com>	<4E156252.2030308@earthspike.net>	<CAP1_MOBAFpYqJNKhgWzv3aYJeKbj+RJDQJCaQByVcrMVp_s8RA@mail.gmail.com>	<CAE=3qOZaqoa1zt1_q_gaYzME=DOJYqhW4i7afhrBYc4PxViq5g@mail.gmail.com>	<C06E47096A4E434AB7466C6309AC9FE5@dell9400>	<4E163411.50507@earthspike.net>
	<CAA8_NKCs_CJKEtHG9CpDZvK9vXevGtjEA18tNoEZ0yzKc2kmYg@mail.gmail.com>
Message-ID: <4E1DFD11.40805@earthspike.net>

Thanks, Gavin. That would be really useful.

John

On 08/07/11 19:51, Gavin Henry wrote:
> Hi John,
>
> We set this up for a customer using a Sangoma card and FusionPBX not
> long ago. I can dig out the config next week when back in the office.
> It's for ISDN30e but should help.
>
> Thanks.
>
> On 7/7/11, John<freeswitch at earthspike.net>  wrote:
>> Jan,
>>
>> I have tried every combination of unknown, national and international
>> with every combination of number length.  I have also confirmed that by
>> using wanpipemon to capture the D-channel messages.  When I went through
>> them with Wireshark and compared them with Q.931, everything was being
>> set correctly by the Sangoma card/drivers.  I am at the point now where
>> I need someone who has worked with BT ISDN switches to tell me what they
>> can accept.  It seems that this is a national secret over here, or
>> instead that they can accept almost anything and my lines have been
>> configured to override the outgoing calling number with the base
>> number.  Either way, the only way in which I think I am now going to get
>> a result is by booking a fault.  Your helpful comment about modern
>> switches being able to work this out also confirms that I am probably
>> not looking at a Q.931 formatting error on my part, but on a blockage in
>> the exchange.  I don't have a problem with the called number, for example.
>>
>> John
>>
>> On 07/07/11 23:09, Jan Berger wrote:
>>> I must admit I don't remember what CLIP and CLOP is -- and my Q.931
>>> experience is getting a bit rusty...
>>>
>>> Can you get a snoop of L3 out and in so I can see what you send and
>>> what the switch responds back?
>>>
>>> I prefer to use Called and Calling -- The numbers contain a few bits
>>> that tell what number this is. The most common are unknown, national
>>> and international. These bits must match the actual number you send.
>>> The 10 digit number is a unknown, the 11 digit is a national.
>>> International start with 00 nn
>>>
>>> On top of that you will face that modern switches are quite capable
>>> and able to configure whatever behaviour they want per line with
>>> regards to called/calling -- so you need to find out exactly what the
>>> switch expect on both called and caller for outgoing to behave as you
>>> want.
>>>
>>> /Jan
>>>
>>>>> On our PRI systems we are sending 10 digits (2071231234), and on the
>>>>> BRI system we are sending 11 digits (02071231234).
>>>>>



From yungwei at resolvity.com  Thu Jul 14 00:47:20 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 13 Jul 2011 16:47:20 -0400
Subject: [Freeswitch-users] multiple active grammars supported?
In-Reply-To: <33095823FD21DF429B481B5163264B7950CBC034F1@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950CBC034F1@VMBX102.ihostexchange.net>
Message-ID: <33095823FD21DF429B481B5163264B7950CBC03597@VMBX102.ihostexchange.net>

According to http://jira.freeswitch.org/browse/FS-2906, FS already supports multiple active grammars.
Now I'm wondering how to take advantages of it from a javascript program. 
Thanks.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei Chen
Sent: Wednesday, July 13, 2011 11:29 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] multiple active grammars supported?

Hi,

I tried the pizza demo shipped with freeswitch, and I am wondering if multiple active grammars are supported.

For example:
question: was this the pizza you had in mind?
response: yes/no (defined in grammar 1) Or 
          help   (defined in grammar 2; this case will be handled differently)

Thanks.

_______________________________________________
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 jcasale at activenetwerx.com  Thu Jul 14 00:47:45 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 13 Jul 2011 20:47:45 +0000
Subject: [Freeswitch-users] SPA-2102 ring issue
Message-ID: <assp.0175c3b9ba.92097A6A775D5147B1078E3F15430B928AA012@prato.activenetwerx.local>

I have an SPA-2102 a client uses that I can't replace with something good.
It appears to ring for several seconds after another user answers the incoming
call from some other sip phone. The SPA-3102 appears to have some params
that look applicable to tune this but those don't appear in this unit. I don't have
an easy way to setup a syslog server remotely, so short of driving out to this
place, does anyone know off hand what may cause this so I may give it a try?

thanks,
jlc


From jack at livecall.com  Thu Jul 14 00:45:44 2011
From: jack at livecall.com (Jack)
Date: Wed, 13 Jul 2011 13:45:44 -0700
Subject: [Freeswitch-users] rtmp_url folder
Message-ID: <4E1E03F8.7050506@livecall.com>

Freeswitch.html has a spot to set the variable rtmp_url.   What should 
be in the folder it points to?
Thanks,
  Jack



From daniel at danielknoll.de  Thu Jul 14 01:17:43 2011
From: daniel at danielknoll.de (Daniel Knoll)
Date: Wed, 13 Jul 2011 23:17:43 +0200
Subject: [Freeswitch-users] session.recordFile create empy wav in while
	loop
In-Reply-To: <CACREEeZFOKXHDTu-5YEsP1UUN64Tj+tLYP=hj95F67=gjZV3vA@mail.gmail.com>
References: <CACREEeaJ8On1Oo2K64PMisbJE+jASFC9tLTxXNZg71bS+3vVig@mail.gmail.com>
	<CAFiqYumaqz-tP9=bjSemCgjaxThfki9DNDFKkDEH7seL7ADHxQ@mail.gmail.com>
	<CACREEeZFOKXHDTu-5YEsP1UUN64Tj+tLYP=hj95F67=gjZV3vA@mail.gmail.com>
Message-ID: <699A1DD4-6D3B-441C-9B01-8DF1997F0FAE@danielknoll.de>

Any Ideas for my problem? 
The question is, why is the audio file empty if i re-run in while loop?

Thanks
Daniel


Am 13.07.2011 um 11:24 schrieb Daniel Knoll:

> Hi Steve, thanks for fast answer. The Channel is't hung up, because I
> stream in the same loop an Audio File
> 
> here is the code, the session.recordFile is executed in the second
> run, because, the filesize is changing for the file at first run.
> 
> do {
> 	var rd = session.uuid;
> 	session.streamFile("/dev/shm/freeswitch/sounds/conf-asrintro.wav", "");
> 	session.recordFile("/dev/shm/"+rd+".wav", "", "", 10, 500, 3);
>       	system("sox -v 1.7 /dev/shm/"+rd+".wav -r 16000 /dev/shm/"+rd+".flac");
> 	var fd = new File("|/usr/bin/googlespeech.sh /dev/shm/"+rd+".flac");
> 	fd.open("read");
> 	var dtmf = fd.read("8");
> 	dtmf = dtmf.replace(/[^0-9]/g, '');
> 	console_log("notice", "DTMF: "+ dtmf +"\n");
> 	roomExists = checkRoom(dtmf);
> 	if(!roomExists){
> 		retry_count++;
> 		session.streamFile("/dev/shm/freeswitch/sounds/conf-invalid.wav", "");
> 		var dtmf = "";
> 	}
> } while(!roomExists && retry_count < 5);
> 
> 
> I googled at this and found some guys he has also the same problem
> http://asterisk.voicemeup.com/viewtopic.php?p=62294&sid=8360229099e219df50bbb016cd43cbfd
> 
> Can you help me ?
> 
> Thanx Daniel
> 
> 
> 
> 
> 
> 2011/7/13, Steven Ayre <steveayre at gmail.com>:
>> Are you able to share the code?
>> 
>> Are you checking session.ready in the while loop's condition? Might the
>> channel be hung up when it reaches the 2nd recordFile?
>> 
>> -Steve
>> 
>> 
>> On 13 July 2011 10:01, Daniel Knoll <daniel at danielknoll.de> wrote:
>> 
>>> Hey Guys,
>>> 
>>> i have a strange Problem with session.recordFile in a while loop and i
>>> don't know how to solve it.
>>> The first run in the loop is absolutly ok, session.recordFile is
>>> creating a good clean wavefile.
>>> but in the second, third ..  run, every wav file has a size of 47724
>>> byte and no audio content.
>>> 
>>> Please help me, because it is an urgent case.
>>> 
>>> Thankx for getting help.
>>> Daniel
>>> 
>>> _______________________________________________
>>> 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
>>> 
>> 
> 
> 
> -- 
> Daniel Knoll
> Liberdastr.. 9
> 12047 Berlin
> 
> fon +49 (0)179 20 16 50 8
> mail daniel at danielknoll.de
> web www.danielknoll.de



From steveayre at gmail.com  Thu Jul 14 01:27:52 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 13 Jul 2011 22:27:52 +0100
Subject: [Freeswitch-users] session.recordFile create empy wav in while
	loop
In-Reply-To: <699A1DD4-6D3B-441C-9B01-8DF1997F0FAE@danielknoll.de>
References: <CACREEeaJ8On1Oo2K64PMisbJE+jASFC9tLTxXNZg71bS+3vVig@mail.gmail.com>
	<CAFiqYumaqz-tP9=bjSemCgjaxThfki9DNDFKkDEH7seL7ADHxQ@mail.gmail.com>
	<CACREEeZFOKXHDTu-5YEsP1UUN64Tj+tLYP=hj95F67=gjZV3vA@mail.gmail.com>
	<699A1DD4-6D3B-441C-9B01-8DF1997F0FAE@danielknoll.de>
Message-ID: <CAFiqYunwXoK2RbX26toMon0Swfsd51jx2ynjAS+CpyytAHnmiA@mail.gmail.com>

I'd need to set up a testcase to confirm it, but I think it's probably
that recordFile blocks and doesn't return until it's finished
recording at the end of the call. As a result on the next loop
iteration there'll be no call so nothing to record.

-Steve



On 13 July 2011 22:17, Daniel Knoll <daniel at danielknoll.de> wrote:
> Any Ideas for my problem?
> The question is, why is the audio file empty if i re-run in while loop?
>
> Thanks
> Daniel
>
>
> Am 13.07.2011 um 11:24 schrieb Daniel Knoll:
>
>> Hi Steve, thanks for fast answer. The Channel is't hung up, because I
>> stream in the same loop an Audio File
>>
>> here is the code, the session.recordFile is executed in the second
>> run, because, the filesize is changing for the file at first run.
>>
>> do {
>> ? ? ? var rd = session.uuid;
>> ? ? ? session.streamFile("/dev/shm/freeswitch/sounds/conf-asrintro.wav", "");
>> ? ? ? session.recordFile("/dev/shm/"+rd+".wav", "", "", 10, 500, 3);
>> ? ? ? ? ? ? ? system("sox -v 1.7 /dev/shm/"+rd+".wav -r 16000 /dev/shm/"+rd+".flac");
>> ? ? ? var fd = new File("|/usr/bin/googlespeech.sh /dev/shm/"+rd+".flac");
>> ? ? ? fd.open("read");
>> ? ? ? var dtmf = fd.read("8");
>> ? ? ? dtmf = dtmf.replace(/[^0-9]/g, '');
>> ? ? ? console_log("notice", "DTMF: "+ dtmf +"\n");
>> ? ? ? roomExists = checkRoom(dtmf);
>> ? ? ? if(!roomExists){
>> ? ? ? ? ? ? ? retry_count++;
>> ? ? ? ? ? ? ? session.streamFile("/dev/shm/freeswitch/sounds/conf-invalid.wav", "");
>> ? ? ? ? ? ? ? var dtmf = "";
>> ? ? ? }
>> } while(!roomExists && retry_count < 5);
>>
>>
>> I googled at this and found some guys he has also the same problem
>> http://asterisk.voicemeup.com/viewtopic.php?p=62294&sid=8360229099e219df50bbb016cd43cbfd
>>
>> Can you help me ?
>>
>> Thanx Daniel
>>
>>
>>
>>
>>
>> 2011/7/13, Steven Ayre <steveayre at gmail.com>:
>>> Are you able to share the code?
>>>
>>> Are you checking session.ready in the while loop's condition? Might the
>>> channel be hung up when it reaches the 2nd recordFile?
>>>
>>> -Steve
>>>
>>>
>>> On 13 July 2011 10:01, Daniel Knoll <daniel at danielknoll.de> wrote:
>>>
>>>> Hey Guys,
>>>>
>>>> i have a strange Problem with session.recordFile in a while loop and i
>>>> don't know how to solve it.
>>>> The first run in the loop is absolutly ok, session.recordFile is
>>>> creating a good clean wavefile.
>>>> but in the second, third .. ?run, every wav file has a size of 47724
>>>> byte and no audio content.
>>>>
>>>> Please help me, because it is an urgent case.
>>>>
>>>> Thankx for getting help.
>>>> Daniel
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>
>>
>>
>> --
>> Daniel Knoll
>> Liberdastr.. 9
>> 12047 Berlin
>>
>> fon +49 (0)179 20 16 50 8
>> mail daniel at danielknoll.de
>> web www.danielknoll.de
>
>
> _______________________________________________
> 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 daniel at danielknoll.de  Thu Jul 14 02:28:28 2011
From: daniel at danielknoll.de (Daniel Knoll)
Date: Thu, 14 Jul 2011 00:28:28 +0200
Subject: [Freeswitch-users] session.recordFile create empy wav in while
	loop
In-Reply-To: <CAFiqYunwXoK2RbX26toMon0Swfsd51jx2ynjAS+CpyytAHnmiA@mail.gmail.com>
References: <CACREEeaJ8On1Oo2K64PMisbJE+jASFC9tLTxXNZg71bS+3vVig@mail.gmail.com>
	<CAFiqYumaqz-tP9=bjSemCgjaxThfki9DNDFKkDEH7seL7ADHxQ@mail.gmail.com>
	<CACREEeZFOKXHDTu-5YEsP1UUN64Tj+tLYP=hj95F67=gjZV3vA@mail.gmail.com>
	<699A1DD4-6D3B-441C-9B01-8DF1997F0FAE@danielknoll.de>
	<CAFiqYunwXoK2RbX26toMon0Swfsd51jx2ynjAS+CpyytAHnmiA@mail.gmail.com>
Message-ID: <6AB0454C-9146-4B67-A5FF-DD4C72314F80@danielknoll.de>

Hi Steve,
Thanx for helping me,
the call is established and the loop can be re-run and re-run. but the new audio file that is created is empty.
i have the same issue if i use session.execute("record", "/dev/shm/"+rd+".wav 10 150");

unfortunately there is no function to stop the record.

If you can help me I would be very thankful

Daniel
 

Am 13.07.2011 um 23:27 schrieb Steven Ayre:

> I'd need to set up a testcase to confirm it, but I think it's probably
> that recordFile blocks and doesn't return until it's finished
> recording at the end of the call. As a result on the next loop
> iteration there'll be no call so nothing to record.
> 
> -Steve
> 
> 
> 
> On 13 July 2011 22:17, Daniel Knoll <daniel at danielknoll.de> wrote:
>> Any Ideas for my problem?
>> The question is, why is the audio file empty if i re-run in while loop?
>> 
>> Thanks
>> Daniel
>> 
>> 
>> Am 13.07.2011 um 11:24 schrieb Daniel Knoll:
>> 
>>> Hi Steve, thanks for fast answer. The Channel is't hung up, because I
>>> stream in the same loop an Audio File
>>> 
>>> here is the code, the session.recordFile is executed in the second
>>> run, because, the filesize is changing for the file at first run.
>>> 
>>> do {
>>>       var rd = session.uuid;
>>>       session.streamFile("/dev/shm/freeswitch/sounds/conf-asrintro.wav", "");
>>>       session.recordFile("/dev/shm/"+rd+".wav", "", "", 10, 500, 3);
>>>               system("sox -v 1.7 /dev/shm/"+rd+".wav -r 16000 /dev/shm/"+rd+".flac");
>>>       var fd = new File("|/usr/bin/googlespeech.sh /dev/shm/"+rd+".flac");
>>>       fd.open("read");
>>>       var dtmf = fd.read("8");
>>>       dtmf = dtmf.replace(/[^0-9]/g, '');
>>>       console_log("notice", "DTMF: "+ dtmf +"\n");
>>>       roomExists = checkRoom(dtmf);
>>>       if(!roomExists){
>>>               retry_count++;
>>>               session.streamFile("/dev/shm/freeswitch/sounds/conf-invalid.wav", "");
>>>               var dtmf = "";
>>>       }
>>> } while(!roomExists && retry_count < 5);
>>> 
>>> 
>>> I googled at this and found some guys he has also the same problem
>>> http://asterisk.voicemeup.com/viewtopic.php?p=62294&sid=8360229099e219df50bbb016cd43cbfd
>>> 
>>> Can you help me ?
>>> 
>>> Thanx Daniel
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 2011/7/13, Steven Ayre <steveayre at gmail.com>:
>>>> Are you able to share the code?
>>>> 
>>>> Are you checking session.ready in the while loop's condition? Might the
>>>> channel be hung up when it reaches the 2nd recordFile?
>>>> 
>>>> -Steve
>>>> 
>>>> 
>>>> On 13 July 2011 10:01, Daniel Knoll <daniel at danielknoll.de> wrote:
>>>> 
>>>>> Hey Guys,
>>>>> 
>>>>> i have a strange Problem with session.recordFile in a while loop and i
>>>>> don't know how to solve it.
>>>>> The first run in the loop is absolutly ok, session.recordFile is
>>>>> creating a good clean wavefile.
>>>>> but in the second, third ..  run, every wav file has a size of 47724
>>>>> byte and no audio content.
>>>>> 
>>>>> Please help me, because it is an urgent case.
>>>>> 
>>>>> Thankx for getting help.
>>>>> Daniel
>>>>> 
>>>>> _______________________________________________
>>>>> 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
>>>>> 
>>>> 
>>> 
>>> 
>>> --
>>> Daniel Knoll
>>> Liberdastr.. 9
>>> 12047 Berlin
>>> 
>>> fon +49 (0)179 20 16 50 8
>>> mail daniel at danielknoll.de
>>> web www.danielknoll.de
>> 
>> 
>> _______________________________________________
>> 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
>> 
> 
> _______________________________________________
> 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 david.ponzone at ipeva.fr  Thu Jul 14 02:49:27 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 14 Jul 2011 00:49:27 +0200
Subject: [Freeswitch-users] How to configure for FS playing voice prompt
	in case the called extension is in a call.
In-Reply-To: <CAOTmbMxJoQTkZ+UnCxtU_VHRHPUU0+sn2dFSa2Q2AMZgAJKXBw@mail.gmail.com>
References: <CAOTmbMzMVtRjUvTj5UgHH7MG=XyHSWRvw7QT6G-tiLFd57H1Vw@mail.gmail.com>
	<CAFiqYuncFULQn=BnWDh2s0m6VhAX3aqN-zPbFaMD19cowfOERA@mail.gmail.com>
	<CAOTmbMwtVZJd_44rs7QH1GATSPygyfMJq2Ao-kgxw1x-L_vkVg@mail.gmail.com>
	<5F28A4A5-726E-44EC-97CA-E00920F95114@ipeva.fr>
	<CAOTmbMxJoQTkZ+UnCxtU_VHRHPUU0+sn2dFSa2Q2AMZgAJKXBw@mail.gmail.com>
Message-ID: <BAA84E25-D711-4D13-A457-6A5DEC888AA9@ipeva.fr>

Charles,

you need to read the wiki with care :)

failure_causes
Controls which failure causes will be considered as a failure to the bridge(s). This will change the values for which continue_on_fail will fail by default unless continue_on_fail is set to true.

This means if you set continue_on_fail to true, failure_causes will be ignored.

Remove:
    <action application="set" data="continue_on_fail=true"/>

Or remove:
    <action application="set" data="failure_causes=USER_BUSY"/>
and set:
    <action application="set" data="continue_on_fail=USER_BUSY"/>


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

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

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




Le 13/07/2011 ? 12:06, fieldpeak a ?crit :

> My phone sends back '486 Busy Here'.
> 
> 2011/7/12 David Ponzone <david.ponzone at ipeva.fr>
> Check what your phone sends back when busy.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 12/07/2011 ? 10:31, fieldpeak a ?crit :
> 
>> Hi Steve,
>> 
>> Thanks for your kindly help, it works with below dial plan...
>> However, for both user busy or no answer, it played the voice prompt...
>> i would like it only prompts in case user busy but not no answer, although i configured the "failure_causes=USER_BUSY" as below dialplan, it looks not help...attached is a log...
>> 
>>     <extension name="Local_Extension">
>>       <condition field="destination_number" expression="^(1\d{3})$">
>>     <action application="set" data="dialed_extension=$1"/>
>>     <action application="export" data="dialed_extension=$1"/>
>>     <action application="set" data="ringback=${us-ring}"/>
>>     <action application="set" data="transfer_ringback=$${hold_music}"/>
>>     <action application="set" data="call_timeout=30"/>
>>     
>>     <action application="set" data="failure_causes=USER_BUSY"/>
>>     
>>     <action application="set" data="hangup_after_bridge=true"/>
>>     
>>     <!-- to enable voice mail, set continue_on_fail=true below -->
>>     <action application="set" data="continue_on_fail=true"/>
>>     <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>     <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>     <action application="bridge" data="sofia/internal/${dialed_extension}%${domain_name}"/>
>>     <action application="answer"/>
>>     
>>     <action application="sleep" data="1000"/>
>>     <!-- play a voice prompt e.g. 'the extension you dialed is busy now, please dial the other exsention' to replacing busy tone when the called extension is in a call... -->    
>>     <action application="play_and_get_digits" data="2 5 3 4000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav other_ext_var \d+"/>
>>     <action application="transfer" data="${other_ext_var} XML default"/>
>>         
>>       </condition>
>>     </extension>
>> 
>> Regards,
>> Charles
>> 
>> 
>> 2011/7/11 Steven Ayre <steveayre at gmail.com>
>> There are two ways....
>> 
>> Most phones will return a USER_BUSY release code if you bridge to a phone that's busy. If the phone does then you can continue_on_fail so that you (pre-)answer the call and play that message after the bridge. Something like:
>> <extension name="Local_Extension">
>>     <condition field="destination_number" expression="^(1\d{3})$">
>>         <action application="set" data="dialed_extension=$1"/>
>>         <action application="export" data="dialed_extension=$1"/>
>>         <action application="set" data="call_timeout=30"/>
>>         <action application="set" data="hangup_after_bridge=true"/> <!-- Hangup on successful call -->
>>         <action application="set" data="continue_on_fail=true"/> <!-- Don't hangup on failed call, continue to IVR prompt -->
>> 
>>         <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>         <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>         <action application="bridge" data="user/${dialed_extension}%${domain_name}"/>
>>         <!-- Put the IVR prompt here, see http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits -->
>>     </condition>
>> </extension>
>> The 2nd option is that you use Limit to determine if another person is already on a call to the phone. That only really works if you run the the only servers that call the phone since it requires tracking all calls to the phone. It will sometimes work better though - especially if the phone has multiple lines. http://wiki.freeswitch.org/wiki/Limit
>> 
>> -Steve
>> 
>> 
>> On 11 July 2011 03:25, fieldpeak <fieldpeak at gmail.com> wrote:
>> Hi Gurus,
>>     
>> Could anyone advise how to realize have FS to play a voice prompt e.g. 'the extension you dialed is busy now, please dial the other exsention' to replacing busy tone when the called extension is in a call...
>> 
>> the details is below,
>> 
>> When calling to FS, the FS will play IVR "welcome to call us, please input the extension number, for operator please press 0",
>> then the caller input the extension number, in case the extension is in a call, the caller will hear busy tone...
>> it needs the system play "the extension you dailed is busy, please dial the other extension", currently my dial plan as following, could anyone advise how to change based on below dial plan or any other new dail plan can realize it...
>> 
>> Thanks a lot!
>> 
>>     <extension name="Local_Extension">
>>       <condition field="destination_number" expression="^(1\d{3})$">
>>     <action application="set" data="dialed_extension=$1"/>
>>     <action application="export" data="dialed_extension=$1"/>
>>     <action application="set" data="ringback=${us-ring}"/>
>>     <action application="set" data="transfer_ringback=$${hold_music}"/>
>>     <action application="set" data="call_timeout=30"/>
>>     <action application="set" data="hangup_after_bridge=true"/>
>>     
>>     <!-- to enable voice mail, set continue_on_fail=true below -->
>>     <action application="set" data="continue_on_fail=false"/>
>>     <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>     <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>     <action application="bridge" data="sofia/internal/${dialed_extension}%${domain_name}"/>
>>     <action application="answer"/>
>>     
>>     <action application="sleep" data="1000"/>
>>     <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
>>       </condition>
>>     </extension>
>>     
>>     
>>     <extension name="IVR" >
>>        <condition field="destination_number" expression="^888$">
>>            <action application="answer"/>
>>            <action application="sleep" data="2000"/>
>>            <action application="ivr" data="IVR"/>
>>        </condition>
>>     </extension>
>>     
>>         <!--  -->
>>     <menu name="IVR"
>>         greet-long="C:/VSWITCH/recordings/greeting_tts.wav"
>>         greet-short="C:/VSWITCH/recordings/greeting_tts.wav"
>>         invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>         exit-sound="voicemail/vm-goodbye.wav"
>>         confirm-macro=""
>>         confirm-key=""
>>         tts-engine="flite"
>>         tts-voice="rms"
>>         confirm-attempts="3"
>>         timeout="10000"
>>         inter-digit-timeout="2000"
>>         max-failures="3"
>>         max-timeouts="3"
>>         digit-len="5">
>>         <entry action="menu-exec-app" digits="/(^\*\d{3,5}$|^\d{3,5}$)/" param="transfer $1 XML default"/>
>>     </menu>
>>     
>> Regards,
>> Charles
>> 
>> _______________________________________________
>> 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
>> 
>> 
>> 
>> _______________________________________________
>> 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
>> 
>> 
>> <log_no_answer.txt>_______________________________________________
>> 
>> 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
> 
> 
> _______________________________________________
> 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
> 
> 
> _______________________________________________
> 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 rzhang at gosilverplus.com  Thu Jul 14 05:24:26 2011
From: rzhang at gosilverplus.com (ran zhang)
Date: Wed, 13 Jul 2011 18:24:26 -0700
Subject: [Freeswitch-users] please help!!! how to limit each FS registered
	user's active calls
Message-ID: <4E1E454A.8040901@gosilverplus.com>

hi all:

            I'm trying to limit each registered user's active calls, so 
if that particular registered user pass the limit, FS stops sending any 
new incoming call to it and it can't make any new calls.

            I looked into 'limit' api, the counter increment everytime a 
call FS receives, regardless of which user it is for, and counter doesnt 
reset itself after the call is finished.






From fieldpeak at gmail.com  Thu Jul 14 06:14:52 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Thu, 14 Jul 2011 10:14:52 +0800
Subject: [Freeswitch-users] How to configure for FS playing voice prompt
 in case the called extension is in a call.
In-Reply-To: <BAA84E25-D711-4D13-A457-6A5DEC888AA9@ipeva.fr>
References: <CAOTmbMzMVtRjUvTj5UgHH7MG=XyHSWRvw7QT6G-tiLFd57H1Vw@mail.gmail.com>
	<CAFiqYuncFULQn=BnWDh2s0m6VhAX3aqN-zPbFaMD19cowfOERA@mail.gmail.com>
	<CAOTmbMwtVZJd_44rs7QH1GATSPygyfMJq2Ao-kgxw1x-L_vkVg@mail.gmail.com>
	<5F28A4A5-726E-44EC-97CA-E00920F95114@ipeva.fr>
	<CAOTmbMxJoQTkZ+UnCxtU_VHRHPUU0+sn2dFSa2Q2AMZgAJKXBw@mail.gmail.com>
	<BAA84E25-D711-4D13-A457-6A5DEC888AA9@ipeva.fr>
Message-ID: <CAOTmbMyJY24gwE-KOPbbcz4B+r5+MBaKVfV9DgNYgXWaVYBQuQ@mail.gmail.com>

Hi David,

It works well with your kindly advise... very appreciated for help! sure, i
need more carefully read the wiki :)

Regards,
Charles

2011/7/14 David Ponzone <david.ponzone at ipeva.fr>

> Charles,
>
> you need to read the wiki with care :)
>
> failure_causes <http://wiki.freeswitch.org/wiki/Variable_failure_causes>
> Controls which failure causes will be considered as a failure to the
> bridge(s). This will change the values for which continue_on_fail<http://wiki.freeswitch.org/wiki/Variable_continue_on_fail> will
> fail by default unless continue_on_fail<http://wiki.freeswitch.org/wiki/Variable_continue_on_fail> is
> set to true.
>
> This means if you set continue_on_fail to true, failure_causes will be
> ignored.
>
> Remove:
>     <action application="set" data="continue_on_fail=true"/>
>
> Or remove:
>     <action application="set" data="failure_causes=USER_BUSY"/>
> and set:
>     <action application="set" data="continue_on_fail=USER_BUSY"/>
>
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 13/07/2011 ? 12:06, fieldpeak a ?crit :
>
> My phone sends back '486 Busy Here'.
>
> 2011/7/12 David Ponzone <david.ponzone at ipeva.fr>
>
>> Check what your phone sends back when busy.
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 12/07/2011 ? 10:31, fieldpeak a ?crit :
>>
>> Hi Steve,
>>
>> Thanks for your kindly help, it works with below dial plan...
>> However, for both user busy or no answer, it played the voice prompt...
>> i would like it only prompts in case user busy but not no answer, although
>> i configured the "failure_causes=USER_BUSY" as below dialplan, it looks not
>> help...attached is a log...
>>
>>     <extension name="Local_Extension">
>>       <condition field="destination_number" expression="^(1\d{3})$">
>>     <action application="set" data="dialed_extension=$1"/>
>>     <action application="export" data="dialed_extension=$1"/>
>>     <action application="set" data="ringback=${us-ring}"/>
>>     <action application="set" data="transfer_ringback=$${hold_music}"/>
>>     <action application="set" data="call_timeout=30"/>
>>
>>     <action application="set" data="failure_causes=USER_BUSY"/>
>>
>>     <action application="set" data="hangup_after_bridge=true"/>
>>
>>     <!-- to enable voice mail, set continue_on_fail=true below -->
>>     <action application="set" data="continue_on_fail=true"/>
>>     <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>     <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>     <action application="bridge"
>> data="sofia/internal/${dialed_extension}%${domain_name}"/>
>>     <action application="answer"/>
>>
>>     <action application="sleep" data="1000"/>
>>     <!-- play a voice prompt e.g. 'the extension you dialed is busy now,
>> please dial the other exsention' to replacing busy tone when the called
>> extension is in a call... -->
>>     <action application="play_and_get_digits" data="2 5 3 4000 #
>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav
>> other_ext_var \d+"/>
>>     <action application="transfer" data="${other_ext_var} XML default"/>
>>
>>       </condition>
>>     </extension>
>>
>> Regards,
>> Charles
>>
>>
>> 2011/7/11 Steven Ayre <steveayre at gmail.com>
>>
>>> There are two ways....
>>>
>>> Most phones will return a USER_BUSY release code if you bridge to a phone
>>> that's busy. If the phone does then you can continue_on_fail so that you
>>> (pre-)answer the call and play that message after the bridge. Something
>>> like:
>>>
>>> <extension name="Local_Extension">
>>>     <condition field="destination_number" expression="^(1\d{3})$">
>>>         <action application="set" data="dialed_extension=$1"/>
>>>         <action application="export" data="dialed_extension=$1"/>
>>>         <action application="set" data="call_timeout=30"/>
>>>         <action application="set" data="hangup_after_bridge=true"/> <!--
>>> Hangup on successful call -->
>>>         <action application="set" data="continue_on_fail=true"/> <!--
>>> Don't hangup on failed call, continue to IVR prompt -->
>>>
>>>         <action application="hash"
>>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>>         <action application="hash"
>>> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>>         <action application="bridge"
>>> data="user/${dialed_extension}%${domain_name}"/>
>>>         <!-- Put the IVR prompt here, see
>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits-->
>>>     </condition>
>>> </extension>
>>>
>>> The 2nd option is that you use Limit to determine if another person is
>>> already on a call to the phone. That only really works if you run the the
>>> only servers that call the phone since it requires tracking all calls to the
>>> phone. It will sometimes work better though - especially if the phone has
>>> multiple lines. http://wiki.freeswitch.org/wiki/Limit
>>>
>>> -Steve
>>>
>>>
>>> On 11 July 2011 03:25, fieldpeak <fieldpeak at gmail.com> wrote:
>>>
>>>> Hi Gurus,
>>>>
>>>> Could anyone advise how to realize have FS to play a voice prompt e.g.
>>>> 'the extension you dialed is busy now, please dial the other exsention' to
>>>> replacing busy tone when the called extension is in a call...
>>>>
>>>> the details is below,
>>>>
>>>> When calling to FS, the FS will play IVR "welcome to call us, please
>>>> input the extension number, for operator please press 0",
>>>> then the caller input the extension number, in case the extension is in
>>>> a call, the caller will hear busy tone...
>>>> it needs the system play "the extension you dailed is busy, please dial
>>>> the other extension", currently my dial plan as following, could anyone
>>>> advise how to change based on below dial plan or any other new dail plan can
>>>> realize it...
>>>>
>>>> Thanks a lot!
>>>>
>>>>     <extension name="Local_Extension">
>>>>       <condition field="destination_number" expression="^(1\d{3})$">
>>>>     <action application="set" data="dialed_extension=$1"/>
>>>>     <action application="export" data="dialed_extension=$1"/>
>>>>     <action application="set" data="ringback=${us-ring}"/>
>>>>     <action application="set" data="transfer_ringback=$${hold_music}"/>
>>>>     <action application="set" data="call_timeout=30"/>
>>>>     <action application="set" data="hangup_after_bridge=true"/>
>>>>
>>>>     <!-- to enable voice mail, set continue_on_fail=true below -->
>>>>     <action application="set" data="continue_on_fail=false"/>
>>>>     <action application="hash"
>>>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>>>     <action application="hash"
>>>> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>>>     <action application="bridge"
>>>> data="sofia/internal/${dialed_extension}%${domain_name}"/>
>>>>     <action application="answer"/>
>>>>
>>>>     <action application="sleep" data="1000"/>
>>>>     <action application="bridge" data="loopback/app=voicemail:default
>>>> ${domain_name} ${dialed_extension}"/>
>>>>       </condition>
>>>>     </extension>
>>>>
>>>>
>>>>     <extension name="IVR" >
>>>>        <condition field="destination_number" expression="^888$">
>>>>            <action application="answer"/>
>>>>            <action application="sleep" data="2000"/>
>>>>            <action application="ivr" data="IVR"/>
>>>>        </condition>
>>>>     </extension>
>>>>
>>>>         <!--  -->
>>>>     <menu name="IVR"
>>>>         greet-long="C:/VSWITCH/recordings/greeting_tts.wav"
>>>>         greet-short="C:/VSWITCH/recordings/greeting_tts.wav"
>>>>         invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>>>         exit-sound="voicemail/vm-goodbye.wav"
>>>>         confirm-macro=""
>>>>         confirm-key=""
>>>>         tts-engine="flite"
>>>>         tts-voice="rms"
>>>>         confirm-attempts="3"
>>>>         timeout="10000"
>>>>         inter-digit-timeout="2000"
>>>>         max-failures="3"
>>>>         max-timeouts="3"
>>>>         digit-len="5">
>>>>         <entry action="menu-exec-app" digits="/(^\*\d{3,5}$|^\d{3,5}$)/"
>>>> param="transfer $1 XML default"/>
>>>>     </menu>
>>>>
>>>> Regards,
>>>> Charles
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>> <log_no_answer.txt>_______________________________________________
>>
>> 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
>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
> _______________________________________________
> 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
>
>
>
> _______________________________________________
> 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 msc at freeswitch.org  Thu Jul 14 10:22:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Jul 2011 23:22:58 -0700
Subject: [Freeswitch-users] please help!!! how to limit each FS
 registered user's active calls
In-Reply-To: <4E1E454A.8040901@gosilverplus.com>
References: <4E1E454A.8040901@gosilverplus.com>
Message-ID: <CAKzWOxVtNTEptQn6n-1POqiF9ntpLV-SsZeP3oufnSOOpFPoLA@mail.gmail.com>

On Wed, Jul 13, 2011 at 6:24 PM, ran zhang <rzhang at gosilverplus.com> wrote:

> hi all:
>
>            I'm trying to limit each registered user's active calls, so
> if that particular registered user pass the limit, FS stops sending any
> new incoming call to it and it can't make any new calls.
>
>            I looked into 'limit' api, the counter increment everytime a
> call FS receives, regardless of which user it is for, and counter doesnt
> reset itself after the call is finished.
>

I presume you mean the limit "dialplan app" and not API. In any case, if you
have the default configs then locate the Local_Extension in
conf/dialplan/default.xml. Add this line right after the condition:

<action application="limit" data="hash ${domain} $1 1 handle_over_limit XML
over_limit_actions"/>

Then add this new file as "limits.xml" in conf/dialplan/ :

<include>


  <context name="over_limit_actions">


    <extension name="oops, too many calls for this one">


      <condition field="destination_number" expression="handle_over_limit">


        <action application="answer"/>


        <action application="playback" data="ivr/ivr-no_no_no.wav"/>


        <action application="playback" data="ivr/ivr-no_no_no.wav"/>


        <action application="playback" data="ivr/ivr-no_no_no.wav"/>


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


      </condition>


    </extension>


  </context>


</include>

Save, then do F6 or reloadxml. Now when you call a local extension it won't
allow more than one call. Note that you can change the value in the limit's
data argument. For example, this would cause a limit of 4 concurrent calls,
sending the 5th call into "oops, too many calls" extension:

<action application="limit" data="hash ${domain} $1 4 handle_over_limit XML
over_limit_actions"/>

The limit application handles everything for you. Note that this only covers
calls made *to* the extension. If you want to handle calls made *from* the
extension you will need to add another limit app to the dialplan. Here's an
example; put it right after the "global" extension in default.xml:

    <!-- set outound caller limit -->


    <extension name="set outbound limit" continue="true">


      <condition field="destination_number" expression="^1?\d{7}"
break="on-false"/>

      <condition field="caller_id_number" expression="^(10[01][0-9])">


        <action application="limit" data="hash ${domain} $1
4 handle_over_limit XML over_limit_actions"/>
        <action application="log" data="INFO Added limit for caller $1"/>


      </condition>


    </extension>



Again, save, press F6 or do reloadxml. Now when this person makes a call to
a 7-digit number or does 1+ 7 or more digits, it will add to his limit
totals. However, if he just calls another 4-digit extension on the system
then it won't add to his limit. (Adjust this as you see fit.)

Let me know if this works for you (it worked for me in my little lab
setting) and I will throw it up on the wiki and maybe add it to the
cookbook.

-MC
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From boris at tagnet.ru  Thu Jul 14 10:55:45 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 14 Jul 2011 12:55:45 +0600
Subject: [Freeswitch-users] DISA, originate & bridge
Message-ID: <4E1E92F1.7070705@tagnet.ru>

Hello!

     I'm trying to modify DISA javascript example to use 'originate' 
instead of 'transfer'.  So the problem is - after originated call is 
answered I see no voice. Thinking this is becase call is not bridged to 
the current session. How may I do this? My peace of code:

osession = new Session("{origination_caller_id_number=7343523000
1,v_ats_dstport=50002}sofia/epbx/89979049898XXX at X.X.X.42:5061");
                 while(osession.ready()) {
                     // Don't know what to write here
                 }

-- 
Regards,
Boris




From jack at livecall.com  Thu Jul 14 12:28:48 2011
From: jack at livecall.com (Jack)
Date: Thu, 14 Jul 2011 01:28:48 -0700
Subject: [Freeswitch-users] Call drops after 30 seconds
In-Reply-To: <BANLkTimBBN8RpAcEiJM6Rj6tegZSeydmUg@mail.gmail.com>
References: <71F1700A-7641-43E5-A7C6-F25A0DC2EBC5@mralston.com>
	<BANLkTikoP+hvtbzDn6fm_O6MmSN--x7TyA@mail.gmail.com>
	<C3BE3091-1685-42D1-ACD1-223B370607B7@mralston.com>
	<BANLkTi=4e1t607G0xCRygHGotxQrsXcnzw@mail.gmail.com>
	<28830F22-9654-45D2-B00A-1E69E32A42F9@mralston.com>
	<BANLkTimBBN8RpAcEiJM6Rj6tegZSeydmUg@mail.gmail.com>
Message-ID: <4E1EA8C0.3030208@livecall.com>

Michael,
  We were having the 30 second drop problem also.  We are not behind a NAT.
We changed both values to $${local_ip_v4}  and it worked great.
Thanks,
Jack

On 6/23/2011 11:32 AM, Michael Collins wrote:
> Matthew,
>
> Did you get resolution on this one? I think you can go to the sip 
> profile and work with these lines:
> <param name="ext-rtp-ip" value="auto-nat"/>
> <param name="ext-sip-ip" value="auto-nat"/>
>
> You can put an IP address in those and try it out. Let us know what 
> happens.
> -MC
>
> On Wed, Jun 22, 2011 at 8:26 AM, Matthew Ralston 
> <freeswitch at mralston.com <mailto:freeswitch at mralston.com>> wrote:
>
>     Hi Chris,
>
>     The FreeSWITCH box does indeed have a private IP address.
>     The Cisco Linksys WAG160N router in front of it has UPnP enabled,
>     not that I particularly trust it.
>
>     So maybe it's worth hard coding the public IP into the Sofia config.
>
>     Where is the best place to put the public IP? The internal.xml and
>     external.xml files both refer to $${local_ip_v4} although I don't
>     see where this comes from. Does FreeSWITCH figure this out itself
>     in some way?
>
>     As I have internal SIP phones, an external SIP provider and
>     (whilst debugging) some external SIP phones (which use the
>     internal profile!!). I'm concerned that if I hard code the public
>     IP address in to the wrong place it will cause problems for the
>     internal SIP phones.
>
>
>     Kind regards,
>
>     Matthew Ralston
>     Web Developer & IT Consultant
>
>     matt at mralston.co.uk <mailto:matt at mralston.co.uk>
>     www.mralston.com <http://www.mralston.com>
>
>     On 22 Jun 2011, at 16:11, Chris Chen wrote:
>
>>     Hi Matthew, continue my last reply here
>>
>>     2) Is your FS server using private IP address?  you have to setup
>>     your FS external SIP/RTP IP address to the proper public IP
>>     address, by either using UPNP enabled router, STUN, or hardcoded
>>     public IP address in sofia profiles.
>>
>>     Please check that.
>>
>>     Thanks,
>>     Chris
>>
>>
>>     On Wed, Jun 22, 2011 at 10:49 AM, Matthew Ralston
>>     <freeswitch at mralston.com <mailto:freeswitch at mralston.com>> wrote:
>>
>>         Hi Chris,
>>
>>         Thanks for the quick reply!
>>
>>         The FreeSWITCH box is in the DMZ of a Cisco Linksys WAG160N.
>>         Best I can tell, the DMZ is doing its job and allowing all
>>         ports through, inbound and outbound, TCP & UDP.
>>
>>         The external SIP phones are a Cisco SPA504G and Bria on the
>>         iPhone. These are behind behind a Cisco ASA5505, which has
>>         policy inspection for SIP switched on, i.e. ALG. I have also
>>         tested with Bria going over 3G (so it's not behind the Cisco
>>         ASA) and had the same problem.
>>
>>         The other scenario we have is some Yealink T20P SIP phones on
>>         the same LAN segment as the FreeSWITCH box. These can make
>>         A-leg only calls into FreeSWITCH (like calling voicemail) and
>>         also calls to other internal SIP phones fine. However when
>>         they make an outbound call the problem happens again. In this
>>         case the b-leg of the calls are sent to an external SIP
>>         provider and get cut after 30 seconds. Incidentally, we also
>>         use the same SIP provider from an Asterisk box in our data
>>         centre and that doesn't have a problem, so I believe the SIP
>>         provider is fine.
>>
>>         Kind regards,
>>
>>
>>         Matthew Ralston
>>         Web Developer & IT Consultant
>>
>>         matt at mralston.co.uk <mailto:matt at mralston.co.uk>
>>         www.mralston.com <http://www.mralston.com/>
>>
>>         On 22 Jun 2011, at 14:21, Chris Chen wrote:
>>
>>>         Hi Matthew, this is typical behavior for the setup of SIP
>>>         behind NAT.
>>>             1) Please provide the exact setup of remote SIP phones,
>>>         what's the router model, does it have SIP ALG enabled, what
>>>         kind  of SIP phones
>>>             2)
>>>
>>>
>>>         On Wed, Jun 22, 2011 at 8:38 AM, Matthew Ralston
>>>         <freeswitch at mralston.com <mailto:freeswitch at mralston.com>>
>>>         wrote:
>>>
>>>             Hi,
>>>
>>>             I'm having a problem at the moment with calls being
>>>             successfully set up, with two-way audio, being
>>>             terminated by FreeSWITCH after 30 seconds.
>>>
>>>             Internal calls (i.e. between SIP phones on the same LAN
>>>             segment as the FreeSWITCH box) work flawlessly.
>>>
>>>             The problem arises when at least one of the handsets is
>>>             located elsewhere on the Internet. This behaviour is
>>>             exhibited under the following circumstances:
>>>
>>>             - A-leg only call, e.g. to voicemail when the handset is
>>>             at another location on the Internet
>>>             - A-leg-B-leg call if one or both of the handsets are at
>>>             another location on the Internet
>>>             - Inbound calls from our external SIP provider
>>>             - Outbound calls to our external SIP provider
>>>
>>>             So it is obvious that the problem is related to the SIP
>>>             going via the Internet, but I'm having trouble
>>>             understanding why.
>>>
>>>             Whilst debugging this problem I have placed the
>>>             FreeSWITCH box is in the DMZ on our router, so there
>>>             should not be any ports blocked. The FreeSWITCH box
>>>             itself is not running a software firewall.
>>>
>>>             The calls themselves are absolutely fine for the first
>>>             30 seconds - each party can hear the other talking fine.
>>>
>>>             The fact that the call is consistently dropped after 30
>>>             seconds (give or take a second or two for PDD) suggests
>>>             that some timeout is being triggered.
>>>
>>>             When FreeSWITCH terminates the call, the following is
>>>             logged to the console:
>>>
>>>             2011-06-22 13:33:50.514941 [DEBUG] sofia.c:4787 Channel
>>>             sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed> entering
>>>             state [terminating][0]
>>>             2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2641
>>>             (sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed>)
>>>             Callstate Change ACTIVE -> HANGUP
>>>             2011-06-22 13:33:50.514941 [NOTICE] sofia.c:5508 Hangup
>>>             sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed>
>>>             [CS_EXECUTE] [NORMAL_UNSPECIFIED]
>>>             2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2657
>>>             Send signal sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed> [KILL]
>>>             2011-06-22 13:33:50.514941 [DEBUG]
>>>             switch_core_session.c:1118 Send signal
>>>             sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed> [BREAK]
>>>             2011-06-22 13:33:50.534966 [DEBUG]
>>>             switch_ivr_play_say.c:1649 done playing file
>>>             2011-06-22 13:33:50.625988 [DEBUG]
>>>             switch_ivr_play_say.c:244 Handle
>>>             play-file:[voicemail/vm-press.wav] (en:en)
>>>             2011-06-22 13:33:50.727027 [DEBUG]
>>>             switch_core_session.c:2063
>>>             sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed> skip
>>>             receive message [APPLICATION_EXEC_COMPLETE] (channel is
>>>             hungup already)
>>>             2011-06-22 13:33:50.727027 [DEBUG]
>>>             switch_core_state_machine.c:371
>>>             (sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed>) State
>>>             EXECUTE going to sleep
>>>             2011-06-22 13:33:50.727027 [DEBUG]
>>>             switch_core_state_machine.c:325
>>>             (sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed>) Running
>>>             State Change CS_HANGUP
>>>             2011-06-22 13:33:50.727027 [DEBUG]
>>>             switch_core_state_machine.c:565
>>>             (sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed>) State HANGUP
>>>             2011-06-22 13:33:50.727027 [DEBUG] mod_sofia.c:458
>>>             Channel sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed> hanging
>>>             up, cause: NORMAL_UNSPECIFIED
>>>             2011-06-22 13:33:50.727027 [DEBUG]
>>>             switch_core_state_machine.c:46
>>>             sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed> Standard
>>>             HANGUP, cause: NORMAL_UNSPECIFIED
>>>             2011-06-22 13:33:50.727027 [DEBUG]
>>>             switch_core_state_machine.c:565
>>>             (sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed>) State
>>>             HANGUP going to sleep
>>>             2011-06-22 13:33:50.727027 [DEBUG]
>>>             switch_core_state_machine.c:356
>>>             (sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed>) State
>>>             Change CS_HANGUP -> CS_REPORTING
>>>             2011-06-22 13:33:50.727027 [DEBUG]
>>>             switch_core_session.c:1118 Send signal
>>>             sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed> [BREAK]
>>>             2011-06-22 13:33:50.727027 [DEBUG]
>>>             switch_core_state_machine.c:325
>>>             (sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed>) Running
>>>             State Change CS_REPORTING
>>>             2011-06-22 13:33:50.727027 [DEBUG]
>>>             switch_core_state_machine.c:625
>>>             (sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed>) State
>>>             REPORTING
>>>             2011-06-22 13:33:50.740064 [DEBUG]
>>>             switch_core_state_machine.c:53
>>>             sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed> Standard
>>>             REPORTING, cause: NORMAL_UNSPECIFIED
>>>             2011-06-22 13:33:50.740064 [DEBUG]
>>>             switch_core_state_machine.c:625
>>>             (sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed>) State
>>>             REPORTING going to sleep
>>>             2011-06-22 13:33:50.740064 [DEBUG]
>>>             switch_core_state_machine.c:350
>>>             (sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed>) State
>>>             Change CS_REPORTING -> CS_DESTROY
>>>             2011-06-22 13:33:50.740064 [DEBUG]
>>>             switch_core_session.c:1118 Send signal
>>>             sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed> [BREAK]
>>>             2011-06-22 13:33:50.740064 [DEBUG]
>>>             switch_core_session.c:1290 Session 5
>>>             (sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed>) Locked,
>>>             Waiting on external entities
>>>             2011-06-22 13:33:50.740064 [NOTICE]
>>>             switch_core_session.c:1308 Session 5
>>>             (sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed>) Ended
>>>             2011-06-22 13:33:50.740064 [NOTICE]
>>>             switch_core_session.c:1310 Close Channel
>>>             sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed> [CS_DESTROY]
>>>             2011-06-22 13:33:50.740064 [DEBUG]
>>>             switch_core_state_machine.c:454
>>>             (sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed>)
>>>             Callstate Change HANGUP -> DOWN
>>>             2011-06-22 13:33:50.740064 [DEBUG]
>>>             switch_core_state_machine.c:457
>>>             (sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed>) Running
>>>             State Change CS_DESTROY
>>>             2011-06-22 13:33:50.740064 [DEBUG]
>>>             switch_core_state_machine.c:467
>>>             (sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed>) State
>>>             DESTROY
>>>             2011-06-22 13:33:50.740064 [DEBUG] mod_sofia.c:363
>>>             sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed> SOFIA DESTROY
>>>             2011-06-22 13:33:50.780056 [DEBUG] switch_nat.c:570
>>>             unmapped public port 31484 protocol UDP to localport 31484
>>>             2011-06-22 13:33:50.840070 [DEBUG] switch_nat.c:570
>>>             unmapped public port 31485 protocol UDP to localport 31485
>>>             2011-06-22 13:33:50.840070 [DEBUG]
>>>             switch_core_state_machine.c:60
>>>             sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed> Standard
>>>             DESTROY
>>>             2011-06-22 13:33:50.840070 [DEBUG]
>>>             switch_core_state_machine.c:467
>>>             (sofia/internal/1006 at public.ip.removed
>>>             <mailto:sofia/internal/1006 at public.ip.removed>) State
>>>             DESTROY going to sleep
>>>
>>>             The above example was from an externally situated SIP
>>>             phone ringing voicemail (4000) on FreeSWITCH.
>>>
>>>             I have experimented changing various timers and timeouts
>>>             in the config of FreeSWITCH (one at a time, being
>>>             careful to put them back afterwards!) but been unable to
>>>             resolve the issue.
>>>
>>>             Incidentally, we have no long term intention of running
>>>             off-site SIP phones with the PBX and I'm hoping not to
>>>             have to leave it in the DMZ either, it's just like that
>>>             for debugging. What is a real issue is the calls to our
>>>             external SIP provider (i.e. outbound calls) being dropped.
>>>
>>>             Any suggestions would be greatly appreciated.
>>>
>>>             Thanks,
>>>
>>>             Matthew Ralston
>>>             Web Developer & IT Consultant
>>>
>>>             matt at mralston.co.uk <mailto:matt at mralston.co.uk>
>>>             www.mralston.com <http://www.mralston.com/>
>>>
>>>
>>>             _______________________________________________
>>>             Join us at ClueCon 2011, Aug 9-11, Chicago
>>>             http://www.cluecon.com <http://www.cluecon.com/>
>>>             877-7-4ACLUE
>>>
>>>             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/>
>>>
>>>
>>>         _______________________________________________
>>>         Join us at ClueCon 2011, Aug 9-11, Chicago
>>>         http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>>
>>>         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/>
>>
>>
>>         _______________________________________________
>>         Join us at ClueCon 2011, Aug 9-11, Chicago
>>         http://www.cluecon.com <http://www.cluecon.com/> 877-7-4ACLUE
>>
>>         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/>
>>
>>
>>     _______________________________________________
>>     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
>>     <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
>
>
>     _______________________________________________
>     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
>     <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
>
>
>
>
> _______________________________________________
> 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 anthony.minessale at gmail.com  Thu Jul 14 12:30:45 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jul 2011 03:30:45 -0500
Subject: [Freeswitch-users] multiple active grammars supported?
In-Reply-To: <33095823FD21DF429B481B5163264B7950CBC03597@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950CBC034F1@VMBX102.ihostexchange.net>
	<33095823FD21DF429B481B5163264B7950CBC03597@VMBX102.ihostexchange.net>
Message-ID: <CAKbxfG_hNkafdnh3Opw=nk8xvjG+L6fxQM5vaD20CJLbmzcSnA@mail.gmail.com>

We need to merge the change into our main git repo first.


On Wed, Jul 13, 2011 at 3:47 PM, Yungwei Chen <yungwei at resolvity.com> wrote:
> According to http://jira.freeswitch.org/browse/FS-2906, FS already supports multiple active grammars.
> Now I'm wondering how to take advantages of it from a javascript program.
> Thanks.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei Chen
> Sent: Wednesday, July 13, 2011 11:29 AM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] multiple active grammars supported?
>
> Hi,
>
> I tried the pizza demo shipped with freeswitch, and I am wondering if multiple active grammars are supported.
>
> For example:
> question: was this the pizza you had in mind?
> response: yes/no (defined in grammar 1) Or
> ? ? ? ? ?help ? (defined in grammar 2; this case will be handled differently)
>
> Thanks.
>
> _______________________________________________
> 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
>
> _______________________________________________
> 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
>



-- 
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 freeswitch at earthspike.net  Thu Jul 14 15:38:55 2011
From: freeswitch at earthspike.net (John)
Date: Thu, 14 Jul 2011 12:38:55 +0100
Subject: [Freeswitch-users] [SOLVED] Re:  UK ISDN outgoing CLI
In-Reply-To: <4E1DFD11.40805@earthspike.net>
References: <4DED4854.6010203@earthspike.net>	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>	<4DF10C83.6020803@earthspike.net>	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>	<4DF1F07A.5000200@earthspike.net>	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>	<4DFB9831.4040709@earthspike.net>	<BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>	<CAP1_MOCq5qRjsgd7XGag9ukjHiG2Se-pXvrghQ7hbB5ciwJK+A@mail.gmail.com>	<4E156252.2030308@earthspike.net>	<CAP1_MOBAFpYqJNKhgWzv3aYJeKbj+RJDQJCaQByVcrMVp_s8RA@mail.gmail.com>	<CAE=3qOZaqoa1zt1_q_gaYzME=DOJYqhW4i7afhrBYc4PxViq5g@mail.gmail.com>	<C06E47096A4E434AB7466C6309AC9FE5@dell9400>	<4E163411.50507@earthspike.net>	<CAA8_NKCs_CJKEtHG9CpDZvK9vXevGtjEA18tNoEZ0yzKc2kmYg@mail.gmail.com>
	<4E1DFD11.40805@earthspike.net>
Message-ID: <4E1ED54F.2070806@earthspike.net>

This has been solved.  BT Openreach identified an issue with the 
exchange configuration that was forcing the lead number to be used as 
the outgoing caller id irrespective of the number from our FS box.  Many 
thanks to those who have offered helpful suggestions; it has allowed me 
to eliminate all alternatives and, ultimately, to articulate the fault 
in terms of non-compliance with Q951.3.

Summary of solution:

    Outgoing NPI: ISDN
    Outgoing TON: national
    Outgoing number format: 10 digit (ie strip leading 0)


Fault report that finally got results (real numbers have been substituted):

    For outgoing calls, SNDDI numbers (654321 or 654322) presented in
    the Calling Number Information Element of the Q931 SETUP packet with
    the screening indicator set to 'user-provided, verified and passed'
    are being overwritten by the BT exchange to the default number
    (654320) with the screening indicator set to 'network provided'. 
    This is contrary to Q951.3 which requires properly constituted
    Calling Numbers to be passed [irrespective of the CLIP supplementary
    service being provided].  The fault can be replicated by dialling
    654320 or 654322 from 654321.  The Caller ID passed should be
    1234654321 but instead is 1234654320.

Helpful diagnosis tools/hints:

    * Anecdotes from former BT engineers suggest that testing CLI with
      mobile phones is bad practice; mobile phone companies pick up some
      strange signalling.  I am not sure that this isn't just folklore
      or due to number-matching algorithms in the handset, but I chose
      to use my home analogue landline for testing.
    * Making calls back to the same set of lines means that you can see
      the full signalling for both outgoing and terminating legs, and
      the only thing in the way is the local exchange.
    * wanpipemon allows Sangoma board users to capture the D-channel
      signalling and then view/process using wireshark/tshark.  This is
      a game-changer.  Syntax is:

      wanpipemon -i w1g1 -pcap -pcap_file isdn.pcap -prot ISDN -full-systime -c  trd

      Replace w1g1 with the wanpipe device name, and the output will go
      to isdn.pcap.  You will need to run one capture for each BRI line
      you have (eg w2g1, w3g1 etc and to isdn1.pcap, isdn2.pcap, etc).

I hope to put something somewhere on the Wiki with this information when 
I get a spare moment (currently scheduled for Jan 2034).

Thanks again to all those who have helped resolve this.

John

On 13/07/11 21:16, John wrote:
> Thanks, Gavin. That would be really useful.
>
> John
>
> On 08/07/11 19:51, Gavin Henry wrote:
>> Hi John,
>>
>> We set this up for a customer using a Sangoma card and FusionPBX not
>> long ago. I can dig out the config next week when back in the office.
>> It's for ISDN30e but should help.
>>
>> Thanks.
>>
>> On 7/7/11, John<freeswitch at earthspike.net>   wrote:
>>> Jan,
>>>
>>> I have tried every combination of unknown, national and international
>>> with every combination of number length.  I have also confirmed that by
>>> using wanpipemon to capture the D-channel messages.  When I went through
>>> them with Wireshark and compared them with Q.931, everything was being
>>> set correctly by the Sangoma card/drivers.  I am at the point now where
>>> I need someone who has worked with BT ISDN switches to tell me what they
>>> can accept.  It seems that this is a national secret over here, or
>>> instead that they can accept almost anything and my lines have been
>>> configured to override the outgoing calling number with the base
>>> number.  Either way, the only way in which I think I am now going to get
>>> a result is by booking a fault.  Your helpful comment about modern
>>> switches being able to work this out also confirms that I am probably
>>> not looking at a Q.931 formatting error on my part, but on a blockage in
>>> the exchange.  I don't have a problem with the called number, for example.
>>>
>>> John
>>>
>>> On 07/07/11 23:09, Jan Berger wrote:
>>>> I must admit I don't remember what CLIP and CLOP is -- and my Q.931
>>>> experience is getting a bit rusty...
>>>>
>>>> Can you get a snoop of L3 out and in so I can see what you send and
>>>> what the switch responds back?
>>>>
>>>> I prefer to use Called and Calling -- The numbers contain a few bits
>>>> that tell what number this is. The most common are unknown, national
>>>> and international. These bits must match the actual number you send.
>>>> The 10 digit number is a unknown, the 11 digit is a national.
>>>> International start with 00 nn
>>>>
>>>> On top of that you will face that modern switches are quite capable
>>>> and able to configure whatever behaviour they want per line with
>>>> regards to called/calling -- so you need to find out exactly what the
>>>> switch expect on both called and caller for outgoing to behave as you
>>>> want.
>>>>
>>>> /Jan
>>>>
>>>>>> On our PRI systems we are sending 10 digits (2071231234), and on the
>>>>>> BRI system we are sending 11 digits (02071231234).
>>>>>>
>
> _______________________________________________
> 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 gmaruzz at gmail.com  Thu Jul 14 16:36:37 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 14 Jul 2011 14:36:37 +0200
Subject: [Freeswitch-users] [SOLVED] Re: UK ISDN outgoing CLI
In-Reply-To: <4E1ED54F.2070806@earthspike.net>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
	<4DF10C83.6020803@earthspike.net>
	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>
	<4DF1F07A.5000200@earthspike.net>
	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>
	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>
	<4DFB9831.4040709@earthspike.net>
	<BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>
	<CAP1_MOCq5qRjsgd7XGag9ukjHiG2Se-pXvrghQ7hbB5ciwJK+A@mail.gmail.com>
	<4E156252.2030308@earthspike.net>
	<CAP1_MOBAFpYqJNKhgWzv3aYJeKbj+RJDQJCaQByVcrMVp_s8RA@mail.gmail.com>
	<CAE=3qOZaqoa1zt1_q_gaYzME=DOJYqhW4i7afhrBYc4PxViq5g@mail.gmail.com>
	<C06E47096A4E434AB7466C6309AC9FE5@dell9400>
	<4E163411.50507@earthspike.net>
	<CAA8_NKCs_CJKEtHG9CpDZvK9vXevGtjEA18tNoEZ0yzKc2kmYg@mail.gmail.com>
	<4E1DFD11.40805@earthspike.net> <4E1ED54F.2070806@earthspike.net>
Message-ID: <CALXCt0r3FT6tjdx6pKfvmOpjz+21eG=rxF0p51nutZZq3Dfx4w@mail.gmail.com>

Thanks a lot for this detailed report.
Please, take a moment to at least cut and paste this mail in the wiki.
If deemed necessary, others will edit/reformat it.
-giovanni

On 7/14/11, John <freeswitch at earthspike.net> wrote:
> This has been solved.  BT Openreach identified an issue with the
> exchange configuration that was forcing the lead number to be used as
> the outgoing caller id irrespective of the number from our FS box.  Many
> thanks to those who have offered helpful suggestions; it has allowed me
> to eliminate all alternatives and, ultimately, to articulate the fault
> in terms of non-compliance with Q951.3.
>
> Summary of solution:
>
>     Outgoing NPI: ISDN
>     Outgoing TON: national
>     Outgoing number format: 10 digit (ie strip leading 0)
>
>
> Fault report that finally got results (real numbers have been substituted):
>
>     For outgoing calls, SNDDI numbers (654321 or 654322) presented in
>     the Calling Number Information Element of the Q931 SETUP packet with
>     the screening indicator set to 'user-provided, verified and passed'
>     are being overwritten by the BT exchange to the default number
>     (654320) with the screening indicator set to 'network provided'.
>     This is contrary to Q951.3 which requires properly constituted
>     Calling Numbers to be passed [irrespective of the CLIP supplementary
>     service being provided].  The fault can be replicated by dialling
>     654320 or 654322 from 654321.  The Caller ID passed should be
>     1234654321 but instead is 1234654320.
>
> Helpful diagnosis tools/hints:
>
>     * Anecdotes from former BT engineers suggest that testing CLI with
>       mobile phones is bad practice; mobile phone companies pick up some
>       strange signalling.  I am not sure that this isn't just folklore
>       or due to number-matching algorithms in the handset, but I chose
>       to use my home analogue landline for testing.
>     * Making calls back to the same set of lines means that you can see
>       the full signalling for both outgoing and terminating legs, and
>       the only thing in the way is the local exchange.
>     * wanpipemon allows Sangoma board users to capture the D-channel
>       signalling and then view/process using wireshark/tshark.  This is
>       a game-changer.  Syntax is:
>
>       wanpipemon -i w1g1 -pcap -pcap_file isdn.pcap -prot ISDN -full-systime
> -c  trd
>
>       Replace w1g1 with the wanpipe device name, and the output will go
>       to isdn.pcap.  You will need to run one capture for each BRI line
>       you have (eg w2g1, w3g1 etc and to isdn1.pcap, isdn2.pcap, etc).
>
> I hope to put something somewhere on the Wiki with this information when
> I get a spare moment (currently scheduled for Jan 2034).
>
> Thanks again to all those who have helped resolve this.
>
> John
>
> On 13/07/11 21:16, John wrote:
>> Thanks, Gavin. That would be really useful.
>>
>> John
>>
>> On 08/07/11 19:51, Gavin Henry wrote:
>>> Hi John,
>>>
>>> We set this up for a customer using a Sangoma card and FusionPBX not
>>> long ago. I can dig out the config next week when back in the office.
>>> It's for ISDN30e but should help.
>>>
>>> Thanks.
>>>
>>> On 7/7/11, John<freeswitch at earthspike.net>   wrote:
>>>> Jan,
>>>>
>>>> I have tried every combination of unknown, national and international
>>>> with every combination of number length.  I have also confirmed that by
>>>> using wanpipemon to capture the D-channel messages.  When I went through
>>>> them with Wireshark and compared them with Q.931, everything was being
>>>> set correctly by the Sangoma card/drivers.  I am at the point now where
>>>> I need someone who has worked with BT ISDN switches to tell me what they
>>>> can accept.  It seems that this is a national secret over here, or
>>>> instead that they can accept almost anything and my lines have been
>>>> configured to override the outgoing calling number with the base
>>>> number.  Either way, the only way in which I think I am now going to get
>>>> a result is by booking a fault.  Your helpful comment about modern
>>>> switches being able to work this out also confirms that I am probably
>>>> not looking at a Q.931 formatting error on my part, but on a blockage in
>>>> the exchange.  I don't have a problem with the called number, for
>>>> example.
>>>>
>>>> John
>>>>
>>>> On 07/07/11 23:09, Jan Berger wrote:
>>>>> I must admit I don't remember what CLIP and CLOP is -- and my Q.931
>>>>> experience is getting a bit rusty...
>>>>>
>>>>> Can you get a snoop of L3 out and in so I can see what you send and
>>>>> what the switch responds back?
>>>>>
>>>>> I prefer to use Called and Calling -- The numbers contain a few bits
>>>>> that tell what number this is. The most common are unknown, national
>>>>> and international. These bits must match the actual number you send.
>>>>> The 10 digit number is a unknown, the 11 digit is a national.
>>>>> International start with 00 nn
>>>>>
>>>>> On top of that you will face that modern switches are quite capable
>>>>> and able to configure whatever behaviour they want per line with
>>>>> regards to called/calling -- so you need to find out exactly what the
>>>>> switch expect on both called and caller for outgoing to behave as you
>>>>> want.
>>>>>
>>>>> /Jan
>>>>>
>>>>>>> On our PRI systems we are sending 10 digits (2071231234), and on the
>>>>>>> BRI system we are sending 11 digits (02071231234).
>>>>>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From fabio.bigliardi at gmail.com  Thu Jul 14 17:06:14 2011
From: fabio.bigliardi at gmail.com (Fabio Bigliardi)
Date: Thu, 14 Jul 2011 15:06:14 +0200
Subject: [Freeswitch-users] Problem with pause
Message-ID: <CABtP9QjCdwZKLS94F4Lv=kYrXSqC3CFvsWZqNqeB=5Lhj=Nm9g@mail.gmail.com>

Hi all,

I'm trying to have an audio stream played back on a channel and to
pause/resume it through API commands on event socket.

Below you can find the sequence of commands and outputs:

$ telnet 127.0.0.1 8021
Trying 127.0.0.1...
Connected to 127.0.0.1.
Escape character is '^]'.
Content-Type: auth/request

auth ClueCon

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

api originate user/1003 &playback(${sounds_dir}/jazz.wav)

Content-Type: api/response
Content-Length: 41

+OK 8fcdb494-82a4-42c4-b011-ecca1da0ade2
api pause 8fcdb494-82a4-42c4-b011-ecca1da0ade2 on

Content-Type: api/response
Content-Length: 14

*-ERR no reply*
**

*
*

The "pause" command is successful (the stream is really paused) but the api
response indicates an error.

Same output with the CLI.

Is it correct?

Thank you very much.

Fabio Bigliardi
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From tculjaga at gmail.com  Thu Jul 14 17:14:54 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 14 Jul 2011 15:14:54 +0200
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAKbxfG9CG0Q11guXN=ONM+Vg8oc1do_ABdn1=6Wh8DywO-0a4w@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
	<CAKbxfG9CG0Q11guXN=ONM+Vg8oc1do_ABdn1=6Wh8DywO-0a4w@mail.gmail.com>
Message-ID: <CAJoKKdUnh_pYWtyUWYz=G8P7Fwz1pjhPfTmPciRzHZWdFmCH8A@mail.gmail.com>

On Wed, Jul 13, 2011 at 5:48 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> 4 days is a long time.
> At the moment you decide to post here you should try latest again to be
> safe.
>


I'm sorry for the late response... been on holidays and all i get is a very
limited access to the internet... Thank you for the prompt response!



>
> Are you saying you are using sipp or something to call this extension?
>
>

yes, I'm using sipp to generate traffic ... on a rate of 10 CPS and seeing
that kind of issue... very simple scenario that I always use before going
live in production.


Currently, I'm updating to the latest git while I'm writing this e-mail.

commit 2932c1fa17c589f3e47178553f78195fe08cf22d
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Thu Jul 14 00:17:05 2011 -0500

    FS-3386 Try this revision please


I will upgrade and will perform the same test again ...


T.
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From tculjaga at gmail.com  Thu Jul 14 17:15:40 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 14 Jul 2011 15:15:40 +0200
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAKbxfG-F-VKpe1xSWUYbyQS1-W+4HSG-+6PpWwRrEPF+a2XGpg@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
	<CAKbxfG9CG0Q11guXN=ONM+Vg8oc1do_ABdn1=6Wh8DywO-0a4w@mail.gmail.com>
	<CAKbxfG-F-VKpe1xSWUYbyQS1-W+4HSG-+6PpWwRrEPF+a2XGpg@mail.gmail.com>
Message-ID: <CAJoKKdVJpK_bzE5i1B3rt1QTrpsuKfsPa+KCxiFRKvBY_ORqPg@mail.gmail.com>

On Wed, Jul 13, 2011 at 5:52 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> oh and btw
>
> you should be filing a jira not posting it to here
>
>
I will open a jira ticket in a few .. sorry for the inconvenience.
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From jmoran at secureachsystems.com  Thu Jul 14 17:21:08 2011
From: jmoran at secureachsystems.com (Jason Moran)
Date: Thu, 14 Jul 2011 09:21:08 -0400
Subject: [Freeswitch-users] session.recordFile create empy wav in
	whileloop
References: <CACREEeaJ8On1Oo2K64PMisbJE+jASFC9tLTxXNZg71bS+3vVig@mail.gmail.com><CAFiqYumaqz-tP9=bjSemCgjaxThfki9DNDFKkDEH7seL7ADHxQ@mail.gmail.com><CACREEeZFOKXHDTu-5YEsP1UUN64Tj+tLYP=hj95F67=gjZV3vA@mail.gmail.com><699A1DD4-6D3B-441C-9B01-8DF1997F0FAE@danielknoll.de><CAFiqYunwXoK2RbX26toMon0Swfsd51jx2ynjAS+CpyytAHnmiA@mail.gmail.com>
	<6AB0454C-9146-4B67-A5FF-DD4C72314F80@danielknoll.de>
Message-ID: <361E98F99D3CC3439EED59BC1924ED695083DD@SERVER2003.SecuReachSystems.local>

I thought you could stop the record via an action in your callback
function?
For example if you used this format:
session.recordFile(fileName, callbackFunction, "", maxreclen,
silencethreshold, silencehits);

Let's say you passed in "onRecordDTMF" as your callbackFunction:
session.recordFile("/dev/shm/"+rd+".wav", onRecordDTMF, "", 10, 500, 3);

function onRecordDTMF(session, type, digits, arg)
{
    if (digits.digit == "#") {
        return false;
    }
    else {
        return true;
    }
}

The recording would end in 2 possible ways: Let it have the silence
timeout or allow a force recording end with the "#" DTMF symbol.  Your
recording message could indicate to hit the "#" to end recording (or
allow the silence to stop it).

-Jason

-----Original Message-----
From: Daniel Knoll [mailto:daniel at danielknoll.de] 
Sent: Wednesday, July 13, 2011 6:28 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] session.recordFile create empy wav in
whileloop

Hi Steve,
Thanx for helping me,
the call is established and the loop can be re-run and re-run. but the
new audio file that is created is empty.
i have the same issue if i use session.execute("record",
"/dev/shm/"+rd+".wav 10 150");

unfortunately there is no function to stop the record.

If you can help me I would be very thankful

Daniel
 

Am 13.07.2011 um 23:27 schrieb Steven Ayre:

> I'd need to set up a testcase to confirm it, but I think it's probably
> that recordFile blocks and doesn't return until it's finished
> recording at the end of the call. As a result on the next loop
> iteration there'll be no call so nothing to record.
> 
> -Steve
> 
> 
> 
> On 13 July 2011 22:17, Daniel Knoll <daniel at danielknoll.de> wrote:
>> Any Ideas for my problem?
>> The question is, why is the audio file empty if i re-run in while
loop?
>> 
>> Thanks
>> Daniel
>> 
>> 
>> Am 13.07.2011 um 11:24 schrieb Daniel Knoll:
>> 
>>> Hi Steve, thanks for fast answer. The Channel is't hung up, because
I
>>> stream in the same loop an Audio File
>>> 
>>> here is the code, the session.recordFile is executed in the second
>>> run, because, the filesize is changing for the file at first run.
>>> 
>>> do {
>>>       var rd = session.uuid;
>>>
session.streamFile("/dev/shm/freeswitch/sounds/conf-asrintro.wav", "");
>>>       session.recordFile("/dev/shm/"+rd+".wav", "", "", 10, 500, 3);
>>>               system("sox -v 1.7 /dev/shm/"+rd+".wav -r 16000
/dev/shm/"+rd+".flac");
>>>       var fd = new File("|/usr/bin/googlespeech.sh
/dev/shm/"+rd+".flac");
>>>       fd.open("read");
>>>       var dtmf = fd.read("8");
>>>       dtmf = dtmf.replace(/[^0-9]/g, '');
>>>       console_log("notice", "DTMF: "+ dtmf +"\n");
>>>       roomExists = checkRoom(dtmf);
>>>       if(!roomExists){
>>>               retry_count++;
>>>
session.streamFile("/dev/shm/freeswitch/sounds/conf-invalid.wav", "");
>>>               var dtmf = "";
>>>       }
>>> } while(!roomExists && retry_count < 5);
>>> 
>>> 
>>> I googled at this and found some guys he has also the same problem
>>>
http://asterisk.voicemeup.com/viewtopic.php?p=62294&sid=8360229099e219df
50bbb016cd43cbfd
>>> 
>>> Can you help me ?
>>> 
>>> Thanx Daniel
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 2011/7/13, Steven Ayre <steveayre at gmail.com>:
>>>> Are you able to share the code?
>>>> 
>>>> Are you checking session.ready in the while loop's condition? Might
the
>>>> channel be hung up when it reaches the 2nd recordFile?
>>>> 
>>>> -Steve
>>>> 
>>>> 
>>>> On 13 July 2011 10:01, Daniel Knoll <daniel at danielknoll.de> wrote:
>>>> 
>>>>> Hey Guys,
>>>>> 
>>>>> i have a strange Problem with session.recordFile in a while loop
and i
>>>>> don't know how to solve it.
>>>>> The first run in the loop is absolutly ok, session.recordFile is
>>>>> creating a good clean wavefile.
>>>>> but in the second, third ..  run, every wav file has a size of
47724
>>>>> byte and no audio content.
>>>>> 
>>>>> Please help me, because it is an urgent case.
>>>>> 
>>>>> Thankx for getting help.
>>>>> Daniel
>>>>> 
>>>>> _______________________________________________
>>>>> 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
>>>>> 
>>>> 
>>> 
>>> 
>>> --
>>> Daniel Knoll
>>> Liberdastr.. 9
>>> 12047 Berlin
>>> 
>>> fon +49 (0)179 20 16 50 8
>>> mail daniel at danielknoll.de
>>> web www.danielknoll.de
>> 
>> 
>> _______________________________________________
>> 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
>> 
> 
> _______________________________________________
> 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 tculjaga at gmail.com  Thu Jul 14 17:23:08 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 14 Jul 2011 15:23:08 +0200
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAFiqYukgW1UOgxNUZs8rPvWNmXx5sDVmGS_D+r_xdLMsXg_i4Q@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
	<CAFiqYukgW1UOgxNUZs8rPvWNmXx5sDVmGS_D+r_xdLMsXg_i4Q@mail.gmail.com>
Message-ID: <CAJoKKdX8J68pmUogKgp1Ew4ACpkHntiHoZ2Rpx5JNhepsOVVvw@mail.gmail.com>

On Wed, Jul 13, 2011 at 6:17 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Does "fsctl reclaim_mem" help at all?
>
> -Steve
>
> Hi Steven,
 I just did on one of the affected servers ... what should i get as a result
?
memory returned, freed ?


anyhow, after 5 minutes observation the memory keeps growing up... but looks
like i still have time :)

$ sudo cat /proc/`ps aux | grep freeswitch | grep -v grep | awk '{print
$2}'`/smaps | egrep "(heap)" -A 7
110ee000-2086d000 rw-p 110ee000 00:00 0
[heap]
Size:            253436 kB
Rss:             253240 kB
Shared_Clean:         0 kB
Shared_Dirty:         0 kB
Private_Clean:        0 kB
Private_Dirty:   253240 kB
Swap:        0 kB
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From jmoran at secureachsystems.com  Thu Jul 14 17:26:38 2011
From: jmoran at secureachsystems.com (Jason Moran)
Date: Thu, 14 Jul 2011 09:26:38 -0400
Subject: [Freeswitch-users] session.recordFile create empy wav in
	whileloop
References: <CACREEeaJ8On1Oo2K64PMisbJE+jASFC9tLTxXNZg71bS+3vVig@mail.gmail.com><CAFiqYumaqz-tP9=bjSemCgjaxThfki9DNDFKkDEH7seL7ADHxQ@mail.gmail.com><CACREEeZFOKXHDTu-5YEsP1UUN64Tj+tLYP=hj95F67=gjZV3vA@mail.gmail.com><699A1DD4-6D3B-441C-9B01-8DF1997F0FAE@danielknoll.de><CAFiqYunwXoK2RbX26toMon0Swfsd51jx2ynjAS+CpyytAHnmiA@mail.gmail.com>
	<6AB0454C-9146-4B67-A5FF-DD4C72314F80@danielknoll.de>
Message-ID: <361E98F99D3CC3439EED59BC1924ED695083DE@SERVER2003.SecuReachSystems.local>

One other thing to check - make sure you aren't having problems with
open file handles.  You might have to play around with issuing an
fd.close() at the appropriate time or else try incrementing the file
name (and other permutations) to ensure you aren't blocking yourself
from writing the same file.

-Jason

-----Original Message-----
From: Daniel Knoll [mailto:daniel at danielknoll.de] 
Sent: Wednesday, July 13, 2011 6:28 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] session.recordFile create empy wav in
whileloop

Hi Steve,
Thanx for helping me,
the call is established and the loop can be re-run and re-run. but the
new audio file that is created is empty.
i have the same issue if i use session.execute("record",
"/dev/shm/"+rd+".wav 10 150");

unfortunately there is no function to stop the record.

If you can help me I would be very thankful

Daniel
 

Am 13.07.2011 um 23:27 schrieb Steven Ayre:

> I'd need to set up a testcase to confirm it, but I think it's probably
> that recordFile blocks and doesn't return until it's finished
> recording at the end of the call. As a result on the next loop
> iteration there'll be no call so nothing to record.
> 
> -Steve
> 
> 
> 
> On 13 July 2011 22:17, Daniel Knoll <daniel at danielknoll.de> wrote:
>> Any Ideas for my problem?
>> The question is, why is the audio file empty if i re-run in while
loop?
>> 
>> Thanks
>> Daniel
>> 
>> 
>> Am 13.07.2011 um 11:24 schrieb Daniel Knoll:
>> 
>>> Hi Steve, thanks for fast answer. The Channel is't hung up, because
I
>>> stream in the same loop an Audio File
>>> 
>>> here is the code, the session.recordFile is executed in the second
>>> run, because, the filesize is changing for the file at first run.
>>> 
>>> do {
>>>       var rd = session.uuid;
>>>
session.streamFile("/dev/shm/freeswitch/sounds/conf-asrintro.wav", "");
>>>       session.recordFile("/dev/shm/"+rd+".wav", "", "", 10, 500, 3);
>>>               system("sox -v 1.7 /dev/shm/"+rd+".wav -r 16000
/dev/shm/"+rd+".flac");
>>>       var fd = new File("|/usr/bin/googlespeech.sh
/dev/shm/"+rd+".flac");
>>>       fd.open("read");
>>>       var dtmf = fd.read("8");
>>>       dtmf = dtmf.replace(/[^0-9]/g, '');
>>>       console_log("notice", "DTMF: "+ dtmf +"\n");
>>>       roomExists = checkRoom(dtmf);
>>>       if(!roomExists){
>>>               retry_count++;
>>>
session.streamFile("/dev/shm/freeswitch/sounds/conf-invalid.wav", "");
>>>               var dtmf = "";
>>>       }
>>> } while(!roomExists && retry_count < 5);
>>> 
>>> 
>>> I googled at this and found some guys he has also the same problem
>>>
http://asterisk.voicemeup.com/viewtopic.php?p=62294&sid=8360229099e219df
50bbb016cd43cbfd
>>> 
>>> Can you help me ?
>>> 
>>> Thanx Daniel
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 2011/7/13, Steven Ayre <steveayre at gmail.com>:
>>>> Are you able to share the code?
>>>> 
>>>> Are you checking session.ready in the while loop's condition? Might
the
>>>> channel be hung up when it reaches the 2nd recordFile?
>>>> 
>>>> -Steve
>>>> 
>>>> 
>>>> On 13 July 2011 10:01, Daniel Knoll <daniel at danielknoll.de> wrote:
>>>> 
>>>>> Hey Guys,
>>>>> 
>>>>> i have a strange Problem with session.recordFile in a while loop
and i
>>>>> don't know how to solve it.
>>>>> The first run in the loop is absolutly ok, session.recordFile is
>>>>> creating a good clean wavefile.
>>>>> but in the second, third ..  run, every wav file has a size of
47724
>>>>> byte and no audio content.
>>>>> 
>>>>> Please help me, because it is an urgent case.
>>>>> 
>>>>> Thankx for getting help.
>>>>> Daniel
>>>>> 
>>>>> _______________________________________________
>>>>> 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
>>>>> 
>>>> 
>>> 
>>> 
>>> --
>>> Daniel Knoll
>>> Liberdastr.. 9
>>> 12047 Berlin
>>> 
>>> fon +49 (0)179 20 16 50 8
>>> mail daniel at danielknoll.de
>>> web www.danielknoll.de
>> 
>> 
>> _______________________________________________
>> 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
>> 
> 
> _______________________________________________
> 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 steveayre at gmail.com  Thu Jul 14 17:35:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 14 Jul 2011 14:35:33 +0100
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAJoKKdX8J68pmUogKgp1Ew4ACpkHntiHoZ2Rpx5JNhepsOVVvw@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
	<CAFiqYukgW1UOgxNUZs8rPvWNmXx5sDVmGS_D+r_xdLMsXg_i4Q@mail.gmail.com>
	<CAJoKKdX8J68pmUogKgp1Ew4ACpkHntiHoZ2Rpx5JNhepsOVVvw@mail.gmail.com>
Message-ID: <CAFiqYunZUs0d-kQVTxXEvuqVveiri6WFFW6oT3UYujPGoa2H-w@mail.gmail.com>

It -might- return some unused memory to the system when you run it,
but it wouldn't stop it growing again.

-Steve



On 14 July 2011 14:23, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
>
> On Wed, Jul 13, 2011 at 6:17 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> Does "fsctl reclaim_mem" help at all?
>>
>> -Steve
>>
> Hi Steven,
> ?I just did on one of the affected servers ... what should i get as a result
> ?
> memory returned, freed ?
>
>
> anyhow, after 5 minutes observation the memory keeps growing up... but looks
> like i still have time :)
>
> $ sudo cat /proc/`ps aux | grep freeswitch | grep -v grep | awk '{print
> $2}'`/smaps | egrep "(heap)" -A 7
> 110ee000-2086d000 rw-p 110ee000 00:00 0
> [heap]
> Size:??????????? 253436 kB
> Rss:???????????? 253240 kB
> Shared_Clean:???????? 0 kB
> Shared_Dirty:???????? 0 kB
> Private_Clean:??????? 0 kB
> Private_Dirty:?? 253240 kB
> Swap:??????? 0 kB
>
>
>
> _______________________________________________
> 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 shouldbeq931 at gmail.com  Thu Jul 14 17:48:31 2011
From: shouldbeq931 at gmail.com (shouldbe q931)
Date: Thu, 14 Jul 2011 14:48:31 +0100
Subject: [Freeswitch-users] [SOLVED] Re: UK ISDN outgoing CLI
In-Reply-To: <4E1ED54F.2070806@earthspike.net>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
	<4DF10C83.6020803@earthspike.net>
	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>
	<4DF1F07A.5000200@earthspike.net>
	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>
	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>
	<4DFB9831.4040709@earthspike.net>
	<BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>
	<CAP1_MOCq5qRjsgd7XGag9ukjHiG2Se-pXvrghQ7hbB5ciwJK+A@mail.gmail.com>
	<4E156252.2030308@earthspike.net>
	<CAP1_MOBAFpYqJNKhgWzv3aYJeKbj+RJDQJCaQByVcrMVp_s8RA@mail.gmail.com>
	<CAE=3qOZaqoa1zt1_q_gaYzME=DOJYqhW4i7afhrBYc4PxViq5g@mail.gmail.com>
	<C06E47096A4E434AB7466C6309AC9FE5@dell9400>
	<4E163411.50507@earthspike.net>
	<CAA8_NKCs_CJKEtHG9CpDZvK9vXevGtjEA18tNoEZ0yzKc2kmYg@mail.gmail.com>
	<4E1DFD11.40805@earthspike.net> <4E1ED54F.2070806@earthspike.net>
Message-ID: <CAP1_MOABgzJMabATDnqeLhOe816RiQQ+QLGnO2JhbEBctPi70A@mail.gmail.com>

On Thu, Jul 14, 2011 at 12:38 PM, John <freeswitch at earthspike.net> wrote:
> This has been solved.? BT Openreach identified an issue with the exchange
> configuration that was forcing the lead number to be used as the outgoing
> caller id irrespective of the number from our FS box.? Many thanks to those
> who have offered helpful suggestions; it has allowed me to eliminate all
> alternatives and, ultimately, to articulate the fault in terms of
> non-compliance with Q951.3.
>

Excellent news!

"issue with the exchange configuration" is all too common :-(


From michal.zubac at comgate.cz  Thu Jul 14 18:01:54 2011
From: michal.zubac at comgate.cz (=?UTF-8?B?TWljaGFsIFp1YsOhxI0=?=)
Date: Thu, 14 Jul 2011 16:01:54 +0200
Subject: [Freeswitch-users] ftmod_libPRI outgoing calls channel selection
Message-ID: <4E1EF6D2.7040705@comgate.cz>

Hi.

I've noticed that outgoing call channels in freetdm/ftmod_libpri are not 
selected exclusively in Q.931 SETUP messages. Channel identification 
"exclusive" bit is 0 so it only means "preferred".
In some cases freeswitch selected channel which was not ready for 
dialing yet and ISDN operator answered with another channel in PROGRESS 
message. So actual call state messages were passed trough another 
channel and it resulted with lots of VETO and consecutive RESET 
messages, because of channel state confusion in FreeSwitch.

I think that it's better to force channel selection and in some cases 
choose wrong channel and get info about it. It's definitely better, then 
choose something, assume that it is correct although actual 
communication is done on another channel and such calls are "doomed"...

I've attached patch, in which we force selection of ISDN channel. Maybe 
better solution will be some configuration variable, which will control 
this behaviour. Consider.

Does anybody have similar experience and if so, how do you deal with it? 
Or am I mistaken in some of my conclusions?

-- 
Michal Zubac
ComGate Interactive s.r.o.

Prague Marina Office Center
Jankovcova 1596/14a
17000 Praha 7, Czech Republic

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From tculjaga at gmail.com  Thu Jul 14 18:06:34 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 14 Jul 2011 16:06:34 +0200
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAFiqYunZUs0d-kQVTxXEvuqVveiri6WFFW6oT3UYujPGoa2H-w@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
	<CAFiqYukgW1UOgxNUZs8rPvWNmXx5sDVmGS_D+r_xdLMsXg_i4Q@mail.gmail.com>
	<CAJoKKdX8J68pmUogKgp1Ew4ACpkHntiHoZ2Rpx5JNhepsOVVvw@mail.gmail.com>
	<CAFiqYunZUs0d-kQVTxXEvuqVveiri6WFFW6oT3UYujPGoa2H-w@mail.gmail.com>
Message-ID: <CAJoKKdXhHyQUXKCzaDc3GSu5m9A3Jdv2m9K=FaY7vt=W1OD5Qg@mail.gmail.com>

On Thu, Jul 14, 2011 at 3:35 PM, Steven Ayre <steveayre at gmail.com> wrote:

> It -might- return some unused memory to the system when you run it,
> but it wouldn't stop it growing again.
>
> -Steve
>
> im performing the test now ... will be able to tell.

btw: how to be sure im running the correct fs ?.... i mean ... the version
from command line returns just "FreeSWITCH Version 1.0.head (git-)" ... any
chance it can return the commit-id  or something ?
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From steveayre at gmail.com  Thu Jul 14 18:27:28 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 14 Jul 2011 15:27:28 +0100
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAJoKKdXhHyQUXKCzaDc3GSu5m9A3Jdv2m9K=FaY7vt=W1OD5Qg@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
	<CAFiqYukgW1UOgxNUZs8rPvWNmXx5sDVmGS_D+r_xdLMsXg_i4Q@mail.gmail.com>
	<CAJoKKdX8J68pmUogKgp1Ew4ACpkHntiHoZ2Rpx5JNhepsOVVvw@mail.gmail.com>
	<CAFiqYunZUs0d-kQVTxXEvuqVveiri6WFFW6oT3UYujPGoa2H-w@mail.gmail.com>
	<CAJoKKdXhHyQUXKCzaDc3GSu5m9A3Jdv2m9K=FaY7vt=W1OD5Qg@mail.gmail.com>
Message-ID: <CAFiqYumuca1OMQ13di425JFjs2dhPeLAi2e0FPu-kG2odJSVqQ@mail.gmail.com>

By the way, if it makes any difference it still doesn't fix the leak,
that'll still need tracking down.

The truncated git version means you're on an old version of git.
Update to 1.7. It shouldn't cause any problems other than breaking the
version number.

In the git checkout (if you still have it) "git log | head" should
show you the latest revision you had checked out.

-Steve



On 14 July 2011 15:06, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
>
> On Thu, Jul 14, 2011 at 3:35 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> It -might- return some unused memory to the system when you run it,
>> but it wouldn't stop it growing again.
>>
>> -Steve
>>
> im performing the test now ... will be able to tell.
>
> btw: how to be sure im running the correct fs ?.... i mean ... the version
> from command line returns just "FreeSWITCH Version 1.0.head (git-)" ... any
> chance it can return the commit-id? or something ?
>
> _______________________________________________
> 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 dimskraft at gmail.com  Thu Jul 14 12:43:46 2011
From: dimskraft at gmail.com (Dmitry Kravchenko)
Date: Thu, 14 Jul 2011 12:43:46 +0400
Subject: [Freeswitch-users] How to run flex client correctly?
Message-ID: <CADDV4gwUwgP6qoj5FG78enij0P1Uj=gQFWezDZpqM2kDaD00fw@mail.gmail.com>

Hi!


How to run flex client correctly?


If I just put a repository files from here
http://fisheye.freeswitch.org/browse/freeswitch.git/clients/flex into
webserver root, and open freeswitch.html, I get damaged web page with
apparently no design. Looks like stylesheets not work. No question about
allowing microphone appears.


If I open swf file in flashplayer, it is just blank.


Changing flashvars to correct freeswitch installation URL does not help.
Running mxmlc does not help (although it runs without errors).


How correctly working client should look like?


Where to put freeswitch credentials?


Thanks.
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From manjiri05_deshpande at yahoo.co.in  Thu Jul 14 16:24:46 2011
From: manjiri05_deshpande at yahoo.co.in (Manjiri Deshpande)
Date: Thu, 14 Jul 2011 17:54:46 +0530 (IST)
Subject: [Freeswitch-users] Cisco 7960 registration issue using mod_skinny
In-Reply-To: <mailman.12842.1310560306.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.12842.1310560306.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <1310646286.6873.YahooMailNeo@web95908.mail.in.yahoo.com>



Hi,
?
From last week I was trying to register Cisco phone with FreeSwitch?using mod_skinny.I am using solarwinds as?TFTP server?. But registration is not successful.
Phone cycles through the process of?TFTP?download and sending register message along with some more SKINNY messages to FreeSwitch.
Also on?FreeSwitch Console I get message as "UCM-Closed-TCP" 

Attached are the?wireshark capture for phone and FreeSwitch.log file
From?wireshark capture we can clearly see that phone sends "RegisterMessage" SKINNY?message in a cycle. This is the consistent flow.
Also attaching SEP[MAC].cnf.xml file which I am using.

?
Ph IP?Addr : 192.1681.1.205 
FreeSwitchIP?Addr : 192.168.1.116

Please have a look at this issue and let me know if I am missing anything here

Thanks,
Manjiri
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Subject: Cisco 7960 registration issue using mod_skinny
Date: Wed, 13 Jul 2011 18:00:33 +0530 (IST)
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From boris at tagnet.ru  Thu Jul 14 19:06:59 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 14 Jul 2011 21:06:59 +0600
Subject: [Freeswitch-users] Javascript from javascript == session freeze
Message-ID: <4E1F0613.2010007@tagnet.ru>

Hello!


     I modified my DISA javascript to use 'originate' back to my FS with 
collected number. The context where the call is placed calls another 
simple javascript application. And when application is called my session 
freezes. If I remove call to javascript application all is ok. So am I 
missed something in docs and I can not call this way or should I open JIRA?

-- 
Regards,
Boris




From anthony.minessale at gmail.com  Thu Jul 14 19:11:30 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jul 2011 10:11:30 -0500
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAFiqYumuca1OMQ13di425JFjs2dhPeLAi2e0FPu-kG2odJSVqQ@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
	<CAFiqYukgW1UOgxNUZs8rPvWNmXx5sDVmGS_D+r_xdLMsXg_i4Q@mail.gmail.com>
	<CAJoKKdX8J68pmUogKgp1Ew4ACpkHntiHoZ2Rpx5JNhepsOVVvw@mail.gmail.com>
	<CAFiqYunZUs0d-kQVTxXEvuqVveiri6WFFW6oT3UYujPGoa2H-w@mail.gmail.com>
	<CAJoKKdXhHyQUXKCzaDc3GSu5m9A3Jdv2m9K=FaY7vt=W1OD5Qg@mail.gmail.com>
	<CAFiqYumuca1OMQ13di425JFjs2dhPeLAi2e0FPu-kG2odJSVqQ@mail.gmail.com>
Message-ID: <CAKbxfG-GSbSm-wq5+YnT0jvtiiP32xJDgc9pqV1TUXj69CwDcg@mail.gmail.com>

you can use gigs before you stop growing in some cases esp on sipp test.
you need a lot more than 5 min to judge it.
more like dozen or more hours.


On Thu, Jul 14, 2011 at 9:27 AM, Steven Ayre <steveayre at gmail.com> wrote:
> By the way, if it makes any difference it still doesn't fix the leak,
> that'll still need tracking down.
>
> The truncated git version means you're on an old version of git.
> Update to 1.7. It shouldn't cause any problems other than breaking the
> version number.
>
> In the git checkout (if you still have it) "git log | head" should
> show you the latest revision you had checked out.
>
> -Steve
>
>
>
> On 14 July 2011 15:06, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>>
>>
>> On Thu, Jul 14, 2011 at 3:35 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>> It -might- return some unused memory to the system when you run it,
>>> but it wouldn't stop it growing again.
>>>
>>> -Steve
>>>
>> im performing the test now ... will be able to tell.
>>
>> btw: how to be sure im running the correct fs ?.... i mean ... the version
>> from command line returns just "FreeSWITCH Version 1.0.head (git-)" ... any
>> chance it can return the commit-id? or something ?
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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
>



-- 
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 gavin.henry at gmail.com  Thu Jul 14 19:15:42 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Thu, 14 Jul 2011 16:15:42 +0100
Subject: [Freeswitch-users] [SOLVED] Re: UK ISDN outgoing CLI
In-Reply-To: <4E1ED54F.2070806@earthspike.net>
References: <4DED4854.6010203@earthspike.net>
	<BANLkTimP23FVs8Vcg+pbxnfBucrQoJur2w@mail.gmail.com>
	<4DF10C83.6020803@earthspike.net>
	<BANLkTikCEKqPTFZLDxF+h940RnmizHYLBA@mail.gmail.com>
	<4DF1F07A.5000200@earthspike.net>
	<BANLkTin5+puYLD7OtreiRxQZ5i8xd5KppQ@mail.gmail.com>
	<BANLkTi=8p2hy6TuPbtQd4tr9oREemVzUhg@mail.gmail.com>
	<4DFB9831.4040709@earthspike.net>
	<BANLkTinXZo=e-Cd+uL=mX21jXZNYBWdHuA@mail.gmail.com>
	<CAP1_MOCq5qRjsgd7XGag9ukjHiG2Se-pXvrghQ7hbB5ciwJK+A@mail.gmail.com>
	<4E156252.2030308@earthspike.net>
	<CAP1_MOBAFpYqJNKhgWzv3aYJeKbj+RJDQJCaQByVcrMVp_s8RA@mail.gmail.com>
	<CAE=3qOZaqoa1zt1_q_gaYzME=DOJYqhW4i7afhrBYc4PxViq5g@mail.gmail.com>
	<C06E47096A4E434AB7466C6309AC9FE5@dell9400>
	<4E163411.50507@earthspike.net>
	<CAA8_NKCs_CJKEtHG9CpDZvK9vXevGtjEA18tNoEZ0yzKc2kmYg@mail.gmail.com>
	<4E1DFD11.40805@earthspike.net> <4E1ED54F.2070806@earthspike.net>
Message-ID: <CAA8_NKDc_me3YXw2nsXv+BChkpD6ysuKEU_zQkshUKHQsqr8Wg@mail.gmail.com>

And I'd just dug out that config! Well done though. Glad it was their fault!


From tculjaga at gmail.com  Thu Jul 14 19:48:30 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 14 Jul 2011 17:48:30 +0200
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAKbxfG-GSbSm-wq5+YnT0jvtiiP32xJDgc9pqV1TUXj69CwDcg@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
	<CAFiqYukgW1UOgxNUZs8rPvWNmXx5sDVmGS_D+r_xdLMsXg_i4Q@mail.gmail.com>
	<CAJoKKdX8J68pmUogKgp1Ew4ACpkHntiHoZ2Rpx5JNhepsOVVvw@mail.gmail.com>
	<CAFiqYunZUs0d-kQVTxXEvuqVveiri6WFFW6oT3UYujPGoa2H-w@mail.gmail.com>
	<CAJoKKdXhHyQUXKCzaDc3GSu5m9A3Jdv2m9K=FaY7vt=W1OD5Qg@mail.gmail.com>
	<CAFiqYumuca1OMQ13di425JFjs2dhPeLAi2e0FPu-kG2odJSVqQ@mail.gmail.com>
	<CAKbxfG-GSbSm-wq5+YnT0jvtiiP32xJDgc9pqV1TUXj69CwDcg@mail.gmail.com>
Message-ID: <CAJoKKdUqV_ygenNTd+Z3hnoC2eXeiPuMOM2PvaChEm3kZXgj0A@mail.gmail.com>

On Thu, Jul 14, 2011 at 5:11 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you can use gigs before you stop growing in some cases esp on sipp test.
> you need a lot more than 5 min to judge it.
> more like dozen or more hours.
>
> yap, im running it for a very long time ... the example was just a snip ...
and i hit the swap right now :=) .... after 2 hours of sipp at 30 CPS....

start: 2011-07-14 12:45:43
stop: 2011-07-14 14:31:13


if i use less load it will grow slowly but it will end into swap... its just
a matter of time.

So, how can we troubleshoot this ?

im going to open a jira to move this out of Users Help mailing list...
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From daniel at danielknoll.de  Thu Jul 14 19:49:39 2011
From: daniel at danielknoll.de (Daniel Knoll)
Date: Thu, 14 Jul 2011 17:49:39 +0200
Subject: [Freeswitch-users] session.recordFile create empy wav in
	whileloop
In-Reply-To: <361E98F99D3CC3439EED59BC1924ED695083DE@SERVER2003.SecuReachSystems.local>
References: <CACREEeaJ8On1Oo2K64PMisbJE+jASFC9tLTxXNZg71bS+3vVig@mail.gmail.com><CAFiqYumaqz-tP9=bjSemCgjaxThfki9DNDFKkDEH7seL7ADHxQ@mail.gmail.com><CACREEeZFOKXHDTu-5YEsP1UUN64Tj+tLYP=hj95F67=gjZV3vA@mail.gmail.com><699A1DD4-6D3B-441C-9B01-8DF1997F0FAE@danielknoll.de><CAFiqYunwXoK2RbX26toMon0Swfsd51jx2ynjAS+CpyytAHnmiA@mail.gmail.com>
	<6AB0454C-9146-4B67-A5FF-DD4C72314F80@danielknoll.de>
	<361E98F99D3CC3439EED59BC1924ED695083DE@SERVER2003.SecuReachSystems.local>
Message-ID: <BD0ED492-D527-49D7-B169-4D0644B91DE6@danielknoll.de>

Hello Jason,

Thank you for your help. Both of them improved the code. Closing the open file handle is very good, because i saw in the process view many "sh" processes.
But the error is another. 

I increase the time of silence to 7 instead of 3.
it seems to be that in this time that session.streamFile is executed (3sec. audio file) , the next instruction session.recordFile is executed, because the first 5 seconds of audio files in the second run are silence.

I have no answer about this control.
for the first my changes works for me, but with a little bit more waiting time.  


Daniel

Am 14.07.2011 um 15:26 schrieb Jason Moran:

> One other thing to check - make sure you aren't having problems with
> open file handles.  You might have to play around with issuing an
> fd.close() at the appropriate time or else try incrementing the file
> name (and other permutations) to ensure you aren't blocking yourself
> from writing the same file.
> 
> -Jason
> 
> -----Original Message-----
> From: Daniel Knoll [mailto:daniel at danielknoll.de] 
> Sent: Wednesday, July 13, 2011 6:28 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] session.recordFile create empy wav in
> whileloop
> 
> Hi Steve,
> Thanx for helping me,
> the call is established and the loop can be re-run and re-run. but the
> new audio file that is created is empty.
> i have the same issue if i use session.execute("record",
> "/dev/shm/"+rd+".wav 10 150");
> 
> unfortunately there is no function to stop the record.
> 
> If you can help me I would be very thankful
> 
> Daniel
> 
> 
> Am 13.07.2011 um 23:27 schrieb Steven Ayre:
> 
>> I'd need to set up a testcase to confirm it, but I think it's probably
>> that recordFile blocks and doesn't return until it's finished
>> recording at the end of the call. As a result on the next loop
>> iteration there'll be no call so nothing to record.
>> 
>> -Steve
>> 
>> 
>> 
>> On 13 July 2011 22:17, Daniel Knoll <daniel at danielknoll.de> wrote:
>>> Any Ideas for my problem?
>>> The question is, why is the audio file empty if i re-run in while
> loop?
>>> 
>>> Thanks
>>> Daniel
>>> 
>>> 
>>> Am 13.07.2011 um 11:24 schrieb Daniel Knoll:
>>> 
>>>> Hi Steve, thanks for fast answer. The Channel is't hung up, because
> I
>>>> stream in the same loop an Audio File
>>>> 
>>>> here is the code, the session.recordFile is executed in the second
>>>> run, because, the filesize is changing for the file at first run.
>>>> 
>>>> do {
>>>>      var rd = session.uuid;
>>>> 
> session.streamFile("/dev/shm/freeswitch/sounds/conf-asrintro.wav", "");
>>>>      session.recordFile("/dev/shm/"+rd+".wav", "", "", 10, 500, 3);
>>>>              system("sox -v 1.7 /dev/shm/"+rd+".wav -r 16000
> /dev/shm/"+rd+".flac");
>>>>      var fd = new File("|/usr/bin/googlespeech.sh
> /dev/shm/"+rd+".flac");
>>>>      fd.open("read");
>>>>      var dtmf = fd.read("8");
>>>>      dtmf = dtmf.replace(/[^0-9]/g, '');
>>>>      console_log("notice", "DTMF: "+ dtmf +"\n");
>>>>      roomExists = checkRoom(dtmf);
>>>>      if(!roomExists){
>>>>              retry_count++;
>>>> 
> session.streamFile("/dev/shm/freeswitch/sounds/conf-invalid.wav", "");
>>>>              var dtmf = "";
>>>>      }
>>>> } while(!roomExists && retry_count < 5);
>>>> 
>>>> 
>>>> I googled at this and found some guys he has also the same problem
>>>> 
> http://asterisk.voicemeup.com/viewtopic.php?p=62294&sid=8360229099e219df
> 50bbb016cd43cbfd
>>>> 
>>>> Can you help me ?
>>>> 
>>>> Thanx Daniel
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 2011/7/13, Steven Ayre <steveayre at gmail.com>:
>>>>> Are you able to share the code?
>>>>> 
>>>>> Are you checking session.ready in the while loop's condition? Might
> the
>>>>> channel be hung up when it reaches the 2nd recordFile?
>>>>> 
>>>>> -Steve
>>>>> 
>>>>> 
>>>>> On 13 July 2011 10:01, Daniel Knoll <daniel at danielknoll.de> wrote:
>>>>> 
>>>>>> Hey Guys,
>>>>>> 
>>>>>> i have a strange Problem with session.recordFile in a while loop
> and i
>>>>>> don't know how to solve it.
>>>>>> The first run in the loop is absolutly ok, session.recordFile is
>>>>>> creating a good clean wavefile.
>>>>>> but in the second, third ..  run, every wav file has a size of
> 47724
>>>>>> byte and no audio content.
>>>>>> 
>>>>>> Please help me, because it is an urgent case.
>>>>>> 
>>>>>> Thankx for getting help.
>>>>>> Daniel
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 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
>>>>>> 
>>>>> 
>>>> 
>>>> 
>>>> --
>>>> Daniel Knoll
>>>> Liberdastr.. 9
>>>> 12047 Berlin
>>>> 
>>>> fon +49 (0)179 20 16 50 8
>>>> mail daniel at danielknoll.de
>>>> web www.danielknoll.de
>>> 
>>> 
>>> _______________________________________________
>>> 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
>>> 
>> 
>> _______________________________________________
>> 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
> 
> 
> 
> 
> _______________________________________________
> 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 anthony.minessale at gmail.com  Thu Jul 14 19:56:12 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Jul 2011 10:56:12 -0500
Subject: [Freeswitch-users] session.recordFile create empy wav in
	whileloop
In-Reply-To: <BD0ED492-D527-49D7-B169-4D0644B91DE6@danielknoll.de>
References: <CACREEeaJ8On1Oo2K64PMisbJE+jASFC9tLTxXNZg71bS+3vVig@mail.gmail.com>
	<CAFiqYumaqz-tP9=bjSemCgjaxThfki9DNDFKkDEH7seL7ADHxQ@mail.gmail.com>
	<CACREEeZFOKXHDTu-5YEsP1UUN64Tj+tLYP=hj95F67=gjZV3vA@mail.gmail.com>
	<699A1DD4-6D3B-441C-9B01-8DF1997F0FAE@danielknoll.de>
	<CAFiqYunwXoK2RbX26toMon0Swfsd51jx2ynjAS+CpyytAHnmiA@mail.gmail.com>
	<6AB0454C-9146-4B67-A5FF-DD4C72314F80@danielknoll.de>
	<361E98F99D3CC3439EED59BC1924ED695083DE@SERVER2003.SecuReachSystems.local>
	<BD0ED492-D527-49D7-B169-4D0644B91DE6@danielknoll.de>
Message-ID: <CAKbxfG9vOTnPxpAtNqJ=0jxL7ayAD=1JKrR0c504m-Os_4MgdQ@mail.gmail.com>

seems like speechtools.jm and mod_flite "see pizza demo" would be a
little easier.

Anyway,
Try session.execute("sleep", "100"); each time before you record to
flush the audio buffers.

On Thu, Jul 14, 2011 at 10:49 AM, Daniel Knoll <daniel at danielknoll.de> wrote:
> Hello Jason,
> Thank you for your help. Both of them improved the code. Closing the open
> file handle is very good, because i saw in the process view many "sh"
> processes.
> But the error is another.
> I increase the time of silence to 7 instead of 3.
> it seems to be that in this time that?session.streamFile is executed (3sec.
> audio file) , the next instruction?session.recordFile is executed, because
> the first 5 seconds of audio files in the second run are silence.
> I have no answer about this control.
> for the first my changes works for me, but with a little bit more waiting
> time.
>
> Daniel
> Am 14.07.2011 um 15:26 schrieb Jason Moran:
>
> One other thing to check - make sure you aren't having problems with
> open file handles. ?You might have to play around with issuing an
> fd.close() at the appropriate time or else try incrementing the file
> name (and other permutations) to ensure you aren't blocking yourself
> from writing the same file.
>
> -Jason
>
> -----Original Message-----
> From: Daniel Knoll [mailto:daniel at danielknoll.de]
> Sent: Wednesday, July 13, 2011 6:28 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] session.recordFile create empy wav in
> whileloop
>
> Hi Steve,
> Thanx for helping me,
> the call is established and the loop can be re-run and re-run. but the
> new audio file that is created is empty.
> i have the same issue if i use session.execute("record",
> "/dev/shm/"+rd+".wav 10 150");
>
> unfortunately there is no function to stop the record.
>
> If you can help me I would be very thankful
>
> Daniel
>
>
> Am 13.07.2011 um 23:27 schrieb Steven Ayre:
>
> I'd need to set up a testcase to confirm it, but I think it's probably
>
> that recordFile blocks and doesn't return until it's finished
>
> recording at the end of the call. As a result on the next loop
>
> iteration there'll be no call so nothing to record.
>
> -Steve
>
>
>
> On 13 July 2011 22:17, Daniel Knoll <daniel at danielknoll.de> wrote:
>
> Any Ideas for my problem?
>
> The question is, why is the audio file empty if i re-run in while
>
> loop?
>
> Thanks
>
> Daniel
>
>
> Am 13.07.2011 um 11:24 schrieb Daniel Knoll:
>
> Hi Steve, thanks for fast answer. The Channel is't hung up, because
>
> I
>
> stream in the same loop an Audio File
>
> here is the code, the session.recordFile is executed in the second
>
> run, because, the filesize is changing for the file at first run.
>
> do {
>
> ?????var rd = session.uuid;
>
> session.streamFile("/dev/shm/freeswitch/sounds/conf-asrintro.wav", "");
>
> ?????session.recordFile("/dev/shm/"+rd+".wav", "", "", 10, 500, 3);
>
> ?????????????system("sox -v 1.7 /dev/shm/"+rd+".wav -r 16000
>
> /dev/shm/"+rd+".flac");
>
> ?????var fd = new File("|/usr/bin/googlespeech.sh
>
> /dev/shm/"+rd+".flac");
>
> ?????fd.open("read");
>
> ?????var dtmf = fd.read("8");
>
> ?????dtmf = dtmf.replace(/[^0-9]/g, '');
>
> ?????console_log("notice", "DTMF: "+ dtmf +"\n");
>
> ?????roomExists = checkRoom(dtmf);
>
> ?????if(!roomExists){
>
> ?????????????retry_count++;
>
> session.streamFile("/dev/shm/freeswitch/sounds/conf-invalid.wav", "");
>
> ?????????????var dtmf = "";
>
> ?????}
>
> } while(!roomExists && retry_count < 5);
>
>
> I googled at this and found some guys he has also the same problem
>
> http://asterisk.voicemeup.com/viewtopic.php?p=62294&sid=8360229099e219df
> 50bbb016cd43cbfd
>
> Can you help me ?
>
> Thanx Daniel
>
>
>
>
>
> 2011/7/13, Steven Ayre <steveayre at gmail.com>:
>
> Are you able to share the code?
>
> Are you checking session.ready in the while loop's condition? Might
>
> the
>
> channel be hung up when it reaches the 2nd recordFile?
>
> -Steve
>
>
> On 13 July 2011 10:01, Daniel Knoll <daniel at danielknoll.de> wrote:
>
> Hey Guys,
>
> i have a strange Problem with session.recordFile in a while loop
>
> and i
>
> don't know how to solve it.
>
> The first run in the loop is absolutly ok, session.recordFile is
>
> creating a good clean wavefile.
>
> but in the second, third .. ?run, every wav file has a size of
>
> 47724
>
> byte and no audio content.
>
> Please help me, because it is an urgent case.
>
> Thankx for getting help.
>
> Daniel
>
> _______________________________________________
>
> 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
>
>
>
>
> --
>
> Daniel Knoll
>
> Liberdastr.. 9
>
> 12047 Berlin
>
> fon +49 (0)179 20 16 50 8
>
> mail daniel at danielknoll.de
>
> web www.danielknoll.de
>
>
> _______________________________________________
>
> 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
>
>
> _______________________________________________
>
> 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
>
>
>
>
> _______________________________________________
> 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
>
>
> _______________________________________________
> 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
>
>



-- 
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 daniel at danielknoll.de  Thu Jul 14 20:18:24 2011
From: daniel at danielknoll.de (Daniel Knoll)
Date: Thu, 14 Jul 2011 18:18:24 +0200
Subject: [Freeswitch-users] session.recordFile create empy wav in
	whileloop
In-Reply-To: <CAKbxfG9vOTnPxpAtNqJ=0jxL7ayAD=1JKrR0c504m-Os_4MgdQ@mail.gmail.com>
References: <CACREEeaJ8On1Oo2K64PMisbJE+jASFC9tLTxXNZg71bS+3vVig@mail.gmail.com>
	<CAFiqYumaqz-tP9=bjSemCgjaxThfki9DNDFKkDEH7seL7ADHxQ@mail.gmail.com>
	<CACREEeZFOKXHDTu-5YEsP1UUN64Tj+tLYP=hj95F67=gjZV3vA@mail.gmail.com>
	<699A1DD4-6D3B-441C-9B01-8DF1997F0FAE@danielknoll.de>
	<CAFiqYunwXoK2RbX26toMon0Swfsd51jx2ynjAS+CpyytAHnmiA@mail.gmail.com>
	<6AB0454C-9146-4B67-A5FF-DD4C72314F80@danielknoll.de>
	<361E98F99D3CC3439EED59BC1924ED695083DE@SERVER2003.SecuReachSystems.local>
	<BD0ED492-D527-49D7-B169-4D0644B91DE6@danielknoll.de>
	<CAKbxfG9vOTnPxpAtNqJ=0jxL7ayAD=1JKrR0c504m-Os_4MgdQ@mail.gmail.com>
Message-ID: <02E9A93F-7A7E-4288-99F1-FD100FE1C946@danielknoll.de>

Hey Anthony,
flushing the audio buffer before recording was the right key. Works well.

Thank you, Steve and Jason for helping me.
I'm happy like a child.

Daniel


Am 14.07.2011 um 17:56 schrieb Anthony Minessale:

> seems like speechtools.jm and mod_flite "see pizza demo" would be a
> little easier.
> 
> Anyway,
> Try session.execute("sleep", "100"); each time before you record to
> flush the audio buffers.
> 
> On Thu, Jul 14, 2011 at 10:49 AM, Daniel Knoll <daniel at danielknoll.de> wrote:
>> Hello Jason,
>> Thank you for your help. Both of them improved the code. Closing the open
>> file handle is very good, because i saw in the process view many "sh"
>> processes.
>> But the error is another.
>> I increase the time of silence to 7 instead of 3.
>> it seems to be that in this time that session.streamFile is executed (3sec.
>> audio file) , the next instruction session.recordFile is executed, because
>> the first 5 seconds of audio files in the second run are silence.
>> I have no answer about this control.
>> for the first my changes works for me, but with a little bit more waiting
>> time.
>> 
>> Daniel
>> Am 14.07.2011 um 15:26 schrieb Jason Moran:
>> 
>> One other thing to check - make sure you aren't having problems with
>> open file handles.  You might have to play around with issuing an
>> fd.close() at the appropriate time or else try incrementing the file
>> name (and other permutations) to ensure you aren't blocking yourself
>> from writing the same file.
>> 
>> -Jason
>> 
>> -----Original Message-----
>> From: Daniel Knoll [mailto:daniel at danielknoll.de]
>> Sent: Wednesday, July 13, 2011 6:28 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] session.recordFile create empy wav in
>> whileloop
>> 
>> Hi Steve,
>> Thanx for helping me,
>> the call is established and the loop can be re-run and re-run. but the
>> new audio file that is created is empty.
>> i have the same issue if i use session.execute("record",
>> "/dev/shm/"+rd+".wav 10 150");
>> 
>> unfortunately there is no function to stop the record.
>> 
>> If you can help me I would be very thankful
>> 
>> Daniel
>> 
>> 
>> Am 13.07.2011 um 23:27 schrieb Steven Ayre:
>> 
>> I'd need to set up a testcase to confirm it, but I think it's probably
>> 
>> that recordFile blocks and doesn't return until it's finished
>> 
>> recording at the end of the call. As a result on the next loop
>> 
>> iteration there'll be no call so nothing to record.
>> 
>> -Steve
>> 
>> 
>> 
>> On 13 July 2011 22:17, Daniel Knoll <daniel at danielknoll.de> wrote:
>> 
>> Any Ideas for my problem?
>> 
>> The question is, why is the audio file empty if i re-run in while
>> 
>> loop?
>> 
>> Thanks
>> 
>> Daniel
>> 
>> 
>> Am 13.07.2011 um 11:24 schrieb Daniel Knoll:
>> 
>> Hi Steve, thanks for fast answer. The Channel is't hung up, because
>> 
>> I
>> 
>> stream in the same loop an Audio File
>> 
>> here is the code, the session.recordFile is executed in the second
>> 
>> run, because, the filesize is changing for the file at first run.
>> 
>> do {
>> 
>>      var rd = session.uuid;
>> 
>> session.streamFile("/dev/shm/freeswitch/sounds/conf-asrintro.wav", "");
>> 
>>      session.recordFile("/dev/shm/"+rd+".wav", "", "", 10, 500, 3);
>> 
>>              system("sox -v 1.7 /dev/shm/"+rd+".wav -r 16000
>> 
>> /dev/shm/"+rd+".flac");
>> 
>>      var fd = new File("|/usr/bin/googlespeech.sh
>> 
>> /dev/shm/"+rd+".flac");
>> 
>>      fd.open("read");
>> 
>>      var dtmf = fd.read("8");
>> 
>>      dtmf = dtmf.replace(/[^0-9]/g, '');
>> 
>>      console_log("notice", "DTMF: "+ dtmf +"\n");
>> 
>>      roomExists = checkRoom(dtmf);
>> 
>>      if(!roomExists){
>> 
>>              retry_count++;
>> 
>> session.streamFile("/dev/shm/freeswitch/sounds/conf-invalid.wav", "");
>> 
>>              var dtmf = "";
>> 
>>      }
>> 
>> } while(!roomExists && retry_count < 5);
>> 
>> 
>> I googled at this and found some guys he has also the same problem
>> 
>> http://asterisk.voicemeup.com/viewtopic.php?p=62294&sid=8360229099e219df
>> 50bbb016cd43cbfd
>> 
>> Can you help me ?
>> 
>> Thanx Daniel
>> 
>> 
>> 
>> 
>> 
>> 2011/7/13, Steven Ayre <steveayre at gmail.com>:
>> 
>> Are you able to share the code?
>> 
>> Are you checking session.ready in the while loop's condition? Might
>> 
>> the
>> 
>> channel be hung up when it reaches the 2nd recordFile?
>> 
>> -Steve
>> 
>> 
>> On 13 July 2011 10:01, Daniel Knoll <daniel at danielknoll.de> wrote:
>> 
>> Hey Guys,
>> 
>> i have a strange Problem with session.recordFile in a while loop
>> 
>> and i
>> 
>> don't know how to solve it.
>> 
>> The first run in the loop is absolutly ok, session.recordFile is
>> 
>> creating a good clean wavefile.
>> 
>> but in the second, third ..  run, every wav file has a size of
>> 
>> 47724
>> 
>> byte and no audio content.
>> 
>> Please help me, because it is an urgent case.
>> 
>> Thankx for getting help.
>> 
>> Daniel
>> 
>> _______________________________________________
>> 
>> 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
>> 
>> 
>> 
>> 
>> --
>> 
>> Daniel Knoll
>> 
>> Liberdastr.. 9
>> 
>> 12047 Berlin
>> 
>> fon +49 (0)179 20 16 50 8
>> 
>> mail daniel at danielknoll.de
>> 
>> web www.danielknoll.de
>> 
>> 
>> _______________________________________________
>> 
>> 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
>> 
>> 
>> _______________________________________________
>> 
>> 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
>> 
>> 
>> 
>> 
>> _______________________________________________
>> 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
>> 
>> 
>> _______________________________________________
>> 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
>> 
>> 
> 
> 
> 
> -- 
> 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 math.parent at gmail.com  Thu Jul 14 20:27:37 2011
From: math.parent at gmail.com (Mathieu Parent)
Date: Thu, 14 Jul 2011 18:27:37 +0200
Subject: [Freeswitch-users] Cisco 7960 registration issue using
	mod_skinny
In-Reply-To: <1310646286.6873.YahooMailNeo@web95908.mail.in.yahoo.com>
References: <mailman.12842.1310560306.24555.freeswitch-users@lists.freeswitch.org>
	<1310646286.6873.YahooMailNeo@web95908.mail.in.yahoo.com>
Message-ID: <CAFX5sbxa2_ZrxH_HpBssEBvvapJQwU3zgtGbVeqO+R8HummT9A@mail.gmail.com>

(NB: you only need to send the mail to
freeswitch-users at lists.freeswitch.org, no -owner)

Hello,

2011/7/14 Manjiri Deshpande <manjiri05_deshpande at yahoo.co.in>:
>
> Hi,
>
> From last week I was trying to register Cisco phone with FreeSwitch?using
> mod_skinny.I am using solarwinds as?TFTP server?. But registration is not
> successful.
> Phone cycles through the process of?TFTP?download and sending register
> message along with some more SKINNY messages to FreeSwitch.
> Also on?FreeSwitch Console I get message as "UCM-Closed-TCP"

Thanks for this feedback.

Can you create a Jira for this bug?

You probably use an arch which is not x86 nor x86_64. I need more info
on this arch and compiler

Regards

> Attached are the?wireshark capture for phone and FreeSwitch.log file
> From?wireshark capture we can clearly see that phone sends "RegisterMessage"
> SKINNY?message in a cycle. This is the consistent flow.
> Also attaching SEP[MAC].cnf.xml file which I am using.
>
>
> Ph IP?Addr : 192.1681.1.205
> FreeSwitchIP?Addr : 192.168.1.116
>
> Please have a look at this issue and let me know if I am missing anything
> here
>
> Thanks,
> Manjiri


-- 
Mathieu Parent


From psilvao at gmail.com  Thu Jul 14 20:32:16 2011
From: psilvao at gmail.com (Pablo Silva)
Date: Thu, 14 Jul 2011 12:32:16 -0400
Subject: [Freeswitch-users] (no subject)
Message-ID: <CAF1in6RLB_ZLOE2R_Lhdpo5EiSWoVwQgT0mXg8V12S5T+10CTg@mail.gmail.com>

unsub


From dave at dchorton.com  Thu Jul 14 20:33:04 2011
From: dave at dchorton.com (Dave Horton)
Date: Thu, 14 Jul 2011 12:33:04 -0400
Subject: [Freeswitch-users] mod_directory questions (prompt missing)
Message-ID: <DB01FDD4-810D-45FD-BE24-D5C81D3A8863@dchorton.com>

I'm trying to use mod_directory and after building and installing am having a couple of problems in my testing:

First, it appears that there is a prompt missing:

2011-07-14 12:13:49.801293 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[directory/dir-press.wav] (en:en)
2011-07-14 12:13:49.801293 [ERR] mod_sndfile.c:194 Error Opening File [/usr/local/freeswitch/sounds/en/us/callie/directory/dir-press.wav] [System error : No such file or directory.]

All of the other prompts appear to be there (under /sounds/en/us/callie/directory/) but this one is missing for some reason.  I did a standard build from latest git as of a few weeks ago.  Is this a known issue, and is there somewhere I could get this prompt?

Secondly, right after I enter the DTMF digits for someone's last name it it reports that there is 1 match and spells out the name, etc, (and encounters the missing prompt) it then immediately tries to to a transfer to the extension that is currently active (which I named 'company_directory') rather than waiting for my DTMF input confirming that I want to be connected to the matching extension for the name I entered.  See log snippet below.  The way I have set this up for testing is that I added an option to the main ivr that does a menu-exec-app and transfers to the company_directory extension, which then executes the 'directory' app.  I'm not sure why the directory app finished running before collecting my response (as to whether I wanted to search again or connect to the matching extension) nor why it then tried to do a transfer (again) to 'company_directory' and then hung up.  


2011-07-14 12:13:48.513298 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[directory/dir-to_select_entry.wav] (en:en)
2011-07-14 12:13:48.533305 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
2011-07-14 12:13:49.681295 [DEBUG] switch_ivr_play_say.c:1649 done playing file
2011-07-14 12:13:49.921288 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en]
2011-07-14 12:13:49.921288 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[directory/dir-no_more_results.wav] (en:en)
2011-07-14 12:13:49.941293 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
2011-07-14 12:13:50.989241 [DEBUG] switch_ivr_play_say.c:1649 done playing file
EXECUTE sofia/external/+15083084809 at 65.162.239.203 transfer(company_directory enum)
2011-07-14 12:13:51.109194 [DEBUG] switch_ivr.c:1622 (sofia/external/+15083084809 at 65.162.239.203) State Change CS_EXECUTE -> CS_ROUTING
2011-07-14 12:13:51.109194 [DEBUG] switch_core_session.c:1118 Send signal sofia/external/+15083084809 at 65.162.239.203 [BREAK]
2011-07-14 12:13:51.109194 [DEBUG] switch_core_session.c:711 Send signal sofia/external/+15083084809 at 65.162.239.203 [BREAK]
2011-07-14 12:13:51.109194 [NOTICE] switch_ivr.c:1628 Transfer sofia/external/+15083084809 at 65.162.239.203 to enum[company_directory at default]

From vedran.zeljeznak at gmail.com  Thu Jul 14 20:47:30 2011
From: vedran.zeljeznak at gmail.com (Vedran Zeljeznak)
Date: Thu, 14 Jul 2011 18:47:30 +0200
Subject: [Freeswitch-users] No contact header on 302 message
Message-ID: <CALsBW=rP+h031N3jOTuO=EjpY8xjvf0FPD0mMA4uiiurvg+s7w@mail.gmail.com>

hello,

i'm having trouble with 302 messages, it looks like FreeSwitch is not
setting Contact header on Leg-A 302 message when 302 message occurs on
Leg-B.

There is a valid Contact header in 302 message on Leg-B. I've posted
FreeSwitch console output on: http://pastebin.freeswitch.org/16806.

Does anybody know how can I tell FreeSwitch to set Contact header on
Leg-A 302 message from Contact header in 302 message from Leg-B?

---
Vedran Zeljeznak


From msc at freeswitch.org  Thu Jul 14 20:49:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Jul 2011 09:49:49 -0700
Subject: [Freeswitch-users] Problem with pause
In-Reply-To: <CABtP9QjCdwZKLS94F4Lv=kYrXSqC3CFvsWZqNqeB=5Lhj=Nm9g@mail.gmail.com>
References: <CABtP9QjCdwZKLS94F4Lv=kYrXSqC3CFvsWZqNqeB=5Lhj=Nm9g@mail.gmail.com>
Message-ID: <CAKzWOxUbPy46J7tJApzM99TFJyMokBQx_WMrh8MyyghLFqUBDQ@mail.gmail.com>

I see this, too. I looked in the source code and it looks like there's no
explicit "+OK" anywhere. However, the "-ERR no reply" looks to me like what
would happen on a successful call to the pause API. I think you're okay to
assume that a "no reply" means your pause on/off worked.

-MC

On Thu, Jul 14, 2011 at 6:06 AM, Fabio Bigliardi
<fabio.bigliardi at gmail.com>wrote:

> Hi all,
>
> I'm trying to have an audio stream played back on a channel and to
> pause/resume it through API commands on event socket.
>
> Below you can find the sequence of commands and outputs:
>
> $ telnet 127.0.0.1 8021
> Trying 127.0.0.1...
> Connected to 127.0.0.1.
> Escape character is '^]'.
> Content-Type: auth/request
>
> auth ClueCon
>
> Content-Type: command/reply
> Reply-Text: +OK accepte
>
> api originate user/1003 &playback(${sounds_dir}/jazz.wav)
>
> Content-Type: api/response
> Content-Length: 41
>
> +OK 8fcdb494-82a4-42c4-b011-ecca1da0ade2
> api pause 8fcdb494-82a4-42c4-b011-ecca1da0ade2 on
>
> Content-Type: api/response
> Content-Length: 14
>
> *-ERR no reply*
> **
>
> *
> *
>
> The "pause" command is successful (the stream is really paused) but the api
> response indicates an error.
>
> Same output with the CLI.
>
> Is it correct?
>
> Thank you very much.
>
> Fabio Bigliardi
>
> _______________________________________________
> 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 msc at freeswitch.org  Thu Jul 14 21:01:32 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Jul 2011 10:01:32 -0700
Subject: [Freeswitch-users] mod_directory questions (prompt missing)
In-Reply-To: <DB01FDD4-810D-45FD-BE24-D5C81D3A8863@dchorton.com>
References: <DB01FDD4-810D-45FD-BE24-D5C81D3A8863@dchorton.com>
Message-ID: <CAKzWOxUb+K1K+4Xa6tSEVmL+J8A8_ujYE9EQNvhH7Db5j5rdHA@mail.gmail.com>

Get the latest git rev and be sure to update the conf/lang/en/dir/sounds.xml
file. It contains the correct file name for the "press.wav" file.

-MC

On Thu, Jul 14, 2011 at 9:33 AM, Dave Horton <dave at dchorton.com> wrote:

> I'm trying to use mod_directory and after building and installing am having
> a couple of problems in my testing:
>
> First, it appears that there is a prompt missing:
>
> 2011-07-14 12:13:49.801293 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[directory/dir-press.wav] (en:en)
> 2011-07-14 12:13:49.801293 [ERR] mod_sndfile.c:194 Error Opening File
> [/usr/local/freeswitch/sounds/en/us/callie/directory/dir-press.wav] [System
> error : No such file or directory.]
>
> All of the other prompts appear to be there (under
> /sounds/en/us/callie/directory/) but this one is missing for some reason.  I
> did a standard build from latest git as of a few weeks ago.  Is this a known
> issue, and is there somewhere I could get this prompt?
>
> Secondly, right after I enter the DTMF digits for someone's last name it it
> reports that there is 1 match and spells out the name, etc, (and encounters
> the missing prompt) it then immediately tries to to a transfer to the
> extension that is currently active (which I named 'company_directory')
> rather than waiting for my DTMF input confirming that I want to be connected
> to the matching extension for the name I entered.  See log snippet below.
>  The way I have set this up for testing is that I added an option to the
> main ivr that does a menu-exec-app and transfers to the company_directory
> extension, which then executes the 'directory' app.  I'm not sure why the
> directory app finished running before collecting my response (as to whether
> I wanted to search again or connect to the matching extension) nor why it
> then tried to do a transfer (again) to 'company_directory' and then hung up.
>
>
> 2011-07-14 12:13:48.513298 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[directory/dir-to_select_entry.wav] (en:en)
> 2011-07-14 12:13:48.533305 [DEBUG] switch_ivr_play_say.c:1279 Codec
> Activated L16 at 8000hz 1 channels 20ms
> 2011-07-14 12:13:49.681295 [DEBUG] switch_ivr_play_say.c:1649 done playing
> file
> 2011-07-14 12:13:49.921288 [DEBUG] switch_ivr_play_say.c:67 No language
> specified - Using [en]
> 2011-07-14 12:13:49.921288 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[directory/dir-no_more_results.wav] (en:en)
> 2011-07-14 12:13:49.941293 [DEBUG] switch_ivr_play_say.c:1279 Codec
> Activated L16 at 8000hz 1 channels 20ms
> 2011-07-14 12:13:50.989241 [DEBUG] switch_ivr_play_say.c:1649 done playing
> file
> EXECUTE sofia/external/+15083084809 at 65.162.239.203transfer(company_directory enum)
> 2011-07-14 12:13:51.109194 [DEBUG] switch_ivr.c:1622 (sofia/external/+
> 15083084809 at 65.162.239.203) State Change CS_EXECUTE -> CS_ROUTING
> 2011-07-14 12:13:51.109194 [DEBUG] switch_core_session.c:1118 Send signal
> sofia/external/+15083084809 at 65.162.239.203 [BREAK]
> 2011-07-14 12:13:51.109194 [DEBUG] switch_core_session.c:711 Send signal
> sofia/external/+15083084809 at 65.162.239.203 [BREAK]
> 2011-07-14 12:13:51.109194 [NOTICE] switch_ivr.c:1628 Transfer
> sofia/external/+15083084809 at 65.162.239.203 to
> enum[company_directory at default]
> _______________________________________________
> 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 boris at tagnet.ru  Thu Jul 14 21:10:34 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 14 Jul 2011 23:10:34 +0600
Subject: [Freeswitch-users] Same peace of code works with Javascript and not
	LUA
Message-ID: <4E1F230A.9080104@tagnet.ru>

Hello!

     I'm trying to migrate DISA from Javascript to LUA. This peace of 
code works fine in Javascript:

ostr = "{ignore_early_media=true" +
            ",origination_caller_id_number=" + 
session.getVariable("caller_id_number") +
                     "}sofia/ipbx/50004#" + digits + "@192.168.1.1:5060";
osession = new Session( ostr );
bridge(session, osession);
osession.hangup();

With LUA not:
         originate_string =
             "{ignore_early_media=true" ..
             ",origination_caller_id_number=" .. 
session:getVariable("caller_id_number") ..
             "}" ..
             "sofia/ipbx/50004#" ..
             digits .. "@192.168.1.1:5060";
         originate_session = freeswitch.Session( originate_string );
             freeswitch.bridge(session, originate_session);
             originate_session:hangup();

The call is droped when remote answers. There is an error in log: 
bridge: session not ready. I tried to use
     if( originate_session:ready() ) then
             freeswitch.bridge(session, originate_session);
             originate_session:hangup();
    end

with no luck :(. Please, help me. What am I doing wrong?

-- 
Regards,
Boris




From boris at tagnet.ru  Thu Jul 14 21:18:46 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 14 Jul 2011 23:18:46 +0600
Subject: [Freeswitch-users] Same peace of code works with Javascript and
 not LUA
In-Reply-To: <4E1F230A.9080104@tagnet.ru>
References: <4E1F230A.9080104@tagnet.ru>
Message-ID: <4E1F24F6.5080101@tagnet.ru>

Hello!

     I found a problem. originate_session must not to be local.
> Hello!
>
>       I'm trying to migrate DISA from Javascript to LUA. This peace of
> code works fine in Javascript:
>
> ostr = "{ignore_early_media=true" +
>              ",origination_caller_id_number=" +
> session.getVariable("caller_id_number") +
>                       "}sofia/ipbx/50004#" + digits + "@192.168.1.1:5060";
> osession = new Session( ostr );
> bridge(session, osession);
> osession.hangup();
>
> With LUA not:
>           originate_string =
>               "{ignore_early_media=true" ..
>               ",origination_caller_id_number=" ..
> session:getVariable("caller_id_number") ..
>               "}" ..
>               "sofia/ipbx/50004#" ..
>               digits .. "@192.168.1.1:5060";
>           originate_session = freeswitch.Session( originate_string );
>               freeswitch.bridge(session, originate_session);
>               originate_session:hangup();
>
> The call is droped when remote answers. There is an error in log:
> bridge: session not ready. I tried to use
>       if( originate_session:ready() ) then
>               freeswitch.bridge(session, originate_session);
>               originate_session:hangup();
>      end
>
> with no luck :(. Please, help me. What am I doing wrong?
>


-- 
Regards,
Boris




From steveayre at gmail.com  Thu Jul 14 21:29:17 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 14 Jul 2011 18:29:17 +0100
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAF1in6RLB_ZLOE2R_Lhdpo5EiSWoVwQgT0mXg8V12S5T+10CTg@mail.gmail.com>
References: <CAF1in6RLB_ZLOE2R_Lhdpo5EiSWoVwQgT0mXg8V12S5T+10CTg@mail.gmail.com>
Message-ID: <CAFiqYumsZCLfOOUB6W-+7R0oYPa0kS4URfA8_X3W8-WZJ-+j6Q@mail.gmail.com>

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



On 14 July 2011 17:32, Pablo Silva <psilvao at gmail.com> wrote:
> unsub
>
> _______________________________________________
> 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 boris at tagnet.ru  Thu Jul 14 21:39:11 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 14 Jul 2011 23:39:11 +0600
Subject: [Freeswitch-users] Same peace of code works with Javascript and
 not LUA
In-Reply-To: <4E1F24F6.5080101@tagnet.ru>
References: <4E1F230A.9080104@tagnet.ru> <4E1F24F6.5080101@tagnet.ru>
Message-ID: <4E1F29BF.2030209@tagnet.ru>

And another problem found. If the remote party answers immediately all 
is ok. But if remote waits more then 8-10 seconds before answer - the 
call is dropped when answered. Can't understand where to look for a problem.
> Hello!
>
>       I found a problem. originate_session must not to be local.
>> Hello!
>>
>>        I'm trying to migrate DISA from Javascript to LUA. This peace of
>> code works fine in Javascript:
>>
>> ostr = "{ignore_early_media=true" +
>>               ",origination_caller_id_number=" +
>> session.getVariable("caller_id_number") +
>>                        "}sofia/ipbx/50004#" + digits + "@192.168.1.1:5060";
>> osession = new Session( ostr );
>> bridge(session, osession);
>> osession.hangup();
>>
>> With LUA not:
>>            originate_string =
>>                "{ignore_early_media=true" ..
>>                ",origination_caller_id_number=" ..
>> session:getVariable("caller_id_number") ..
>>                "}" ..
>>                "sofia/ipbx/50004#" ..
>>                digits .. "@192.168.1.1:5060";
>>            originate_session = freeswitch.Session( originate_string );
>>                freeswitch.bridge(session, originate_session);
>>                originate_session:hangup();
>>
>> The call is droped when remote answers. There is an error in log:
>> bridge: session not ready. I tried to use
>>        if( originate_session:ready() ) then
>>                freeswitch.bridge(session, originate_session);
>>                originate_session:hangup();
>>       end
>>
>> with no luck :(. Please, help me. What am I doing wrong?
>>
>


-- 
Regards,
Boris




From msc at freeswitch.org  Thu Jul 14 21:43:54 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Jul 2011 10:43:54 -0700
Subject: [Freeswitch-users] Same peace of code works with Javascript and
 not LUA
In-Reply-To: <4E1F29BF.2030209@tagnet.ru>
References: <4E1F230A.9080104@tagnet.ru> <4E1F24F6.5080101@tagnet.ru>
	<4E1F29BF.2030209@tagnet.ru>
Message-ID: <CAKzWOxV_mtSxRR0XULjY6QC92wXnnrHL9eq-ZyPMeBoKX9eVrw@mail.gmail.com>

get a console log w/ siptrace of that scenario and put it on pastebin. the
gang here will take a look.
-MC

On Thu, Jul 14, 2011 at 10:39 AM, Boris Kovalenko <boris at tagnet.ru> wrote:

> And another problem found. If the remote party answers immediately all
> is ok. But if remote waits more then 8-10 seconds before answer - the
> call is dropped when answered. Can't understand where to look for a
> problem.
> > Hello!
> >
> >       I found a problem. originate_session must not to be local.
> >> Hello!
> >>
> >>        I'm trying to migrate DISA from Javascript to LUA. This peace of
> >> code works fine in Javascript:
> >>
> >> ostr = "{ignore_early_media=true" +
> >>               ",origination_caller_id_number=" +
> >> session.getVariable("caller_id_number") +
> >>                        "}sofia/ipbx/50004#" + digits + "@
> 192.168.1.1:5060";
> >> osession = new Session( ostr );
> >> bridge(session, osession);
> >> osession.hangup();
> >>
> >> With LUA not:
> >>            originate_string =
> >>                "{ignore_early_media=true" ..
> >>                ",origination_caller_id_number=" ..
> >> session:getVariable("caller_id_number") ..
> >>                "}" ..
> >>                "sofia/ipbx/50004#" ..
> >>                digits .. "@192.168.1.1:5060";
> >>            originate_session = freeswitch.Session( originate_string );
> >>                freeswitch.bridge(session, originate_session);
> >>                originate_session:hangup();
> >>
> >> The call is droped when remote answers. There is an error in log:
> >> bridge: session not ready. I tried to use
> >>        if( originate_session:ready() ) then
> >>                freeswitch.bridge(session, originate_session);
> >>                originate_session:hangup();
> >>       end
> >>
> >> with no luck :(. Please, help me. What am I doing wrong?
> >>
> >
>
>
> --
> Regards,
> Boris
>
>
>
> _______________________________________________
> 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 boris at tagnet.ru  Thu Jul 14 22:11:49 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 15 Jul 2011 00:11:49 +0600
Subject: [Freeswitch-users] Same peace of code works with Javascript and
 not LUA
In-Reply-To: <CAKzWOxV_mtSxRR0XULjY6QC92wXnnrHL9eq-ZyPMeBoKX9eVrw@mail.gmail.com>
References: <4E1F230A.9080104@tagnet.ru> <4E1F24F6.5080101@tagnet.ru>
	<4E1F29BF.2030209@tagnet.ru>
	<CAKzWOxV_mtSxRR0XULjY6QC92wXnnrHL9eq-ZyPMeBoKX9eVrw@mail.gmail.com>
Message-ID: <4E1F3165.70403@tagnet.ru>

Hello!

     Here it is http://pastebin.freeswitch.org/16809
     Hope somebody help
> get a console log w/ siptrace of that scenario and put it on pastebin. 
> the gang here will take a look.
> -MC
>
> On Thu, Jul 14, 2011 at 10:39 AM, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     And another problem found. If the remote party answers immediately all
>     is ok. But if remote waits more then 8-10 seconds before answer - the
>     call is dropped when answered. Can't understand where to look for
>     a problem.
>     > Hello!
>     >
>     >       I found a problem. originate_session must not to be local.
>     >> Hello!
>     >>
>     >>        I'm trying to migrate DISA from Javascript to LUA. This
>     peace of
>     >> code works fine in Javascript:
>     >>
>     >> ostr = "{ignore_early_media=true" +
>     >>               ",origination_caller_id_number=" +
>     >> session.getVariable("caller_id_number") +
>     >>                        "}sofia/ipbx/50004#" + digits +
>     "@192.168.1.1:5060 <http://192.168.1.1:5060>";
>     >> osession = new Session( ostr );
>     >> bridge(session, osession);
>     >> osession.hangup();
>     >>
>     >> With LUA not:
>     >>            originate_string =
>     >>                "{ignore_early_media=true" ..
>     >>                ",origination_caller_id_number=" ..
>     >> session:getVariable("caller_id_number") ..
>     >>                "}" ..
>     >>                "sofia/ipbx/50004#" ..
>     >>                digits .. "@192.168.1.1:5060
>     <http://192.168.1.1:5060>";
>     >>            originate_session = freeswitch.Session(
>     originate_string );
>     >>                freeswitch.bridge(session, originate_session);
>     >>                originate_session:hangup();
>     >>
>     >> The call is droped when remote answers. There is an error in log:
>     >> bridge: session not ready. I tried to use
>     >>        if( originate_session:ready() ) then
>     >>                freeswitch.bridge(session, originate_session);
>     >>                originate_session:hangup();
>     >>       end
>     >>
>     >> with no luck :(. Please, help me. What am I doing wrong?
>     >>
>     >
>
>
>     --
>     Regards,
>     Boris
>
>
>
>     _______________________________________________
>     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
>     <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
>
>
>
>
> _______________________________________________
> 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


-- 
Regards,
Boris


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From rzhang at gosilverplus.com  Thu Jul 14 22:38:44 2011
From: rzhang at gosilverplus.com (ran zhang)
Date: Thu, 14 Jul 2011 11:38:44 -0700
Subject: [Freeswitch-users] please help!!! how to limit each FS
 registered user's active calls
In-Reply-To: <CAKzWOxVtNTEptQn6n-1POqiF9ntpLV-SsZeP3oufnSOOpFPoLA@mail.gmail.com>
References: <4E1E454A.8040901@gosilverplus.com>
	<CAKzWOxVtNTEptQn6n-1POqiF9ntpLV-SsZeP3oufnSOOpFPoLA@mail.gmail.com>
Message-ID: <4E1F37B4.9050802@gosilverplus.com>

Mike:

     I have added 'limits.xml' as u said.   If i use 'hash', FS 
generates error "limit subsystem hash not found!", so i changed to 'db', 
following is my local extension in my dialplan, i changed '$1' to '11' 
since i just want to limit user/11, but its still limiting incoming 
calls on all the extensions.


    Also, is creating 'limits.xml' required?

    Would it work if i can just add 'handle_over_limit' extension in 
'default.xml', omit the 'XML over_limit_actions' part in the limit action?




<extension name="PBX Extension">
<condition field="destination_number" expression="^(1[0-9]{1})$">

<action application="limit" data="db ${domain} 11 1 handle_over_limit 
XML over_limit_actions"/>
<action application="bridge" data="user/$1"/>

</condition>
</extension>
On 7/13/2011 11:22 PM, Michael Collins wrote:
>
>
> On Wed, Jul 13, 2011 at 6:24 PM, ran zhang <rzhang at gosilverplus.com 
> <mailto:rzhang at gosilverplus.com>> wrote:
>
>     hi all:
>
>                I'm trying to limit each registered user's active calls, so
>     if that particular registered user pass the limit, FS stops
>     sending any
>     new incoming call to it and it can't make any new calls.
>
>                I looked into 'limit' api, the counter increment
>     everytime a
>     call FS receives, regardless of which user it is for, and counter
>     doesnt
>     reset itself after the call is finished.
>
>
> I presume you mean the limit "dialplan app" and not API. In any case, 
> if you have the default configs then locate the Local_Extension in 
> conf/dialplan/default.xml. Add this line right after the condition:
>
> <action application="limit" data="hash ${domain} $1 1 
> handle_over_limit XML over_limit_actions"/>
>
> Then add this new file as "limits.xml" in conf/dialplan/ :
>
> <include>
> <context name="over_limit_actions">
> <extension name="oops, too many calls for this one">
> <condition field="destination_number" expression="handle_over_limit">
> <action application="answer"/>
> <action application="playback" data="ivr/ivr-no_no_no.wav"/>
> <action application="playback" data="ivr/ivr-no_no_no.wav"/>
> <action application="playback" data="ivr/ivr-no_no_no.wav"/>
> <action application="hangup" data="USER_BUSY"/>
> </condition>
> </extension>
> </context>
> </include>
>
> Save, then do F6 or reloadxml. Now when you call a local extension it 
> won't allow more than one call. Note that you can change the value in 
> the limit's data argument. For example, this would cause a limit of 4 
> concurrent calls, sending the 5th call into "oops, too many calls" 
> extension:
>
> <action application="limit" data="hash ${domain} $1 4 
> handle_over_limit XML over_limit_actions"/>
>
> The limit application handles everything for you. Note that this only 
> covers calls made *to* the extension. If you want to handle calls made 
> *from* the extension you will need to add another limit app to the 
> dialplan. Here's an example; put it right after the "global" extension 
> in default.xml:
>
> <!-- set outound caller limit -->
> <extension name="set outbound limit" continue="true">
> <condition field="destination_number" expression="^1?\d{7}" 
> break="on-false"/>
> <condition field="caller_id_number" expression="^(10[01][0-9])">
> <action application="limit" data="hash ${domain} $1 
> 4 handle_over_limit XML over_limit_actions"/>
> <action application="log" data="INFO Added limit for caller $1"/>
> </condition>
> </extension>
> Again, save, press F6 or do reloadxml. Now when this person makes a 
> call to a 7-digit number or does 1+ 7 or more digits, it will add to 
> his limit totals. However, if he just calls another 4-digit extension 
> on the system then it won't add to his limit. (Adjust this as you see 
> fit.)
>
> Let me know if this works for you (it worked for me in my little lab 
> setting) and I will throw it up on the wiki and maybe add it to the 
> cookbook.
>
> -MC
>
>
>
>
> _______________________________________________
> 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 msc at freeswitch.org  Thu Jul 14 23:54:32 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Jul 2011 12:54:32 -0700
Subject: [Freeswitch-users] please help!!! how to limit each FS
 registered user's active calls
In-Reply-To: <4E1F37B4.9050802@gosilverplus.com>
References: <4E1E454A.8040901@gosilverplus.com>
	<CAKzWOxVtNTEptQn6n-1POqiF9ntpLV-SsZeP3oufnSOOpFPoLA@mail.gmail.com>
	<4E1F37B4.9050802@gosilverplus.com>
Message-ID: <CAKzWOxWi39ySot2eGhAnhwrDF2hmts=x1O5AnQEG5fEd0Rmy4g@mail.gmail.com>

You may be on an older freeswitch version. Type the command "version" at
fs_cli and see what it says.

Also, if you want *only* to limit calls made to user 11 then you need to
have a separate extension that handles just the limit for 11. Instead of
trying to do the limit in the same extension where you do the bridge, just
create a separate extension and put it near the top of your dialplan:

<extension name="limit user 11" continue="true">
  <condition field="destination_number" expression="^(11)$">
        <action application="limit" data="db ${domain} $1 1
handle_over_limit XML over_limit_actions"/>
  </condition>
</extension>

Take the limit app out of the "PBX extension" and have just the bridge app
and anything else you normally do there.

Also, you can put the "handle_over_limit" extension in your same context as
the rest of your stuff, but personally I like to have special stuff like
that in its own context so that the dialplan doesn't have to evaluate so
many extensions.

-MC

On Thu, Jul 14, 2011 at 11:38 AM, ran zhang <rzhang at gosilverplus.com> wrote:

> **
> Mike:
>
>     I have added 'limits.xml' as u said.   If i use 'hash', FS generates
> error "limit subsystem hash not found!", so i changed to 'db', following is
> my local extension in my dialplan, i changed '$1' to '11' since i just want
> to limit user/11, but its still limiting incoming calls on all the
> extensions.
>
>
>    Also, is creating 'limits.xml' required?
>
>    Would it work if i can just add 'handle_over_limit' extension in
> 'default.xml', omit the 'XML over_limit_actions' part in the limit action?
>
>
>
>
> <extension name="PBX Extension">
>       <condition field="destination_number" expression="^(1[0-9]{1})$">
>
>         <action application="limit" data="db ${domain} 11 1
> handle_over_limit XML over_limit_actions"/>
>         <action application="bridge" data="user/$1"/>
>
>       </condition>
> </extension>
> On 7/13/2011 11:22 PM, Michael Collins wrote:
>
>
>
> On Wed, Jul 13, 2011 at 6:24 PM, ran zhang <rzhang at gosilverplus.com>wrote:
>
>> hi all:
>>
>>            I'm trying to limit each registered user's active calls, so
>> if that particular registered user pass the limit, FS stops sending any
>> new incoming call to it and it can't make any new calls.
>>
>>            I looked into 'limit' api, the counter increment everytime a
>> call FS receives, regardless of which user it is for, and counter doesnt
>> reset itself after the call is finished.
>>
>
>  I presume you mean the limit "dialplan app" and not API. In any case, if
> you have the default configs then locate the Local_Extension in
> conf/dialplan/default.xml. Add this line right after the condition:
>
>  <action application="limit" data="hash ${domain} $1 1 handle_over_limit
> XML over_limit_actions"/>
>
>  Then add this new file as "limits.xml" in conf/dialplan/ :
>
>  <include>
>
>
>   <context name="over_limit_actions">
>
>
>     <extension name="oops, too many calls for this one">
>
>
>       <condition field="destination_number" expression="handle_over_limit">
>
>
>         <action application="answer"/>
>
>
>         <action application="playback" data="ivr/ivr-no_no_no.wav"/>
>
>
>         <action application="playback" data="ivr/ivr-no_no_no.wav"/>
>
>
>         <action application="playback" data="ivr/ivr-no_no_no.wav"/>
>
>
>         <action application="hangup" data="USER_BUSY"/>
>
>
>       </condition>
>
>
>     </extension>
>
>
>   </context>
>
>
> </include>
>
>  Save, then do F6 or reloadxml. Now when you call a local extension it
> won't allow more than one call. Note that you can change the value in the
> limit's data argument. For example, this would cause a limit of 4 concurrent
> calls, sending the 5th call into "oops, too many calls" extension:
>
>  <action application="limit" data="hash ${domain} $1 4 handle_over_limit
> XML over_limit_actions"/>
>
>  The limit application handles everything for you. Note that this only
> covers calls made *to* the extension. If you want to handle calls made
> *from* the extension you will need to add another limit app to the dialplan.
> Here's an example; put it right after the "global" extension in default.xml:
>
>      <!-- set outound caller limit -->
>
>
>     <extension name="set outbound limit" continue="true">
>
>
>       <condition field="destination_number" expression="^1?\d{7}"
> break="on-false"/>
>
>       <condition field="caller_id_number" expression="^(10[01][0-9])">
>
>
>         <action application="limit" data="hash ${domain} $1
> 4 handle_over_limit XML over_limit_actions"/>
>         <action application="log" data="INFO Added limit for caller $1"/>
>
>
>       </condition>
>
>
>     </extension>
>
>
>
>  Again, save, press F6 or do reloadxml. Now when this person makes a call
> to a 7-digit number or does 1+ 7 or more digits, it will add to his limit
> totals. However, if he just calls another 4-digit extension on the system
> then it won't add to his limit. (Adjust this as you see fit.)
>
>  Let me know if this works for you (it worked for me in my little lab
> setting) and I will throw it up on the wiki and maybe add it to the
> cookbook.
>
>  -MC
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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
>
>
>
> _______________________________________________
> 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 steveayre at gmail.com  Thu Jul 14 23:57:14 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 14 Jul 2011 20:57:14 +0100
Subject: [Freeswitch-users] please help!!! how to limit each FS
 registered user's active calls
In-Reply-To: <4E1F37B4.9050802@gosilverplus.com>
References: <4E1E454A.8040901@gosilverplus.com>
	<CAKzWOxVtNTEptQn6n-1POqiF9ntpLV-SsZeP3oufnSOOpFPoLA@mail.gmail.com>
	<4E1F37B4.9050802@gosilverplus.com>
Message-ID: <CAFiqYukke6By+ugg7PYkN=DcxUWZFQLP_JJt7Xz9hO5WgkkjfQ@mail.gmail.com>

"limit subsystem hash not found!" means you have not loaded mod_hash

As for changing the limit line from $1 to 11 what you've done is any
calls to other numbers such as 05 are also registered as a call to 11.
Therefore a call to 05 will prevent a call to 11.

If you want to handle 11 differently from the others, you'll need to
put it in a different extension:

<extension name="PBX Extension 11">
    <condition field="destination_number" expression="^(11)$">
        <action application="limit" data="db ${domain} $1 1
handle_over_limit XML over_limit_actions"/>
        <action application="bridge" data="user/$1"/>
    </condition>
</extension>
<extension name="PBX Extension">
    <condition field="destination_number" expression="^(1[0-9]{1})$">
        <action application="bridge" data="user/$1"/>
    </condition>
</extension>

-Steve



On 14 July 2011 19:38, ran zhang <rzhang at gosilverplus.com> wrote:
> Mike:
>
> ??? I have added 'limits.xml' as u said.?? If i use 'hash', FS generates
> error "limit subsystem hash not found!", so i changed to 'db', following is
> my local extension in my dialplan, i changed '$1' to '11' since i just want
> to limit user/11, but its still limiting incoming calls on all the
> extensions.
>
>
> ?? Also, is creating 'limits.xml' required?
>
> ?? Would it work if i can just add 'handle_over_limit' extension in
> 'default.xml', omit the 'XML over_limit_actions' part in the limit action?
>
>
>
>
> <extension name="PBX Extension">
> ????? <condition field="destination_number" expression="^(1[0-9]{1})$">
>
> ??????? <action application="limit" data="db ${domain} 11 1
> handle_over_limit XML over_limit_actions"/>
> ??????? <action application="bridge" data="user/$1"/>
>
> ????? </condition>
> </extension>
> On 7/13/2011 11:22 PM, Michael Collins wrote:
>
> On Wed, Jul 13, 2011 at 6:24 PM, ran zhang <rzhang at gosilverplus.com> wrote:
>>
>> hi all:
>>
>> ? ? ? ? ? ?I'm trying to limit each registered user's active calls, so
>> if that particular registered user pass the limit, FS stops sending any
>> new incoming call to it and it can't make any new calls.
>>
>> ? ? ? ? ? ?I looked into 'limit' api, the counter increment everytime a
>> call FS receives, regardless of which user it is for, and counter doesnt
>> reset itself after the call is finished.
>
> I presume you mean the limit "dialplan app" and not API. In any case, if you
> have the default configs then locate the Local_Extension in
> conf/dialplan/default.xml. Add this line right after the condition:
> <action application="limit"?data="hash ${domain} $1 1 handle_over_limit XML
> over_limit_actions"/>
> Then add this new file as "limits.xml" in conf/dialplan/ :
> <include>
>
>
> ? <context name="over_limit_actions">
>
>
> ? ? <extension name="oops, too many calls for this one">
>
>
> ? ? ? <condition field="destination_number" expression="handle_over_limit">
>
>
> ? ? ? ? <action application="answer"/>
>
>
> ? ? ? ? <action application="playback" data="ivr/ivr-no_no_no.wav"/>
>
>
> ? ? ? ? <action application="playback" data="ivr/ivr-no_no_no.wav"/>
>
>
> ? ? ? ? <action application="playback" data="ivr/ivr-no_no_no.wav"/>
>
>
> ? ? ? ? <action application="hangup" data="USER_BUSY"/>
>
>
> ? ? ? </condition>
>
>
> ? ? </extension>
>
>
> ? </context>
>
>
> </include>
> Save, then do F6 or reloadxml. Now when you call a local extension it won't
> allow more than one call. Note that you can change the value in the limit's
> data argument. For example, this would cause a limit of 4 concurrent calls,
> sending the 5th call into "oops, too many calls" extension:
> <action application="limit"?data="hash ${domain} $1 4 handle_over_limit XML
> over_limit_actions"/>
> The limit application handles everything for you. Note that this only covers
> calls made *to* the extension. If you want to handle calls made *from* the
> extension you will need to add another limit app to the dialplan. Here's an
> example; put it right after the "global" extension in default.xml:
> ? ? <!-- set outound caller limit -->
>
>
> ? ? <extension name="set outbound limit" continue="true">
>
>
> ? ? ? <condition field="destination_number" expression="^1?\d{7}"
> break="on-false"/>
>
> ? ? ? <condition field="caller_id_number" expression="^(10[01][0-9])">
>
>
> ? ? ? ? <action application="limit" data="hash ${domain} $1
> 4?handle_over_limit XML over_limit_actions"/>
> ? ? ? ? <action application="log" data="INFO Added limit for caller $1"/>
>
>
> ? ? ? </condition>
>
>
> ? ? </extension>
>
>
>
> Again, save, press F6 or do reloadxml. Now when this person makes a call to
> a 7-digit number or does 1+ 7 or more digits, it will add to his limit
> totals. However, if he just calls another 4-digit extension on the system
> then it won't add to his limit. (Adjust this as you see fit.)
> Let me know if this works for you (it worked for me in my little lab
> setting) and I will throw it up on the wiki and maybe add it to the
> cookbook.
> -MC
>
>
>
> _______________________________________________
> 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
>
>
> _______________________________________________
> 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 Duane.Gilbert at patlive.com  Fri Jul 15 02:29:40 2011
From: Duane.Gilbert at patlive.com (Duane Gilbert)
Date: Thu, 14 Jul 2011 18:29:40 -0400
Subject: [Freeswitch-users] VAD Detection on Outbound Call
In-Reply-To: <4E15E832.3030803@gmail.com>
References: <CAKzWOxXgDVPWdQH90y9OQz0kJP8UMT_MeKfhWWki78cui-WHrA@mail.gmail.com>
	<4E15E832.3030803@gmail.com>
Message-ID: <BD87D3B1554F614190EB6E54CE95DDB604F30BCA@mail2.patlive.local>

Is there a way either using the Dialplan or Lua script to detect when
Voice Activity(VAD) is present on an outbound call?

 

I am using the group_confirm_key/group_confirm_file variables to prompt
the user after the outbound call is answered, but it would be nice to
not start the prompt until after Voice Activity is detected on the line.

 

- Duane

 

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From mi.ke at null.net  Fri Jul 15 02:31:23 2011
From: mi.ke at null.net (Mi Ke)
Date: Thu, 14 Jul 2011 22:31:23 +0000
Subject: [Freeswitch-users] non-blocking bridge in Lua
Message-ID: <20110714223123.232360@gmx.com>

Hi all !

 I want to make a Lua app which will start with Leg A, collect input, initiate outbound Leg B and when it answers, periodically play decreasing series of beeps to Leg A depending on how many minutes left to disconnect (remaining time queried via odbc). 

 My initial idea was to issue a non-blocking bridge command after Leg B session originate returns ring-ready and then to do the rest in while loop, but the bridge called by api:executeString("bgapi bridge legB_session legA_session") just returns Job UUID without actually bridging sessions. A usual freeswitch.bridge call blocks further script execution until unbridged, disallowing me to check Leg B states when it's active.

 Is there any elegant way to acheive my goals using Lua/Dialplan apps ? To create an app starting with Leg A and constantly watching for Leg B existense/states, playing corresponding feedback messages to Leg A ?

 Thanks in advance for yours hints/responses

 Mike
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From avi at avimarcus.net  Fri Jul 15 02:48:05 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 15 Jul 2011 01:48:05 +0300
Subject: [Freeswitch-users] non-blocking bridge in Lua
In-Reply-To: <20110714223123.232360@gmx.com>
References: <20110714223123.232360@gmx.com>
Message-ID: <CALNZuwnsMh-W5V+NEFHXy=T_W5M5Dn2GSz7_OXHkG73oDUwDLw@mail.gmail.com>

Some options:
You can set an execute_on_answer on the leg B to run your new lua script
http://wiki.freeswitch.org/wiki/Variable_execute_on_answer

<http://wiki.freeswitch.org/wiki/Variable_execute_on_answer>Or, you can
schedule all this stuff in the originate string, with either
execute_on_answer or export nolocal:api_on_answer=sched_transfer or the
like.

-Avi Marcus


On Fri, Jul 15, 2011 at 1:31 AM, Mi Ke <mi.ke at null.net> wrote:

>  Hi all !
>
> I want to make a Lua app which will start with Leg A, collect input,
> initiate outbound Leg B and when it answers, periodically play decreasing
> series of beeps to Leg A depending on how many minutes left to disconnect
> (remaining time queried via odbc).
>
> My initial idea was to issue a non-blocking bridge command after Leg B
> session originate returns ring-ready and then to do the rest in while loop,
> but the bridge called by api:executeString("bgapi bridge legB_session
> legA_session") just returns Job UUID without actually bridging sessions. A
> usual freeswitch.bridge call blocks further script execution until
> unbridged, disallowing me to check Leg B states when it's active.
>
> Is there any elegant way to acheive my goals using Lua/Dialplan apps ? To
> create an app starting with Leg A and constantly watching for Leg B
> existense/states, playing corresponding feedback messages to Leg A ?
>
> Thanks in advance for yours hints/responses
>
> Mike
>
>
> _______________________________________________
> 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 msc at freeswitch.org  Fri Jul 15 03:33:51 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Jul 2011 16:33:51 -0700
Subject: [Freeswitch-users] VAD Detection on Outbound Call
In-Reply-To: <BD87D3B1554F614190EB6E54CE95DDB604F30BCA@mail2.patlive.local>
References: <CAKzWOxXgDVPWdQH90y9OQz0kJP8UMT_MeKfhWWki78cui-WHrA@mail.gmail.com>
	<4E15E832.3030803@gmail.com>
	<BD87D3B1554F614190EB6E54CE95DDB604F30BCA@mail2.patlive.local>
Message-ID: <CAKzWOxXoX_SRLQJ_H_-QVW5a=M_y0ge0R7exPS5WO1yEFu8Dew@mail.gmail.com>

So you don't want to start the prompt at 500ms after you hear the answer,
but instead, until you hear some voice activity? Just curious - what problem
does that solve?

-MC

On Thu, Jul 14, 2011 at 3:29 PM, Duane Gilbert <Duane.Gilbert at patlive.com>wrote:

> Is there a way either using the Dialplan or Lua script to detect when Voice
> Activity(VAD) is present on an outbound call?****
>
> ** **
>
> I am using the group_confirm_key/group_confirm_file variables to prompt the
> user after the outbound call is answered, but it would be nice to not start
> the prompt until after Voice Activity is detected on the line.****
>
> ** **
>
> - Duane****
>
> ** **
>
> _______________________________________________
> 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 dujinfang at gmail.com  Fri Jul 15 04:04:30 2011
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 15 Jul 2011 08:04:30 +0800
Subject: [Freeswitch-users] rtmp_url folder
In-Reply-To: <4E1E03F8.7050506@livecall.com>
References: <4E1E03F8.7050506@livecall.com>
Message-ID: <DBE14866595D4F2497A94ED20649B448@gmail.com>

I think point to /phone as in the example should be ok

-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn
Sent with Sparrow (http://www.sparrowmailapp.com)

On Thursday, July 14, 2011 at 4:45 AM, Jack wrote:

> Freeswitch.html has a spot to set the variable rtmp_url. What should 
> be in the folder it points to?
> Thanks,
>  Jack
> 
> 
> _______________________________________________
> 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 (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 vedran.zeljeznak at gmail.com  Fri Jul 15 09:56:01 2011
From: vedran.zeljeznak at gmail.com (Vedran Zeljeznak)
Date: Fri, 15 Jul 2011 07:56:01 +0200
Subject: [Freeswitch-users] No contact header on 302 message
In-Reply-To: <CALsBW=rP+h031N3jOTuO=EjpY8xjvf0FPD0mMA4uiiurvg+s7w@mail.gmail.com>
References: <CALsBW=rP+h031N3jOTuO=EjpY8xjvf0FPD0mMA4uiiurvg+s7w@mail.gmail.com>
Message-ID: <CALsBW=o6L8JCfnjjh-y__q0KTrYxEU_f8Trw_pYuZ=zQD+AZ2g@mail.gmail.com>

i've also tried to set "manual-redirect" parameter to true in my
internal sip profile but FreeSwitch behaved the same as before setting
this param.

Has anybody seen this kind of behavior with "SIP/2.0 302 Moved
Temporarily" message?

---
Vedran Zeljeznak



On Thu, Jul 14, 2011 at 6:47 PM, Vedran Zeljeznak
<vedran.zeljeznak at gmail.com> wrote:
> hello,
>
> i'm having trouble with 302 messages, it looks like FreeSwitch is not
> setting Contact header on Leg-A 302 message when 302 message occurs on
> Leg-B.
>
> There is a valid Contact header in 302 message on Leg-B. I've posted
> FreeSwitch console output on: http://pastebin.freeswitch.org/16806.
>
> Does anybody know how can I tell FreeSwitch to set Contact header on
> Leg-A 302 message from Contact header in 302 message from Leg-B?
>
> ---
> Vedran Zeljeznak
>


From jaybinks at gmail.com  Fri Jul 15 10:54:06 2011
From: jaybinks at gmail.com (jay binks)
Date: Fri, 15 Jul 2011 16:54:06 +1000
Subject: [Freeswitch-users] stripping sip-rh headers at the edge of my
	network
Message-ID: <CAO47HpdG8qJeH4uoXDFB31Jf7_1oNVbmoJQ52DUS6WcmxuAj=Q@mail.gmail.com>

Inside my network I use sip_ph , sip_rh & sip_h headers to pass data around
between boxes.

however I need to strip these off at the edge ( SBC's running Freeswitch
also )

will unset work with these headers in the 200 OK ( and other sip messages )
?
if not, how can I achieve this ?

-- 
Sincerely

Jay
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From anthony.minessale at gmail.com  Fri Jul 15 12:20:24 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jul 2011 03:20:24 -0500
Subject: [Freeswitch-users] Same peace of code works with Javascript and
 not LUA
In-Reply-To: <4E1F3165.70403@tagnet.ru>
References: <4E1F230A.9080104@tagnet.ru> <4E1F24F6.5080101@tagnet.ru>
	<4E1F29BF.2030209@tagnet.ru>
	<CAKzWOxV_mtSxRR0XULjY6QC92wXnnrHL9eq-ZyPMeBoKX9eVrw@mail.gmail.com>
	<4E1F3165.70403@tagnet.ru>
Message-ID: <CAKbxfG8y+hgJ5kAzKpWGeeozOx0j+j=Zzd3AWGhCBVto9_FRHw@mail.gmail.com>

after you create it

session:setAutoHangup(0)

or it will hangup when the script exits

On Thu, Jul 14, 2011 at 1:11 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ??? Here it is http://pastebin.freeswitch.org/16809
> ??? Hope somebody help
>
> get a console log w/ siptrace of that scenario and put it on pastebin. the
> gang here will take a look.
> -MC
>
> On Thu, Jul 14, 2011 at 10:39 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
>>
>> And another problem found. If the remote party answers immediately all
>> is ok. But if remote waits more then 8-10 seconds before answer - the
>> call is dropped when answered. Can't understand where to look for a
>> problem.
>> > Hello!
>> >
>> > ? ? ? I found a problem. originate_session must not to be local.
>> >> Hello!
>> >>
>> >> ? ? ? ?I'm trying to migrate DISA from Javascript to LUA. This peace of
>> >> code works fine in Javascript:
>> >>
>> >> ostr = "{ignore_early_media=true" +
>> >> ? ? ? ? ? ? ? ",origination_caller_id_number=" +
>> >> session.getVariable("caller_id_number") +
>> >> ? ? ? ? ? ? ? ? ? ? ? ?"}sofia/ipbx/50004#" + digits +
>> >> "@192.168.1.1:5060";
>> >> osession = new Session( ostr );
>> >> bridge(session, osession);
>> >> osession.hangup();
>> >>
>> >> With LUA not:
>> >> ? ? ? ? ? ?originate_string =
>> >> ? ? ? ? ? ? ? ?"{ignore_early_media=true" ..
>> >> ? ? ? ? ? ? ? ?",origination_caller_id_number=" ..
>> >> session:getVariable("caller_id_number") ..
>> >> ? ? ? ? ? ? ? ?"}" ..
>> >> ? ? ? ? ? ? ? ?"sofia/ipbx/50004#" ..
>> >> ? ? ? ? ? ? ? ?digits .. "@192.168.1.1:5060";
>> >> ? ? ? ? ? ?originate_session = freeswitch.Session( originate_string );
>> >> ? ? ? ? ? ? ? ?freeswitch.bridge(session, originate_session);
>> >> ? ? ? ? ? ? ? ?originate_session:hangup();
>> >>
>> >> The call is droped when remote answers. There is an error in log:
>> >> bridge: session not ready. I tried to use
>> >> ? ? ? ?if( originate_session:ready() ) then
>> >> ? ? ? ? ? ? ? ?freeswitch.bridge(session, originate_session);
>> >> ? ? ? ? ? ? ? ?originate_session:hangup();
>> >> ? ? ? end
>> >>
>> >> with no luck :(. Please, help me. What am I doing wrong?
>> >>
>> >
>>
>>
>> --
>> Regards,
>> Boris
>>
>>
>>
>> _______________________________________________
>> 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
>
>
>
> _______________________________________________
> 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
>
>
> --
> Regards,
> Boris
>
>
> _______________________________________________
> 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
>
>



-- 
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 manjiri05_deshpande at yahoo.co.in  Fri Jul 15 13:35:18 2011
From: manjiri05_deshpande at yahoo.co.in (Manjiri Deshpande)
Date: Fri, 15 Jul 2011 15:05:18 +0530 (IST)
Subject: [Freeswitch-users] FreeSwitch does not send HOLD Reinvite on other
	leg.
Message-ID: <1310722518.24590.YahooMailNeo@web95905.mail.in.yahoo.com>

Scenario:

A -> FS -> B

A puts the calls on hold ( ie sending a reINVITE with media ip 0.0.0.0 ).

FS does not send this re INVITE to leg B.Leg B continues to send RTP??to FS and call disconnects sometime afterwards if call is not removed from hold. Disconnect times have been as short as 3 minutes and as long as 30 minutes. 
?
Test calls have included background noise from Leg B?and seen noise in trace as normal RTP packets. Call does not seem to disconnect if this background noise is present. Once background noise disappears from Leg B ,FreeSwitch call will disconnect.
?
Why Leg B sends RTP packets sometimes ,so that call is not getting disconnected before 30 mins.
Also If RTP packets are not there,then also call should not be disconnected before 30 mins.
?
Thanks,
Manjiri
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From boris at tagnet.ru  Fri Jul 15 13:42:02 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 15 Jul 2011 15:42:02 +0600
Subject: [Freeswitch-users] Same peace of code works with Javascript and
 not LUA
In-Reply-To: <CAKbxfG8y+hgJ5kAzKpWGeeozOx0j+j=Zzd3AWGhCBVto9_FRHw@mail.gmail.com>
References: <4E1F230A.9080104@tagnet.ru> <4E1F24F6.5080101@tagnet.ru>
	<4E1F29BF.2030209@tagnet.ru>
	<CAKzWOxV_mtSxRR0XULjY6QC92wXnnrHL9eq-ZyPMeBoKX9eVrw@mail.gmail.com>
	<4E1F3165.70403@tagnet.ru>
	<CAKbxfG8y+hgJ5kAzKpWGeeozOx0j+j=Zzd3AWGhCBVto9_FRHw@mail.gmail.com>
Message-ID: <4E200B6A.1060705@tagnet.ru>

Hello!

     I tried, no success.

> after you create it
>
> session:setAutoHangup(0)
>
> or it will hangup when the script exits
>
> On Thu, Jul 14, 2011 at 1:11 PM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> Hello!
>>
>>      Here it is http://pastebin.freeswitch.org/16809
>>      Hope somebody help
>>
>> get a console log w/ siptrace of that scenario and put it on pastebin. the
>> gang here will take a look.
>> -MC
>>
>> On Thu, Jul 14, 2011 at 10:39 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>>> And another problem found. If the remote party answers immediately all
>>> is ok. But if remote waits more then 8-10 seconds before answer - the
>>> call is dropped when answered. Can't understand where to look for a
>>> problem.
>>>> Hello!
>>>>
>>>>        I found a problem. originate_session must not to be local.
>>>>> Hello!
>>>>>
>>>>>         I'm trying to migrate DISA from Javascript to LUA. This peace of
>>>>> code works fine in Javascript:
>>>>>
>>>>> ostr = "{ignore_early_media=true" +
>>>>>                ",origination_caller_id_number=" +
>>>>> session.getVariable("caller_id_number") +
>>>>>                         "}sofia/ipbx/50004#" + digits +
>>>>> "@192.168.1.1:5060";
>>>>> osession = new Session( ostr );
>>>>> bridge(session, osession);
>>>>> osession.hangup();
>>>>>
>>>>> With LUA not:
>>>>>             originate_string =
>>>>>                 "{ignore_early_media=true" ..
>>>>>                 ",origination_caller_id_number=" ..
>>>>> session:getVariable("caller_id_number") ..
>>>>>                 "}" ..
>>>>>                 "sofia/ipbx/50004#" ..
>>>>>                 digits .. "@192.168.1.1:5060";
>>>>>             originate_session = freeswitch.Session( originate_string );
>>>>>                 freeswitch.bridge(session, originate_session);
>>>>>                 originate_session:hangup();
>>>>>
>>>>> The call is droped when remote answers. There is an error in log:
>>>>> bridge: session not ready. I tried to use
>>>>>         if( originate_session:ready() ) then
>>>>>                 freeswitch.bridge(session, originate_session);
>>>>>                 originate_session:hangup();
>>>>>        end
>>>>>
>>>>> with no luck :(. Please, help me. What am I doing wrong?
>>>>>
>>>
>>> --
>>> Regards,
>>> Boris
>>>
>>>
>>>
>>> _______________________________________________
>>> 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
>>
>>
>> _______________________________________________
>> 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
>>
>>
>> --
>> Regards,
>> Boris
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
>


-- 
Regards,
Boris




From Duane.Gilbert at patlive.com  Fri Jul 15 14:29:44 2011
From: Duane.Gilbert at patlive.com (Duane Gilbert)
Date: Fri, 15 Jul 2011 06:29:44 -0400
Subject: [Freeswitch-users] VAD Detection on Outbound Call
In-Reply-To: <CAKzWOxXoX_SRLQJ_H_-QVW5a=M_y0ge0R7exPS5WO1yEFu8Dew@mail.gmail.com>
References: <CAKzWOxXgDVPWdQH90y9OQz0kJP8UMT_MeKfhWWki78cui-WHrA@mail.gmail.com><4E15E832.3030803@gmail.com><BD87D3B1554F614190EB6E54CE95DDB604F30BCA@mail2.patlive.local>
	<CAKzWOxXoX_SRLQJ_H_-QVW5a=M_y0ge0R7exPS5WO1yEFu8Dew@mail.gmail.com>
Message-ID: <BD87D3B1554F614190EB6E54CE95DDB604F30BCE@mail2.patlive.local>

We are migrating an application from a platform that had enhanced call
analysis builtin(Voice and Answering Machine Detection).  We used the
Voice Detection to determine when play information to the callee about
the inbound caller.  Waiting an arbitrary amount of time after the call
is answered solves the problem in most cases, but we had complaints that
the information was played too fast or not fast enough after the call
was answered.  Waiting for Voice Activity seems to satisfy those
complaints.

 

- Duane

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Thursday, July 14, 2011 7:34 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] VAD Detection on Outbound Call

 

So you don't want to start the prompt at 500ms after you hear the
answer, but instead, until you hear some voice activity? Just curious -
what problem does that solve?

 

-MC

On Thu, Jul 14, 2011 at 3:29 PM, Duane Gilbert
<Duane.Gilbert at patlive.com> wrote:

Is there a way either using the Dialplan or Lua script to detect when
Voice Activity(VAD) is present on an outbound call?

 

I am using the group_confirm_key/group_confirm_file variables to prompt
the user after the outbound call is answered, but it would be nice to
not start the prompt until after Voice Activity is detected on the line.

 

- Duane

 


_______________________________________________
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 david.ponzone at ipeva.fr  Fri Jul 15 15:22:09 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 15 Jul 2011 13:22:09 +0200
Subject: [Freeswitch-users] FreeSwitch does not send HOLD Reinvite on
	other leg.
In-Reply-To: <1310722518.24590.YahooMailNeo@web95905.mail.in.yahoo.com>
References: <1310722518.24590.YahooMailNeo@web95905.mail.in.yahoo.com>
Message-ID: <FF145CE6-6931-49BA-A5E4-2D442E74947E@ipeva.fr>

There are some information that you need to give to the list:

-we need to know what are A and B (remote switches or phones registered locally, ...)
-which element triggers the disconnect ?

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

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

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




Le 15/07/2011 ? 11:35, Manjiri Deshpande a ?crit :

> Scenario:
> A -> FS -> B
> 
> A puts the calls on hold ( ie sending a reINVITE with media ip 0.0.0.0 ).
> 
> FS does not send this re INVITE to leg B.Leg B continues to send RTP  to FS and call disconnects sometime afterwards if call is not removed from hold. Disconnect times have been as short as 3 minutes and as long as 30 minutes.
>  
> Test calls have included background noise from Leg B and seen noise in trace as normal RTP packets. Call does not seem to disconnect if this background noise is present. Once background noise disappears from Leg B ,FreeSwitch call will disconnect.
>  
> Why Leg B sends RTP packets sometimes ,so that call is not getting disconnected before 30 mins.
> Also If RTP packets are not there,then also call should not be disconnected before 30 mins.
>  
> Thanks,
> Manjiri
> _______________________________________________
> 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 member at linkedin.com  Fri Jul 15 18:57:27 2011
From: member at linkedin.com (Steve Butterfield via LinkedIn)
Date: Fri, 15 Jul 2011 14:57:27 +0000 (UTC)
Subject: [Freeswitch-users] Invitation to connect on LinkedIn
Message-ID: <732969226.1337606.1310741847105.JavaMail.app@ela4-bed77.prod>

LinkedIn
------------




    Steve Butterfield requested to add you as a connection on LinkedIn:
  
------------------------------------------

Zohair,

I'd like to add you to my professional network on LinkedIn.

- Steve

Accept invitation from Steve Butterfield
http://www.linkedin.com/e/xbphn8-gq59uq0k-1h/vPtmrrfmcGvVxWv84eLqdl0FlSkzdWO84S6qfAKHOSUhfgvFvWjMOLQ/blk/I1521565314_3/1BpC5vrmRLoRZcjkkZt5YCpnlOt3RApnhMpmdzgmhxrSNBszYPnPgNcPkSdj4Odj59bRZVs4RKmAppbPATczwUczAQdjcLrCBxbOYWrSlI/EML_comm_afe/

View invitation from Steve Butterfield
http://www.linkedin.com/e/xbphn8-gq59uq0k-1h/vPtmrrfmcGvVxWv84eLqdl0FlSkzdWO84S6qfAKHOSUhfgvFvWjMOLQ/blk/I1521565314_3/3dvd34PdjoRcj8RckALqnpPbOYWrSlI/svi/ 
------------------------------------------

DID YOU KNOW you can conduct a more credible and powerful reference check using LinkedIn? Enter the company name and years of employment or the prospective employee to find their colleagues that are also in your network. This provides you with a more balanced set of feedback to evaluate that new hire.
http://www.linkedin.com/e/xbphn8-gq59uq0k-1h/rsr/inv-27/

 
-- 
(c) 2011, LinkedIn Corporation
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From gohar.ahmed at vopium.com  Fri Jul 15 09:21:46 2011
From: gohar.ahmed at vopium.com (Gohar Ahmed)
Date: Fri, 15 Jul 2011 10:21:46 +0500
Subject: [Freeswitch-users] rtmp_url folder
In-Reply-To: <DBE14866595D4F2497A94ED20649B448@gmail.com>
References: <4E1E03F8.7050506@livecall.com>
	<DBE14866595D4F2497A94ED20649B448@gmail.com>
Message-ID: <017c01cc42af$18f45ec0$4add1c40$@ahmed@vopium.com>

It doesn?t matter , you can remove the ?/blabla? part from rtmp://ip.of.server and it?ll still connect. Atleast It worked for me.

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Seven Du
Sent: Friday, July 15, 2011 5:05 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] rtmp_url folder

 

I think point to /phone as in the example should be ok

-- 
Seven Du

About: http://about.me/dujinfang

Blog: http://www.dujinfang.com

Proj:  http://www.freeswitch.org.cn

Sent with Sparrow <http://www.sparrowmailapp.com> 

On Thursday, July 14, 2011 at 4:45 AM, Jack wrote:

Freeswitch.html has a spot to set the variable rtmp_url. What should 
be in the folder it points to?
Thanks,
Jack


_______________________________________________
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 anthony.minessale at gmail.com  Fri Jul 15 19:23:28 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Jul 2011 10:23:28 -0500
Subject: [Freeswitch-users] Same peace of code works with Javascript and
 not LUA
In-Reply-To: <4E200B6A.1060705@tagnet.ru>
References: <4E1F230A.9080104@tagnet.ru> <4E1F24F6.5080101@tagnet.ru>
	<4E1F29BF.2030209@tagnet.ru>
	<CAKzWOxV_mtSxRR0XULjY6QC92wXnnrHL9eq-ZyPMeBoKX9eVrw@mail.gmail.com>
	<4E1F3165.70403@tagnet.ru>
	<CAKbxfG8y+hgJ5kAzKpWGeeozOx0j+j=Zzd3AWGhCBVto9_FRHw@mail.gmail.com>
	<4E200B6A.1060705@tagnet.ru>
Message-ID: <CAKbxfG9CdSLryeCVnpq9L=x+DT2kF-K7xic7mnh2zpGOKqi8+g@mail.gmail.com>

Try harder..
On Jul 15, 2011 4:43 AM, "Boris Kovalenko" <boris at tagnet.ru> wrote:
> Hello!
>
> I tried, no success.
>
>> after you create it
>>
>> session:setAutoHangup(0)
>>
>> or it will hangup when the script exits
>>
>> On Thu, Jul 14, 2011 at 1:11 PM, Boris Kovalenko<boris at tagnet.ru> wrote:
>>> Hello!
>>>
>>> Here it is http://pastebin.freeswitch.org/16809
>>> Hope somebody help
>>>
>>> get a console log w/ siptrace of that scenario and put it on pastebin.
the
>>> gang here will take a look.
>>> -MC
>>>
>>> On Thu, Jul 14, 2011 at 10:39 AM, Boris Kovalenko<boris at tagnet.ru>
wrote:
>>>> And another problem found. If the remote party answers immediately all
>>>> is ok. But if remote waits more then 8-10 seconds before answer - the
>>>> call is dropped when answered. Can't understand where to look for a
>>>> problem.
>>>>> Hello!
>>>>>
>>>>> I found a problem. originate_session must not to be local.
>>>>>> Hello!
>>>>>>
>>>>>> I'm trying to migrate DISA from Javascript to LUA. This peace of
>>>>>> code works fine in Javascript:
>>>>>>
>>>>>> ostr = "{ignore_early_media=true" +
>>>>>> ",origination_caller_id_number=" +
>>>>>> session.getVariable("caller_id_number") +
>>>>>> "}sofia/ipbx/50004#" + digits +
>>>>>> "@192.168.1.1:5060";
>>>>>> osession = new Session( ostr );
>>>>>> bridge(session, osession);
>>>>>> osession.hangup();
>>>>>>
>>>>>> With LUA not:
>>>>>> originate_string =
>>>>>> "{ignore_early_media=true" ..
>>>>>> ",origination_caller_id_number=" ..
>>>>>> session:getVariable("caller_id_number") ..
>>>>>> "}" ..
>>>>>> "sofia/ipbx/50004#" ..
>>>>>> digits .. "@192.168.1.1:5060";
>>>>>> originate_session = freeswitch.Session( originate_string );
>>>>>> freeswitch.bridge(session, originate_session);
>>>>>> originate_session:hangup();
>>>>>>
>>>>>> The call is droped when remote answers. There is an error in log:
>>>>>> bridge: session not ready. I tried to use
>>>>>> if( originate_session:ready() ) then
>>>>>> freeswitch.bridge(session, originate_session);
>>>>>> originate_session:hangup();
>>>>>> end
>>>>>>
>>>>>> with no luck :(. Please, help me. What am I doing wrong?
>>>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Boris
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>> --
>>> Regards,
>>> Boris
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>>
>
>
> --
> Regards,
> Boris
>
>
>
> _______________________________________________
> 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 msc at freeswitch.org  Fri Jul 15 19:27:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Jul 2011 08:27:26 -0700
Subject: [Freeswitch-users] Same peace of code works with Javascript and
 not LUA
In-Reply-To: <4E200B6A.1060705@tagnet.ru>
References: <4E1F230A.9080104@tagnet.ru> <4E1F24F6.5080101@tagnet.ru>
	<4E1F29BF.2030209@tagnet.ru>
	<CAKzWOxV_mtSxRR0XULjY6QC92wXnnrHL9eq-ZyPMeBoKX9eVrw@mail.gmail.com>
	<4E1F3165.70403@tagnet.ru>
	<CAKbxfG8y+hgJ5kAzKpWGeeozOx0j+j=Zzd3AWGhCBVto9_FRHw@mail.gmail.com>
	<4E200B6A.1060705@tagnet.ru>
Message-ID: <CAKzWOxVaUFs6vZ6Vsztn_juuMO5yrgxGZ1ukKa33Ou26jh8k6A@mail.gmail.com>

On Fri, Jul 15, 2011 at 2:42 AM, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
>     I tried, no success.
>
> Maybe you could pastebin your dialplan extension and the Lua script so that
we can see exactly what you are doing.

-MC
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From yungwei at resolvity.com  Fri Jul 15 19:36:00 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Fri, 15 Jul 2011 11:36:00 -0400
Subject: [Freeswitch-users] multiple active grammars supported?
In-Reply-To: <CAKbxfG_hNkafdnh3Opw=nk8xvjG+L6fxQM5vaD20CJLbmzcSnA@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950CBC034F1@VMBX102.ihostexchange.net>
	<33095823FD21DF429B481B5163264B7950CBC03597@VMBX102.ihostexchange.net>
	<CAKbxfG_hNkafdnh3Opw=nk8xvjG+L6fxQM5vaD20CJLbmzcSnA@mail.gmail.com>
Message-ID: <33095823FD21DF429B481B5163264B7950CBC03826@VMBX102.ihostexchange.net>

I checked the source code of the freeswitch I have, and it looks like this patch, http://jira.freeswitch.org/browse/FS-2906, has been merged.

-----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, July 14, 2011 3:31 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] multiple active grammars supported?

We need to merge the change into our main git repo first.


On Wed, Jul 13, 2011 at 3:47 PM, Yungwei Chen <yungwei at resolvity.com> wrote:
> According to http://jira.freeswitch.org/browse/FS-2906, FS already supports multiple active grammars.
> Now I'm wondering how to take advantages of it from a javascript program.
> Thanks.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei Chen
> Sent: Wednesday, July 13, 2011 11:29 AM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] multiple active grammars supported?
>
> Hi,
>
> I tried the pizza demo shipped with freeswitch, and I am wondering if multiple active grammars are supported.
>
> For example:
> question: was this the pizza you had in mind?
> response: yes/no (defined in grammar 1) Or
> ? ? ? ? ?help ? (defined in grammar 2; this case will be handled differently)
>
> Thanks.
>
> _______________________________________________
> 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
>
> _______________________________________________
> 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
>



-- 
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 jeff at jefflenk.com  Fri Jul 15 20:05:53 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 15 Jul 2011 09:05:53 -0700 (PDT)
Subject: [Freeswitch-users] multiple active grammars supported?
In-Reply-To: <33095823FD21DF429B481B5163264B7950CBC03826@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950CBC034F1@VMBX102.ihostexchange.net>
	<33095823FD21DF429B481B5163264B7950CBC03597@VMBX102.ihostexchange.net>
	<CAKbxfG_hNkafdnh3Opw=nk8xvjG+L6fxQM5vaD20CJLbmzcSnA@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950CBC03826@VMBX102.ihostexchange.net>
Message-ID: <1310745953606-6587363.post@n2.nabble.com>

Yes it was merged. You will need to add support for the multiple grammar
implementation to the SpeechTools.jm file. When that code was authored
multiple grammar support was not available.

Please open a Jira with your suggested patch and maybe we can get it
contributed.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/multiple-active-grammars-supported-tp6579722p6587363.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From adam.kelloway at newpace.ca  Fri Jul 15 20:32:38 2011
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Fri, 15 Jul 2011 13:32:38 -0300
Subject: [Freeswitch-users] FS and SIP Proxy
Message-ID: <4E206BA6.3020701@newpace.ca>

Hi there,

I have a FS installation that is on a private network, with a  SIP proxy 
in front of it that has an assigned public IP. I would like to be able 
to make calls from a public network to this FS installation, but I am 
not able to fully establish SIP and RTP sessions. I have been playing 
around with the NAT features of FS without any luck. In this scenario, 
do you think it would be necessary for the SIP proxy to handle the NAT 
traversal to ensure that the SIP responses reach the SIP endpoint? For 
RTP, would the proxy need to modify the SDP response that FS creates, so 
that it includes the external IP address necessary to establish the 
session? What I am getting at is whether the proxy needs to be not a 
proxy at all, but more like some kind of SBC, that also has RTP proxy 
capabilities. It would be necessary to keep this proxy/SBC in the signal 
path for both sessions.

Any guidance on this scenario would be appreciated.

Thanks,

Adam


From jack at livecall.com  Fri Jul 15 20:46:37 2011
From: jack at livecall.com (Jack)
Date: Fri, 15 Jul 2011 09:46:37 -0700
Subject: [Freeswitch-users] rtmp_url folder
In-Reply-To: <017c01cc42af$18f45ec0$4add1c40$@ahmed@vopium.com>
References: <4E1E03F8.7050506@livecall.com>
	<DBE14866595D4F2497A94ED20649B448@gmail.com>
	<017c01cc42af$18f45ec0$4add1c40$@ahmed@vopium.com>
Message-ID: <4E206EED.6060009@livecall.com>

Thanks, it does connect with just the sip ip .  My client shows 
connected, but I am not getting a keypad and nothing happens when I 
click new call or login.???
I am serving the web page from windows iis6 .
I am getting an error in jquery.hotkeys-0.7.9.min.js line 124:

 1. Uncaught TypeError: Cannot read property 'combi' of null
     1.


Has anyone else experienced this and found a solution?



On 7/14/2011 10:21 PM, Gohar Ahmed wrote:
>
> It doesn't matter , you can remove the "/blabla" part from 
> rtmp://ip.of.server and it'll still connect. Atleast It worked for me.
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Seven Du
> *Sent:* Friday, July 15, 2011 5:05 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] rtmp_url folder
>
> I think point to /phone as in the example should be ok
>
> -- 
> Seven Du
>
> About: http://about.me/dujinfang
>
> Blog: http://www.dujinfang.com
>
> Proj: http://www.freeswitch.org.cn
>
> Sent with Sparrow <http://www.sparrowmailapp.com>
>
> On Thursday, July 14, 2011 at 4:45 AM, Jack wrote:
>
>     Freeswitch.html has a spot to set the variable rtmp_url. What should
>     be in the folder it points to?
>     Thanks,
>     Jack
>
>
>     _______________________________________________
>     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
>     <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
>
>
>
> _______________________________________________
> 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 boris at tagnet.ru  Fri Jul 15 21:05:15 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 15 Jul 2011 23:05:15 +0600
Subject: [Freeswitch-users] Same peace of code works with Javascript and
 not LUA
In-Reply-To: <CAKbxfG9CdSLryeCVnpq9L=x+DT2kF-K7xic7mnh2zpGOKqi8+g@mail.gmail.com>
References: <4E1F230A.9080104@tagnet.ru> <4E1F24F6.5080101@tagnet.ru>
	<4E1F29BF.2030209@tagnet.ru>
	<CAKzWOxV_mtSxRR0XULjY6QC92wXnnrHL9eq-ZyPMeBoKX9eVrw@mail.gmail.com>
	<4E1F3165.70403@tagnet.ru>
	<CAKbxfG8y+hgJ5kAzKpWGeeozOx0j+j=Zzd3AWGhCBVto9_FRHw@mail.gmail.com>
	<4E200B6A.1060705@tagnet.ru>
	<CAKbxfG9CdSLryeCVnpq9L=x+DT2kF-K7xic7mnh2zpGOKqi8+g@mail.gmail.com>
Message-ID: <4E20734B.70307@tagnet.ru>

:) Nice joke


Ok. Pastebin: http://pastebin.freeswitch.org/16818

The extension is:
<extension name="ext_office_disa2">
<condition field="destination_number" expression="^(73435230022)$">
<action application="set" data="disa_msg_enterpin=$${base_dir}/sound
s/ru/RU/elena/ivr/8000/ivr-please_enter_pin_followed_by_pound.wav"/>
<action application="set" data="disa_msg_wrongpin=$${base_dir}/sound
s/ru/RU/elena/ivr/8000/ivr-pin_or_extension_is-invalid.wav"/>
<action application="set" data="disa_msg_enterext=$${base_dir}/sound
s/ru/RU/elena/ivr/8000/ivr-enter_ext.wav"/>
<action application="lua" data="common/disa.lua"/>
<!-- <action application="javascript" data="common/disa2.js"/> -->
</condition>
</extension>


LUA script:
--[[
]]

-- if you choose not to require a pin then then you may want to add a 
dialplan c
ondition for a specific caller id
local   pin = "111111"; -- make sure to change the PIN number.

local   digitmaxlength = 0;
local   initialTimeout = 7500;
local   interdigitTimeout = 5000;
local   absoluteTimeout = 30000;

local   msg_enterpin;
local   msg_wrongPin;
local   msg_enterext;

function disa()
     if( not session:ready() ) then
         return
     end


     msg_enterpin = session:getVariable("disa_msg_enterpin");
     msg_wrongpin = session:getVariable("disa_msg_wrongpin");
     msg_enterext = session:getVariable("disa_msg_enterext");

     if( msg_enterpin == nil or msg_wrongpin == nil or msg_enterext == 
nil ) then
         freeswitch.consoleLog("ALERT", "DISA: Please set necessary 
variables");
         return false;
     end

     session:answer();
     session:sleep( 500 );

     -- session:execute("start_dtmf", "");

     if( string.len( pin ) > 0 ) then
         digitmaxlength = 6;

         session:flushDigits();
         digits = session:playAndGetDigits(digitmaxlength, 
digitmaxlength, 1, int
erdigitTimeout, "#", msg_enterpin, msg_wrongpin, "\\d+", absoluteTimeout);
         freeswitch.consoleLog( "info", "DISA.lua pin: " .. digits .. 
"\n" );
     end

     if( digits == pin or string.len( pin ) == 0 ) then
         local   ru_ring = session:getVariable("ru-ring");
         local   originate_string;

         originate_session       = nil;

         session:setVariable("ringback", ru_ring);               -- set 
to ringto
ne
         session:setVariable("transfer_ringback", ru_ring);      -- set 
to ringto
ne
         session:setVariable("hangup_after_bridge", "true");
         session:setVariable("v_numbering_plan", "RU");

         digits = "";    -- clear dtmf digits to prepare for next dtmf 
request
         digitmaxlength = 20;
         session:flushDigits();
         digits = session:playAndGetDigits(6, digitmaxlength, 1, 
interdigitTimeou
t, "#", msg_enterext, "", "\\d+", absoluteTimeout);
         if( string.len( digits ) == 0 ) then
             return false;
         end
         freeswitch.consoleLog( "info", "DISA.lua Collected: " .. digits 
.. "\n"
);
         originate_string =
             "{ignore_early_media=true" ..
             ",origination_caller_id_number=" .. 
session:getVariable("caller_id_n
umber") ..
             ",originate_timeout=90" ..
             -- ",uuid=" .. session:get_uuid() ..
             "}" ..
             "sofia/ipbx/" .. session:getVariable("v_ats_srcport") .. "#" ..
             digits .. "@192.168.1.1:5060";

         originate_session = freeswitch.Session( originate_string, 
session );

         session:setAutoHangup( false );
         originate_session:setAutoHangup( false );

         if( originate_session:ready() ) then

             freeswitch.bridge(session, originate_session);
             -- originate_session:hangup();
         end
     end
end

local   digits = "";

disa();

if( session:ready() ) then
     session:hangup();
end


The extension where originate is placed:
<extension name="ext_domestic_e164">
<condition field="destination_number" expression="^(7[0-9]{10})$">
<action application="log" data="NOTICE [top.ctx] - ext_domestic"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=false"/>
<action application="set" data="failure_causes=USER_BUSY,NO_ANSWER"/>
<action application="set" data="fail_on_single_reject=true"/>
<!-- Least Cost Routing scheme -->
<action application="lcr" data="$1 lcr_tagnet"/>
<action application="set" data="call_timeout=60"/>
<action application="set" data="bridge_answer_timeout=60"/>
<action application="bridge" data="${lcr_auto_route}"/>
<action application="hangup"/>
</condition>
</extension>

> Try harder..
>
> On Jul 15, 2011 4:43 AM, "Boris Kovalenko" <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
> > Hello!
> >
> > I tried, no success.
> >
> >> after you create it
> >>
> >> session:setAutoHangup(0)
> >>
> >> or it will hangup when the script exits
> >>
> >> On Thu, Jul 14, 2011 at 1:11 PM, Boris Kovalenko<boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
> >>> Hello!
> >>>
> >>> Here it is http://pastebin.freeswitch.org/16809
> >>> Hope somebody help
> >>>
> >>> get a console log w/ siptrace of that scenario and put it on 
> pastebin. the
> >>> gang here will take a look.
> >>> -MC
> >>>
> >>> On Thu, Jul 14, 2011 at 10:39 AM, Boris Kovalenko<boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
> >>>> And another problem found. If the remote party answers 
> immediately all
> >>>> is ok. But if remote waits more then 8-10 seconds before answer - the
> >>>> call is dropped when answered. Can't understand where to look for a
> >>>> problem.
> >>>>> Hello!
> >>>>>
> >>>>> I found a problem. originate_session must not to be local.
> >>>>>> Hello!
> >>>>>>
> >>>>>> I'm trying to migrate DISA from Javascript to LUA. This peace of
> >>>>>> code works fine in Javascript:
> >>>>>>
> >>>>>> ostr = "{ignore_early_media=true" +
> >>>>>> ",origination_caller_id_number=" +
> >>>>>> session.getVariable("caller_id_number") +
> >>>>>> "}sofia/ipbx/50004#" + digits +
> >>>>>> "@192.168.1.1:5060 <http://192.168.1.1:5060>";
> >>>>>> osession = new Session( ostr );
> >>>>>> bridge(session, osession);
> >>>>>> osession.hangup();
> >>>>>>
> >>>>>> With LUA not:
> >>>>>> originate_string =
> >>>>>> "{ignore_early_media=true" ..
> >>>>>> ",origination_caller_id_number=" ..
> >>>>>> session:getVariable("caller_id_number") ..
> >>>>>> "}" ..
> >>>>>> "sofia/ipbx/50004#" ..
> >>>>>> digits .. "@192.168.1.1:5060 <http://192.168.1.1:5060>";
> >>>>>> originate_session = freeswitch.Session( originate_string );
> >>>>>> freeswitch.bridge(session, originate_session);
> >>>>>> originate_session:hangup();
> >>>>>>
> >>>>>> The call is droped when remote answers. There is an error in log:
> >>>>>> bridge: session not ready. I tried to use
> >>>>>> if( originate_session:ready() ) then
> >>>>>> freeswitch.bridge(session, originate_session);
> >>>>>> originate_session:hangup();
> >>>>>> end
> >>>>>>
> >>>>>> with no luck :(. Please, help me. What am I doing wrong?
> >>>>>>
> >>>>
> >>>> --
> >>>> Regards,
> >>>> Boris
> >>>>
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> 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 
> <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
> >>>
> >>>
> >>> _______________________________________________
> >>> 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 
> <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
> >>>
> >>>
> >>> --
> >>> Regards,
> >>> Boris
> >>>
> >>>
> >>> _______________________________________________
> >>> 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 
> <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
> >>>
> >>>
> >>
> >>
> >
> >
> > --
> > Regards,
> > Boris
> >
> >
> >
> > _______________________________________________
> > 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 
> <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
>
>
> _______________________________________________
> 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


-- 
Regards,
Boris


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From mi.ke at null.net  Fri Jul 15 21:42:34 2011
From: mi.ke at null.net (Mi Ke)
Date: Fri, 15 Jul 2011 17:42:34 +0000
Subject: [Freeswitch-users] non-blocking bridge in Lua
Message-ID: <20110715174234.232370@gmx.com>

Many thanks, Avi ! The following code worked for me:

 local ob_session = freeswitch.Session("{return_ring_ready=true,execute_on_answer=lua ob_answer.lua}" .. called_id,session)

 if (ob_session:ready()) then
 freeswitch.bridge(session,ob_session)
 else
 --- process failure here
 end

 All channel data related to the bridge and both legs are available from ob_answer.lua via session:getVariable()

 All the best / Mike

----- Original Message -----
From: Avi Marcus
Sent: 07/15/11 01:48 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] non-blocking bridge in Lua

 Some options: You can set an execute_on_answer on the leg B to run your new lua script 
http://wiki.freeswitch.org/wiki/Variable_execute_on_answer 

 Or, you can schedule all this stuff in the originate string, with either execute_on_answer or export nolocal:api_on_answer=sched_transfer or the like.

-Avi Marcus

 On Fri, Jul 15, 2011 at 1:31 AM, Mi Ke <http://mi.ke @http://null.net > wrote:

Hi all !

 I want to make a Lua app which will start with Leg A, collect input, initiate outbound Leg B and when it answers, periodically play decreasing series of beeps to Leg A depending on how many minutes left to disconnect (remaining time queried via odbc). 

 My initial idea was to issue a non-blocking bridge command after Leg B session originate returns ring-ready and then to do the rest in while loop, but the bridge called by api:executeString("bgapi bridge legB_session legA_session") just returns Job UUID without actually bridging sessions. A usual freeswitch.bridge call blocks further script execution until unbridged, disallowing me to check Leg B states when it's active.

 Is there any elegant way to acheive my goals using Lua/Dialplan apps ? To create an app starting with Leg A and constantly watching for Leg B existense/states, playing corresponding feedback messages to Leg A ?

 Thanks in advance for yours hints/responses

 Mike

 _______________________________________________
 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 msc at freeswitch.org  Fri Jul 15 21:54:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Jul 2011 10:54:36 -0700
Subject: [Freeswitch-users] Same peace of code works with Javascript and
 not LUA
In-Reply-To: <4E20734B.70307@tagnet.ru>
References: <4E1F230A.9080104@tagnet.ru> <4E1F24F6.5080101@tagnet.ru>
	<4E1F29BF.2030209@tagnet.ru>
	<CAKzWOxV_mtSxRR0XULjY6QC92wXnnrHL9eq-ZyPMeBoKX9eVrw@mail.gmail.com>
	<4E1F3165.70403@tagnet.ru>
	<CAKbxfG8y+hgJ5kAzKpWGeeozOx0j+j=Zzd3AWGhCBVto9_FRHw@mail.gmail.com>
	<4E200B6A.1060705@tagnet.ru>
	<CAKbxfG9CdSLryeCVnpq9L=x+DT2kF-K7xic7mnh2zpGOKqi8+g@mail.gmail.com>
	<4E20734B.70307@tagnet.ru>
Message-ID: <CAKzWOxWtGvOzEihsph9H2-SMar+L5CkvuZ0rZpEM1dHAdxmhrA@mail.gmail.com>

Okay, turn off the sip trace and turn on debug level logging:

sofia global siptrace off
console loglevel 7

Test call again, pastebin log please.
-MC

On Fri, Jul 15, 2011 at 10:05 AM, Boris Kovalenko <boris at tagnet.ru> wrote:

>  :) Nice joke
>
>
> Ok. Pastebin: http://pastebin.freeswitch.org/16818
>
> The extension is:
>     <extension name="ext_office_disa2">
>         <condition field="destination_number" expression="^(73435230022)$">
>             <action application="set"
> data="disa_msg_enterpin=$${base_dir}/sound
> s/ru/RU/elena/ivr/8000/ivr-please_enter_pin_followed_by_pound.wav"/>
>             <action application="set"
> data="disa_msg_wrongpin=$${base_dir}/sound
> s/ru/RU/elena/ivr/8000/ivr-pin_or_extension_is-invalid.wav"/>
>             <action application="set"
> data="disa_msg_enterext=$${base_dir}/sound
> s/ru/RU/elena/ivr/8000/ivr-enter_ext.wav"/>
>             <action application="lua" data="common/disa.lua"/>
>             <!-- <action application="javascript" data="common/disa2.js"/>
> -->
>         </condition>
>     </extension>
>
>
> LUA script:
> --[[
> ]]
>
> -- if you choose not to require a pin then then you may want to add a
> dialplan c
> ondition for a specific caller id
> local   pin = "111111"; -- make sure to change the PIN number.
>
> local   digitmaxlength = 0;
> local   initialTimeout = 7500;
> local   interdigitTimeout = 5000;
> local   absoluteTimeout = 30000;
>
> local   msg_enterpin;
> local   msg_wrongPin;
> local   msg_enterext;
>
> function disa()
>     if( not session:ready() ) then
>         return
>     end
>
>
>     msg_enterpin = session:getVariable("disa_msg_enterpin");
>     msg_wrongpin = session:getVariable("disa_msg_wrongpin");
>     msg_enterext = session:getVariable("disa_msg_enterext");
>
>     if( msg_enterpin == nil or msg_wrongpin == nil or msg_enterext == nil )
> then
>         freeswitch.consoleLog("ALERT", "DISA: Please set necessary
> variables");
>         return false;
>     end
>
>     session:answer();
>     session:sleep( 500 );
>
>     -- session:execute("start_dtmf", "");
>
>     if( string.len( pin ) > 0 ) then
>         digitmaxlength = 6;
>
>         session:flushDigits();
>         digits = session:playAndGetDigits(digitmaxlength, digitmaxlength,
> 1, int
> erdigitTimeout, "#", msg_enterpin, msg_wrongpin, "\\d+", absoluteTimeout);
>         freeswitch.consoleLog( "info", "DISA.lua pin: " .. digits .. "\n"
> );
>     end
>
>     if( digits == pin or string.len( pin ) == 0 ) then
>         local   ru_ring = session:getVariable("ru-ring");
>         local   originate_string;
>
>         originate_session       = nil;
>
>         session:setVariable("ringback", ru_ring);               -- set to
> ringto
> ne
>         session:setVariable("transfer_ringback", ru_ring);      -- set to
> ringto
> ne
>         session:setVariable("hangup_after_bridge", "true");
>         session:setVariable("v_numbering_plan", "RU");
>
>         digits = "";    -- clear dtmf digits to prepare for next dtmf
> request
>         digitmaxlength = 20;
>         session:flushDigits();
>         digits = session:playAndGetDigits(6, digitmaxlength, 1,
> interdigitTimeou
> t, "#", msg_enterext, "", "\\d+", absoluteTimeout);
>         if( string.len( digits ) == 0 ) then
>             return false;
>         end
>         freeswitch.consoleLog( "info", "DISA.lua Collected: " .. digits ..
> "\n"
> );
>         originate_string =
>             "{ignore_early_media=true" ..
>             ",origination_caller_id_number=" ..
> session:getVariable("caller_id_n
> umber") ..
>             ",originate_timeout=90" ..
>             -- ",uuid=" .. session:get_uuid() ..
>             "}" ..
>             "sofia/ipbx/" .. session:getVariable("v_ats_srcport") .. "#" ..
>             digits .. "@192.168.1.1:5060";
>
>         originate_session = freeswitch.Session( originate_string, session
> );
>
>         session:setAutoHangup( false );
>         originate_session:setAutoHangup( false );
>
>         if( originate_session:ready() ) then
>
>             freeswitch.bridge(session, originate_session);
>             -- originate_session:hangup();
>         end
>     end
> end
>
> local   digits = "";
>
> disa();
>
> if( session:ready() ) then
>     session:hangup();
> end
>
>
> The extension where originate is placed:
>     <extension name="ext_domestic_e164">
>         <condition field="destination_number" expression="^(7[0-9]{10})$">
>             <action application="log" data="NOTICE [top.ctx] -
> ext_domestic"/>
>             <action application="set" data="hangup_after_bridge=true"/>
>             <action application="set" data="continue_on_fail=false"/>
>             <action application="set"
> data="failure_causes=USER_BUSY,NO_ANSWER"/>
>             <action application="set" data="fail_on_single_reject=true"/>
>             <!-- Least Cost Routing scheme -->
>             <action application="lcr" data="$1 lcr_tagnet"/>
>             <action application="set" data="call_timeout=60"/>
>             <action application="set" data="bridge_answer_timeout=60"/>
>             <action application="bridge" data="${lcr_auto_route}"/>
>             <action application="hangup"/>
>         </condition>
>     </extension>
>
>  Try harder..
> On Jul 15, 2011 4:43 AM, "Boris Kovalenko" <boris at tagnet.ru> wrote:
> > Hello!
> >
> > I tried, no success.
> >
> >> after you create it
> >>
> >> session:setAutoHangup(0)
> >>
> >> or it will hangup when the script exits
> >>
> >> On Thu, Jul 14, 2011 at 1:11 PM, Boris Kovalenko<boris at tagnet.ru>
> wrote:
> >>> Hello!
> >>>
> >>> Here it is http://pastebin.freeswitch.org/16809
> >>> Hope somebody help
> >>>
> >>> get a console log w/ siptrace of that scenario and put it on pastebin.
> the
> >>> gang here will take a look.
> >>> -MC
> >>>
> >>> On Thu, Jul 14, 2011 at 10:39 AM, Boris Kovalenko<boris at tagnet.ru>
> wrote:
> >>>> And another problem found. If the remote party answers immediately all
> >>>> is ok. But if remote waits more then 8-10 seconds before answer - the
> >>>> call is dropped when answered. Can't understand where to look for a
> >>>> problem.
> >>>>> Hello!
> >>>>>
> >>>>> I found a problem. originate_session must not to be local.
> >>>>>> Hello!
> >>>>>>
> >>>>>> I'm trying to migrate DISA from Javascript to LUA. This peace of
> >>>>>> code works fine in Javascript:
> >>>>>>
> >>>>>> ostr = "{ignore_early_media=true" +
> >>>>>> ",origination_caller_id_number=" +
> >>>>>> session.getVariable("caller_id_number") +
> >>>>>> "}sofia/ipbx/50004#" + digits +
> >>>>>> "@192.168.1.1:5060";
> >>>>>> osession = new Session( ostr );
> >>>>>> bridge(session, osession);
> >>>>>> osession.hangup();
> >>>>>>
> >>>>>> With LUA not:
> >>>>>> originate_string =
> >>>>>> "{ignore_early_media=true" ..
> >>>>>> ",origination_caller_id_number=" ..
> >>>>>> session:getVariable("caller_id_number") ..
> >>>>>> "}" ..
> >>>>>> "sofia/ipbx/50004#" ..
> >>>>>> digits .. "@192.168.1.1:5060";
> >>>>>> originate_session = freeswitch.Session( originate_string );
> >>>>>> freeswitch.bridge(session, originate_session);
> >>>>>> originate_session:hangup();
> >>>>>>
> >>>>>> The call is droped when remote answers. There is an error in log:
> >>>>>> bridge: session not ready. I tried to use
> >>>>>> if( originate_session:ready() ) then
> >>>>>> freeswitch.bridge(session, originate_session);
> >>>>>> originate_session:hangup();
> >>>>>> end
> >>>>>>
> >>>>>> with no luck :(. Please, help me. What am I doing wrong?
> >>>>>>
> >>>>
> >>>> --
> >>>> Regards,
> >>>> Boris
> >>>>
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> 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
> >>>
> >>>
> >>> _______________________________________________
> >>> 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
> >>>
> >>>
> >>> --
> >>> Regards,
> >>> Boris
> >>>
> >>>
> >>> _______________________________________________
> >>> 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
> >>>
> >>>
> >>
> >>
> >
> >
> > --
> > Regards,
> > Boris
> >
> >
> >
> > _______________________________________________
> > 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
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> 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
>
>
>
> --
> Regards,
> Boris
>
>
>
> _______________________________________________
> 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 anthony.minessale at gmail.com  Sat Jul 16 12:10:35 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 16 Jul 2011 03:10:35 -0500
Subject: [Freeswitch-users] Same peace of code works with Javascript and
 not LUA
In-Reply-To: <CAKzWOxWtGvOzEihsph9H2-SMar+L5CkvuZ0rZpEM1dHAdxmhrA@mail.gmail.com>
References: <4E1F230A.9080104@tagnet.ru> <4E1F24F6.5080101@tagnet.ru>
	<4E1F29BF.2030209@tagnet.ru>
	<CAKzWOxV_mtSxRR0XULjY6QC92wXnnrHL9eq-ZyPMeBoKX9eVrw@mail.gmail.com>
	<4E1F3165.70403@tagnet.ru>
	<CAKbxfG8y+hgJ5kAzKpWGeeozOx0j+j=Zzd3AWGhCBVto9_FRHw@mail.gmail.com>
	<4E200B6A.1060705@tagnet.ru>
	<CAKbxfG9CdSLryeCVnpq9L=x+DT2kF-K7xic7mnh2zpGOKqi8+g@mail.gmail.com>
	<4E20734B.70307@tagnet.ru>
	<CAKzWOxWtGvOzEihsph9H2-SMar+L5CkvuZ0rZpEM1dHAdxmhrA@mail.gmail.com>
Message-ID: <CAKbxfG-vY4=q_1Gb2GrR5=AfJEsrBtzhk4jiqKH8fFSDrzhtHg@mail.gmail.com>

your line numbers indicate you are on an old revision update before
you ask for help

On Fri, Jul 15, 2011 at 12:54 PM, Michael Collins <msc at freeswitch.org> wrote:
> Okay, turn off the sip trace and turn on debug level logging:
> sofia global siptrace off
> console loglevel 7
> Test call again, pastebin log please.
> -MC
>
> On Fri, Jul 15, 2011 at 10:05 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
>>
>> :) Nice joke
>>
>>
>> Ok. Pastebin: http://pastebin.freeswitch.org/16818
>>
>> The extension is:
>> ??? <extension name="ext_office_disa2">
>> ??????? <condition field="destination_number"
>> expression="^(73435230022)$">
>> ??????????? <action application="set"
>> data="disa_msg_enterpin=$${base_dir}/sound
>> s/ru/RU/elena/ivr/8000/ivr-please_enter_pin_followed_by_pound.wav"/>
>> ??????????? <action application="set"
>> data="disa_msg_wrongpin=$${base_dir}/sound
>> s/ru/RU/elena/ivr/8000/ivr-pin_or_extension_is-invalid.wav"/>
>> ??????????? <action application="set"
>> data="disa_msg_enterext=$${base_dir}/sound
>> s/ru/RU/elena/ivr/8000/ivr-enter_ext.wav"/>
>> ??????????? <action application="lua" data="common/disa.lua"/>
>> ??????????? <!-- <action application="javascript" data="common/disa2.js"/>
>> -->
>> ??????? </condition>
>> ??? </extension>
>>
>>
>> LUA script:
>> --[[
>> ]]
>>
>> -- if you choose not to require a pin then then you may want to add a
>> dialplan c
>> ondition for a specific caller id
>> local?? pin = "111111"; -- make sure to change the PIN number.
>>
>> local?? digitmaxlength = 0;
>> local?? initialTimeout = 7500;
>> local?? interdigitTimeout = 5000;
>> local?? absoluteTimeout = 30000;
>>
>> local?? msg_enterpin;
>> local?? msg_wrongPin;
>> local?? msg_enterext;
>>
>> function disa()
>> ??? if( not session:ready() ) then
>> ??????? return
>> ??? end
>>
>>
>> ??? msg_enterpin = session:getVariable("disa_msg_enterpin");
>> ??? msg_wrongpin = session:getVariable("disa_msg_wrongpin");
>> ??? msg_enterext = session:getVariable("disa_msg_enterext");
>>
>> ??? if( msg_enterpin == nil or msg_wrongpin == nil or msg_enterext == nil
>> ) then
>> ??????? freeswitch.consoleLog("ALERT", "DISA: Please set necessary
>> variables");
>> ??????? return false;
>> ??? end
>>
>> ??? session:answer();
>> ??? session:sleep( 500 );
>>
>> ??? -- session:execute("start_dtmf", "");
>>
>> ??? if( string.len( pin ) > 0 ) then
>> ??????? digitmaxlength = 6;
>>
>> ??????? session:flushDigits();
>> ??????? digits = session:playAndGetDigits(digitmaxlength, digitmaxlength,
>> 1, int
>> erdigitTimeout, "#", msg_enterpin, msg_wrongpin, "\\d+", absoluteTimeout);
>> ??????? freeswitch.consoleLog( "info", "DISA.lua pin: " .. digits .. "\n"
>> );
>> ??? end
>>
>> ??? if( digits == pin or string.len( pin ) == 0 ) then
>> ??????? local?? ru_ring = session:getVariable("ru-ring");
>> ??????? local?? originate_string;
>>
>> ??????? originate_session?????? = nil;
>>
>> ??????? session:setVariable("ringback", ru_ring);?????????????? -- set to
>> ringto
>> ne
>> ??????? session:setVariable("transfer_ringback", ru_ring);????? -- set to
>> ringto
>> ne
>> ??????? session:setVariable("hangup_after_bridge", "true");
>> ??????? session:setVariable("v_numbering_plan", "RU");
>>
>> ??????? digits = "";??? -- clear dtmf digits to prepare for next dtmf
>> request
>> ??????? digitmaxlength = 20;
>> ??????? session:flushDigits();
>> ??????? digits = session:playAndGetDigits(6, digitmaxlength, 1,
>> interdigitTimeou
>> t, "#", msg_enterext, "", "\\d+", absoluteTimeout);
>> ??????? if( string.len( digits ) == 0 ) then
>> ??????????? return false;
>> ??????? end
>> ??????? freeswitch.consoleLog( "info", "DISA.lua Collected: " .. digits ..
>> "\n"
>> );
>> ??????? originate_string =
>> ??????????? "{ignore_early_media=true" ..
>> ??????????? ",origination_caller_id_number=" ..
>> session:getVariable("caller_id_n
>> umber") ..
>> ??????????? ",originate_timeout=90" ..
>> ??????????? -- ",uuid=" .. session:get_uuid() ..
>> ??????????? "}" ..
>> ??????????? "sofia/ipbx/" .. session:getVariable("v_ats_srcport") .. "#"
>> ..
>> ??????????? digits .. "@192.168.1.1:5060";
>>
>> ??????? originate_session = freeswitch.Session( originate_string, session
>> );
>>
>> ??????? session:setAutoHangup( false );
>> ??????? originate_session:setAutoHangup( false );
>>
>> ??????? if( originate_session:ready() ) then
>>
>> ??????????? freeswitch.bridge(session, originate_session);
>> ??????????? -- originate_session:hangup();
>> ??????? end
>> ??? end
>> end
>>
>> local?? digits = "";
>>
>> disa();
>>
>> if( session:ready() ) then
>> ??? session:hangup();
>> end
>>
>>
>> The extension where originate is placed:
>> ??? <extension name="ext_domestic_e164">
>> ??????? <condition field="destination_number" expression="^(7[0-9]{10})$">
>> ??????????? <action application="log" data="NOTICE [top.ctx] -
>> ext_domestic"/>
>> ??????????? <action application="set" data="hangup_after_bridge=true"/>
>> ??????????? <action application="set" data="continue_on_fail=false"/>
>> ??????????? <action application="set"
>> data="failure_causes=USER_BUSY,NO_ANSWER"/>
>> ??????????? <action application="set" data="fail_on_single_reject=true"/>
>> ??????????? <!-- Least Cost Routing scheme -->
>> ??????????? <action application="lcr" data="$1 lcr_tagnet"/>
>> ??????????? <action application="set" data="call_timeout=60"/>
>> ??????????? <action application="set" data="bridge_answer_timeout=60"/>
>> ??????????? <action application="bridge" data="${lcr_auto_route}"/>
>> ??????????? <action application="hangup"/>
>> ??????? </condition>
>> ??? </extension>
>>
>> Try harder..
>>
>> On Jul 15, 2011 4:43 AM, "Boris Kovalenko" <boris at tagnet.ru> wrote:
>> > Hello!
>> >
>> > I tried, no success.
>> >
>> >> after you create it
>> >>
>> >> session:setAutoHangup(0)
>> >>
>> >> or it will hangup when the script exits
>> >>
>> >> On Thu, Jul 14, 2011 at 1:11 PM, Boris Kovalenko<boris at tagnet.ru>
>> >> wrote:
>> >>> Hello!
>> >>>
>> >>> Here it is http://pastebin.freeswitch.org/16809
>> >>> Hope somebody help
>> >>>
>> >>> get a console log w/ siptrace of that scenario and put it on pastebin.
>> >>> the
>> >>> gang here will take a look.
>> >>> -MC
>> >>>
>> >>> On Thu, Jul 14, 2011 at 10:39 AM, Boris Kovalenko<boris at tagnet.ru>
>> >>> wrote:
>> >>>> And another problem found. If the remote party answers immediately
>> >>>> all
>> >>>> is ok. But if remote waits more then 8-10 seconds before answer - the
>> >>>> call is dropped when answered. Can't understand where to look for a
>> >>>> problem.
>> >>>>> Hello!
>> >>>>>
>> >>>>> I found a problem. originate_session must not to be local.
>> >>>>>> Hello!
>> >>>>>>
>> >>>>>> I'm trying to migrate DISA from Javascript to LUA. This peace of
>> >>>>>> code works fine in Javascript:
>> >>>>>>
>> >>>>>> ostr = "{ignore_early_media=true" +
>> >>>>>> ",origination_caller_id_number=" +
>> >>>>>> session.getVariable("caller_id_number") +
>> >>>>>> "}sofia/ipbx/50004#" + digits +
>> >>>>>> "@192.168.1.1:5060";
>> >>>>>> osession = new Session( ostr );
>> >>>>>> bridge(session, osession);
>> >>>>>> osession.hangup();
>> >>>>>>
>> >>>>>> With LUA not:
>> >>>>>> originate_string =
>> >>>>>> "{ignore_early_media=true" ..
>> >>>>>> ",origination_caller_id_number=" ..
>> >>>>>> session:getVariable("caller_id_number") ..
>> >>>>>> "}" ..
>> >>>>>> "sofia/ipbx/50004#" ..
>> >>>>>> digits .. "@192.168.1.1:5060";
>> >>>>>> originate_session = freeswitch.Session( originate_string );
>> >>>>>> freeswitch.bridge(session, originate_session);
>> >>>>>> originate_session:hangup();
>> >>>>>>
>> >>>>>> The call is droped when remote answers. There is an error in log:
>> >>>>>> bridge: session not ready. I tried to use
>> >>>>>> if( originate_session:ready() ) then
>> >>>>>> freeswitch.bridge(session, originate_session);
>> >>>>>> originate_session:hangup();
>> >>>>>> end
>> >>>>>>
>> >>>>>> with no luck :(. Please, help me. What am I doing wrong?
>> >>>>>>
>> >>>>
>> >>>> --
>> >>>> Regards,
>> >>>> Boris
>> >>>>
>> >>>>
>> >>>>
>> >>>> _______________________________________________
>> >>>> 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
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> 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
>> >>>
>> >>>
>> >>> --
>> >>> Regards,
>> >>> Boris
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> 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
>> >>>
>> >>>
>> >>
>> >>
>> >
>> >
>> > --
>> > Regards,
>> > Boris
>> >
>> >
>> >
>> > _______________________________________________
>> > 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
>>
>>
>> _______________________________________________
>> 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
>>
>>
>> --
>> Regards,
>> Boris
>>
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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
>
>



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


From boris at tagnet.ru  Sat Jul 16 17:02:41 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 16 Jul 2011 19:02:41 +0600
Subject: [Freeswitch-users] Same peace of code works with Javascript and
 not LUA
In-Reply-To: <CAKbxfG-vY4=q_1Gb2GrR5=AfJEsrBtzhk4jiqKH8fFSDrzhtHg@mail.gmail.com>
References: <4E1F230A.9080104@tagnet.ru> <4E1F24F6.5080101@tagnet.ru>
	<4E1F29BF.2030209@tagnet.ru>
	<CAKzWOxV_mtSxRR0XULjY6QC92wXnnrHL9eq-ZyPMeBoKX9eVrw@mail.gmail.com>
	<4E1F3165.70403@tagnet.ru>
	<CAKbxfG8y+hgJ5kAzKpWGeeozOx0j+j=Zzd3AWGhCBVto9_FRHw@mail.gmail.com>
	<4E200B6A.1060705@tagnet.ru>
	<CAKbxfG9CdSLryeCVnpq9L=x+DT2kF-K7xic7mnh2zpGOKqi8+g@mail.gmail.com>
	<4E20734B.70307@tagnet.ru>
	<CAKzWOxWtGvOzEihsph9H2-SMar+L5CkvuZ0rZpEM1dHAdxmhrA@mail.gmail.com>
	<CAKbxfG-vY4=q_1Gb2GrR5=AfJEsrBtzhk4jiqKH8fFSDrzhtHg@mail.gmail.com>
Message-ID: <4E218BF1.3040806@tagnet.ru>

Hello!

     Thank You, Anthony. I've updated to the latest git and script is 
working now! :)

> your line numbers indicate you are on an old revision update before
> you ask for help
>
> On Fri, Jul 15, 2011 at 12:54 PM, Michael Collins<msc at freeswitch.org>  wrote:
>> Okay, turn off the sip trace and turn on debug level logging:
>> sofia global siptrace off
>> console loglevel 7
>> Test call again, pastebin log please.
>> -MC
>>
>> On Fri, Jul 15, 2011 at 10:05 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>>> :) Nice joke
>>>
>>>
>>> Ok. Pastebin: http://pastebin.freeswitch.org/16818
>>>
>>> The extension is:
>>>      <extension name="ext_office_disa2">
>>>          <condition field="destination_number"
>>> expression="^(73435230022)$">
>>>              <action application="set"
>>> data="disa_msg_enterpin=$${base_dir}/sound
>>> s/ru/RU/elena/ivr/8000/ivr-please_enter_pin_followed_by_pound.wav"/>
>>>              <action application="set"
>>> data="disa_msg_wrongpin=$${base_dir}/sound
>>> s/ru/RU/elena/ivr/8000/ivr-pin_or_extension_is-invalid.wav"/>
>>>              <action application="set"
>>> data="disa_msg_enterext=$${base_dir}/sound
>>> s/ru/RU/elena/ivr/8000/ivr-enter_ext.wav"/>
>>>              <action application="lua" data="common/disa.lua"/>
>>>              <!--<action application="javascript" data="common/disa2.js"/>
>>> -->
>>>          </condition>
>>>      </extension>
>>>
>>>
>>> LUA script:
>>> --[[
>>> ]]
>>>
>>> -- if you choose not to require a pin then then you may want to add a
>>> dialplan c
>>> ondition for a specific caller id
>>> local   pin = "111111"; -- make sure to change the PIN number.
>>>
>>> local   digitmaxlength = 0;
>>> local   initialTimeout = 7500;
>>> local   interdigitTimeout = 5000;
>>> local   absoluteTimeout = 30000;
>>>
>>> local   msg_enterpin;
>>> local   msg_wrongPin;
>>> local   msg_enterext;
>>>
>>> function disa()
>>>      if( not session:ready() ) then
>>>          return
>>>      end
>>>
>>>
>>>      msg_enterpin = session:getVariable("disa_msg_enterpin");
>>>      msg_wrongpin = session:getVariable("disa_msg_wrongpin");
>>>      msg_enterext = session:getVariable("disa_msg_enterext");
>>>
>>>      if( msg_enterpin == nil or msg_wrongpin == nil or msg_enterext == nil
>>> ) then
>>>          freeswitch.consoleLog("ALERT", "DISA: Please set necessary
>>> variables");
>>>          return false;
>>>      end
>>>
>>>      session:answer();
>>>      session:sleep( 500 );
>>>
>>>      -- session:execute("start_dtmf", "");
>>>
>>>      if( string.len( pin )>  0 ) then
>>>          digitmaxlength = 6;
>>>
>>>          session:flushDigits();
>>>          digits = session:playAndGetDigits(digitmaxlength, digitmaxlength,
>>> 1, int
>>> erdigitTimeout, "#", msg_enterpin, msg_wrongpin, "\\d+", absoluteTimeout);
>>>          freeswitch.consoleLog( "info", "DISA.lua pin: " .. digits .. "\n"
>>> );
>>>      end
>>>
>>>      if( digits == pin or string.len( pin ) == 0 ) then
>>>          local   ru_ring = session:getVariable("ru-ring");
>>>          local   originate_string;
>>>
>>>          originate_session       = nil;
>>>
>>>          session:setVariable("ringback", ru_ring);               -- set to
>>> ringto
>>> ne
>>>          session:setVariable("transfer_ringback", ru_ring);      -- set to
>>> ringto
>>> ne
>>>          session:setVariable("hangup_after_bridge", "true");
>>>          session:setVariable("v_numbering_plan", "RU");
>>>
>>>          digits = "";    -- clear dtmf digits to prepare for next dtmf
>>> request
>>>          digitmaxlength = 20;
>>>          session:flushDigits();
>>>          digits = session:playAndGetDigits(6, digitmaxlength, 1,
>>> interdigitTimeou
>>> t, "#", msg_enterext, "", "\\d+", absoluteTimeout);
>>>          if( string.len( digits ) == 0 ) then
>>>              return false;
>>>          end
>>>          freeswitch.consoleLog( "info", "DISA.lua Collected: " .. digits ..
>>> "\n"
>>> );
>>>          originate_string =
>>>              "{ignore_early_media=true" ..
>>>              ",origination_caller_id_number=" ..
>>> session:getVariable("caller_id_n
>>> umber") ..
>>>              ",originate_timeout=90" ..
>>>              -- ",uuid=" .. session:get_uuid() ..
>>>              "}" ..
>>>              "sofia/ipbx/" .. session:getVariable("v_ats_srcport") .. "#"
>>> ..
>>>              digits .. "@192.168.1.1:5060";
>>>
>>>          originate_session = freeswitch.Session( originate_string, session
>>> );
>>>
>>>          session:setAutoHangup( false );
>>>          originate_session:setAutoHangup( false );
>>>
>>>          if( originate_session:ready() ) then
>>>
>>>              freeswitch.bridge(session, originate_session);
>>>              -- originate_session:hangup();
>>>          end
>>>      end
>>> end
>>>
>>> local   digits = "";
>>>
>>> disa();
>>>
>>> if( session:ready() ) then
>>>      session:hangup();
>>> end
>>>
>>>
>>> The extension where originate is placed:
>>>      <extension name="ext_domestic_e164">
>>>          <condition field="destination_number" expression="^(7[0-9]{10})$">
>>>              <action application="log" data="NOTICE [top.ctx] -
>>> ext_domestic"/>
>>>              <action application="set" data="hangup_after_bridge=true"/>
>>>              <action application="set" data="continue_on_fail=false"/>
>>>              <action application="set"
>>> data="failure_causes=USER_BUSY,NO_ANSWER"/>
>>>              <action application="set" data="fail_on_single_reject=true"/>
>>>              <!-- Least Cost Routing scheme -->
>>>              <action application="lcr" data="$1 lcr_tagnet"/>
>>>              <action application="set" data="call_timeout=60"/>
>>>              <action application="set" data="bridge_answer_timeout=60"/>
>>>              <action application="bridge" data="${lcr_auto_route}"/>
>>>              <action application="hangup"/>
>>>          </condition>
>>>      </extension>
>>>
>>> Try harder..
>>>
>>> On Jul 15, 2011 4:43 AM, "Boris Kovalenko"<boris at tagnet.ru>  wrote:
>>>> Hello!
>>>>
>>>> I tried, no success.
>>>>
>>>>> after you create it
>>>>>
>>>>> session:setAutoHangup(0)
>>>>>
>>>>> or it will hangup when the script exits
>>>>>
>>>>> On Thu, Jul 14, 2011 at 1:11 PM, Boris Kovalenko<boris at tagnet.ru>
>>>>> wrote:
>>>>>> Hello!
>>>>>>
>>>>>> Here it is http://pastebin.freeswitch.org/16809
>>>>>> Hope somebody help
>>>>>>
>>>>>> get a console log w/ siptrace of that scenario and put it on pastebin.
>>>>>> the
>>>>>> gang here will take a look.
>>>>>> -MC
>>>>>>
>>>>>> On Thu, Jul 14, 2011 at 10:39 AM, Boris Kovalenko<boris at tagnet.ru>
>>>>>> wrote:
>>>>>>> And another problem found. If the remote party answers immediately
>>>>>>> all
>>>>>>> is ok. But if remote waits more then 8-10 seconds before answer - the
>>>>>>> call is dropped when answered. Can't understand where to look for a
>>>>>>> problem.
>>>>>>>> Hello!
>>>>>>>>
>>>>>>>> I found a problem. originate_session must not to be local.
>>>>>>>>> Hello!
>>>>>>>>>
>>>>>>>>> I'm trying to migrate DISA from Javascript to LUA. This peace of
>>>>>>>>> code works fine in Javascript:
>>>>>>>>>
>>>>>>>>> ostr = "{ignore_early_media=true" +
>>>>>>>>> ",origination_caller_id_number=" +
>>>>>>>>> session.getVariable("caller_id_number") +
>>>>>>>>> "}sofia/ipbx/50004#" + digits +
>>>>>>>>> "@192.168.1.1:5060";
>>>>>>>>> osession = new Session( ostr );
>>>>>>>>> bridge(session, osession);
>>>>>>>>> osession.hangup();
>>>>>>>>>
>>>>>>>>> With LUA not:
>>>>>>>>> originate_string =
>>>>>>>>> "{ignore_early_media=true" ..
>>>>>>>>> ",origination_caller_id_number=" ..
>>>>>>>>> session:getVariable("caller_id_number") ..
>>>>>>>>> "}" ..
>>>>>>>>> "sofia/ipbx/50004#" ..
>>>>>>>>> digits .. "@192.168.1.1:5060";
>>>>>>>>> originate_session = freeswitch.Session( originate_string );
>>>>>>>>> freeswitch.bridge(session, originate_session);
>>>>>>>>> originate_session:hangup();
>>>>>>>>>
>>>>>>>>> The call is droped when remote answers. There is an error in log:
>>>>>>>>> bridge: session not ready. I tried to use
>>>>>>>>> if( originate_session:ready() ) then
>>>>>>>>> freeswitch.bridge(session, originate_session);
>>>>>>>>> originate_session:hangup();
>>>>>>>>> end
>>>>>>>>>
>>>>>>>>> with no luck :(. Please, help me. What am I doing wrong?
>>>>>>>>>
>>>>>>> --
>>>>>>> Regards,
>>>>>>> Boris
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> 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
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Regards,
>>>>>> Boris
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> --
>>>> Regards,
>>>> Boris
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>> --
>>> Regards,
>>> Boris
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
>


-- 
Regards,
Boris




From mi.ke at null.net  Sat Jul 16 19:18:02 2011
From: mi.ke at null.net (Mi Ke)
Date: Sat, 16 Jul 2011 15:18:02 +0000
Subject: [Freeswitch-users] playing audio to the *bridged* legs
Message-ID: <20110716151803.232360@gmx.com>

Hi All !

 Is it possible to play notification audio (wav or tgml) to one of *bridged* legs (without conferencing them) with Lua or any other way?

 session:streamFile executed on the leg gives audio neither on A or B while log shows file was played OK ...

 Thanks / Mike
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From avi at avimarcus.net  Sat Jul 16 22:12:33 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 16 Jul 2011 21:12:33 +0300
Subject: [Freeswitch-users] playing audio to the *bridged* legs
In-Reply-To: <20110716151803.232360@gmx.com>
References: <20110716151803.232360@gmx.com>
Message-ID: <CALNZuwmekSSdhJnXx6rif3KruAwU9fB7Gqn0sZ0+CTo0hXsLGQ@mail.gmail.com>

There's sched_broadcast which has leg options-
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_broadcast
 <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_broadcast>Or
uuid_broadcast http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
You can use the first as an execute, or either one as an API command and
specify the UUID.

-Avi


On Sat, Jul 16, 2011 at 6:18 PM, Mi Ke <mi.ke at null.net> wrote:

> Hi All !
>
> Is it possible to play notification audio (wav or tgml) to one of
> *bridged* legs (without conferencing them) with Lua or any other way?
>
> session:streamFile executed on the leg gives audio neither on A or B while
> log shows file was played OK ...
>
> Thanks / Mike
> _______________________________________________
> 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 jcasale at activenetwerx.com  Sun Jul 17 01:06:17 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 16 Jul 2011 21:06:17 +0000
Subject: [Freeswitch-users] Music on hold spec file
Message-ID: <assp.01780d1961.92097A6A775D5147B1078E3F15430B928B699E@prato.activenetwerx.local>

I whipped up a quick spec file to build these, there's an empty base package
which pulls in the 8/16/32/48 Khz sub packages. How does one go abouts
submitting this?

Thanks,
jlc


From brian at freeswitch.org  Sun Jul 17 01:26:08 2011
From: brian at freeswitch.org (Brian West)
Date: Sat, 16 Jul 2011 16:26:08 -0500
Subject: [Freeswitch-users] Music on hold spec file
In-Reply-To: <assp.01780d1961.92097A6A775D5147B1078E3F15430B928B699E@prato.activenetwerx.local>
References: <assp.01780d1961.92097A6A775D5147B1078E3F15430B928B699E@prato.activenetwerx.local>
Message-ID: <8A4D2223-A81E-4192-B456-CD62F6FABB8D@freeswitch.org>

jira.freeswitch.org

/b

On Jul 16, 2011, at 4:06 PM, Joseph L. Casale wrote:

> I whipped up a quick spec file to build these, there's an empty base package
> which pulls in the 8/16/32/48 Khz sub packages. How does one go abouts
> submitting this?
> 
> Thanks,
> jlc
> 
> _______________________________________________
> 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 bwibowo at gmail.com  Sun Jul 17 16:58:11 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Sun, 17 Jul 2011 19:58:11 +0700
Subject: [Freeswitch-users] incoming call to FS
Message-ID: <CAKnRR0ZkKt=QjB5a7eKdtFvYW+pq2SR7u-AFXr1TNPfGNfdObg@mail.gmail.com>

hi
i have sip server and want to connect the sip server to FS then call will be
routed using mod_dingaling, mod_dingaling already active
sipserver--->FS---mod_dingaling

from FS it will be incoming sip call, i already define sipserver arameter in
external sip_profile, then make call but i cant see any call coming from
fs_cli
anything missing in my config?

regards

budi wibowo
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From va_mclean at yahoo.com  Sun Jul 17 18:00:38 2011
From: va_mclean at yahoo.com (yahoo2003)
Date: Sun, 17 Jul 2011 07:00:38 -0700 (PDT)
Subject: [Freeswitch-users] Google Voice for Windows version of FS
In-Reply-To: <1310571922569-6579538.post@n2.nabble.com>
References: <1310468651.13286.YahooMailNeo@web121612.mail.ne1.yahoo.com>
	<1310479518002-6575189.post@n2.nabble.com>
	<1310482854.79304.YahooMailNeo@web121610.mail.ne1.yahoo.com>
	<1310483885842-6575490.post@n2.nabble.com>
	<1310511452.58175.YahooMailNeo@web121618.mail.ne1.yahoo.com>
	<1310571922569-6579538.post@n2.nabble.com>
Message-ID: <1310911238220-6591923.post@n2.nabble.com>

Thanks for everyone's help, I have Gtalk working for FreeSwitch (the problem
was "gnutls" was not included correctly by specifying "HAVE_GNUTLS").

I and others experienced one problem, when making Gtalk outgoing calls and
hanging up before the other party answers, the phone will continue to ring
for couple of minutes.

Is this known issue? Are there solutions for it?

Thanks


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Google-Voice-for-Windows-version-of-FS-tp6574607p6591923.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From michel.daggelinckx at gmail.com  Sun Jul 17 18:54:18 2011
From: michel.daggelinckx at gmail.com (Michel Daggelinckx)
Date: Sun, 17 Jul 2011 16:54:18 +0200
Subject: [Freeswitch-users] tdm support
Message-ID: <CAFYZzbuzLyx0Q4dzykMofgNDZ=EfhuU2UchquRju1ZD4oXYtwQ@mail.gmail.com>

Is there a module in the works to configure freetdm cards also with support
for dahdi tools and wanpipeconfig.


Michel
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From curriegrad2004 at gmail.com  Sun Jul 17 19:32:43 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 17 Jul 2011 08:32:43 -0700
Subject: [Freeswitch-users] Google Voice for Windows version of FS
In-Reply-To: <1310911238220-6591923.post@n2.nabble.com>
References: <1310468651.13286.YahooMailNeo@web121612.mail.ne1.yahoo.com>
	<1310479518002-6575189.post@n2.nabble.com>
	<1310482854.79304.YahooMailNeo@web121610.mail.ne1.yahoo.com>
	<1310483885842-6575490.post@n2.nabble.com>
	<1310511452.58175.YahooMailNeo@web121618.mail.ne1.yahoo.com>
	<1310571922569-6579538.post@n2.nabble.com>
	<1310911238220-6591923.post@n2.nabble.com>
Message-ID: <CAMWiMy9u-c5isDsB-chGM8vm4Fazxtpz-uKVz+86q6R+kpk54A@mail.gmail.com>

Unfortunately it is a known issue. It's more of google's problem than
the module's problem itself.

On Sun, Jul 17, 2011 at 7:00 AM, yahoo2003 <va_mclean at yahoo.com> wrote:
> Thanks for everyone's help, I have Gtalk working for FreeSwitch (the problem
> was "gnutls" was not included correctly by specifying "HAVE_GNUTLS").
>
> I and others experienced one problem, when making Gtalk outgoing calls and
> hanging up before the other party answers, the phone will continue to ring
> for couple of minutes.
>
> Is this known issue? Are there solutions for it?
>
> Thanks
>
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Google-Voice-for-Windows-version-of-FS-tp6574607p6591923.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> 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 brian at freeswitch.org  Sun Jul 17 22:06:21 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 17 Jul 2011 13:06:21 -0500
Subject: [Freeswitch-users] tdm support
In-Reply-To: <CAFYZzbuzLyx0Q4dzykMofgNDZ=EfhuU2UchquRju1ZD4oXYtwQ@mail.gmail.com>
References: <CAFYZzbuzLyx0Q4dzykMofgNDZ=EfhuU2UchquRju1ZD4oXYtwQ@mail.gmail.com>
Message-ID: <41CAD084-8B12-4888-9CB2-046B54D73BA7@freeswitch.org>

Why would mod_freetdm need to know about either of those?  

/b

On Jul 17, 2011, at 9:54 AM, Michel Daggelinckx wrote:

> Is there a module in the works to configure freetdm cards also with support
> for dahdi tools and wanpipeconfig.
> 
> 
> Michel

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From michel.daggelinckx at gmail.com  Sun Jul 17 22:24:23 2011
From: michel.daggelinckx at gmail.com (Michel Daggelinckx)
Date: Sun, 17 Jul 2011 20:24:23 +0200
Subject: [Freeswitch-users] tdm support
In-Reply-To: <41CAD084-8B12-4888-9CB2-046B54D73BA7@freeswitch.org>
References: <CAFYZzbuzLyx0Q4dzykMofgNDZ=EfhuU2UchquRju1ZD4oXYtwQ@mail.gmail.com>
	<41CAD084-8B12-4888-9CB2-046B54D73BA7@freeswitch.org>
Message-ID: <CAFYZzbtihP3g8fPDSrx4LLbxY=4yEVT6ZNWeYuB8aC8xExCAqQ@mail.gmail.com>

oops,
wrong list
sorry


On Sun, Jul 17, 2011 at 8:06 PM, Brian West <brian at freeswitch.org> wrote:

> Why would mod_freetdm need to know about either of those?
>
> /b
>
> On Jul 17, 2011, at 9:54 AM, Michel Daggelinckx wrote:
>
> Is there a module in the works to configure freetdm cards also with support
> for dahdi tools and wanpipeconfig.
>
>
> Michel
>
>
>
> _______________________________________________
> 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 roger.castaldo at gmail.com  Mon Jul 18 04:47:29 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Sun, 17 Jul 2011 20:47:29 -0400
Subject: [Freeswitch-users] .Net interface
Message-ID: <CAAYf2Lt5QJ+WjzBxY257XsfH021dAMDvGqACosbKwmPt8NOjjg@mail.gmail.com>

Hi everyone this is just to let everyone know that I have developed a C#
based .Net socket library to allow a freeswitch server to be interfaced with
using the event socket and outbound socket.  The library is hosted on google
code, the link is http://code.google.com/p/freeswitch-socket-dotnet/
If anyone is interested in helping with the development please let me know.
Feel free to use it and expect updates to it as time permits, my goal is to
try and make the library somewhat idiot proof by implementing all available
calls that could be handled on the server side when controlling a phone
call.
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From manjiri05_deshpande at yahoo.co.in  Mon Jul 18 06:39:48 2011
From: manjiri05_deshpande at yahoo.co.in (Manjiri Deshpande)
Date: Mon, 18 Jul 2011 08:09:48 +0530 (IST)
Subject: [Freeswitch-users] FreeSwitch does not send HOLD Reinvite on
	other leg.
In-Reply-To: <FF145CE6-6931-49BA-A5E4-2D442E74947E@ipeva.fr>
References: <1310722518.24590.YahooMailNeo@web95905.mail.in.yahoo.com>
	<FF145CE6-6931-49BA-A5E4-2D442E74947E@ipeva.fr>
Message-ID: <1310956788.11356.YahooMailNeo@web95908.mail.in.yahoo.com>

A and B are phones registered locally.I am using win2k8 R2 platform for testing these calls.
FS sends "BYE" message to both the legs .
I have set "rtp-hold-timeout-sec" as 30 mins

From: David Ponzone <david.ponzone at ipeva.fr>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Friday, 15 July 2011 4:52 PM
Subject: Re: [Freeswitch-users] FreeSwitch does not send HOLD Reinvite on other leg.


There are some information that you need to give to the list:
-we need to know what are A and B (remote switches or phones registered locally, ...) 
-which element triggers the disconnect ?


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

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

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




Le 15/07/2011 ? 11:35, Manjiri Deshpande a ?crit :

Scenario:
>
>A -> FS -> B
>
>A puts the calls on hold ( ie sending a reINVITE with media ip 0.0.0.0 ).
>
>FS does not send this re INVITE to leg B.Leg B continues to send RTP??to FS and call disconnects sometime afterwards if call is not removed from hold. Disconnect times have been as short as 3 minutes and as long as 30 minutes. 
>
>Test calls have included background noise from Leg B?and seen noise in trace as normal RTP packets. Call does not seem to disconnect if this background noise is present. Once background noise disappears from Leg B ,FreeSwitch call will disconnect.
>
>Why Leg B sends RTP packets sometimes ,so that call is not getting disconnected before 30 mins.
>Also If RTP packets are not there,then also call should not be disconnected before 30 mins.
>
>Thanks,
>Manjiri_______________________________________________
>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
>

_______________________________________________
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 ravi.4indra at gmail.com  Mon Jul 18 00:23:23 2011
From: ravi.4indra at gmail.com (Ravindra Kondiparthi)
Date: Sun, 17 Jul 2011 13:23:23 -0700
Subject: [Freeswitch-users] Error While Installing free switch on UBuntu
	11.04
Message-ID: <CAHhQwYDcz=h9EVyFNZH-2p-wOgODrocTBrGWUbznCxvU4YhX=g@mail.gmail.com>

Hi,

 I am getting the below error while installing free switch on UBuntu
11.04.Any help in resolving the error is appreciated.


Compiling src/switch_core_sqldb.c ...
cc1: warnings being treated as errors
src/switch_core_sqldb.c: In function '_switch_cache_db_get_db_handle':
src/switch_core_sqldb.c:314:47: error: comparison between
'switch_odbc_status_t' and 'enum <anonymous>'
src/switch_core_sqldb.c: In function 'switch_cache_db_execute_sql_real':
src/switch_core_sqldb.c:389:90: error: comparison between 'switch_status_t'
and 'enum <anonymous>'
make[1]: *** [libfreeswitch_la-switch_core_sqldb.lo] Error 1
make: *** [all] Error 2




thanks
Ravi
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From curriegrad2004 at gmail.com  Mon Jul 18 08:30:13 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 17 Jul 2011 21:30:13 -0700
Subject: [Freeswitch-users] Error While Installing free switch on UBuntu
	11.04
In-Reply-To: <CAHhQwYDcz=h9EVyFNZH-2p-wOgODrocTBrGWUbznCxvU4YhX=g@mail.gmail.com>
References: <CAHhQwYDcz=h9EVyFNZH-2p-wOgODrocTBrGWUbznCxvU4YhX=g@mail.gmail.com>
Message-ID: <CAMWiMy9jtApsrnyMZDBE53i+PysAM9wkN6ARv_zq4SQ-9LUWHA@mail.gmail.com>

-Werror on the core lib is enabled... If you can find a way to mess
with the build system, you could probably can get it to build without
the option, but I can't remember off the top of my head on how to get
-Werror off the build...

On Sun, Jul 17, 2011 at 1:23 PM, Ravindra Kondiparthi
<ravi.4indra at gmail.com> wrote:
> Hi,
>
> ?I am getting the below error while installing free switch on UBuntu
> 11.04.Any help in resolving the error is appreciated.
>
>
> Compiling src/switch_core_sqldb.c ...
> cc1: warnings being treated as errors
> src/switch_core_sqldb.c: In function '_switch_cache_db_get_db_handle':
> src/switch_core_sqldb.c:314:47: error: comparison between
> 'switch_odbc_status_t' and 'enum <anonymous>'
> src/switch_core_sqldb.c: In function 'switch_cache_db_execute_sql_real':
> src/switch_core_sqldb.c:389:90: error: comparison between 'switch_status_t'
> and 'enum <anonymous>'
> make[1]: *** [libfreeswitch_la-switch_core_sqldb.lo] Error 1
> make: *** [all] Error 2
>
>
>
>
> thanks
> Ravi
>
> _______________________________________________
> 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 steveayre at gmail.com  Mon Jul 18 11:58:38 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 18 Jul 2011 08:58:38 +0100
Subject: [Freeswitch-users] .Net interface
In-Reply-To: <CAAYf2Lt5QJ+WjzBxY257XsfH021dAMDvGqACosbKwmPt8NOjjg@mail.gmail.com>
References: <CAAYf2Lt5QJ+WjzBxY257XsfH021dAMDvGqACosbKwmPt8NOjjg@mail.gmail.com>
Message-ID: <CAFiqYuk3J1452vfF7tgJgJh_rEt73LgtVdQW6BJnwokSc+A3XQ@mail.gmail.com>

Thanks for the contribution...

Hate to tell you this though, but do you do know there's already one
in the FreeSWITCH trunk? Look in libs/esl/managed of the git checkout.

Although yours might of course provide slightly different
functionality so could still be useful in addition to the official
one.

-Steve


On 18 July 2011 01:47, Roger Castaldo <roger.castaldo at gmail.com> wrote:
> Hi everyone this is just to let everyone know that I have developed a C#
> based .Net socket library to allow a freeswitch server to be interfaced with
> using the event socket and outbound socket.? The library is hosted on google
> code, the link is http://code.google.com/p/freeswitch-socket-dotnet/
> If anyone is interested in helping with the development please let me know.
> Feel free to use it and expect updates to it as time permits, my goal is to
> try and make the library somewhat idiot proof by implementing all available
> calls that could be handled on the server side when controlling a phone
> call.
>
> _______________________________________________
> 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 mays.david at gmail.com  Mon Jul 18 13:18:26 2011
From: mays.david at gmail.com (David Ma)
Date: Mon, 18 Jul 2011 17:18:26 +0800
Subject: [Freeswitch-users] Leg-A is automatically disconnected on Leg-B
 orginate failure
In-Reply-To: <BANLkTikyyAZaJzQx2tbiVOxHxx5mCbUdvQ@mail.gmail.com>
References: <1308213151543-6482192.post@n2.nabble.com>
	<BANLkTi=xYgBTPVGxVGmk9f2=bmX5X2vF=Q@mail.gmail.com>
	<BANLkTikhPmjN5HeeJHBnQt1ivA8zbtCnYA@mail.gmail.com>
	<BANLkTi=m7oMix1ZRC9URtRuAbsG2fWyhMQ@mail.gmail.com>
	<BANLkTimxBU8m311LVKMWcFOjukwLg+ZUGQ@mail.gmail.com>
	<BANLkTinZJf3BxB7QHefnmOvxQmVW2r4dTg@mail.gmail.com>
	<BANLkTimiExLew_1cjMuQVpF3CKkR8mrGXQ@mail.gmail.com>
	<BANLkTi=ZvOZ99qVxUeJAtpjtwP5r=hfVkg@mail.gmail.com>
	<BANLkTimj4iSWFiHuBSKskz7NjF2bCEWnaw@mail.gmail.com>
	<BANLkTikyyAZaJzQx2tbiVOxHxx5mCbUdvQ@mail.gmail.com>
Message-ID: <CAMLBmRLtAeU9m6HwOW3o9FA2pZFRBiXk2P82Z7msv218B4AxCA@mail.gmail.com>

Hi Anthony,

I've tried channel variable uuid_bridge_continue_on_cancel=true. However,
this doesn't solve the problem. The log is in the following on pastebin,
please kindly help investigate why.

http://pastebin.freeswitch.org/16839

Thanks a lot!

Best regards,
D.Ma

On Wed, Jun 29, 2011 at 10:53 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> The only side effect is that it will give you the behavior you want in
> this particular situation.
>
>
> On Wed, Jun 29, 2011 at 4:35 AM, David Ma <mays.david at gmail.com> wrote:
> > Hi Anthony,
> >
> > Thanks very much for the information. I appreciate your advice. It is
> great
> > to learn about such a parameter.
> >
> > FS wiki has only a little description about this parameter. Is there any
> > side effect, or caution for using such a parameter?
> >
> > Thanks,
> > D.Ma
> >
> > On Wed, Jun 29, 2011 at 12:08 AM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> The cases where you have this problem arise because you are
> >> originating to the B leg who answers 183 (early media) then hangs up
> >> before the call was answered.
> >>
> >> set the channel variable uuid_bridge_continue_on_cancel=true on the A
> >> leg to change the behavior to what you want.
> >>
> >>
> >> On Mon, Jun 27, 2011 at 10:09 AM, Michael Collins <msc at freeswitch.org>
> >> wrote:
> >> > This makes my eyes bleed. Can you please put this on
> >> > pastebin.freeswitch.org? Use "FreeSWITCH Log" as the syntax
> highlight.
> >> > -MC
> >> >
> >> > On Sun, Jun 26, 2011 at 11:27 PM, David Ma <mays.david at gmail.com>
> wrote:
> >> >>
> >> >> Hi Michael,
> >> >>
> >>
> >> --
> >> 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
> >
> >
> > _______________________________________________
> > 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
> >
> >
>
>
>
> --
> 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
>
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From mays.david at gmail.com  Mon Jul 18 13:25:01 2011
From: mays.david at gmail.com (David Ma)
Date: Mon, 18 Jul 2011 17:25:01 +0800
Subject: [Freeswitch-users] Leg-A is automatically disconnected on Leg-B
 orginate failure
In-Reply-To: <CAMLBmRLtAeU9m6HwOW3o9FA2pZFRBiXk2P82Z7msv218B4AxCA@mail.gmail.com>
References: <1308213151543-6482192.post@n2.nabble.com>
	<BANLkTi=xYgBTPVGxVGmk9f2=bmX5X2vF=Q@mail.gmail.com>
	<BANLkTikhPmjN5HeeJHBnQt1ivA8zbtCnYA@mail.gmail.com>
	<BANLkTi=m7oMix1ZRC9URtRuAbsG2fWyhMQ@mail.gmail.com>
	<BANLkTimxBU8m311LVKMWcFOjukwLg+ZUGQ@mail.gmail.com>
	<BANLkTinZJf3BxB7QHefnmOvxQmVW2r4dTg@mail.gmail.com>
	<BANLkTimiExLew_1cjMuQVpF3CKkR8mrGXQ@mail.gmail.com>
	<BANLkTi=ZvOZ99qVxUeJAtpjtwP5r=hfVkg@mail.gmail.com>
	<BANLkTimj4iSWFiHuBSKskz7NjF2bCEWnaw@mail.gmail.com>
	<BANLkTikyyAZaJzQx2tbiVOxHxx5mCbUdvQ@mail.gmail.com>
	<CAMLBmRLtAeU9m6HwOW3o9FA2pZFRBiXk2P82Z7msv218B4AxCA@mail.gmail.com>
Message-ID: <CAMLBmR+yqetjWU+yGOypLtwOj8Ryz8rbEp41tW0yHS53w-pgWg@mail.gmail.com>

Hello Anthony,

The same log in better formatting is here:

http://pastebin.freeswitch.org/16840

Regards,
D.Ma

On Mon, Jul 18, 2011 at 5:18 PM, David Ma <mays.david at gmail.com> wrote:

> Hi Anthony,
>
> I've tried channel variable uuid_bridge_continue_on_cancel=true. However,
> this doesn't solve the problem. The log is in the following on pastebin,
> please kindly help investigate why.
>
> http://pastebin.freeswitch.org/16839
>
> Thanks a lot!
>
> Best regards,
> D.Ma
>
>
> On Wed, Jun 29, 2011 at 10:53 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> The only side effect is that it will give you the behavior you want in
>> this particular situation.
>>
>>
>> On Wed, Jun 29, 2011 at 4:35 AM, David Ma <mays.david at gmail.com> wrote:
>> > Hi Anthony,
>> >
>> > Thanks very much for the information. I appreciate your advice. It is
>> great
>> > to learn about such a parameter.
>> >
>> > FS wiki has only a little description about this parameter. Is there any
>> > side effect, or caution for using such a parameter?
>> >
>> > Thanks,
>> > D.Ma
>> >
>> > On Wed, Jun 29, 2011 at 12:08 AM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> The cases where you have this problem arise because you are
>> >> originating to the B leg who answers 183 (early media) then hangs up
>> >> before the call was answered.
>> >>
>> >> set the channel variable uuid_bridge_continue_on_cancel=true on the A
>> >> leg to change the behavior to what you want.
>> >>
>> >>
>> >> On Mon, Jun 27, 2011 at 10:09 AM, Michael Collins <msc at freeswitch.org>
>> >> wrote:
>> >> > This makes my eyes bleed. Can you please put this on
>> >> > pastebin.freeswitch.org? Use "FreeSWITCH Log" as the syntax
>> highlight.
>> >> > -MC
>> >> >
>> >> > On Sun, Jun 26, 2011 at 11:27 PM, David Ma <mays.david at gmail.com>
>> wrote:
>> >> >>
>> >> >> Hi Michael,
>> >> >>
>> >>
>> >> --
>> >> 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
>> >
>> >
>> > _______________________________________________
>> > 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
>> >
>> >
>>
>>
>>
>> --
>> 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
>>
>
>
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From tgraziano at myitdepartment.net  Mon Jul 18 14:48:37 2011
From: tgraziano at myitdepartment.net (Tony Graziano)
Date: Mon, 18 Jul 2011 06:48:37 -0400
Subject: [Freeswitch-users] Looking for US based providers offering t.38
	that work with freeswitch
Message-ID: <CAMgKNJUYwgPF6YoF5ckgLwYZfd4OfzOT0wJ0hneOBKp5KSHXLQ@mail.gmail.com>

So far we have only been able to find a provider using sonus switches
for this. Is anyone successfully using a trunk provider to receive
faxes via t.38 in freeswitch they are willing to share?


Thanks,

Tony


From roger.castaldo at gmail.com  Mon Jul 18 15:31:48 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Mon, 18 Jul 2011 07:31:48 -0400
Subject: [Freeswitch-users] .Net interface
In-Reply-To: <CAFiqYuk3J1452vfF7tgJgJh_rEt73LgtVdQW6BJnwokSc+A3XQ@mail.gmail.com>
References: <CAAYf2Lt5QJ+WjzBxY257XsfH021dAMDvGqACosbKwmPt8NOjjg@mail.gmail.com>
	<CAFiqYuk3J1452vfF7tgJgJh_rEt73LgtVdQW6BJnwokSc+A3XQ@mail.gmail.com>
Message-ID: <CAAYf2LuJYW7GMGa0DmrnsOjw-zQQjqEvmkVvvS0u2SzcX=kB7Q@mail.gmail.com>

Yes there is a contributed library made of the esl which is a managed
wrapper of the cpp code.  The library I have designed and built requires no
external references it is a pure socket implementation that is compiled
independently of freeswitch to allow interfacing with freeswitch and is an
attempt to wrap all command calls in easy to use function calls to allow
less freeswitch experienced .Net developers to interface with freeswitch
even easier.

On Mon, Jul 18, 2011 at 3:58 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Thanks for the contribution...
>
> Hate to tell you this though, but do you do know there's already one
> in the FreeSWITCH trunk? Look in libs/esl/managed of the git checkout.
>
> Although yours might of course provide slightly different
> functionality so could still be useful in addition to the official
> one.
>
> -Steve
>
>
> On 18 July 2011 01:47, Roger Castaldo <roger.castaldo at gmail.com> wrote:
> > Hi everyone this is just to let everyone know that I have developed a C#
> > based .Net socket library to allow a freeswitch server to be interfaced
> with
> > using the event socket and outbound socket.  The library is hosted on
> google
> > code, the link is http://code.google.com/p/freeswitch-socket-dotnet/
> > If anyone is interested in helping with the development please let me
> know.
> > Feel free to use it and expect updates to it as time permits, my goal is
> to
> > try and make the library somewhat idiot proof by implementing all
> available
> > calls that could be handled on the server side when controlling a phone
> > call.
> >
> > _______________________________________________
> > 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
> >
> >
>
> _______________________________________________
> 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 cmrienzo at gmail.com  Mon Jul 18 15:46:57 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Mon, 18 Jul 2011 07:46:57 -0400
Subject: [Freeswitch-users] .Net interface
In-Reply-To: <CAAYf2LuJYW7GMGa0DmrnsOjw-zQQjqEvmkVvvS0u2SzcX=kB7Q@mail.gmail.com>
References: <CAAYf2Lt5QJ+WjzBxY257XsfH021dAMDvGqACosbKwmPt8NOjjg@mail.gmail.com>
	<CAFiqYuk3J1452vfF7tgJgJh_rEt73LgtVdQW6BJnwokSc+A3XQ@mail.gmail.com>
	<CAAYf2LuJYW7GMGa0DmrnsOjw-zQQjqEvmkVvvS0u2SzcX=kB7Q@mail.gmail.com>
Message-ID: <CAD-qRkiA5ht+ubua60Q0hjZbRLTWTrOoMuTR6uYc+1HffrDvEw@mail.gmail.com>

ESL is a library compiled separately from FreeSWITCH.  There are no
dependencies between them.



On Mon, Jul 18, 2011 at 7:31 AM, Roger Castaldo <roger.castaldo at gmail.com>wrote:

> Yes there is a contributed library made of the esl which is a managed
> wrapper of the cpp code.  The library I have designed and built requires no
> external references it is a pure socket implementation that is compiled
> independently of freeswitch to allow interfacing with freeswitch and is an
> attempt to wrap all command calls in easy to use function calls to allow
> less freeswitch experienced .Net developers to interface with freeswitch
> even easier.
>
>
> On Mon, Jul 18, 2011 at 3:58 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Thanks for the contribution...
>>
>> Hate to tell you this though, but do you do know there's already one
>> in the FreeSWITCH trunk? Look in libs/esl/managed of the git checkout.
>>
>> Although yours might of course provide slightly different
>> functionality so could still be useful in addition to the official
>> one.
>>
>> -Steve
>>
>>
>> On 18 July 2011 01:47, Roger Castaldo <roger.castaldo at gmail.com> wrote:
>> > Hi everyone this is just to let everyone know that I have developed a C#
>> > based .Net socket library to allow a freeswitch server to be interfaced
>> with
>> > using the event socket and outbound socket.  The library is hosted on
>> google
>> > code, the link is http://code.google.com/p/freeswitch-socket-dotnet/
>> > If anyone is interested in helping with the development please let me
>> know.
>> > Feel free to use it and expect updates to it as time permits, my goal is
>> to
>> > try and make the library somewhat idiot proof by implementing all
>> available
>> > calls that could be handled on the server side when controlling a phone
>> > call.
>> >
>> > _______________________________________________
>> > 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
>> >
>> >
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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 jaybinks at gmail.com  Mon Jul 18 16:29:18 2011
From: jaybinks at gmail.com (jay binks)
Date: Mon, 18 Jul 2011 22:29:18 +1000
Subject: [Freeswitch-users] stripping sip-rh headers at the edge of my
	network
In-Reply-To: <CAO47HpdG8qJeH4uoXDFB31Jf7_1oNVbmoJQ52DUS6WcmxuAj=Q@mail.gmail.com>
References: <CAO47HpdG8qJeH4uoXDFB31Jf7_1oNVbmoJQ52DUS6WcmxuAj=Q@mail.gmail.com>
Message-ID: <CAO47HpfcjUJj_9UVO8dFEQq8M6gMaZs5DKvUP4H_hW8tkC324A@mail.gmail.com>

hmm this isnt good..  no replies..

so, seems I cant strip these at all with freeswitch..

using unset and even trying to overwrite the value like this

<action application="set" data="sip_ph_X-someheader=blah"/>
<action application="set" data="sip_rh_X-someheader=blah"/>
<action application="set" data="sip_h_X-someheader=blah"/>

fails...
since FS just splats on top of whatever I set , when the sip message comes
in from upstream..
(  in this case in the 180 / 183 progress )

I need a way to strip headers from 180/183 .. preferably I could have a
regex to filter any starting with X :)
someone please point me in the right direction

Jay





On Fri, Jul 15, 2011 at 4:54 PM, jay binks <jaybinks at gmail.com> wrote:

> Inside my network I use sip_ph , sip_rh & sip_h headers to pass data around
> between boxes.
>
> however I need to strip these off at the edge ( SBC's running Freeswitch
> also )
>
> will unset work with these headers in the 200 OK ( and other sip messages )
> ?
> if not, how can I achieve this ?
>
> --
> Sincerely
>
> Jay
>



-- 
Sincerely

Jay
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From leonardo.bidinoto at voicetechnology.com.br  Mon Jul 18 17:21:08 2011
From: leonardo.bidinoto at voicetechnology.com.br (Leonardo P. Bidinoto)
Date: Mon, 18 Jul 2011 10:21:08 -0300
Subject: [Freeswitch-users] Hanged up callers doubt
In-Reply-To: <793379B8-D1FC-47F4-8387-4E2789A69C72@ipeva.fr>
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Message-ID: <CALV5c5NGFX6feJAf9bXMwWju-wCdLSp6Gi8r4aQ63eC4x_vwqg@mail.gmail.com>

Hello FS Community,

i found a solution for this issue and i'm giving some feedback about how I
avoided this ghost callers issue.

I discovered the scenario when ghost start to happen. When two different ESL
connections try to execute commands with the same channel with small
interval with each other, like one command is processing and the other
connection send another command to the same channel.
FS channel keeps stuck and the symptom is the appearance of ghost calls in
the system. Even when the user hangs up, the channel keeps in the memory. As
I described in my posts, after while this issue crashes FS. I discovered it
after a stress test with tons of calls running and my software management
trying to command those lines simultaneity.
There is no way to fix this after ghost appearance, only restarting FS.

The issue is generated by one command executed by ESL connection thru
mod_socket waiting for the response from the channel(the channel is inside a
conference) and another ESL connection try to execute another one like
conference_kick or conference_relate.
To avoid this situation I created a semaphore to block simultaneous command
execution over the same channel and stacking the other commands to wait for
the response of the previous command. That was tricky because you need to
set the semaphore right after you get the command, otherwise you will get
ghost callers on extreme conditions (I.E: High traffic)

I hope this hint could be useful to the other users.

PS:. Before sending this email, i checked with the lasted FS
version(git-db5f504 2011-07-17 17-00-38 -0400), and verify that
'conference_kick' command was no longer causing the channel get stuck, but
'conference_relate' command still showing the same problem.


2011/6/3 David Ponzone <david.ponzone at ipeva.fr>

> I guess there is no way to drop stuck calls. That's why they are stuck.
> I think you should find a way to solve the root cause.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 03/06/2011 ? 14:31, Leonardo P. Bidinoto a ?crit :
>
> Hi David,
>
> Yes, indeed that should work, BUT it doesn't if an user inside is stucked(i
> get a "false" from "uuid_exists").
> With "conference kick", all users inside will be dropped, but the stucked
> ones will remain, and with that the conference isnt destroyed.
>
>
>
> 2011/6/2 David Ponzone <david.ponzone at ipeva.fr>
>
>> Sure, that one is easy:
>>
>> from fs_cli:
>>
>> conference <confname> kick all
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 02/06/2011 ? 21:33, Leonardo P. Bidinoto a ?crit :
>>
>> Hi,
>>
>> About this issue, could be a way to destroy the conference from FS, in a
>> attempt to remove the stucked channels from the "show channels " and
>> "conference list"?
>>
>> 2011/5/17 Leonardo P. Bidinoto <leonardo.bidinoto at voicetechnology.com.br>
>>
>>> Sure. Im sending a pcap file made by tcpdump and one that i made by
>>> ngrep. In both files, it was registering whats happening when i stuck the
>>> channel by hanging up while using a ESL connection inside a conference(app
>>> socket 8085 sync full). I did a "conference kick" command in this channel
>>> while its was waiting to close the ESL connection.
>>>
>>>
>>>
>>> 2011/5/16 Michael Collins <msc at freeswitch.org>
>>>
>>>> Can you tcpdump or otherwise capture the traffic on port 8085? I am
>>>> curious what is happening with that.
>>>> -MC
>>>>
>>>>
>>>> On Mon, May 16, 2011 at 12:12 PM, Leonardo P. Bidinoto <
>>>> leonardo.bidinoto at voicetechnology.com.br> wrote:
>>>>
>>>>> hehe, ok michael.
>>>>>
>>>>> here is the pastebin link:
>>>>> http://pastebin.freeswitch.org/16303
>>>>>
>>>>>
>>>>> 2011/5/13 Michael Collins <msc at freeswitch.org>
>>>>>
>>>>>> Pastebin this info and select "FreeSWITCH Log" as the syntax
>>>>>> highlighting. I need the colorized output to read logs. (I'm getting older
>>>>>> and it's hard for me to ready black and white in an email.)
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>> On Fri, May 13, 2011 at 7:32 AM, Leonardo P. Bidinoto <
>>>>>> leonardo.bidinoto at voicetechnology.com.br> wrote:
>>>>>>
>>>>>>> Hi Michael,
>>>>>>>
>>>>>>> Just succeeded to reproduce the problem.
>>>>>>>
>>>>>>> The condition is: when a channel inside a conference is using a ESL
>>>>>>> connection(lets call it "A") through socket application and another ESL
>>>>>>> connection(lets call it "B") executes a command with this channel, the "B"
>>>>>>> ESL connection will wait the "A" ESL connection close to execute its
>>>>>>> command.
>>>>>>> If the channel hangs up before the "A" ESL connection is closed, then
>>>>>>> "B" ESL command will never be executed and the stucked channel will still be
>>>>>>> there, into sofia and the conference too.
>>>>>>> To verify that, just do "show channels" and "conference list". with
>>>>>>> "uuid_exists" command, return "false".
>>>>>>>
>>>>>>> Here are the actions done by the channel before get stucked:
>>>>>>>
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.187321
>>>>>>> [NOTICE] switch_channel.c:816 New Channel sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 [16e09413-9cb0-4011-a635-f91933a35c0f]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154entering state [received][100]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] sofia.c:4772 Remote SDP:
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] sofia_glue.c:4656 Audio Codec Compare
>>>>>>> [BV32:107:16000:20:0]/[PCMU:0:8000:20:64000]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] sofia_glue.c:4656 Audio Codec Compare
>>>>>>> [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] sofia_glue.c:4656 Audio Codec Compare
>>>>>>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] sofia_glue.c:2788 Set Codec sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 PCMU/8000 20 ms 160 samples 64000 bits
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] sofia_glue.c:4770 Set 2833 dtmf send/recv payload to 101
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] sofia.c:4943 (sofia/external/1000123402 at 192.168.0.154) State
>>>>>>> Change CS_NEW -> CS_INIT
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] switch_core_session.c:1114 Send signal sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] switch_core_state_machine.c:325 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) Running State Change CS_INIT
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] switch_core_state_machine.c:361 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) State INIT
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] mod_sofia.c:84 sofia/external/1000123402 at 192.168.0.154 SOFIA
>>>>>>> INIT
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] mod_sofia.c:124 (sofia/external/1000123402 at 192.168.0.154)
>>>>>>> State Change CS_INIT -> CS_ROUTING
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] switch_core_session.c:1114 Send signal sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] switch_core_state_machine.c:361 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) State INIT going to sleep
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] switch_core_state_machine.c:325 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) Running State Change CS_ROUTING
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] switch_channel.c:1665 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) Callstate Change DOWN -> RINGING
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] switch_core_state_machine.c:364 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) State ROUTING
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154SOFIA ROUTING
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [DEBUG] switch_core_state_machine.c:77 sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Standard ROUTING
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.188323
>>>>>>> [INFO] mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in
>>>>>>> context public
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 parsing [public->unloop] continue=false
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Regex (PASS) [unloop] ${unroll_loops}(true)
>>>>>>> =~ /^true$/ break=on-false
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Regex (FAIL) [unloop] ${sip_looped_call}()
>>>>>>> =~ /^true$/ break=on-false
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 parsing [public->public_extensions]
>>>>>>> continue=false
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Regex (PASS) [public_extensions]
>>>>>>> destination_number(1234567890) =~ /^(\d*)$/ break=on-false
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Action transfer(1234567890 XML default)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328
>>>>>>> [DEBUG] switch_core_state_machine.c:119 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) State Change CS_ROUTING -> CS_EXECUTE
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328
>>>>>>> [DEBUG] switch_core_session.c:1114 Send signal sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328
>>>>>>> [DEBUG] switch_core_state_machine.c:364 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) State ROUTING going to sleep
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328
>>>>>>> [DEBUG] switch_core_state_machine.c:325 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) Running State Change CS_EXECUTE
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328
>>>>>>> [DEBUG] switch_core_state_machine.c:371 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) State EXECUTE
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.190328
>>>>>>> [DEBUG] mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154SOFIA EXECUTE
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>>> [DEBUG] switch_core_state_machine.c:157 sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Standard EXECUTE
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 transfer(1234567890 XML default)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>>> [DEBUG] switch_ivr.c:1597 (sofia/external/1000123402 at 192.168.0.154)
>>>>>>> State Change CS_EXECUTE -> CS_ROUTING
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>>> [DEBUG] switch_core_session.c:1114 Send signal sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>>> [DEBUG] switch_core_session.c:707 Send signal sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>>> [NOTICE] switch_ivr.c:1603 Transfer sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 to XML[1234567890 at default]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>>> [DEBUG] switch_core_state_machine.c:371 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) State EXECUTE going to sleep
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>>> [DEBUG] switch_core_state_machine.c:325 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) Running State Change CS_ROUTING
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>>> [DEBUG] switch_core_state_machine.c:364 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) State ROUTING
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>>> [DEBUG] mod_sofia.c:147 sofia/external/1000123402 at 192.168.0.154SOFIA ROUTING
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>>> [DEBUG] switch_core_state_machine.c:77 sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Standard ROUTING
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.191327
>>>>>>> [INFO] mod_dialplan_xml.c:331 Processing Phone1 <1000123402>->1234567890 in
>>>>>>> context default
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 parsing [default->flex] continue=false
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Regex (PASS) [flex]
>>>>>>> destination_number(1234567890) =~ /^(\d+)$/ break=on-false
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Action log(INFO VOICE received
>>>>>>> dest=1234567890)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Action set(playback_terminators=#)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Action log(INFO Let's do some ivrd, shall
>>>>>>> we?)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Action
>>>>>>> set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f Dialplan: sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Action socket(localhost:8084 full)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325
>>>>>>> [DEBUG] switch_core_state_machine.c:119 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) State Change CS_ROUTING -> CS_EXECUTE
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325
>>>>>>> [DEBUG] switch_core_session.c:1114 Send signal sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325
>>>>>>> [DEBUG] switch_core_state_machine.c:364 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) State ROUTING going to sleep
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325
>>>>>>> [DEBUG] switch_core_state_machine.c:325 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) Running State Change CS_EXECUTE
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325
>>>>>>> [DEBUG] switch_core_state_machine.c:371 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) State EXECUTE
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325
>>>>>>> [DEBUG] mod_sofia.c:240 sofia/external/1000123402 at 192.168.0.154SOFIA EXECUTE
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325
>>>>>>> [DEBUG] switch_core_state_machine.c:157 sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Standard EXECUTE
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 log(INFO VOICE received dest=1234567890)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.193325
>>>>>>> [INFO] mod_dptools.c:1184 VOICE received dest=1234567890
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 set(playback_terminators=#)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331
>>>>>>> [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154SET [playback_terminators]=[#]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 log(INFO Let's do some ivrd, shall we?)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331
>>>>>>> [INFO] mod_dptools.c:1184 Let's do some ivrd, shall we?
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154set(ivr_path=/usr/local/ipconf/fs/daemon/ivr.rb)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.194331
>>>>>>> [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154SET [ivr_path]=[/usr/local/ipconf/fs/daemon/ivr.rb]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 socket(localhost:8084 full)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309
>>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute answer()
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 answer()
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309
>>>>>>> [DEBUG] sofia_glue.c:3022 AUDIO RTP [sofia/external/
>>>>>>> 1000123402 at 192.168.0.154] 192.168.0.154 port 24232 -> 192.168.0.111
>>>>>>> port 4046 codec: 0 ms: 20
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.930309
>>>>>>> [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313
>>>>>>> [DEBUG] sofia_glue.c:3284 Set 2833 dtmf send payload to 101
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313
>>>>>>> [DEBUG] sofia_glue.c:3289 Set 2833 dtmf receive payload to 101
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313
>>>>>>> [DEBUG] mod_sofia.c:681 Local SDP sofia/external/
>>>>>>> 1000123402 at 192.168.0.154:
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313
>>>>>>> [DEBUG] switch_core_session.c:707 Send signal sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313
>>>>>>> [DEBUG] switch_channel.c:2827 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) Callstate Change RINGING -> ACTIVE
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.932313
>>>>>>> [NOTICE] mod_dptools.c:930 Channel [sofia/external/
>>>>>>> 1000123402 at 192.168.0.154] has been answered
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:47.933312
>>>>>>> [DEBUG] sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154entering state [completed][200]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.040285
>>>>>>> [DEBUG] sofia.c:4761 Channel sofia/external/1000123402 at 192.168.0.154entering state [ready][200]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294
>>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute
>>>>>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/WelcomeToTheConferenceSystem.wav)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.082294
>>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>>> channels 20ms
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.114277
>>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281
>>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute read(1 1
>>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav flex_digits 5000
>>>>>>> )
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 read(1 1
>>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Mainmenu.wav flex_digits 5000
>>>>>>> )
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:51:48.155281
>>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>>> channels 20ms
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670
>>>>>>> [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 1:1120
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.893670
>>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674
>>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute set(flex_digits)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.897674
>>>>>>> [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154SET [flex_digits]=[UNDEF]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673
>>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute read(1 1
>>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav
>>>>>>> flex_digits 5000 )
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 read(1 1
>>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/PleaseEnterYourPassCodeNow.wav
>>>>>>> flex_digits 5000 )
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:00.940673
>>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>>> channels 20ms
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662
>>>>>>> [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 8:640
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.113662
>>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663
>>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute set(flex_digits)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.116663
>>>>>>> [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154SET [flex_digits]=[UNDEF]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663
>>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute read(1 11
>>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 5000
>>>>>>> #,*)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 read(1 11
>>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 5000
>>>>>>> #,*)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.117663
>>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>>> channels 20ms
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:01.153658
>>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.053615
>>>>>>> [DEBUG] switch_rtp.c:3280 RTP RECV DTMF #:960
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618
>>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute set(flex_digits)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.055618
>>>>>>> [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154SET [flex_digits]=[UNDEF]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613
>>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute
>>>>>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/InternalHelp.wav)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.146613
>>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>>> channels 20ms
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603
>>>>>>> [DEBUG] switch_rtp.c:3280 RTP RECV DTMF #:800
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603
>>>>>>> [DEBUG] mod_dptools.c:1664 Digit #
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.333603
>>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617
>>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute
>>>>>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/YouAreTheOnlyPersonOnThisConference.wav)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:02.350617
>>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>>> channels 20ms
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.253520
>>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.322510
>>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute conference(15646 at teste
>>>>>>> +flags{waste})
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 conference(15646 at teste+flags{waste})
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492
>>>>>>> [DEBUG] mod_conference.c:5582 Raw Codec Activation Success L16 at 8000hz1 channel 20ms
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492
>>>>>>> [DEBUG] mod_conference.c:5627 Raw Codec Activation Success L16 at 8000hz1 channel 20ms
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:04.813492
>>>>>>> [DEBUG] switch_core_codec.c:116 sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Push codec L16:70
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469
>>>>>>> [DEBUG] switch_core_session.c:707 Send signal sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:05.280469
>>>>>>> [DEBUG] mod_conference.c:2557 Setup timer soft success interval: 20
>>>>>>> samples: 160
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.573928
>>>>>>> [DEBUG] switch_rtp.c:3280 RTP RECV DTMF *:960
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929
>>>>>>> [DEBUG]  mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:16.593929
>>>>>>> [DEBUG] switch_core_codec.c:141 sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Restore previous codec PCMU:0.
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927
>>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute read(1 1
>>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 read(1 1
>>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.097927
>>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>>> channels 20ms
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.133923
>>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.373913
>>>>>>> [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 1:1120
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912
>>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute set(flex_digits)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.375912
>>>>>>> [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154SET [flex_digits]=[UNDEF]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911
>>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute
>>>>>>> playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154playback(/usr/local/freeswitch/sounds/flex/app32/teste/SelfMuteOn.wav)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:17.490911
>>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>>> channels 20ms
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.233824
>>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828
>>>>>>> [DEBUG] switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154destroy/unlink session from object
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:19.235828
>>>>>>> [DEBUG] switch_core_codec.c:116 sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Push codec L16:70
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.433668
>>>>>>> [DEBUG] switch_rtp.c:3280 RTP RECV DTMF *:800
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667
>>>>>>> [DEBUG]  mod_conference.c:2021 Execute app: socket, localhost:8085 sync full
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:43.453667
>>>>>>> [DEBUG] switch_core_codec.c:141 sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Restore previous codec PCMU:0.
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 socket(localhost:8085 sync full)
>>>>>>>
>>>>>>> ====================================================================================================================================================
>>>>>>> While Inside this connection, a "conference 15646 kick [member_id of
>>>>>>> this channels]" command is executed by a fs_cli console and get stuck while
>>>>>>> waiting response.
>>>>>>>
>>>>>>> ====================================================================================================================================================
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850
>>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute read(1 1
>>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 read(1 1
>>>>>>> /usr/local/freeswitch/sounds/flex/app32/teste/Silence.wav flex_digits 2000 )
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.000850
>>>>>>> [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1
>>>>>>> channels 20ms
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.033846
>>>>>>> [DEBUG] switch_ivr_play_say.c:1649 done playing file
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.553821
>>>>>>> [DEBUG] switch_rtp.c:3280 RTP RECV DTMF 1:960
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825
>>>>>>> [DEBUG] switch_ivr.c:561 sofia/external/1000123402 at 192.168.0.154Command Execute set(flex_digits)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f EXECUTE sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 set(flex_digits)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:44.555825
>>>>>>> [DEBUG] mod_dptools.c:1060 sofia/external/1000123402 at 192.168.0.154SET [flex_digits]=[UNDEF]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244
>>>>>>> [DEBUG] switch_channel.c:2560 (sofia/external/
>>>>>>> 1000123402 at 192.168.0.154) Callstate Change ACTIVE -> HANGUP
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244
>>>>>>> [NOTICE] sofia.c:538 Hangup sofia/external/1000123402 at 192.168.0.154[CS_EXECUTE] [NORMAL_CLEARING]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244
>>>>>>> [DEBUG] switch_channel.c:2576 Send signal sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 [KILL]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.674244
>>>>>>> [DEBUG] switch_core_session.c:1114 Send signal sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 [BREAK]
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246
>>>>>>> [DEBUG] switch_core_session.c:2057 sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 skip receive message
>>>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246
>>>>>>> [DEBUG] switch_cpp.cpp:988 sofia/external/1000123402 at 192.168.0.154destroy/unlink session from object
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246
>>>>>>> [DEBUG] switch_core_session.c:2057 sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 skip receive message
>>>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246
>>>>>>> [DEBUG] switch_core_codec.c:116 sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 Push codec L16:70
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.675246
>>>>>>> [DEBUG] mod_conference.c:2815 Channel leaving conference, cause:
>>>>>>> NORMAL_CLEARING
>>>>>>> 16e09413-9cb0-4011-a635-f91933a35c0f 2011-05-13 10:52:56.757239
>>>>>>> [DEBUG] mod_conference.c:6104 sofia/external/
>>>>>>> 1000123402 at 192.168.0.154 skip receive message [UNBRIDGE] (channel is
>>>>>>> hungup already)
>>>>>>>
>>>>>>> I hope this info helps.
>>>>>>>
>>>>>>> 2011/5/12 Michael Collins <msc at freeswitch.org>
>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Thu, May 12, 2011 at 12:27 PM, Leonardo P. Bidinoto <
>>>>>>>> leonardo.bidinoto at voicetechnology.com.br> wrote:
>>>>>>>>
>>>>>>>>> Hi Michael,
>>>>>>>>>
>>>>>>>>> Im not using to any cdr module.
>>>>>>>>
>>>>>>>>
>>>>>>>> I would recommend that you do several things:
>>>>>>>>
>>>>>>>> #1 - update to latest git
>>>>>>>> #2 - rotate logs
>>>>>>>> #3 - enable uuid logging (see logfile.conf.xml - param name "uuid")
>>>>>>>> #4 - reproduce the symptom with a single call (if possible)
>>>>>>>> #5 - pastebin the log for the uuid in question and link to it in
>>>>>>>> this thread
>>>>>>>>
>>>>>>>> From there hopefully we'll get a clue as to what is happening.
>>>>>>>> -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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Leonardo Pires Bidinoto
>>>>>>> Voice Technology
>>>>>>> www.voicetechnology.com.br
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Leonardo Pires Bidinoto
>>>>> Voice Technology
>>>>> www.voicetechnology.com.br
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Leonardo Pires Bidinoto
>>> Voice Technology
>>> www.voicetechnology.com.br
>>>
>>
>>
>>
>> --
>> Leonardo Pires Bidinoto
>> Voice Technology
>> www.voicetechnology.com.br
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Leonardo Pires Bidinoto
> Voice Technology
> www.voicetechnology.com.br
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Leonardo Pires Bidinoto
Voice Technology
www.voicetechnology.com.br
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From roger.castaldo at gmail.com  Mon Jul 18 17:22:21 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Mon, 18 Jul 2011 09:22:21 -0400
Subject: [Freeswitch-users] .Net interface
In-Reply-To: <CAD-qRkiA5ht+ubua60Q0hjZbRLTWTrOoMuTR6uYc+1HffrDvEw@mail.gmail.com>
References: <CAAYf2Lt5QJ+WjzBxY257XsfH021dAMDvGqACosbKwmPt8NOjjg@mail.gmail.com>
	<CAFiqYuk3J1452vfF7tgJgJh_rEt73LgtVdQW6BJnwokSc+A3XQ@mail.gmail.com>
	<CAAYf2LuJYW7GMGa0DmrnsOjw-zQQjqEvmkVvvS0u2SzcX=kB7Q@mail.gmail.com>
	<CAD-qRkiA5ht+ubua60Q0hjZbRLTWTrOoMuTR6uYc+1HffrDvEw@mail.gmail.com>
Message-ID: <CAAYf2LtMCBWzxDgpgHM3X4gDoj5tDTJJLLKnWC4jjxbFw=eMNw@mail.gmail.com>

No No what I meant is the .Net ESL library looks to me to be just a wrapper
to wrap around an unmanaged dll whereas the library I have written is
straight C# only code, no wrapped management or anything it is a pure .Net
dll that only requires you to use at least .Net 2.5.

On Mon, Jul 18, 2011 at 7:46 AM, Christopher Rienzo <cmrienzo at gmail.com>wrote:

> ESL is a library compiled separately from FreeSWITCH.  There are no
> dependencies between them.
>
>
>
>
> On Mon, Jul 18, 2011 at 7:31 AM, Roger Castaldo <roger.castaldo at gmail.com>wrote:
>
>> Yes there is a contributed library made of the esl which is a managed
>> wrapper of the cpp code.  The library I have designed and built requires no
>> external references it is a pure socket implementation that is compiled
>> independently of freeswitch to allow interfacing with freeswitch and is an
>> attempt to wrap all command calls in easy to use function calls to allow
>> less freeswitch experienced .Net developers to interface with freeswitch
>> even easier.
>>
>>
>> On Mon, Jul 18, 2011 at 3:58 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> Thanks for the contribution...
>>>
>>> Hate to tell you this though, but do you do know there's already one
>>> in the FreeSWITCH trunk? Look in libs/esl/managed of the git checkout.
>>>
>>> Although yours might of course provide slightly different
>>> functionality so could still be useful in addition to the official
>>> one.
>>>
>>> -Steve
>>>
>>>
>>> On 18 July 2011 01:47, Roger Castaldo <roger.castaldo at gmail.com> wrote:
>>> > Hi everyone this is just to let everyone know that I have developed a
>>> C#
>>> > based .Net socket library to allow a freeswitch server to be interfaced
>>> with
>>> > using the event socket and outbound socket.  The library is hosted on
>>> google
>>> > code, the link is http://code.google.com/p/freeswitch-socket-dotnet/
>>> > If anyone is interested in helping with the development please let me
>>> know.
>>> > Feel free to use it and expect updates to it as time permits, my goal
>>> is to
>>> > try and make the library somewhat idiot proof by implementing all
>>> available
>>> > calls that could be handled on the server side when controlling a phone
>>> > call.
>>> >
>>> > _______________________________________________
>>> > 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
>>> >
>>> >
>>>
>>> _______________________________________________
>>> 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
>>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 jeff at jefflenk.com  Mon Jul 18 17:46:26 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 18 Jul 2011 06:46:26 -0700 (PDT)
Subject: [Freeswitch-users] .Net interface
In-Reply-To: <CAAYf2LtMCBWzxDgpgHM3X4gDoj5tDTJJLLKnWC4jjxbFw=eMNw@mail.gmail.com>
References: <CAAYf2Lt5QJ+WjzBxY257XsfH021dAMDvGqACosbKwmPt8NOjjg@mail.gmail.com>
	<CAFiqYuk3J1452vfF7tgJgJh_rEt73LgtVdQW6BJnwokSc+A3XQ@mail.gmail.com>
	<CAAYf2LuJYW7GMGa0DmrnsOjw-zQQjqEvmkVvvS0u2SzcX=kB7Q@mail.gmail.com>
	<CAD-qRkiA5ht+ubua60Q0hjZbRLTWTrOoMuTR6uYc+1HffrDvEw@mail.gmail.com>
	<CAAYf2LtMCBWzxDgpgHM3X4gDoj5tDTJJLLKnWC4jjxbFw=eMNw@mail.gmail.com>
Message-ID: <1310996786408-6594779.post@n2.nabble.com>

There is also an already existing pure c# interface in git contrib too -
verifier\EventSocket\trunk
Haven't looked at it for a while so I'm not sure if its status.

--
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From jeff at jefflenk.com  Mon Jul 18 18:12:16 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 18 Jul 2011 07:12:16 -0700 (PDT)
Subject: [Freeswitch-users] Google Voice for Windows version of FS
In-Reply-To: <1310911238220-6591923.post@n2.nabble.com>
References: <1310468651.13286.YahooMailNeo@web121612.mail.ne1.yahoo.com>
	<1310479518002-6575189.post@n2.nabble.com>
	<1310482854.79304.YahooMailNeo@web121610.mail.ne1.yahoo.com>
	<1310483885842-6575490.post@n2.nabble.com>
	<1310511452.58175.YahooMailNeo@web121618.mail.ne1.yahoo.com>
	<1310571922569-6579538.post@n2.nabble.com>
	<1310911238220-6591923.post@n2.nabble.com>
Message-ID: <1310998336606-6594878.post@n2.nabble.com>

Yahoo2003,

I was hoping you might take a few minutes if there were any notes you had
when you built with vs2010 and make a few improvments to the Wiki for
GnuTls?

Thanks
Jeff

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Google-Voice-for-Windows-version-of-FS-tp6574607p6594878.html
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From jeff at jefflenk.com  Mon Jul 18 19:20:13 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 18 Jul 2011 08:20:13 -0700 (PDT)
Subject: [Freeswitch-users] Hanged up callers doubt
In-Reply-To: <CALV5c5NGFX6feJAf9bXMwWju-wCdLSp6Gi8r4aQ63eC4x_vwqg@mail.gmail.com>
References: <BANLkTi=WN97QMJcpwyEGQNvRxBW1c=UsuQ@mail.gmail.com>
	<BANLkTiky4M2ZgTMS228-DJ6oaEsmkpG19g@mail.gmail.com>
	<BANLkTimzsNxVyFpLf6x8X4E7P58FYwdHng@mail.gmail.com>
	<BANLkTinYynZsX6=zg-yyeP7hEQ7-q5J9+Q@mail.gmail.com>
	<BANLkTim=ueSmFA_ug2MbbLFMBh0362_2Sg@mail.gmail.com>
	<BANLkTikV_-=Zkv8zOQVcWyg5Gc06UtRxjw@mail.gmail.com>
	<2C18EE80-BB65-4A93-9A44-627B6154DD5F@ipeva.fr>
	<BANLkTimYO9kV2VFcraTCjn4uLOtAu44R3Q@mail.gmail.com>
	<793379B8-D1FC-47F4-8387-4E2789A69C72@ipeva.fr>
	<CALV5c5NGFX6feJAf9bXMwWju-wCdLSp6Gi8r4aQ63eC4x_vwqg@mail.gmail.com>
Message-ID: <1311002413967-6595161.post@n2.nabble.com>

Please report this issue to Jira along with what you did to reproduce the
problem and steps taken to fix.


--
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From ijurado at econcept.es  Mon Jul 18 17:07:11 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Mon, 18 Jul 2011 15:07:11 +0200
Subject: [Freeswitch-users] Transfers and billsecs
Message-ID: <CABj_V-kBEwXo08RhESMhUaHJrUFZ28heFfFEG51xw5z5Ri3WbA@mail.gmail.com>

Hello all,

First of all, I want to apologize for the length of this message.  I
wouldn't like to waste anyone's time.

I'm new to this list and also fairly new to the VoIP world.  I work for
a company that provides some basic telephony services using FreeSWITCH.
Recelty, we sketched up a plan to extend the features of our service to
the customers; so we started experimenting first.

Our setup is a basic HTTP server that controls FreeSWITCH through
mod_xml_curl and processes call information afterwards, with the
information provided by mod_xml_cdr.  In general, each customer has its
own context and we are trying to come along with a general billing
system that tries to be as fair as possible.

In order to reduce billing system complexity, we are completing channel
information from the dialplan so we can inspect each CDR generated
independently, and be able to discard it or process it on its own,
without having to check other related CDRs.

Obviously, this is becoming harder and harder after we started dealing
with call transfers, either attended or blind.  Consider the following
scenario:

  1. We have four customers: A, B, C and D.

  2. A calls B, the complete billing information is in the a-leg (from A
     to FS) and it will be charged as a call to B.

  3. After some time (say 20 seconds), B makes a blind transfer to C.
     So a new b-leg is created (from FS to C), the channel between FS
     and B is closed and a CDR is generated.  Conceptually, this is a
     call from B to C which should be charged to B, not to A.

  4. Again, after some time (say 30 seconds), C makes a blind transfer
     to D.  Similar procedure: new channel from FS to D and the channel
     from FS to C terminates, generating a CDR.  This time, we should
     charge a call from C to D.

  5. After 15 seconds, A or D hang up and two more CDRs are generated.

In the end, the billing system should charge:

  - The call between A and B, which had a duration of 65 seconds.
  - The call between B and C, which had a duration of 45 seconds.
  - The call between C and D, which had a duration of 15 seconds.

And here is where the problems arise, because only the a-leg (from A to
FS) contains the proper value of <billsec> for the "logical" call
between A and B.  The CDR of the FS to C channel has a <billsec> of 30
seconds and the one for the FS to D channel has 15 seconds (all aprox).

However, all CDRs appear to have the same <call_uuid> so they can be
easily related.  The problem is that we would like to avoid this CDR
dependency if we can, mainly because the background process that treats
with CDR information is stateless at the moment.

The purpose of this message is to survey about what other people is
doing about this, as it seems to be the source of some headaches
according to the list archives.  I think that, in the end, we will have
to mix up different CDRs but we may be missing some hidden functionality
of FreeSWITCH.

Thank you for your time :-)

P.S: Testing the same call flow, but with attended transfers it is even
more complex in terms of CDR processing, but we would like to have some
idea on how to solve this one first.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From sip98765 at googlemail.com  Mon Jul 18 17:24:18 2011
From: sip98765 at googlemail.com (max)
Date: Mon, 18 Jul 2011 15:24:18 +0200
Subject: [Freeswitch-users] How to creat an outgoing media channal for
	non-SIP?
Message-ID: <1310995458.1678.30.camel@anyong>

Hello,

I want to write a quick and dirty extension as a prove of concept for
Freeswitch translating SIP into another protocol(e.g. proprietary RUF
XML-RTP for calls). For that I chose to use mod_erlang_event to handle
the handshake and the other standards server. The media should be
handled completely by Freeswitch. Please see the attached picture.
The focus on this project is to get used to Erlang as well as to
Freeswitch.

To the problem:
While creating an outgoing media channel via Erlang to the non SIP I got
stuck. Can I use originate in that context and if yes, how? If no, is
there any other command?

Thank you,
Max
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From sip98765 at googlemail.com  Mon Jul 18 19:09:33 2011
From: sip98765 at googlemail.com (max)
Date: Mon, 18 Jul 2011 17:09:33 +0200
Subject: [Freeswitch-users] How to creat an outgoing media channal for
	non-SIP?
Message-ID: <1311001773.19823.4.camel@anyong>

Hello,


First of all I am sorry if you receive this mail double. I sent it but I
didn't get a copy on the mailing list.
I want to write a quick and dirty extension as a prove of concept for
Freeswitch translating SIP into another protocol(e.g. proprietary RUF
XML-RTP for calls). For that I chose to use mod_erlang_event to handle
the handshake and the other standards server. The media should be
handled completely by Freeswitch.
The focus on this project is to get used to Erlang as well as to
Freeswitch so solutions should contain this two parts.

To the problem:
While creating an outgoing media channel via Erlang to the non SIP I got
stuck. Can I use originate in that context and if yes, how? If no, is
there any other command to realise that?
Also I would need to know how to control/close this media channel.

Thank you,
Max 



From spencer at 5ninesolutions.com  Mon Jul 18 20:03:16 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 18 Jul 2011 09:03:16 -0700
Subject: [Freeswitch-users] Looking for US based providers offering t.38
	that work with freeswitch
In-Reply-To: <CAMgKNJUYwgPF6YoF5ckgLwYZfd4OfzOT0wJ0hneOBKp5KSHXLQ@mail.gmail.com>
References: <CAMgKNJUYwgPF6YoF5ckgLwYZfd4OfzOT0wJ0hneOBKp5KSHXLQ@mail.gmail.com>
Message-ID: <7290A820-C7DB-4A3D-90C3-63485F0B9212@5ninesolutions.com>

Sotel Systems.  They use an OpenSIPS and FreeSWITCH based platform in front of Level 3 and never proxy media.  And their support is fantastic.

Spencer
 
On Jul 18, 2011, at 3:48 AM, Tony Graziano wrote:

> So far we have only been able to find a provider using sonus switches
> for this. Is anyone successfully using a trunk provider to receive
> faxes via t.38 in freeswitch they are willing to share?
> 
> 
> Thanks,
> 
> Tony
> 
> _______________________________________________
> 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 steveayre at gmail.com  Mon Jul 18 20:06:41 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 18 Jul 2011 17:06:41 +0100
Subject: [Freeswitch-users] How to creat an outgoing media channal for
	non-SIP?
In-Reply-To: <1310995458.1678.30.camel@anyong>
References: <1310995458.1678.30.camel@anyong>
Message-ID: <CAFiqYu=SM9bNYVv7dByw_sC+Bpx6VLSjMdgbEYUM=AtLdHhoHQ@mail.gmail.com>

Ordinarily endpoint modules implement a signalling endpoint. They
register a dialstring prefix and handlers for generic call states. The
either handle media themselves (eg mod_skypopen) or use the freeswitch
rtp stack (mod_sofia). The freeswitch core handles all the
interoperation between any combination of protocols between modules
using this endpoint interface.

mod_erlang_event doesn't support that interface. It sounds to me like
you would want to create a new mod_erlang_endpoint module implementing
that interface written in erlang. Since sessions require certain
callbacks for the call states and for reading/writing media you would
need to implement that interface as its out of the scope of the events
subsystem.

FreeSWITCH would then handle all calls through the dialplan completely
irrespective of which protocol is being used to receive the call
(though channel variables will let you see which and do
module-specific stuff) and let you bridge to any of the protocols
freeswitch supports.

-Steve


On 18 July 2011 14:24, max <sip98765 at googlemail.com> wrote:
> Hello,
>
> I want to write a quick and dirty extension as a prove of concept for
> Freeswitch translating SIP into another protocol(e.g. proprietary RUF
> XML-RTP for calls). For that I chose to use mod_erlang_event to handle
> the handshake and the other standards server. The media should be
> handled completely by Freeswitch. Please see the attached picture.
> The focus on this project is to get used to Erlang as well as to
> Freeswitch.
>
> To the problem:
> While creating an outgoing media channel via Erlang to the non SIP I got
> stuck. Can I use originate in that context and if yes, how? If no, is
> there any other command?
>
> Thank you,
> Max
>
> _______________________________________________
> 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 frank at rosengart.de  Mon Jul 18 20:22:48 2011
From: frank at rosengart.de (Frank Rosengart)
Date: Mon, 18 Jul 2011 18:22:48 +0200
Subject: [Freeswitch-users] BLF (on Snom)
Message-ID: <4E245DD8.3080604@rosengart.de>

Hi,

I'm tearing my hair about getting FS & BLF to work with Snom 360s.
I think it should work, but I'm stuck with debugging. Maybe someone can
help.


What I know:
The whole setup was working fine on Asterisk 1.4, including BLF.

The phone is configured with a function key
 => BLF, <to_be_monitored>@<mydomain>

The phone sends SUBSCRIBE to FS and receives a '202 Accepted'

The sqlite db lists this subscription with SELECT * FROM sip_subscriptions.

How can I trigger FS to send the NOTIFY dialog-info-XML?

* I tried on the cli to send
  presence in to_be_monitored at mydomain inuse inuse

and
  <action application="set" data="result=${presence(in
to_be_monitored at mydomain inuse inuse)}"/>

as well as the

<param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>

or setting the presence_id variable from the dialplan, like suggested in
the "persistant BLF' ML thread a while ago.

There is no NOTIFY packet leaving FS, or any reaction of LED on the phone.


Thanks for any hints.

Frank



From brian at freeswitch.org  Mon Jul 18 20:27:02 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 Jul 2011 11:27:02 -0500
Subject: [Freeswitch-users] BLF (on Snom)
In-Reply-To: <4E245DD8.3080604@rosengart.de>
References: <4E245DD8.3080604@rosengart.de>
Message-ID: <7521EAF2-ADEF-465A-9917-7DB86F2B9C54@freeswitch.org>

It'll just work if you have manage_presence enabled on the profile? try the default configs it'll just work.

/b

On Jul 18, 2011, at 11:22 AM, Frank Rosengart wrote:

> Hi,
> 
> I'm tearing my hair about getting FS & BLF to work with Snom 360s.
> I think it should work, but I'm stuck with debugging. Maybe someone can
> help.
> 
> 
> What I know:
> The whole setup was working fine on Asterisk 1.4, including BLF.
> 
> The phone is configured with a function key
> => BLF, <to_be_monitored>@<mydomain>
> 
> The phone sends SUBSCRIBE to FS and receives a '202 Accepted'
> 
> The sqlite db lists this subscription with SELECT * FROM sip_subscriptions.
> 
> How can I trigger FS to send the NOTIFY dialog-info-XML?
> 
> * I tried on the cli to send
>  presence in to_be_monitored at mydomain inuse inuse
> 
> and
>  <action application="set" data="result=${presence(in
> to_be_monitored at mydomain inuse inuse)}"/>
> 
> as well as the
> 
> <param name="dial-string"
> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
> 
> or setting the presence_id variable from the dialplan, like suggested in
> the "persistant BLF' ML thread a while ago.
> 
> There is no NOTIFY packet leaving FS, or any reaction of LED on the phone.
> 
> 
> Thanks for any hints.
> 
> Frank
> 
> 
> _______________________________________________
> 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 spencer at 5ninesolutions.com  Mon Jul 18 21:03:31 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 18 Jul 2011 10:03:31 -0700
Subject: [Freeswitch-users] RTP Error
Message-ID: <684D78F9-46E9-402A-AB0B-BC0DD0A735E7@5ninesolutions.com>

Hello all,
I have been getting the following error in my logs, very intermittently.  Any idea what this means and where I could look to resolve it?  This instance was built from git on July 8.

2011-07-18 11:50:39.855581 [ERR] sofia.c:5748 RTP Error!
2011-07-18 11:50:44.459669 [ERR] sofia.c:5748 RTP Error!


Thanks,
Spencer



From sip98765 at googlemail.com  Mon Jul 18 21:14:12 2011
From: sip98765 at googlemail.com (max)
Date: Mon, 18 Jul 2011 19:14:12 +0200
Subject: [Freeswitch-users] How to creat an outgoing media channal for
 non-SIP?
In-Reply-To: <CAFiqYu=SM9bNYVv7dByw_sC+Bpx6VLSjMdgbEYUM=AtLdHhoHQ@mail.gmail.com>
References: <1310995458.1678.30.camel@anyong>
	<CAFiqYu=SM9bNYVv7dByw_sC+Bpx6VLSjMdgbEYUM=AtLdHhoHQ@mail.gmail.com>
Message-ID: <1311009252.19823.31.camel@anyong>

Hello Steve,

Thank you for your quick response. I appreciate the information
provided. You stopped me from running in the wrong direction.

I got you that way that it is impossible, if I'm not going to write
another mod_erlang_endpoint.

The session handling for the other Protocol is as simple and I thought I
can just tell Freeswitch to handle the Sip session. The only missing
thing was to get the non SIP media data into Freeswitch (attached to the
call, what would probably be exactly what the new endpoint would have to
do) by using the rtp stack.

regards,
Max


Am Montag, den 18.07.2011, 17:06 +0100 schrieb Steven Ayre:
> Ordinarily endpoint modules implement a signalling endpoint. They
> register a dialstring prefix and handlers for generic call states. The
> either handle media themselves (eg mod_skypopen) or use the freeswitch
> rtp stack (mod_sofia). The freeswitch core handles all the
> interoperation between any combination of protocols between modules
> using this endpoint interface.
> 
> mod_erlang_event doesn't support that interface. It sounds to me like
> you would want to create a new mod_erlang_endpoint module implementing
> that interface written in erlang. Since sessions require certain
> callbacks for the call states and for reading/writing media you would
> need to implement that interface as its out of the scope of the events
> subsystem.
> 
> FreeSWITCH would then handle all calls through the dialplan completely
> irrespective of which protocol is being used to receive the call
> (though channel variables will let you see which and do
> module-specific stuff) and let you bridge to any of the protocols
> freeswitch supports.
> 
> -Steve
> 
> 
> On 18 July 2011 14:24, max <sip98765 at googlemail.com> wrote:
> > Hello,
> >
> > I want to write a quick and dirty extension as a prove of concept for
> > Freeswitch translating SIP into another protocol(e.g. proprietary RUF
> > XML-RTP for calls). For that I chose to use mod_erlang_event to handle
> > the handshake and the other standards server. The media should be
> > handled completely by Freeswitch. Please see the attached picture.
> > The focus on this project is to get used to Erlang as well as to
> > Freeswitch.
> >
> > To the problem:
> > While creating an outgoing media channel via Erlang to the non SIP I got
> > stuck. Can I use originate in that context and if yes, how? If no, is
> > there any other command?
> >
> > Thank you,
> > Max
> >
> > _______________________________________________
> > 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
> >
> >
> 
> _______________________________________________
> 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 djbinter at gmail.com  Mon Jul 18 22:22:26 2011
From: djbinter at gmail.com (DJB International)
Date: Mon, 18 Jul 2011 11:22:26 -0700
Subject: [Freeswitch-users] stripping sip-rh headers at the edge of my
	network
In-Reply-To: <CAO47HpfcjUJj_9UVO8dFEQq8M6gMaZs5DKvUP4H_hW8tkC324A@mail.gmail.com>
References: <CAO47HpdG8qJeH4uoXDFB31Jf7_1oNVbmoJQ52DUS6WcmxuAj=Q@mail.gmail.com>
	<CAO47HpfcjUJj_9UVO8dFEQq8M6gMaZs5DKvUP4H_hW8tkC324A@mail.gmail.com>
Message-ID: <CAAB3jWv-i9ABnNn6fom+VSTGEvaCQhrfyGJ5dyWQ1DO9p1n=WA@mail.gmail.com>

I thought "unset" should work:
http://wiki.freeswitch.org/wiki/Strip_SIP_Headers



On Mon, Jul 18, 2011 at 5:29 AM, jay binks <jaybinks at gmail.com> wrote:

> hmm this isnt good..  no replies..
>
> so, seems I cant strip these at all with freeswitch..
>
> using unset and even trying to overwrite the value like this
>
> <action application="set" data="sip_ph_X-someheader=blah"/>
> <action application="set" data="sip_rh_X-someheader=blah"/>
> <action application="set" data="sip_h_X-someheader=blah"/>
>
> fails...
> since FS just splats on top of whatever I set , when the sip message comes
> in from upstream..
> (  in this case in the 180 / 183 progress )
>
> I need a way to strip headers from 180/183 .. preferably I could have a
> regex to filter any starting with X :)
> someone please point me in the right direction
>
> Jay
>
>
>
>
>
>
> On Fri, Jul 15, 2011 at 4:54 PM, jay binks <jaybinks at gmail.com> wrote:
>
>> Inside my network I use sip_ph , sip_rh & sip_h headers to pass data
>> around between boxes.
>>
>> however I need to strip these off at the edge ( SBC's running Freeswitch
>> also )
>>
>> will unset work with these headers in the 200 OK ( and other sip messages
>> ) ?
>> if not, how can I achieve this ?
>>
>> --
>> Sincerely
>>
>> Jay
>>
>
>
>
> --
> Sincerely
>
> Jay
>
> _______________________________________________
> 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 acrow at integrafin.co.uk  Mon Jul 18 22:25:24 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 18 Jul 2011 19:25:24 +0100
Subject: [Freeswitch-users] rfc 4662
Message-ID: <4E247A94.6030605@integrafin.co.uk>

Hi,

Does freeswitch support RFC4662 SIP resource lists notification (on 
phones it seems to be called Eventlist BLF)? It would seem a good thing 
to reduce traffic for monitoring presence/status.

There seems to be at least one post on Trixbox forums indicating it 
does, however I can find nothing in the Wiki.

Thanks,

Alex


From spautz2 at telefaks.biz  Mon Jul 18 22:13:53 2011
From: spautz2 at telefaks.biz (David Spautz)
Date: Mon, 18 Jul 2011 20:13:53 +0200
Subject: [Freeswitch-users] domain-Element in SIP-Profiles
Message-ID: <4E2477E1.8020001@telefaks.biz>

Hi FreeSWITCH users,

currently, i am write my bachelor thesis about FreeSWITCH and the 
communication with external application servers. In this regard,
I am faced with the question of what function the <domains>-element of 
sofia profiles should have. What is the concept behind it?
And what are the differences to the domain attribute in the <profile 
name='...' domain='...'>-element of sofia.conf.xml file?

It would be helpful if someone could describe one or more scenarios to 
use this domains-element.

Is it a way to register the SIP Profile to all user specified gateways 
in addition?


<domains>
<!-- indicator to parse the directory for domains with parse="true" to 
get gateways-->
<!--<domain name="$${domain}" parse="true"/>-->
<!-- indicator to parse the directory for domains with parse="true" to 
get gateways and alias every domain to this profile -->
<!--<domain name="all" alias="true" parse="true"/>-->
<domain name="all" alias="true" parse="false"/>
</domains>


Thanks for all help


David
Anh?ren
Umschrift



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From frank at rosengart.de  Mon Jul 18 23:19:26 2011
From: frank at rosengart.de (Frank Rosengart)
Date: Mon, 18 Jul 2011 21:19:26 +0200
Subject: [Freeswitch-users] BLF (on Snom)
In-Reply-To: <7521EAF2-ADEF-465A-9917-7DB86F2B9C54@freeswitch.org>
References: <4E245DD8.3080604@rosengart.de>
	<7521EAF2-ADEF-465A-9917-7DB86F2B9C54@freeswitch.org>
Message-ID: <4E24873E.9050908@rosengart.de>

On 07/18/2011 06:27 PM, Brian West wrote:
> It'll just work if you have manage_presence enabled on the profile?
> try the default configs it'll just work.

Sorry, I forget to mention. Of course, it's enabled. Otherwise the
sqlite db wouldn't have the subscription stored.

Is there any description how the whole subscription thing works? I guess
FS is looking up the sqlite on every presence event. And if there is a
matching subscription for user at domain, a notification is triggered. Correct?
The sofia global debug presence is far too verbose to be useful...

Thanks
Frank


From msc at freeswitch.org  Mon Jul 18 23:24:07 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Jul 2011 12:24:07 -0700
Subject: [Freeswitch-users] incoming call to FS
In-Reply-To: <CAKnRR0ZkKt=QjB5a7eKdtFvYW+pq2SR7u-AFXr1TNPfGNfdObg@mail.gmail.com>
References: <CAKnRR0ZkKt=QjB5a7eKdtFvYW+pq2SR7u-AFXr1TNPfGNfdObg@mail.gmail.com>
Message-ID: <CAKzWOxV_2PwB0Z5H9KY6c=unxAr1u-08KS1W+d4HthSA2sjYvw@mail.gmail.com>

On Sun, Jul 17, 2011 at 5:58 AM, budi wibowo <bwibowo at gmail.com> wrote:

> hi
> i have sip server and want to connect the sip server to FS then call will
> be routed using mod_dingaling, mod_dingaling already active
> sipserver--->FS---mod_dingaling
>
> from FS it will be incoming sip call, i already define sipserver arameter
> in external sip_profile, then make call but i cant see any call coming from
> fs_cli
> anything missing in my config?
>
Without seeing your config or a debug log it is difficult to tell.
-MC
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From msc at freeswitch.org  Mon Jul 18 23:34:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Jul 2011 12:34:17 -0700
Subject: [Freeswitch-users] Leg-A is automatically disconnected on Leg-B
 orginate failure
In-Reply-To: <CAMLBmR+yqetjWU+yGOypLtwOj8Ryz8rbEp41tW0yHS53w-pgWg@mail.gmail.com>
References: <1308213151543-6482192.post@n2.nabble.com>
	<BANLkTi=xYgBTPVGxVGmk9f2=bmX5X2vF=Q@mail.gmail.com>
	<BANLkTikhPmjN5HeeJHBnQt1ivA8zbtCnYA@mail.gmail.com>
	<BANLkTi=m7oMix1ZRC9URtRuAbsG2fWyhMQ@mail.gmail.com>
	<BANLkTimxBU8m311LVKMWcFOjukwLg+ZUGQ@mail.gmail.com>
	<BANLkTinZJf3BxB7QHefnmOvxQmVW2r4dTg@mail.gmail.com>
	<BANLkTimiExLew_1cjMuQVpF3CKkR8mrGXQ@mail.gmail.com>
	<BANLkTi=ZvOZ99qVxUeJAtpjtwP5r=hfVkg@mail.gmail.com>
	<BANLkTimj4iSWFiHuBSKskz7NjF2bCEWnaw@mail.gmail.com>
	<BANLkTikyyAZaJzQx2tbiVOxHxx5mCbUdvQ@mail.gmail.com>
	<CAMLBmRLtAeU9m6HwOW3o9FA2pZFRBiXk2P82Z7msv218B4AxCA@mail.gmail.com>
	<CAMLBmR+yqetjWU+yGOypLtwOj8Ryz8rbEp41tW0yHS53w-pgWg@mail.gmail.com>
Message-ID: <CAKzWOxXAx3LpL0pA2cF4=0QsEKF_e=QhVp5PRua9EbnwLDCABg@mail.gmail.com>

FYI,

Tony is on vacation for a bit, so we'll have to do our best without him.
Could you please do two things? First, turn off the sofia debugging. The SIP
trace is fine, but all that other stuff is just noise. Second, can you show
the call log from the very beginning, including any dialplan hunting.

Thanks,
MC

On Mon, Jul 18, 2011 at 2:25 AM, David Ma <mays.david at gmail.com> wrote:

> Hello Anthony,
>
> The same log in better formatting is here:
>
> http://pastebin.freeswitch.org/16840
>
> Regards,
> D.Ma
>
>
> On Mon, Jul 18, 2011 at 5:18 PM, David Ma <mays.david at gmail.com> wrote:
>
>> Hi Anthony,
>>
>> I've tried channel variable uuid_bridge_continue_on_cancel=true. However,
>> this doesn't solve the problem. The log is in the following on pastebin,
>> please kindly help investigate why.
>>
>> http://pastebin.freeswitch.org/16839
>>
>> Thanks a lot!
>>
>> Best regards,
>> D.Ma
>>
>>
>> On Wed, Jun 29, 2011 at 10:53 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> The only side effect is that it will give you the behavior you want in
>>> this particular situation.
>>>
>>>
>>> On Wed, Jun 29, 2011 at 4:35 AM, David Ma <mays.david at gmail.com> wrote:
>>> > Hi Anthony,
>>> >
>>> > Thanks very much for the information. I appreciate your advice. It is
>>> great
>>> > to learn about such a parameter.
>>> >
>>> > FS wiki has only a little description about this parameter. Is there
>>> any
>>> > side effect, or caution for using such a parameter?
>>> >
>>> > Thanks,
>>> > D.Ma
>>> >
>>> > On Wed, Jun 29, 2011 at 12:08 AM, Anthony Minessale
>>> > <anthony.minessale at gmail.com> wrote:
>>> >>
>>> >> The cases where you have this problem arise because you are
>>> >> originating to the B leg who answers 183 (early media) then hangs up
>>> >> before the call was answered.
>>> >>
>>> >> set the channel variable uuid_bridge_continue_on_cancel=true on the A
>>> >> leg to change the behavior to what you want.
>>> >>
>>> >>
>>> >> On Mon, Jun 27, 2011 at 10:09 AM, Michael Collins <msc at freeswitch.org
>>> >
>>> >> wrote:
>>> >> > This makes my eyes bleed. Can you please put this on
>>> >> > pastebin.freeswitch.org? Use "FreeSWITCH Log" as the syntax
>>> highlight.
>>> >> > -MC
>>> >> >
>>> >> > On Sun, Jun 26, 2011 at 11:27 PM, David Ma <mays.david at gmail.com>
>>> wrote:
>>> >> >>
>>> >> >> Hi Michael,
>>> >> >>
>>> >>
>>> >> --
>>> >> 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
>>> >
>>> >
>>> > _______________________________________________
>>> > 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
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> 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
>>>
>>
>>
>
> _______________________________________________
> 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 acrow at integrafin.co.uk  Tue Jul 19 00:25:48 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 18 Jul 2011 21:25:48 +0100
Subject: [Freeswitch-users] BLF (on Snom)
In-Reply-To: <4E245DD8.3080604@rosengart.de>
References: <4E245DD8.3080604@rosengart.de>
Message-ID: <4E2496CC.4050904@integrafin.co.uk>

On 18/07/11 17:22, Frank Rosengart wrote:
> Hi,
>
> I'm tearing my hair about getting FS&  BLF to work with Snom 360s.
> I think it should work, but I'm stuck with debugging. Maybe someone can
> help.
>

Hi Frank, all,

I have been testing with Snom 370s for a while. With <param 
name="manage-presence" value="true"/> in the SIP profile it seems to 
work as expected. However compared to Asterisk I do find the resultant 
pickup of a ringing BLF (either using Extension <extension no.> or BLF 
<extension no.|**> on the function keys on the Snom web config) to be a 
bit slow. It takes 2-3 seconds from pressing the flashing button for the 
call to be picked up.

I think the above delay is due to the time it takes for the original 
b-leg to be removed. Is there any way to speed this up?

Cheers

Alex

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From yungwei at resolvity.com  Tue Jul 19 00:29:02 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Mon, 18 Jul 2011 16:29:02 -0400
Subject: [Freeswitch-users] What does "delete interp" do?
Message-ID: <33095823FD21DF429B481B5163264B7950CBC03B6E@VMBX102.ihostexchange.net>

Hi,

I came across the following statement in SpeechTools.jm, and I'm wondering what it does. Thanks.

delete interp;


From cthompson at stonecracker.com  Tue Jul 19 01:51:35 2011
From: cthompson at stonecracker.com (Chris Thompson)
Date: Mon, 18 Jul 2011 21:51:35 +0000 (UTC)
Subject: [Freeswitch-users] please help!!! how to set flag 'endconf' in
	bridging conference
References: <E826C94A3C6BE44198FAF4D66B5AB4D81BDA7E5903@VA3DIAXVS351.RED001.local>
	<4E0CB983.8060407@gosilverplus.com>
	<BANLkTikb-xbvLwzsSv7nQJZncx8y_v+hpw@mail.gmail.com>
	<4E0CE1B7.60609@gosilverplus.com>
	<BANLkTi=Cmuz6a3K8smK3Nr46E2ptg=pZdQ@mail.gmail.com>
	<4E0D233B.6010609@gosilverplus.com>
	<BANLkTinmHti_769mQ5Xbs4sV6U0em-VZaw@mail.gmail.com>
	<4E0DF997.2060105@gosilverplus.com>
	<BANLkTimpd--c8JabTrd1ki3rFFt+tZY3bg@mail.gmail.com>
	<4E0DFCEE.5010203@gosilverplus.com>
	<BANLkTi=URVjLSShw+gmnijEkRNfrGiy0XQ@mail.gmail.com>
Message-ID: <loom.20110718T234800-548@post.gmane.org>

Michael Collins <msc at ...> writes:

> 
> 
> On Fri, Jul 1, 2011 at 9:59 AM, ran zhang <rzhang <at> gosilverplus.com> wrote:
> 
> 
>   
>     
>     
>   
>   
>     I think most people don't understand my problem, let me just
>     clarify:I want a way so only when either one of the?
>       first 2 members of the bridging conference leaves the conference,?
>       
>       the conference will be terminated even if there are 2 or more
>       members in it.
> 
> 
> 
> The problem with using the endconf flag is that *all* members with that flag
must leave before the conference is torn down. I don't believe that this feature
is presently in FS but with a feature request and possibly a modest bounty it
could possibly be added. Something like a flag named "endconfabsolute" that
would end the conference no matter what. Now your other problem was trying to do
a "bridging" conference and also set the endconf flag and you were getting
errors. I was able to reproduce the symptoms you saw, but I have not
investigated any further at this point.If anyone knows the trick to making the
members added to a "bridging" conference also have the endconf flag set (or any
flags for that matter) please let me know and we can get it documented. -MC
> 
> 
> 


The "endconf" flag can be set from the dialplan application call

IE:

    <extension name="wb_conferences_participant">
      <condition field="destination_number" expression="^(31\d{2})0$">
        <action application="answer"/>
        <action application="conference" data="$1-${domain_name}@wideband"/>
      </condition>
    </extension>

    <extension name="wb_conferences_moderator">
      <condition field="destination_number" expression="^(31\d{2})1$">
        <action application="answer"/>
        <action application="conference"
data="$1-${domain_name}@wideband+flags{moderator,endconf}"/>
      </condition>
    </extension>

PS: Bottom posting is awful.



From jaybinks at gmail.com  Tue Jul 19 03:12:34 2011
From: jaybinks at gmail.com (jay binks)
Date: Tue, 19 Jul 2011 09:12:34 +1000
Subject: [Freeswitch-users] stripping sip-rh headers at the edge of my
	network
In-Reply-To: <CAAB3jWv-i9ABnNn6fom+VSTGEvaCQhrfyGJ5dyWQ1DO9p1n=WA@mail.gmail.com>
References: <CAO47HpdG8qJeH4uoXDFB31Jf7_1oNVbmoJQ52DUS6WcmxuAj=Q@mail.gmail.com>
	<CAO47HpfcjUJj_9UVO8dFEQq8M6gMaZs5DKvUP4H_hW8tkC324A@mail.gmail.com>
	<CAAB3jWv-i9ABnNn6fom+VSTGEvaCQhrfyGJ5dyWQ1DO9p1n=WA@mail.gmail.com>
Message-ID: <CAO47Hpcupf1ZDc8Gv8WAmu+abd2F8V=RVYhjAFCeDfPRfM0wZw@mail.gmail.com>

nope.... unset dosnt work..
for sip_ph and sip_rh :(

Jay

On Tue, Jul 19, 2011 at 4:22 AM, DJB International <djbinter at gmail.com>wrote:

> I thought "unset" should work:
> http://wiki.freeswitch.org/wiki/Strip_SIP_Headers
>
>
>
> On Mon, Jul 18, 2011 at 5:29 AM, jay binks <jaybinks at gmail.com> wrote:
>
>> hmm this isnt good..  no replies..
>>
>> so, seems I cant strip these at all with freeswitch..
>>
>> using unset and even trying to overwrite the value like this
>>
>> <action application="set" data="sip_ph_X-someheader=blah"/>
>> <action application="set" data="sip_rh_X-someheader=blah"/>
>> <action application="set" data="sip_h_X-someheader=blah"/>
>>
>> fails...
>> since FS just splats on top of whatever I set , when the sip message comes
>> in from upstream..
>> (  in this case in the 180 / 183 progress )
>>
>> I need a way to strip headers from 180/183 .. preferably I could have a
>> regex to filter any starting with X :)
>> someone please point me in the right direction
>>
>> Jay
>>
>>
>>
>>
>>
>>
>> On Fri, Jul 15, 2011 at 4:54 PM, jay binks <jaybinks at gmail.com> wrote:
>>
>>> Inside my network I use sip_ph , sip_rh & sip_h headers to pass data
>>> around between boxes.
>>>
>>> however I need to strip these off at the edge ( SBC's running Freeswitch
>>> also )
>>>
>>> will unset work with these headers in the 200 OK ( and other sip messages
>>> ) ?
>>> if not, how can I achieve this ?
>>>
>>> --
>>> Sincerely
>>>
>>> Jay
>>>
>>
>>
>>
>> --
>> Sincerely
>>
>> Jay
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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
>
>


-- 
Sincerely

Jay
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From mays.david at gmail.com  Tue Jul 19 07:06:45 2011
From: mays.david at gmail.com (David Ma)
Date: Tue, 19 Jul 2011 11:06:45 +0800
Subject: [Freeswitch-users] Leg-A is automatically disconnected on Leg-B
 orginate failure
In-Reply-To: <CAKzWOxXAx3LpL0pA2cF4=0QsEKF_e=QhVp5PRua9EbnwLDCABg@mail.gmail.com>
References: <1308213151543-6482192.post@n2.nabble.com>
	<BANLkTi=xYgBTPVGxVGmk9f2=bmX5X2vF=Q@mail.gmail.com>
	<BANLkTikhPmjN5HeeJHBnQt1ivA8zbtCnYA@mail.gmail.com>
	<BANLkTi=m7oMix1ZRC9URtRuAbsG2fWyhMQ@mail.gmail.com>
	<BANLkTimxBU8m311LVKMWcFOjukwLg+ZUGQ@mail.gmail.com>
	<BANLkTinZJf3BxB7QHefnmOvxQmVW2r4dTg@mail.gmail.com>
	<BANLkTimiExLew_1cjMuQVpF3CKkR8mrGXQ@mail.gmail.com>
	<BANLkTi=ZvOZ99qVxUeJAtpjtwP5r=hfVkg@mail.gmail.com>
	<BANLkTimj4iSWFiHuBSKskz7NjF2bCEWnaw@mail.gmail.com>
	<BANLkTikyyAZaJzQx2tbiVOxHxx5mCbUdvQ@mail.gmail.com>
	<CAMLBmRLtAeU9m6HwOW3o9FA2pZFRBiXk2P82Z7msv218B4AxCA@mail.gmail.com>
	<CAMLBmR+yqetjWU+yGOypLtwOj8Ryz8rbEp41tW0yHS53w-pgWg@mail.gmail.com>
	<CAKzWOxXAx3LpL0pA2cF4=0QsEKF_e=QhVp5PRua9EbnwLDCABg@mail.gmail.com>
Message-ID: <CAMLBmRKSM3r_zzqGFofx9fqRtuMZCB0hri=p2PZ8zfy5FrRYOg@mail.gmail.com>

Hi Michael,


Thanks for the prompt response. I've pastebined the log here without SIP
messages.

http://pastebin.freeswitch.org/16846

The log is all that for a complete call. *NO dialplan was applied*. It is a
callback logic and both legs are outbound. The channel variable *
uuid_bridge_continue_on_cancel*=*true *is set. But leg-A still hangs up
automatically on leg-B dialing failure (DESTINATION_OUT_OF_ORDER). NOTE the
2 legs are bridged on reception of *progress-media* from leg-B.

Please let me know if you need additional input.

Thanks,
D.Ma

On Tue, Jul 19, 2011 at 3:34 AM, Michael Collins <msc at freeswitch.org> wrote:

> FYI,
>
> Tony is on vacation for a bit, so we'll have to do our best without him.
> Could you please do two things? First, turn off the sofia debugging. The SIP
> trace is fine, but all that other stuff is just noise. Second, can you show
> the call log from the very beginning, including any dialplan hunting.
>
> Thanks,
> MC
>
>
> On Mon, Jul 18, 2011 at 2:25 AM, David Ma <mays.david at gmail.com> wrote:
>
>> Hello Anthony,
>>
>> The same log in better formatting is here:
>>
>> http://pastebin.freeswitch.org/16840
>>
>> Regards,
>> D.Ma
>>
>>
>> On Mon, Jul 18, 2011 at 5:18 PM, David Ma <mays.david at gmail.com> wrote:
>>
>>> Hi Anthony,
>>>
>>> I've tried channel variable uuid_bridge_continue_on_cancel=true. However,
>>> this doesn't solve the problem. The log is in the following on pastebin,
>>> please kindly help investigate why.
>>>
>>> http://pastebin.freeswitch.org/16839
>>>
>>> Thanks a lot!
>>>
>>> Best regards,
>>> D.Ma
>>>
>>>
>>> On Wed, Jun 29, 2011 at 10:53 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> The only side effect is that it will give you the behavior you want in
>>>> this particular situation.
>>>>
>>>>
>>>> On Wed, Jun 29, 2011 at 4:35 AM, David Ma <mays.david at gmail.com> wrote:
>>>> > Hi Anthony,
>>>> >
>>>> > Thanks very much for the information. I appreciate your advice. It is
>>>> great
>>>> > to learn about such a parameter.
>>>> >
>>>> > FS wiki has only a little description about this parameter. Is there
>>>> any
>>>> > side effect, or caution for using such a parameter?
>>>> >
>>>> > Thanks,
>>>> > D.Ma
>>>> >
>>>> > On Wed, Jun 29, 2011 at 12:08 AM, Anthony Minessale
>>>> > <anthony.minessale at gmail.com> wrote:
>>>> >>
>>>> >> The cases where you have this problem arise because you are
>>>> >> originating to the B leg who answers 183 (early media) then hangs up
>>>> >> before the call was answered.
>>>> >>
>>>> >> set the channel variable uuid_bridge_continue_on_cancel=true on the A
>>>> >> leg to change the behavior to what you want.
>>>> >>
>>>> >>
>>>> >> On Mon, Jun 27, 2011 at 10:09 AM, Michael Collins <
>>>> msc at freeswitch.org>
>>>> >> wrote:
>>>> >> > This makes my eyes bleed. Can you please put this on
>>>> >> > pastebin.freeswitch.org? Use "FreeSWITCH Log" as the syntax
>>>> highlight.
>>>> >> > -MC
>>>> >> >
>>>> >> > On Sun, Jun 26, 2011 at 11:27 PM, David Ma <mays.david at gmail.com>
>>>> wrote:
>>>> >> >>
>>>> >> >> Hi Michael,
>>>> >> >>
>>>> >>
>>>> >> --
>>>> >> 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
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > 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
>>>> >
>>>> >
>>>>
>>>>
>>>>
>>>> --
>>>> 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
>>>>
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 va_mclean at yahoo.com  Tue Jul 19 07:22:54 2011
From: va_mclean at yahoo.com (Mclean Va)
Date: Mon, 18 Jul 2011 20:22:54 -0700 (PDT)
Subject: [Freeswitch-users] Google Voice for Windows version of FS
In-Reply-To: <1310998336606-6594878.post@n2.nabble.com>
Message-ID: <1311045774.68169.YahooMailClassic@web121608.mail.ne1.yahoo.com>

Jeff,
I'll after we have it completely working. ?Currently we are having couple of people testing it and are having Gtalk out hangup issues (called phone still rang when calling phone hang up).
May be some Gtalk-in issues.
Asterisk worked fine for Gtalk-out hangup.
Y.
?
--- On Mon, 7/18/11, Jeff Lenk <jeff at jefflenk.com> wrote:

From: Jeff Lenk <jeff at jefflenk.com>
Subject: Re: [Freeswitch-users] Google Voice for Windows version of FS
To: freeswitch-users at lists.freeswitch.org
Date: Monday, July 18, 2011, 7:12 AM

Yahoo2003,

I was hoping you might take a few minutes if there were any notes you had
when you built with vs2010 and make a few improvments to the Wiki for
GnuTls?

Thanks
Jeff

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Google-Voice-for-Windows-version-of-FS-tp6574607p6594878.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
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 jaybinks at gmail.com  Tue Jul 19 09:15:10 2011
From: jaybinks at gmail.com (jay binks)
Date: Tue, 19 Jul 2011 15:15:10 +1000
Subject: [Freeswitch-users] stripping sip-rh headers at the edge of my
	network
In-Reply-To: <CAO47Hpcupf1ZDc8Gv8WAmu+abd2F8V=RVYhjAFCeDfPRfM0wZw@mail.gmail.com>
References: <CAO47HpdG8qJeH4uoXDFB31Jf7_1oNVbmoJQ52DUS6WcmxuAj=Q@mail.gmail.com>
	<CAO47HpfcjUJj_9UVO8dFEQq8M6gMaZs5DKvUP4H_hW8tkC324A@mail.gmail.com>
	<CAAB3jWv-i9ABnNn6fom+VSTGEvaCQhrfyGJ5dyWQ1DO9p1n=WA@mail.gmail.com>
	<CAO47Hpcupf1ZDc8Gv8WAmu+abd2F8V=RVYhjAFCeDfPRfM0wZw@mail.gmail.com>
Message-ID: <CAO47HpdensYVuVk5q_Cz0dts5wsfxxYj74ARv7BVfQAMi4k12w@mail.gmail.com>

ok so I just wrote a patch for this ..
http://jira.freeswitch.org/browse/FS-3439

hope someone picks it up and applies it asap :)

Jay

On Tue, Jul 19, 2011 at 9:12 AM, jay binks <jaybinks at gmail.com> wrote:

> nope.... unset dosnt work..
> for sip_ph and sip_rh :(
>
> Jay
>
>
> On Tue, Jul 19, 2011 at 4:22 AM, DJB International <djbinter at gmail.com>wrote:
>
>> I thought "unset" should work:
>> http://wiki.freeswitch.org/wiki/Strip_SIP_Headers
>>
>>
>>
>> On Mon, Jul 18, 2011 at 5:29 AM, jay binks <jaybinks at gmail.com> wrote:
>>
>>> hmm this isnt good..  no replies..
>>>
>>> so, seems I cant strip these at all with freeswitch..
>>>
>>> using unset and even trying to overwrite the value like this
>>>
>>> <action application="set" data="sip_ph_X-someheader=blah"/>
>>> <action application="set" data="sip_rh_X-someheader=blah"/>
>>> <action application="set" data="sip_h_X-someheader=blah"/>
>>>
>>> fails...
>>> since FS just splats on top of whatever I set , when the sip message
>>> comes in from upstream..
>>> (  in this case in the 180 / 183 progress )
>>>
>>> I need a way to strip headers from 180/183 .. preferably I could have a
>>> regex to filter any starting with X :)
>>> someone please point me in the right direction
>>>
>>> Jay
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Fri, Jul 15, 2011 at 4:54 PM, jay binks <jaybinks at gmail.com> wrote:
>>>
>>>> Inside my network I use sip_ph , sip_rh & sip_h headers to pass data
>>>> around between boxes.
>>>>
>>>> however I need to strip these off at the edge ( SBC's running Freeswitch
>>>> also )
>>>>
>>>> will unset work with these headers in the 200 OK ( and other sip
>>>> messages ) ?
>>>> if not, how can I achieve this ?
>>>>
>>>> --
>>>> Sincerely
>>>>
>>>> Jay
>>>>
>>>
>>>
>>>
>>> --
>>> Sincerely
>>>
>>> Jay
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
>
> --
> Sincerely
>
> Jay
>



-- 
Sincerely

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

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

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

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

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

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

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

Sincerely

Jay Binks
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From steveayre at gmail.com  Tue Jul 19 12:00:27 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 19 Jul 2011 09:00:27 +0100
Subject: [Freeswitch-users] RTP Error
In-Reply-To: <684D78F9-46E9-402A-AB0B-BC0DD0A735E7@5ninesolutions.com>
References: <684D78F9-46E9-402A-AB0B-BC0DD0A735E7@5ninesolutions.com>
Message-ID: <CAFiqYum0ERGxU1sDD36EaQJ8teSGyhLC_g7T05JKhmE=f9q1gQ@mail.gmail.com>

Start off by seeing if you can reproduce it on the latest Git.

If you can, some more context would be useful. Your line numbers are
out-of-date and there are several places which generate that message.
If you set logging to debug level then you should see a message
shortly before the error indicating what it might be doing.

-Steve


On 18 July 2011 18:03, Spencer Thomason <spencer at 5ninesolutions.com> wrote:
> Hello all,
> I have been getting the following error in my logs, very intermittently. ?Any idea what this means and where I could look to resolve it? ?This instance was built from git on July 8.
>
> 2011-07-18 11:50:39.855581 [ERR] sofia.c:5748 RTP Error!
> 2011-07-18 11:50:44.459669 [ERR] sofia.c:5748 RTP Error!
>
>
> Thanks,
> Spencer
>
>
> _______________________________________________
> 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 daniel at danielknoll.de  Tue Jul 19 13:17:39 2011
From: daniel at danielknoll.de (Daniel Knoll)
Date: Tue, 19 Jul 2011 11:17:39 +0200
Subject: [Freeswitch-users] X-PRE-PROCESS and Dialplan XML Module
Message-ID: <CACREEeYXb+Y3FddJ+aFDE-xRqNDPCfKBH-hC7BM8hrk2A8qQ+w@mail.gmail.com>

Hi FreeSwitch Group,
i'm a little bit confused about the X-PRE-PROCESS  and XML Dialplan Module.

I'm using a dynamic dialplan to include in FreeSwitch with
<X-PRE-PROCESS cmd="exec" data="wget -qO -
http://www.domain.com/dialplan.php" />

This don't work if i place X-PRE-PROCESS in at the top from public.xml
outsite the <include> tag

<include>
   <context name="public">
     <extension name="123456789">
        <condition field="destination_number" expression="123456789">
            <action application="set" data="hangup_after_bridge=true"/>
            <action application="javascript" data="app.js"/>
       </condition>
    </extension>
  </context>
</include>


but

this works if X-PRE-PROCESS is in the <include> tag, but it's not
clean XML, because the root is missing

<extension name="123456789">
   <condition field="destination_number" expression="123456789">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="javascript" data="app.js"/>
   </condition>
</extension>


My Question is what is the right anatomy of a XML to include my
dialplan in the public.xml with X-PRE-PROCESS and exec wget ?


Thanks for your answers.
Daniel


From avi at avimarcus.net  Tue Jul 19 13:24:13 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 19 Jul 2011 12:24:13 +0300
Subject: [Freeswitch-users] Urgent Feature Request - RTCP report blocks
In-Reply-To: <CAO47HpcdY-PSitfd_2njqJNa8OH6aOMP06_oqAvveqZugyFf-A@mail.gmail.com>
References: <CAO47HpcdY-PSitfd_2njqJNa8OH6aOMP06_oqAvveqZugyFf-A@mail.gmail.com>
Message-ID: <CALNZuwkJsna26wNLb6E1K__JP-SDpZPh_81t=6V67mibcfczSg@mail.gmail.com>

You might want to check out voipmonitor, it logs most of that information
(as it saves a complete pcap of each call...)

http://wiki.freeswitch.org/wiki/Voipmonitor
 <http://wiki.freeswitch.org/wiki/Voipmonitor>
-Avi


On Tue, Jul 19, 2011 at 10:55 AM, jay binks <jaybinks at gmail.com> wrote:

> I NEED to get Freeswitch generating RTCP Report Blocks
> ( with all associated data - packet_loss, jitter etc )
>
> Im happy to consider a bounty on this requirement as I
> need this done as quickly as humanly possible.
>
> in preference to only filling my request,
> a complete RTCP implementation would be great however currently
> I simply need RTCP packet generation with Packet Loss Jitter and RTT
> information.
>
> we need this is to fulfill an interop test we are trying to get done, so as
> you can imagine
> time is of the essence.
>
> please contact me if you wish to work on this, to be eligible for bounty
> payment
> the code will need to be accepted into the FS Git tree and pass my testing.
> ( however I would like to test asap, even before inclusion in git )
>
> I would like your guidance, as to what bounty you think this would require.
> please contact me for bounty approval.
> this email does not constitute acceptance of any bounty / work.
>
> Sincerely
>
> Jay Binks
>
> _______________________________________________
> 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 steveayre at gmail.com  Tue Jul 19 13:36:45 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 19 Jul 2011 10:36:45 +0100
Subject: [Freeswitch-users] Urgent Feature Request - RTCP report blocks
In-Reply-To: <CALNZuwkJsna26wNLb6E1K__JP-SDpZPh_81t=6V67mibcfczSg@mail.gmail.com>
References: <CAO47HpcdY-PSitfd_2njqJNa8OH6aOMP06_oqAvveqZugyFf-A@mail.gmail.com>
	<CALNZuwkJsna26wNLb6E1K__JP-SDpZPh_81t=6V67mibcfczSg@mail.gmail.com>
Message-ID: <CAFiqYumvLki4hkKyS3YZwoC8=1PhUQmd=VaqM_zuA5JusDXcpQ@mail.gmail.com>

AviMarcus,

He says he needs it to interop with some device, not to monitor the
results himself...

-Steve



On 19 July 2011 10:24, Avi Marcus <avi at avimarcus.net> wrote:
> You might want to check out voipmonitor, it logs most of that information
> (as it saves a complete pcap of each call...)
> http://wiki.freeswitch.org/wiki/Voipmonitor
> -Avi
>
> On Tue, Jul 19, 2011 at 10:55 AM, jay binks <jaybinks at gmail.com> wrote:
>>
>> I NEED to get Freeswitch generating RTCP Report Blocks
>> ( with all associated data - packet_loss, jitter etc )
>>
>> Im happy to consider a bounty on this requirement as I
>> need this done as quickly as humanly possible.
>>
>> in preference to only filling my request,
>> a complete RTCP implementation would be great however currently
>> I simply need RTCP packet generation with Packet Loss Jitter and RTT
>> information.
>>
>> we need this is to fulfill an interop test we are trying to get done, so
>> as you can imagine
>> time is of the essence.
>>
>> please contact me if you wish to work on this, to be eligible for bounty
>> payment
>> the code will need to be accepted into the FS Git tree and pass my
>> testing.
>> ( however I would like to test asap, even before inclusion in git )
>>
>> I would like your guidance, as to what bounty you think this would
>> require.
>> please contact me for bounty approval.
>> this email does not constitute acceptance of any bounty / work.
>>
>> Sincerely
>>
>> Jay Binks
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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 gcd at i.ph  Tue Jul 19 14:19:32 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 19 Jul 2011 18:19:32 +0800
Subject: [Freeswitch-users] X-PRE-PROCESS and Dialplan XML Module
In-Reply-To: <CACREEeYXb+Y3FddJ+aFDE-xRqNDPCfKBH-hC7BM8hrk2A8qQ+w@mail.gmail.com>
References: <CACREEeYXb+Y3FddJ+aFDE-xRqNDPCfKBH-hC7BM8hrk2A8qQ+w@mail.gmail.com>
Message-ID: <CAHKqMb7obu6Ob6j=ei8p=wQhvvrNdZ+TD-ZHxtBrPgBz9wrfKw@mail.gmail.com>

hi daniel,

the X-PRE-PROCESS only purpose is to read the *.xml files in the said
directory at that point before proceeding further down. cool way of
organizing configuration files. nothing else.

if you want dynamic dialplan, use XML/Curl.

On Tue, Jul 19, 2011 at 5:17 PM, Daniel Knoll <daniel at danielknoll.de> wrote:

> Hi FreeSwitch Group,
> i'm a little bit confused about the X-PRE-PROCESS  and XML Dialplan Module.
>
> I'm using a dynamic dialplan to include in FreeSwitch with
> <X-PRE-PROCESS cmd="exec" data="wget -qO -
> http://www.domain.com/dialplan.php" />
>
> This don't work if i place X-PRE-PROCESS in at the top from public.xml
> outsite the <include> tag
>
> <include>
>   <context name="public">
>     <extension name="123456789">
>        <condition field="destination_number" expression="123456789">
>            <action application="set" data="hangup_after_bridge=true"/>
>            <action application="javascript" data="app.js"/>
>       </condition>
>    </extension>
>  </context>
> </include>
>
>
> but
>
> this works if X-PRE-PROCESS is in the <include> tag, but it's not
> clean XML, because the root is missing
>
> <extension name="123456789">
>   <condition field="destination_number" expression="123456789">
>        <action application="set" data="hangup_after_bridge=true"/>
>        <action application="javascript" data="app.js"/>
>   </condition>
> </extension>
>
>
> My Question is what is the right anatomy of a XML to include my
> dialplan in the public.xml with X-PRE-PROCESS and exec wget ?
>
>
> Thanks for your answers.
> Daniel
>
> _______________________________________________
> 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 jaybinks at gmail.com  Tue Jul 19 14:43:14 2011
From: jaybinks at gmail.com (jay binks)
Date: Tue, 19 Jul 2011 20:43:14 +1000
Subject: [Freeswitch-users] Urgent Feature Request - RTCP report blocks
In-Reply-To: <CALNZuwkJsna26wNLb6E1K__JP-SDpZPh_81t=6V67mibcfczSg@mail.gmail.com>
References: <CAO47HpcdY-PSitfd_2njqJNa8OH6aOMP06_oqAvveqZugyFf-A@mail.gmail.com>
	<CALNZuwkJsna26wNLb6E1K__JP-SDpZPh_81t=6V67mibcfczSg@mail.gmail.com>
Message-ID: <CAO47Hpfe5gEVtVA8LfAjaAEsm=1rvr72fZXCOvJj3nY6Q-0UrQ@mail.gmail.com>

Yup , ive looked at voip monitor and its quite good.
infact Ive provided some patches too it.

however it dosnt help with an interop requirement specifying that I must
provide RTCP
complete with jitter and RTT information.. :(

Jay



On Tue, Jul 19, 2011 at 7:24 PM, Avi Marcus <avi at avimarcus.net> wrote:

> You might want to check out voipmonitor, it logs most of that information
> (as it saves a complete pcap of each call...)
>
> http://wiki.freeswitch.org/wiki/Voipmonitor
>  <http://wiki.freeswitch.org/wiki/Voipmonitor>
> -Avi
>
>
> On Tue, Jul 19, 2011 at 10:55 AM, jay binks <jaybinks at gmail.com> wrote:
>
>> I NEED to get Freeswitch generating RTCP Report Blocks
>> ( with all associated data - packet_loss, jitter etc )
>>
>> Im happy to consider a bounty on this requirement as I
>> need this done as quickly as humanly possible.
>>
>> in preference to only filling my request,
>> a complete RTCP implementation would be great however currently
>> I simply need RTCP packet generation with Packet Loss Jitter and RTT
>> information.
>>
>> we need this is to fulfill an interop test we are trying to get done, so
>> as you can imagine
>> time is of the essence.
>>
>> please contact me if you wish to work on this, to be eligible for bounty
>> payment
>> the code will need to be accepted into the FS Git tree and pass my
>> testing.
>> ( however I would like to test asap, even before inclusion in git )
>>
>> I would like your guidance, as to what bounty you think this would
>> require.
>> please contact me for bounty approval.
>> this email does not constitute acceptance of any bounty / work.
>>
>> Sincerely
>>
>> Jay Binks
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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
>
>


-- 
Sincerely

Jay
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From boris at tagnet.ru  Tue Jul 19 14:45:49 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 19 Jul 2011 16:45:49 +0600
Subject: [Freeswitch-users] Can't get fax working
Message-ID: <4E25605D.7050809@tagnet.ru>

Hello!

     I need help with fax configuration. My network is:

PSTN E1 --- Cisco AS5350 -- Freeswitch -- Audiocodes MP-114 -- Fax

FS version: FreeSWITCH Version 1.0.head (git-062f07f 2011-07-16 12-35-25 
+0200)
Audiocodes firmware: 6.00A.038.004
I can't get faxes working :( In FS profiles there is a parameter <param 
name="t38-passthru" value="true"/>
Cisco DP fax settings:
fax-relay ecm disable
  fax rate 9600
  fax protocol t38 ls-redundancy 2 hs-redundancy 1 fallback pass-through 
g711alaw
  no vad

On Audiocodes there is also T38 enabled (as I think) but there is no 
T38. There are messages on MP-114:
ErrMgs=16 T38Decoder received Non-T38 Packet
and
||

Modem Relay Is Not Supported! Forcing Bypass Mode


Please, help! What information I need to collect to help solve the problem?

-- 
Regards,
Boris


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From avi at avimarcus.net  Tue Jul 19 15:02:38 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 19 Jul 2011 14:02:38 +0300
Subject: [Freeswitch-users] X-PRE-PROCESS and Dialplan XML Module
In-Reply-To: <CAHKqMb7obu6Ob6j=ei8p=wQhvvrNdZ+TD-ZHxtBrPgBz9wrfKw@mail.gmail.com>
References: <CACREEeYXb+Y3FddJ+aFDE-xRqNDPCfKBH-hC7BM8hrk2A8qQ+w@mail.gmail.com>
	<CAHKqMb7obu6Ob6j=ei8p=wQhvvrNdZ+TD-ZHxtBrPgBz9wrfKw@mail.gmail.com>
Message-ID: <CALNZuw=uQaHC028EU9+BFfn2gBx5ucTvtcVRwqWaD4FUSPsTDg@mail.gmail.com>

Yes, afaik, the x-pre-process is done BEFORE the XML is parsed (which is why
you can't comment it out) and just does a wget and replaces itself with the
file it grabbed.

-Avi


On Tue, Jul 19, 2011 at 1:19 PM, Nandy Dagondon <gcd at i.ph> wrote:

> hi daniel,
>
> the X-PRE-PROCESS only purpose is to read the *.xml files in the said
> directory at that point before proceeding further down. cool way of
> organizing configuration files. nothing else.
>
> if you want dynamic dialplan, use XML/Curl.
>
> On Tue, Jul 19, 2011 at 5:17 PM, Daniel Knoll <daniel at danielknoll.de>wrote:
>
>> Hi FreeSwitch Group,
>> i'm a little bit confused about the X-PRE-PROCESS  and XML Dialplan
>> Module.
>>
>> I'm using a dynamic dialplan to include in FreeSwitch with
>> <X-PRE-PROCESS cmd="exec" data="wget -qO -
>> http://www.domain.com/dialplan.php" />
>>
>> This don't work if i place X-PRE-PROCESS in at the top from public.xml
>> outsite the <include> tag
>>
>> <include>
>>   <context name="public">
>>     <extension name="123456789">
>>        <condition field="destination_number" expression="123456789">
>>            <action application="set" data="hangup_after_bridge=true"/>
>>            <action application="javascript" data="app.js"/>
>>       </condition>
>>    </extension>
>>  </context>
>> </include>
>>
>>
>> but
>>
>> this works if X-PRE-PROCESS is in the <include> tag, but it's not
>> clean XML, because the root is missing
>>
>> <extension name="123456789">
>>   <condition field="destination_number" expression="123456789">
>>        <action application="set" data="hangup_after_bridge=true"/>
>>        <action application="javascript" data="app.js"/>
>>   </condition>
>> </extension>
>>
>>
>> My Question is what is the right anatomy of a XML to include my
>> dialplan in the public.xml with X-PRE-PROCESS and exec wget ?
>>
>>
>> Thanks for your answers.
>> Daniel
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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 steveayre at gmail.com  Tue Jul 19 15:15:30 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 19 Jul 2011 12:15:30 +0100
Subject: [Freeswitch-users] X-PRE-PROCESS and Dialplan XML Module
In-Reply-To: <CALNZuw=uQaHC028EU9+BFfn2gBx5ucTvtcVRwqWaD4FUSPsTDg@mail.gmail.com>
References: <CACREEeYXb+Y3FddJ+aFDE-xRqNDPCfKBH-hC7BM8hrk2A8qQ+w@mail.gmail.com>
	<CAHKqMb7obu6Ob6j=ei8p=wQhvvrNdZ+TD-ZHxtBrPgBz9wrfKw@mail.gmail.com>
	<CALNZuw=uQaHC028EU9+BFfn2gBx5ucTvtcVRwqWaD4FUSPsTDg@mail.gmail.com>
Message-ID: <CAFiqYunQYhoGqxDjZQn4KXMptEtQiRq06d9v48-wCEh1xUf=OA@mail.gmail.com>

Correct


On 19 July 2011 12:02, Avi Marcus <avi at avimarcus.net> wrote:
> Yes, afaik, the x-pre-process is done BEFORE the XML is parsed (which is why
> you can't comment it out) and just does a wget and replaces itself with the
> file it grabbed.
> -Avi
>
> On Tue, Jul 19, 2011 at 1:19 PM, Nandy Dagondon <gcd at i.ph> wrote:
>>
>> hi daniel,
>> the X-PRE-PROCESS only purpose is to read the *.xml files in the said
>> directory at that point before proceeding further down. cool way of
>> organizing configuration files. nothing else.
>> if you want dynamic dialplan, use XML/Curl.
>> On Tue, Jul 19, 2011 at 5:17 PM, Daniel Knoll <daniel at danielknoll.de>
>> wrote:
>>>
>>> Hi FreeSwitch Group,
>>> i'm a little bit confused about the X-PRE-PROCESS ?and XML Dialplan
>>> Module.
>>>
>>> I'm using a dynamic dialplan to include in FreeSwitch with
>>> <X-PRE-PROCESS cmd="exec" data="wget -qO -
>>> http://www.domain.com/dialplan.php" />
>>>
>>> This don't work if i place X-PRE-PROCESS in at the top from public.xml
>>> outsite the <include> tag
>>>
>>> <include>
>>> ? <context name="public">
>>> ? ? <extension name="123456789">
>>> ? ? ? ?<condition field="destination_number" expression="123456789">
>>> ? ? ? ? ? ?<action application="set" data="hangup_after_bridge=true"/>
>>> ? ? ? ? ? ?<action application="javascript" data="app.js"/>
>>> ? ? ? </condition>
>>> ? ?</extension>
>>> ?</context>
>>> </include>
>>>
>>>
>>> but
>>>
>>> this works if X-PRE-PROCESS is in the <include> tag, but it's not
>>> clean XML, because the root is missing
>>>
>>> <extension name="123456789">
>>> ? <condition field="destination_number" expression="123456789">
>>> ? ? ? ?<action application="set" data="hangup_after_bridge=true"/>
>>> ? ? ? ?<action application="javascript" data="app.js"/>
>>> ? </condition>
>>> </extension>
>>>
>>>
>>> My Question is what is the right anatomy of a XML to include my
>>> dialplan in the public.xml with X-PRE-PROCESS and exec wget ?
>>>
>>>
>>> Thanks for your answers.
>>> Daniel
>>>
>>> _______________________________________________
>>> 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
>>
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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 dimskraft at gmail.com  Tue Jul 19 15:53:36 2011
From: dimskraft at gmail.com (Dmitry Kravchenko)
Date: Tue, 19 Jul 2011 15:53:36 +0400
Subject: [Freeswitch-users] How to authencticate with mod_rtmp?
Message-ID: <CADDV4gzH8gW1cP_K3vMgOtouACRs4PNQSRKC9aVd-m0uwxTjQw@mail.gmail.com>

Hi!

It is said in book that user registry is single and centralized for all
components (p. 64).

But while trying to login via FLEX telephone to default number of 1000 with
password of 1234, I get "Authentication failed" message.

In freeswitch CLI I see message

2011-07-19 15:46:51.473828 [WARNING] rtmp.c:227 Authentication failed for
1000 at 192.168.10.196

where 192.168.10.196 is true IP of my freeswitch machine.

Connection with X-Lite goes OK with message

2011-07-19 15:50:19.473829 [WARNING] sofia_reg.c:1337 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [1000 at 192.168.10.196] from ip
192.168.10.56

If I put wrong password into X-Lite then I get

2011-07-19 15:50:19.493774 [WARNING] sofia_reg.c:1295 SIP auth failure
(REGISTER) on sofia profile 'internal' for [1000 at 192.168.10.196] from ip
192.168.10.56

So my conclusion is that SIP phone is treates as internal profile by
default. Is it possible to set RTMP profiles as internal too?

Or may be I should better create exteral accounts to authenticate with
RTMP?

Thanks.
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From daniel at danielknoll.de  Tue Jul 19 16:16:50 2011
From: daniel at danielknoll.de (Daniel Knoll)
Date: Tue, 19 Jul 2011 14:16:50 +0200
Subject: [Freeswitch-users] X-PRE-PROCESS and Dialplan XML Module
In-Reply-To: <CAHKqMb7obu6Ob6j=ei8p=wQhvvrNdZ+TD-ZHxtBrPgBz9wrfKw@mail.gmail.com>
References: <CACREEeYXb+Y3FddJ+aFDE-xRqNDPCfKBH-hC7BM8hrk2A8qQ+w@mail.gmail.com>
	<CAHKqMb7obu6Ob6j=ei8p=wQhvvrNdZ+TD-ZHxtBrPgBz9wrfKw@mail.gmail.com>
Message-ID: <CACREEeaWtjiKGHbsw4d9WOT3UkXUJzxhmVYA7B5_=JyzPE5Fxw@mail.gmail.com>

Hi All,

i used XML/Curl module, but the problem at this module is, that he ask
the webserver for every call. It is better to have static diaplans (in
my case these have the most call traffic) and after parsing static
plan he ask the webserver.

My X-PRE-PROCESS Option is described in XML/Curl Wiki under Sektion
"Storing Static Dialplans."

I found a little solution, but I think it is very dirty

I use this without having "header('Content-Type: text/xml'); " in my php

<extension name="123456789">
   <condition field="destination_number" expression="123456789">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="javascript" data="app.js"/>
   </condition>
</extension>

If you had a solution, that -not- every call made a http request (only
if don't find extension in static dialplan), I'm happy about that.

Thanks for help
Daniel


2011/7/19, Nandy Dagondon <gcd at i.ph>:
> hi daniel,
>
> the X-PRE-PROCESS only purpose is to read the *.xml files in the said
> directory at that point before proceeding further down. cool way of
> organizing configuration files. nothing else.
>
> if you want dynamic dialplan, use XML/Curl.
>
> On Tue, Jul 19, 2011 at 5:17 PM, Daniel Knoll <daniel at danielknoll.de> wrote:
>
>> Hi FreeSwitch Group,
>> i'm a little bit confused about the X-PRE-PROCESS  and XML Dialplan
>> Module.
>>
>> I'm using a dynamic dialplan to include in FreeSwitch with
>> <X-PRE-PROCESS cmd="exec" data="wget -qO -
>> http://www.domain.com/dialplan.php" />
>>
>> This don't work if i place X-PRE-PROCESS in at the top from public.xml
>> outsite the <include> tag
>>
>> <include>
>>   <context name="public">
>>     <extension name="123456789">
>>        <condition field="destination_number" expression="123456789">
>>            <action application="set" data="hangup_after_bridge=true"/>
>>            <action application="javascript" data="app.js"/>
>>       </condition>
>>    </extension>
>>  </context>
>> </include>
>>
>>
>> but
>>
>> this works if X-PRE-PROCESS is in the <include> tag, but it's not
>> clean XML, because the root is missing
>>
>> <extension name="123456789">
>>   <condition field="destination_number" expression="123456789">
>>        <action application="set" data="hangup_after_bridge=true"/>
>>        <action application="javascript" data="app.js"/>
>>   </condition>
>> </extension>
>>
>>
>> My Question is what is the right anatomy of a XML to include my
>> dialplan in the public.xml with X-PRE-PROCESS and exec wget ?
>>
>>
>> Thanks for your answers.
>> Daniel
>>
>> _______________________________________________
>> 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
>>
>


-- 
Daniel Knoll
Liberdastr.. 9
12047 Berlin

fon +49 (0)179 20 16 50 8
mail daniel at danielknoll.de
web www.danielknoll.de


From sascha.daniels at amooma.de  Tue Jul 19 16:43:56 2011
From: sascha.daniels at amooma.de (Sascha Daniels)
Date: Tue, 19 Jul 2011 14:43:56 +0200
Subject: [Freeswitch-users] vm-alternate-greet-id is ignored
Message-ID: <4E257C0C.7040103@amooma.de>

Hi together,

I need to set a different Number in voicemail greetings.

Just for testing purpose I tried to set it without a variable.

xml.condition( :field => 'destination_number', :expression =>
'^-vbox-(.+)$' ) {
                    xml.action( :application => 'answer', :data =>
'voicemail_authorized=true' )
                    xml.param( :name => 'vm-alternate-greet-id', :value
=> '4444'  )
                    xml.action( :application => 'voicemail', :data =>
'default ${domain_name} $1' )
                }

Unfortunately the real sip account is played.

Did I get the documentation wrong?

I am using "FreeSWITCH version: 1.0.head (git-2e651c8 2011-07-03
22-35-44 -0500)"

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



From fabio.bigliardi at gmail.com  Tue Jul 19 16:47:09 2011
From: fabio.bigliardi at gmail.com (Fabio Bigliardi)
Date: Tue, 19 Jul 2011 14:47:09 +0200
Subject: [Freeswitch-users] API conference invite group
Message-ID: <CABtP9QiLqZXUwGv9udZ7gpyHo=XLHT_DcpGMDnvmYPU7LUjf2Q@mail.gmail.com>

Hi all,
I would like to be able to invite a  group to a conference through an API
command.

In xml dialplan this can be achieved through the following action:

<action application="conference_set_auto_outcall"
data="${group_call(my_group}"/>

In the API, I suppose the right command is "dial":

api conference my_conf dial <group dial string>

But how can I specify group dial string?

Thank you for your support.

Best regards,

Fabio Bigliardi
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From boris at tagnet.ru  Tue Jul 19 18:31:09 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 19 Jul 2011 20:31:09 +0600
Subject: [Freeswitch-users] Can't get fax working
In-Reply-To: <4E25605D.7050809@tagnet.ru>
References: <4E25605D.7050809@tagnet.ru>
Message-ID: <4E25952D.6080106@tagnet.ru>

Hello!

     After digging I found that problem is with Audiocodes.
Kapanga SoftPhone --- FS -- Cisco 5350 -- PSTN -- fax is working
Kapanga SoftPhone -- FS -- VOIP clould --- fax is working
Kapanga SoftPhone -- FS -- AudioCodes MP-11x -- fax is not working....

So may be there is somebody who may help with AudioCodes configuration?
> Hello!
>
>     I need help with fax configuration. My network is:
>
> PSTN E1 --- Cisco AS5350 -- Freeswitch -- Audiocodes MP-114 -- Fax
>
> FS version: FreeSWITCH Version 1.0.head (git-062f07f 2011-07-16 
> 12-35-25 +0200)
> Audiocodes firmware: 6.00A.038.004
> I can't get faxes working :( In FS profiles there is a parameter 
> <param name="t38-passthru" value="true"/>
> Cisco DP fax settings:
> fax-relay ecm disable
>  fax rate 9600
>  fax protocol t38 ls-redundancy 2 hs-redundancy 1 fallback 
> pass-through g711alaw
>  no vad
>
> On Audiocodes there is also T38 enabled (as I think) but there is no 
> T38. There are messages on MP-114:
> ErrMgs=16 T38Decoder received Non-T38 Packet
> and
> ||
> Modem Relay Is Not Supported! Forcing Bypass Mode
>
>
> Please, help! What information I need to collect to help solve the problem?
> -- 
> Regards,
> Boris
>
>
>
> _______________________________________________
> 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


-- 
Regards,
Boris


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From Nabble at slickdeals.endjunk.com  Tue Jul 19 18:52:44 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 19 Jul 2011 07:52:44 -0700 (PDT)
Subject: [Freeswitch-users] incoming call to FS
In-Reply-To: <CAKnRR0ZkKt=QjB5a7eKdtFvYW+pq2SR7u-AFXr1TNPfGNfdObg@mail.gmail.com>
References: <CAKnRR0ZkKt=QjB5a7eKdtFvYW+pq2SR7u-AFXr1TNPfGNfdObg@mail.gmail.com>
Message-ID: <1311087164892-6599055.post@n2.nabble.com>


budi wibowo wrote:
> 
> hi
> i have sip server and want to connect the sip server to FS then call will
> be
> routed using mod_dingaling, mod_dingaling already active
> sipserver--->FS---mod_dingaling
> 
> from FS it will be incoming sip call, i already define sipserver arameter
> in
> external sip_profile, then make call but i cant see any call coming from
> fs_cli
> anything missing in my config?
Just a thought if you can confirm this issue with a direct call to your SIP
line that is handled by your FS system? Also, while at it, try to call
directly to your GV line handled by mod_dingaling on your FS system. In
either case, observe if there is some incoming traffics.

-----
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/incoming-call-to-FS-tp6591832p6599055.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fabio.bigliardi at gmail.com  Tue Jul 19 18:55:52 2011
From: fabio.bigliardi at gmail.com (Fabio Bigliardi)
Date: Tue, 19 Jul 2011 16:55:52 +0200
Subject: [Freeswitch-users] Problem with API command expand
Message-ID: <CABtP9QhjuS+2yfw1p9xyqZSFbn8d9yGCw10N9c9PK5MAD0awSA@mail.gmail.com>

Hi all,

I would like to add users to a group through an API command.
I've tried the following syntax:

api expand group insert:01@${domain_name}:${sofia_contact(1001@
${domain_name})}

But the output of the sqlite command:

select * from group_data;

is the following:

1LGL0Q1|01 at 192.168.25.61|sofia/internal/sip

instead of the complete and correct one:

1LGL0Q1|01 at 192.168.25.61|sofia/internal/sip:1001 at 192.168.25.61:5070

What am I doing wrong?

Many thanks.

FB
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From Suneel.Papineni at mettoni.com  Tue Jul 19 19:35:37 2011
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Tue, 19 Jul 2011 16:35:37 +0100
Subject: [Freeswitch-users] Event_socket api is not working on Ring_Ready
	and Sleep configured
Message-ID: <3181A30B8C35AB4AA8577B78DDF461380838B292@nickel.mettonigroup.com>

Hi,

 

I am trying to respond an incoming call to Freeswitch with 180 Ringing
for 3 minutes. During this time I am sending a redirect event through
event sockets to Divert the call to another number (in fact my intention
is to send 302 here). After sending the API command, my application has
received "Reply-Text: +OK", but Freeswitch didn't send 302. I checked in
Freeswitch logs and Wireshark traces as well. Call is just in ringing
state for 3 minutes and then 480 was sent by FS.

 

My configuration is as follows:

<extension name="TrunkLine">

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

      <action application="ring_ready"/>

        <action application="sleep" data="180000"/>

   </condition>

  </extension>

 

Could someone please explain me why events are taken effective and help
me with any suggestion.

 

Hope this is somewhat a similar question which Mr. Steve Richardson
asked with subject [Freeswitch-users] dialplan ring_ready/sleep question
<http://lists.freeswitch.org/pipermail/freeswitch-users/2010-November/06
5635.html> . I am sorry, as I couldn't find answer, I am asking my
question.

 

Thanks & Regards

Suneel

 

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From wstephen80 at gmail.com  Tue Jul 19 19:42:33 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 19 Jul 2011 17:42:33 +0200
Subject: [Freeswitch-users] CRIT error in the log: Error creating HANDLE
Message-ID: <CAO4LeU0MN3U8QQOC1kx8vhawryAL+bCC5PqaTGPwE63hFzd32w@mail.gmail.com>

After a git pull, I'm having many CRIT error in the log that I never seen
before.

All errors are similar to:

2011-07-19 17:34:40.271709 [CRIT] sofia_glue.c:2206 Error creating HANDLE!
url_str=[sip:2348163907756 at 213.166.103.1]
call_id=[N/A]
to_str=[<sip:2348163907756 at 213.166.103.1>]
from_str=["Anonymous\" <sip:971587421 at 62.196.58.2>]
invite_contact=[<sip:mod_sofia at 62.196.58.2:5060>]

Any advice on that?
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From wstephen80 at gmail.com  Tue Jul 19 19:50:11 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 19 Jul 2011 17:50:11 +0200
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAJoKKdUqV_ygenNTd+Z3hnoC2eXeiPuMOM2PvaChEm3kZXgj0A@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
	<CAFiqYukgW1UOgxNUZs8rPvWNmXx5sDVmGS_D+r_xdLMsXg_i4Q@mail.gmail.com>
	<CAJoKKdX8J68pmUogKgp1Ew4ACpkHntiHoZ2Rpx5JNhepsOVVvw@mail.gmail.com>
	<CAFiqYunZUs0d-kQVTxXEvuqVveiri6WFFW6oT3UYujPGoa2H-w@mail.gmail.com>
	<CAJoKKdXhHyQUXKCzaDc3GSu5m9A3Jdv2m9K=FaY7vt=W1OD5Qg@mail.gmail.com>
	<CAFiqYumuca1OMQ13di425JFjs2dhPeLAi2e0FPu-kG2odJSVqQ@mail.gmail.com>
	<CAKbxfG-GSbSm-wq5+YnT0jvtiiP32xJDgc9pqV1TUXj69CwDcg@mail.gmail.com>
	<CAJoKKdUqV_ygenNTd+Z3hnoC2eXeiPuMOM2PvaChEm3kZXgj0A@mail.gmail.com>
Message-ID: <CAO4LeU3SKmO4oSHMthnKzhV2w489JRMTxdd3N5hFmdDgbrTyNA@mail.gmail.com>

I'm using commit:

commit 130e1c87746b0596460358d77d6aabbfe41a0072
Author: Jeff Lenk <jeff at jefflenk.com>
Date:   Sat Jul 16 19:13:27 2011 -0500

and after 2 days and more then 3600000 sessions, the memory seems to be
stable so the problem is probably fixed.
I'll update this info in the next days.

Stephen

On Thu, Jul 14, 2011 at 5:48 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

>
>
> On Thu, Jul 14, 2011 at 5:11 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> you can use gigs before you stop growing in some cases esp on sipp test.
>> you need a lot more than 5 min to judge it.
>> more like dozen or more hours.
>>
>> yap, im running it for a very long time ... the example was just a snip
> ... and i hit the swap right now :=) .... after 2 hours of sipp at 30
> CPS....
>
> start: 2011-07-14 12:45:43
> stop: 2011-07-14 14:31:13
>
>
> if i use less load it will grow slowly but it will end into swap... its
> just a matter of time.
>
> So, how can we troubleshoot this ?
>
> im going to open a jira to move this out of Users Help mailing list...
>
> _______________________________________________
> 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 msc at freeswitch.org  Tue Jul 19 21:38:43 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Jul 2011 10:38:43 -0700
Subject: [Freeswitch-users] API conference invite group
In-Reply-To: <CABtP9QiLqZXUwGv9udZ7gpyHo=XLHT_DcpGMDnvmYPU7LUjf2Q@mail.gmail.com>
References: <CABtP9QiLqZXUwGv9udZ7gpyHo=XLHT_DcpGMDnvmYPU7LUjf2Q@mail.gmail.com>
Message-ID: <CAKzWOxXv-K1riCO_DUk9JbEudVaWP6c9m5VuskaeqqzJ+dcE8g@mail.gmail.com>

On Tue, Jul 19, 2011 at 5:47 AM, Fabio Bigliardi
<fabio.bigliardi at gmail.com>wrote:

> Hi all,
> I would like to be able to invite a  group to a conference through an API
> command.
>
> In xml dialplan this can be achieved through the following action:
>
> <action application="conference_set_auto_outcall"
> data="${group_call(my_group}"/>
>
> In the API, I suppose the right command is "dial":
>
> api conference my_conf dial <group dial string>
>
> But how can I specify group dial string?
>

Try this:

expand api conference my_conf dial group_call(my_group)

-MC


>
> Thank you for your support.
>
> Best regards,
>
> Fabio Bigliardi
>
> _______________________________________________
> 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 mi.ke at null.net  Tue Jul 19 22:07:13 2011
From: mi.ke at null.net (Mi Ke)
Date: Tue, 19 Jul 2011 18:07:13 +0000
Subject: [Freeswitch-users] playing audio to the *bridged* legs
Message-ID: <20110719180714.183000@gmx.com>

Thanks again, Avi, uuid_broadcast worked for me ! However an audio exchange between legs stops while file/tgml plays (even when played to single bridged leg) Can that issue be resolved by using conference only ?

 Thanks / Mike

----- Original Message -----
From: Avi Marcus
Sent: 07/16/11 09:12 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] playing audio to the *bridged* legs

 There's sched_broadcast which has leg options- http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_broadcast 
 Or uuid_broadcast http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast 
 You can use the first as an execute, or either one as an API command and specify the UUID.

 -Avi

 On Sat, Jul 16, 2011 at 6:18 PM, Mi Ke <http://mi.ke @http://null.net > wrote:
Hi All !

 Is it possible to play notification audio (wav or tgml) to one of *bridged* legs (without conferencing them) with Lua or any other way?

 session:streamFile executed on the leg gives audio neither on A or B while log shows file was played OK ...

 Thanks / Mike
 _______________________________________________
 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 msc at freeswitch.org  Tue Jul 19 22:18:18 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Jul 2011 11:18:18 -0700
Subject: [Freeswitch-users] Problem with API command expand
In-Reply-To: <CABtP9QhjuS+2yfw1p9xyqZSFbn8d9yGCw10N9c9PK5MAD0awSA@mail.gmail.com>
References: <CABtP9QhjuS+2yfw1p9xyqZSFbn8d9yGCw10N9c9PK5MAD0awSA@mail.gmail.com>
Message-ID: <CAKzWOxVrj-ZOqn6fvG7NjPPmbJBVdiuoCOypPty0ptNdBQESDA@mail.gmail.com>

What version of FS are you running? On the latest git I cannot do "api
expand" without getting a syntax error. However, this worked perfectly:

expand group insert:01@${domain_name}:'${sofia_contact(1001@
${domain_name})}'

Note the single quotes. The first colon in the URL returned by sofia_contact
is acting as a delimiter. Use the single quotes to prevent that.

-MC

On Tue, Jul 19, 2011 at 7:55 AM, Fabio Bigliardi
<fabio.bigliardi at gmail.com>wrote:

> Hi all,
>
> I would like to add users to a group through an API command.
> I've tried the following syntax:
>
> api expand group insert:01@${domain_name}:${sofia_contact(1001@
> ${domain_name})}
>
> But the output of the sqlite command:
>
> select * from group_data;
>
> is the following:
>
> 1LGL0Q1|01 at 192.168.25.61|sofia/internal/sip
>
> instead of the complete and correct one:
>
> 1LGL0Q1|01 at 192.168.25.61|sofia/internal/sip:1001 at 192.168.25.61:5070
>
> What am I doing wrong?
>
> Many thanks.
>
> FB
>
>
>
>
>
>
> _______________________________________________
> 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 msc at freeswitch.org  Tue Jul 19 22:30:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Jul 2011 11:30:36 -0700
Subject: [Freeswitch-users] X-PRE-PROCESS and Dialplan XML Module
In-Reply-To: <CACREEeaWtjiKGHbsw4d9WOT3UkXUJzxhmVYA7B5_=JyzPE5Fxw@mail.gmail.com>
References: <CACREEeYXb+Y3FddJ+aFDE-xRqNDPCfKBH-hC7BM8hrk2A8qQ+w@mail.gmail.com>
	<CAHKqMb7obu6Ob6j=ei8p=wQhvvrNdZ+TD-ZHxtBrPgBz9wrfKw@mail.gmail.com>
	<CACREEeaWtjiKGHbsw4d9WOT3UkXUJzxhmVYA7B5_=JyzPE5Fxw@mail.gmail.com>
Message-ID: <CAKzWOxU-Arh+O3ALk+_-2SNOQV1ZoPYPgMqeOs_7GaXEKm=MHw@mail.gmail.com>

On Tue, Jul 19, 2011 at 5:16 AM, Daniel Knoll <daniel at danielknoll.de> wrote:

> Hi All,
>
> i used XML/Curl module, but the problem at this module is, that he ask
> the webserver for every call. It is better to have static diaplans (in
> my case these have the most call traffic) and after parsing static
> plan he ask the webserver.
>

Just how much traffic are you talking about here? Unless it's many hundreds
of calls per second then a local web server should easily be able to handle
the request load. Believe me, the amount of resources that you will save by
using static XML first and then falling back to dynamic is minimal.

-MC
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From david.ponzone at ipeva.fr  Tue Jul 19 22:53:45 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 19 Jul 2011 20:53:45 +0200
Subject: [Freeswitch-users] Can't get fax working
In-Reply-To: <4E25952D.6080106@tagnet.ru>
References: <4E25605D.7050809@tagnet.ru> <4E25952D.6080106@tagnet.ru>
Message-ID: <9A383774-1B89-4D11-A459-174453254271@ipeva.fr>

Boris, what is your primary voice codec on the Audiocodes ?
If it is G711A, try setting G729 instead.

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

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

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




Le 19/07/2011 ? 16:31, Boris Kovalenko a ?crit :

> Hello!
> 
>     After digging I found that problem is with Audiocodes. 
> Kapanga SoftPhone --- FS -- Cisco 5350 -- PSTN -- fax is working
> Kapanga SoftPhone -- FS -- VOIP clould --- fax is working
> Kapanga SoftPhone -- FS -- AudioCodes MP-11x -- fax is not working....
> 
> So may be there is somebody who may help with AudioCodes configuration?
>> Hello!
>> 
>>     I need help with fax configuration. My network is:
>> 
>> PSTN E1 --- Cisco AS5350 -- Freeswitch -- Audiocodes MP-114 -- Fax
>> 
>> FS version: FreeSWITCH Version 1.0.head (git-062f07f 2011-07-16 12-35-25 +0200)
>> Audiocodes firmware: 6.00A.038.004
>> I can't get faxes working :( In FS profiles there is a parameter <param name="t38-passthru" value="true"/>
>> Cisco DP fax settings:
>> fax-relay ecm disable
>>  fax rate 9600
>>  fax protocol t38 ls-redundancy 2 hs-redundancy 1 fallback pass-through g711alaw
>>  no vad
>> 
>> On Audiocodes there is also T38 enabled (as I think) but there is no T38. There are messages on MP-114:
>> ErrMgs=16 T38Decoder received Non-T38 Packet
>> and
>>  Modem Relay Is Not Supported! Forcing Bypass Mode
>> 
>> 
>> Please, help! What information I need to collect to help solve the problem?
>> -- 
>> Regards,
>> Boris
>> 
>> 
>> 
>> _______________________________________________
>> 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
> 
> 
> -- 
> Regards,
> Boris
> 
> _______________________________________________
> 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 brian at freeswitch.org  Wed Jul 20 00:18:23 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Jul 2011 15:18:23 -0500
Subject: [Freeswitch-users] X-PRE-PROCESS and Dialplan XML Module
In-Reply-To: <CAKzWOxU-Arh+O3ALk+_-2SNOQV1ZoPYPgMqeOs_7GaXEKm=MHw@mail.gmail.com>
References: <CACREEeYXb+Y3FddJ+aFDE-xRqNDPCfKBH-hC7BM8hrk2A8qQ+w@mail.gmail.com>
	<CAHKqMb7obu6Ob6j=ei8p=wQhvvrNdZ+TD-ZHxtBrPgBz9wrfKw@mail.gmail.com>
	<CACREEeaWtjiKGHbsw4d9WOT3UkXUJzxhmVYA7B5_=JyzPE5Fxw@mail.gmail.com>
	<CAKzWOxU-Arh+O3ALk+_-2SNOQV1ZoPYPgMqeOs_7GaXEKm=MHw@mail.gmail.com>
Message-ID: <A92B9627-BC7E-4CC0-B59A-31E491F20AFE@freeswitch.org>

Well if you really really wanna go crazy? this is not well documented:

<?xml version="1.0"?>
<document type="freeswitch/xml">
<X-PRE-PROCESS cmd="exec" data="/path/to/my_uber_leet_script_that_barfs_all_configs_to_stdout.pl"/>
</document>

:P

/b
PS Linux only.

On Jul 19, 2011, at 1:30 PM, Michael Collins wrote:

> 
> Just how much traffic are you talking about here? Unless it's many hundreds
> of calls per second then a local web server should easily be able to handle
> the request load. Believe me, the amount of resources that you will save by
> using static XML first and then falling back to dynamic is minimal.
> 
> -MC



From daniel at danielknoll.de  Wed Jul 20 00:38:32 2011
From: daniel at danielknoll.de (Daniel Knoll)
Date: Tue, 19 Jul 2011 22:38:32 +0200
Subject: [Freeswitch-users] X-PRE-PROCESS and Dialplan XML Module
In-Reply-To: <A92B9627-BC7E-4CC0-B59A-31E491F20AFE@freeswitch.org>
References: <CACREEeYXb+Y3FddJ+aFDE-xRqNDPCfKBH-hC7BM8hrk2A8qQ+w@mail.gmail.com>
	<CAHKqMb7obu6Ob6j=ei8p=wQhvvrNdZ+TD-ZHxtBrPgBz9wrfKw@mail.gmail.com>
	<CACREEeaWtjiKGHbsw4d9WOT3UkXUJzxhmVYA7B5_=JyzPE5Fxw@mail.gmail.com>
	<CAKzWOxU-Arh+O3ALk+_-2SNOQV1ZoPYPgMqeOs_7GaXEKm=MHw@mail.gmail.com>
	<A92B9627-BC7E-4CC0-B59A-31E491F20AFE@freeswitch.org>
Message-ID: <525C2940-ACA2-4766-A6FB-E5D2BD3FF46D@danielknoll.de>

hahahaha. ok have understood. 
if even you say that ironic, it can't be a good solution. Now i use XML/curl, it's good, that i have 2 Webservers in a load balancing. 
I Think it's always good to show many opinions. The Mailing list is pretty much good way for that.

Thanks to all
Daniel

Am 19.07.2011 um 22:18 schrieb Brian West:

> Well if you really really wanna go crazy? this is not well documented:
> 
> <?xml version="1.0"?>
> <document type="freeswitch/xml">
> <X-PRE-PROCESS cmd="exec" data="/path/to/my_uber_leet_script_that_barfs_all_configs_to_stdout.pl"/>
> </document>
> 
> :P
> 
> /b
> PS Linux only.
> 
> On Jul 19, 2011, at 1:30 PM, Michael Collins wrote:
> 
>> 
>> Just how much traffic are you talking about here? Unless it's many hundreds
>> of calls per second then a local web server should easily be able to handle
>> the request load. Believe me, the amount of resources that you will save by
>> using static XML first and then falling back to dynamic is minimal.
>> 
>> -MC
> 
> 
> _______________________________________________
> 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 bwibowo at gmail.com  Wed Jul 20 03:24:22 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Wed, 20 Jul 2011 06:24:22 +0700
Subject: [Freeswitch-users] incoming call to FS
In-Reply-To: <CAKzWOxV_2PwB0Z5H9KY6c=unxAr1u-08KS1W+d4HthSA2sjYvw@mail.gmail.com>
References: <CAKnRR0ZkKt=QjB5a7eKdtFvYW+pq2SR7u-AFXr1TNPfGNfdObg@mail.gmail.com>
	<CAKzWOxV_2PwB0Z5H9KY6c=unxAr1u-08KS1W+d4HthSA2sjYvw@mail.gmail.com>
Message-ID: <CAKnRR0aW6BetQ69WOz6O12GpZDfj4DUkBOEoLB5+i8b8v=Qo5A@mail.gmail.com>

hi michael
after sometimes, now working thx.
i have question, is there any possibility to change external sip to 5060
instead of 5080?
why in FS design the port is different for internal and external profile?

thx
budi


On Tue, Jul 19, 2011 at 2:24 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Sun, Jul 17, 2011 at 5:58 AM, budi wibowo <bwibowo at gmail.com> wrote:
>
>> hi
>> i have sip server and want to connect the sip server to FS then call will
>> be routed using mod_dingaling, mod_dingaling already active
>> sipserver--->FS---mod_dingaling
>>
>> from FS it will be incoming sip call, i already define sipserver arameter
>> in external sip_profile, then make call but i cant see any call coming from
>> fs_cli
>> anything missing in my config?
>>
> Without seeing your config or a debug log it is difficult to tell.
> -MC
>
> _______________________________________________
> 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 mitch.johnson7 at gmail.com  Wed Jul 20 04:16:00 2011
From: mitch.johnson7 at gmail.com (Mitch Johnson)
Date: Tue, 19 Jul 2011 20:16:00 -0400
Subject: [Freeswitch-users] One way calling issue over an internal trunk
In-Reply-To: <mailman.13315.1311098867.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.13315.1311098867.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <85C0FE5C-56F3-482F-8D9C-1E9E9F7058DB@gmail.com>

I have a connection between my FreeSWITCH and a Cisco CallManager:

This is from the default.xml

 <extension name="Dial to CM8">
      <condition field="destination_number" expression="^(800\d)$">
        <action application="bridge" data="sofia/internal/$1 at 172.16.200.100"/>
      </condition>
     </extension>


The phone I dial on the CallManager PBX is number 8000.  As soon as it dials it fails and spews out the debug below.  However, I can call from the CallManager to a phone on FreeSWITCH.

Can anyone give me some ideas on how to find out what's wrong?  A method of troubleshooting this issue.  I don't even see the traffic make it to the CallManager.

I've done debugs and gone through them line by line.  I can call phone to phone on the FreeSWITCH.

I'm suspecting that the FreeSWITCH is looking in the wrong area to make the call, but I'm not sure how to verify that.

Many Thanks, Mitch

2011-07-19 19:21:47.662473 [DEBUG] sofia.c:6418 IP 192.168.60.104 Approved by acl "domains[]". Access Granted.
2011-07-19 19:21:47.662473 [NOTICE] switch_channel.c:812 New Channel sofia/internal/1002 at 172.16.200.60 [45a003c6-5a6b-430b-9214-a2279b4acf7e]
2011-07-19 19:21:47.664595 [DEBUG] sofia.c:4701 Channel sofia/internal/1002 at 172.16.200.60 entering state [received][100]
2011-07-19 19:21:47.664595 [DEBUG] sofia.c:4712 Remote SDP:
v=0
o=- 3520107294 3520107294 IN IP4 68.225.42.67
s=cpc_med
c=IN IP4 68.225.42.67
t=0 0
m=audio 18460 RTP/AVP 0 8 104 3 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15

2011-07-19 19:21:47.664595 [DEBUG] sofia_glue.c:4601 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000]
2011-07-19 19:21:47.664595 [DEBUG] sofia_glue.c:4601 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000]
2011-07-19 19:21:47.664595 [DEBUG] sofia_glue.c:4601 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
2011-07-19 19:21:47.664595 [DEBUG] sofia_glue.c:4601 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-07-19 19:21:47.664595 [DEBUG] sofia_glue.c:2760 Set Codec sofia/internal/1002 at 172.16.200.60 PCMU/8000 20 ms 160 samples 64000 bits
2011-07-19 19:21:47.664595 [DEBUG] sofia_glue.c:4713 Set 2833 dtmf send/recv payload to 96
2011-07-19 19:21:47.664595 [DEBUG] sofia.c:4879 (sofia/internal/1002 at 172.16.200.60) State Change CS_NEW -> CS_INIT
2011-07-19 19:21:47.664595 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 at 172.16.200.60 [BREAK]
2011-07-19 19:21:47.667744 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1002 at 172.16.200.60) Running State Change CS_INIT
2011-07-19 19:21:47.667744 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/1002 at 172.16.200.60) State INIT
2011-07-19 19:21:47.667744 [DEBUG] mod_sofia.c:84 sofia/internal/1002 at 172.16.200.60 SOFIA INIT
2011-07-19 19:21:47.667744 [DEBUG] mod_sofia.c:124 (sofia/internal/1002 at 172.16.200.60) State Change CS_INIT -> CS_ROUTING
2011-07-19 19:21:47.667744 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 at 172.16.200.60 [BREAK]
2011-07-19 19:21:47.667744 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/1002 at 172.16.200.60) State INIT going to sleep
2011-07-19 19:21:47.667744 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1002 at 172.16.200.60) Running State Change CS_ROUTING
2011-07-19 19:21:47.667744 [DEBUG] switch_channel.c:1664 (sofia/internal/1002 at 172.16.200.60) Callstate Change DOWN -> RINGING
2011-07-19 19:21:47.669630 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/1002 at 172.16.200.60) State ROUTING
2011-07-19 19:21:47.669630 [DEBUG] mod_sofia.c:147 sofia/internal/1002 at 172.16.200.60 SOFIA ROUTING
2011-07-19 19:21:47.669630 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1002 at 172.16.200.60 Standard ROUTING
2011-07-19 19:21:47.669630 [INFO] mod_dialplan_xml.c:331 Processing iPhone <1002>->8000 in context public
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [public->unloop] continue=false
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [public->outside_call] continue=true
Dialplan: sofia/internal/1002 at 172.16.200.60 Absolute Condition [outside_call]
Dialplan: sofia/internal/1002 at 172.16.200.60 Action set(outside_call=true) 
Dialplan: sofia/internal/1002 at 172.16.200.60 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [public->call_debug] continue=true
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [public->public_extensions] continue=false
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (FAIL) [public_extensions] destination_number(8000) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [public->Calls from CM8] continue=false
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (PASS) [Calls from CM8] destination_number(8000) =~ /^(800\d)$/ break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 Action transfer(8000 XML default) 
2011-07-19 19:21:47.671552 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1002 at 172.16.200.60) State Change CS_ROUTING -> CS_EXECUTE
2011-07-19 19:21:47.671552 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 at 172.16.200.60 [BREAK]
2011-07-19 19:21:47.671552 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/1002 at 172.16.200.60) State ROUTING going to sleep
2011-07-19 19:21:47.671552 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1002 at 172.16.200.60) Running State Change CS_EXECUTE
2011-07-19 19:21:47.671552 [DEBUG] switch_core_state_machine.c:366 (sofia/internal/1002 at 172.16.200.60) State EXECUTE
2011-07-19 19:21:47.672781 [DEBUG] mod_sofia.c:240 sofia/internal/1002 at 172.16.200.60 SOFIA EXECUTE
2011-07-19 19:21:47.672781 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1002 at 172.16.200.60 Standard EXECUTE
EXECUTE sofia/internal/1002 at 172.16.200.60 set(outside_call=true)
2011-07-19 19:21:47.672781 [DEBUG] mod_dptools.c:1059 sofia/internal/1002 at 172.16.200.60 SET [outside_call]=[true]
EXECUTE sofia/internal/1002 at 172.16.200.60 set(RFC2822_DATE=Tue, 19 Jul 2011 19:21:47 -0400)
2011-07-19 19:21:47.672781 [DEBUG] mod_dptools.c:1059 sofia/internal/1002 at 172.16.200.60 SET [RFC2822_DATE]=[Tue, 19 Jul 2011 19:21:47 -0400]
EXECUTE sofia/internal/1002 at 172.16.200.60 transfer(8000 XML default)
2011-07-19 19:21:47.674475 [DEBUG] switch_ivr.c:1600 (sofia/internal/1002 at 172.16.200.60) State Change CS_EXECUTE -> CS_ROUTING
2011-07-19 19:21:47.674475 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 at 172.16.200.60 [BREAK]
2011-07-19 19:21:47.674475 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/1002 at 172.16.200.60 [BREAK]
2011-07-19 19:21:47.674475 [NOTICE] switch_ivr.c:1606 Transfer sofia/internal/1002 at 172.16.200.60 to XML[8000 at default]
2011-07-19 19:21:47.676173 [DEBUG] switch_core_state_machine.c:366 (sofia/internal/1002 at 172.16.200.60) State EXECUTE going to sleep
2011-07-19 19:21:47.676173 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1002 at 172.16.200.60) Running State Change CS_ROUTING
2011-07-19 19:21:47.676173 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/1002 at 172.16.200.60) State ROUTING
2011-07-19 19:21:47.676173 [DEBUG] mod_sofia.c:147 sofia/internal/1002 at 172.16.200.60 SOFIA ROUTING
2011-07-19 19:21:47.676173 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1002 at 172.16.200.60 Standard ROUTING
2011-07-19 19:21:47.676173 [INFO] mod_dialplan_xml.c:331 Processing iPhone <1002>->8000 in context default
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [default->unloop] continue=false
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [default->tod_example] continue=true
Dialplan: sofia/internal/1002 at 172.16.200.60 Date/Time Match (FAIL) [tod_example] break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [default->holiday_example] continue=true
Dialplan: sofia/internal/1002 at 172.16.200.60 Date/Time Match (FAIL) [holiday_example] break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [default->global-intercept] continue=false
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (FAIL) [global-intercept] destination_number(8000) =~ /^886$/ break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [default->group-intercept] continue=false
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (FAIL) [group-intercept] destination_number(8000) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [default->intercept-ext] continue=false
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (FAIL) [intercept-ext] destination_number(8000) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [default->redial] continue=false
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (FAIL) [redial] destination_number(8000) =~ /^(redial|870)$/ break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [default->global] continue=true
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: sofia/internal/1002 at 172.16.200.60 Absolute Condition [global]
Dialplan: sofia/internal/1002 at 172.16.200.60 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) 
Dialplan: sofia/internal/1002 at 172.16.200.60 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) 
Dialplan: sofia/internal/1002 at 172.16.200.60 Action hash(insert/${domain_name}-last_dial/global/${uuid}) 
Dialplan: sofia/internal/1002 at 172.16.200.60 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [default->snom-demo-2] continue=false
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (FAIL) [snom-demo-2] destination_number(8000) =~ /^9001$/ break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [default->snom-demo-1] continue=false
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (FAIL) [snom-demo-1] destination_number(8000) =~ /^9000$/ break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [default->eavesdrop] continue=false
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (FAIL) [eavesdrop] destination_number(8000) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [default->eavesdrop] continue=false
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (FAIL) [eavesdrop] destination_number(8000) =~ /^779$/ break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [default->call_return] continue=false
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (FAIL) [call_return] destination_number(8000) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 parsing [default->del-group] continue=false
Dialplan: sofia/internal/1002 at 172.16.200.60 Regex (PASS) [del-group] destination_number(8000) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/internal/1002 at 172.16.200.60 Action answer() 
Dialplan: sofia/internal/1002 at 172.16.200.60 Action group(delete:00@${domain_name}:${sofia_contact(${sip_from_user}@${domain_name})}) 
Dialplan: sofia/internal/1002 at 172.16.200.60 Action gentones(%(1000, 0, 320)) 
2011-07-19 19:21:47.680477 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1002 at 172.16.200.60) State Change CS_ROUTING -> CS_EXECUTE
2011-07-19 19:21:47.680477 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 at 172.16.200.60 [BREAK]
2011-07-19 19:21:47.680477 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/1002 at 172.16.200.60) State ROUTING going to sleep
2011-07-19 19:21:47.680477 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1002 at 172.16.200.60) Running State Change CS_EXECUTE
2011-07-19 19:21:47.680477 [DEBUG] switch_core_state_machine.c:366 (sofia/internal/1002 at 172.16.200.60) State EXECUTE
2011-07-19 19:21:47.681477 [DEBUG] mod_sofia.c:240 sofia/internal/1002 at 172.16.200.60 SOFIA EXECUTE
2011-07-19 19:21:47.681477 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1002 at 172.16.200.60 Standard EXECUTE
EXECUTE sofia/internal/1002 at 172.16.200.60 hash(insert/172.16.200.60-spymap/1002/45a003c6-5a6b-430b-9214-a2279b4acf7e)
EXECUTE sofia/internal/1002 at 172.16.200.60 hash(insert/172.16.200.60-last_dial/1002/8000)
EXECUTE sofia/internal/1002 at 172.16.200.60 hash(insert/172.16.200.60-last_dial/global/45a003c6-5a6b-430b-9214-a2279b4acf7e)
EXECUTE sofia/internal/1002 at 172.16.200.60 set(RFC2822_DATE=Tue, 19 Jul 2011 19:21:47 -0400)
2011-07-19 19:21:47.683508 [DEBUG] mod_dptools.c:1059 sofia/internal/1002 at 172.16.200.60 SET [RFC2822_DATE]=[Tue, 19 Jul 2011 19:21:47 -0400]
EXECUTE sofia/internal/1002 at 172.16.200.60 answer()
2011-07-19 19:21:47.686486 [DEBUG] sofia_glue.c:2990 AUDIO RTP [sofia/internal/1002 at 172.16.200.60] 172.16.200.60 port 16574 -> 68.225.42.67 port 18460 codec: 0 ms: 20
2011-07-19 19:21:47.686486 [DEBUG] switch_rtp.c:1621 Starting timer [soft] 160 bytes per 20ms
2011-07-19 19:21:47.687953 [DEBUG] sofia_glue.c:3245 Set 2833 dtmf send payload to 96
2011-07-19 19:21:47.687953 [DEBUG] sofia_glue.c:3250 Set 2833 dtmf receive payload to 96
2011-07-19 19:21:47.687953 [DEBUG] mod_sofia.c:681 Local SDP sofia/internal/1002 at 172.16.200.60:
v=0
o=FreeSWITCH 1311101133 1311101134 IN IP4 172.16.200.60
s=FreeSWITCH
c=IN IP4 172.16.200.60
t=0 0
m=audio 16574 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2011-07-19 19:21:47.689648 [DEBUG] sofia.c:4701 Channel sofia/internal/1002 at 172.16.200.60 entering state [completed][200]
2011-07-19 19:21:47.689648 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/1002 at 172.16.200.60 [BREAK]
2011-07-19 19:21:47.689648 [DEBUG] switch_channel.c:2795 (sofia/internal/1002 at 172.16.200.60) Callstate Change RINGING -> ACTIVE
2011-07-19 19:21:47.690588 [NOTICE] mod_dptools.c:929 Channel [sofia/internal/1002 at 172.16.200.60] has been answered
EXECUTE sofia/internal/1002 at 172.16.200.60 group(delete:00 at 172.16.200.60:sofia/internal/sip:1002 at 192.168.60.104:50167;transport=UDP;fs_nat=yes;fs_path=sip%3A1002%40192.168.60.104%3A50167%3Btransport%3DUDP)
EXECUTE sofia/internal/1002 at 172.16.200.60 gentones(%(1000, 0, 320))
2011-07-19 19:21:47.849590 [DEBUG] sofia.c:4701 Channel sofia/internal/1002 at 172.16.200.60 entering state [ready][200]
2011-07-19 19:21:48.745448 [NOTICE] switch_core_state_machine.c:189 sofia/internal/1002 at 172.16.200.60 has executed the last dialplan instruction, hanging up.
2011-07-19 19:21:48.745448 [DEBUG] switch_channel.c:2545 (sofia/internal/1002 at 172.16.200.60) Callstate Change ACTIVE -> HANGUP
2011-07-19 19:21:48.745448 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/internal/1002 at 172.16.200.60 [CS_EXECUTE] [NORMAL_CLEARING]
2011-07-19 19:21:48.745448 [DEBUG] switch_channel.c:2561 Send signal sofia/internal/1002 at 172.16.200.60 [KILL]
2011-07-19 19:21:48.745448 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 at 172.16.200.60 [BREAK]
2011-07-19 19:21:48.745448 [DEBUG] switch_core_state_machine.c:366 (sofia/internal/1002 at 172.16.200.60) State EXECUTE going to sleep
2011-07-19 19:21:48.745448 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1002 at 172.16.200.60) Running State Change CS_HANGUP
2011-07-19 19:21:48.745448 [DEBUG] switch_core_state_machine.c:560 (sofia/internal/1002 at 172.16.200.60) State HANGUP
2011-07-19 19:21:48.745448 [DEBUG] mod_sofia.c:457 Channel sofia/internal/1002 at 172.16.200.60 hanging up, cause: NORMAL_CLEARING
2011-07-19 19:21:48.745448 [DEBUG] mod_sofia.c:500 Sending BYE to sofia/internal/1002 at 172.16.200.60
2011-07-19 19:21:48.745448 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1002 at 172.16.200.60 Standard HANGUP, cause: NORMAL_CLEARING
2011-07-19 19:21:48.745448 [DEBUG] switch_core_state_machine.c:560 (sofia/internal/1002 at 172.16.200.60) State HANGUP going to sleep
2011-07-19 19:21:48.745448 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/1002 at 172.16.200.60) State Change CS_HANGUP -> CS_REPORTING
2011-07-19 19:21:48.745448 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 at 172.16.200.60 [BREAK]
2011-07-19 19:21:48.745448 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1002 at 172.16.200.60) Running State Change CS_REPORTING
2011-07-19 19:21:48.745448 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/1002 at 172.16.200.60) State REPORTING
2011-07-19 19:21:48.745448 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1002 at 172.16.200.60 Standard REPORTING, cause: NORMAL_CLEARING
2011-07-19 19:21:48.745448 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/1002 at 172.16.200.60) State REPORTING going to sleep
2011-07-19 19:21:48.745448 [DEBUG] switch_core_state_machine.c:345 (sofia/internal/1002 at 172.16.200.60) State Change CS_REPORTING -> CS_DESTROY
2011-07-19 19:21:48.745448 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 at 172.16.200.60 [BREAK]
2011-07-19 19:21:48.745448 [DEBUG] switch_core_session.c:1288 Session 39 (sofia/internal/1002 at 172.16.200.60) Locked, Waiting on external entities
2011-07-19 19:21:48.745448 [NOTICE] switch_core_session.c:1306 Session 39 (sofia/internal/1002 at 172.16.200.60) Ended
2011-07-19 19:21:48.745448 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/1002 at 172.16.200.60 [CS_DESTROY]
2011-07-19 19:21:48.745448 [DEBUG] switch_core_state_machine.c:449 (sofia/internal/1002 at 172.16.200.60) Callstate Change HANGUP -> DOWN
2011-07-19 19:21:48.745448 [DEBUG] switch_core_state_machine.c:452 (sofia/internal/1002 at 172.16.200.60) Running State Change CS_DESTROY
2011-07-19 19:21:48.745448 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/1002 at 172.16.200.60) State DESTROY
2011-07-19 19:21:48.745448 [DEBUG] mod_sofia.c:362 sofia/internal/1002 at 172.16.200.60 SOFIA DESTROY
2011-07-19 19:21:48.745448 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1002 at 172.16.200.60 Standard DESTROY
2011-07-19 19:21:48.745448 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/1002 at 172.16.200.60) State DESTROY going to sleep



From david.villasmil.work at gmail.com  Wed Jul 20 04:17:32 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 20 Jul 2011 02:17:32 +0200
Subject: [Freeswitch-users] X-PRE-PROCESS and Dialplan XML Module
In-Reply-To: <A92B9627-BC7E-4CC0-B59A-31E491F20AFE@freeswitch.org>
References: <CACREEeYXb+Y3FddJ+aFDE-xRqNDPCfKBH-hC7BM8hrk2A8qQ+w@mail.gmail.com>
	<CAHKqMb7obu6Ob6j=ei8p=wQhvvrNdZ+TD-ZHxtBrPgBz9wrfKw@mail.gmail.com>
	<CACREEeaWtjiKGHbsw4d9WOT3UkXUJzxhmVYA7B5_=JyzPE5Fxw@mail.gmail.com>
	<CAKzWOxU-Arh+O3ALk+_-2SNOQV1ZoPYPgMqeOs_7GaXEKm=MHw@mail.gmail.com>
	<A92B9627-BC7E-4CC0-B59A-31E491F20AFE@freeswitch.org>
Message-ID: <CAFGRPVqkxKzP2BbBCnt9mBHyP_ZOAEP-ADiNOBdVtWoVx9Tv7A@mail.gmail.com>

This is incredibly cool! why isn't this documented!? :P

thanks for this!

David

On Tue, Jul 19, 2011 at 10:18 PM, Brian West <brian at freeswitch.org> wrote:

> Well if you really really wanna go crazy? this is not well documented:
>
> <?xml version="1.0"?>
> <document type="freeswitch/xml">
> <X-PRE-PROCESS cmd="exec" data="/path/to/
> my_uber_leet_script_that_barfs_all_configs_to_stdout.pl"/>
> </document>
>
> :P
>
> /b
> PS Linux only.
>
> On Jul 19, 2011, at 1:30 PM, Michael Collins wrote:
>
> >
> > Just how much traffic are you talking about here? Unless it's many
> hundreds
> > of calls per second then a local web server should easily be able to
> handle
> > the request load. Believe me, the amount of resources that you will save
> by
> > using static XML first and then falling back to dynamic is minimal.
> >
> > -MC
>
>
> _______________________________________________
> 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 david.ponzone at ipeva.fr  Wed Jul 20 04:30:40 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 20 Jul 2011 02:30:40 +0200
Subject: [Freeswitch-users] incoming call to FS
In-Reply-To: <CAKnRR0aW6BetQ69WOz6O12GpZDfj4DUkBOEoLB5+i8b8v=Qo5A@mail.gmail.com>
References: <CAKnRR0ZkKt=QjB5a7eKdtFvYW+pq2SR7u-AFXr1TNPfGNfdObg@mail.gmail.com>
	<CAKzWOxV_2PwB0Z5H9KY6c=unxAr1u-08KS1W+d4HthSA2sjYvw@mail.gmail.com>
	<CAKnRR0aW6BetQ69WOz6O12GpZDfj4DUkBOEoLB5+i8b8v=Qo5A@mail.gmail.com>
Message-ID: <F9654662-BCA0-47D9-BB3E-CFFB8F93F947@ipeva.fr>

Budi,

you may change ports as you wish.
Check vars.xml.

why ?
Because you could need to do different things: authenticate everything coming in through internal, but through external, just IP-auth gateways sending you trafic, etc...

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

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

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




Le 20/07/2011 ? 01:24, budi wibowo a ?crit :

> hi michael 
> after sometimes, now working thx.
> i have question, is there any possibility to change external sip to 5060 instead of 5080?
> why in FS design the port is different for internal and external profile?
> 
> thx
> budi 
> 
> 
> On Tue, Jul 19, 2011 at 2:24 AM, Michael Collins <msc at freeswitch.org> wrote:
> 
> 
> On Sun, Jul 17, 2011 at 5:58 AM, budi wibowo <bwibowo at gmail.com> wrote:
> hi 
> i have sip server and want to connect the sip server to FS then call will be routed using mod_dingaling, mod_dingaling already active
> sipserver--->FS---mod_dingaling
> 
> from FS it will be incoming sip call, i already define sipserver arameter in external sip_profile, then make call but i cant see any call coming from fs_cli 
> anything missing in my config?
> Without seeing your config or a debug log it is difficult to tell.
> -MC
> 
> _______________________________________________
> 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
> 
> 
> _______________________________________________
> 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 brian at freeswitch.org  Wed Jul 20 04:55:02 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Jul 2011 19:55:02 -0500
Subject: [Freeswitch-users] One way calling issue over an internal trunk
In-Reply-To: <85C0FE5C-56F3-482F-8D9C-1E9E9F7058DB@gmail.com>
References: <mailman.13315.1311098867.24555.freeswitch-users@lists.freeswitch.org>
	<85C0FE5C-56F3-482F-8D9C-1E9E9F7058DB@gmail.com>
Message-ID: <9828BFD4-CFF7-47D1-BD99-107E99C40F38@freeswitch.org>

you need to set the ext-sip-ip and ext-rtp-ip and deal with setting local-network-acl to the proper network/mask because you're answering with RFC1918 media IP's to the cisco? its retarded and will just start sending to those IP's and they'll usually never make it back to you.

/b

On Jul 19, 2011, at 7:16 PM, Mitch Johnson wrote:

> I have a connection between my FreeSWITCH and a Cisco CallManager:
> 
> This is from the default.xml
> 
> <extension name="Dial to CM8">
>      <condition field="destination_number" expression="^(800\d)$">
>        <action application="bridge" data="sofia/internal/$1 at 172.16.200.100"/>
>      </condition>
>     </extension>

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From brian at freeswitch.org  Wed Jul 20 04:55:38 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Jul 2011 19:55:38 -0500
Subject: [Freeswitch-users] X-PRE-PROCESS and Dialplan XML Module
In-Reply-To: <CAFGRPVqkxKzP2BbBCnt9mBHyP_ZOAEP-ADiNOBdVtWoVx9Tv7A@mail.gmail.com>
References: <CACREEeYXb+Y3FddJ+aFDE-xRqNDPCfKBH-hC7BM8hrk2A8qQ+w@mail.gmail.com>
	<CAHKqMb7obu6Ob6j=ei8p=wQhvvrNdZ+TD-ZHxtBrPgBz9wrfKw@mail.gmail.com>
	<CACREEeaWtjiKGHbsw4d9WOT3UkXUJzxhmVYA7B5_=JyzPE5Fxw@mail.gmail.com>
	<CAKzWOxU-Arh+O3ALk+_-2SNOQV1ZoPYPgMqeOs_7GaXEKm=MHw@mail.gmail.com>
	<A92B9627-BC7E-4CC0-B59A-31E491F20AFE@freeswitch.org>
	<CAFGRPVqkxKzP2BbBCnt9mBHyP_ZOAEP-ADiNOBdVtWoVx9Tv7A@mail.gmail.com>
Message-ID: <F4772298-D830-44C0-A2E5-ED5DE5B3BF86@freeswitch.org>

Linux only? see scripts/perl/  I have two scripts that use this method.

/b

On Jul 19, 2011, at 7:17 PM, David Villasmil wrote:

> This is incredibly cool! why isn't this documented!? :P

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From boris at tagnet.ru  Wed Jul 20 07:29:18 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Wed, 20 Jul 2011 09:29:18 +0600
Subject: [Freeswitch-users] Can't get fax working
In-Reply-To: <9A383774-1B89-4D11-A459-174453254271@ipeva.fr>
References: <4E25605D.7050809@tagnet.ru> <4E25952D.6080106@tagnet.ru>
	<9A383774-1B89-4D11-A459-174453254271@ipeva.fr>
Message-ID: <4E264B8E.8040008@tagnet.ru>

Hello!

     Doesn't help, David. Or may be I don't understood what primary 
codec is? I changed the order of codecs in tel profile.

> Boris, what is your primary voice codec on the Audiocodes ?
> If it is G711A, try setting G729 instead.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service ClientIPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 19/07/2011 ? 16:31, Boris Kovalenko a ?crit :
>
>> Hello!
>>
>>     After digging I found that problem is with Audiocodes.
>> Kapanga SoftPhone --- FS -- Cisco 5350 -- PSTN -- fax is working
>> Kapanga SoftPhone -- FS -- VOIP clould --- fax is working
>> Kapanga SoftPhone -- FS -- AudioCodes MP-11x -- fax is not working....
>>
>> So may be there is somebody who may help with AudioCodes configuration?
>>> Hello!
>>>
>>>     I need help with fax configuration. My network is:
>>>
>>> PSTN E1 --- Cisco AS5350 -- Freeswitch -- Audiocodes MP-114 -- Fax
>>>
>>> FS version: FreeSWITCH Version 1.0.head (git-062f07f 2011-07-16 
>>> 12-35-25 +0200)
>>> Audiocodes firmware: 6.00A.038.004
>>> I can't get faxes working :( In FS profiles there is a parameter 
>>> <param name="t38-passthru" value="true"/>
>>> Cisco DP fax settings:
>>> fax-relay ecm disable
>>>  fax rate 9600
>>>  fax protocol t38 ls-redundancy 2 hs-redundancy 1 fallback 
>>> pass-through g711alaw
>>>  no vad
>>>
>>> On Audiocodes there is also T38 enabled (as I think) but there is no 
>>> T38. There are messages on MP-114:
>>> ErrMgs=16 T38Decoder received Non-T38 Packet
>>> and
>>> ||
>>> Modem Relay Is Not Supported! Forcing Bypass Mode
>>>
>>>
>>> Please, help! What information I need to collect to help solve the problem?
>>> -- 
>>> Regards,
>>> Boris
>>>
>>>
>>>
>>> _______________________________________________
>>> 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
>>
>>
>> -- 
>> Regards,
>> Boris
>>
>> _______________________________________________
>> 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 
>> <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
>
>
>
> _______________________________________________
> 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


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005

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From valery.kalinin at gmail.com  Wed Jul 20 08:01:23 2011
From: valery.kalinin at gmail.com (Valery Kalinin)
Date: Wed, 20 Jul 2011 10:01:23 +0600
Subject: [Freeswitch-users] FreeTDM viewer
Message-ID: <CAF=swh-E5=Utv9YF2ZATYF=GeSqtJRC685pMUbJiyio3v5qXBA@mail.gmail.com>

I'm write small web utility for FreeTDM channel statuses view:

FreeTDM viewer
Web-based online view FreeTDM spans & channels statuses.

Download from:
https://sites.google.com/site/freeswitched/home/downloads

Installation guide:
    Upload files (ftdmv.php & jquery-1.6.1.min.js) to your web-server with
PHP support.
    Check modules.conf and event_socket.conf for enable module
mod_event_socket
    Change IP address, port, password for FreeSWITCH server and FreeTDM
spans in ftdmv.php file
    Enter in browser http://you-server-name/ftdmv.php
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From kheimerl at cs.berkeley.edu  Wed Jul 20 10:00:17 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Tue, 19 Jul 2011 23:00:17 -0700
Subject: [Freeswitch-users] Using chat command in a dialplan
Message-ID: <CACyT-3m7nMTCwav_j_CVqtL-V+T9vFcyGwsrD6uatJ+bkVti3Q@mail.gmail.com>

I feel somewhat embarrassed, but I can't figure out how to cause a
chat event to go off from the dialplan. I've tried:

<action application="chat"
data="sip|${username}|${destination_number}@${domain_name}|${msg_body}"/>

and

<action application="eval" data="chat
sip|${username}|${destination_number}@${domain_name}|${msg_body}"/>

and neither worked. The wiki says it's part of the API
(http://wiki.freeswitch.org/wiki/Mod_dptools) but I'm pretty sure
that's just a command-line API. Is there a way to run a command-line
call from the dialplan?

Any direction would be appreciated.


From fdelawarde at wirelessmundi.com  Wed Jul 20 12:14:40 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Wed, 20 Jul 2011 10:14:40 +0200
Subject: [Freeswitch-users] Using chat command in a dialplan
In-Reply-To: <CACyT-3m7nMTCwav_j_CVqtL-V+T9vFcyGwsrD6uatJ+bkVti3Q@mail.gmail.com>
References: <CACyT-3m7nMTCwav_j_CVqtL-V+T9vFcyGwsrD6uatJ+bkVti3Q@mail.gmail.com>
Message-ID: <1311149680.32170.264.camel@luna.madrid.commsmundi.com>

Hi,

Take a look at:
http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan


Fran?ois.

On Tue, 2011-07-19 at 23:00 -0700, Kurtis Heimerl wrote: 
> I feel somewhat embarrassed, but I can't figure out how to cause a
> chat event to go off from the dialplan. I've tried:
> 
> <action application="chat"
> data="sip|${username}|${destination_number}@${domain_name}|${msg_body}"/>
> 
> and
> 
> <action application="eval" data="chat
> sip|${username}|${destination_number}@${domain_name}|${msg_body}"/>
> 
> and neither worked. The wiki says it's part of the API
> (http://wiki.freeswitch.org/wiki/Mod_dptools) but I'm pretty sure
> that's just a command-line API. Is there a way to run a command-line
> call from the dialplan?
> 
> Any direction would be appreciated.
> 
> _______________________________________________
> 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 steveayre at gmail.com  Wed Jul 20 12:21:48 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 20 Jul 2011 09:21:48 +0100
Subject: [Freeswitch-users] Using chat command in a dialplan
In-Reply-To: <1311149680.32170.264.camel@luna.madrid.commsmundi.com>
References: <CACyT-3m7nMTCwav_j_CVqtL-V+T9vFcyGwsrD6uatJ+bkVti3Q@mail.gmail.com>
	<1311149680.32170.264.camel@luna.madrid.commsmundi.com>
Message-ID: <83217555-012B-4861-AAB9-045B2E78660D@gmail.com>

Kurtis,

*Any* api can be run from the dialplan. It's just a slightly different syntax - the link Francois gave you will show you how.

Steve on iPhone

On 20 Jul 2011, at 09:14, Fran?ois Delawarde <fdelawarde at wirelessmundi.com> wrote:

> Hi,
> 
> Take a look at:
> http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
> 
> 
> Fran?ois.
> 
> On Tue, 2011-07-19 at 23:00 -0700, Kurtis Heimerl wrote: 
>> I feel somewhat embarrassed, but I can't figure out how to cause a
>> chat event to go off from the dialplan. I've tried:
>> 
>> <action application="chat"
>> data="sip|${username}|${destination_number}@${domain_name}|${msg_body}"/>
>> 
>> and
>> 
>> <action application="eval" data="chat
>> sip|${username}|${destination_number}@${domain_name}|${msg_body}"/>
>> 
>> and neither worked. The wiki says it's part of the API
>> (http://wiki.freeswitch.org/wiki/Mod_dptools) but I'm pretty sure
>> that's just a command-line API. Is there a way to run a command-line
>> call from the dialplan?
>> 
>> Any direction would be appreciated.
>> 
>> _______________________________________________
>> 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
> 
> 
> 
> _______________________________________________
> 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 yehavi.bourvine at gmail.com  Wed Jul 20 12:22:31 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 20 Jul 2011 11:22:31 +0300
Subject: [Freeswitch-users] Can't get fax working
In-Reply-To: <4E25605D.7050809@tagnet.ru>
References: <4E25605D.7050809@tagnet.ru>
Message-ID: <CAAtWKeHYhfoY2OT4X0QsaFv5uZ-f=MYGxCrHQCU9t-PRn85TaA@mail.gmail.com>

Hello Boris,

  It seems like your Audiocodes is misconfigured. I had a different problem
which might be (or not) related to your problem: Both FS and Audiocodes
asked for T.38 and it didn't work.

Do you have T38 enabled in FreeSwitch? If so, try the following at your MP:
- At the Fax/Modem screen disable all fax and modems protocols.
- At all other screens set the fax protocol to be T.38
- Add the T.38 codec.

I hope this helps.

                           __Yehavi:


2011/7/19 Boris Kovalenko <boris at tagnet.ru>

>  Hello!
>
>     I need help with fax configuration. My network is:
>
> PSTN E1 --- Cisco AS5350 -- Freeswitch -- Audiocodes MP-114 -- Fax
>
> FS version: FreeSWITCH Version 1.0.head (git-062f07f 2011-07-16 12-35-25
> +0200)
> Audiocodes firmware: 6.00A.038.004
> I can't get faxes working :( In FS profiles there is a parameter <param
> name="t38-passthru" value="true"/>
> Cisco DP fax settings:
> fax-relay ecm disable
>  fax rate 9600
>  fax protocol t38 ls-redundancy 2 hs-redundancy 1 fallback pass-through
> g711alaw
>  no vad
>
> On Audiocodes there is also T38 enabled (as I think) but there is no T38.
> There are messages on MP-114:
> ErrMgs=16 T38Decoder received Non-T38 Packet
> and
>
> Modem Relay Is Not Supported! Forcing Bypass Mode
>
>
> Please, help! What information I need to collect to help solve the problem?
>
> --
> Regards,
> Boris
>
>
>
> _______________________________________________
> 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 boris at tagnet.ru  Wed Jul 20 12:38:48 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Wed, 20 Jul 2011 14:38:48 +0600
Subject: [Freeswitch-users] Can't get fax working
In-Reply-To: <CAAtWKeHYhfoY2OT4X0QsaFv5uZ-f=MYGxCrHQCU9t-PRn85TaA@mail.gmail.com>
References: <4E25605D.7050809@tagnet.ru>
	<CAAtWKeHYhfoY2OT4X0QsaFv5uZ-f=MYGxCrHQCU9t-PRn85TaA@mail.gmail.com>
Message-ID: <4E269418.6050206@tagnet.ru>

Hello!
> Hello Boris,
>   It seems like your Audiocodes is misconfigured. I had a different 
> problem which might be (or not) related to your problem: Both FS and 
> Audiocodes asked for T.38 and it didn't work.
> Do you have T38 enabled in FreeSwitch? If so, try the following at 
> your MP:
I think I have. This is my configuration:
<configuration name="fax.conf" description="FAX application configuration">
<settings>
<param name="use-ecm"                   value="false"/>
<param name="verbose"                   value="true"/>
<param name="disable-v17"               value="false"/>
<param name="ident"                     value="SpanDSP Fax Ident"/>
<param name="header"                    value="SpanDSP Fax Header"/>
<param name="spool-dir"                 value="/tmp"/>
<param name="file-prefix"               value="faxrx"/>
<param name="enable-t38"                value="true"/>
<param name="enable-t38-request"        value="false"/>
</settings>
</configuration>

Also I have <param name="t38-passthru" value="true"/> in every sip 
profile. Also
<!-- <param name="inbound-bypass-media" value="true"/> -->
<!-- <param name="inbound-proxy-media" value="true"/> -->
parameters are commented. Is this right and enought configuration for T38?

> - At the Fax/Modem screen disable all fax and modems protocols.
> - At all other screens set the fax protocol to be T.38
> - Add the T.38 codec.
Tried with no success.
> I hope this helps.
>                            __Yehavi:
>
>
> 2011/7/19 Boris Kovalenko <boris at tagnet.ru <mailto:boris at tagnet.ru>>
>
>     Hello!
>
>         I need help with fax configuration. My network is:
>
>     PSTN E1 --- Cisco AS5350 -- Freeswitch -- Audiocodes MP-114 -- Fax
>
>     FS version: FreeSWITCH Version 1.0.head (git-062f07f 2011-07-16
>     12-35-25 +0200)
>     Audiocodes firmware: 6.00A.038.004
>     I can't get faxes working :( In FS profiles there is a parameter
>     <param name="t38-passthru" value="true"/>
>     Cisco DP fax settings:
>     fax-relay ecm disable
>      fax rate 9600
>      fax protocol t38 ls-redundancy 2 hs-redundancy 1 fallback
>     pass-through g711alaw
>      no vad
>
>     On Audiocodes there is also T38 enabled (as I think) but there is
>     no T38. There are messages on MP-114:
>     ErrMgs=16 T38Decoder received Non-T38 Packet
>     and
>     ||
>
>     Modem Relay Is Not Supported! Forcing Bypass Mode
>
>
>     Please, help! What information I need to collect to help solve the problem?
>
>     -- 
>     Regards,
>     Boris
>
>
>
>     _______________________________________________
>     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
>     <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
>
>
>
>
> _______________________________________________
> 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


-- 
Regards,
Boris


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From saeedahmad1981 at gmail.com  Wed Jul 20 13:11:27 2011
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Wed, 20 Jul 2011 11:11:27 +0200
Subject: [Freeswitch-users] absolute_codec_string just process one codec
Message-ID: <4E269BBF.1010805@gmail.com>

Hi,

If i set absoluteye_codec_string in dialplan with multiple codecs, like 
G729,PCMA,PCMU, then just first codec will be processed, rest will be 
ignored.

Example: {absolute_codec_string=${ep_codec_string}, i even tried using 
it like: absolute_codec_string='${ep_codec_string}'
or absolute_codec_string=\'${ep_codec_string}\'
or absolute_codec_string=\\'${ep_codec_string}\\'
(as suggested by bkw_ on IRC)

Console log shows like that:

EXECUTE sofia/profile/1234567 at 1.2.2.3 
bridge({absolute_codec_string=PCMA at 8000h@20i at 64000b,PCMU at 8000h@20i at 64000b,G729 at 8000h@20i at 8000b,leg_time_out=
10,process_cdr=b_only}[SupplierPrefix=]sofia/fstest/456789 at 7.2.1.4)


2011-07-20 11:03:59.448455 [DEBUG] switch_ivr_originate.c:1873 Parsing 
global variables
2011-07-20 11:03:59.448455 [DEBUG] switch_event.c:1170 Parsing variable 
[absolute_codec_string]=[PCMA at 8000h@20i at 64000b]
2011-07-20 11:03:59.448455 [DEBUG] switch_event.c:1170 Parsing variable 
[leg_time_out]=[10]
2011-07-20 11:03:59.448455 [DEBUG] switch_event.c:1170 Parsing variable 
[process_cdr]=[b_only]

Am i doing something wrong or should i open a JIRA?


-- 

Kind Regards

Saeed Ahmed

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From steveayre at gmail.com  Wed Jul 20 14:16:32 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 20 Jul 2011 11:16:32 +0100
Subject: [Freeswitch-users] absolute_codec_string just process one codec
In-Reply-To: <4E269BBF.1010805@gmail.com>
References: <4E269BBF.1010805@gmail.com>
Message-ID: <CAFiqYukBJ-BBFZAp99weP=UCxcqJBj9EV=Xrx2iSqtzM_0HCHw@mail.gmail.com>

You are... but it's perhaps not a very well documented feature so you
can be forgiven...

The problem is that , separates variable assignments in {} and []
blocks. So what you're using is actually setting the following
variables:
    absolute_codec_string=PCMA at 8000h@20i at 64000b
    PCMU at 8000h@20i at 64000b=
    G729 at 8000h@20i at 8000b=
    leg_time_out=10
    process_cdr=b_only
which hopefully makes it obvious why it's only using the first codec.

It's a known problem (http://jira.freeswitch.org/browse/FS-2126) and
there *is* a workaround for it. There's a special syntax that lets you
use a character other than comma for separating the codec names. This
actually applies to any variable values that are set in the {} and []
blocks and the special character gets replaced by a comma during the
variable assignment.

The syntax is to put a prefix of ^^: at the start of the value and
replace the commas in the value with :
You can use any (almost) character you like though, it uses the one
after the initial ^^.

As a short example: [absolute_codec_string=^^:PCMU:PCMA]

For your example you'll want to pass in:
{absolute_codec_string=^^:PCMA at 8000h@20i at 64000b:PCMU at 8000h@20i at 64000b:G729 at 8000h@20i at 8000b,leg_time_out=10,process_cdr=b_only}

Since you appear to be getting that from a variable you'll probably
want to find some way of substituting the , for the :

-Steve



On 20 July 2011 10:11, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> Hi,
>
> If i set absoluteye_codec_string in dialplan with multiple codecs, like
> G729,PCMA,PCMU, then just first codec will be processed, rest will be
> ignored.
>
> Example: {absolute_codec_string=${ep_codec_string}, i even tried using it
> like: absolute_codec_string='${ep_codec_string}'
> or absolute_codec_string=\'${ep_codec_string}\'
> or absolute_codec_string=\\'${ep_codec_string}\\'
> (as suggested by bkw_ on IRC)
>
> Console log shows like that:
>
> EXECUTE sofia/profile/1234567 at 1.2.2.3
> bridge({absolute_codec_string=PCMA at 8000h@20i at 64000b,PCMU at 8000h@20i at 64000b,G729 at 8000h@20i at 8000b,leg_time_out=
> 10,process_cdr=b_only}[SupplierPrefix=]sofia/fstest/456789 at 7.2.1.4)
>
>
> 2011-07-20 11:03:59.448455 [DEBUG] switch_ivr_originate.c:1873 Parsing
> global variables
> 2011-07-20 11:03:59.448455 [DEBUG] switch_event.c:1170 Parsing variable
> [absolute_codec_string]=[PCMA at 8000h@20i at 64000b]
> 2011-07-20 11:03:59.448455 [DEBUG] switch_event.c:1170 Parsing variable
> [leg_time_out]=[10]
> 2011-07-20 11:03:59.448455 [DEBUG] switch_event.c:1170 Parsing variable
> [process_cdr]=[b_only]
>
> Am i doing something wrong or should i open a JIRA?
>
>
> --
>
> Kind Regards
>
> Saeed Ahmed
>
> _______________________________________________
> 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 fabio.bigliardi at gmail.com  Wed Jul 20 14:55:52 2011
From: fabio.bigliardi at gmail.com (Fabio Bigliardi)
Date: Wed, 20 Jul 2011 12:55:52 +0200
Subject: [Freeswitch-users] API conference invite group
In-Reply-To: <CAKzWOxXv-K1riCO_DUk9JbEudVaWP6c9m5VuskaeqqzJ+dcE8g@mail.gmail.com>
References: <CABtP9QiLqZXUwGv9udZ7gpyHo=XLHT_DcpGMDnvmYPU7LUjf2Q@mail.gmail.com>
	<CAKzWOxXv-K1riCO_DUk9JbEudVaWP6c9m5VuskaeqqzJ+dcE8g@mail.gmail.com>
Message-ID: <CABtP9QhBOwZWBSrZHvimtVUCr4w4mVMbiZ0mO7J6MLA4u2F_rw@mail.gmail.com>

Hi Michael,

I've tried the command:

api expand conference ds_totale dial group_call(01@${domain_name})

But the output is:

Content-Type: api/response
Content-Length: 47

Call Requested: result: [CHAN_NOT_IMPLEMENTED]

Freeswitch log:
2011-07-20 12:28:49.304970 [ERR] switch_core_session.c:413 Could not locate
channel type group_call(01 at 192.168.25.61)
2011-07-20 12:28:49.304970 [NOTICE] switch_ivr_originate.c:2453 Cannot
create outgoing channel of type [group_call(01 at 192.168.25.61)] cause:
[CHAN_NOT_IMPLEMENTED]
2011-07-20 12:28:49.304970 [DEBUG] switch_ivr_originate.c:3318 Originate
Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
2011-07-20 12:28:49.304970 [ERR] mod_conference.c:5169 Cannot create
outgoing channel, cause: CHAN_NOT_IMPLEMENTED

I think that this is related to the fact that the group is created in sqlite
DB and not in the xml dialplan. So I've tried:

api expand conference ds_totale dial group call:'01@${domain_name}'

But the output is:

Content-Type: api/response
Content-Length: 47

Call Requested: result: [NO_ROUTE_DESTINATION]

Freeswitch log:
2011-07-20 12:36:49.964965 [DEBUG] switch_ivr_originate.c:1879 Parsing
global variables
2011-07-20 12:36:49.964965 [NOTICE] switch_ivr_originate.c:2453 Cannot
create outgoing channel of type [error] cause: [NO_ROUTE_DESTINATION]
2011-07-20 12:36:49.964965 [DEBUG] switch_ivr_originate.c:3318 Originate
Resulted in Error Cause: 3 [NO_ROUTE_DESTINATION]
2011-07-20 12:36:49.964965 [NOTICE] switch_ivr_originate.c:2453 Cannot
create outgoing channel of type [group] cause: [NO_ROUTE_DESTINATION]
2011-07-20 12:36:49.964965 [DEBUG] switch_ivr_originate.c:3318 Originate
Resulted in Error Cause: 3 [NO_ROUTE_DESTINATION]
2011-07-20 12:36:49.964965 [ERR] mod_conference.c:5169 Cannot create
outgoing channel, cause: NO_ROUTE_DESTINATION

It seems a problem in parsing; in fact the command:

api expand group call:'01@${domain_name}'

gives the following output:

Content-Type: api/response
Content-Length: 85

sofia/internal/sip:1001 at 192.168.25.61:5070,sofia/internal/
sip:1016 at 192.168.25.31:5060

So I've tried:

api expand conference ds_totale dial sofia/internal/
sip:1001 at 192.168.25.61:5070,sofia/internal/sip:1016 at 192.168.25.31:5060

The output is:
Content-Type: api/response
Content-Length: 34

Call Requested: result: [SUCCESS]

But only 1016 is added to the conference while 1001 is hanged up with cause:
LOSE_RACE.

So the questions are:

1. What is wrong with the syntax of the following command?

 api expand conference ds_totale dial group call:'01@${domain_name}'

2. How can it be avoided that parties which "loose the race" are hanged up?


Thank you very much.

F. Bigliardi



2011/7/19 Michael Collins <msc at freeswitch.org>

>
>
> On Tue, Jul 19, 2011 at 5:47 AM, Fabio Bigliardi <
> fabio.bigliardi at gmail.com> wrote:
>
>> Hi all,
>> I would like to be able to invite a  group to a conference through an API
>> command.
>>
>> In xml dialplan this can be achieved through the following action:
>>
>> <action application="conference_set_auto_outcall"
>> data="${group_call(my_group}"/>
>>
>> In the API, I suppose the right command is "dial":
>>
>> api conference my_conf dial <group dial string>
>>
>> But how can I specify group dial string?
>>
>
> Try this:
>
> expand api conference my_conf dial group_call(my_group)
>
> -MC
>
>
>>
>> Thank you for your support.
>>
>> Best regards,
>>
>> Fabio Bigliardi
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 saeedahmad1981 at gmail.com  Wed Jul 20 14:57:46 2011
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Wed, 20 Jul 2011 12:57:46 +0200
Subject: [Freeswitch-users] absolute_codec_string just process one codec
In-Reply-To: <CAFiqYukBJ-BBFZAp99weP=UCxcqJBj9EV=Xrx2iSqtzM_0HCHw@mail.gmail.com>
References: <4E269BBF.1010805@gmail.com>
	<CAFiqYukBJ-BBFZAp99weP=UCxcqJBj9EV=Xrx2iSqtzM_0HCHw@mail.gmail.com>
Message-ID: <CAE9h+47DHayCo2uTjT9d7mnL3aC_gfQ1wq=bgQmm+jNzXRfFgQ@mail.gmail.com>

Thanks Steve for a quick reply.

now i understand the mystery behind :)

I'll try to find a way to replace , with : in  ${ep_codec_string}.

thanks for detailed explanation.

On Wed, Jul 20, 2011 at 12:16 PM, Steven Ayre <steveayre at gmail.com> wrote:

> You are... but it's perhaps not a very well documented feature so you
> can be forgiven...
>
> The problem is that , separates variable assignments in {} and []
> blocks. So what you're using is actually setting the following
> variables:
>     absolute_codec_string=PCMA at 8000h@20i at 64000b
>    PCMU at 8000h@20i at 64000b=
>    G729 at 8000h@20i at 8000b=
>    leg_time_out=10
>    process_cdr=b_only
> which hopefully makes it obvious why it's only using the first codec.
>
> It's a known problem (http://jira.freeswitch.org/browse/FS-2126) and
> there *is* a workaround for it. There's a special syntax that lets you
> use a character other than comma for separating the codec names. This
> actually applies to any variable values that are set in the {} and []
> blocks and the special character gets replaced by a comma during the
> variable assignment.
>
> The syntax is to put a prefix of ^^: at the start of the value and
> replace the commas in the value with :
> You can use any (almost) character you like though, it uses the one
> after the initial ^^.
>
> As a short example: [absolute_codec_string=^^:PCMU:PCMA]
>
> For your example you'll want to pass in:
> {absolute_codec_string=^^:PCMA at 8000h@20i at 64000b:PCMU at 8000h@20i at 64000b
> :G729 at 8000h@20i at 8000b,leg_time_out=10,process_cdr=b_only}
>
> Since you appear to be getting that from a variable you'll probably
> want to find some way of substituting the , for the :
>
> -Steve
>
>
>
> On 20 July 2011 10:11, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> > Hi,
> >
> > If i set absoluteye_codec_string in dialplan with multiple codecs, like
> > G729,PCMA,PCMU, then just first codec will be processed, rest will be
> > ignored.
> >
> > Example: {absolute_codec_string=${ep_codec_string}, i even tried using it
> > like: absolute_codec_string='${ep_codec_string}'
> > or absolute_codec_string=\'${ep_codec_string}\'
> > or absolute_codec_string=\\'${ep_codec_string}\\'
> > (as suggested by bkw_ on IRC)
> >
> > Console log shows like that:
> >
> > EXECUTE sofia/profile/1234567 at 1.2.2.3
> > bridge({absolute_codec_string=PCMA at 8000h@20i at 64000b,PCMU at 8000h
> @20i at 64000b,G729 at 8000h@20i at 8000b,leg_time_out=
> > 10,process_cdr=b_only}[SupplierPrefix=]sofia/fstest/456789 at 7.2.1.4)
> >
> >
> > 2011-07-20 11:03:59.448455 [DEBUG] switch_ivr_originate.c:1873 Parsing
> > global variables
> > 2011-07-20 11:03:59.448455 [DEBUG] switch_event.c:1170 Parsing variable
> > [absolute_codec_string]=[PCMA at 8000h@20i at 64000b]
> > 2011-07-20 11:03:59.448455 [DEBUG] switch_event.c:1170 Parsing variable
> > [leg_time_out]=[10]
> > 2011-07-20 11:03:59.448455 [DEBUG] switch_event.c:1170 Parsing variable
> > [process_cdr]=[b_only]
> >
> > Am i doing something wrong or should i open a JIRA?
> >
> >
> > --
> >
> > Kind Regards
> >
> > Saeed Ahmed
> >
> > _______________________________________________
> > 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
> >
> >
>
> _______________________________________________
> 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 steveayre at gmail.com  Wed Jul 20 16:03:20 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 20 Jul 2011 13:03:20 +0100
Subject: [Freeswitch-users] API conference invite group
In-Reply-To: <CABtP9QhBOwZWBSrZHvimtVUCr4w4mVMbiZ0mO7J6MLA4u2F_rw@mail.gmail.com>
References: <CABtP9QiLqZXUwGv9udZ7gpyHo=XLHT_DcpGMDnvmYPU7LUjf2Q@mail.gmail.com>
	<CAKzWOxXv-K1riCO_DUk9JbEudVaWP6c9m5VuskaeqqzJ+dcE8g@mail.gmail.com>
	<CABtP9QhBOwZWBSrZHvimtVUCr4w4mVMbiZ0mO7J6MLA4u2F_rw@mail.gmail.com>
Message-ID: <CAFiqYuk4Za30-jLmj5QKiWMtkyoUCeZ203T8rTFkMNOt2P3eyg@mail.gmail.com>

I think you want to be dialing using the group/ prefix:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#group

Try: api expand conference ds_totale dial group/01@${domain_name}

All dialstrings start with chantype/ which identifies which endpoint
module to use to dial. CHAN_NOT_IMPLEMENTED is the error given if that
part was not recognised.

Since it looks like you might be trying to connect using ESL directly,
look in libs/esl in your git checkout. You'll find esl libraries there
in a range of languages that'll handle the protocol for you.

-Steve





On 20 July 2011 11:55, Fabio Bigliardi <fabio.bigliardi at gmail.com> wrote:
> Hi Michael,
>
> I've tried the command:
>
> api expand conference ds_totale dial group_call(01@${domain_name})
>
> But the output is:
>
> Content-Type: api/response
> Content-Length: 47
>
> Call Requested: result: [CHAN_NOT_IMPLEMENTED]
>
> Freeswitch log:
> 2011-07-20 12:28:49.304970 [ERR] switch_core_session.c:413 Could not locate
> channel type group_call(01 at 192.168.25.61)
> 2011-07-20 12:28:49.304970 [NOTICE] switch_ivr_originate.c:2453 Cannot
> create outgoing channel of type [group_call(01 at 192.168.25.61)] cause:
> [CHAN_NOT_IMPLEMENTED]
> 2011-07-20 12:28:49.304970 [DEBUG] switch_ivr_originate.c:3318 Originate
> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
> 2011-07-20 12:28:49.304970 [ERR] mod_conference.c:5169 Cannot create
> outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>
> I think that this is related to the fact that the group is created in sqlite
> DB and not in the xml dialplan. So I've tried:
>
> api expand conference ds_totale dial group call:'01@${domain_name}'
>
> But the output is:
>
> Content-Type: api/response
> Content-Length: 47
>
> Call Requested: result: [NO_ROUTE_DESTINATION]
>
> Freeswitch log:
> 2011-07-20 12:36:49.964965 [DEBUG] switch_ivr_originate.c:1879 Parsing
> global variables
> 2011-07-20 12:36:49.964965 [NOTICE] switch_ivr_originate.c:2453 Cannot
> create outgoing channel of type [error] cause: [NO_ROUTE_DESTINATION]
> 2011-07-20 12:36:49.964965 [DEBUG] switch_ivr_originate.c:3318 Originate
> Resulted in Error Cause: 3 [NO_ROUTE_DESTINATION]
> 2011-07-20 12:36:49.964965 [NOTICE] switch_ivr_originate.c:2453 Cannot
> create outgoing channel of type [group] cause: [NO_ROUTE_DESTINATION]
> 2011-07-20 12:36:49.964965 [DEBUG] switch_ivr_originate.c:3318 Originate
> Resulted in Error Cause: 3 [NO_ROUTE_DESTINATION]
> 2011-07-20 12:36:49.964965 [ERR] mod_conference.c:5169 Cannot create
> outgoing channel, cause: NO_ROUTE_DESTINATION
>
> It seems a problem in parsing; in fact the command:
>
> api expand group call:'01@${domain_name}'
>
> gives the following output:
>
> Content-Type: api/response
> Content-Length: 85
>
> sofia/internal/sip:1001 at 192.168.25.61:5070,sofia/internal/sip:1016 at 192.168.25.31:5060
>
> So I've tried:
>
> api expand conference ds_totale dial
> sofia/internal/sip:1001 at 192.168.25.61:5070,sofia/internal/sip:1016 at 192.168.25.31:5060
>
> The output is:
> Content-Type: api/response
> Content-Length: 34
>
> Call Requested: result: [SUCCESS]
>
> But only 1016 is added to the conference while 1001 is hanged up with cause:
> LOSE_RACE.
>
> So the questions are:
>
> 1. What is wrong with the syntax of the following command?
>
> ?api expand conference ds_totale dial group call:'01@${domain_name}'
>
> 2. How can it be avoided that parties which "loose the race" are hanged up?
>
>
> Thank you very much.
>
> F. Bigliardi
>
>
>
> 2011/7/19 Michael Collins <msc at freeswitch.org>
>>
>>
>> On Tue, Jul 19, 2011 at 5:47 AM, Fabio Bigliardi
>> <fabio.bigliardi at gmail.com> wrote:
>>>
>>> Hi all,
>>> I would like to be able to invite a? group to a conference through an API
>>> command.
>>>
>>> In xml dialplan this can be achieved through the following action:
>>>
>>> <action application="conference_set_auto_outcall"
>>> data="${group_call(my_group}"/>
>>>
>>> In the API, I suppose the right command is "dial":
>>>
>>> api conference my_conf dial <group dial string>
>>>
>>> But how can I specify group dial string?
>>
>> Try this:
>> expand api conference my_conf dial group_call(my_group)
>> -MC
>>
>>>
>>> Thank you for your support.
>>>
>>> Best regards,
>>>
>>> Fabio Bigliardi
>>>
>>> _______________________________________________
>>> 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
>>>
>>
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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 david.villasmil.work at gmail.com  Wed Jul 20 16:30:41 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 20 Jul 2011 14:30:41 +0200
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <CAGKOgbK=JzEpWmR8=nFq_x47mFFy96nWRRkMZrogY0hxOMnr2g@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
	<163DDA0D-0B14-4623-B48B-5E9EEE907AED@gmail.com>
	<CAFYZzbu9Nh3Ojb26G_pAiFn9MVmm59qeO8BL02_WLX5Fj0mfUg@mail.gmail.com>
	<CAKzWOxX2CrDESPMOWMtCgSDxt1R2S3=70aDsTrRM2_p6isfkvA@mail.gmail.com>
	<CAGKOgbK=JzEpWmR8=nFq_x47mFFy96nWRRkMZrogY0hxOMnr2g@mail.gmail.com>
Message-ID: <CAFGRPVoPVa8ir-18cBLYDzRNftO96Jd=uJ7-xiZQUFsikUwBkA@mail.gmail.com>

Hello guys,

I finally set up a git at github. I would like to have a couple testers,
anyone?

Please bear in mind:

- There's NO installation script. But it is all scripts and web pages, etc.
- The web interface is in spanish, i still need to translate that.
- The web design is by no means "cool", it is just functional.
- There will for sure be a thousand bugs, which will need to be fixed.

Contact me whoever's interested.


Thanks all

David

On Wed, Jul 6, 2011 at 3:18 PM, Umair Bari <umairbari at gmail.com> wrote:

> Signup at: https://github.com/signup/free
>
> Create a new repository: https://github.com/repositories/new
> Help: http://help.github.com/create-a-repo/
>
> A nice help with images:
> Windows: http://help.github.com/win-set-up-git/
> Mac: http://help.github.com/mac-set-up-git/
> Linux: http://help.github.com/linux-set-up-git/
>
> On Tue, Jul 5, 2011 at 11:21 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> FYI, Ray is out this week, so for now you might want to throw it up on
>> github for now.
>> -MC
>>
>>
>> On Mon, Jul 4, 2011 at 3:59 PM, Michel Daggelinckx <
>> michel.daggelinckx at gmail.com> wrote:
>>
>>> Ask intralanman for acces to the FS contrib
>>>
>>> On Mon, Jul 4, 2011 at 7:10 PM, Mohammad Emran <monemran at gmail.com>wrote:
>>>
>>>> We can put on google code or git hub.
>>>>
>>>> Sent from my iPad
>>>>
>>>> On 4 Jul 2011, at 19:12, David Villasmil <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>> Hello All,
>>>>
>>>> I think I'm as ready as i can be to publish this...
>>>> Can someone guide me into publishing via GIT?
>>>>
>>>> thanks
>>>>
>>>> David
>>>>
>>>> On Thu, Feb 24, 2011 at 12:39 PM, David Villasmil <<david.villasmil.work at gmail.com>
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>>> Hello Guys,
>>>>>
>>>>> I'm finishing a "complete" wholesale application created on freeswitch
>>>>> and I was wondering whether it would be a good idea to put it up on the
>>>>> wiki. I just don't know how.
>>>>>
>>>>> Features include all the following parameters configurable via web
>>>>> interface:
>>>>>
>>>>> - Profile creation based on server IP where traffic is received. You
>>>>> can have multiple IPs, system will create multiple profiles/diaplans so it
>>>>> can differentiate.
>>>>>      - i.e. offer to the same customer a "gold" routing on IP1 and
>>>>> cheap routing on IP2
>>>>>
>>>>> - Customer add/modify/delete
>>>>>      - IP source
>>>>>      - Rates for client routes based on areacode
>>>>>      - Prepaid or postpaid.
>>>>>           - When cutomer balance is 0, no more calls are allowed.
>>>>>      - limit max channels
>>>>>      - Media by-pass
>>>>>           - When by-passed, customer and provider will exchange RTPs
>>>>> directly. Else, server will be in the middle.
>>>>>
>>>>> - Provider add/modify/delete
>>>>>      - costs for provider routes based on areacode
>>>>>      - limit max channels
>>>>>
>>>>> - Routing based on areacode, gives great granularity.
>>>>>
>>>>> - Routes can be assigned multiple gateways/providers which can in turn
>>>>> be distributed based on weigth. Includes overflow to next configured GW.
>>>>>
>>>>> - Basic financial report generation (totals) by customer/provider
>>>>>
>>>>> - Basic traffic ASR/ACD report (totals) by cutomer/provider
>>>>>
>>>>> - Basic user administration. (No access level, only total access)
>>>>>
>>>>> - CDR export to csv file.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> I also have a prepaid card app... no web interface on that one
>>>>> though...
>>>>>
>>>>> Thanks all
>>>>>
>>>>>
>>>>> David
>>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
>
> --
>
> Thanks & Regards,
>
> Umair Bari
>
> _______________________________________________
> 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 vipkilla at gmail.com  Wed Jul 20 16:44:57 2011
From: vipkilla at gmail.com (vip killa)
Date: Wed, 20 Jul 2011 08:44:57 -0400
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <CAFGRPVoPVa8ir-18cBLYDzRNftO96Jd=uJ7-xiZQUFsikUwBkA@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
	<163DDA0D-0B14-4623-B48B-5E9EEE907AED@gmail.com>
	<CAFYZzbu9Nh3Ojb26G_pAiFn9MVmm59qeO8BL02_WLX5Fj0mfUg@mail.gmail.com>
	<CAKzWOxX2CrDESPMOWMtCgSDxt1R2S3=70aDsTrRM2_p6isfkvA@mail.gmail.com>
	<CAGKOgbK=JzEpWmR8=nFq_x47mFFy96nWRRkMZrogY0hxOMnr2g@mail.gmail.com>
	<CAFGRPVoPVa8ir-18cBLYDzRNftO96Jd=uJ7-xiZQUFsikUwBkA@mail.gmail.com>
Message-ID: <CAC-LwPNmqm5xcEV83Bu8ROvT3z2YwUxk9p52OG=E5sM_h3uLZQ@mail.gmail.com>

can you provide a link to the github?

On Wed, Jul 20, 2011 at 8:30 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello guys,
>
> I finally set up a git at github. I would like to have a couple testers,
> anyone?
>
> Please bear in mind:
>
> - There's NO installation script. But it is all scripts and web pages, etc.
> - The web interface is in spanish, i still need to translate that.
> - The web design is by no means "cool", it is just functional.
> - There will for sure be a thousand bugs, which will need to be fixed.
>
> Contact me whoever's interested.
>
>
> Thanks all
>
> David
>
>
> On Wed, Jul 6, 2011 at 3:18 PM, Umair Bari <umairbari at gmail.com> wrote:
>
>> Signup at: https://github.com/signup/free
>>
>> Create a new repository: https://github.com/repositories/new
>> Help: http://help.github.com/create-a-repo/
>>
>> A nice help with images:
>> Windows: http://help.github.com/win-set-up-git/
>> Mac: http://help.github.com/mac-set-up-git/
>> Linux: http://help.github.com/linux-set-up-git/
>>
>> On Tue, Jul 5, 2011 at 11:21 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> FYI, Ray is out this week, so for now you might want to throw it up on
>>> github for now.
>>> -MC
>>>
>>>
>>> On Mon, Jul 4, 2011 at 3:59 PM, Michel Daggelinckx <
>>> michel.daggelinckx at gmail.com> wrote:
>>>
>>>> Ask intralanman for acces to the FS contrib
>>>>
>>>> On Mon, Jul 4, 2011 at 7:10 PM, Mohammad Emran <monemran at gmail.com>wrote:
>>>>
>>>>> We can put on google code or git hub.
>>>>>
>>>>> Sent from my iPad
>>>>>
>>>>> On 4 Jul 2011, at 19:12, David Villasmil <
>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>
>>>>> Hello All,
>>>>>
>>>>> I think I'm as ready as i can be to publish this...
>>>>> Can someone guide me into publishing via GIT?
>>>>>
>>>>> thanks
>>>>>
>>>>> David
>>>>>
>>>>> On Thu, Feb 24, 2011 at 12:39 PM, David Villasmil <<david.villasmil.work at gmail.com>
>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>
>>>>>> Hello Guys,
>>>>>>
>>>>>> I'm finishing a "complete" wholesale application created on freeswitch
>>>>>> and I was wondering whether it would be a good idea to put it up on the
>>>>>> wiki. I just don't know how.
>>>>>>
>>>>>> Features include all the following parameters configurable via web
>>>>>> interface:
>>>>>>
>>>>>> - Profile creation based on server IP where traffic is received. You
>>>>>> can have multiple IPs, system will create multiple profiles/diaplans so it
>>>>>> can differentiate.
>>>>>>      - i.e. offer to the same customer a "gold" routing on IP1 and
>>>>>> cheap routing on IP2
>>>>>>
>>>>>> - Customer add/modify/delete
>>>>>>      - IP source
>>>>>>      - Rates for client routes based on areacode
>>>>>>      - Prepaid or postpaid.
>>>>>>           - When cutomer balance is 0, no more calls are allowed.
>>>>>>      - limit max channels
>>>>>>      - Media by-pass
>>>>>>           - When by-passed, customer and provider will exchange RTPs
>>>>>> directly. Else, server will be in the middle.
>>>>>>
>>>>>> - Provider add/modify/delete
>>>>>>      - costs for provider routes based on areacode
>>>>>>      - limit max channels
>>>>>>
>>>>>> - Routing based on areacode, gives great granularity.
>>>>>>
>>>>>> - Routes can be assigned multiple gateways/providers which can in turn
>>>>>> be distributed based on weigth. Includes overflow to next configured GW.
>>>>>>
>>>>>> - Basic financial report generation (totals) by customer/provider
>>>>>>
>>>>>> - Basic traffic ASR/ACD report (totals) by cutomer/provider
>>>>>>
>>>>>> - Basic user administration. (No access level, only total access)
>>>>>>
>>>>>> - CDR export to csv file.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> I also have a prepaid card app... no web interface on that one
>>>>>> though...
>>>>>>
>>>>>> Thanks all
>>>>>>
>>>>>>
>>>>>> David
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>>
>> --
>>
>> Thanks & Regards,
>>
>> Umair Bari
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 fabio.bigliardi at gmail.com  Wed Jul 20 16:45:13 2011
From: fabio.bigliardi at gmail.com (Fabio Bigliardi)
Date: Wed, 20 Jul 2011 14:45:13 +0200
Subject: [Freeswitch-users] API conference invite group
In-Reply-To: <CAFiqYuk4Za30-jLmj5QKiWMtkyoUCeZ203T8rTFkMNOt2P3eyg@mail.gmail.com>
References: <CABtP9QiLqZXUwGv9udZ7gpyHo=XLHT_DcpGMDnvmYPU7LUjf2Q@mail.gmail.com>
	<CAKzWOxXv-K1riCO_DUk9JbEudVaWP6c9m5VuskaeqqzJ+dcE8g@mail.gmail.com>
	<CABtP9QhBOwZWBSrZHvimtVUCr4w4mVMbiZ0mO7J6MLA4u2F_rw@mail.gmail.com>
	<CAFiqYuk4Za30-jLmj5QKiWMtkyoUCeZ203T8rTFkMNOt2P3eyg@mail.gmail.com>
Message-ID: <CABtP9QjXwDoia-64TtuGavSCdAmsHNgzqdiZ7PvcEXp=D-9+OA@mail.gmail.com>

It works only with groups defined in XML directory and not with groups in
sqlite DB.
Only one member of the group is answered; the others are hanged up with
cause: LOSE_RACE.

-Fabio


2011/7/20 Steven Ayre <steveayre at gmail.com>

> I think you want to be dialing using the group/ prefix:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#group
>
> Try: api expand conference ds_totale dial group/01@${domain_name}
>
> All dialstrings start with chantype/ which identifies which endpoint
> module to use to dial. CHAN_NOT_IMPLEMENTED is the error given if that
> part was not recognised.
>
> Since it looks like you might be trying to connect using ESL directly,
> look in libs/esl in your git checkout. You'll find esl libraries there
> in a range of languages that'll handle the protocol for you.
>
> -Steve
>
>
>
>
>
> On 20 July 2011 11:55, Fabio Bigliardi <fabio.bigliardi at gmail.com> wrote:
> > Hi Michael,
> >
> > I've tried the command:
> >
> > api expand conference ds_totale dial group_call(01@${domain_name})
> >
> > But the output is:
> >
> > Content-Type: api/response
> > Content-Length: 47
> >
> > Call Requested: result: [CHAN_NOT_IMPLEMENTED]
> >
> > Freeswitch log:
> > 2011-07-20 12:28:49.304970 [ERR] switch_core_session.c:413 Could not
> locate
> > channel type group_call(01 at 192.168.25.61)
> > 2011-07-20 12:28:49.304970 [NOTICE] switch_ivr_originate.c:2453 Cannot
> > create outgoing channel of type [group_call(01 at 192.168.25.61)] cause:
> > [CHAN_NOT_IMPLEMENTED]
> > 2011-07-20 12:28:49.304970 [DEBUG] switch_ivr_originate.c:3318 Originate
> > Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
> > 2011-07-20 12:28:49.304970 [ERR] mod_conference.c:5169 Cannot create
> > outgoing channel, cause: CHAN_NOT_IMPLEMENTED
> >
> > I think that this is related to the fact that the group is created in
> sqlite
> > DB and not in the xml dialplan. So I've tried:
> >
> > api expand conference ds_totale dial group call:'01@${domain_name}'
> >
> > But the output is:
> >
> > Content-Type: api/response
> > Content-Length: 47
> >
> > Call Requested: result: [NO_ROUTE_DESTINATION]
> >
> > Freeswitch log:
> > 2011-07-20 12:36:49.964965 [DEBUG] switch_ivr_originate.c:1879 Parsing
> > global variables
> > 2011-07-20 12:36:49.964965 [NOTICE] switch_ivr_originate.c:2453 Cannot
> > create outgoing channel of type [error] cause: [NO_ROUTE_DESTINATION]
> > 2011-07-20 12:36:49.964965 [DEBUG] switch_ivr_originate.c:3318 Originate
> > Resulted in Error Cause: 3 [NO_ROUTE_DESTINATION]
> > 2011-07-20 12:36:49.964965 [NOTICE] switch_ivr_originate.c:2453 Cannot
> > create outgoing channel of type [group] cause: [NO_ROUTE_DESTINATION]
> > 2011-07-20 12:36:49.964965 [DEBUG] switch_ivr_originate.c:3318 Originate
> > Resulted in Error Cause: 3 [NO_ROUTE_DESTINATION]
> > 2011-07-20 12:36:49.964965 [ERR] mod_conference.c:5169 Cannot create
> > outgoing channel, cause: NO_ROUTE_DESTINATION
> >
> > It seems a problem in parsing; in fact the command:
> >
> > api expand group call:'01@${domain_name}'
> >
> > gives the following output:
> >
> > Content-Type: api/response
> > Content-Length: 85
> >
> > sofia/internal/sip:1001 at 192.168.25.61:5070,sofia/internal/
> sip:1016 at 192.168.25.31:5060
> >
> > So I've tried:
> >
> > api expand conference ds_totale dial
> > sofia/internal/sip:1001 at 192.168.25.61:5070,sofia/internal/
> sip:1016 at 192.168.25.31:5060
> >
> > The output is:
> > Content-Type: api/response
> > Content-Length: 34
> >
> > Call Requested: result: [SUCCESS]
> >
> > But only 1016 is added to the conference while 1001 is hanged up with
> cause:
> > LOSE_RACE.
> >
> > So the questions are:
> >
> > 1. What is wrong with the syntax of the following command?
> >
> >  api expand conference ds_totale dial group call:'01@${domain_name}'
> >
> > 2. How can it be avoided that parties which "loose the race" are hanged
> up?
> >
> >
> > Thank you very much.
> >
> > F. Bigliardi
> >
> >
> >
> > 2011/7/19 Michael Collins <msc at freeswitch.org>
> >>
> >>
> >> On Tue, Jul 19, 2011 at 5:47 AM, Fabio Bigliardi
> >> <fabio.bigliardi at gmail.com> wrote:
> >>>
> >>> Hi all,
> >>> I would like to be able to invite a  group to a conference through an
> API
> >>> command.
> >>>
> >>> In xml dialplan this can be achieved through the following action:
> >>>
> >>> <action application="conference_set_auto_outcall"
> >>> data="${group_call(my_group}"/>
> >>>
> >>> In the API, I suppose the right command is "dial":
> >>>
> >>> api conference my_conf dial <group dial string>
> >>>
> >>> But how can I specify group dial string?
> >>
> >> Try this:
> >> expand api conference my_conf dial group_call(my_group)
> >> -MC
> >>
> >>>
> >>> Thank you for your support.
> >>>
> >>> Best regards,
> >>>
> >>> Fabio Bigliardi
> >>>
> >>> _______________________________________________
> >>> 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
> >>>
> >>
> >>
> >> _______________________________________________
> >> 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
> >>
> >
> >
> > _______________________________________________
> > 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
> >
> >
>
> _______________________________________________
> 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 avi at avimarcus.net  Wed Jul 20 16:55:36 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 20 Jul 2011 15:55:36 +0300
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <CAC-LwPNmqm5xcEV83Bu8ROvT3z2YwUxk9p52OG=E5sM_h3uLZQ@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
	<163DDA0D-0B14-4623-B48B-5E9EEE907AED@gmail.com>
	<CAFYZzbu9Nh3Ojb26G_pAiFn9MVmm59qeO8BL02_WLX5Fj0mfUg@mail.gmail.com>
	<CAKzWOxX2CrDESPMOWMtCgSDxt1R2S3=70aDsTrRM2_p6isfkvA@mail.gmail.com>
	<CAGKOgbK=JzEpWmR8=nFq_x47mFFy96nWRRkMZrogY0hxOMnr2g@mail.gmail.com>
	<CAFGRPVoPVa8ir-18cBLYDzRNftO96Jd=uJ7-xiZQUFsikUwBkA@mail.gmail.com>
	<CAC-LwPNmqm5xcEV83Bu8ROvT3z2YwUxk9p52OG=E5sM_h3uLZQ@mail.gmail.com>
Message-ID: <CALNZuwkR-=wrwVg6GKjuaD31UmdEuS5iEr1=pyTyQqWMsc7SCA@mail.gmail.com>

It was.. deleted? off the wiki:
https://github.com/davidcsi/FreeSWITCH-Billing

-Avi


On Wed, Jul 20, 2011 at 3:44 PM, vip killa <vipkilla at gmail.com> wrote:

> can you provide a link to the github?
>
>
> On Wed, Jul 20, 2011 at 8:30 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello guys,
>>
>> I finally set up a git at github. I would like to have a couple testers,
>> anyone?
>>
>> Please bear in mind:
>>
>> - There's NO installation script. But it is all scripts and web pages,
>> etc.
>> - The web interface is in spanish, i still need to translate that.
>> - The web design is by no means "cool", it is just functional.
>> - There will for sure be a thousand bugs, which will need to be fixed.
>>
>> Contact me whoever's interested.
>>
>>
>> Thanks all
>>
>> David
>>
>>
>> On Wed, Jul 6, 2011 at 3:18 PM, Umair Bari <umairbari at gmail.com> wrote:
>>
>>> Signup at: https://github.com/signup/free
>>>
>>> Create a new repository: https://github.com/repositories/new
>>> Help: http://help.github.com/create-a-repo/
>>>
>>> A nice help with images:
>>> Windows: http://help.github.com/win-set-up-git/
>>> Mac: http://help.github.com/mac-set-up-git/
>>> Linux: http://help.github.com/linux-set-up-git/
>>>
>>> On Tue, Jul 5, 2011 at 11:21 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> FYI, Ray is out this week, so for now you might want to throw it up on
>>>> github for now.
>>>> -MC
>>>>
>>>>
>>>> On Mon, Jul 4, 2011 at 3:59 PM, Michel Daggelinckx <
>>>> michel.daggelinckx at gmail.com> wrote:
>>>>
>>>>> Ask intralanman for acces to the FS contrib
>>>>>
>>>>> On Mon, Jul 4, 2011 at 7:10 PM, Mohammad Emran <monemran at gmail.com>wrote:
>>>>>
>>>>>> We can put on google code or git hub.
>>>>>>
>>>>>> Sent from my iPad
>>>>>>
>>>>>> On 4 Jul 2011, at 19:12, David Villasmil <
>>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>>
>>>>>> Hello All,
>>>>>>
>>>>>> I think I'm as ready as i can be to publish this...
>>>>>> Can someone guide me into publishing via GIT?
>>>>>>
>>>>>> thanks
>>>>>>
>>>>>> David
>>>>>>
>>>>>> On Thu, Feb 24, 2011 at 12:39 PM, David Villasmil <<david.villasmil.work at gmail.com>
>>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>>
>>>>>>> Hello Guys,
>>>>>>>
>>>>>>> I'm finishing a "complete" wholesale application created on
>>>>>>> freeswitch and I was wondering whether it would be a good idea to put it up
>>>>>>> on the wiki. I just don't know how.
>>>>>>>
>>>>>>> Features include all the following parameters configurable via web
>>>>>>> interface:
>>>>>>>
>>>>>>> - Profile creation based on server IP where traffic is received. You
>>>>>>> can have multiple IPs, system will create multiple profiles/diaplans so it
>>>>>>> can differentiate.
>>>>>>>      - i.e. offer to the same customer a "gold" routing on IP1 and
>>>>>>> cheap routing on IP2
>>>>>>>
>>>>>>> - Customer add/modify/delete
>>>>>>>      - IP source
>>>>>>>      - Rates for client routes based on areacode
>>>>>>>      - Prepaid or postpaid.
>>>>>>>           - When cutomer balance is 0, no more calls are allowed.
>>>>>>>      - limit max channels
>>>>>>>      - Media by-pass
>>>>>>>           - When by-passed, customer and provider will exchange RTPs
>>>>>>> directly. Else, server will be in the middle.
>>>>>>>
>>>>>>> - Provider add/modify/delete
>>>>>>>      - costs for provider routes based on areacode
>>>>>>>      - limit max channels
>>>>>>>
>>>>>>> - Routing based on areacode, gives great granularity.
>>>>>>>
>>>>>>> - Routes can be assigned multiple gateways/providers which can in
>>>>>>> turn be distributed based on weigth. Includes overflow to next configured
>>>>>>> GW.
>>>>>>>
>>>>>>> - Basic financial report generation (totals) by customer/provider
>>>>>>>
>>>>>>> - Basic traffic ASR/ACD report (totals) by cutomer/provider
>>>>>>>
>>>>>>> - Basic user administration. (No access level, only total access)
>>>>>>>
>>>>>>> - CDR export to csv file.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> I also have a prepaid card app... no web interface on that one
>>>>>>> though...
>>>>>>>
>>>>>>> Thanks all
>>>>>>>
>>>>>>>
>>>>>>> David
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>>
>>> --
>>>
>>> Thanks & Regards,
>>>
>>> Umair Bari
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 david.villasmil.work at gmail.com  Wed Jul 20 17:26:09 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 20 Jul 2011 15:26:09 +0200
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <CALNZuwkR-=wrwVg6GKjuaD31UmdEuS5iEr1=pyTyQqWMsc7SCA@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
	<163DDA0D-0B14-4623-B48B-5E9EEE907AED@gmail.com>
	<CAFYZzbu9Nh3Ojb26G_pAiFn9MVmm59qeO8BL02_WLX5Fj0mfUg@mail.gmail.com>
	<CAKzWOxX2CrDESPMOWMtCgSDxt1R2S3=70aDsTrRM2_p6isfkvA@mail.gmail.com>
	<CAGKOgbK=JzEpWmR8=nFq_x47mFFy96nWRRkMZrogY0hxOMnr2g@mail.gmail.com>
	<CAFGRPVoPVa8ir-18cBLYDzRNftO96Jd=uJ7-xiZQUFsikUwBkA@mail.gmail.com>
	<CAC-LwPNmqm5xcEV83Bu8ROvT3z2YwUxk9p52OG=E5sM_h3uLZQ@mail.gmail.com>
	<CALNZuwkR-=wrwVg6GKjuaD31UmdEuS5iEr1=pyTyQqWMsc7SCA@mail.gmail.com>
Message-ID: <CAFGRPVpEGht-AnoSaiXeFyMvpuC1L8nrZdCAj=LQORcUAywPCg@mail.gmail.com>

Yep, that's it. I removed it  because i first want a few people to test it.

Go ahead guys! let's see what happens

David

On Wed, Jul 20, 2011 at 2:55 PM, Avi Marcus <avi at avimarcus.net> wrote:

> It was.. deleted? off the wiki:
> https://github.com/davidcsi/FreeSWITCH-Billing
>
> -Avi
>
>
> On Wed, Jul 20, 2011 at 3:44 PM, vip killa <vipkilla at gmail.com> wrote:
>
>> can you provide a link to the github?
>>
>>
>> On Wed, Jul 20, 2011 at 8:30 AM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Hello guys,
>>>
>>> I finally set up a git at github. I would like to have a couple testers,
>>> anyone?
>>>
>>> Please bear in mind:
>>>
>>> - There's NO installation script. But it is all scripts and web pages,
>>> etc.
>>> - The web interface is in spanish, i still need to translate that.
>>> - The web design is by no means "cool", it is just functional.
>>> - There will for sure be a thousand bugs, which will need to be fixed.
>>>
>>> Contact me whoever's interested.
>>>
>>>
>>> Thanks all
>>>
>>> David
>>>
>>>
>>> On Wed, Jul 6, 2011 at 3:18 PM, Umair Bari <umairbari at gmail.com> wrote:
>>>
>>>> Signup at: https://github.com/signup/free
>>>>
>>>> Create a new repository: https://github.com/repositories/new
>>>> Help: http://help.github.com/create-a-repo/
>>>>
>>>> A nice help with images:
>>>> Windows: http://help.github.com/win-set-up-git/
>>>> Mac: http://help.github.com/mac-set-up-git/
>>>> Linux: http://help.github.com/linux-set-up-git/
>>>>
>>>> On Tue, Jul 5, 2011 at 11:21 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> FYI, Ray is out this week, so for now you might want to throw it up on
>>>>> github for now.
>>>>> -MC
>>>>>
>>>>>
>>>>> On Mon, Jul 4, 2011 at 3:59 PM, Michel Daggelinckx <
>>>>> michel.daggelinckx at gmail.com> wrote:
>>>>>
>>>>>> Ask intralanman for acces to the FS contrib
>>>>>>
>>>>>> On Mon, Jul 4, 2011 at 7:10 PM, Mohammad Emran <monemran at gmail.com>wrote:
>>>>>>
>>>>>>> We can put on google code or git hub.
>>>>>>>
>>>>>>> Sent from my iPad
>>>>>>>
>>>>>>> On 4 Jul 2011, at 19:12, David Villasmil <
>>>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>>>
>>>>>>> Hello All,
>>>>>>>
>>>>>>> I think I'm as ready as i can be to publish this...
>>>>>>> Can someone guide me into publishing via GIT?
>>>>>>>
>>>>>>> thanks
>>>>>>>
>>>>>>> David
>>>>>>>
>>>>>>> On Thu, Feb 24, 2011 at 12:39 PM, David Villasmil <<david.villasmil.work at gmail.com>
>>>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>>>
>>>>>>>> Hello Guys,
>>>>>>>>
>>>>>>>> I'm finishing a "complete" wholesale application created on
>>>>>>>> freeswitch and I was wondering whether it would be a good idea to put it up
>>>>>>>> on the wiki. I just don't know how.
>>>>>>>>
>>>>>>>> Features include all the following parameters configurable via web
>>>>>>>> interface:
>>>>>>>>
>>>>>>>> - Profile creation based on server IP where traffic is received. You
>>>>>>>> can have multiple IPs, system will create multiple profiles/diaplans so it
>>>>>>>> can differentiate.
>>>>>>>>      - i.e. offer to the same customer a "gold" routing on IP1 and
>>>>>>>> cheap routing on IP2
>>>>>>>>
>>>>>>>> - Customer add/modify/delete
>>>>>>>>      - IP source
>>>>>>>>      - Rates for client routes based on areacode
>>>>>>>>      - Prepaid or postpaid.
>>>>>>>>           - When cutomer balance is 0, no more calls are allowed.
>>>>>>>>      - limit max channels
>>>>>>>>      - Media by-pass
>>>>>>>>           - When by-passed, customer and provider will exchange RTPs
>>>>>>>> directly. Else, server will be in the middle.
>>>>>>>>
>>>>>>>> - Provider add/modify/delete
>>>>>>>>      - costs for provider routes based on areacode
>>>>>>>>      - limit max channels
>>>>>>>>
>>>>>>>> - Routing based on areacode, gives great granularity.
>>>>>>>>
>>>>>>>> - Routes can be assigned multiple gateways/providers which can in
>>>>>>>> turn be distributed based on weigth. Includes overflow to next configured
>>>>>>>> GW.
>>>>>>>>
>>>>>>>> - Basic financial report generation (totals) by customer/provider
>>>>>>>>
>>>>>>>> - Basic traffic ASR/ACD report (totals) by cutomer/provider
>>>>>>>>
>>>>>>>> - Basic user administration. (No access level, only total access)
>>>>>>>>
>>>>>>>> - CDR export to csv file.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> I also have a prepaid card app... no web interface on that one
>>>>>>>> though...
>>>>>>>>
>>>>>>>> Thanks all
>>>>>>>>
>>>>>>>>
>>>>>>>> David
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> 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
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> 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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> Thanks & Regards,
>>>>
>>>> Umair Bari
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 Nabble at slickdeals.endjunk.com  Wed Jul 20 17:40:22 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 20 Jul 2011 06:40:22 -0700 (PDT)
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <CAFGRPVpEGht-AnoSaiXeFyMvpuC1L8nrZdCAj=LQORcUAywPCg@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
	<163DDA0D-0B14-4623-B48B-5E9EEE907AED@gmail.com>
	<CAFYZzbu9Nh3Ojb26G_pAiFn9MVmm59qeO8BL02_WLX5Fj0mfUg@mail.gmail.com>
	<CAKzWOxX2CrDESPMOWMtCgSDxt1R2S3=70aDsTrRM2_p6isfkvA@mail.gmail.com>
	<CAGKOgbK=JzEpWmR8=nFq_x47mFFy96nWRRkMZrogY0hxOMnr2g@mail.gmail.com>
	<CAFGRPVoPVa8ir-18cBLYDzRNftO96Jd=uJ7-xiZQUFsikUwBkA@mail.gmail.com>
	<CAC-LwPNmqm5xcEV83Bu8ROvT3z2YwUxk9p52OG=E5sM_h3uLZQ@mail.gmail.com>
	<CALNZuwkR-=wrwVg6GKjuaD31UmdEuS5iEr1=pyTyQqWMsc7SCA@mail.gmail.com>
	<CAFGRPVpEGht-AnoSaiXeFyMvpuC1L8nrZdCAj=LQORcUAywPCg@mail.gmail.com>
Message-ID: <1311169222908-6602699.post@n2.nabble.com>


David Villasmil wrote:
> Go ahead guys! let's see what happens
> 
> David
For a small scale system, perhaps it is not bad to include an option to
support SQLite3. Just a thought.

-----
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/Complete-wholesale-app-in-freeswitch-tp6060011p6602699.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Wed Jul 20 17:45:26 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 20 Jul 2011 14:45:26 +0100
Subject: [Freeswitch-users] API conference invite group
In-Reply-To: <CABtP9QjXwDoia-64TtuGavSCdAmsHNgzqdiZ7PvcEXp=D-9+OA@mail.gmail.com>
References: <CABtP9QiLqZXUwGv9udZ7gpyHo=XLHT_DcpGMDnvmYPU7LUjf2Q@mail.gmail.com>
	<CAKzWOxXv-K1riCO_DUk9JbEudVaWP6c9m5VuskaeqqzJ+dcE8g@mail.gmail.com>
	<CABtP9QhBOwZWBSrZHvimtVUCr4w4mVMbiZ0mO7J6MLA4u2F_rw@mail.gmail.com>
	<CAFiqYuk4Za30-jLmj5QKiWMtkyoUCeZ203T8rTFkMNOt2P3eyg@mail.gmail.com>
	<CABtP9QjXwDoia-64TtuGavSCdAmsHNgzqdiZ7PvcEXp=D-9+OA@mail.gmail.com>
Message-ID: <CAFiqYukZ-jzg8NAsJgn-6ugOwXahbd3JG8ZpPrbH8=ggs1rvmw@mail.gmail.com>

> Only one member of the group is answered; the others are hanged up with
> cause: LOSE_RACE.

That's normal. An originate only ever gets answered by one party.

-Steve



On 20 July 2011 13:45, Fabio Bigliardi <fabio.bigliardi at gmail.com> wrote:
> It works only with groups defined in XML directory and not with groups in
> sqlite DB.
> Only one member of the group is answered; the others are hanged up with
> cause: LOSE_RACE.
>
> -Fabio
>
>
> 2011/7/20 Steven Ayre <steveayre at gmail.com>
>>
>> I think you want to be dialing using the group/ prefix:
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#group
>>
>> Try: api expand conference ds_totale dial group/01@${domain_name}
>>
>> All dialstrings start with chantype/ which identifies which endpoint
>> module to use to dial. CHAN_NOT_IMPLEMENTED is the error given if that
>> part was not recognised.
>>
>> Since it looks like you might be trying to connect using ESL directly,
>> look in libs/esl in your git checkout. You'll find esl libraries there
>> in a range of languages that'll handle the protocol for you.
>>
>> -Steve
>>
>>
>>
>>
>>
>> On 20 July 2011 11:55, Fabio Bigliardi <fabio.bigliardi at gmail.com> wrote:
>> > Hi Michael,
>> >
>> > I've tried the command:
>> >
>> > api expand conference ds_totale dial group_call(01@${domain_name})
>> >
>> > But the output is:
>> >
>> > Content-Type: api/response
>> > Content-Length: 47
>> >
>> > Call Requested: result: [CHAN_NOT_IMPLEMENTED]
>> >
>> > Freeswitch log:
>> > 2011-07-20 12:28:49.304970 [ERR] switch_core_session.c:413 Could not
>> > locate
>> > channel type group_call(01 at 192.168.25.61)
>> > 2011-07-20 12:28:49.304970 [NOTICE] switch_ivr_originate.c:2453 Cannot
>> > create outgoing channel of type [group_call(01 at 192.168.25.61)] cause:
>> > [CHAN_NOT_IMPLEMENTED]
>> > 2011-07-20 12:28:49.304970 [DEBUG] switch_ivr_originate.c:3318 Originate
>> > Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>> > 2011-07-20 12:28:49.304970 [ERR] mod_conference.c:5169 Cannot create
>> > outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>> >
>> > I think that this is related to the fact that the group is created in
>> > sqlite
>> > DB and not in the xml dialplan. So I've tried:
>> >
>> > api expand conference ds_totale dial group call:'01@${domain_name}'
>> >
>> > But the output is:
>> >
>> > Content-Type: api/response
>> > Content-Length: 47
>> >
>> > Call Requested: result: [NO_ROUTE_DESTINATION]
>> >
>> > Freeswitch log:
>> > 2011-07-20 12:36:49.964965 [DEBUG] switch_ivr_originate.c:1879 Parsing
>> > global variables
>> > 2011-07-20 12:36:49.964965 [NOTICE] switch_ivr_originate.c:2453 Cannot
>> > create outgoing channel of type [error] cause: [NO_ROUTE_DESTINATION]
>> > 2011-07-20 12:36:49.964965 [DEBUG] switch_ivr_originate.c:3318 Originate
>> > Resulted in Error Cause: 3 [NO_ROUTE_DESTINATION]
>> > 2011-07-20 12:36:49.964965 [NOTICE] switch_ivr_originate.c:2453 Cannot
>> > create outgoing channel of type [group] cause: [NO_ROUTE_DESTINATION]
>> > 2011-07-20 12:36:49.964965 [DEBUG] switch_ivr_originate.c:3318 Originate
>> > Resulted in Error Cause: 3 [NO_ROUTE_DESTINATION]
>> > 2011-07-20 12:36:49.964965 [ERR] mod_conference.c:5169 Cannot create
>> > outgoing channel, cause: NO_ROUTE_DESTINATION
>> >
>> > It seems a problem in parsing; in fact the command:
>> >
>> > api expand group call:'01@${domain_name}'
>> >
>> > gives the following output:
>> >
>> > Content-Type: api/response
>> > Content-Length: 85
>> >
>> >
>> > sofia/internal/sip:1001 at 192.168.25.61:5070,sofia/internal/sip:1016 at 192.168.25.31:5060
>> >
>> > So I've tried:
>> >
>> > api expand conference ds_totale dial
>> >
>> > sofia/internal/sip:1001 at 192.168.25.61:5070,sofia/internal/sip:1016 at 192.168.25.31:5060
>> >
>> > The output is:
>> > Content-Type: api/response
>> > Content-Length: 34
>> >
>> > Call Requested: result: [SUCCESS]
>> >
>> > But only 1016 is added to the conference while 1001 is hanged up with
>> > cause:
>> > LOSE_RACE.
>> >
>> > So the questions are:
>> >
>> > 1. What is wrong with the syntax of the following command?
>> >
>> > ?api expand conference ds_totale dial group call:'01@${domain_name}'
>> >
>> > 2. How can it be avoided that parties which "loose the race" are hanged
>> > up?
>> >
>> >
>> > Thank you very much.
>> >
>> > F. Bigliardi
>> >
>> >
>> >
>> > 2011/7/19 Michael Collins <msc at freeswitch.org>
>> >>
>> >>
>> >> On Tue, Jul 19, 2011 at 5:47 AM, Fabio Bigliardi
>> >> <fabio.bigliardi at gmail.com> wrote:
>> >>>
>> >>> Hi all,
>> >>> I would like to be able to invite a? group to a conference through an
>> >>> API
>> >>> command.
>> >>>
>> >>> In xml dialplan this can be achieved through the following action:
>> >>>
>> >>> <action application="conference_set_auto_outcall"
>> >>> data="${group_call(my_group}"/>
>> >>>
>> >>> In the API, I suppose the right command is "dial":
>> >>>
>> >>> api conference my_conf dial <group dial string>
>> >>>
>> >>> But how can I specify group dial string?
>> >>
>> >> Try this:
>> >> expand api conference my_conf dial group_call(my_group)
>> >> -MC
>> >>
>> >>>
>> >>> Thank you for your support.
>> >>>
>> >>> Best regards,
>> >>>
>> >>> Fabio Bigliardi
>> >>>
>> >>> _______________________________________________
>> >>> 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
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> 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
>> >>
>> >
>> >
>> > _______________________________________________
>> > 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
>> >
>> >
>>
>> _______________________________________________
>> 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
>
>
> _______________________________________________
> 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 steveayre at gmail.com  Wed Jul 20 17:49:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 20 Jul 2011 14:49:54 +0100
Subject: [Freeswitch-users] API conference invite group
In-Reply-To: <CAFiqYukZ-jzg8NAsJgn-6ugOwXahbd3JG8ZpPrbH8=ggs1rvmw@mail.gmail.com>
References: <CABtP9QiLqZXUwGv9udZ7gpyHo=XLHT_DcpGMDnvmYPU7LUjf2Q@mail.gmail.com>
	<CAKzWOxXv-K1riCO_DUk9JbEudVaWP6c9m5VuskaeqqzJ+dcE8g@mail.gmail.com>
	<CABtP9QhBOwZWBSrZHvimtVUCr4w4mVMbiZ0mO7J6MLA4u2F_rw@mail.gmail.com>
	<CAFiqYuk4Za30-jLmj5QKiWMtkyoUCeZ203T8rTFkMNOt2P3eyg@mail.gmail.com>
	<CABtP9QjXwDoia-64TtuGavSCdAmsHNgzqdiZ7PvcEXp=D-9+OA@mail.gmail.com>
	<CAFiqYukZ-jzg8NAsJgn-6ugOwXahbd3JG8ZpPrbH8=ggs1rvmw@mail.gmail.com>
Message-ID: <CAFiqYukOui44Q=2ovDO18DpR=kqCs3uvT7RVVX1pVWq9htRLDg@mail.gmail.com>

Since you're trying to invite all the members into the conference, I
think the better way to approach this is:

- 'api groupcall <group>@<domain>'
- Read reply to get list of group members and then for each member execute:
- 'api conference <confname> dial <dialstring>'

-Steve


On 20 July 2011 14:45, Steven Ayre <steveayre at gmail.com> wrote:
>> Only one member of the group is answered; the others are hanged up with
>> cause: LOSE_RACE.
>
> That's normal. An originate only ever gets answered by one party.
>
> -Steve
>
>
>
> On 20 July 2011 13:45, Fabio Bigliardi <fabio.bigliardi at gmail.com> wrote:
>> It works only with groups defined in XML directory and not with groups in
>> sqlite DB.
>> Only one member of the group is answered; the others are hanged up with
>> cause: LOSE_RACE.
>>
>> -Fabio
>>
>>
>> 2011/7/20 Steven Ayre <steveayre at gmail.com>
>>>
>>> I think you want to be dialing using the group/ prefix:
>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#group
>>>
>>> Try: api expand conference ds_totale dial group/01@${domain_name}
>>>
>>> All dialstrings start with chantype/ which identifies which endpoint
>>> module to use to dial. CHAN_NOT_IMPLEMENTED is the error given if that
>>> part was not recognised.
>>>
>>> Since it looks like you might be trying to connect using ESL directly,
>>> look in libs/esl in your git checkout. You'll find esl libraries there
>>> in a range of languages that'll handle the protocol for you.
>>>
>>> -Steve
>>>
>>>
>>>
>>>
>>>
>>> On 20 July 2011 11:55, Fabio Bigliardi <fabio.bigliardi at gmail.com> wrote:
>>> > Hi Michael,
>>> >
>>> > I've tried the command:
>>> >
>>> > api expand conference ds_totale dial group_call(01@${domain_name})
>>> >
>>> > But the output is:
>>> >
>>> > Content-Type: api/response
>>> > Content-Length: 47
>>> >
>>> > Call Requested: result: [CHAN_NOT_IMPLEMENTED]
>>> >
>>> > Freeswitch log:
>>> > 2011-07-20 12:28:49.304970 [ERR] switch_core_session.c:413 Could not
>>> > locate
>>> > channel type group_call(01 at 192.168.25.61)
>>> > 2011-07-20 12:28:49.304970 [NOTICE] switch_ivr_originate.c:2453 Cannot
>>> > create outgoing channel of type [group_call(01 at 192.168.25.61)] cause:
>>> > [CHAN_NOT_IMPLEMENTED]
>>> > 2011-07-20 12:28:49.304970 [DEBUG] switch_ivr_originate.c:3318 Originate
>>> > Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
>>> > 2011-07-20 12:28:49.304970 [ERR] mod_conference.c:5169 Cannot create
>>> > outgoing channel, cause: CHAN_NOT_IMPLEMENTED
>>> >
>>> > I think that this is related to the fact that the group is created in
>>> > sqlite
>>> > DB and not in the xml dialplan. So I've tried:
>>> >
>>> > api expand conference ds_totale dial group call:'01@${domain_name}'
>>> >
>>> > But the output is:
>>> >
>>> > Content-Type: api/response
>>> > Content-Length: 47
>>> >
>>> > Call Requested: result: [NO_ROUTE_DESTINATION]
>>> >
>>> > Freeswitch log:
>>> > 2011-07-20 12:36:49.964965 [DEBUG] switch_ivr_originate.c:1879 Parsing
>>> > global variables
>>> > 2011-07-20 12:36:49.964965 [NOTICE] switch_ivr_originate.c:2453 Cannot
>>> > create outgoing channel of type [error] cause: [NO_ROUTE_DESTINATION]
>>> > 2011-07-20 12:36:49.964965 [DEBUG] switch_ivr_originate.c:3318 Originate
>>> > Resulted in Error Cause: 3 [NO_ROUTE_DESTINATION]
>>> > 2011-07-20 12:36:49.964965 [NOTICE] switch_ivr_originate.c:2453 Cannot
>>> > create outgoing channel of type [group] cause: [NO_ROUTE_DESTINATION]
>>> > 2011-07-20 12:36:49.964965 [DEBUG] switch_ivr_originate.c:3318 Originate
>>> > Resulted in Error Cause: 3 [NO_ROUTE_DESTINATION]
>>> > 2011-07-20 12:36:49.964965 [ERR] mod_conference.c:5169 Cannot create
>>> > outgoing channel, cause: NO_ROUTE_DESTINATION
>>> >
>>> > It seems a problem in parsing; in fact the command:
>>> >
>>> > api expand group call:'01@${domain_name}'
>>> >
>>> > gives the following output:
>>> >
>>> > Content-Type: api/response
>>> > Content-Length: 85
>>> >
>>> >
>>> > sofia/internal/sip:1001 at 192.168.25.61:5070,sofia/internal/sip:1016 at 192.168.25.31:5060
>>> >
>>> > So I've tried:
>>> >
>>> > api expand conference ds_totale dial
>>> >
>>> > sofia/internal/sip:1001 at 192.168.25.61:5070,sofia/internal/sip:1016 at 192.168.25.31:5060
>>> >
>>> > The output is:
>>> > Content-Type: api/response
>>> > Content-Length: 34
>>> >
>>> > Call Requested: result: [SUCCESS]
>>> >
>>> > But only 1016 is added to the conference while 1001 is hanged up with
>>> > cause:
>>> > LOSE_RACE.
>>> >
>>> > So the questions are:
>>> >
>>> > 1. What is wrong with the syntax of the following command?
>>> >
>>> > ?api expand conference ds_totale dial group call:'01@${domain_name}'
>>> >
>>> > 2. How can it be avoided that parties which "loose the race" are hanged
>>> > up?
>>> >
>>> >
>>> > Thank you very much.
>>> >
>>> > F. Bigliardi
>>> >
>>> >
>>> >
>>> > 2011/7/19 Michael Collins <msc at freeswitch.org>
>>> >>
>>> >>
>>> >> On Tue, Jul 19, 2011 at 5:47 AM, Fabio Bigliardi
>>> >> <fabio.bigliardi at gmail.com> wrote:
>>> >>>
>>> >>> Hi all,
>>> >>> I would like to be able to invite a? group to a conference through an
>>> >>> API
>>> >>> command.
>>> >>>
>>> >>> In xml dialplan this can be achieved through the following action:
>>> >>>
>>> >>> <action application="conference_set_auto_outcall"
>>> >>> data="${group_call(my_group}"/>
>>> >>>
>>> >>> In the API, I suppose the right command is "dial":
>>> >>>
>>> >>> api conference my_conf dial <group dial string>
>>> >>>
>>> >>> But how can I specify group dial string?
>>> >>
>>> >> Try this:
>>> >> expand api conference my_conf dial group_call(my_group)
>>> >> -MC
>>> >>
>>> >>>
>>> >>> Thank you for your support.
>>> >>>
>>> >>> Best regards,
>>> >>>
>>> >>> Fabio Bigliardi
>>> >>>
>>> >>> _______________________________________________
>>> >>> 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
>>> >>>
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> 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
>>> >>
>>> >
>>> >
>>> > _______________________________________________
>>> > 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
>>> >
>>> >
>>>
>>> _______________________________________________
>>> 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
>>
>>
>> _______________________________________________
>> 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 dave9876 at gmx.com  Wed Jul 20 11:34:19 2011
From: dave9876 at gmx.com (dave9876 at gmx.com)
Date: Wed, 20 Jul 2011 07:34:19 +0000
Subject: [Freeswitch-users] Different REGISTER and INVITE From:
Message-ID: <20110720073419.73470@gmx.com>

Hi,
I'm trying to test a gateway with FS.

The GW requires me to register with "From: 12345_01 at gw.example.com", but when calling I need to have "From: +12345 at gw.example.com" (with "+" and without "_01")

If I try from the console:
originate {origination_caller_id_number='+12345'}sofia/gateway/gw.example.com/+10000 5000

I still get "From: 12345_01 at gw.example.com", since I have to have "param name="from-user" value="12345_01" set in my GW profile.

Any ideas on how to force a different "From" on the INVITE?

I tried calling from a softphone registered with FS, but then I had trouble with the from-domain instead.

Any help is much appreciated.


Dave


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

Hello all!

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

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

-Michael
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From steveayre at gmail.com  Wed Jul 20 19:55:24 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 20 Jul 2011 16:55:24 +0100
Subject: [Freeswitch-users] Different REGISTER and INVITE From:
In-Reply-To: <20110720073419.73470@gmx.com>
References: <20110720073419.73470@gmx.com>
Message-ID: <CAFiqYu=sybnzKtKjfBNtHFtLKe9-wD-13T5jwvNJU==uWg_5yA@mail.gmail.com>

Try setting this in the gateway config:

    <!--/// replace the INVITE from user with the channel's caller-id ///-->
    <!--/// this is false by default because outbound calls generally
need username in the INVITE ///-->
    <!--Use the callerid of an inbound call in the from field on
outbound calls via this gateway -->
    <param name="caller-id-in-from" value="true"/>

You'll need to do 'sofia profile NAME killgw NAME', 'sofia profile NAME rescan'.

It'll then use the caller id in the From for the INVITE, so this
should then work:
originate {origination_caller_id_number='+12345'}sofia/gateway/gw.example.com/+10000
5000

-Steve




On 20 July 2011 08:34,  <dave9876 at gmx.com> wrote:
> Hi,
> I'm trying to test a gateway with FS.
>
> The GW requires me to register with "From: 12345_01 at gw.example.com", but when calling I need to have "From: +12345 at gw.example.com" (with "+" and without "_01")
>
> If I try from the console:
> originate {origination_caller_id_number='+12345'}sofia/gateway/gw.example.com/+10000 5000
>ld
> I still get "From: 12345_01 at gw.example.com", since I have to have "param name="from-user" value="12345_01" set in my GW profile.
>
> Any ideas on how to force a different "From" on the INVITE?
>
> I tried calling from a softphone registered with FS, but then I had trouble with the from-domain instead.
>
> Any help is much appreciated.
>
>
> Dave
>
> _______________________________________________
> 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 adavidm at gmail.com  Wed Jul 20 19:48:44 2011
From: adavidm at gmail.com (David Martin)
Date: Wed, 20 Jul 2011 16:48:44 +0100
Subject: [Freeswitch-users] Cisco CUCM 6.1 and Freeswitch SIP trunk
Message-ID: <CA+ViVc99bqe2e8AAy-ROxLA_LPumQv7q50ygPNQjf_KMzypKAg@mail.gmail.com>

All,

Firstly apologies if this is not the correct place to ask, but I am having
problems getting CUCM 6.1 to talk with freeswitch over a SIP trunk (I only
need inbound from Cisco -> Freeswitch). I have enabled 3pcc, but am getting
the following sofia trace:

http://pastebin.com/g9VmvxFe

Can anyone point me in the right direction? I have also tried enabling MTP
on the Cisco side, but this does not make any difference.

Thanks in advance.

David
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From cdunkel at harris.com  Wed Jul 20 20:21:03 2011
From: cdunkel at harris.com (Dunkel, Christopher)
Date: Wed, 20 Jul 2011 16:21:03 +0000
Subject: [Freeswitch-users] SIP Presence Subscription Management
Message-ID: <A68E65C355992E42A4136563912136AB235EEEFF@MLBMXUS21.cs.myharris.net>

I'm having an issue with SIP presence through Freeswitch and I'm hoping that someone might be able to help me determine if my problem is with Freeswitch or with the client(s) I'm using.

The system I'm using includes Freeswitch (version 1.0.6) running on a Linux box and two soft phone clients running on iPhone and Android (PJSip on iPhone and Linphone on Android).  The problem that I'm having appears the same on both clients, which leads me to believe that this may be a Freeswitch config problem.

The clients are able to send a SUBSCRIBE message and receive a NOTIFY immediately when one of the subscribed users registers with Freeswitch.  They also receive the offline NOTIFY when the user unregisters from Freeswitch.  However, an online NOTIFY message is not sent when the user comes back online.  It seems as if Freeswitch is clearing all presence subscriptions for a particular user when they unregister from the server.  Is there any way to prevent Freeswitch from clearing these subscribes unless specifically directed to do so?

Also, on a possibly related subject, I am only able to receive "online" and "offline" presence notifications.  I only receive "offline" presence notifications when a user unregisters from Freeswitch.  Attempting to send any other presence value from one of the clients (i.e. "busy", "be right back", "offline", etc...) only results in an "online" notification being sent.  Has anyone else seen similar behavior?

Thanks,
Chris
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From msc at freeswitch.org  Wed Jul 20 22:16:41 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Jul 2011 11:16:41 -0700
Subject: [Freeswitch-users] FreeTDM viewer
In-Reply-To: <CAF=swh-E5=Utv9YF2ZATYF=GeSqtJRC685pMUbJiyio3v5qXBA@mail.gmail.com>
References: <CAF=swh-E5=Utv9YF2ZATYF=GeSqtJRC685pMUbJiyio3v5qXBA@mail.gmail.com>
Message-ID: <CAKzWOxW6fL4hxGpqHEyNr-YfzrGjVMqHWTuL6LstW2+7P=9ToQ@mail.gmail.com>

Valery,

Thanks for giving back! We appreciate it when people let us know what
they're doing with FreeSWITCH. I don't have any TDM-based systems at the
moment but others among us do so they will no doubt appreciate having more
tools available.

Thanks,
MC

On Tue, Jul 19, 2011 at 9:01 PM, Valery Kalinin <valery.kalinin at gmail.com>wrote:

> I'm write small web utility for FreeTDM channel statuses view:
>
> FreeTDM viewer
> Web-based online view FreeTDM spans & channels statuses.
>
> Download from:
> https://sites.google.com/site/freeswitched/home/downloads
>
> Installation guide:
>     Upload files (ftdmv.php & jquery-1.6.1.min.js) to your web-server with
> PHP support.
>     Check modules.conf and event_socket.conf for enable module
> mod_event_socket
>     Change IP address, port, password for FreeSWITCH server and FreeTDM
> spans in ftdmv.php file
>     Enter in browser http://you-server-name/ftdmv.php
>
>
> _______________________________________________
> 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 kris at kriskinc.com  Wed Jul 20 22:22:11 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 20 Jul 2011 14:22:11 -0400
Subject: [Freeswitch-users] Cisco CUCM 6.1 and Freeswitch SIP trunk
In-Reply-To: <CA+ViVc99bqe2e8AAy-ROxLA_LPumQv7q50ygPNQjf_KMzypKAg@mail.gmail.com>
References: <CA+ViVc99bqe2e8AAy-ROxLA_LPumQv7q50ygPNQjf_KMzypKAg@mail.gmail.com>
Message-ID: <CAKDfjgd0R0Sb_BYpmyHLjmv-oPajQ_uw_5ev2ioPDrXXM_3qpw@mail.gmail.com>

That trace is missing the INVITE that starts the transaction.

On Wed, Jul 20, 2011 at 11:48 AM, David Martin <adavidm at gmail.com> wrote:
> All,
> Firstly apologies if this is not the correct place to ask, but I am having
> problems getting CUCM 6.1 to talk with freeswitch over a SIP trunk (I only
> need inbound from Cisco -> Freeswitch). I have enabled 3pcc, but am getting
> the following sofia trace:
> http://pastebin.com/g9VmvxFe
> Can anyone point me in the right direction? I have also tried enabling MTP
> on the Cisco side, but this does not make any difference.
> Thanks in advance.
> David
> _______________________________________________
> 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
>
>



-- 
Kristian Kielhofner


From freeswitch at earthspike.net  Wed Jul 20 22:44:22 2011
From: freeswitch at earthspike.net (John)
Date: Wed, 20 Jul 2011 19:44:22 +0100
Subject: [Freeswitch-users] FreeTDM viewer
In-Reply-To: <CAKzWOxW6fL4hxGpqHEyNr-YfzrGjVMqHWTuL6LstW2+7P=9ToQ@mail.gmail.com>
References: <CAF=swh-E5=Utv9YF2ZATYF=GeSqtJRC685pMUbJiyio3v5qXBA@mail.gmail.com>
	<CAKzWOxW6fL4hxGpqHEyNr-YfzrGjVMqHWTuL6LstW2+7P=9ToQ@mail.gmail.com>
Message-ID: <4E272206.3090606@earthspike.net>

Valery,

My thanks as well.  Although I will not use it directly, it is a good
demonstration of something similar to what I want to do later to monitor
my BRI trunks.

Thank you,

John

On 20/07/11 19:16, Michael Collins wrote:
> Valery,
>
> Thanks for giving back! We appreciate it when people let us know what
> they're doing with FreeSWITCH. I don't have any TDM-based systems at
> the moment but others among us do so they will no doubt appreciate
> having more tools available.
>
> Thanks,
> MC
>
> On Tue, Jul 19, 2011 at 9:01 PM, Valery Kalinin
> <valery.kalinin at gmail.com <mailto:valery.kalinin at gmail.com>> wrote:
>
>     I'm write small web utility for FreeTDM channel statuses view:
>
>     FreeTDM viewer
>     Web-based online view FreeTDM spans & channels statuses.
>
>     Download from:
>     https://sites.google.com/site/freeswitched/home/downloads
>
>     Installation guide:
>         Upload files (ftdmv.php & jquery-1.6.1.min.js) to your
>     web-server with PHP support.
>         Check modules.conf and event_socket.conf for enable module
>     mod_event_socket
>         Change IP address, port, password for FreeSWITCH server and
>     FreeTDM spans in ftdmv.php file
>         Enter in browser http://you-server-name/ftdmv.php
>
>
>     _______________________________________________
>     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
>     <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
>
>
>
>
> _______________________________________________
> 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 msc at freeswitch.org  Thu Jul 21 00:31:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Jul 2011 13:31:21 -0700
Subject: [Freeswitch-users] ClueCon 2011 - Sharing rooms
Message-ID: <CAKzWOxUmDZyqSmTgummnScQmo5WaiVEhmK2eX3SC=X4MHKdCGg@mail.gmail.com>

Hello all!

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

Thanks,
Michael
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From mike.burlingame at me.com  Thu Jul 21 00:36:44 2011
From: mike.burlingame at me.com (Mike Burlingame)
Date: Wed, 20 Jul 2011 13:36:44 -0700
Subject: [Freeswitch-users] Call disconnect after transfer
Message-ID: <50F5BA53-5C2B-4378-9702-7E7512C57F4A@me.com>

I am having a small issue with calls getting disconnected after transfer from a customer signaling is posted at http://pastebin.freeswitch.org/16869

but basic call flow is call is sent to user - user answers - user transfer the call to another user - invite is sent to Freeswitch with out SDP - 200 OK provided to customer with SDP - ACK received from customer with SDP - Freeswitch disconnects call with [INCOMPATIBLE_DESTINATION]

I have 3PCC enabled in FreeSwitch via sofia.conf.xml  <param name="enable-3pcc" value="true"/>
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From rhill3221 at hotmail.com  Thu Jul 21 01:50:49 2011
From: rhill3221 at hotmail.com (Rob Hill)
Date: Wed, 20 Jul 2011 16:50:49 -0500
Subject: [Freeswitch-users] Google voice terminating outbound calls
Message-ID: <SNT138-w55C08D46F533B37FDE13F5BE4C0@phx.gbl>


I'm using mod dingaling to connect via Google Voice.  Incoming and outgoing calls were working fine until just a few days ago, now outgoing calls are terminated after ~15 seconds.  Incoming calls are still fine.

Here is the disconnect info :
2011-07-20 17:36:20.485777 [INFO] libdingaling.c:1371 SecRECV:
-------------------------------------------------------------------------------
<iq from="+1312xxxxxxx at voice.google.com/srvres-MTAuMTc3LjEyMi4xNDo5ODgz" to="r at gmail.com/talkCC63ED4F" id="jingle:10.177.122.14-2416924:1:3F7BB3A2" type="set">
  <ses:session type="terminate" id="1514126677" initiator="r at gmail.com/talkCC63ED4F" xmlns:ses="http://www.google.com/session">
    <ses:reason>
      <ses:timeout></ses:timeout>
      <ses:text>Session timed out</ses:text>
    </ses:reason>
    <pho:recipient-unavailable xmlns:pho="http://www.google.com/session/phone"></pho:recipient-unavailable>
  </ses:session>
</iq>

To make matters more confusing I've tested my configuration on two different hosts and outgoing calls work fine on the second host (using the exact same configuration files).
Any ideas where this might be going wrong?

a

O
 		 	   		  
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From adavidm at gmail.com  Thu Jul 21 02:54:40 2011
From: adavidm at gmail.com (David Martin)
Date: Wed, 20 Jul 2011 23:54:40 +0100
Subject: [Freeswitch-users] Cisco CUCM 6.1 and Freeswitch SIP trunk
In-Reply-To: <CAKDfjgd0R0Sb_BYpmyHLjmv-oPajQ_uw_5ev2ioPDrXXM_3qpw@mail.gmail.com>
References: <CA+ViVc99bqe2e8AAy-ROxLA_LPumQv7q50ygPNQjf_KMzypKAg@mail.gmail.com>
	<CAKDfjgd0R0Sb_BYpmyHLjmv-oPajQ_uw_5ev2ioPDrXXM_3qpw@mail.gmail.com>
Message-ID: <80E395DA-B3F2-42CE-AB9F-098096B32B45@gmail.com>

Forgive my confusion, I'm new to freeswitch. Does the invite not start at line 7 on the pastebin?

David

Sent from my iPhone

On 20 Jul 2011, at 07:22 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:

> That trace is missing the INVITE that starts the transaction.
> 
> On Wed, Jul 20, 2011 at 11:48 AM, David Martin <adavidm at gmail.com> wrote:
>> All,
>> Firstly apologies if this is not the correct place to ask, but I am having
>> problems getting CUCM 6.1 to talk with freeswitch over a SIP trunk (I only
>> need inbound from Cisco -> Freeswitch). I have enabled 3pcc, but am getting
>> the following sofia trace:
>> http://pastebin.com/g9VmvxFe
>> Can anyone point me in the right direction? I have also tried enabling MTP
>> on the Cisco side, but this does not make any difference.
>> Thanks in advance.
>> David
>> _______________________________________________
>> 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
>> 
>> 
> 
> 
> 
> -- 
> Kristian Kielhofner
> 
> _______________________________________________
> 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 lakersman2006 at yahoo.com  Thu Jul 21 03:45:33 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Wed, 20 Jul 2011 16:45:33 -0700 (PDT)
Subject: [Freeswitch-users] invoking php scripts from dialplan
Message-ID: <1311205533.33756.YahooMailClassic@web161011.mail.bf1.yahoo.com>

Hi,

I wanted to know is there another method of invoking a php script from the dialplan besides using the following because I still cannot get the script to run for wahtever reason. Yes, I have opened up the ports on my firewall. When I run the php script directly from console it outputs fine, but i cannot get it to output via the dialplan. 

<!-- dialplan.xml -->
<extension name="outbound-socket">
<condition field="destination_number" expression="^55(522)$">
<action application="set" data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/>
<action application="socket" data="127.0.0.1:8021 async full"/>
</condition>
</extension>

//I then ran fs_ivrd with this command:
/usr/local/freeswitch/bin/fs_ivrd -h 127.0.0.1 -p 8021


//my php script: ivrd-demo.php
#!/usr/bin/php -q

<?php

// set a couple of things so we dont kill the system
ob_implicit_flush(true);
set_time_limit(30);

// Open stdin so we can read the data in
$in = fopen("php://stdin", "r");

// Connect
echo "connect\n\n";

// Answer
echo "sendmsg\n";
echo "call-command: execute\n";
echo "execute-app-name: answer\n\n";

// Play a prompt
echo "sendmsg\n";
echo "call-command: execute\n";
echo "execute-app-name: playback\n";
echo "execute-app-arg: /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n";

// Wait
sleep(5);

// Hangup
echo "sendmsg\n";
echo "call-command: hangup\n\n";

fclose($in);

?> 


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From bryansmart at bryansmart.com  Thu Jul 21 04:17:01 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Wed, 20 Jul 2011 20:17:01 -0400
Subject: [Freeswitch-users] Pizza Demo Grammar
Message-ID: <45543322-DEDF-4125-ADB9-EABA63C446C5@bryansmart.com>

Hi.

Trying to install the Pizza demo.

The Wiki says to download the grammar from www.bkw.org, but that entire site seems to be down.

Are these files available somewhere else? I tried to find them with Google, but only found references to the Wiki page and references to it in FS updates.

Thanks
Bryan



From brian at freeswitch.org  Thu Jul 21 04:19:58 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Jul 2011 19:19:58 -0500
Subject: [Freeswitch-users] Pizza Demo Grammar
In-Reply-To: <45543322-DEDF-4125-ADB9-EABA63C446C5@bryansmart.com>
References: <45543322-DEDF-4125-ADB9-EABA63C446C5@bryansmart.com>
Message-ID: <C35D5B41-1661-462A-9621-84D11DE60B28@freeswitch.org>

Yah my box died this morning.  So i'm in need of a new one.  Anyone wish to donate to my new linux box cause?  paypal brian at bkw.org ? I think the hard drive is intact so i'll get the data back up as soon as I get a new box.

Thanks,
/b

On Jul 20, 2011, at 7:17 PM, Bryan Smart wrote:

> Hi.
> 
> Trying to install the Pizza demo.
> 
> The Wiki says to download the grammar from www.bkw.org, but that entire site seems to be down.
> 
> Are these files available somewhere else? I tried to find them with Google, but only found references to the Wiki page and references to it in FS updates.
> 
> Thanks
> Bryan
> 
> 
> _______________________________________________
> 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 cmrienzo at gmail.com  Thu Jul 21 05:19:39 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 20 Jul 2011 21:19:39 -0400
Subject: [Freeswitch-users] Cisco CUCM 6.1 and Freeswitch SIP trunk
In-Reply-To: <80E395DA-B3F2-42CE-AB9F-098096B32B45@gmail.com>
References: <CA+ViVc99bqe2e8AAy-ROxLA_LPumQv7q50ygPNQjf_KMzypKAg@mail.gmail.com>
	<CAKDfjgd0R0Sb_BYpmyHLjmv-oPajQ_uw_5ev2ioPDrXXM_3qpw@mail.gmail.com>
	<80E395DA-B3F2-42CE-AB9F-098096B32B45@gmail.com>
Message-ID: <CAD-qRkjh2-St09KcqbMEA=muQw5tzHckV=TVVMRdFHLYqLr6OQ@mail.gmail.com>

Line 7 is the 200 OK response to the INVITE.


On Wed, Jul 20, 2011 at 6:54 PM, David Martin <adavidm at gmail.com> wrote:

> Forgive my confusion, I'm new to freeswitch. Does the invite not start at
> line 7 on the pastebin?
>
> David
>
> Sent from my iPhone
>
> On 20 Jul 2011, at 07:22 PM, Kristian Kielhofner <kris at kriskinc.com>
> wrote:
>
> > That trace is missing the INVITE that starts the transaction.
> >
> > On Wed, Jul 20, 2011 at 11:48 AM, David Martin <adavidm at gmail.com>
> wrote:
> >> All,
> >> Firstly apologies if this is not the correct place to ask, but I am
> having
> >> problems getting CUCM 6.1 to talk with freeswitch over a SIP trunk (I
> only
> >> need inbound from Cisco -> Freeswitch). I have enabled 3pcc, but am
> getting
> >> the following sofia trace:
> >> http://pastebin.com/g9VmvxFe
> >> Can anyone point me in the right direction? I have also tried enabling
> MTP
> >> on the Cisco side, but this does not make any difference.
> >> Thanks in advance.
> >> David
> >> _______________________________________________
> >> 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
> >>
> >>
> >
> >
> >
> > --
> > Kristian Kielhofner
> >
> > _______________________________________________
> > 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
>
> _______________________________________________
> 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 kheimerl at cs.berkeley.edu  Thu Jul 21 06:11:58 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Wed, 20 Jul 2011 19:11:58 -0700
Subject: [Freeswitch-users] Using chat command in a dialplan
In-Reply-To: <83217555-012B-4861-AAB9-045B2E78660D@gmail.com>
References: <CACyT-3m7nMTCwav_j_CVqtL-V+T9vFcyGwsrD6uatJ+bkVti3Q@mail.gmail.com>
	<1311149680.32170.264.camel@luna.madrid.commsmundi.com>
	<83217555-012B-4861-AAB9-045B2E78660D@gmail.com>
Message-ID: <CACyT-3mASOUzN4kX_L4X734Jp1wL5nTbpi-Ytd4TuF-nJA3xaw@mail.gmail.com>

Exactly what I was looking for. Thanks!

On Wed, Jul 20, 2011 at 1:21 AM, Steven Ayre <steveayre at gmail.com> wrote:
> Kurtis,
>
> *Any* api can be run from the dialplan. It's just a slightly different syntax - the link Francois gave you will show you how.
>
> Steve on iPhone
>
> On 20 Jul 2011, at 09:14, Fran?ois Delawarde <fdelawarde at wirelessmundi.com> wrote:
>
>> Hi,
>>
>> Take a look at:
>> http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
>>
>>
>> Fran?ois.
>>
>> On Tue, 2011-07-19 at 23:00 -0700, Kurtis Heimerl wrote:
>>> I feel somewhat embarrassed, but I can't figure out how to cause a
>>> chat event to go off from the dialplan. I've tried:
>>>
>>> <action application="chat"
>>> data="sip|${username}|${destination_number}@${domain_name}|${msg_body}"/>
>>>
>>> and
>>>
>>> <action application="eval" data="chat
>>> sip|${username}|${destination_number}@${domain_name}|${msg_body}"/>
>>>
>>> and neither worked. The wiki says it's part of the API
>>> (http://wiki.freeswitch.org/wiki/Mod_dptools) but I'm pretty sure
>>> that's just a command-line API. Is there a way to run a command-line
>>> call from the dialplan?
>>>
>>> Any direction would be appreciated.
>>>
>>> _______________________________________________
>>> 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
>>
>>
>>
>> _______________________________________________
>> 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
>
> _______________________________________________
> 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 kheimerl at cs.berkeley.edu  Thu Jul 21 10:37:58 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Wed, 20 Jul 2011 23:37:58 -0700
Subject: [Freeswitch-users] Using chat command in a dialplan
In-Reply-To: <CACyT-3mASOUzN4kX_L4X734Jp1wL5nTbpi-Ytd4TuF-nJA3xaw@mail.gmail.com>
References: <CACyT-3m7nMTCwav_j_CVqtL-V+T9vFcyGwsrD6uatJ+bkVti3Q@mail.gmail.com>
	<1311149680.32170.264.camel@luna.madrid.commsmundi.com>
	<83217555-012B-4861-AAB9-045B2E78660D@gmail.com>
	<CACyT-3mASOUzN4kX_L4X734Jp1wL5nTbpi-Ytd4TuF-nJA3xaw@mail.gmail.com>
Message-ID: <CACyT-3nBYP88Ve3oTC6R1WYSPKZkvY08iVTuSxfmm9miFgYEtA@mail.gmail.com>

I'm now having a similar problem with using "originate" to create a
new call inside a dialplan. This one is particularly perplexing (and
small enough not to warrant pastebin)

Here's the important bits of the dialplan:
	<action application="log" data="(user/${username} ${destination_number})"/>
	<action application="set"
data="api_result=${originate(user/${username}
${destination_number})}"/>

and the log output
EXECUTE 9199 log((user/1304 at 192.168.1.144 9199))
2011-07-20 23:17:12.955419 [DEBUG] mod_dptools.c:1202 9199)
EXECUTE 9199 set(api_result=-USAGE <call url>
<exten>|&<application_name>(<app_args>) [<dialplan>] [<context>]
[<cid_name>] [<cid_num>] [<timeout_sec>]
)

So it's somehow mangling the originate commands. If I run the
originate command directly (with the arguments logged!) it works just
fine.
originate user/1304 at 192.168.1.144 9199

What am I missing here? This seems so simple...  but I'm at my wit's end.

Thanks for any direction in advance!


On Wed, Jul 20, 2011 at 7:11 PM, Kurtis Heimerl
<kheimerl at cs.berkeley.edu> wrote:
> Exactly what I was looking for. Thanks!
>
> On Wed, Jul 20, 2011 at 1:21 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> Kurtis,
>>
>> *Any* api can be run from the dialplan. It's just a slightly different syntax - the link Francois gave you will show you how.
>>
>> Steve on iPhone
>>
>> On 20 Jul 2011, at 09:14, Fran?ois Delawarde <fdelawarde at wirelessmundi.com> wrote:
>>
>>> Hi,
>>>
>>> Take a look at:
>>> http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
>>>
>>>
>>> Fran?ois.
>>>
>>> On Tue, 2011-07-19 at 23:00 -0700, Kurtis Heimerl wrote:
>>>> I feel somewhat embarrassed, but I can't figure out how to cause a
>>>> chat event to go off from the dialplan. I've tried:
>>>>
>>>> <action application="chat"
>>>> data="sip|${username}|${destination_number}@${domain_name}|${msg_body}"/>
>>>>
>>>> and
>>>>
>>>> <action application="eval" data="chat
>>>> sip|${username}|${destination_number}@${domain_name}|${msg_body}"/>
>>>>
>>>> and neither worked. The wiki says it's part of the API
>>>> (http://wiki.freeswitch.org/wiki/Mod_dptools) but I'm pretty sure
>>>> that's just a command-line API. Is there a way to run a command-line
>>>> call from the dialplan?
>>>>
>>>> Any direction would be appreciated.
>>>>
>>>> _______________________________________________
>>>> 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
>>>
>>>
>>>
>>> _______________________________________________
>>> 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
>>
>> _______________________________________________
>> 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 dave9876 at gmx.com  Thu Jul 21 11:19:42 2011
From: dave9876 at gmx.com (dave9876 at gmx.com)
Date: Thu, 21 Jul 2011 07:19:42 +0000
Subject: [Freeswitch-users] Different REGISTER and INVITE From:
Message-ID: <20110721071942.81270@gmx.com>

Hi,
 Thanks! I thought I had tried that, but apparently not properly.

 The next problem was that I needed P-Asserted-Identity set to "12345_01". 

 I added "{sip_h_P-Asserted-Identity='sip:12345_01 at gw.example.com',..." and it works fine.

Thanks a lot!


 Dave



----- Original Message -----
From: Steven Ayre
Sent: 07/20/11 05:55 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Different REGISTER and INVITE From:

 Try setting this in the gateway config: <!--/// replace the INVITE from user with the channel's caller-id ///--> <!--/// this is false by default because outbound calls generally need username in the INVITE ///--> <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway --> <param name="caller-id-in-from" value="true"/> You'll need to do 'sofia profile NAME killgw NAME', 'sofia profile NAME rescan'. It'll then use the caller id in the From for the INVITE, so this should then work: originate {origination_caller_id_number='+12345'}sofia/gateway/gw.example.com/+10000 5000 -Steve On 20 July 2011 08:34, <dave9876 at gmx.com> wrote: > Hi, > I'm trying to test a gateway with FS. > > The GW requires me to register with "From: 12345_01 at gw.example.com", but when calling I need to have "From: +12345 at gw.example.com" (with "+" and without "_01") > > If I try from the console: > originate {origination_caller_id_number='+12345'}sofia/gateway/gw.example.com/+10000 5000 >ld > I still get "From: 12345_01 at gw.example.com", since I have to have "param name="from-user" value="12345_01" set in my GW profile. > > Any ideas on how to force a different "From" on the INVITE? > > I tried calling from a softphone registered with FS, but then I had trouble with the from-domain instead. > > Any help is much appreciated. > > > Dave > _______________________________________________ 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 adavidm at gmail.com  Thu Jul 21 11:45:11 2011
From: adavidm at gmail.com (David Martin)
Date: Thu, 21 Jul 2011 08:45:11 +0100
Subject: [Freeswitch-users] Cisco CUCM 6.1 and Freeswitch SIP trunk
In-Reply-To: <CAD-qRkjh2-St09KcqbMEA=muQw5tzHckV=TVVMRdFHLYqLr6OQ@mail.gmail.com>
References: <CA+ViVc99bqe2e8AAy-ROxLA_LPumQv7q50ygPNQjf_KMzypKAg@mail.gmail.com>
	<CAKDfjgd0R0Sb_BYpmyHLjmv-oPajQ_uw_5ev2ioPDrXXM_3qpw@mail.gmail.com>
	<80E395DA-B3F2-42CE-AB9F-098096B32B45@gmail.com>
	<CAD-qRkjh2-St09KcqbMEA=muQw5tzHckV=TVVMRdFHLYqLr6OQ@mail.gmail.com>
Message-ID: <CA+ViVc9ydZfjWwTVVkDxRugAC+s4iV9TPRLEU3v=h7emfnf98Q@mail.gmail.com>

Gotcha, not sure how I missed the INVITE the first time, but here is a new
pastebin that should have everything:

http://pastebin.com/T3eqms8N

thanks in advance once again.

David



On 21 July 2011 02:19, Christopher Rienzo <cmrienzo at gmail.com> wrote:

> Line 7 is the 200 OK response to the INVITE.
>
>
>
> On Wed, Jul 20, 2011 at 6:54 PM, David Martin <adavidm at gmail.com> wrote:
>
>> Forgive my confusion, I'm new to freeswitch. Does the invite not start at
>> line 7 on the pastebin?
>>
>> David
>>
>> Sent from my iPhone
>>
>> On 20 Jul 2011, at 07:22 PM, Kristian Kielhofner <kris at kriskinc.com>
>> wrote:
>>
>> > That trace is missing the INVITE that starts the transaction.
>> >
>> > On Wed, Jul 20, 2011 at 11:48 AM, David Martin <adavidm at gmail.com>
>> wrote:
>> >> All,
>> >> Firstly apologies if this is not the correct place to ask, but I am
>> having
>> >> problems getting CUCM 6.1 to talk with freeswitch over a SIP trunk (I
>> only
>> >> need inbound from Cisco -> Freeswitch). I have enabled 3pcc, but am
>> getting
>> >> the following sofia trace:
>> >> http://pastebin.com/g9VmvxFe
>> >> Can anyone point me in the right direction? I have also tried enabling
>> MTP
>> >> on the Cisco side, but this does not make any difference.
>> >> Thanks in advance.
>> >> David
>> >> _______________________________________________
>> >> 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
>> >>
>> >>
>> >
>> >
>> >
>> > --
>> > Kristian Kielhofner
>> >
>> > _______________________________________________
>> > 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
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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 steveayre at gmail.com  Thu Jul 21 12:23:58 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 21 Jul 2011 09:23:58 +0100
Subject: [Freeswitch-users] Using chat command in a dialplan
In-Reply-To: <CACyT-3nBYP88Ve3oTC6R1WYSPKZkvY08iVTuSxfmm9miFgYEtA@mail.gmail.com>
References: <CACyT-3m7nMTCwav_j_CVqtL-V+T9vFcyGwsrD6uatJ+bkVti3Q@mail.gmail.com>
	<1311149680.32170.264.camel@luna.madrid.commsmundi.com>
	<83217555-012B-4861-AAB9-045B2E78660D@gmail.com>
	<CACyT-3mASOUzN4kX_L4X734Jp1wL5nTbpi-Ytd4TuF-nJA3xaw@mail.gmail.com>
	<CACyT-3nBYP88Ve3oTC6R1WYSPKZkvY08iVTuSxfmm9miFgYEtA@mail.gmail.com>
Message-ID: <CAFiqYumBRYwpNamX_ig525Hbo+v+s+uVZZ8httTu7Vd7LYk9Qg@mail.gmail.com>

Are you trying to join the 2 calls together? If so, use bridge:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge

-Steve


On 21 July 2011 07:37, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:

> I'm now having a similar problem with using "originate" to create a
> new call inside a dialplan. This one is particularly perplexing (and
> small enough not to warrant pastebin)
>
> Here's the important bits of the dialplan:
>        <action application="log" data="(user/${username}
> ${destination_number})"/>
>        <action application="set"
> data="api_result=${originate(user/${username}
> ${destination_number})}"/>
>
> and the log output
> EXECUTE 9199 log((user/1304 at 192.168.1.144 9199))
> 2011-07-20 23:17:12.955419 [DEBUG] mod_dptools.c:1202 9199)
> EXECUTE 9199 set(api_result=-USAGE <call url>
> <exten>|&<application_name>(<app_args>) [<dialplan>] [<context>]
> [<cid_name>] [<cid_num>] [<timeout_sec>]
> )
>
> So it's somehow mangling the originate commands. If I run the
> originate command directly (with the arguments logged!) it works just
> fine.
> originate user/1304 at 192.168.1.144 9199
>
> What am I missing here? This seems so simple...  but I'm at my wit's end.
>
> Thanks for any direction in advance!
>
>
> On Wed, Jul 20, 2011 at 7:11 PM, Kurtis Heimerl
> <kheimerl at cs.berkeley.edu> wrote:
> > Exactly what I was looking for. Thanks!
> >
> > On Wed, Jul 20, 2011 at 1:21 AM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >> Kurtis,
> >>
> >> *Any* api can be run from the dialplan. It's just a slightly different
> syntax - the link Francois gave you will show you how.
> >>
> >> Steve on iPhone
> >>
> >> On 20 Jul 2011, at 09:14, Fran?ois Delawarde <
> fdelawarde at wirelessmundi.com> wrote:
> >>
> >>> Hi,
> >>>
> >>> Take a look at:
> >>> http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
> >>>
> >>>
> >>> Fran?ois.
> >>>
> >>> On Tue, 2011-07-19 at 23:00 -0700, Kurtis Heimerl wrote:
> >>>> I feel somewhat embarrassed, but I can't figure out how to cause a
> >>>> chat event to go off from the dialplan. I've tried:
> >>>>
> >>>> <action application="chat"
> >>>> data="sip|${username}|${destination_number}@
> ${domain_name}|${msg_body}"/>
> >>>>
> >>>> and
> >>>>
> >>>> <action application="eval" data="chat
> >>>> sip|${username}|${destination_number}@${domain_name}|${msg_body}"/>
> >>>>
> >>>> and neither worked. The wiki says it's part of the API
> >>>> (http://wiki.freeswitch.org/wiki/Mod_dptools) but I'm pretty sure
> >>>> that's just a command-line API. Is there a way to run a command-line
> >>>> call from the dialplan?
> >>>>
> >>>> Any direction would be appreciated.
> >>>>
> >>>> _______________________________________________
> >>>> 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
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> 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
> >>
> >> _______________________________________________
> >> 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
> >>
> >
>
> _______________________________________________
> 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 michel.daggelinckx at gmail.com  Thu Jul 21 13:18:48 2011
From: michel.daggelinckx at gmail.com (Michel Daggelinckx)
Date: Thu, 21 Jul 2011 11:18:48 +0200
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <CAFGRPVoPVa8ir-18cBLYDzRNftO96Jd=uJ7-xiZQUFsikUwBkA@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
	<163DDA0D-0B14-4623-B48B-5E9EEE907AED@gmail.com>
	<CAFYZzbu9Nh3Ojb26G_pAiFn9MVmm59qeO8BL02_WLX5Fj0mfUg@mail.gmail.com>
	<CAKzWOxX2CrDESPMOWMtCgSDxt1R2S3=70aDsTrRM2_p6isfkvA@mail.gmail.com>
	<CAGKOgbK=JzEpWmR8=nFq_x47mFFy96nWRRkMZrogY0hxOMnr2g@mail.gmail.com>
	<CAFGRPVoPVa8ir-18cBLYDzRNftO96Jd=uJ7-xiZQUFsikUwBkA@mail.gmail.com>
Message-ID: <CAFYZzbuPhHK0MO-SsfHzLO1ZxT=aa9Vjb7OEGtjjSkjBSw7j7g@mail.gmail.com>

i want to test

On Wed, Jul 20, 2011 at 2:30 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello guys,
>
> I finally set up a git at github. I would like to have a couple testers,
> anyone?
>
> Please bear in mind:
>
> - There's NO installation script. But it is all scripts and web pages, etc.
> - The web interface is in spanish, i still need to translate that.
> - The web design is by no means "cool", it is just functional.
> - There will for sure be a thousand bugs, which will need to be fixed.
>
> Contact me whoever's interested.
>
>
> Thanks all
>
> David
>
>
> On Wed, Jul 6, 2011 at 3:18 PM, Umair Bari <umairbari at gmail.com> wrote:
>
>> Signup at: https://github.com/signup/free
>>
>> Create a new repository: https://github.com/repositories/new
>> Help: http://help.github.com/create-a-repo/
>>
>> A nice help with images:
>> Windows: http://help.github.com/win-set-up-git/
>> Mac: http://help.github.com/mac-set-up-git/
>> Linux: http://help.github.com/linux-set-up-git/
>>
>> On Tue, Jul 5, 2011 at 11:21 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> FYI, Ray is out this week, so for now you might want to throw it up on
>>> github for now.
>>> -MC
>>>
>>>
>>> On Mon, Jul 4, 2011 at 3:59 PM, Michel Daggelinckx <
>>> michel.daggelinckx at gmail.com> wrote:
>>>
>>>> Ask intralanman for acces to the FS contrib
>>>>
>>>> On Mon, Jul 4, 2011 at 7:10 PM, Mohammad Emran <monemran at gmail.com>wrote:
>>>>
>>>>> We can put on google code or git hub.
>>>>>
>>>>> Sent from my iPad
>>>>>
>>>>> On 4 Jul 2011, at 19:12, David Villasmil <
>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>
>>>>> Hello All,
>>>>>
>>>>> I think I'm as ready as i can be to publish this...
>>>>> Can someone guide me into publishing via GIT?
>>>>>
>>>>> thanks
>>>>>
>>>>> David
>>>>>
>>>>> On Thu, Feb 24, 2011 at 12:39 PM, David Villasmil <<david.villasmil.work at gmail.com>
>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>
>>>>>> Hello Guys,
>>>>>>
>>>>>> I'm finishing a "complete" wholesale application created on freeswitch
>>>>>> and I was wondering whether it would be a good idea to put it up on the
>>>>>> wiki. I just don't know how.
>>>>>>
>>>>>> Features include all the following parameters configurable via web
>>>>>> interface:
>>>>>>
>>>>>> - Profile creation based on server IP where traffic is received. You
>>>>>> can have multiple IPs, system will create multiple profiles/diaplans so it
>>>>>> can differentiate.
>>>>>>      - i.e. offer to the same customer a "gold" routing on IP1 and
>>>>>> cheap routing on IP2
>>>>>>
>>>>>> - Customer add/modify/delete
>>>>>>      - IP source
>>>>>>      - Rates for client routes based on areacode
>>>>>>      - Prepaid or postpaid.
>>>>>>           - When cutomer balance is 0, no more calls are allowed.
>>>>>>      - limit max channels
>>>>>>      - Media by-pass
>>>>>>           - When by-passed, customer and provider will exchange RTPs
>>>>>> directly. Else, server will be in the middle.
>>>>>>
>>>>>> - Provider add/modify/delete
>>>>>>      - costs for provider routes based on areacode
>>>>>>      - limit max channels
>>>>>>
>>>>>> - Routing based on areacode, gives great granularity.
>>>>>>
>>>>>> - Routes can be assigned multiple gateways/providers which can in turn
>>>>>> be distributed based on weigth. Includes overflow to next configured GW.
>>>>>>
>>>>>> - Basic financial report generation (totals) by customer/provider
>>>>>>
>>>>>> - Basic traffic ASR/ACD report (totals) by cutomer/provider
>>>>>>
>>>>>> - Basic user administration. (No access level, only total access)
>>>>>>
>>>>>> - CDR export to csv file.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> I also have a prepaid card app... no web interface on that one
>>>>>> though...
>>>>>>
>>>>>> Thanks all
>>>>>>
>>>>>>
>>>>>> David
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>>
>> --
>>
>> Thanks & Regards,
>>
>> Umair Bari
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 kris at kriskinc.com  Thu Jul 21 14:13:17 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 21 Jul 2011 06:13:17 -0400
Subject: [Freeswitch-users] Cisco CUCM 6.1 and Freeswitch SIP trunk
In-Reply-To: <CA+ViVc9ydZfjWwTVVkDxRugAC+s4iV9TPRLEU3v=h7emfnf98Q@mail.gmail.com>
References: <CA+ViVc99bqe2e8AAy-ROxLA_LPumQv7q50ygPNQjf_KMzypKAg@mail.gmail.com>
	<CAKDfjgd0R0Sb_BYpmyHLjmv-oPajQ_uw_5ev2ioPDrXXM_3qpw@mail.gmail.com>
	<80E395DA-B3F2-42CE-AB9F-098096B32B45@gmail.com>
	<CAD-qRkjh2-St09KcqbMEA=muQw5tzHckV=TVVMRdFHLYqLr6OQ@mail.gmail.com>
	<CA+ViVc9ydZfjWwTVVkDxRugAC+s4iV9TPRLEU3v=h7emfnf98Q@mail.gmail.com>
Message-ID: <CAKDfjgfh2162ysaDc3-91-qNR4wV9R5K-Ds-OtSYOXdgiDZD9w@mail.gmail.com>

David,

  The problem here is that even with 3pcc enabled the ACK from
CallManager doesn't have an SDP.  CallManager never provides an SDP
during this entire trace.  Not in the INVITE (preferred) and not in
the ACK (as I already mentioned).

  This is odd, even for CallManager.  Are you sure you have MTP enabled:

http://blog.krisk.org/2010/05/another-sip-gotcha-cisco.html

On Thu, Jul 21, 2011 at 3:45 AM, David Martin <adavidm at gmail.com> wrote:
> Gotcha, not sure how I missed the INVITE the first time, but here is a new
> pastebin that should have everything:
> http://pastebin.com/T3eqms8N
> thanks in advance once again.
> David
>
>
> On 21 July 2011 02:19, Christopher Rienzo <cmrienzo at gmail.com> wrote:
>>
>> Line 7 is the 200 OK response to the INVITE.
>>
>>
>> On Wed, Jul 20, 2011 at 6:54 PM, David Martin <adavidm at gmail.com> wrote:
>>>
>>> Forgive my confusion, I'm new to freeswitch. Does the invite not start at
>>> line 7 on the pastebin?
>>>
>>> David
>>>
>>> Sent from my iPhone
>>>
>>> On 20 Jul 2011, at 07:22 PM, Kristian Kielhofner <kris at kriskinc.com>
>>> wrote:
>>>
>>> > That trace is missing the INVITE that starts the transaction.
>>> >
>>> > On Wed, Jul 20, 2011 at 11:48 AM, David Martin <adavidm at gmail.com>
>>> > wrote:
>>> >> All,
>>> >> Firstly apologies if this is not the correct place to ask, but I am
>>> >> having
>>> >> problems getting CUCM 6.1 to talk with freeswitch over a SIP trunk (I
>>> >> only
>>> >> need inbound from Cisco -> Freeswitch). I have enabled 3pcc, but am
>>> >> getting
>>> >> the following sofia trace:
>>> >> http://pastebin.com/g9VmvxFe
>>> >> Can anyone point me in the right direction? I have also tried enabling
>>> >> MTP
>>> >> on the Cisco side, but this does not make any difference.
>>> >> Thanks in advance.
>>> >> David
>>> >> _______________________________________________
>>> >> 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
>>> >>
>>> >>
>>> >
>>> >
>>> >
>>> > --
>>> > Kristian Kielhofner
>>> >
>>> > _______________________________________________
>>> > 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
>>>
>>> _______________________________________________
>>> 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
>>
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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
>
>



-- 
Kristian Kielhofner


From avi at avimarcus.net  Thu Jul 21 15:39:07 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 21 Jul 2011 14:39:07 +0300
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <CAFYZzbuPhHK0MO-SsfHzLO1ZxT=aa9Vjb7OEGtjjSkjBSw7j7g@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
	<163DDA0D-0B14-4623-B48B-5E9EEE907AED@gmail.com>
	<CAFYZzbu9Nh3Ojb26G_pAiFn9MVmm59qeO8BL02_WLX5Fj0mfUg@mail.gmail.com>
	<CAKzWOxX2CrDESPMOWMtCgSDxt1R2S3=70aDsTrRM2_p6isfkvA@mail.gmail.com>
	<CAGKOgbK=JzEpWmR8=nFq_x47mFFy96nWRRkMZrogY0hxOMnr2g@mail.gmail.com>
	<CAFGRPVoPVa8ir-18cBLYDzRNftO96Jd=uJ7-xiZQUFsikUwBkA@mail.gmail.com>
	<CAFYZzbuPhHK0MO-SsfHzLO1ZxT=aa9Vjb7OEGtjjSkjBSw7j7g@mail.gmail.com>
Message-ID: <CALNZuw=CwmjbpqOZrSPJhzMkaNbJ3JAC0e8O+zwXOiJqHJqKvQ@mail.gmail.com>

I took a quick look at the Lua, and it looks like you bride within the lua
scripts.
That means the lua script keeps running during the whole bridge.
At scale, you might start to see the overhead as an issue.

Instead you can bridge to a custom static extension, e.g.
"bridge_callingcard_$number"  that then uses all the variables and completes
the bridge, and launches whatever lua scripts needs to be run afterwards
from there.

-Avi

On Wed, Jul 20, 2011 at 2:30 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello guys,
>>
>> I finally set up a git at github. I would like to have a couple testers,
>> anyone?
>>
>> Please bear in mind:
>>
>> - There's NO installation script. But it is all scripts and web pages,
>> etc.
>> - The web interface is in spanish, i still need to translate that.
>> - The web design is by no means "cool", it is just functional.
>> - There will for sure be a thousand bugs, which will need to be fixed.
>>
>> Contact me whoever's interested.
>>
>>
>> Thanks all
>>
>> David
>>
>
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From adavidm at gmail.com  Thu Jul 21 16:10:31 2011
From: adavidm at gmail.com (David Martin)
Date: Thu, 21 Jul 2011 13:10:31 +0100
Subject: [Freeswitch-users] Cisco CUCM 6.1 and Freeswitch SIP trunk
In-Reply-To: <CAKDfjgfh2162ysaDc3-91-qNR4wV9R5K-Ds-OtSYOXdgiDZD9w@mail.gmail.com>
References: <CA+ViVc99bqe2e8AAy-ROxLA_LPumQv7q50ygPNQjf_KMzypKAg@mail.gmail.com>
	<CAKDfjgd0R0Sb_BYpmyHLjmv-oPajQ_uw_5ev2ioPDrXXM_3qpw@mail.gmail.com>
	<80E395DA-B3F2-42CE-AB9F-098096B32B45@gmail.com>
	<CAD-qRkjh2-St09KcqbMEA=muQw5tzHckV=TVVMRdFHLYqLr6OQ@mail.gmail.com>
	<CA+ViVc9ydZfjWwTVVkDxRugAC+s4iV9TPRLEU3v=h7emfnf98Q@mail.gmail.com>
	<CAKDfjgfh2162ysaDc3-91-qNR4wV9R5K-Ds-OtSYOXdgiDZD9w@mail.gmail.com>
Message-ID: <CA+ViVc_U+iEifVcwb=n=keR4EO8rTV-SGFEu7XO0q=NSzykGqw@mail.gmail.com>

Kristian,

Here are some screenshots from the trunk config.

http://img200.imageshack.us/img200/9405/siptrunk.png

and for the sip profile

http://img191.imageshack.us/img191/3863/sipprofile.png

I have tried the early media option both ways with the same result. Is the
MTP payload type correct?

Appreciate your help on this.

David


On 21 July 2011 11:13, Kristian Kielhofner <kris at kriskinc.com> wrote:

> David,
>
>  The problem here is that even with 3pcc enabled the ACK from
> CallManager doesn't have an SDP.  CallManager never provides an SDP
> during this entire trace.  Not in the INVITE (preferred) and not in
> the ACK (as I already mentioned).
>
>  This is odd, even for CallManager.  Are you sure you have MTP enabled:
>
> http://blog.krisk.org/2010/05/another-sip-gotcha-cisco.html
>
> On Thu, Jul 21, 2011 at 3:45 AM, David Martin <adavidm at gmail.com> wrote:
> > Gotcha, not sure how I missed the INVITE the first time, but here is a
> new
> > pastebin that should have everything:
> > http://pastebin.com/T3eqms8N
> > thanks in advance once again.
> > David
> >
> >
> > On 21 July 2011 02:19, Christopher Rienzo <cmrienzo at gmail.com> wrote:
> >>
> >> Line 7 is the 200 OK response to the INVITE.
> >>
> >>
> >> On Wed, Jul 20, 2011 at 6:54 PM, David Martin <adavidm at gmail.com>
> wrote:
> >>>
> >>> Forgive my confusion, I'm new to freeswitch. Does the invite not start
> at
> >>> line 7 on the pastebin?
> >>>
> >>> David
> >>>
> >>> Sent from my iPhone
> >>>
> >>> On 20 Jul 2011, at 07:22 PM, Kristian Kielhofner <kris at kriskinc.com>
> >>> wrote:
> >>>
> >>> > That trace is missing the INVITE that starts the transaction.
> >>> >
> >>> > On Wed, Jul 20, 2011 at 11:48 AM, David Martin <adavidm at gmail.com>
> >>> > wrote:
> >>> >> All,
> >>> >> Firstly apologies if this is not the correct place to ask, but I am
> >>> >> having
> >>> >> problems getting CUCM 6.1 to talk with freeswitch over a SIP trunk
> (I
> >>> >> only
> >>> >> need inbound from Cisco -> Freeswitch). I have enabled 3pcc, but am
> >>> >> getting
> >>> >> the following sofia trace:
> >>> >> http://pastebin.com/g9VmvxFe
> >>> >> Can anyone point me in the right direction? I have also tried
> enabling
> >>> >> MTP
> >>> >> on the Cisco side, but this does not make any difference.
> >>> >> Thanks in advance.
> >>> >> David
> >>> >> _______________________________________________
> >>> >> 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
> >>> >>
> >>> >>
> >>> >
> >>> >
> >>> >
> >>> > --
> >>> > Kristian Kielhofner
> >>> >
> >>> > _______________________________________________
> >>> > 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
> >>>
> >>> _______________________________________________
> >>> 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
> >>
> >>
> >> _______________________________________________
> >> 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
> >>
> >
> >
> > _______________________________________________
> > 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
> >
> >
>
>
>
> --
> Kristian Kielhofner
>
> _______________________________________________
> 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 david.villasmil.work at gmail.com  Thu Jul 21 16:12:27 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 21 Jul 2011 14:12:27 +0200
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <CALNZuw=CwmjbpqOZrSPJhzMkaNbJ3JAC0e8O+zwXOiJqHJqKvQ@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
	<163DDA0D-0B14-4623-B48B-5E9EEE907AED@gmail.com>
	<CAFYZzbu9Nh3Ojb26G_pAiFn9MVmm59qeO8BL02_WLX5Fj0mfUg@mail.gmail.com>
	<CAKzWOxX2CrDESPMOWMtCgSDxt1R2S3=70aDsTrRM2_p6isfkvA@mail.gmail.com>
	<CAGKOgbK=JzEpWmR8=nFq_x47mFFy96nWRRkMZrogY0hxOMnr2g@mail.gmail.com>
	<CAFGRPVoPVa8ir-18cBLYDzRNftO96Jd=uJ7-xiZQUFsikUwBkA@mail.gmail.com>
	<CAFYZzbuPhHK0MO-SsfHzLO1ZxT=aa9Vjb7OEGtjjSkjBSw7j7g@mail.gmail.com>
	<CALNZuw=CwmjbpqOZrSPJhzMkaNbJ3JAC0e8O+zwXOiJqHJqKvQ@mail.gmail.com>
Message-ID: <CAFGRPVpBFYs1ZcxTnRVhPeLdGJ8HMo1pJGfMhEm3b3_0RWBWtQ@mail.gmail.com>

Avi,

I have indeed been looking into the overhead, even though it doesn't becomes
until very high load. Good idea, I will look into that, will i not lose all
variables set by the first lua?

thanks for the tip!

David

On Thu, Jul 21, 2011 at 1:39 PM, Avi Marcus <avi at avimarcus.net> wrote:

> I took a quick look at the Lua, and it looks like you bride within the lua
> scripts.
> That means the lua script keeps running during the whole bridge.
> At scale, you might start to see the overhead as an issue.
>
> Instead you can bridge to a custom static extension, e.g.
> "bridge_callingcard_$number"  that then uses all the variables and completes
> the bridge, and launches whatever lua scripts needs to be run afterwards
> from there.
>
> -Avi
>
> On Wed, Jul 20, 2011 at 2:30 PM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Hello guys,
>>>
>>> I finally set up a git at github. I would like to have a couple testers,
>>> anyone?
>>>
>>> Please bear in mind:
>>>
>>> - There's NO installation script. But it is all scripts and web pages,
>>> etc.
>>> - The web interface is in spanish, i still need to translate that.
>>> - The web design is by no means "cool", it is just functional.
>>> - There will for sure be a thousand bugs, which will need to be fixed.
>>>
>>> Contact me whoever's interested.
>>>
>>>
>>> Thanks all
>>>
>>> David
>>>
>>
> _______________________________________________
> 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 adavidm at gmail.com  Thu Jul 21 16:46:59 2011
From: adavidm at gmail.com (David Martin)
Date: Thu, 21 Jul 2011 13:46:59 +0100
Subject: [Freeswitch-users] Cisco CUCM 6.1 and Freeswitch SIP trunk
In-Reply-To: <CA+ViVc_U+iEifVcwb=n=keR4EO8rTV-SGFEu7XO0q=NSzykGqw@mail.gmail.com>
References: <CA+ViVc99bqe2e8AAy-ROxLA_LPumQv7q50ygPNQjf_KMzypKAg@mail.gmail.com>
	<CAKDfjgd0R0Sb_BYpmyHLjmv-oPajQ_uw_5ev2ioPDrXXM_3qpw@mail.gmail.com>
	<80E395DA-B3F2-42CE-AB9F-098096B32B45@gmail.com>
	<CAD-qRkjh2-St09KcqbMEA=muQw5tzHckV=TVVMRdFHLYqLr6OQ@mail.gmail.com>
	<CA+ViVc9ydZfjWwTVVkDxRugAC+s4iV9TPRLEU3v=h7emfnf98Q@mail.gmail.com>
	<CAKDfjgfh2162ysaDc3-91-qNR4wV9R5K-Ds-OtSYOXdgiDZD9w@mail.gmail.com>
	<CA+ViVc_U+iEifVcwb=n=keR4EO8rTV-SGFEu7XO0q=NSzykGqw@mail.gmail.com>
Message-ID: <CA+ViVc-oY3ZV6AYiDkHeSVJucjm=pLnha0=gW4P2A91Ux=TUSQ@mail.gmail.com>

For completeness, here is the INVITE with "Disable Early Media" unchecked:

   ------------------------------------------------------------------------
   INVITE sip:65131111 at 10.201.27.20:5060 SIP/2.0
   Date: Thu, 21 Jul 2011 12:42:02 GMT
   Call-Info: <sip:10.200.170.3:5060
>;method="NOTIFY;Event=telephone-event;Duration=500"
   Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY, PUBLISH
   From: "Ryan Girdlestone" <sip:64105554 at 10.200.170.3
>;tag=90509204-400f-4cb6-9be7-cb5410a83a11-54376643
   Allow-Events: presence, kpml
   Supported: timer,replaces
   Min-SE:  1800
   Remote-Party-ID: "Ryan Girdlestone" <sip:64105554 at 10.200.170.3
>;party=calling;screen=yes;privacy=off
   Content-Length: 0
   User-Agent: Cisco-CUCM6.1
   To: <sip:65131111 at 10.201.27.20>
   Contact: <sip:64105554 at 10.200.170.3:5060>
   Expires: 180
   Call-ID: d3b6ce80-e2811e9a-33-3aac80a at 10.200.170.3
   Via: SIP/2.0/UDP 10.200.170.3:5060;branch=z9hG4bK4d55790bf1
   CSeq: 101 INVITE
   Session-Expires:  1800
   Max-Forwards: 70




On 21 July 2011 13:10, David Martin <adavidm at gmail.com> wrote:

> Kristian,
>
> Here are some screenshots from the trunk config.
>
> http://img200.imageshack.us/img200/9405/siptrunk.png
>
> and for the sip profile
>
> http://img191.imageshack.us/img191/3863/sipprofile.png
>
> I have tried the early media option both ways with the same result. Is the
> MTP payload type correct?
>
> Appreciate your help on this.
>
> David
>
>
> On 21 July 2011 11:13, Kristian Kielhofner <kris at kriskinc.com> wrote:
>
>> David,
>>
>>  The problem here is that even with 3pcc enabled the ACK from
>> CallManager doesn't have an SDP.  CallManager never provides an SDP
>> during this entire trace.  Not in the INVITE (preferred) and not in
>> the ACK (as I already mentioned).
>>
>>  This is odd, even for CallManager.  Are you sure you have MTP enabled:
>>
>> http://blog.krisk.org/2010/05/another-sip-gotcha-cisco.html
>>
>> On Thu, Jul 21, 2011 at 3:45 AM, David Martin <adavidm at gmail.com> wrote:
>> > Gotcha, not sure how I missed the INVITE the first time, but here is a
>> new
>> > pastebin that should have everything:
>> > http://pastebin.com/T3eqms8N
>> > thanks in advance once again.
>> > David
>> >
>> >
>> > On 21 July 2011 02:19, Christopher Rienzo <cmrienzo at gmail.com> wrote:
>> >>
>> >> Line 7 is the 200 OK response to the INVITE.
>> >>
>> >>
>> >> On Wed, Jul 20, 2011 at 6:54 PM, David Martin <adavidm at gmail.com>
>> wrote:
>> >>>
>> >>> Forgive my confusion, I'm new to freeswitch. Does the invite not start
>> at
>> >>> line 7 on the pastebin?
>> >>>
>> >>> David
>> >>>
>> >>> Sent from my iPhone
>> >>>
>> >>> On 20 Jul 2011, at 07:22 PM, Kristian Kielhofner <kris at kriskinc.com>
>> >>> wrote:
>> >>>
>> >>> > That trace is missing the INVITE that starts the transaction.
>> >>> >
>> >>> > On Wed, Jul 20, 2011 at 11:48 AM, David Martin <adavidm at gmail.com>
>> >>> > wrote:
>> >>> >> All,
>> >>> >> Firstly apologies if this is not the correct place to ask, but I am
>> >>> >> having
>> >>> >> problems getting CUCM 6.1 to talk with freeswitch over a SIP trunk
>> (I
>> >>> >> only
>> >>> >> need inbound from Cisco -> Freeswitch). I have enabled 3pcc, but am
>> >>> >> getting
>> >>> >> the following sofia trace:
>> >>> >> http://pastebin.com/g9VmvxFe
>> >>> >> Can anyone point me in the right direction? I have also tried
>> enabling
>> >>> >> MTP
>> >>> >> on the Cisco side, but this does not make any difference.
>> >>> >> Thanks in advance.
>> >>> >> David
>> >>> >> _______________________________________________
>> >>> >> 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
>> >>> >>
>> >>> >>
>> >>> >
>> >>> >
>> >>> >
>> >>> > --
>> >>> > Kristian Kielhofner
>> >>> >
>> >>> > _______________________________________________
>> >>> > 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
>> >>>
>> >>> _______________________________________________
>> >>> 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
>> >>
>> >>
>> >> _______________________________________________
>> >> 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
>> >>
>> >
>> >
>> > _______________________________________________
>> > 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
>> >
>> >
>>
>>
>>
>> --
>> Kristian Kielhofner
>>
>> _______________________________________________
>> 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 boris at tagnet.ru  Thu Jul 21 17:01:17 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 21 Jul 2011 19:01:17 +0600
Subject: [Freeswitch-users] Can't get fax working
In-Reply-To: <4E25605D.7050809@tagnet.ru>
References: <4E25605D.7050809@tagnet.ru>
Message-ID: <4E28231D.3030603@tagnet.ru>

Hello!

     May be a little detailed description of problem. Network configuration
NC1: Kapanga Softphone -- FS -- Audiocodes MP-114 - FAX
NC2: Kapanga Softphone -- FS -- Cisco 5350 - PSTN - FAX

With NC1 I call to fax. If I will wait for Kapanga automaticaly detects 
T38 mode, the fax is _always_ success. If I force Kapanga to start T38 
(by pressing fax key), the fax mostly fails.

With NC2 no matter will or not I press fax key. The fax is always success.
Hope this may help.

> Hello!
>
>     I need help with fax configuration. My network is:
>
> PSTN E1 --- Cisco AS5350 -- Freeswitch -- Audiocodes MP-114 -- Fax
>
> FS version: FreeSWITCH Version 1.0.head (git-062f07f 2011-07-16 
> 12-35-25 +0200)
> Audiocodes firmware: 6.00A.038.004
> I can't get faxes working :( In FS profiles there is a parameter 
> <param name="t38-passthru" value="true"/>
> Cisco DP fax settings:
> fax-relay ecm disable
>  fax rate 9600
>  fax protocol t38 ls-redundancy 2 hs-redundancy 1 fallback 
> pass-through g711alaw
>  no vad
>
> On Audiocodes there is also T38 enabled (as I think) but there is no 
> T38. There are messages on MP-114:
> ErrMgs=16 T38Decoder received Non-T38 Packet
> and
> ||
> Modem Relay Is Not Supported! Forcing Bypass Mode
>
>
> Please, help! What information I need to collect to help solve the problem?
> -- 
> Regards,
> Boris
>
>
>
> _______________________________________________
> 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


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005

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From avi at avimarcus.net  Thu Jul 21 17:11:34 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 21 Jul 2011 16:11:34 +0300
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <CAFGRPVpBFYs1ZcxTnRVhPeLdGJ8HMo1pJGfMhEm3b3_0RWBWtQ@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
	<163DDA0D-0B14-4623-B48B-5E9EEE907AED@gmail.com>
	<CAFYZzbu9Nh3Ojb26G_pAiFn9MVmm59qeO8BL02_WLX5Fj0mfUg@mail.gmail.com>
	<CAKzWOxX2CrDESPMOWMtCgSDxt1R2S3=70aDsTrRM2_p6isfkvA@mail.gmail.com>
	<CAGKOgbK=JzEpWmR8=nFq_x47mFFy96nWRRkMZrogY0hxOMnr2g@mail.gmail.com>
	<CAFGRPVoPVa8ir-18cBLYDzRNftO96Jd=uJ7-xiZQUFsikUwBkA@mail.gmail.com>
	<CAFYZzbuPhHK0MO-SsfHzLO1ZxT=aa9Vjb7OEGtjjSkjBSw7j7g@mail.gmail.com>
	<CALNZuw=CwmjbpqOZrSPJhzMkaNbJ3JAC0e8O+zwXOiJqHJqKvQ@mail.gmail.com>
	<CAFGRPVpBFYs1ZcxTnRVhPeLdGJ8HMo1pJGfMhEm3b3_0RWBWtQ@mail.gmail.com>
Message-ID: <CALNZuwmK0DzZJoYgWBu=fF5ECu14hVGR-+rmzSkoNswAorYzMQ@mail.gmail.com>

If you are having lua "set" the variable, then those are actually SET in the
channel for FreeSWITCH.
Any other lua-specific variables though will not be carried over.

-Avi


On Thu, Jul 21, 2011 at 3:12 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Avi,
>
> I have indeed been looking into the overhead, even though it doesn't
> becomes until very high load. Good idea, I will look into that, will i not
> lose all variables set by the first lua?
>
> thanks for the tip!
>
> David
>
> On Thu, Jul 21, 2011 at 1:39 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> I took a quick look at the Lua, and it looks like you bride within the lua
>> scripts.
>> That means the lua script keeps running during the whole bridge.
>> At scale, you might start to see the overhead as an issue.
>>
>> Instead you can bridge to a custom static extension, e.g.
>> "bridge_callingcard_$number"  that then uses all the variables and completes
>> the bridge, and launches whatever lua scripts needs to be run afterwards
>> from there.
>>
>> -Avi
>>
>> On Wed, Jul 20, 2011 at 2:30 PM, David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>> Hello guys,
>>>>
>>>> I finally set up a git at github. I would like to have a couple testers,
>>>> anyone?
>>>>
>>>> Please bear in mind:
>>>>
>>>> - There's NO installation script. But it is all scripts and web pages,
>>>> etc.
>>>> - The web interface is in spanish, i still need to translate that.
>>>> - The web design is by no means "cool", it is just functional.
>>>> - There will for sure be a thousand bugs, which will need to be fixed.
>>>>
>>>> Contact me whoever's interested.
>>>>
>>>>
>>>> Thanks all
>>>>
>>>> David
>>>>
>>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 david.villasmil.work at gmail.com  Thu Jul 21 17:43:38 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 21 Jul 2011 15:43:38 +0200
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <CALNZuwmK0DzZJoYgWBu=fF5ECu14hVGR-+rmzSkoNswAorYzMQ@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<CAFGRPVrnGq2jyQ40WFuWjina2odM6iVwDL2Pz_9WYQ7i8pHnxg@mail.gmail.com>
	<163DDA0D-0B14-4623-B48B-5E9EEE907AED@gmail.com>
	<CAFYZzbu9Nh3Ojb26G_pAiFn9MVmm59qeO8BL02_WLX5Fj0mfUg@mail.gmail.com>
	<CAKzWOxX2CrDESPMOWMtCgSDxt1R2S3=70aDsTrRM2_p6isfkvA@mail.gmail.com>
	<CAGKOgbK=JzEpWmR8=nFq_x47mFFy96nWRRkMZrogY0hxOMnr2g@mail.gmail.com>
	<CAFGRPVoPVa8ir-18cBLYDzRNftO96Jd=uJ7-xiZQUFsikUwBkA@mail.gmail.com>
	<CAFYZzbuPhHK0MO-SsfHzLO1ZxT=aa9Vjb7OEGtjjSkjBSw7j7g@mail.gmail.com>
	<CALNZuw=CwmjbpqOZrSPJhzMkaNbJ3JAC0e8O+zwXOiJqHJqKvQ@mail.gmail.com>
	<CAFGRPVpBFYs1ZcxTnRVhPeLdGJ8HMo1pJGfMhEm3b3_0RWBWtQ@mail.gmail.com>
	<CALNZuwmK0DzZJoYgWBu=fF5ECu14hVGR-+rmzSkoNswAorYzMQ@mail.gmail.com>
Message-ID: <CAFGRPVrnHL+e378h7XeXjvE2ohdC0GfpoGD4Qns1Wrtx0W27Jg@mail.gmail.com>

Ok, perfect. I assumed as much, since all variables are posted to the cdr
anyway.

thanks Avi

David

On Thu, Jul 21, 2011 at 3:11 PM, Avi Marcus <avi at avimarcus.net> wrote:

> If you are having lua "set" the variable, then those are actually SET in
> the channel for FreeSWITCH.
> Any other lua-specific variables though will not be carried over.
>
> -Avi
>
>
> On Thu, Jul 21, 2011 at 3:12 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Avi,
>>
>> I have indeed been looking into the overhead, even though it doesn't
>> becomes until very high load. Good idea, I will look into that, will i not
>> lose all variables set by the first lua?
>>
>> thanks for the tip!
>>
>> David
>>
>> On Thu, Jul 21, 2011 at 1:39 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> I took a quick look at the Lua, and it looks like you bride within the
>>> lua scripts.
>>> That means the lua script keeps running during the whole bridge.
>>> At scale, you might start to see the overhead as an issue.
>>>
>>> Instead you can bridge to a custom static extension, e.g.
>>> "bridge_callingcard_$number"  that then uses all the variables and completes
>>> the bridge, and launches whatever lua scripts needs to be run afterwards
>>> from there.
>>>
>>> -Avi
>>>
>>> On Wed, Jul 20, 2011 at 2:30 PM, David Villasmil <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>>> Hello guys,
>>>>>
>>>>> I finally set up a git at github. I would like to have a couple
>>>>> testers, anyone?
>>>>>
>>>>> Please bear in mind:
>>>>>
>>>>> - There's NO installation script. But it is all scripts and web pages,
>>>>> etc.
>>>>> - The web interface is in spanish, i still need to translate that.
>>>>> - The web design is by no means "cool", it is just functional.
>>>>> - There will for sure be a thousand bugs, which will need to be fixed.
>>>>>
>>>>> Contact me whoever's interested.
>>>>>
>>>>>
>>>>> Thanks all
>>>>>
>>>>> David
>>>>>
>>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 boris at tagnet.ru  Thu Jul 21 21:17:00 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 21 Jul 2011 23:17:00 +0600
Subject: [Freeswitch-users] Can't get fax working
In-Reply-To: <4E25605D.7050809@tagnet.ru>
References: <4E25605D.7050809@tagnet.ru>
Message-ID: <4E285F0C.6040403@tagnet.ru>

Hello!


     I collected tcpdump between Freeswitch and Audiocodes MP-114 and 
found something strange. The sequence of packets is:
FS  INVITE with SDP
MP 100 Trying
MP 180 Ringing
MP 200 OK with SDP
FS/MP RTP/RTP
MP ReINVITE with SDP
FS 100 Trying
---- long pause and call disconnected
MP BYE
FS 200 OK with SDP !!!!! CSeq: INVITE !!!!!
FS 200 OK CSeq BYE
FS 200 OK with SDP !!!!! CSeq: INVITE !!!!!
FS 200 OK with SDP !!!!! CSeq: INVITE !!!!!
FS 200 OK with SDP !!!!! CSeq: INVITE !!!!!

Thinking this is a problem of T.38 operation. Sometimes, IMHO, 
Freeswitch is buffering??? 200 OK response and session can not be 
established. Where may I place dump for gurus may look at it?

> Hello!
>
>     I need help with fax configuration. My network is:
>
> PSTN E1 --- Cisco AS5350 -- Freeswitch -- Audiocodes MP-114 -- Fax
>
> FS version: FreeSWITCH Version 1.0.head (git-062f07f 2011-07-16 
> 12-35-25 +0200)
> Audiocodes firmware: 6.00A.038.004
> I can't get faxes working :( In FS profiles there is a parameter 
> <param name="t38-passthru" value="true"/>
> Cisco DP fax settings:
> fax-relay ecm disable
>  fax rate 9600
>  fax protocol t38 ls-redundancy 2 hs-redundancy 1 fallback 
> pass-through g711alaw
>  no vad
>
> On Audiocodes there is also T38 enabled (as I think) but there is no 
> T38. There are messages on MP-114:
> ErrMgs=16 T38Decoder received Non-T38 Packet
> and
> ||
> Modem Relay Is Not Supported! Forcing Bypass Mode
>
>
> Please, help! What information I need to collect to help solve the problem?
> -- 
> Regards,
> Boris
>
>
>
> _______________________________________________
> 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


-- 
Regards,
Boris


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From yehavi.bourvine at gmail.com  Thu Jul 21 22:33:16 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 21 Jul 2011 21:33:16 +0300
Subject: [Freeswitch-users] Can't get fax working
In-Reply-To: <4E285F0C.6040403@tagnet.ru>
References: <4E25605D.7050809@tagnet.ru>
	<4E285F0C.6040403@tagnet.ru>
Message-ID: <CAAtWKeG7kv6dKu9BS81cG76nkjbs3Wa6amPMw9U-eve7qBa-cw@mail.gmail.com>

Hello Boris,

  That's exactly the problem I had with my MP-124. If you send me its
configuration file I will look on it next week and compare it to mine.
It is related to T.38 re-invite.

                              __Yehavi:


2011/7/21 Boris Kovalenko <boris at tagnet.ru>

>  Hello!
>
>
>     I collected tcpdump between Freeswitch and Audiocodes MP-114 and found
> something strange. The sequence of packets is:
> FS  INVITE with SDP
> MP 100 Trying
> MP 180 Ringing
> MP 200 OK with SDP
> FS/MP RTP/RTP
> MP ReINVITE with SDP
> FS 100 Trying
> ---- long pause and call disconnected
> MP BYE
> FS 200 OK with SDP !!!!! CSeq: INVITE !!!!!
> FS 200 OK CSeq BYE
> FS 200 OK with SDP !!!!! CSeq: INVITE !!!!!
> FS 200 OK with SDP !!!!! CSeq: INVITE !!!!!
> FS 200 OK with SDP !!!!! CSeq: INVITE !!!!!
>
> Thinking this is a problem of T.38 operation. Sometimes, IMHO, Freeswitch
> is buffering??? 200 OK response and session can not be established. Where
> may I place dump for gurus may look at it?
>
>  Hello!
>
>     I need help with fax configuration. My network is:
>
> PSTN E1 --- Cisco AS5350 -- Freeswitch -- Audiocodes MP-114 -- Fax
>
> FS version: FreeSWITCH Version 1.0.head (git-062f07f 2011-07-16 12-35-25
> +0200)
> Audiocodes firmware: 6.00A.038.004
> I can't get faxes working :( In FS profiles there is a parameter <param
> name="t38-passthru" value="true"/>
> Cisco DP fax settings:
> fax-relay ecm disable
>  fax rate 9600
>  fax protocol t38 ls-redundancy 2 hs-redundancy 1 fallback pass-through
> g711alaw
>  no vad
>
> On Audiocodes there is also T38 enabled (as I think) but there is no T38.
> There are messages on MP-114:
> ErrMgs=16 T38Decoder received Non-T38 Packet
> and
>
> Modem Relay Is Not Supported! Forcing Bypass Mode
>
>
> Please, help! What information I need to collect to help solve the problem?
>
> --
> Regards,
> Boris
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> 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
>
>
>
> --
> Regards,
> Boris
>
>
>
> _______________________________________________
> 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 kris at kriskinc.com  Thu Jul 21 23:44:47 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 21 Jul 2011 15:44:47 -0400
Subject: [Freeswitch-users] Cisco CUCM 6.1 and Freeswitch SIP trunk
In-Reply-To: <CA+ViVc-oY3ZV6AYiDkHeSVJucjm=pLnha0=gW4P2A91Ux=TUSQ@mail.gmail.com>
References: <CA+ViVc99bqe2e8AAy-ROxLA_LPumQv7q50ygPNQjf_KMzypKAg@mail.gmail.com>
	<CAKDfjgd0R0Sb_BYpmyHLjmv-oPajQ_uw_5ev2ioPDrXXM_3qpw@mail.gmail.com>
	<80E395DA-B3F2-42CE-AB9F-098096B32B45@gmail.com>
	<CAD-qRkjh2-St09KcqbMEA=muQw5tzHckV=TVVMRdFHLYqLr6OQ@mail.gmail.com>
	<CA+ViVc9ydZfjWwTVVkDxRugAC+s4iV9TPRLEU3v=h7emfnf98Q@mail.gmail.com>
	<CAKDfjgfh2162ysaDc3-91-qNR4wV9R5K-Ds-OtSYOXdgiDZD9w@mail.gmail.com>
	<CA+ViVc_U+iEifVcwb=n=keR4EO8rTV-SGFEu7XO0q=NSzykGqw@mail.gmail.com>
	<CA+ViVc-oY3ZV6AYiDkHeSVJucjm=pLnha0=gW4P2A91Ux=TUSQ@mail.gmail.com>
Message-ID: <CAKDfjgczmND40J-ygWUCsJ64sSb+USenco87MFmm9mMTRGmRNg@mail.gmail.com>

The issue now is CUCM never sends an SDP.  Not in the INVITE, not in
the ACK.  If the INVITE contained an SDP you'd be fine with the
default configuration.  If it didn't but the ACK did you'd be fine
with 3pcc enabled.

Right now CUCM isn't providing either.  That's your problem, it's
somewhere on the Cisco side, and I've never seen it before.
Unfortunately I don't have any specific recommendations for you.

On Thu, Jul 21, 2011 at 8:46 AM, David Martin <adavidm at gmail.com> wrote:
> For completeness, here is the INVITE with "Disable Early Media" unchecked:
> ? ?------------------------------------------------------------------------
> ? ?INVITE sip:65131111 at 10.201.27.20:5060 SIP/2.0
> ? ?Date: Thu, 21 Jul 2011 12:42:02 GMT
> ? ?Call-Info:
> <sip:10.200.170.3:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
> ? ?Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY, PUBLISH
> ? ?From: "Ryan Girdlestone"
> <sip:64105554 at 10.200.170.3>;tag=90509204-400f-4cb6-9be7-cb5410a83a11-54376643
> ? ?Allow-Events: presence, kpml
> ? ?Supported: timer,replaces
> ? ?Min-SE: ?1800
> ? ?Remote-Party-ID: "Ryan Girdlestone"
> <sip:64105554 at 10.200.170.3>;party=calling;screen=yes;privacy=off
> ? ?Content-Length: 0
> ? ?User-Agent: Cisco-CUCM6.1
> ? ?To: <sip:65131111 at 10.201.27.20>
> ? ?Contact: <sip:64105554 at 10.200.170.3:5060>
> ? ?Expires: 180
> ? ?Call-ID: d3b6ce80-e2811e9a-33-3aac80a at 10.200.170.3
> ? ?Via: SIP/2.0/UDP 10.200.170.3:5060;branch=z9hG4bK4d55790bf1
> ? ?CSeq: 101 INVITE
> ? ?Session-Expires: ?1800
> ? ?Max-Forwards: 70
>
>
>
> On 21 July 2011 13:10, David Martin <adavidm at gmail.com> wrote:
>>
>> Kristian,
>> Here are some screenshots from the trunk config.
>> http://img200.imageshack.us/img200/9405/siptrunk.png
>> and for the sip profile
>> http://img191.imageshack.us/img191/3863/sipprofile.png
>> I have tried the early media option both ways with the same result. Is the
>> MTP payload type correct?
>> Appreciate?your help on this.
>> David
>>
>> On 21 July 2011 11:13, Kristian Kielhofner <kris at kriskinc.com> wrote:
>>>
>>> David,
>>>
>>> ?The problem here is that even with 3pcc enabled the ACK from
>>> CallManager doesn't have an SDP. ?CallManager never provides an SDP
>>> during this entire trace. ?Not in the INVITE (preferred) and not in
>>> the ACK (as I already mentioned).
>>>
>>> ?This is odd, even for CallManager. ?Are you sure you have MTP enabled:
>>>
>>> http://blog.krisk.org/2010/05/another-sip-gotcha-cisco.html
>>>
>>> On Thu, Jul 21, 2011 at 3:45 AM, David Martin <adavidm at gmail.com> wrote:
>>> > Gotcha, not sure how I missed the INVITE the first time, but here is a
>>> > new
>>> > pastebin that should have everything:
>>> > http://pastebin.com/T3eqms8N
>>> > thanks in advance once again.
>>> > David
>>> >
>>> >
>>> > On 21 July 2011 02:19, Christopher Rienzo <cmrienzo at gmail.com> wrote:
>>> >>
>>> >> Line 7 is the 200 OK response to the INVITE.
>>> >>
>>> >>
>>> >> On Wed, Jul 20, 2011 at 6:54 PM, David Martin <adavidm at gmail.com>
>>> >> wrote:
>>> >>>
>>> >>> Forgive my confusion, I'm new to freeswitch. Does the invite not
>>> >>> start at
>>> >>> line 7 on the pastebin?
>>> >>>
>>> >>> David
>>> >>>
>>> >>> Sent from my iPhone
>>> >>>
>>> >>> On 20 Jul 2011, at 07:22 PM, Kristian Kielhofner <kris at kriskinc.com>
>>> >>> wrote:
>>> >>>
>>> >>> > That trace is missing the INVITE that starts the transaction.
>>> >>> >
>>> >>> > On Wed, Jul 20, 2011 at 11:48 AM, David Martin <adavidm at gmail.com>
>>> >>> > wrote:
>>> >>> >> All,
>>> >>> >> Firstly apologies if this is not the correct place to ask, but I
>>> >>> >> am
>>> >>> >> having
>>> >>> >> problems getting CUCM 6.1 to talk with freeswitch over a SIP trunk
>>> >>> >> (I
>>> >>> >> only
>>> >>> >> need inbound from Cisco -> Freeswitch). I have enabled 3pcc, but
>>> >>> >> am
>>> >>> >> getting
>>> >>> >> the following sofia trace:
>>> >>> >> http://pastebin.com/g9VmvxFe
>>> >>> >> Can anyone point me in the right direction? I have also tried
>>> >>> >> enabling
>>> >>> >> MTP
>>> >>> >> on the Cisco side, but this does not make any difference.
>>> >>> >> Thanks in advance.
>>> >>> >> David
>>> >>> >> _______________________________________________
>>> >>> >> 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
>>> >>> >>
>>> >>> >>
>>> >>> >
>>> >>> >
>>> >>> >
>>> >>> > --
>>> >>> > Kristian Kielhofner
>>> >>> >
>>> >>> > _______________________________________________
>>> >>> > 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
>>> >>>
>>> >>> _______________________________________________
>>> >>> 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
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> 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
>>> >>
>>> >
>>> >
>>> > _______________________________________________
>>> > 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
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Kristian Kielhofner
>>>
>>> _______________________________________________
>>> 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
>>
>
>
> _______________________________________________
> 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
>
>



-- 
Kristian Kielhofner


From adavidm at gmail.com  Fri Jul 22 02:28:38 2011
From: adavidm at gmail.com (David Martin)
Date: Thu, 21 Jul 2011 23:28:38 +0100
Subject: [Freeswitch-users] Cisco CUCM 6.1 and Freeswitch SIP trunk
In-Reply-To: <CAKDfjgczmND40J-ygWUCsJ64sSb+USenco87MFmm9mMTRGmRNg@mail.gmail.com>
References: <CA+ViVc99bqe2e8AAy-ROxLA_LPumQv7q50ygPNQjf_KMzypKAg@mail.gmail.com>
	<CAKDfjgd0R0Sb_BYpmyHLjmv-oPajQ_uw_5ev2ioPDrXXM_3qpw@mail.gmail.com>
	<80E395DA-B3F2-42CE-AB9F-098096B32B45@gmail.com>
	<CAD-qRkjh2-St09KcqbMEA=muQw5tzHckV=TVVMRdFHLYqLr6OQ@mail.gmail.com>
	<CA+ViVc9ydZfjWwTVVkDxRugAC+s4iV9TPRLEU3v=h7emfnf98Q@mail.gmail.com>
	<CAKDfjgfh2162ysaDc3-91-qNR4wV9R5K-Ds-OtSYOXdgiDZD9w@mail.gmail.com>
	<CA+ViVc_U+iEifVcwb=n=keR4EO8rTV-SGFEu7XO0q=NSzykGqw@mail.gmail.com>
	<CA+ViVc-oY3ZV6AYiDkHeSVJucjm=pLnha0=gW4P2A91Ux=TUSQ@mail.gmail.com>
	<CAKDfjgczmND40J-ygWUCsJ64sSb+USenco87MFmm9mMTRGmRNg@mail.gmail.com>
Message-ID: <FAAAC9F6-ACCE-4A17-BCEE-EED2FE2DAE1A@gmail.com>

Thanks for your assistance. I might try mod_opal and use h323 instead then.

Thanks again.

David

Sent from my iPad

On 21 Jul 2011, at 20:44, Kristian Kielhofner <kris at kriskinc.com> wrote:

> The issue now is CUCM never sends an SDP.  Not in the INVITE, not in
> the ACK.  If the INVITE contained an SDP you'd be fine with the
> default configuration.  If it didn't but the ACK did you'd be fine
> with 3pcc enabled.
> 
> Right now CUCM isn't providing either.  That's your problem, it's
> somewhere on the Cisco side, and I've never seen it before.
> Unfortunately I don't have any specific recommendations for you.
> 
> On Thu, Jul 21, 2011 at 8:46 AM, David Martin <adavidm at gmail.com> wrote:
>> For completeness, here is the INVITE with "Disable Early Media" unchecked:
>>    ------------------------------------------------------------------------
>>    INVITE sip:65131111 at 10.201.27.20:5060 SIP/2.0
>>    Date: Thu, 21 Jul 2011 12:42:02 GMT
>>    Call-Info:
>> <sip:10.200.170.3:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
>>    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY, PUBLISH
>>    From: "Ryan Girdlestone"
>> <sip:64105554 at 10.200.170.3>;tag=90509204-400f-4cb6-9be7-cb5410a83a11-54376643
>>    Allow-Events: presence, kpml
>>    Supported: timer,replaces
>>    Min-SE:  1800
>>    Remote-Party-ID: "Ryan Girdlestone"
>> <sip:64105554 at 10.200.170.3>;party=calling;screen=yes;privacy=off
>>    Content-Length: 0
>>    User-Agent: Cisco-CUCM6.1
>>    To: <sip:65131111 at 10.201.27.20>
>>    Contact: <sip:64105554 at 10.200.170.3:5060>
>>    Expires: 180
>>    Call-ID: d3b6ce80-e2811e9a-33-3aac80a at 10.200.170.3
>>    Via: SIP/2.0/UDP 10.200.170.3:5060;branch=z9hG4bK4d55790bf1
>>    CSeq: 101 INVITE
>>    Session-Expires:  1800
>>    Max-Forwards: 70
>> 
>> 
>> 
>> On 21 July 2011 13:10, David Martin <adavidm at gmail.com> wrote:
>>> 
>>> Kristian,
>>> Here are some screenshots from the trunk config.
>>> http://img200.imageshack.us/img200/9405/siptrunk.png
>>> and for the sip profile
>>> http://img191.imageshack.us/img191/3863/sipprofile.png
>>> I have tried the early media option both ways with the same result. Is the
>>> MTP payload type correct?
>>> Appreciate your help on this.
>>> David
>>> 
>>> On 21 July 2011 11:13, Kristian Kielhofner <kris at kriskinc.com> wrote:
>>>> 
>>>> David,
>>>> 
>>>>  The problem here is that even with 3pcc enabled the ACK from
>>>> CallManager doesn't have an SDP.  CallManager never provides an SDP
>>>> during this entire trace.  Not in the INVITE (preferred) and not in
>>>> the ACK (as I already mentioned).
>>>> 
>>>>  This is odd, even for CallManager.  Are you sure you have MTP enabled:
>>>> 
>>>> http://blog.krisk.org/2010/05/another-sip-gotcha-cisco.html
>>>> 
>>>> On Thu, Jul 21, 2011 at 3:45 AM, David Martin <adavidm at gmail.com> wrote:
>>>>> Gotcha, not sure how I missed the INVITE the first time, but here is a
>>>>> new
>>>>> pastebin that should have everything:
>>>>> http://pastebin.com/T3eqms8N
>>>>> thanks in advance once again.
>>>>> David
>>>>> 
>>>>> 
>>>>> On 21 July 2011 02:19, Christopher Rienzo <cmrienzo at gmail.com> wrote:
>>>>>> 
>>>>>> Line 7 is the 200 OK response to the INVITE.
>>>>>> 
>>>>>> 
>>>>>> On Wed, Jul 20, 2011 at 6:54 PM, David Martin <adavidm at gmail.com>
>>>>>> wrote:
>>>>>>> 
>>>>>>> Forgive my confusion, I'm new to freeswitch. Does the invite not
>>>>>>> start at
>>>>>>> line 7 on the pastebin?
>>>>>>> 
>>>>>>> David
>>>>>>> 
>>>>>>> Sent from my iPhone
>>>>>>> 
>>>>>>> On 20 Jul 2011, at 07:22 PM, Kristian Kielhofner <kris at kriskinc.com>
>>>>>>> wrote:
>>>>>>> 
>>>>>>>> That trace is missing the INVITE that starts the transaction.
>>>>>>>> 
>>>>>>>> On Wed, Jul 20, 2011 at 11:48 AM, David Martin <adavidm at gmail.com>
>>>>>>>> wrote:
>>>>>>>>> All,
>>>>>>>>> Firstly apologies if this is not the correct place to ask, but I
>>>>>>>>> am
>>>>>>>>> having
>>>>>>>>> problems getting CUCM 6.1 to talk with freeswitch over a SIP trunk
>>>>>>>>> (I
>>>>>>>>> only
>>>>>>>>> need inbound from Cisco -> Freeswitch). I have enabled 3pcc, but
>>>>>>>>> am
>>>>>>>>> getting
>>>>>>>>> the following sofia trace:
>>>>>>>>> http://pastebin.com/g9VmvxFe
>>>>>>>>> Can anyone point me in the right direction? I have also tried
>>>>>>>>> enabling
>>>>>>>>> MTP
>>>>>>>>> on the Cisco side, but this does not make any difference.
>>>>>>>>> Thanks in advance.
>>>>>>>>> David
>>>>>>>>> _______________________________________________
>>>>>>>>> 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
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> --
>>>>>>>> Kristian Kielhofner
>>>>>>>> 
>>>>>>>> _______________________________________________
>>>>>>>> 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
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 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
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 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
>>>>>> 
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> 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
>>>>> 
>>>>> 
>>>> 
>>>> 
>>>> 
>>>> --
>>>> Kristian Kielhofner
>>>> 
>>>> _______________________________________________
>>>> 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
>>> 
>> 
>> 
>> _______________________________________________
>> 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
>> 
>> 
> 
> 
> 
> -- 
> Kristian Kielhofner
> 
> _______________________________________________
> 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 kheimerl at cs.berkeley.edu  Fri Jul 22 02:47:55 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Thu, 21 Jul 2011 15:47:55 -0700
Subject: [Freeswitch-users] Using chat command in a dialplan
In-Reply-To: <CAFiqYumBRYwpNamX_ig525Hbo+v+s+uVZZ8httTu7Vd7LYk9Qg@mail.gmail.com>
References: <CACyT-3m7nMTCwav_j_CVqtL-V+T9vFcyGwsrD6uatJ+bkVti3Q@mail.gmail.com>
	<1311149680.32170.264.camel@luna.madrid.commsmundi.com>
	<83217555-012B-4861-AAB9-045B2E78660D@gmail.com>
	<CACyT-3mASOUzN4kX_L4X734Jp1wL5nTbpi-Ytd4TuF-nJA3xaw@mail.gmail.com>
	<CACyT-3nBYP88Ve3oTC6R1WYSPKZkvY08iVTuSxfmm9miFgYEtA@mail.gmail.com>
	<CAFiqYumBRYwpNamX_ig525Hbo+v+s+uVZZ8httTu7Vd7LYk9Qg@mail.gmail.com>
Message-ID: <CACyT-3=9J+Saqi_c6KuNAjNWyZpgQF+ao3GuUZhLUUD4mw+hog@mail.gmail.com>

Unfortunately, I am not. I'm working on a system that causes dialplan
actions when a chat is received by FS. That's what's going on here,
basically. I want someone to text 9199 and have FS immediately call
them back. I think "originate" is the right way to do that, as I need
to create a brand new session, not bridge to an existing one.

Anyhow, any other ideas on why this isn't working?

On Thu, Jul 21, 2011 at 1:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
> Are you trying to join the 2 calls together? If so, use bridge:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge
>
> -Steve
>
>
> On 21 July 2011 07:37, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>>
>> I'm now having a similar problem with using "originate" to create a
>> new call inside a dialplan. This one is particularly perplexing (and
>> small enough not to warrant pastebin)
>>
>> Here's the important bits of the dialplan:
>> ? ? ? ?<action application="log" data="(user/${username}
>> ${destination_number})"/>
>> ? ? ? ?<action application="set"
>> data="api_result=${originate(user/${username}
>> ${destination_number})}"/>
>>
>> and the log output
>> EXECUTE 9199 log((user/1304 at 192.168.1.144 9199))
>> 2011-07-20 23:17:12.955419 [DEBUG] mod_dptools.c:1202 9199)
>> EXECUTE 9199 set(api_result=-USAGE <call url>
>> <exten>|&<application_name>(<app_args>) [<dialplan>] [<context>]
>> [<cid_name>] [<cid_num>] [<timeout_sec>]
>> )
>>
>> So it's somehow mangling the originate commands. If I run the
>> originate command directly (with the arguments logged!) it works just
>> fine.
>> originate user/1304 at 192.168.1.144 9199
>>
>> What am I missing here? This seems so simple... ?but I'm at my wit's end.
>>
>> Thanks for any direction in advance!
>>
>>
>> On Wed, Jul 20, 2011 at 7:11 PM, Kurtis Heimerl
>> <kheimerl at cs.berkeley.edu> wrote:
>> > Exactly what I was looking for. Thanks!
>> >
>> > On Wed, Jul 20, 2011 at 1:21 AM, Steven Ayre <steveayre at gmail.com>
>> > wrote:
>> >> Kurtis,
>> >>
>> >> *Any* api can be run from the dialplan. It's just a slightly different
>> >> syntax - the link Francois gave you will show you how.
>> >>
>> >> Steve on iPhone
>> >>
>> >> On 20 Jul 2011, at 09:14, Fran?ois Delawarde
>> >> <fdelawarde at wirelessmundi.com> wrote:
>> >>
>> >>> Hi,
>> >>>
>> >>> Take a look at:
>> >>> http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
>> >>>
>> >>>
>> >>> Fran?ois.
>> >>>
>> >>> On Tue, 2011-07-19 at 23:00 -0700, Kurtis Heimerl wrote:
>> >>>> I feel somewhat embarrassed, but I can't figure out how to cause a
>> >>>> chat event to go off from the dialplan. I've tried:
>> >>>>
>> >>>> <action application="chat"
>> >>>>
>> >>>> data="sip|${username}|${destination_number}@${domain_name}|${msg_body}"/>
>> >>>>
>> >>>> and
>> >>>>
>> >>>> <action application="eval" data="chat
>> >>>> sip|${username}|${destination_number}@${domain_name}|${msg_body}"/>
>> >>>>
>> >>>> and neither worked. The wiki says it's part of the API
>> >>>> (http://wiki.freeswitch.org/wiki/Mod_dptools) but I'm pretty sure
>> >>>> that's just a command-line API. Is there a way to run a command-line
>> >>>> call from the dialplan?
>> >>>>
>> >>>> Any direction would be appreciated.
>> >>>>
>> >>>> _______________________________________________
>> >>>> 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
>> >>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> 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
>> >>
>> >> _______________________________________________
>> >> 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
>> >>
>> >
>>
>> _______________________________________________
>> 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
>
>
> _______________________________________________
> 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 lakersman2006 at yahoo.com  Fri Jul 22 05:10:04 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Thu, 21 Jul 2011 18:10:04 -0700 (PDT)
Subject: [Freeswitch-users] How to get call bridge status?
Message-ID: <1311297004.51793.YahooMailClassic@web161007.mail.bf1.yahoo.com>

I am writing a perl script that does a call bridge and I would like to know what is the status of the call bridge once the bridge application as finished execution. I am using the following but do not know if it is the correct way since on NO_ANSWER call states the sip code is blank, but shouldn't it return SIP:480 according to this http://wiki.freeswitch.org/wiki/Hangup_causes?

$sipcode = $session->getVariable("proto_specific_hangup_cause");
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From boris at tagnet.ru  Fri Jul 22 07:36:26 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 22 Jul 2011 09:36:26 +0600
Subject: [Freeswitch-users] Can't get fax working
In-Reply-To: <CAAtWKeG7kv6dKu9BS81cG76nkjbs3Wa6amPMw9U-eve7qBa-cw@mail.gmail.com>
References: <4E25605D.7050809@tagnet.ru> <4E285F0C.6040403@tagnet.ru>
	<CAAtWKeG7kv6dKu9BS81cG76nkjbs3Wa6amPMw9U-eve7qBa-cw@mail.gmail.com>
Message-ID: <4E28F03A.6030808@tagnet.ru>

Hello!

     This is one of I tried.

> Hello Boris,
>   That's exactly the problem I had with my MP-124. If you send me its 
> configuration file I will look on it next week and compare it to mine.
> It is related to T.38 re-invite.
>                               __Yehavi:
>
>
> 2011/7/21 Boris Kovalenko <boris at tagnet.ru <mailto:boris at tagnet.ru>>
>
>     Hello!
>
>
>         I collected tcpdump between Freeswitch and Audiocodes MP-114
>     and found something strange. The sequence of packets is:
>     FS  INVITE with SDP
>     MP 100 Trying
>     MP 180 Ringing
>     MP 200 OK with SDP
>     FS/MP RTP/RTP
>     MP ReINVITE with SDP
>     FS 100 Trying
>     ---- long pause and call disconnected
>     MP BYE
>     FS 200 OK with SDP !!!!! CSeq: INVITE !!!!!
>     FS 200 OK CSeq BYE
>     FS 200 OK with SDP !!!!! CSeq: INVITE !!!!!
>     FS 200 OK with SDP !!!!! CSeq: INVITE !!!!!
>     FS 200 OK with SDP !!!!! CSeq: INVITE !!!!!
>
>     Thinking this is a problem of T.38 operation. Sometimes, IMHO,
>     Freeswitch is buffering??? 200 OK response and session can not be
>     established. Where may I place dump for gurus may look at it?
>
>>     Hello!
>>
>>         I need help with fax configuration. My network is:
>>
>>     PSTN E1 --- Cisco AS5350 -- Freeswitch -- Audiocodes MP-114 -- Fax
>>
>>     FS version: FreeSWITCH Version 1.0.head (git-062f07f 2011-07-16
>>     12-35-25 +0200)
>>     Audiocodes firmware: 6.00A.038.004
>>     I can't get faxes working :( In FS profiles there is a parameter
>>     <param name="t38-passthru" value="true"/>
>>     Cisco DP fax settings:
>>     fax-relay ecm disable
>>      fax rate 9600
>>      fax protocol t38 ls-redundancy 2 hs-redundancy 1 fallback
>>     pass-through g711alaw
>>      no vad
>>
>>     On Audiocodes there is also T38 enabled (as I think) but there is
>>     no T38. There are messages on MP-114:
>>     ErrMgs=16 T38Decoder received Non-T38 Packet
>>     and
>>     ||
>>     Modem Relay Is Not Supported! Forcing Bypass Mode
>>
>>
>>     Please, help! What information I need to collect to help solve the problem?
>>     -- 
>>     Regards,
>>     Boris
>>
>>
>>
>>     _______________________________________________
>>     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  <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
>
>
>     -- 
>     Regards,
>     Boris
>
>
>
>     _______________________________________________
>     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
>     <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
>
>
>
>
> _______________________________________________
> 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


-- 
Regards,
Boris


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From kbdfck at gmail.com  Fri Jul 22 11:24:57 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Fri, 22 Jul 2011 11:24:57 +0400
Subject: [Freeswitch-users] DTMF conversion / regeneration with
	bind_digit_action ?
Message-ID: <CA+GmxbYDfzr8TGhsgQJW-z2W1T60+7TAv63gD054ArQGZW9eYQ@mail.gmail.com>

Hi All

DTMF troubles again :(

We are trying to use Linksys  PAP2T / SPA8000 with FS on internal profile
and Audiocodes Mediant 2000 as gateway on external. When ATAs set to RFC2833
and proxy media is set on both profiles, everything works fine. We get
RFC2833 on both sides.
If we turn proxy media off to use freeswitch DTMF features like attended
transfer by *7 or another in-call DTMF features, things go worse.

PAP2T/SPA8000 convert inband DTMF from phone to RFC2833, but there are some
inband clips still coming into audio channel. Freeswitch processes RFC2833
as dtmf-type set to 2833 in internal profile, but since we use
bind_digit_action, little inband clip of DTMF gets to other side earlier
than regenerated 2833, which is delayed by FS in bind_digit_action /
bind_meta_app. BTW, bind_meta_app introduces less delay than
bind_digit_action.

As the result, after media goes through mediant2000, on PSTN side we have
double DTMF - little inband clip and generated RFC2833 with really big gap
between them. If some local transfers has been done via internal sip
profile, more channels are linked in chain, so delay gets even bigger.

When we set DTMF to inband on ATAs and internal profile dtmf-type is set to
inband, freeswitch detects inband and in-call features work, but after
passing inband DTMF through, FS for some reason also generates RFC2833 on
external profile, and then Mediant passes inband part and re-generates
rfc2833 DTMF part to PSTN. Again, double DTMF on PSTN side :(

Can inband <-> rfc2833 conversion be done on FS side with inband DTMF
suppression? And what can be done with in-call DTMF detection functions to
make  FS pass rfc2833 through as fast as it receives it without delay to
make no gap between inband clips and regenerated DTMF?

I really appreciate any help. I'm trying to handle this in different ways
for two month with no result :(


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From steveayre at gmail.com  Fri Jul 22 11:58:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 22 Jul 2011 08:58:01 +0100
Subject: [Freeswitch-users] How to get call bridge status?
In-Reply-To: <1311297004.51793.YahooMailClassic@web161007.mail.bf1.yahoo.com>
References: <1311297004.51793.YahooMailClassic@web161007.mail.bf1.yahoo.com>
Message-ID: <CAFiqYu=MNHUALvEjO9Db4ng_JQi8_2PjHrOQpx0KkUOQ33ZwZA@mail.gmail.com>

What do you get in bridge_hangup_cause?

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

-Steve



On 22 July 2011 02:10, Sam <lakersman2006 at yahoo.com> wrote:

> I am writing a perl script that does a call bridge and I would like to know
> what is the status of the call bridge once the bridge application as
> finished execution. I am using the following but do not know if it is the
> correct way since on NO_ANSWER call states the sip code is blank, but
> shouldn't it return SIP:480 according to this
> http://wiki.freeswitch.org/wiki/Hangup_causes?
>
> $sipcode = $session->getVariable("proto_specific_hangup_cause");
>
> _______________________________________________
> 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 brokendash at gmail.com  Fri Jul 22 12:04:28 2011
From: brokendash at gmail.com (broken dash)
Date: Fri, 22 Jul 2011 03:04:28 -0500
Subject: [Freeswitch-users] Pizza Demo Grammar
In-Reply-To: <C35D5B41-1661-462A-9621-84D11DE60B28@freeswitch.org>
References: <45543322-DEDF-4125-ADB9-EABA63C446C5@bryansmart.com>
	<C35D5B41-1661-462A-9621-84D11DE60B28@freeswitch.org>
Message-ID: <CABNzcD8YJ-QhR6reQUGhVse3JeOkLfpi1yBeVrCq-_RaqSFRcQ@mail.gmail.com>

uhm, can you  attach the files to this thread? or post them in txt
form to the thread or something.

B


On Wed, Jul 20, 2011 at 7:19 PM, Brian West <brian at freeswitch.org> wrote:
> Yah my box died this morning. ?So i'm in need of a new one. ?Anyone wish to donate to my new linux box cause? ?paypal brian at bkw.org ? I think the hard drive is intact so i'll get the data back up as soon as I get a new box.
>
> Thanks,
> /b
>
> On Jul 20, 2011, at 7:17 PM, Bryan Smart wrote:
>
>> Hi.
>>
>> Trying to install the Pizza demo.
>>
>> The Wiki says to download the grammar from www.bkw.org, but that entire site seems to be down.
>>
>> Are these files available somewhere else? I tried to find them with Google, but only found references to the Wiki page and references to it in FS updates.
>>
>> Thanks
>> Bryan
>>
>>
>> _______________________________________________
>> 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
>
>
> _______________________________________________
> 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 steveayre at gmail.com  Fri Jul 22 12:04:08 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 22 Jul 2011 09:04:08 +0100
Subject: [Freeswitch-users] DTMF conversion / regeneration with
 bind_digit_action ?
In-Reply-To: <CA+GmxbYDfzr8TGhsgQJW-z2W1T60+7TAv63gD054ArQGZW9eYQ@mail.gmail.com>
References: <CA+GmxbYDfzr8TGhsgQJW-z2W1T60+7TAv63gD054ArQGZW9eYQ@mail.gmail.com>
Message-ID: <CAFiqYuk1smWYJ1Z3=yrk11sqxeQvi97-DOLwH9sLv8P5UavVqA@mail.gmail.com>

>
> PAP2T/SPA8000 convert inband DTMF from phone to RFC2833, but there are some
> inband clips still coming into audio channel.
>

Which is normal for any device... they generate RFC2833 digits but leave the
inband DTMF intact.



> Can inband <-> rfc2833 conversion be done on FS side with inband DTMF
> suppression
>

No. It detects the tones but doesn't remove them.


Freeswitch processes RFC2833 as dtmf-type set to 2833 in internal profile,
> but since we use bind_digit_action, little inband clip of DTMF gets to other
> side earlier than regenerated 2833
>

FreeSWITCH should only detect inband DTMF if you're calling the start_dtmf
app. Are you running that app? If so, don't. It's not needed for out-of-band
DTMF such as RFC2833, only for detecting inband DTMF. If you're running it
on a channel that's getting both inband and RFC2833 digits you'll get
duplicate digits.


FS for some reason also generates RFC2833 on external profile
>

What is the dtmf-type on the external profile? If that's rfc2833 it'll be
generating that on the outbound leg even though it's set inband on the
receiving profile.


-Steve



On 22 July 2011 08:24, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> Hi All
>
> DTMF troubles again :(
>
> We are trying to use Linksys  PAP2T / SPA8000 with FS on internal profile
> and Audiocodes Mediant 2000 as gateway on external. When ATAs set to RFC2833
> and proxy media is set on both profiles, everything works fine. We get
> RFC2833 on both sides.
> If we turn proxy media off to use freeswitch DTMF features like attended
> transfer by *7 or another in-call DTMF features, things go worse.
>
> PAP2T/SPA8000 convert inband DTMF from phone to RFC2833, but there are some
> inband clips still coming into audio channel. Freeswitch processes RFC2833
> as dtmf-type set to 2833 in internal profile, but since we use
> bind_digit_action, little inband clip of DTMF gets to other side earlier
> than regenerated 2833, which is delayed by FS in bind_digit_action /
> bind_meta_app. BTW, bind_meta_app introduces less delay than
> bind_digit_action.
>
> As the result, after media goes through mediant2000, on PSTN side we have
> double DTMF - little inband clip and generated RFC2833 with really big gap
> between them. If some local transfers has been done via internal sip
> profile, more channels are linked in chain, so delay gets even bigger.
>
> When we set DTMF to inband on ATAs and internal profile dtmf-type is set to
> inband, freeswitch detects inband and in-call features work, but after
> passing inband DTMF through, FS for some reason also generates RFC2833 on
> external profile, and then Mediant passes inband part and re-generates
> rfc2833 DTMF part to PSTN. Again, double DTMF on PSTN side :(
>
> Can inband <-> rfc2833 conversion be done on FS side with inband DTMF
> suppression? And what can be done with in-call DTMF detection functions to
> make  FS pass rfc2833 through as fast as it receives it without delay to
> make no gap between inband clips and regenerated DTMF?
>
> I really appreciate any help. I'm trying to handle this in different ways
> for two month with no result :(
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> 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 steveayre at gmail.com  Fri Jul 22 12:05:45 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 22 Jul 2011 09:05:45 +0100
Subject: [Freeswitch-users] Using chat command in a dialplan
In-Reply-To: <CACyT-3=9J+Saqi_c6KuNAjNWyZpgQF+ao3GuUZhLUUD4mw+hog@mail.gmail.com>
References: <CACyT-3m7nMTCwav_j_CVqtL-V+T9vFcyGwsrD6uatJ+bkVti3Q@mail.gmail.com>
	<1311149680.32170.264.camel@luna.madrid.commsmundi.com>
	<83217555-012B-4861-AAB9-045B2E78660D@gmail.com>
	<CACyT-3mASOUzN4kX_L4X734Jp1wL5nTbpi-Ytd4TuF-nJA3xaw@mail.gmail.com>
	<CACyT-3nBYP88Ve3oTC6R1WYSPKZkvY08iVTuSxfmm9miFgYEtA@mail.gmail.com>
	<CAFiqYumBRYwpNamX_ig525Hbo+v+s+uVZZ8httTu7Vd7LYk9Qg@mail.gmail.com>
	<CACyT-3=9J+Saqi_c6KuNAjNWyZpgQF+ao3GuUZhLUUD4mw+hog@mail.gmail.com>
Message-ID: <CAFiqYunELHt2ti+x4OBO=osyrjqn-HQRKhfrjeJV6Mvjpo87Kg@mail.gmail.com>

Never tried it from a dialplan although I'm sure it should work. But you
should be able to do it through Lua etc.

Maybe you could apply a quick patch to originate in mod_commands to log the
arguments it's being passed - perhaps it's being mangled somehow.

-Steve


On 21 July 2011 23:47, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:

> Unfortunately, I am not. I'm working on a system that causes dialplan
> actions when a chat is received by FS. That's what's going on here,
> basically. I want someone to text 9199 and have FS immediately call
> them back. I think "originate" is the right way to do that, as I need
> to create a brand new session, not bridge to an existing one.
>
> Anyhow, any other ideas on why this isn't working?
>
> On Thu, Jul 21, 2011 at 1:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
> > Are you trying to join the 2 calls together? If so, use bridge:
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge
> >
> > -Steve
> >
> >
> > On 21 July 2011 07:37, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
> >>
> >> I'm now having a similar problem with using "originate" to create a
> >> new call inside a dialplan. This one is particularly perplexing (and
> >> small enough not to warrant pastebin)
> >>
> >> Here's the important bits of the dialplan:
> >>        <action application="log" data="(user/${username}
> >> ${destination_number})"/>
> >>        <action application="set"
> >> data="api_result=${originate(user/${username}
> >> ${destination_number})}"/>
> >>
> >> and the log output
> >> EXECUTE 9199 log((user/1304 at 192.168.1.144 9199))
> >> 2011-07-20 23:17:12.955419 [DEBUG] mod_dptools.c:1202 9199)
> >> EXECUTE 9199 set(api_result=-USAGE <call url>
> >> <exten>|&<application_name>(<app_args>) [<dialplan>] [<context>]
> >> [<cid_name>] [<cid_num>] [<timeout_sec>]
> >> )
> >>
> >> So it's somehow mangling the originate commands. If I run the
> >> originate command directly (with the arguments logged!) it works just
> >> fine.
> >> originate user/1304 at 192.168.1.144 9199
> >>
> >> What am I missing here? This seems so simple...  but I'm at my wit's
> end.
> >>
> >> Thanks for any direction in advance!
> >>
> >>
> >> On Wed, Jul 20, 2011 at 7:11 PM, Kurtis Heimerl
> >> <kheimerl at cs.berkeley.edu> wrote:
> >> > Exactly what I was looking for. Thanks!
> >> >
> >> > On Wed, Jul 20, 2011 at 1:21 AM, Steven Ayre <steveayre at gmail.com>
> >> > wrote:
> >> >> Kurtis,
> >> >>
> >> >> *Any* api can be run from the dialplan. It's just a slightly
> different
> >> >> syntax - the link Francois gave you will show you how.
> >> >>
> >> >> Steve on iPhone
> >> >>
> >> >> On 20 Jul 2011, at 09:14, Fran?ois Delawarde
> >> >> <fdelawarde at wirelessmundi.com> wrote:
> >> >>
> >> >>> Hi,
> >> >>>
> >> >>> Take a look at:
> >> >>> http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
> >> >>>
> >> >>>
> >> >>> Fran?ois.
> >> >>>
> >> >>> On Tue, 2011-07-19 at 23:00 -0700, Kurtis Heimerl wrote:
> >> >>>> I feel somewhat embarrassed, but I can't figure out how to cause a
> >> >>>> chat event to go off from the dialplan. I've tried:
> >> >>>>
> >> >>>> <action application="chat"
> >> >>>>
> >> >>>> data="sip|${username}|${destination_number}@
> ${domain_name}|${msg_body}"/>
> >> >>>>
> >> >>>> and
> >> >>>>
> >> >>>> <action application="eval" data="chat
> >> >>>> sip|${username}|${destination_number}@
> ${domain_name}|${msg_body}"/>
> >> >>>>
> >> >>>> and neither worked. The wiki says it's part of the API
> >> >>>> (http://wiki.freeswitch.org/wiki/Mod_dptools) but I'm pretty sure
> >> >>>> that's just a command-line API. Is there a way to run a
> command-line
> >> >>>> call from the dialplan?
> >> >>>>
> >> >>>> Any direction would be appreciated.
> >> >>>>
> >> >>>> _______________________________________________
> >> >>>> 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
> >> >>>
> >> >>>
> >> >>>
> >> >>> _______________________________________________
> >> >>> 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
> >> >>
> >> >> _______________________________________________
> >> >> 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
> >> >>
> >> >
> >>
> >> _______________________________________________
> >> 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
> >
> >
> > _______________________________________________
> > 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
> >
> >
>
> _______________________________________________
> 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 kheimerl at cs.berkeley.edu  Fri Jul 22 13:20:28 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Fri, 22 Jul 2011 02:20:28 -0700
Subject: [Freeswitch-users] Using chat command in a dialplan
In-Reply-To: <CAFiqYunELHt2ti+x4OBO=osyrjqn-HQRKhfrjeJV6Mvjpo87Kg@mail.gmail.com>
References: <CACyT-3m7nMTCwav_j_CVqtL-V+T9vFcyGwsrD6uatJ+bkVti3Q@mail.gmail.com>
	<1311149680.32170.264.camel@luna.madrid.commsmundi.com>
	<83217555-012B-4861-AAB9-045B2E78660D@gmail.com>
	<CACyT-3mASOUzN4kX_L4X734Jp1wL5nTbpi-Ytd4TuF-nJA3xaw@mail.gmail.com>
	<CACyT-3nBYP88Ve3oTC6R1WYSPKZkvY08iVTuSxfmm9miFgYEtA@mail.gmail.com>
	<CAFiqYumBRYwpNamX_ig525Hbo+v+s+uVZZ8httTu7Vd7LYk9Qg@mail.gmail.com>
	<CACyT-3=9J+Saqi_c6KuNAjNWyZpgQF+ao3GuUZhLUUD4mw+hog@mail.gmail.com>
	<CAFiqYunELHt2ti+x4OBO=osyrjqn-HQRKhfrjeJV6Mvjpo87Kg@mail.gmail.com>
Message-ID: <CACyT-3nXTDh8U7mSd4NOJAhXdW_YNx2BQCqK2-_3OsHoZDYENg@mail.gmail.com>

That's a good plan. I'll try that out this weekend.

On Fri, Jul 22, 2011 at 1:05 AM, Steven Ayre <steveayre at gmail.com> wrote:
> Never tried it from a dialplan although I'm sure it should work. But you
> should be able to do it through Lua etc.
>
> Maybe you could apply a quick patch to originate in mod_commands to log the
> arguments it's being passed - perhaps it's being mangled somehow.
>
> -Steve
>
>
> On 21 July 2011 23:47, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>>
>> Unfortunately, I am not. I'm working on a system that causes dialplan
>> actions when a chat is received by FS. That's what's going on here,
>> basically. I want someone to text 9199 and have FS immediately call
>> them back. I think "originate" is the right way to do that, as I need
>> to create a brand new session, not bridge to an existing one.
>>
>> Anyhow, any other ideas on why this isn't working?
>>
>> On Thu, Jul 21, 2011 at 1:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> > Are you trying to join the 2 calls together? If so, use bridge:
>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge
>> >
>> > -Steve
>> >
>> >
>> > On 21 July 2011 07:37, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>> >>
>> >> I'm now having a similar problem with using "originate" to create a
>> >> new call inside a dialplan. This one is particularly perplexing (and
>> >> small enough not to warrant pastebin)
>> >>
>> >> Here's the important bits of the dialplan:
>> >> ? ? ? ?<action application="log" data="(user/${username}
>> >> ${destination_number})"/>
>> >> ? ? ? ?<action application="set"
>> >> data="api_result=${originate(user/${username}
>> >> ${destination_number})}"/>
>> >>
>> >> and the log output
>> >> EXECUTE 9199 log((user/1304 at 192.168.1.144 9199))
>> >> 2011-07-20 23:17:12.955419 [DEBUG] mod_dptools.c:1202 9199)
>> >> EXECUTE 9199 set(api_result=-USAGE <call url>
>> >> <exten>|&<application_name>(<app_args>) [<dialplan>] [<context>]
>> >> [<cid_name>] [<cid_num>] [<timeout_sec>]
>> >> )
>> >>
>> >> So it's somehow mangling the originate commands. If I run the
>> >> originate command directly (with the arguments logged!) it works just
>> >> fine.
>> >> originate user/1304 at 192.168.1.144 9199
>> >>
>> >> What am I missing here? This seems so simple... ?but I'm at my wit's
>> >> end.
>> >>
>> >> Thanks for any direction in advance!
>> >>
>> >>
>> >> On Wed, Jul 20, 2011 at 7:11 PM, Kurtis Heimerl
>> >> <kheimerl at cs.berkeley.edu> wrote:
>> >> > Exactly what I was looking for. Thanks!
>> >> >
>> >> > On Wed, Jul 20, 2011 at 1:21 AM, Steven Ayre <steveayre at gmail.com>
>> >> > wrote:
>> >> >> Kurtis,
>> >> >>
>> >> >> *Any* api can be run from the dialplan. It's just a slightly
>> >> >> different
>> >> >> syntax - the link Francois gave you will show you how.
>> >> >>
>> >> >> Steve on iPhone
>> >> >>
>> >> >> On 20 Jul 2011, at 09:14, Fran?ois Delawarde
>> >> >> <fdelawarde at wirelessmundi.com> wrote:
>> >> >>
>> >> >>> Hi,
>> >> >>>
>> >> >>> Take a look at:
>> >> >>> http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
>> >> >>>
>> >> >>>
>> >> >>> Fran?ois.
>> >> >>>
>> >> >>> On Tue, 2011-07-19 at 23:00 -0700, Kurtis Heimerl wrote:
>> >> >>>> I feel somewhat embarrassed, but I can't figure out how to cause a
>> >> >>>> chat event to go off from the dialplan. I've tried:
>> >> >>>>
>> >> >>>> <action application="chat"
>> >> >>>>
>> >> >>>>
>> >> >>>> data="sip|${username}|${destination_number}@${domain_name}|${msg_body}"/>
>> >> >>>>
>> >> >>>> and
>> >> >>>>
>> >> >>>> <action application="eval" data="chat
>> >> >>>>
>> >> >>>> sip|${username}|${destination_number}@${domain_name}|${msg_body}"/>
>> >> >>>>
>> >> >>>> and neither worked. The wiki says it's part of the API
>> >> >>>> (http://wiki.freeswitch.org/wiki/Mod_dptools) but I'm pretty sure
>> >> >>>> that's just a command-line API. Is there a way to run a
>> >> >>>> command-line
>> >> >>>> call from the dialplan?
>> >> >>>>
>> >> >>>> Any direction would be appreciated.
>> >> >>>>
>> >> >>>> _______________________________________________
>> >> >>>> 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
>> >> >>>
>> >> >>>
>> >> >>>
>> >> >>> _______________________________________________
>> >> >>> 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
>> >> >>
>> >> >> _______________________________________________
>> >> >> 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
>> >> >>
>> >> >
>> >>
>> >> _______________________________________________
>> >> 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
>> >
>> >
>> > _______________________________________________
>> > 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
>> >
>> >
>>
>> _______________________________________________
>> 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
>
>
> _______________________________________________
> 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 Nabble at slickdeals.endjunk.com  Fri Jul 22 15:21:57 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 22 Jul 2011 04:21:57 -0700 (PDT)
Subject: [Freeswitch-users] Pizza Demo Grammar
In-Reply-To: <C35D5B41-1661-462A-9621-84D11DE60B28@freeswitch.org>
References: <45543322-DEDF-4125-ADB9-EABA63C446C5@bryansmart.com>
	<C35D5B41-1661-462A-9621-84D11DE60B28@freeswitch.org>
Message-ID: <1311333717717-6610132.post@n2.nabble.com>


Brian West wrote:
> I think the hard drive is intact so i'll get the data back up as soon as I
> get a new box.
At least, that is a piece of good news that you can still retrieve the data.
Once you have it, I am sure you won't have any more problem to share when
you upload it to FS contrib section. 

As with your dead computer, there is probably a chance you can still
resurrect it. If its components, i.e. mobo, cards, CPU cooling system,
inside power supply, etc., have collected dusts, you may want to clean them
using air pressure system or use a non-electrostatic brush to dust them away
in an open area (preferably in the morning outside the house when the air is
still cool and damn). If that still doesn't work, chances are replacing the
old power supply will a better one will resurrect your system. Sometimes
your local Fry's Electronics stores in Alpharetta, GA and/or Duluth, GA have
some power supply on sale very inexpensive. Check out the 
http://www.facebook.com/fryselectronics?sk=app_7146470109 Fry's Electronics
ads on AJC  (courtesy of FaceBook).

-----
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/Pizza-Demo-Grammar-tp6605004p6610132.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kris at kriskinc.com  Fri Jul 22 17:42:55 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 22 Jul 2011 09:42:55 -0400
Subject: [Freeswitch-users] Cisco CUCM 6.1 and Freeswitch SIP trunk
In-Reply-To: <FAAAC9F6-ACCE-4A17-BCEE-EED2FE2DAE1A@gmail.com>
References: <CA+ViVc99bqe2e8AAy-ROxLA_LPumQv7q50ygPNQjf_KMzypKAg@mail.gmail.com>
	<CAKDfjgd0R0Sb_BYpmyHLjmv-oPajQ_uw_5ev2ioPDrXXM_3qpw@mail.gmail.com>
	<80E395DA-B3F2-42CE-AB9F-098096B32B45@gmail.com>
	<CAD-qRkjh2-St09KcqbMEA=muQw5tzHckV=TVVMRdFHLYqLr6OQ@mail.gmail.com>
	<CA+ViVc9ydZfjWwTVVkDxRugAC+s4iV9TPRLEU3v=h7emfnf98Q@mail.gmail.com>
	<CAKDfjgfh2162ysaDc3-91-qNR4wV9R5K-Ds-OtSYOXdgiDZD9w@mail.gmail.com>
	<CA+ViVc_U+iEifVcwb=n=keR4EO8rTV-SGFEu7XO0q=NSzykGqw@mail.gmail.com>
	<CA+ViVc-oY3ZV6AYiDkHeSVJucjm=pLnha0=gW4P2A91Ux=TUSQ@mail.gmail.com>
	<CAKDfjgczmND40J-ygWUCsJ64sSb+USenco87MFmm9mMTRGmRNg@mail.gmail.com>
	<FAAAC9F6-ACCE-4A17-BCEE-EED2FE2DAE1A@gmail.com>
Message-ID: <CAKDfjgcGU7ea4HmC726gHQu2R+jTW6G=u8_Shqyv30L=NuriOA@mail.gmail.com>

David,

  I'd hate to see you have to go that route (it carries its own
headaches).  You should try sticking with SIP.

  This is really more of a Cisco question.  Have you tried asking on

http://puck.nether.net/mailman/listinfo/cisco-nsp

-or-

http://puck.nether.net/mailman/listinfo/cisco-voip

  Even Voiceops may have some clues:

http://puck.nether.net/mailman/listinfo/voiceops

  Of course if you find the answer here it would be nice to have you
come back and let us know what you needed to do to CUCM to get it to
act sane...

On Thu, Jul 21, 2011 at 6:28 PM, David Martin <adavidm at gmail.com> wrote:
> Thanks for your assistance. I might try mod_opal and use h323 instead then.
>
> Thanks again.
>
> David
>
> Sent from my iPad
>
> On 21 Jul 2011, at 20:44, Kristian Kielhofner <kris at kriskinc.com> wrote:
>
>> The issue now is CUCM never sends an SDP. ?Not in the INVITE, not in
>> the ACK. ?If the INVITE contained an SDP you'd be fine with the
>> default configuration. ?If it didn't but the ACK did you'd be fine
>> with 3pcc enabled.
>>
>> Right now CUCM isn't providing either. ?That's your problem, it's
>> somewhere on the Cisco side, and I've never seen it before.
>> Unfortunately I don't have any specific recommendations for you.
>>
>> On Thu, Jul 21, 2011 at 8:46 AM, David Martin <adavidm at gmail.com> wrote:
>>> For completeness, here is the INVITE with "Disable Early Media" unchecked:
>>> ? ?------------------------------------------------------------------------
>>> ? ?INVITE sip:65131111 at 10.201.27.20:5060 SIP/2.0
>>> ? ?Date: Thu, 21 Jul 2011 12:42:02 GMT
>>> ? ?Call-Info:
>>> <sip:10.200.170.3:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
>>> ? ?Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>> SUBSCRIBE, NOTIFY, PUBLISH
>>> ? ?From: "Ryan Girdlestone"
>>> <sip:64105554 at 10.200.170.3>;tag=90509204-400f-4cb6-9be7-cb5410a83a11-54376643
>>> ? ?Allow-Events: presence, kpml
>>> ? ?Supported: timer,replaces
>>> ? ?Min-SE: ?1800
>>> ? ?Remote-Party-ID: "Ryan Girdlestone"
>>> <sip:64105554 at 10.200.170.3>;party=calling;screen=yes;privacy=off
>>> ? ?Content-Length: 0
>>> ? ?User-Agent: Cisco-CUCM6.1
>>> ? ?To: <sip:65131111 at 10.201.27.20>
>>> ? ?Contact: <sip:64105554 at 10.200.170.3:5060>
>>> ? ?Expires: 180
>>> ? ?Call-ID: d3b6ce80-e2811e9a-33-3aac80a at 10.200.170.3
>>> ? ?Via: SIP/2.0/UDP 10.200.170.3:5060;branch=z9hG4bK4d55790bf1
>>> ? ?CSeq: 101 INVITE
>>> ? ?Session-Expires: ?1800
>>> ? ?Max-Forwards: 70
>>>
>>>
>>>
>>> On 21 July 2011 13:10, David Martin <adavidm at gmail.com> wrote:
>>>>
>>>> Kristian,
>>>> Here are some screenshots from the trunk config.
>>>> http://img200.imageshack.us/img200/9405/siptrunk.png
>>>> and for the sip profile
>>>> http://img191.imageshack.us/img191/3863/sipprofile.png
>>>> I have tried the early media option both ways with the same result. Is the
>>>> MTP payload type correct?
>>>> Appreciate your help on this.
>>>> David
>>>>
>>>> On 21 July 2011 11:13, Kristian Kielhofner <kris at kriskinc.com> wrote:
>>>>>
>>>>> David,
>>>>>
>>>>> ?The problem here is that even with 3pcc enabled the ACK from
>>>>> CallManager doesn't have an SDP. ?CallManager never provides an SDP
>>>>> during this entire trace. ?Not in the INVITE (preferred) and not in
>>>>> the ACK (as I already mentioned).
>>>>>
>>>>> ?This is odd, even for CallManager. ?Are you sure you have MTP enabled:
>>>>>
>>>>> http://blog.krisk.org/2010/05/another-sip-gotcha-cisco.html
>>>>>
>>>>> On Thu, Jul 21, 2011 at 3:45 AM, David Martin <adavidm at gmail.com> wrote:
>>>>>> Gotcha, not sure how I missed the INVITE the first time, but here is a
>>>>>> new
>>>>>> pastebin that should have everything:
>>>>>> http://pastebin.com/T3eqms8N
>>>>>> thanks in advance once again.
>>>>>> David
>>>>>>
>>>>>>
>>>>>> On 21 July 2011 02:19, Christopher Rienzo <cmrienzo at gmail.com> wrote:
>>>>>>>
>>>>>>> Line 7 is the 200 OK response to the INVITE.
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Jul 20, 2011 at 6:54 PM, David Martin <adavidm at gmail.com>
>>>>>>> wrote:
>>>>>>>>
>>>>>>>> Forgive my confusion, I'm new to freeswitch. Does the invite not
>>>>>>>> start at
>>>>>>>> line 7 on the pastebin?
>>>>>>>>
>>>>>>>> David
>>>>>>>>
>>>>>>>> Sent from my iPhone
>>>>>>>>
>>>>>>>> On 20 Jul 2011, at 07:22 PM, Kristian Kielhofner <kris at kriskinc.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>>> That trace is missing the INVITE that starts the transaction.
>>>>>>>>>
>>>>>>>>> On Wed, Jul 20, 2011 at 11:48 AM, David Martin <adavidm at gmail.com>
>>>>>>>>> wrote:
>>>>>>>>>> All,
>>>>>>>>>> Firstly apologies if this is not the correct place to ask, but I
>>>>>>>>>> am
>>>>>>>>>> having
>>>>>>>>>> problems getting CUCM 6.1 to talk with freeswitch over a SIP trunk
>>>>>>>>>> (I
>>>>>>>>>> only
>>>>>>>>>> need inbound from Cisco -> Freeswitch). I have enabled 3pcc, but
>>>>>>>>>> am
>>>>>>>>>> getting
>>>>>>>>>> the following sofia trace:
>>>>>>>>>> http://pastebin.com/g9VmvxFe
>>>>>>>>>> Can anyone point me in the right direction? I have also tried
>>>>>>>>>> enabling
>>>>>>>>>> MTP
>>>>>>>>>> on the Cisco side, but this does not make any difference.
>>>>>>>>>> Thanks in advance.
>>>>>>>>>> David
>>>>>>>>>> _______________________________________________
>>>>>>>>>> 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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> Kristian Kielhofner
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> 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
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> 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
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> 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
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Kristian Kielhofner
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>>
>>
>> --
>> Kristian Kielhofner
>>
>> _______________________________________________
>> 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
>
> _______________________________________________
> 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
>



-- 
Kristian Kielhofner


From lakersman2006 at yahoo.com  Fri Jul 22 20:25:07 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Fri, 22 Jul 2011 09:25:07 -0700 (PDT)
Subject: [Freeswitch-users] How to get call bridge status?
In-Reply-To: <CAFiqYu=MNHUALvEjO9Db4ng_JQi8_2PjHrOQpx0KkUOQ33ZwZA@mail.gmail.com>
Message-ID: <1311351907.82029.YahooMailClassic@web161003.mail.bf1.yahoo.com>

Steve,

When I tried "bridge_hangup_cause" I still receive a blank even though I should be receiving something. Any other suggestions?

--- On Fri, 7/22/11, Steven Ayre <steveayre at gmail.com> wrote:

From: Steven Ayre <steveayre at gmail.com>
Subject: Re: [Freeswitch-users] How to get call bridge status?
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Friday, July 22, 2011, 12:58 AM

What do you get in bridge_hangup_cause?

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

-Steve





On 22 July 2011 02:10, Sam <lakersman2006 at yahoo.com> wrote:


I am writing a perl script that does a call bridge and I would like to know what is the status of the call bridge once the bridge application as finished execution. I am using the following but do not know if it is the correct way since on NO_ANSWER call states the sip code is blank, but shouldn't it return SIP:480 according to this http://wiki.freeswitch.org/wiki/Hangup_causes?



$sipcode = $session->getVariable("proto_specific_hangup_cause");

_______________________________________________

Join us at ClueCon 2011, Aug 9-11, Chicago

http://www.cluecon.com 877-7-4ACLUE



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FreeSWITCH-users at lists.freeswitch.org

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

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

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From lakersman2006 at yahoo.com  Fri Jul 22 22:31:31 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Fri, 22 Jul 2011 11:31:31 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk dial status
Message-ID: <1311359491.89428.YahooMailClassic@web161018.mail.bf1.yahoo.com>

Hi,

I am trying to port over an Asterisk AGI script that does a call bridge to Freeswitch using perl, I wanted to know what is the equivalent of the asterisk dial status like: ANSWER, NO_ANSWER, CANCEL, BUSY, CONGESTION, CHANUNAVAIL ? I wanted to know if what methods can be used to gets these statues?
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From Hector.Geraldino at ip-soft.net  Fri Jul 22 22:57:50 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Fri, 22 Jul 2011 14:57:50 -0400
Subject: [Freeswitch-users] Asterisk dial status
In-Reply-To: <1311359491.89428.YahooMailClassic@web161018.mail.bf1.yahoo.com>
References: <1311359491.89428.YahooMailClassic@web161018.mail.bf1.yahoo.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD021FD89DA5@NY1-EXMB-01.ip-soft.net>

Hi Sam,

My guess is that you can subscribe to the events of the channel and evaluate the channel events. Look at:

http://wiki.freeswitch.org/wiki/Event_List

specifically the channel events list.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sam
Sent: Friday, July 22, 2011 2:32 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Asterisk dial status

Hi,

I am trying to port over an Asterisk AGI script that does a call bridge to Freeswitch using perl, I wanted to know what is the equivalent of the asterisk dial status like: ANSWER, NO_ANSWER, CANCEL, BUSY, CONGESTION, CHANUNAVAIL ? I wanted to know if what methods can be used to gets these statues?


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From lakersman2006 at yahoo.com  Sat Jul 23 00:23:00 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Fri, 22 Jul 2011 13:23:00 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk dial status
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD021FD89DA5@NY1-EXMB-01.ip-soft.net>
Message-ID: <1311366180.83556.YahooMailClassic@web161016.mail.bf1.yahoo.com>

Sam,

Does this require event sockets? Or how would I check these events?

--- On Fri, 7/22/11, Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:

From: Hector Geraldino <Hector.Geraldino at ip-soft.net>
Subject: Re: [Freeswitch-users] Asterisk dial status
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Friday, July 22, 2011, 11:57 AM

Hi Sam, ?My guess is that you can subscribe to the events of the channel and evaluate the channel events. Look at: ?http://wiki.freeswitch.org/wiki/Event_List ?specifically the channel events list. ?From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sam
Sent: Friday, July 22, 2011 2:32 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Asterisk dial status ?Hi,

I am trying to port over an Asterisk AGI script that does a call bridge to Freeswitch using perl, I wanted to know what is the equivalent of the asterisk dial status like: ANSWER, NO_ANSWER, CANCEL, BUSY, CONGESTION, CHANUNAVAIL ? I wanted to know if what methods can be used to gets these statues? ?
-----Inline Attachment Follows-----

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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From jeff at jefflenk.com  Sat Jul 23 01:02:41 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 22 Jul 2011 14:02:41 -0700 (PDT)
Subject: [Freeswitch-users] Can't get fax working
In-Reply-To: <4E28F03A.6030808@tagnet.ru>
References: <4E25605D.7050809@tagnet.ru> <4E285F0C.6040403@tagnet.ru>
	<CAAtWKeG7kv6dKu9BS81cG76nkjbs3Wa6amPMw9U-eve7qBa-cw@mail.gmail.com>
	<4E28F03A.6030808@tagnet.ru>
Message-ID: <1311368561787-6611816.post@n2.nabble.com>

Please make current and try this again there was a change to git that
probably fixes this.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-get-fax-working-tp6598322p6611816.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Sat Jul 23 01:06:18 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 22 Jul 2011 16:06:18 -0500
Subject: [Freeswitch-users] Can't get fax working
In-Reply-To: <1311368561787-6611816.post@n2.nabble.com>
References: <4E25605D.7050809@tagnet.ru> <4E285F0C.6040403@tagnet.ru>
	<CAAtWKeG7kv6dKu9BS81cG76nkjbs3Wa6amPMw9U-eve7qBa-cw@mail.gmail.com>
	<4E28F03A.6030808@tagnet.ru>
	<1311368561787-6611816.post@n2.nabble.com>
Message-ID: <C8A24193-A4E2-4EF8-94DC-8344F6CCBE1D@freeswitch.org>

lock the MP124 to 9600 bps.  otherwise it won't work.

/b

On Jul 22, 2011, at 4:02 PM, Jeff Lenk wrote:

> Please make current and try this again there was a change to git that
> probably fixes this.

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From Hector.Geraldino at ip-soft.net  Sat Jul 23 01:28:27 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Fri, 22 Jul 2011 17:28:27 -0400
Subject: [Freeswitch-users] Asterisk dial status
In-Reply-To: <1311366180.83556.YahooMailClassic@web161016.mail.bf1.yahoo.com>
References: <6A6B4C284AD15042B429EB9D904544AD021FD89DA5@NY1-EXMB-01.ip-soft.net>
	<1311366180.83556.YahooMailClassic@web161016.mail.bf1.yahoo.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD021FD89DC2@NY1-EXMB-01.ip-soft.net>

It does.

You can check the events by connecting to FS on inbound mode, or using FS outbound socket connection to an application designed by you.

As you already have a perl script to interact with FS, you can modify it to connect to FS (inbound mode) and listen for events. You can subscribe to the events of a particular call, or filter an specific set of events. I haven't used Asterisk before so I'm not sure how the AGI fits on the FS architecture.

You can get more info at:
http://wiki.freeswitch.org/wiki/Mod_event_socket

And a working demo at:
http://wiki.freeswitch.org/wiki/Perl_freeswitch_client_example



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sam
Sent: Friday, July 22, 2011 4:23 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Asterisk dial status

Sam,

Does this require event sockets? Or how would I check these events?

--- On Fri, 7/22/11, Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:

From: Hector Geraldino <Hector.Geraldino at ip-soft.net>
Subject: Re: [Freeswitch-users] Asterisk dial status
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Friday, July 22, 2011, 11:57 AM

Hi Sam,



My guess is that you can subscribe to the events of the channel and evaluate the channel events. Look at:



http://wiki.freeswitch.org/wiki/Event_List



specifically the channel events list.



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sam
Sent: Friday, July 22, 2011 2:32 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Asterisk dial status



Hi,

I am trying to port over an Asterisk AGI script that does a call bridge to Freeswitch using perl, I wanted to know what is the equivalent of the asterisk dial status like: ANSWER, NO_ANSWER, CANCEL, BUSY, CONGESTION, CHANUNAVAIL ? I wanted to know if what methods can be used to gets these statues?




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From adavidm at gmail.com  Sat Jul 23 02:23:05 2011
From: adavidm at gmail.com (David Martin)
Date: Fri, 22 Jul 2011 23:23:05 +0100
Subject: [Freeswitch-users] Cisco CUCM 6.1 and Freeswitch SIP trunk
In-Reply-To: <CAKDfjgcGU7ea4HmC726gHQu2R+jTW6G=u8_Shqyv30L=NuriOA@mail.gmail.com>
References: <CA+ViVc99bqe2e8AAy-ROxLA_LPumQv7q50ygPNQjf_KMzypKAg@mail.gmail.com>
	<CAKDfjgd0R0Sb_BYpmyHLjmv-oPajQ_uw_5ev2ioPDrXXM_3qpw@mail.gmail.com>
	<80E395DA-B3F2-42CE-AB9F-098096B32B45@gmail.com>
	<CAD-qRkjh2-St09KcqbMEA=muQw5tzHckV=TVVMRdFHLYqLr6OQ@mail.gmail.com>
	<CA+ViVc9ydZfjWwTVVkDxRugAC+s4iV9TPRLEU3v=h7emfnf98Q@mail.gmail.com>
	<CAKDfjgfh2162ysaDc3-91-qNR4wV9R5K-Ds-OtSYOXdgiDZD9w@mail.gmail.com>
	<CA+ViVc_U+iEifVcwb=n=keR4EO8rTV-SGFEu7XO0q=NSzykGqw@mail.gmail.com>
	<CA+ViVc-oY3ZV6AYiDkHeSVJucjm=pLnha0=gW4P2A91Ux=TUSQ@mail.gmail.com>
	<CAKDfjgczmND40J-ygWUCsJ64sSb+USenco87MFmm9mMTRGmRNg@mail.gmail.com>
	<FAAAC9F6-ACCE-4A17-BCEE-EED2FE2DAE1A@gmail.com>
	<CAKDfjgcGU7ea4HmC726gHQu2R+jTW6G=u8_Shqyv30L=NuriOA@mail.gmail.com>
Message-ID: <DC79FD7E-539B-43D7-89B1-33B6304607EB@gmail.com>

Kristian,

I'm starting to see some of that pain already! It's not wasted effort though as I have removed the ubuntu packages and built freeswitch from git so I can more easily try out different modules and versions of software. Getting mod_h323 or mod_opal to compile is proving more tricky but I can tinker with that at the same time as pursuing the SIP route, which I would definitely prefer.

I will take your advice and post the question to the lists. I'm also going to raise a call with our cisco partner, we pay them enough money for support so they can earn it for a change. I will let the list know what I find out and if I get it fixed I will write something up.

Thanks yet again

David

Sent from my iPad

On 22 Jul 2011, at 14:42, Kristian Kielhofner <kris at kriskinc.com> wrote:

> David,
> 
>  I'd hate to see you have to go that route (it carries its own
> headaches).  You should try sticking with SIP.
> 
>  This is really more of a Cisco question.  Have you tried asking on
> 
> http://puck.nether.net/mailman/listinfo/cisco-nsp
> 
> -or-
> 
> http://puck.nether.net/mailman/listinfo/cisco-voip
> 
>  Even Voiceops may have some clues:
> 
> http://puck.nether.net/mailman/listinfo/voiceops
> 
>  Of course if you find the answer here it would be nice to have you
> come back and let us know what you needed to do to CUCM to get it to
> act sane...
> 
> On Thu, Jul 21, 2011 at 6:28 PM, David Martin <adavidm at gmail.com> wrote:
>> Thanks for your assistance. I might try mod_opal and use h323 instead then.
>> 
>> Thanks again.
>> 
>> David
>> 
>> Sent from my iPad
>> 
>> On 21 Jul 2011, at 20:44, Kristian Kielhofner <kris at kriskinc.com> wrote:
>> 
>>> The issue now is CUCM never sends an SDP.  Not in the INVITE, not in
>>> the ACK.  If the INVITE contained an SDP you'd be fine with the
>>> default configuration.  If it didn't but the ACK did you'd be fine
>>> with 3pcc enabled.
>>> 
>>> Right now CUCM isn't providing either.  That's your problem, it's
>>> somewhere on the Cisco side, and I've never seen it before.
>>> Unfortunately I don't have any specific recommendations for you.
>>> 
>>> On Thu, Jul 21, 2011 at 8:46 AM, David Martin <adavidm at gmail.com> wrote:
>>>> For completeness, here is the INVITE with "Disable Early Media" unchecked:
>>>>    ------------------------------------------------------------------------
>>>>    INVITE sip:65131111 at 10.201.27.20:5060 SIP/2.0
>>>>    Date: Thu, 21 Jul 2011 12:42:02 GMT
>>>>    Call-Info:
>>>> <sip:10.200.170.3:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
>>>>    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>> SUBSCRIBE, NOTIFY, PUBLISH
>>>>    From: "Ryan Girdlestone"
>>>> <sip:64105554 at 10.200.170.3>;tag=90509204-400f-4cb6-9be7-cb5410a83a11-54376643
>>>>    Allow-Events: presence, kpml
>>>>    Supported: timer,replaces
>>>>    Min-SE:  1800
>>>>    Remote-Party-ID: "Ryan Girdlestone"
>>>> <sip:64105554 at 10.200.170.3>;party=calling;screen=yes;privacy=off
>>>>    Content-Length: 0
>>>>    User-Agent: Cisco-CUCM6.1
>>>>    To: <sip:65131111 at 10.201.27.20>
>>>>    Contact: <sip:64105554 at 10.200.170.3:5060>
>>>>    Expires: 180
>>>>    Call-ID: d3b6ce80-e2811e9a-33-3aac80a at 10.200.170.3
>>>>    Via: SIP/2.0/UDP 10.200.170.3:5060;branch=z9hG4bK4d55790bf1
>>>>    CSeq: 101 INVITE
>>>>    Session-Expires:  1800
>>>>    Max-Forwards: 70
>>>> 
>>>> 
>>>> 
>>>> On 21 July 2011 13:10, David Martin <adavidm at gmail.com> wrote:
>>>>> 
>>>>> Kristian,
>>>>> Here are some screenshots from the trunk config.
>>>>> http://img200.imageshack.us/img200/9405/siptrunk.png
>>>>> and for the sip profile
>>>>> http://img191.imageshack.us/img191/3863/sipprofile.png
>>>>> I have tried the early media option both ways with the same result. Is the
>>>>> MTP payload type correct?
>>>>> Appreciate your help on this.
>>>>> David
>>>>> 
>>>>> On 21 July 2011 11:13, Kristian Kielhofner <kris at kriskinc.com> wrote:
>>>>>> 
>>>>>> David,
>>>>>> 
>>>>>>  The problem here is that even with 3pcc enabled the ACK from
>>>>>> CallManager doesn't have an SDP.  CallManager never provides an SDP
>>>>>> during this entire trace.  Not in the INVITE (preferred) and not in
>>>>>> the ACK (as I already mentioned).
>>>>>> 
>>>>>>  This is odd, even for CallManager.  Are you sure you have MTP enabled:
>>>>>> 
>>>>>> http://blog.krisk.org/2010/05/another-sip-gotcha-cisco.html
>>>>>> 
>>>>>> On Thu, Jul 21, 2011 at 3:45 AM, David Martin <adavidm at gmail.com> wrote:
>>>>>>> Gotcha, not sure how I missed the INVITE the first time, but here is a
>>>>>>> new
>>>>>>> pastebin that should have everything:
>>>>>>> http://pastebin.com/T3eqms8N
>>>>>>> thanks in advance once again.
>>>>>>> David
>>>>>>> 
>>>>>>> 
>>>>>>> On 21 July 2011 02:19, Christopher Rienzo <cmrienzo at gmail.com> wrote:
>>>>>>>> 
>>>>>>>> Line 7 is the 200 OK response to the INVITE.
>>>>>>>> 
>>>>>>>> 
>>>>>>>> On Wed, Jul 20, 2011 at 6:54 PM, David Martin <adavidm at gmail.com>
>>>>>>>> wrote:
>>>>>>>>> 
>>>>>>>>> Forgive my confusion, I'm new to freeswitch. Does the invite not
>>>>>>>>> start at
>>>>>>>>> line 7 on the pastebin?
>>>>>>>>> 
>>>>>>>>> David
>>>>>>>>> 
>>>>>>>>> Sent from my iPhone
>>>>>>>>> 
>>>>>>>>> On 20 Jul 2011, at 07:22 PM, Kristian Kielhofner <kris at kriskinc.com>
>>>>>>>>> wrote:
>>>>>>>>> 
>>>>>>>>>> That trace is missing the INVITE that starts the transaction.
>>>>>>>>>> 
>>>>>>>>>> On Wed, Jul 20, 2011 at 11:48 AM, David Martin <adavidm at gmail.com>
>>>>>>>>>> wrote:
>>>>>>>>>>> All,
>>>>>>>>>>> Firstly apologies if this is not the correct place to ask, but I
>>>>>>>>>>> am
>>>>>>>>>>> having
>>>>>>>>>>> problems getting CUCM 6.1 to talk with freeswitch over a SIP trunk
>>>>>>>>>>> (I
>>>>>>>>>>> only
>>>>>>>>>>> need inbound from Cisco -> Freeswitch). I have enabled 3pcc, but
>>>>>>>>>>> am
>>>>>>>>>>> getting
>>>>>>>>>>> the following sofia trace:
>>>>>>>>>>> http://pastebin.com/g9VmvxFe
>>>>>>>>>>> Can anyone point me in the right direction? I have also tried
>>>>>>>>>>> enabling
>>>>>>>>>>> MTP
>>>>>>>>>>> on the Cisco side, but this does not make any difference.
>>>>>>>>>>> Thanks in advance.
>>>>>>>>>>> David
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> 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
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> --
>>>>>>>>>> Kristian Kielhofner
>>>>>>>>>> 
>>>>>>>>>> _______________________________________________
>>>>>>>>>> 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
>>>>>>>>> 
>>>>>>>>> _______________________________________________
>>>>>>>>> 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
>>>>>>>> 
>>>>>>>> 
>>>>>>>> _______________________________________________
>>>>>>>> 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
>>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 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
>>>>>>> 
>>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> --
>>>>>> Kristian Kielhofner
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 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
>>>>> 
>>>> 
>>>> 
>>>> _______________________________________________
>>>> 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
>>>> 
>>>> 
>>> 
>>> 
>>> 
>>> --
>>> Kristian Kielhofner
>>> 
>>> _______________________________________________
>>> 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
>> 
>> _______________________________________________
>> 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
>> 
> 
> 
> 
> -- 
> Kristian Kielhofner
> 
> _______________________________________________
> 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 va_mclean at yahoo.com  Sat Jul 23 02:54:06 2011
From: va_mclean at yahoo.com (yahoo2003)
Date: Fri, 22 Jul 2011 15:54:06 -0700 (PDT)
Subject: [Freeswitch-users] Dingaling/Gtalk hangup problem
Message-ID: <1311375246484-6612074.post@n2.nabble.com>

We have been troubleshooting Dingaling/Gtalk hangup problem for a week, the
problem is:

If A using FS->Gtalk to make calls and hangup before B is answered, B will
continue to ring until either B times out or Google times out in 2 minutes.

If B is answered, A can hangup correctly (ACTIVE->HANGUP).
 
The problem is only for RING->HANGUP.

I tried Windows and Linux version of FS, they behaved the same way.

Asterisk Gtalk hangs up correctly.

The "client.xml" I used was:

<include>
  <profile type="client">
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
    
                           
    
  </profile>
</include>


Any suggestions are appreciated.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dingaling-Gtalk-hangup-problem-tp6612074p6612074.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From slash at nat-tele.com  Thu Jul 21 20:56:04 2011
From: slash at nat-tele.com (slash at nat-tele.com)
Date: Thu, 21 Jul 2011 11:56:04 -0500
Subject: [Freeswitch-users] rxfax on b-leg
Message-ID: <6c1cf419782ba6781928a095d3bc19bc@nat-tele.com>


 Hello!


 I wonder if there is a way to make rxfax recieve fax on outgoing call? 
 The idea is that registered FS user dials out, than tone_detect is used 
 to detect fax tone and start rxfax. The difference is that rxfax should 
 be run on B-leg, instead of A-leg in incoming fax call scenario.

 I've tried to make a test config with outgoing call, rxfax & 
 tone_detect, but when I dial out and remote user starts sending fax to 
 me, FS detects a tone correctly, starts rxfax, but on my leg (A-leg, i 
 hear fax tones) and B-leg (with real fax machine) is hanged up.

 Are there are solutions to explain somehow to rxfax that it should be 
 run on B-leg, not on A-leg?

 Thanks in advance!

 With best regards,
 Kirill.



From gcd at i.ph  Sat Jul 23 14:24:43 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Sat, 23 Jul 2011 18:24:43 +0800
Subject: [Freeswitch-users] rxfax on b-leg
In-Reply-To: <6c1cf419782ba6781928a095d3bc19bc@nat-tele.com>
References: <6c1cf419782ba6781928a095d3bc19bc@nat-tele.com>
Message-ID: <CAHKqMb5_PoYeijC+rEFS6ZMq21SkYjCUnZsiZ1gXD2s7KSu3Ug@mail.gmail.com>

hi kirill

this is working by manually transferring the call to another extension. you
can use the model here but use tone detect app:
http://wiki.freeswitch.org/wiki/Mod_spandsp#Manual_fax_transmission

hope it works!
-nandy

On Fri, Jul 22, 2011 at 12:56 AM, <slash at nat-tele.com> wrote:

>
>  Hello!
>
>
>  I wonder if there is a way to make rxfax recieve fax on outgoing call?
>  The idea is that registered FS user dials out, than tone_detect is used
>  to detect fax tone and start rxfax. The difference is that rxfax should
>  be run on B-leg, instead of A-leg in incoming fax call scenario.
>
>  I've tried to make a test config with outgoing call, rxfax &
>  tone_detect, but when I dial out and remote user starts sending fax to
>  me, FS detects a tone correctly, starts rxfax, but on my leg (A-leg, i
>  hear fax tones) and B-leg (with real fax machine) is hanged up.
>
>  Are there are solutions to explain somehow to rxfax that it should be
>  run on B-leg, not on A-leg?
>
>  Thanks in advance!
>
>  With best regards,
>  Kirill.
>
>
> _______________________________________________
> 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 boris at tagnet.ru  Sat Jul 23 17:10:26 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 23 Jul 2011 19:10:26 +0600
Subject: [Freeswitch-users] Can't get fax working
In-Reply-To: <C8A24193-A4E2-4EF8-94DC-8344F6CCBE1D@freeswitch.org>
References: <4E25605D.7050809@tagnet.ru> <4E285F0C.6040403@tagnet.ru>
	<CAAtWKeG7kv6dKu9BS81cG76nkjbs3Wa6amPMw9U-eve7qBa-cw@mail.gmail.com>
	<4E28F03A.6030808@tagnet.ru>
	<1311368561787-6611816.post@n2.nabble.com>
	<C8A24193-A4E2-4EF8-94DC-8344F6CCBE1D@freeswitch.org>
Message-ID: <4E2AC842.7050102@tagnet.ru>

Hello!

     Yes, it's locked at 9600. Testing the latest git, looks like bug 
already fixed.

> lock the MP124 to 9600 bps.  otherwise it won't work.
>
> /b
>
> On Jul 22, 2011, at 4:02 PM, Jeff Lenk wrote:
>
>> Please make current and try this again there was a change to git that
>> probably fixes this.
>
>
>
> _______________________________________________
> 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


-- 
Regards,
Boris


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From boris at tagnet.ru  Sat Jul 23 17:25:38 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 23 Jul 2011 19:25:38 +0600
Subject: [Freeswitch-users] Can't get fax working
In-Reply-To: <1311368561787-6611816.post@n2.nabble.com>
References: <4E25605D.7050809@tagnet.ru> <4E285F0C.6040403@tagnet.ru>
	<CAAtWKeG7kv6dKu9BS81cG76nkjbs3Wa6amPMw9U-eve7qBa-cw@mail.gmail.com>
	<4E28F03A.6030808@tagnet.ru>
	<1311368561787-6611816.post@n2.nabble.com>
Message-ID: <4E2ACBD2.4040903@tagnet.ru>

Hello!

     Yes, I tested and it works now! Have received 10 of 10 faxes with 
no problems even at 14400 bps! Thanks to all developers for their work 
and very very nice and powerfull software!

> Please make current and try this again there was a change to git that
> probably fixes this.
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-get-fax-working-tp6598322p6611816.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> 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


-- 
Regards,
Boris




From steveu at coppice.org  Sat Jul 23 17:35:53 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 23 Jul 2011 21:35:53 +0800
Subject: [Freeswitch-users] rxfax on b-leg
In-Reply-To: <6c1cf419782ba6781928a095d3bc19bc@nat-tele.com>
References: <6c1cf419782ba6781928a095d3bc19bc@nat-tele.com>
Message-ID: <4E2ACE39.2080007@coppice.org>

On 07/22/2011 12:56 AM, slash at nat-tele.com wrote:
>   Hello!
>
>
>   I wonder if there is a way to make rxfax recieve fax on outgoing call?
>   The idea is that registered FS user dials out, than tone_detect is used
>   to detect fax tone and start rxfax. The difference is that rxfax should
>   be run on B-leg, instead of A-leg in incoming fax call scenario.
>
>   I've tried to make a test config with outgoing call, rxfax&
>   tone_detect, but when I dial out and remote user starts sending fax to
>   me, FS detects a tone correctly, starts rxfax, but on my leg (A-leg, i
>   hear fax tones) and B-leg (with real fax machine) is hanged up.
>
>   Are there are solutions to explain somehow to rxfax that it should be
>   run on B-leg, not on A-leg?
>
>   Thanks in advance!
>
>   With best regards,
>   Kirill.
If you use rxfax at the outgoing end of a call, mod_spandsp expects to 
get that FAX by the polling procedure (i.e. the one you use when you 
dial into a FAX server to collect an information FAX from it). If you 
want to freely act as a simple FAX transmitter or receiver in the middle 
of a call, the current module code may not properly allow for that. As 
someone else said, there are tricks related to call transfers which can 
be used to coax the fax instance into caller or answerer mode. Spandsp 
does pretty much everything the FAX spec allows. If there is demand we 
could provide access to more of that functionality.

Steve



From moises.silva at gmail.com  Sat Jul 23 20:55:55 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 23 Jul 2011 12:55:55 -0400
Subject: [Freeswitch-users] ftmod_libPRI outgoing calls channel selection
In-Reply-To: <4E1EF6D2.7040705@comgate.cz>
References: <4E1EF6D2.7040705@comgate.cz>
Message-ID: <CAA4nhyDNcaPk1G4JBaA=t4Nq5r_7VuQBN5xbmEj7CntKcuXhAA@mail.gmail.com>

On Thu, Jul 14, 2011 at 10:01 AM, Michal Zub?? <michal.zubac at comgate.cz> wrote:
> Hi.
>
> I've noticed that outgoing call channels in freetdm/ftmod_libpri are not
> selected exclusively in Q.931 SETUP messages. Channel identification
> "exclusive" bit is 0 so it only means "preferred".
> In some cases freeswitch selected channel which was not ready for dialing
> yet and ISDN operator answered with another channel in PROGRESS message. So
> actual call state messages were passed trough another channel and it
> resulted with lots of VETO and consecutive RESET messages, because of
> channel state confusion in FreeSwitch.
>
> I think that it's better to force channel selection and in some cases choose
> wrong channel and get info about it. It's definitely better, then choose
> something, assume that it is correct although actual communication is done
> on another channel and such calls are "doomed"...
>
> I've attached patch, in which we force selection of ISDN channel. Maybe
> better solution will be some configuration variable, which will control this
> behaviour. Consider.
>
> Does anybody have similar experience and if so, how do you deal with it? Or
> am I mistaken in some of my conclusions?

Hello Michal,

You are totally correct. FreeTDM as of now assumes exclusive requests
of the channel. It seems is a bug in the libpri module not to do so.
Also the last member should be nonisdn because the caller (most of the
time) will be nonisdn most of the time (although idealy we should
figure that out from mod_freetdm). Don't know what the implications of
that
are though, so I won't change that yet until I verify on the spec.

I just committed your change into git, in the future please report
issues like this in jira.freeswitch.org, thanks!

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 engineerzuhairraza at gmail.com  Sun Jul 24 00:05:07 2011
From: engineerzuhairraza at gmail.com (Zohair Raza)
Date: Sun, 24 Jul 2011 01:05:07 +0500
Subject: [Freeswitch-users] Asterisk dial status
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD021FD89DC2@NY1-EXMB-01.ip-soft.net>
References: <6A6B4C284AD15042B429EB9D904544AD021FD89DA5@NY1-EXMB-01.ip-soft.net>
	<1311366180.83556.YahooMailClassic@web161016.mail.bf1.yahoo.com>
	<6A6B4C284AD15042B429EB9D904544AD021FD89DC2@NY1-EXMB-01.ip-soft.net>
Message-ID: <CAF_PTesUUqeSMHcYFMZXK1NBo9GeWX1NawKXot3k4kVVdR71qg@mail.gmail.com>

Hi,

You can also get these status in xml_curl post.

see http://wiki.freeswitch.org/wiki/Mod_xml_curl
http://wiki.freeswitch.org/wiki/Mod_xml_curl_CGI_example


On Sat, Jul 23, 2011 at 2:28 AM, Hector Geraldino <
Hector.Geraldino at ip-soft.net> wrote:

> It does.****
>
> ** **
>
> You can check the events by connecting to FS on inbound mode, or using FS
> outbound socket connection to an application designed by you. ****
>
> ** **
>
> As you already have a perl script to interact with FS, you can modify it to
> connect to FS (inbound mode) and listen for events. You can subscribe to the
> events of a particular call, or filter an specific set of events. I haven?t
> used Asterisk before so I?m not sure how the AGI fits on the FS
> architecture.****
>
> ** **
>
> You can get more info at:****
>
> http://wiki.freeswitch.org/wiki/Mod_event_socket****
>
> ** **
>
> And a working demo at:****
>
> http://wiki.freeswitch.org/wiki/Perl_freeswitch_client_example****
>
> ** **
>
> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sam
> *Sent:* Friday, July 22, 2011 4:23 PM
> *To:* FreeSWITCH Users Help
>
> *Subject:* Re: [Freeswitch-users] Asterisk dial status****
>
> ** **
>
> Sam,
>
> Does this require event sockets? Or how would I check these events?
>
> --- On *Fri, 7/22/11, Hector Geraldino <Hector.Geraldino at ip-soft.net>*wrote:
> ****
>
>
> From: Hector Geraldino <Hector.Geraldino at ip-soft.net>
> Subject: Re: [Freeswitch-users] Asterisk dial status
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Friday, July 22, 2011, 11:57 AM****
>
> Hi Sam,****
>
>  ****
>
> My guess is that you can subscribe to the events of the channel and
> evaluate the channel events. Look at:****
>
>  ****
>
> http://wiki.freeswitch.org/wiki/Event_List****
>
>  ****
>
> specifically the channel events list.****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sam
> *Sent:* Friday, July 22, 2011 2:32 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Asterisk dial status****
>
>  ****
>
> Hi,
>
> I am trying to port over an Asterisk AGI script that does a call bridge to
> Freeswitch using perl, I wanted to know what is the equivalent of the
> asterisk dial status like: ANSWER, NO_ANSWER, CANCEL, BUSY, CONGESTION,
> CHANUNAVAIL ? I wanted to know if what methods can be used to gets these
> statues?****
>
>  ****
>
>
> -----Inline Attachment Follows-----****
>
> _______________________________________________
> 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://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
> _______________________________________________
> 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
>
>


-- 
Regards,
Zohair Raza

www.zuhair.info

*http://ae.linkedin.com/in/zuhairraza** ***
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From mike.burlingame at me.com  Sun Jul 24 00:12:35 2011
From: mike.burlingame at me.com (Mike Burlingame)
Date: Sat, 23 Jul 2011 13:12:35 -0700
Subject: [Freeswitch-users] Call disconnect after transfer
In-Reply-To: <50F5BA53-5C2B-4378-9702-7E7512C57F4A@me.com>
References: <50F5BA53-5C2B-4378-9702-7E7512C57F4A@me.com>
Message-ID: <F807D32D-3684-4B1F-9216-B97ED1173ADC@me.com>

Correction to the original paste bin URL:

I am having a small issue with calls getting disconnected after transfer from a customer signaling is posted at http://pastebin.freeswitch.org/16868

but basic call flow is call is sent to user - user answers - user transfer the call to another user - invite is sent to Freeswitch with out SDP - 200 OK provided to customer with SDP - ACK received from customer with SDP - Freeswitch disconnects call with [INCOMPATIBLE_DESTINATION]

I have 3PCC enabled in FreeSwitch via sofia.conf.xml  <param name="enable-3pcc" value="true"/>

On Jul 20, 2011, at 1:36 PM, Mike Burlingame wrote:

> I am having a small issue with calls getting disconnected after transfer from a customer signaling is posted at http://pastebin.freeswitch.org/16869
> 
> but basic call flow is call is sent to user - user answers - user transfer the call to another user - invite is sent to Freeswitch with out SDP - 200 OK provided to customer with SDP - ACK received from customer with SDP - Freeswitch disconnects call with [INCOMPATIBLE_DESTINATION]
> 
> I have 3PCC enabled in FreeSwitch via sofia.conf.xml  <param name="enable-3pcc" value="true"/>
> _______________________________________________
> 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 kheimerl at cs.berkeley.edu  Sun Jul 24 02:22:20 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Sat, 23 Jul 2011 15:22:20 -0700
Subject: [Freeswitch-users] Using chat command in a dialplan
In-Reply-To: <CACyT-3nXTDh8U7mSd4NOJAhXdW_YNx2BQCqK2-_3OsHoZDYENg@mail.gmail.com>
References: <CACyT-3m7nMTCwav_j_CVqtL-V+T9vFcyGwsrD6uatJ+bkVti3Q@mail.gmail.com>
	<1311149680.32170.264.camel@luna.madrid.commsmundi.com>
	<83217555-012B-4861-AAB9-045B2E78660D@gmail.com>
	<CACyT-3mASOUzN4kX_L4X734Jp1wL5nTbpi-Ytd4TuF-nJA3xaw@mail.gmail.com>
	<CACyT-3nBYP88Ve3oTC6R1WYSPKZkvY08iVTuSxfmm9miFgYEtA@mail.gmail.com>
	<CAFiqYumBRYwpNamX_ig525Hbo+v+s+uVZZ8httTu7Vd7LYk9Qg@mail.gmail.com>
	<CACyT-3=9J+Saqi_c6KuNAjNWyZpgQF+ao3GuUZhLUUD4mw+hog@mail.gmail.com>
	<CAFiqYunELHt2ti+x4OBO=osyrjqn-HQRKhfrjeJV6Mvjpo87Kg@mail.gmail.com>
	<CACyT-3nXTDh8U7mSd4NOJAhXdW_YNx2BQCqK2-_3OsHoZDYENg@mail.gmail.com>
Message-ID: <CACyT-3=0Q+sVvZ61t_J108QDojbT3CHLqvLjo=yGvKr=4-MwhQ@mail.gmail.com>

Just so the robots get this:

Turns out you cannot initiate an action (e.g. originate) in an
existing session. It should probably give a better error than usage.
However, I modified mod_commands to allow originating in an existing
session and it works, so I'll be continuing this discussion on
freeswitch-dev.

Thanks Steven!

On Fri, Jul 22, 2011 at 2:20 AM, Kurtis Heimerl
<kheimerl at cs.berkeley.edu> wrote:
> That's a good plan. I'll try that out this weekend.
>
> On Fri, Jul 22, 2011 at 1:05 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> Never tried it from a dialplan although I'm sure it should work. But you
>> should be able to do it through Lua etc.
>>
>> Maybe you could apply a quick patch to originate in mod_commands to log the
>> arguments it's being passed - perhaps it's being mangled somehow.
>>
>> -Steve
>>
>>
>> On 21 July 2011 23:47, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>>>
>>> Unfortunately, I am not. I'm working on a system that causes dialplan
>>> actions when a chat is received by FS. That's what's going on here,
>>> basically. I want someone to text 9199 and have FS immediately call
>>> them back. I think "originate" is the right way to do that, as I need
>>> to create a brand new session, not bridge to an existing one.
>>>
>>> Anyhow, any other ideas on why this isn't working?
>>>
>>> On Thu, Jul 21, 2011 at 1:23 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>> > Are you trying to join the 2 calls together? If so, use bridge:
>>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge
>>> >
>>> > -Steve
>>> >
>>> >
>>> > On 21 July 2011 07:37, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>>> >>
>>> >> I'm now having a similar problem with using "originate" to create a
>>> >> new call inside a dialplan. This one is particularly perplexing (and
>>> >> small enough not to warrant pastebin)
>>> >>
>>> >> Here's the important bits of the dialplan:
>>> >> ? ? ? ?<action application="log" data="(user/${username}
>>> >> ${destination_number})"/>
>>> >> ? ? ? ?<action application="set"
>>> >> data="api_result=${originate(user/${username}
>>> >> ${destination_number})}"/>
>>> >>
>>> >> and the log output
>>> >> EXECUTE 9199 log((user/1304 at 192.168.1.144 9199))
>>> >> 2011-07-20 23:17:12.955419 [DEBUG] mod_dptools.c:1202 9199)
>>> >> EXECUTE 9199 set(api_result=-USAGE <call url>
>>> >> <exten>|&<application_name>(<app_args>) [<dialplan>] [<context>]
>>> >> [<cid_name>] [<cid_num>] [<timeout_sec>]
>>> >> )
>>> >>
>>> >> So it's somehow mangling the originate commands. If I run the
>>> >> originate command directly (with the arguments logged!) it works just
>>> >> fine.
>>> >> originate user/1304 at 192.168.1.144 9199
>>> >>
>>> >> What am I missing here? This seems so simple... ?but I'm at my wit's
>>> >> end.
>>> >>
>>> >> Thanks for any direction in advance!
>>> >>
>>> >>
>>> >> On Wed, Jul 20, 2011 at 7:11 PM, Kurtis Heimerl
>>> >> <kheimerl at cs.berkeley.edu> wrote:
>>> >> > Exactly what I was looking for. Thanks!
>>> >> >
>>> >> > On Wed, Jul 20, 2011 at 1:21 AM, Steven Ayre <steveayre at gmail.com>
>>> >> > wrote:
>>> >> >> Kurtis,
>>> >> >>
>>> >> >> *Any* api can be run from the dialplan. It's just a slightly
>>> >> >> different
>>> >> >> syntax - the link Francois gave you will show you how.
>>> >> >>
>>> >> >> Steve on iPhone
>>> >> >>
>>> >> >> On 20 Jul 2011, at 09:14, Fran?ois Delawarde
>>> >> >> <fdelawarde at wirelessmundi.com> wrote:
>>> >> >>
>>> >> >>> Hi,
>>> >> >>>
>>> >> >>> Take a look at:
>>> >> >>> http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
>>> >> >>>
>>> >> >>>
>>> >> >>> Fran?ois.
>>> >> >>>
>>> >> >>> On Tue, 2011-07-19 at 23:00 -0700, Kurtis Heimerl wrote:
>>> >> >>>> I feel somewhat embarrassed, but I can't figure out how to cause a
>>> >> >>>> chat event to go off from the dialplan. I've tried:
>>> >> >>>>
>>> >> >>>> <action application="chat"
>>> >> >>>>
>>> >> >>>>
>>> >> >>>> data="sip|${username}|${destination_number}@${domain_name}|${msg_body}"/>
>>> >> >>>>
>>> >> >>>> and
>>> >> >>>>
>>> >> >>>> <action application="eval" data="chat
>>> >> >>>>
>>> >> >>>> sip|${username}|${destination_number}@${domain_name}|${msg_body}"/>
>>> >> >>>>
>>> >> >>>> and neither worked. The wiki says it's part of the API
>>> >> >>>> (http://wiki.freeswitch.org/wiki/Mod_dptools) but I'm pretty sure
>>> >> >>>> that's just a command-line API. Is there a way to run a
>>> >> >>>> command-line
>>> >> >>>> call from the dialplan?
>>> >> >>>>
>>> >> >>>> Any direction would be appreciated.
>>> >> >>>>
>>> >> >>>> _______________________________________________
>>> >> >>>> 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
>>> >> >>>
>>> >> >>>
>>> >> >>>
>>> >> >>> _______________________________________________
>>> >> >>> 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
>>> >> >>
>>> >> >> _______________________________________________
>>> >> >> 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
>>> >> >>
>>> >> >
>>> >>
>>> >> _______________________________________________
>>> >> 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
>>> >
>>> >
>>> > _______________________________________________
>>> > 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
>>> >
>>> >
>>>
>>> _______________________________________________
>>> 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
>>
>>
>> _______________________________________________
>> 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 tculjaga at gmail.com  Sun Jul 24 12:36:04 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sun, 24 Jul 2011 10:36:04 +0200
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAO4LeU3SKmO4oSHMthnKzhV2w489JRMTxdd3N5hFmdDgbrTyNA@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
	<CAFiqYukgW1UOgxNUZs8rPvWNmXx5sDVmGS_D+r_xdLMsXg_i4Q@mail.gmail.com>
	<CAJoKKdX8J68pmUogKgp1Ew4ACpkHntiHoZ2Rpx5JNhepsOVVvw@mail.gmail.com>
	<CAFiqYunZUs0d-kQVTxXEvuqVveiri6WFFW6oT3UYujPGoa2H-w@mail.gmail.com>
	<CAJoKKdXhHyQUXKCzaDc3GSu5m9A3Jdv2m9K=FaY7vt=W1OD5Qg@mail.gmail.com>
	<CAFiqYumuca1OMQ13di425JFjs2dhPeLAi2e0FPu-kG2odJSVqQ@mail.gmail.com>
	<CAKbxfG-GSbSm-wq5+YnT0jvtiiP32xJDgc9pqV1TUXj69CwDcg@mail.gmail.com>
	<CAJoKKdUqV_ygenNTd+Z3hnoC2eXeiPuMOM2PvaChEm3kZXgj0A@mail.gmail.com>
	<CAO4LeU3SKmO4oSHMthnKzhV2w489JRMTxdd3N5hFmdDgbrTyNA@mail.gmail.com>
Message-ID: <CAJoKKdUosKzjxLwmwjQUoWQHjDKsx=DyBAufSG_HHGSR3DWNAQ@mail.gmail.com>

I just took:

commit 3c74f391f4f4dc03b687f0d6f45dfd19c17555d3
Author: Mathieu Parent <math.parent at gmail.com>
Date:   Fri Jul 22 23:05:56 2011 +0200

and running for 2 days as well.... hitting FS with SIPP at quite a high CPS

its been 14178302 sessions since startup and the memory keeps stable.

DATE                          PID    RSS      %MEM
2011-07-24 12:30:02 13268 239852  5.9


$DAT_TIM,$SIZE,$RSS,$SHARED_CLEAN,$SHARED_DIRTY,$PRIVATE_CLEAN,$PRIVATE_DIRTY,$SWAP
2011-07-24 12:31:11,46752,46644,0,0,0,46644,0


I don't see the problem anymore!


Thanks for you effort.



On Tue, Jul 19, 2011 at 5:50 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> I'm using commit:
>
> commit 130e1c87746b0596460358d77d6aabbfe41a0072
> Author: Jeff Lenk <jeff at jefflenk.com>
> Date:   Sat Jul 16 19:13:27 2011 -0500
>
> and after 2 days and more then 3600000 sessions, the memory seems to be
> stable so the problem is probably fixed.
> I'll update this info in the next days.
>
> Stephen
>
> On Thu, Jul 14, 2011 at 5:48 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>>
>>
>> On Thu, Jul 14, 2011 at 5:11 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> you can use gigs before you stop growing in some cases esp on sipp test.
>>> you need a lot more than 5 min to judge it.
>>> more like dozen or more hours.
>>>
>>> yap, im running it for a very long time ... the example was just a snip
>> ... and i hit the swap right now :=) .... after 2 hours of sipp at 30
>> CPS....
>>
>> start: 2011-07-14 12:45:43
>> stop: 2011-07-14 14:31:13
>>
>>
>> if i use less load it will grow slowly but it will end into swap... its
>> just a matter of time.
>>
>> So, how can we troubleshoot this ?
>>
>> im going to open a jira to move this out of Users Help mailing list...
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 b_ball_henry at hotmail.com  Sun Jul 24 14:36:15 2011
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Sun, 24 Jul 2011 18:36:15 +0800
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAJoKKdUosKzjxLwmwjQUoWQHjDKsx=DyBAufSG_HHGSR3DWNAQ@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
	<CAFiqYukgW1UOgxNUZs8rPvWNmXx5sDVmGS_D+r_xdLMsXg_i4Q@mail.gmail.com>
	<CAJoKKdX8J68pmUogKgp1Ew4ACpkHntiHoZ2Rpx5JNhepsOVVvw@mail.gmail.com>
	<CAFiqYunZUs0d-kQVTxXEvuqVveiri6WFFW6oT3UYujPGoa2H-w@mail.gmail.com>
	<CAJoKKdXhHyQUXKCzaDc3GSu5m9A3Jdv2m9K=FaY7vt=W1OD5Qg@mail.gmail.com>
	<CAFiqYumuca1OMQ13di425JFjs2dhPeLAi2e0FPu-kG2odJSVqQ@mail.gmail.com>
	<CAKbxfG-GSbSm-wq5+YnT0jvtiiP32xJDgc9pqV1TUXj69CwDcg@mail.gmail.com>
	<CAJoKKdUqV_ygenNTd+Z3hnoC2eXeiPuMOM2PvaChEm3kZXgj0A@mail.gmail.com>
	<CAO4LeU3SKmO4oSHMthnKzhV2w489JRMTxdd3N5hFmdDgbrTyNA@mail.gmail.com>
	<CAJoKKdUosKzjxLwmwjQUoWQHjDKsx=DyBAufSG_HHGSR3DWNAQ@mail.gmail.com>
Message-ID: <CAP_G397PRcGq8d7yTOZcJw_EN=mHRw6gD60hqaLseeazK=y60A@mail.gmail.com>

What CPS did you get?

Henry

On Sun, Jul 24, 2011 at 4:36 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> I just took:
>
> commit 3c74f391f4f4dc03b687f0d6f45dfd19c17555d3
> Author: Mathieu Parent <math.parent at gmail.com>
> Date:   Fri Jul 22 23:05:56 2011 +0200
>
> and running for 2 days as well.... hitting FS with SIPP at quite a high CPS
>
> its been 14178302 sessions since startup and the memory keeps stable.
>
> DATE                          PID    RSS      %MEM
> 2011-07-24 12:30:02 13268 239852  5.9
>
>
>
> $DAT_TIM,$SIZE,$RSS,$SHARED_CLEAN,$SHARED_DIRTY,$PRIVATE_CLEAN,$PRIVATE_DIRTY,$SWAP
> 2011-07-24 12:31:11,46752,46644,0,0,0,46644,0
>
>
> I don't see the problem anymore!
>
>
> Thanks for you effort.
>
>
>
> On Tue, Jul 19, 2011 at 5:50 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> I'm using commit:
>>
>> commit 130e1c87746b0596460358d77d6aabbfe41a0072
>> Author: Jeff Lenk <jeff at jefflenk.com>
>> Date:   Sat Jul 16 19:13:27 2011 -0500
>>
>> and after 2 days and more then 3600000 sessions, the memory seems to be
>> stable so the problem is probably fixed.
>> I'll update this info in the next days.
>>
>> Stephen
>>
>> On Thu, Jul 14, 2011 at 5:48 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>>
>>>
>>> On Thu, Jul 14, 2011 at 5:11 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> you can use gigs before you stop growing in some cases esp on sipp test.
>>>> you need a lot more than 5 min to judge it.
>>>> more like dozen or more hours.
>>>>
>>>> yap, im running it for a very long time ... the example was just a snip
>>> ... and i hit the swap right now :=) .... after 2 hours of sipp at 30
>>> CPS....
>>>
>>> start: 2011-07-14 12:45:43
>>> stop: 2011-07-14 14:31:13
>>>
>>>
>>> if i use less load it will grow slowly but it will end into swap... its
>>> just a matter of time.
>>>
>>> So, how can we troubleshoot this ?
>>>
>>> im going to open a jira to move this out of Users Help mailing list...
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 nsirugudi at gmail.com  Sun Jul 24 16:40:30 2011
From: nsirugudi at gmail.com (Narendra Sirugudi)
Date: Sun, 24 Jul 2011 18:10:30 +0530
Subject: [Freeswitch-users] Sofia stack sip rfc conformance.
Message-ID: <CAP18h9oKKw=WMs4Sb9kQ08sBRCwdgZqk9Kd2NLcjd+PMsq8Lxw@mail.gmail.com>

Hi All,

I wanted to know what all SIP standards/rfc(s) does Sofia stack conform to ?

I could not find this informantion anywhere so far.

thanks,
--naren
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From gcd at i.ph  Sun Jul 24 16:49:17 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Sun, 24 Jul 2011 20:49:17 +0800
Subject: [Freeswitch-users] Sofia stack sip rfc conformance.
In-Reply-To: <CAP18h9oKKw=WMs4Sb9kQ08sBRCwdgZqk9Kd2NLcjd+PMsq8Lxw@mail.gmail.com>
References: <CAP18h9oKKw=WMs4Sb9kQ08sBRCwdgZqk9Kd2NLcjd+PMsq8Lxw@mail.gmail.com>
Message-ID: <CAHKqMb5_BX7qHqYhEB_deCVYC-uVj+1Fv98YQq=xL-3=xbKqng@mail.gmail.com>

hi naren,

try it here:
http://sofia-sip.sourceforge.net/

-nandy
On Sun, Jul 24, 2011 at 8:40 PM, Narendra Sirugudi <nsirugudi at gmail.com>wrote:

> Hi All,
>
> I wanted to know what all SIP standards/rfc(s) does Sofia stack conform to
> ?
>
> I could not find this informantion anywhere so far.
>
> thanks,
> --naren
>
> _______________________________________________
> 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 peter.olsson at visionutveckling.se  Sun Jul 24 16:53:24 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 24 Jul 2011 14:53:24 +0200
Subject: [Freeswitch-users] Sofia stack sip rfc conformance.
In-Reply-To: <CAP18h9oKKw=WMs4Sb9kQ08sBRCwdgZqk9Kd2NLcjd+PMsq8Lxw@mail.gmail.com>
References: <CAP18h9oKKw=WMs4Sb9kQ08sBRCwdgZqk9Kd2NLcjd+PMsq8Lxw@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F89@cooper>

I found this, it should be what you're looking for?

http://sofia-sip.sourceforge.net/refdocs/sofia_sip_conformance.html

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Narendra Sirugudi [nsirugudi at gmail.com]
Skickat: den 24 juli 2011 14:40
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] Sofia stack sip rfc conformance.

Hi All,

I wanted to know what all SIP standards/rfc(s) does Sofia stack conform to ?

I could not find this informantion anywhere so far.

thanks,
--naren
!DSPAM:4e2c136232763277319425!


From nsirugudi at gmail.com  Sun Jul 24 16:57:35 2011
From: nsirugudi at gmail.com (Narendra Sirugudi)
Date: Sun, 24 Jul 2011 18:27:35 +0530
Subject: [Freeswitch-users] Sofia stack sip rfc conformance.
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F89@cooper>
References: <CAP18h9oKKw=WMs4Sb9kQ08sBRCwdgZqk9Kd2NLcjd+PMsq8Lxw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F89@cooper>
Message-ID: <CAP18h9qV3D6W9BYFZx+xiZ=sD6=Fm_hAzK+BvjD58-9NoQHRng@mail.gmail.com>

Peter, Nandy,

Thanks for the quick reply.
I got the information i needed.

thanks,
--naren

On Sun, Jul 24, 2011 at 6:23 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> I found this, it should be what you're looking for?
>
> http://sofia-sip.sourceforge.net/refdocs/sofia_sip_conformance.html
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Narendra Sirugudi
> [nsirugudi at gmail.com]
> Skickat: den 24 juli 2011 14:40
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] Sofia stack sip rfc conformance.
>
> Hi All,
>
> I wanted to know what all SIP standards/rfc(s) does Sofia stack conform to
> ?
>
> I could not find this informantion anywhere so far.
>
> thanks,
> --naren
> !DSPAM:4e2c136232763277319425!
>
> _______________________________________________
> 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 tculjaga at gmail.com  Sun Jul 24 17:26:53 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sun, 24 Jul 2011 15:26:53 +0200
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAP_G397PRcGq8d7yTOZcJw_EN=mHRw6gD60hqaLseeazK=y60A@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
	<CAFiqYukgW1UOgxNUZs8rPvWNmXx5sDVmGS_D+r_xdLMsXg_i4Q@mail.gmail.com>
	<CAJoKKdX8J68pmUogKgp1Ew4ACpkHntiHoZ2Rpx5JNhepsOVVvw@mail.gmail.com>
	<CAFiqYunZUs0d-kQVTxXEvuqVveiri6WFFW6oT3UYujPGoa2H-w@mail.gmail.com>
	<CAJoKKdXhHyQUXKCzaDc3GSu5m9A3Jdv2m9K=FaY7vt=W1OD5Qg@mail.gmail.com>
	<CAFiqYumuca1OMQ13di425JFjs2dhPeLAi2e0FPu-kG2odJSVqQ@mail.gmail.com>
	<CAKbxfG-GSbSm-wq5+YnT0jvtiiP32xJDgc9pqV1TUXj69CwDcg@mail.gmail.com>
	<CAJoKKdUqV_ygenNTd+Z3hnoC2eXeiPuMOM2PvaChEm3kZXgj0A@mail.gmail.com>
	<CAO4LeU3SKmO4oSHMthnKzhV2w489JRMTxdd3N5hFmdDgbrTyNA@mail.gmail.com>
	<CAJoKKdUosKzjxLwmwjQUoWQHjDKsx=DyBAufSG_HHGSR3DWNAQ@mail.gmail.com>
	<CAP_G397PRcGq8d7yTOZcJw_EN=mHRw6gD60hqaLseeazK=y60A@mail.gmail.com>
Message-ID: <CAJoKKdVam_C_Xww-hmwRRaxoaBDPfJ-ETJMZjVZYAxfv+9nuNA@mail.gmail.com>

i got peaks of 60 CPS as i was trying to simulate my real traffic pattern.

Anyhow, a year ago o reached like 400 - 450 CPS on an average server


Also, bear in mind it was just signaling (no RTP included)

INVITE =>
       <= 100 Trying
       <= 300 Multiple choices


Also i can provide memory log (5 minute interval) to construct a chart if
someone is interested.

T.



On Sun, Jul 24, 2011 at 12:36 PM, Henry Huang <b_ball_henry at hotmail.com>wrote:

> What CPS did you get?
>
> Henry
>
> On Sun, Jul 24, 2011 at 4:36 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> I just took:
>>
>> commit 3c74f391f4f4dc03b687f0d6f45dfd19c17555d3
>> Author: Mathieu Parent <math.parent at gmail.com>
>> Date:   Fri Jul 22 23:05:56 2011 +0200
>>
>> and running for 2 days as well.... hitting FS with SIPP at quite a high
>> CPS
>>
>> its been 14178302 sessions since startup and the memory keeps stable.
>>
>> DATE                          PID    RSS      %MEM
>> 2011-07-24 12:30:02 13268 239852  5.9
>>
>>
>>
>> $DAT_TIM,$SIZE,$RSS,$SHARED_CLEAN,$SHARED_DIRTY,$PRIVATE_CLEAN,$PRIVATE_DIRTY,$SWAP
>> 2011-07-24 12:31:11,46752,46644,0,0,0,46644,0
>>
>>
>> I don't see the problem anymore!
>>
>>
>> Thanks for you effort.
>>
>>
>>
>> On Tue, Jul 19, 2011 at 5:50 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>
>>> I'm using commit:
>>>
>>> commit 130e1c87746b0596460358d77d6aabbfe41a0072
>>> Author: Jeff Lenk <jeff at jefflenk.com>
>>> Date:   Sat Jul 16 19:13:27 2011 -0500
>>>
>>> and after 2 days and more then 3600000 sessions, the memory seems to be
>>> stable so the problem is probably fixed.
>>> I'll update this info in the next days.
>>>
>>> Stephen
>>>
>>> On Thu, Jul 14, 2011 at 5:48 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>>
>>>>
>>>>
>>>> On Thu, Jul 14, 2011 at 5:11 PM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> you can use gigs before you stop growing in some cases esp on sipp
>>>>> test.
>>>>> you need a lot more than 5 min to judge it.
>>>>> more like dozen or more hours.
>>>>>
>>>>> yap, im running it for a very long time ... the example was just a snip
>>>> ... and i hit the swap right now :=) .... after 2 hours of sipp at 30
>>>> CPS....
>>>>
>>>> start: 2011-07-14 12:45:43
>>>> stop: 2011-07-14 14:31:13
>>>>
>>>>
>>>> if i use less load it will grow slowly but it will end into swap... its
>>>> just a matter of time.
>>>>
>>>> So, how can we troubleshoot this ?
>>>>
>>>> im going to open a jira to move this out of Users Help mailing list...
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 lakersman2006 at yahoo.com  Sun Jul 24 22:45:12 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Sun, 24 Jul 2011 11:45:12 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk dial status
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD021FD89DC2@NY1-EXMB-01.ip-soft.net>
Message-ID: <1311533112.57294.YahooMailClassic@web161004.mail.bf1.yahoo.com>

For the life of me I am still unable to get the bridged call status, when I subscribe to an "event" I am not here is my perl code snippet below, what am I doing wrong?


my $fs = new ESL::ESLconnection($host, $port, $password);

$session->answer();

if ($session->ready ()) 
{
??? #set bridge settings
??? $session->execute("set", "ringback=$ringback_tone");?? 
??? $session->execute("set", "instant_ringback=true");??? ??? 
??? $session->execute("set", "ignore_early_media=false"); 
??? $session->execute("set", "call_timeout=60");??? ??? ??? ??? ??? ??? ??? 
??? $session->execute("set", "progress_timeout=15");
??? $session->execute("set", "origination_caller_id_number=$caller_id");
??? $session->execute("set", "continue_on_fail=false");
??? $session->execute("bridge", "sofia/gateway/provider1/$destination_number");
???? 
??? $fs->sendRecv("event plain ALL");
??? $fs->send("linger");
??? 
??? while($fs->connected())
??? {
??? ??? $data = $fs->recvEvent();
??? ??? 
??? ??? if ( $data =~ m/Answered-State: (\w+)/m ) 
??? ??? {
??? ??? ?? my $state = $1;
??? ??? 
??? ??? ?? #print "Channel state is $state\n";
??? ??? 
??? ??? ?? if ( $state eq 'answered' ) {
??? ??? 
??? ??? ??? ?freeswitch::consoleLog("INFO", "CALL ANSWERED\n");
??? ??? 
??? ??? ?? }
??? ??? ?? else
??? ??? ?? {
??? ??? ?? ??? ?freeswitch::consoleLog("INFO", "NO CALLS ANSWERED\n");
??? ??? ?? }
??? ??? ?}??? 
??? }
}
?? 
$session->hangup();
?
return 1;

--- On Fri, 7/22/11, Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:

From: Hector Geraldino <Hector.Geraldino at ip-soft.net>
Subject: Re: [Freeswitch-users] Asterisk dial status
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Friday, July 22, 2011, 2:28 PM

It does. ?You can check the events by connecting to FS on inbound mode, or using FS outbound socket connection to an application designed by you.  ?As you already have a perl script to interact with FS, you can modify it to connect to FS (inbound mode) and listen for events. You can subscribe to the events of a particular call, or filter an specific set of events. I haven?t used Asterisk before so I?m not sure how the AGI fits on the FS architecture. ?You can get more info at:http://wiki.freeswitch.org/wiki/Mod_event_socket ?And a working demo at: http://wiki.freeswitch.org/wiki/Perl_freeswitch_client_example ? ? ?From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sam
Sent: Friday, July 22, 2011 4:23 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Asterisk dial status ?Sam,

Does this require event sockets? Or how would I check these events?

--- On Fri, 7/22/11, Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:
From: Hector Geraldino <Hector.Geraldino at ip-soft.net>
Subject: Re: [Freeswitch-users] Asterisk dial status
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Friday, July 22, 2011, 11:57 AMHi Sam,?My guess is that you can subscribe to the events of the channel and evaluate the channel events. Look at:?http://wiki.freeswitch.org/wiki/Event_List?specifically the channel events list.?From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sam
Sent: Friday, July 22, 2011 2:32 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Asterisk dial status?Hi,

I am trying to port over an Asterisk AGI script that does a call bridge to Freeswitch using perl, I wanted to know what is the equivalent of the asterisk dial status like: ANSWER, NO_ANSWER, CANCEL, BUSY, CONGESTION, CHANUNAVAIL ? I wanted to know if what methods can be used to gets these statues??
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From peter.olsson at visionutveckling.se  Sun Jul 24 23:01:35 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 24 Jul 2011 21:01:35 +0200
Subject: [Freeswitch-users] Asterisk dial status
In-Reply-To: <1311533112.57294.YahooMailClassic@web161004.mail.bf1.yahoo.com>
References: <6A6B4C284AD15042B429EB9D904544AD021FD89DC2@NY1-EXMB-01.ip-soft.net>,
	<1311533112.57294.YahooMailClassic@web161004.mail.bf1.yahoo.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F8A@cooper>

I've never used Perl for this, but I'm pretty sure that the last line won't return until the bridge is done (because of call finishied or failed).

What exactly do you want to accomplish - I think there is probably an easier way for what you're trying to solve.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Sam [lakersman2006 at yahoo.com]
Skickat: den 24 juli 2011 20:45
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Asterisk dial status

For the life of me I am still unable to get the bridged call status, when I subscribe to an "event" I am not here is my perl code snippet below, what am I doing wrong?


my $fs = new ESL::ESLconnection($host, $port, $password);

$session->answer();

if ($session->ready ())
{
    #set bridge settings
    $session->execute("set", "ringback=$ringback_tone");
    $session->execute("set", "instant_ringback=true");
    $session->execute("set", "ignore_early_media=false");
    $session->execute("set", "call_timeout=60");
    $session->execute("set", "progress_timeout=15");
    $session->execute("set", "origination_caller_id_number=$caller_id");
    $session->execute("set", "continue_on_fail=false");
    $session->execute("bridge", "sofia/gateway/provider1/$destination_number");

    $fs->sendRecv("event plain ALL");
    $fs->send("linger");

    while($fs->connected())
    {
        $data = $fs->recvEvent();

        if ( $data =~ m/Answered-State: (\w+)/m )
        {
           my $state = $1;

           #print "Channel state is $state\n";

           if ( $state eq 'answered' ) {

             freeswitch::consoleLog("INFO", "CALL ANSWERED\n");

           }
           else
           {
                freeswitch::consoleLog("INFO", "NO CALLS ANSWERED\n");
           }
         }
    }
}

$session->hangup();

return 1;

--- On Fri, 7/22/11, Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:

From: Hector Geraldino <Hector.Geraldino at ip-soft.net>
Subject: Re: [Freeswitch-users] Asterisk dial status
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Friday, July 22, 2011, 2:28 PM


It does.



You can check the events by connecting to FS on inbound mode, or using FS outbound socket connection to an application designed by you.



As you already have a perl script to interact with FS, you can modify it to connect to FS (inbound mode) and listen for events. You can subscribe to the events of a particular call, or filter an specific set of events. I haven?t used Asterisk before so I?m not sure how the AGI fits on the FS architecture.



You can get more info at:

http://wiki.freeswitch.org/wiki/Mod_event_socket



And a working demo at:

http://wiki.freeswitch.org/wiki/Perl_freeswitch_client_example







From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sam
Sent: Friday, July 22, 2011 4:23 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Asterisk dial status



Sam,

Does this require event sockets? Or how would I check these events?

--- On Fri, 7/22/11, Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:

From: Hector Geraldino <Hector.Geraldino at ip-soft.net>
Subject: Re: [Freeswitch-users] Asterisk dial status
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Friday, July 22, 2011, 11:57 AM

Hi Sam,



My guess is that you can subscribe to the events of the channel and evaluate the channel events. Look at:



http://wiki.freeswitch.org/wiki/Event_List



specifically the channel events list.



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sam
Sent: Friday, July 22, 2011 2:32 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Asterisk dial status



Hi,

I am trying to port over an Asterisk AGI script that does a call bridge to Freeswitch using perl, I wanted to know what is the equivalent of the asterisk dial status like: ANSWER, NO_ANSWER, CANCEL, BUSY, CONGESTION, CHANUNAVAIL ? I wanted to know if what methods can be used to gets these statues?




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

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




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

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FreeSWITCH-users at lists.freeswitch.org</mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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!DSPAM:4e2c68e932761895611653!


From acrow at integrafin.co.uk  Mon Jul 25 00:16:13 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Sun, 24 Jul 2011 21:16:13 +0100
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAJoKKdVam_C_Xww-hmwRRaxoaBDPfJ-ETJMZjVZYAxfv+9nuNA@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>	<CAFiqYukgW1UOgxNUZs8rPvWNmXx5sDVmGS_D+r_xdLMsXg_i4Q@mail.gmail.com>	<CAJoKKdX8J68pmUogKgp1Ew4ACpkHntiHoZ2Rpx5JNhepsOVVvw@mail.gmail.com>	<CAFiqYunZUs0d-kQVTxXEvuqVveiri6WFFW6oT3UYujPGoa2H-w@mail.gmail.com>	<CAJoKKdXhHyQUXKCzaDc3GSu5m9A3Jdv2m9K=FaY7vt=W1OD5Qg@mail.gmail.com>	<CAFiqYumuca1OMQ13di425JFjs2dhPeLAi2e0FPu-kG2odJSVqQ@mail.gmail.com>	<CAKbxfG-GSbSm-wq5+YnT0jvtiiP32xJDgc9pqV1TUXj69CwDcg@mail.gmail.com>	<CAJoKKdUqV_ygenNTd+Z3hnoC2eXeiPuMOM2PvaChEm3kZXgj0A@mail.gmail.com>	<CAO4LeU3SKmO4oSHMthnKzhV2w489JRMTxdd3N5hFmdDgbrTyNA@mail.gmail.com>	<CAJoKKdUosKzjxLwmwjQUoWQHjDKsx=DyBAufSG_HHGSR3DWNAQ@mail.gmail.com>	<CAP_G397PRcGq8d7yTOZcJw_EN=mHRw6gD60hqaLseeazK=y60A@mail.gmail.com>
	<CAJoKKdVam_C_Xww-hmwRRaxoaBDPfJ-ETJMZjVZYAxfv+9nuNA@mail.gmail.com>
Message-ID: <4E2C7D8D.7040305@integrafin.co.uk>

On 24/07/11 14:26, Tihomir Culjaga wrote:
> i got peaks of 60 CPS as i was trying to simulate my real traffic pattern.
>
> Anyhow, a year ago o reached like 400 - 450 CPS on an average server
>

Apologies, are you implying that FS performance has dropped by a factor 
of 9 in a year?

I hope this isn't the case!

Alex

-- 
This message is intended only for the addressee and may contain
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the message along with any attachments and notify us immediately.

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Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)



From gmaruzz at gmail.com  Mon Jul 25 00:48:35 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 24 Jul 2011 22:48:35 +0200
Subject: [Freeswitch-users] memory leak
In-Reply-To: <4E2C7D8D.7040305@integrafin.co.uk>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
	<CAFiqYukgW1UOgxNUZs8rPvWNmXx5sDVmGS_D+r_xdLMsXg_i4Q@mail.gmail.com>
	<CAJoKKdX8J68pmUogKgp1Ew4ACpkHntiHoZ2Rpx5JNhepsOVVvw@mail.gmail.com>
	<CAFiqYunZUs0d-kQVTxXEvuqVveiri6WFFW6oT3UYujPGoa2H-w@mail.gmail.com>
	<CAJoKKdXhHyQUXKCzaDc3GSu5m9A3Jdv2m9K=FaY7vt=W1OD5Qg@mail.gmail.com>
	<CAFiqYumuca1OMQ13di425JFjs2dhPeLAi2e0FPu-kG2odJSVqQ@mail.gmail.com>
	<CAKbxfG-GSbSm-wq5+YnT0jvtiiP32xJDgc9pqV1TUXj69CwDcg@mail.gmail.com>
	<CAJoKKdUqV_ygenNTd+Z3hnoC2eXeiPuMOM2PvaChEm3kZXgj0A@mail.gmail.com>
	<CAO4LeU3SKmO4oSHMthnKzhV2w489JRMTxdd3N5hFmdDgbrTyNA@mail.gmail.com>
	<CAJoKKdUosKzjxLwmwjQUoWQHjDKsx=DyBAufSG_HHGSR3DWNAQ@mail.gmail.com>
	<CAP_G397PRcGq8d7yTOZcJw_EN=mHRw6gD60hqaLseeazK=y60A@mail.gmail.com>
	<CAJoKKdVam_C_Xww-hmwRRaxoaBDPfJ-ETJMZjVZYAxfv+9nuNA@mail.gmail.com>
	<4E2C7D8D.7040305@integrafin.co.uk>
Message-ID: <CALXCt0oaUWKB9G6D5adVOmWVFyqYZRDzzN0QfAKvWgOm1xW8oQ@mail.gmail.com>

We're trying to be nice toward other similar projects, both closed and
open source.

At Cluecon we'll announce the reduction to 0.5 cpm (calls per minute),
and optional random deadlocks.

-giovanni



On 7/24/11, Alex Crow <acrow at integrafin.co.uk> wrote:
> On 24/07/11 14:26, Tihomir Culjaga wrote:
>> i got peaks of 60 CPS as i was trying to simulate my real traffic pattern.
>>
>> Anyhow, a year ago o reached like 400 - 450 CPS on an average server
>>
>
> Apologies, are you implying that FS performance has dropped by a factor
> of 9 in a year?
>
> I hope this isn't the case!
>
> Alex
>
> --
> This message is intended only for the addressee and may contain
> confidential information.  Unless you are that person, you may not
> disclose its contents or use it in any way and are requested to delete
> the message along with any attachments and notify us immediately.
>
> "Transact" is operated by Integrated Financial Arrangements plc
> Domain House, 5-7 Singer Street, London  EC2A 4BQ
> Tel: (020) 7608 4900 Fax: (020) 7608 5300
> (Registered office: as above; Registered in England and Wales under number:
> 3727592)
> Authorised and regulated by the Financial Services Authority (entered on the
> FSA Register; number: 190856)
>
>
> _______________________________________________
> 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
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From avi at avimarcus.net  Mon Jul 25 01:41:17 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 25 Jul 2011 00:41:17 +0300
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CALXCt0oaUWKB9G6D5adVOmWVFyqYZRDzzN0QfAKvWgOm1xW8oQ@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
	<CAFiqYukgW1UOgxNUZs8rPvWNmXx5sDVmGS_D+r_xdLMsXg_i4Q@mail.gmail.com>
	<CAJoKKdX8J68pmUogKgp1Ew4ACpkHntiHoZ2Rpx5JNhepsOVVvw@mail.gmail.com>
	<CAFiqYunZUs0d-kQVTxXEvuqVveiri6WFFW6oT3UYujPGoa2H-w@mail.gmail.com>
	<CAJoKKdXhHyQUXKCzaDc3GSu5m9A3Jdv2m9K=FaY7vt=W1OD5Qg@mail.gmail.com>
	<CAFiqYumuca1OMQ13di425JFjs2dhPeLAi2e0FPu-kG2odJSVqQ@mail.gmail.com>
	<CAKbxfG-GSbSm-wq5+YnT0jvtiiP32xJDgc9pqV1TUXj69CwDcg@mail.gmail.com>
	<CAJoKKdUqV_ygenNTd+Z3hnoC2eXeiPuMOM2PvaChEm3kZXgj0A@mail.gmail.com>
	<CAO4LeU3SKmO4oSHMthnKzhV2w489JRMTxdd3N5hFmdDgbrTyNA@mail.gmail.com>
	<CAJoKKdUosKzjxLwmwjQUoWQHjDKsx=DyBAufSG_HHGSR3DWNAQ@mail.gmail.com>
	<CAP_G397PRcGq8d7yTOZcJw_EN=mHRw6gD60hqaLseeazK=y60A@mail.gmail.com>
	<CAJoKKdVam_C_Xww-hmwRRaxoaBDPfJ-ETJMZjVZYAxfv+9nuNA@mail.gmail.com>
	<4E2C7D8D.7040305@integrafin.co.uk>
	<CALXCt0oaUWKB9G6D5adVOmWVFyqYZRDzzN0QfAKvWgOm1xW8oQ@mail.gmail.com>
Message-ID: <CALNZuwkiQNN4kKwv=zwKQm-vXihXQcbsjEb2XH5YpiStFYqkZg@mail.gmail.com>

Alex, he said he only tested 50-60 because that's what he has in actual
real-use.
The test he mentioned the year before presumably was full-capacity, rather
than throttling back to test normal-usage.
-Avi



On Sun, Jul 24, 2011 at 11:48 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> We're trying to be nice toward other similar projects, both closed and
> open source.
>
> At Cluecon we'll announce the reduction to 0.5 cpm (calls per minute),
> and optional random deadlocks.
>
> -giovanni
>
>
>
> On 7/24/11, Alex Crow <acrow at integrafin.co.uk> wrote:
> > On 24/07/11 14:26, Tihomir Culjaga wrote:
> >> i got peaks of 60 CPS as i was trying to simulate my real traffic
> pattern.
> >>
> >> Anyhow, a year ago o reached like 400 - 450 CPS on an average server
> >>
> >
> > Apologies, are you implying that FS performance has dropped by a factor
> > of 9 in a year?
> >
> > I hope this isn't the case!
> >
> > Alex
> >
> > --
> > This message is intended only for the addressee and may contain
> > confidential information.  Unless you are that person, you may not
> > disclose its contents or use it in any way and are requested to delete
> > the message along with any attachments and notify us immediately.
> >
> > "Transact" is operated by Integrated Financial Arrangements plc
> > Domain House, 5-7 Singer Street, London  EC2A 4BQ
> > Tel: (020) 7608 4900 Fax: (020) 7608 5300
> > (Registered office: as above; Registered in England and Wales under
> number:
> > 3727592)
> > Authorised and regulated by the Financial Services Authority (entered on
> the
> > FSA Register; number: 190856)
> >
> >
> > _______________________________________________
> > 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
> >
>
> --
> Sent from my mobile device
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lakersman2006 at yahoo.com  Mon Jul 25 02:20:13 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Sun, 24 Jul 2011 15:20:13 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk dial status
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F8A@cooper>
Message-ID: <1311546013.75400.YahooMailClassic@web161007.mail.bf1.yahoo.com>

I just want to be able to get the call statues like ANSWER, NO_ANSWER, BUSY etc. but Freeswitch does not return these statuses after the bridge app is finished.

--- On Sun, 7/24/11, Peter Olsson <peter.olsson at visionutveckling.se> wrote:

From: Peter Olsson <peter.olsson at visionutveckling.se>
Subject: Re: [Freeswitch-users] Asterisk dial status
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Sunday, July 24, 2011, 12:01 PM

I've never used Perl for this, but I'm pretty sure that the last line won't return until the bridge is done (because of call finishied or failed).

What exactly do you want to accomplish - I think there is probably an easier way for what you're trying to solve.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Sam [lakersman2006 at yahoo.com]
Skickat: den 24 juli 2011 20:45
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Asterisk dial status

For the life of me I am still unable to get the bridged call status, when I subscribe to an "event" I am not here is my perl code snippet below, what am I doing wrong?


my $fs = new ESL::ESLconnection($host, $port, $password);

$session->answer();

if ($session->ready ())
{
? ? #set bridge settings
? ? $session->execute("set", "ringback=$ringback_tone");
? ? $session->execute("set", "instant_ringback=true");
? ? $session->execute("set", "ignore_early_media=false");
? ? $session->execute("set", "call_timeout=60");
? ? $session->execute("set", "progress_timeout=15");
? ? $session->execute("set", "origination_caller_id_number=$caller_id");
? ? $session->execute("set", "continue_on_fail=false");
? ? $session->execute("bridge", "sofia/gateway/provider1/$destination_number");

? ? $fs->sendRecv("event plain ALL");
? ? $fs->send("linger");

? ? while($fs->connected())
? ? {
? ? ? ? $data = $fs->recvEvent();

? ? ? ? if ( $data =~ m/Answered-State: (\w+)/m )
? ? ? ? {
? ? ? ? ???my $state = $1;

? ? ? ? ???#print "Channel state is $state\n";

? ? ? ? ???if ( $state eq 'answered' ) {

? ? ? ? ? ???freeswitch::consoleLog("INFO", "CALL ANSWERED\n");

? ? ? ? ???}
? ? ? ? ???else
? ? ? ? ???{
? ? ? ? ? ? ? ? freeswitch::consoleLog("INFO", "NO CALLS ANSWERED\n");
? ? ? ? ???}
? ? ? ???}
? ? }
}

$session->hangup();

return 1;

--- On Fri, 7/22/11, Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:

From: Hector Geraldino <Hector.Geraldino at ip-soft.net>
Subject: Re: [Freeswitch-users] Asterisk dial status
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Friday, July 22, 2011, 2:28 PM


It does.



You can check the events by connecting to FS on inbound mode, or using FS outbound socket connection to an application designed by you.



As you already have a perl script to interact with FS, you can modify it to connect to FS (inbound mode) and listen for events. You can subscribe to the events of a particular call, or filter an specific set of events. I haven?t used Asterisk before so I?m not sure how the AGI fits on the FS architecture.



You can get more info at:

http://wiki.freeswitch.org/wiki/Mod_event_socket



And a working demo at:

http://wiki.freeswitch.org/wiki/Perl_freeswitch_client_example







From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sam
Sent: Friday, July 22, 2011 4:23 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Asterisk dial status



Sam,

Does this require event sockets? Or how would I check these events?

--- On Fri, 7/22/11, Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:

From: Hector Geraldino <Hector.Geraldino at ip-soft.net>
Subject: Re: [Freeswitch-users] Asterisk dial status
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Friday, July 22, 2011, 11:57 AM

Hi Sam,



My guess is that you can subscribe to the events of the channel and evaluate the channel events. Look at:



http://wiki.freeswitch.org/wiki/Event_List



specifically the channel events list.



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sam
Sent: Friday, July 22, 2011 2:32 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Asterisk dial status



Hi,

I am trying to port over an Asterisk AGI script that does a call bridge to Freeswitch using perl, I wanted to know what is the equivalent of the asterisk dial status like: ANSWER, NO_ANSWER, CANCEL, BUSY, CONGESTION, CHANUNAVAIL ? I wanted to know if what methods can be used to gets these statues?




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

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

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




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

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

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FreeSWITCH-users at lists.freeswitch.org</mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4e2c68e932761895611653!

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

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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From wstephen80 at gmail.com  Mon Jul 25 02:20:28 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 25 Jul 2011 00:20:28 +0200
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CAJoKKdUosKzjxLwmwjQUoWQHjDKsx=DyBAufSG_HHGSR3DWNAQ@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>
	<CAFiqYukgW1UOgxNUZs8rPvWNmXx5sDVmGS_D+r_xdLMsXg_i4Q@mail.gmail.com>
	<CAJoKKdX8J68pmUogKgp1Ew4ACpkHntiHoZ2Rpx5JNhepsOVVvw@mail.gmail.com>
	<CAFiqYunZUs0d-kQVTxXEvuqVveiri6WFFW6oT3UYujPGoa2H-w@mail.gmail.com>
	<CAJoKKdXhHyQUXKCzaDc3GSu5m9A3Jdv2m9K=FaY7vt=W1OD5Qg@mail.gmail.com>
	<CAFiqYumuca1OMQ13di425JFjs2dhPeLAi2e0FPu-kG2odJSVqQ@mail.gmail.com>
	<CAKbxfG-GSbSm-wq5+YnT0jvtiiP32xJDgc9pqV1TUXj69CwDcg@mail.gmail.com>
	<CAJoKKdUqV_ygenNTd+Z3hnoC2eXeiPuMOM2PvaChEm3kZXgj0A@mail.gmail.com>
	<CAO4LeU3SKmO4oSHMthnKzhV2w489JRMTxdd3N5hFmdDgbrTyNA@mail.gmail.com>
	<CAJoKKdUosKzjxLwmwjQUoWQHjDKsx=DyBAufSG_HHGSR3DWNAQ@mail.gmail.com>
Message-ID: <CAO4LeU0nMW5w0zcboy1H567hG_dbEY32iOkYL6dDPrk9dr5DMA@mail.gmail.com>

I can confirm that after 7 days and 12,000,000 sessions with real traffic
(production server with sip+isdn and rtp handling) the memory is stable so
the problem is surely fixed.

Stephen

On Sun, Jul 24, 2011 at 10:36 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> I just took:
>
> commit 3c74f391f4f4dc03b687f0d6f45dfd19c17555d3
> Author: Mathieu Parent <math.parent at gmail.com>
> Date:   Fri Jul 22 23:05:56 2011 +0200
>
> and running for 2 days as well.... hitting FS with SIPP at quite a high CPS
>
> its been 14178302 sessions since startup and the memory keeps stable.
>
> DATE                          PID    RSS      %MEM
> 2011-07-24 12:30:02 13268 239852  5.9
>
>
>
> $DAT_TIM,$SIZE,$RSS,$SHARED_CLEAN,$SHARED_DIRTY,$PRIVATE_CLEAN,$PRIVATE_DIRTY,$SWAP
> 2011-07-24 12:31:11,46752,46644,0,0,0,46644,0
>
>
> I don't see the problem anymore!
>
>
> Thanks for you effort.
>
>
>
>
> On Tue, Jul 19, 2011 at 5:50 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> I'm using commit:
>>
>> commit 130e1c87746b0596460358d77d6aabbfe41a0072
>> Author: Jeff Lenk <jeff at jefflenk.com>
>> Date:   Sat Jul 16 19:13:27 2011 -0500
>>
>> and after 2 days and more then 3600000 sessions, the memory seems to be
>> stable so the problem is probably fixed.
>> I'll update this info in the next days.
>>
>> Stephen
>>
>> On Thu, Jul 14, 2011 at 5:48 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>>
>>>
>>> On Thu, Jul 14, 2011 at 5:11 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> you can use gigs before you stop growing in some cases esp on sipp test.
>>>> you need a lot more than 5 min to judge it.
>>>> more like dozen or more hours.
>>>>
>>>> yap, im running it for a very long time ... the example was just a snip
>>> ... and i hit the swap right now :=) .... after 2 hours of sipp at 30
>>> CPS....
>>>
>>> start: 2011-07-14 12:45:43
>>> stop: 2011-07-14 14:31:13
>>>
>>>
>>> if i use less load it will grow slowly but it will end into swap... its
>>> just a matter of time.
>>>
>>> So, how can we troubleshoot this ?
>>>
>>> im going to open a jira to move this out of Users Help mailing list...
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 engineerzuhairraza at gmail.com  Mon Jul 25 02:44:22 2011
From: engineerzuhairraza at gmail.com (Zohair Raza)
Date: Mon, 25 Jul 2011 03:44:22 +0500
Subject: [Freeswitch-users] Asterisk dial status
In-Reply-To: <1311546013.75400.YahooMailClassic@web161007.mail.bf1.yahoo.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F8A@cooper>
	<1311546013.75400.YahooMailClassic@web161007.mail.bf1.yahoo.com>
Message-ID: <CAF_PTeugifrZSsakNYbDjEdhgwyU8K+=4wt9nif-pVmpYO25UQ@mail.gmail.com>

Hi Again,

Sorry I forgot to mention xml_cdr.
*http://wiki.freeswitch.org/wiki/Mod_xml_cdr*

I use it with xml curl and take all after call information through it.

On Mon, Jul 25, 2011 at 3:20 AM, Sam <lakersman2006 at yahoo.com> wrote:

> I just want to be able to get the call statues like ANSWER, NO_ANSWER, BUSY
> etc. but Freeswitch does not return these statuses after the bridge app is
> finished.
>
> --- On *Sun, 7/24/11, Peter Olsson <peter.olsson at visionutveckling.se>*wrote:
>
>
> From: Peter Olsson <peter.olsson at visionutveckling.se>
>
> Subject: Re: [Freeswitch-users] Asterisk dial status
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Sunday, July 24, 2011, 12:01 PM
>
> I've never used Perl for this, but I'm pretty sure that the last line won't
> return until the bridge is done (because of call finishied or failed).
>
> What exactly do you want to accomplish - I think there is probably an
> easier way for what you're trying to solve.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org<http://mc/compose?to=freeswitch-users-bounces at lists.freeswitch.org>[
> freeswitch-users-bounces at lists.freeswitch.org<http://mc/compose?to=freeswitch-users-bounces at lists.freeswitch.org>]
> f?r Sam [lakersman2006 at yahoo.com<http://mc/compose?to=lakersman2006 at yahoo.com>
> ]
>
> Skickat: den 24 juli 2011 20:45
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Asterisk dial status
>
> For the life of me I am still unable to get the bridged call status, when I
> subscribe to an "event" I am not here is my perl code snippet below, what am
> I doing wrong?
>
>
> my $fs = new ESL::ESLconnection($host, $port, $password);
>
> $session->answer();
>
> if ($session->ready ())
> {
>     #set bridge settings
>     $session->execute("set", "ringback=$ringback_tone");
>     $session->execute("set", "instant_ringback=true");
>     $session->execute("set", "ignore_early_media=false");
>     $session->execute("set", "call_timeout=60");
>     $session->execute("set", "progress_timeout=15");
>     $session->execute("set", "origination_caller_id_number=$caller_id");
>     $session->execute("set", "continue_on_fail=false");
>     $session->execute("bridge",
> "sofia/gateway/provider1/$destination_number");
>
>     $fs->sendRecv("event plain ALL");
>     $fs->send("linger");
>
>     while($fs->connected())
>     {
>         $data = $fs->recvEvent();
>
>         if ( $data =~ m/Answered-State: (\w+)/m )
>         {
>            my $state = $1;
>
>            #print "Channel state is $state\n";
>
>            if ( $state eq 'answered' ) {
>
>              freeswitch::consoleLog("INFO", "CALL ANSWERED\n");
>
>            }
>            else
>            {
>                 freeswitch::consoleLog("INFO", "NO CALLS ANSWERED\n");
>            }
>          }
>     }
> }
>
> $session->hangup();
>
> return 1;
>
> --- On Fri, 7/22/11, Hector Geraldino <Hector.Geraldino at ip-soft.net<http://mc/compose?to=Hector.Geraldino at ip-soft.net>>
> wrote:
>
> From: Hector Geraldino <Hector.Geraldino at ip-soft.net<http://mc/compose?to=Hector.Geraldino at ip-soft.net>
> >
> Subject: Re: [Freeswitch-users] Asterisk dial status
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users at lists.freeswitch.org>
> >
> Date: Friday, July 22, 2011, 2:28 PM
>
>
> It does.
>
>
>
> You can check the events by connecting to FS on inbound mode, or using FS
> outbound socket connection to an application designed by you.
>
>
>
> As you already have a perl script to interact with FS, you can modify it to
> connect to FS (inbound mode) and listen for events. You can subscribe to the
> events of a particular call, or filter an specific set of events. I haven?t
> used Asterisk before so I?m not sure how the AGI fits on the FS
> architecture.
>
>
>
> You can get more info at:
>
> http://wiki.freeswitch.org/wiki/Mod_event_socket
>
>
>
> And a working demo at:
>
> http://wiki.freeswitch.org/wiki/Perl_freeswitch_client_example
>
>
>
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org<http://mc/compose?to=freeswitch-users-bounces at lists.freeswitch.org>[mailto:
> freeswitch-users-bounces at lists.freeswitch.org<http://mc/compose?to=freeswitch-users-bounces at lists.freeswitch.org>]
> On Behalf Of Sam
> Sent: Friday, July 22, 2011 4:23 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Asterisk dial status
>
>
>
> Sam,
>
> Does this require event sockets? Or how would I check these events?
>
> --- On Fri, 7/22/11, Hector Geraldino <Hector.Geraldino at ip-soft.net<http://mc/compose?to=Hector.Geraldino at ip-soft.net>>
> wrote:
>
> From: Hector Geraldino <Hector.Geraldino at ip-soft.net<http://mc/compose?to=Hector.Geraldino at ip-soft.net>
> >
> Subject: Re: [Freeswitch-users] Asterisk dial status
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users at lists.freeswitch.org>
> >
> Date: Friday, July 22, 2011, 11:57 AM
>
> Hi Sam,
>
>
>
> My guess is that you can subscribe to the events of the channel and
> evaluate the channel events. Look at:
>
>
>
> http://wiki.freeswitch.org/wiki/Event_List
>
>
>
> specifically the channel events list.
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org<http://mc/compose?to=freeswitch-users-bounces at lists.freeswitch.org>[mailto:
> freeswitch-users-bounces at lists.freeswitch.org<http://mc/compose?to=freeswitch-users-bounces at lists.freeswitch.org>]
> On Behalf Of Sam
> Sent: Friday, July 22, 2011 2:32 PM
> To: freeswitch-users at lists.freeswitch.org<http://mc/compose?to=freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Asterisk dial status
>
>
>
> Hi,
>
> I am trying to port over an Asterisk AGI script that does a call bridge to
> Freeswitch using perl, I wanted to know what is the equivalent of the
> asterisk dial status like: ANSWER, NO_ANSWER, CANCEL, BUSY, CONGESTION,
> CHANUNAVAIL ? I wanted to know if what methods can be used to gets these
> statues?
>
>
>
>
> -----Inline Attachment Follows-----
>
> _______________________________________________
> 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://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -----Inline Attachment Follows-----
>
> _______________________________________________
> 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://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> </mc/compose?to=FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4e2c68e932761895611653!
>
> _______________________________________________
> 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://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> 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
>
>


-- 
Regards,
Zohair Raza

www.zuhair.info

*http://ae.linkedin.com/in/zuhairraza** ***
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From gavin.henry at gmail.com  Mon Jul 25 03:22:34 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 25 Jul 2011 00:22:34 +0100
Subject: [Freeswitch-users] Asterisk dial status
In-Reply-To: <1311533112.57294.YahooMailClassic@web161004.mail.bf1.yahoo.com>
References: <6A6B4C284AD15042B429EB9D904544AD021FD89DC2@NY1-EXMB-01.ip-soft.net>
	<1311533112.57294.YahooMailClassic@web161004.mail.bf1.yahoo.com>
Message-ID: <CAA8_NKCAsHCU2fQM-_+u4xpxpRkBqfBGv+se_X8o1v+D5MF1mA@mail.gmail.com>

Where are you getting $session from?

Print out $e->serialize to get all the keys and use getHeader(''),
after looking at sample code in libs/esl/perl. Tons of examples.

Also subscribe to specific events:

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

I'm using bgapi and originate:

http://wiki.freeswitch.org/wiki/Mod_event_socket#bgapi

Thanks.

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


From mitch.johnson7 at gmail.com  Mon Jul 25 08:08:44 2011
From: mitch.johnson7 at gmail.com (Mitch Johnson)
Date: Mon, 25 Jul 2011 00:08:44 -0400
Subject: [Freeswitch-users] One way calling issue over an internal trunk
In-Reply-To: <mailman.13350.1311132599.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.13350.1311132599.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <90AC5B44-C3E1-48EA-AC45-34C03567CE94@gmail.com>

Sorry for the delayed response, got tied up in other things.  A quick recap. I can dial from my call manager to Freeswitch but cannot dial from freeswitch to call manager.

I wasn't sure what you meant by setting the ext-sip-ip and ext-rtp-ip.  So I replaced the auto-nat with local-network-acl.  I'll always be on the internal network so should never need the nat.

I saw in the acl.conf.xml that it automatically populates all my local subnets.  The networks are correct under the domains of the acl.conf.

I'm not sure where else to look.  Any help would be greatly appreciated.

Thanks so much,

Mitch

On Jul 19, 2011, at 11:29 PM, freeswitch-users-request at lists.freeswitch.org wrote:

> From: Brian West <brian at freeswitch.org>
> Subject: Re: [Freeswitch-users] One way calling issue over an internal trunk
> Date: July 19, 2011 8:55:02 PM EDT
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> you need to set the ext-sip-ip and ext-rtp-ip and deal with setting local-network-acl to the proper network/mask because you're answering with RFC1918 media IP's to the cisco? its retarded and will just start sending to those IP's and they'll usually never make it back to you.
> 
> /b
> 
> On Jul 19, 2011, at 7:16 PM, Mitch Johnson wrote:
> 
>> I have a connection between my FreeSWITCH and a Cisco CallManager:
>> 
>> This is from the default.xml
>> 
>> <extension name="Dial to CM8">
>>      <condition field="destination_number" expression="^(800\d)$">
>>        <action application="bridge" data="sofia/internal/$1 at 172.16.200.100"/>
>>      </condition>
>>     </extension>
> 

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From steveayre at gmail.com  Mon Jul 25 11:05:39 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 25 Jul 2011 08:05:39 +0100
Subject: [Freeswitch-users] One way calling issue over an internal trunk
In-Reply-To: <90AC5B44-C3E1-48EA-AC45-34C03567CE94@gmail.com>
References: <mailman.13350.1311132599.24555.freeswitch-users@lists.freeswitch.org>
	<90AC5B44-C3E1-48EA-AC45-34C03567CE94@gmail.com>
Message-ID: <CAFiqYunfYJsB4rsttPRNPqXDk6zC-QQm5vcvDUGWHG2Q7p=JfA@mail.gmail.com>

>
> I'll always be on the internal network so should never need the nat.
>

Try starting FS with the -nonat option. It'll start faster and skip all the
stuff where it tries to find your external IP and open routes through the
NAT firewall.

On your sip profiles set sip-ip and rtp-ip to your internal IP.

Once you've done that you should have 2-way audio to anywhere on your LAN.

-Steve




On 25 July 2011 05:08, Mitch Johnson <mitch.johnson7 at gmail.com> wrote:

> Sorry for the delayed response, got tied up in other things.  A quick
> recap. I can dial from my call manager to Freeswitch but cannot dial from
> freeswitch to call manager.
>
> I wasn't sure what you meant by setting the ext-sip-ip and ext-rtp-ip.  So
> I replaced the auto-nat with local-network-acl.  I'll always be on the
> internal network so should never need the nat.
>
> I saw in the acl.conf.xml that it automatically populates all my local
> subnets.  The networks are correct under the domains of the acl.conf.
>
> I'm not sure where else to look.  Any help would be greatly appreciated.
>
> Thanks so much,
>
> Mitch
>
> On Jul 19, 2011, at 11:29 PM,
> freeswitch-users-request at lists.freeswitch.org wrote:
>
> *From: *Brian West <brian at freeswitch.org>
> *Subject: **Re: [Freeswitch-users] One way calling issue over an internal
> trunk*
> *Date: *July 19, 2011 8:55:02 PM EDT
> *To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Reply-To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>
> you need to set the ext-sip-ip and ext-rtp-ip and deal with setting
> local-network-acl to the proper network/mask because you're answering with
> RFC1918 media IP's to the cisco? its retarded and will just start sending to
> those IP's and they'll usually never make it back to you.
>
> /b
>
> On Jul 19, 2011, at 7:16 PM, Mitch Johnson wrote:
>
> I have a connection between my FreeSWITCH and a Cisco CallManager:
>
> This is from the default.xml
>
> <extension name="Dial to CM8">
>      <condition field="destination_number" expression="^(800\d)$">
>        <action application="bridge" data="sofia/internal/$1 at 172.16.200.100
> "/>
>      </condition>
>     </extension>
>
>
>
>
> _______________________________________________
> 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 freeswitch-list at puzzled.xs4all.nl  Mon Jul 25 13:03:29 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Mon, 25 Jul 2011 11:03:29 +0200
Subject: [Freeswitch-users] memory leak
In-Reply-To: <CALXCt0oaUWKB9G6D5adVOmWVFyqYZRDzzN0QfAKvWgOm1xW8oQ@mail.gmail.com>
References: <CAJoKKdWZ6FUWs8eus58Y2W4x0J5sgyCxGG_jSNwA=zWy+bfMCQ@mail.gmail.com>	<CAFiqYukgW1UOgxNUZs8rPvWNmXx5sDVmGS_D+r_xdLMsXg_i4Q@mail.gmail.com>	<CAJoKKdX8J68pmUogKgp1Ew4ACpkHntiHoZ2Rpx5JNhepsOVVvw@mail.gmail.com>	<CAFiqYunZUs0d-kQVTxXEvuqVveiri6WFFW6oT3UYujPGoa2H-w@mail.gmail.com>	<CAJoKKdXhHyQUXKCzaDc3GSu5m9A3Jdv2m9K=FaY7vt=W1OD5Qg@mail.gmail.com>	<CAFiqYumuca1OMQ13di425JFjs2dhPeLAi2e0FPu-kG2odJSVqQ@mail.gmail.com>	<CAKbxfG-GSbSm-wq5+YnT0jvtiiP32xJDgc9pqV1TUXj69CwDcg@mail.gmail.com>	<CAJoKKdUqV_ygenNTd+Z3hnoC2eXeiPuMOM2PvaChEm3kZXgj0A@mail.gmail.com>	<CAO4LeU3SKmO4oSHMthnKzhV2w489JRMTxdd3N5hFmdDgbrTyNA@mail.gmail.com>	<CAJoKKdUosKzjxLwmwjQUoWQHjDKsx=DyBAufSG_HHGSR3DWNAQ@mail.gmail.com>	<CAP_G397PRcGq8d7yTOZcJw_EN=mHRw6gD60hqaLseeazK=y60A@mail.gmail.com>	<CAJoKKdVam_C_Xww-hmwRRaxoaBDPfJ-ETJMZjVZYAxfv+9nuNA@mail.gmail.com>	<4E2C7D8D.7040305@integrafin.co.uk>
	<CALXCt0oaUWKB9G6D5adVOmWVFyqYZRDzzN0QfAKvWgOm1xW8oQ@mail.gmail.com>
Message-ID: <4E2D3161.7010301@puzzled.xs4all.nl>

On 07/24/2011 10:48 PM, Giovanni Maruzzelli wrote:
> We're trying to be nice toward other similar projects, both closed and
> open source.
>
> At Cluecon we'll announce the reduction to 0.5 cpm (calls per minute),
> and optional random deadlocks.

Lol, nice one Giovanni :)

Regards,
Patrick


From ijurado at econcept.es  Mon Jul 25 13:17:47 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Mon, 25 Jul 2011 11:17:47 +0200
Subject: [Freeswitch-users] Altering arbitrary channels
Message-ID: <CABj_V-m=MGKeSSYZ+tNU1mvyoKDrf-18s7WEO2XK_tEmafjQnA@mail.gmail.com>

Hi all,

>From the diaplan, is it possible to set a variable in any other channel?
That is, by specifying the channel UUID.

Thanks.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From steveayre at gmail.com  Mon Jul 25 14:42:15 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 25 Jul 2011 11:42:15 +0100
Subject: [Freeswitch-users] Altering arbitrary channels
In-Reply-To: <CABj_V-m=MGKeSSYZ+tNU1mvyoKDrf-18s7WEO2XK_tEmafjQnA@mail.gmail.com>
References: <CABj_V-m=MGKeSSYZ+tNU1mvyoKDrf-18s7WEO2XK_tEmafjQnA@mail.gmail.com>
Message-ID: <CAFiqYumgaRcFofreA+4HB9qc+EaoN4RumNS-j2vGpKJs+7WpBg@mail.gmail.com>

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

On 25 July 2011 10:17, Isaac Jurado <ijurado at econcept.es> wrote:

> Hi all,
>
> >From the diaplan, is it possible to set a variable in any other channel?
> That is, by specifying the channel UUID.
>
> Thanks.
>
> --
> Isaac Jurado
> Internet Busines Solutions eConcept
>
> _______________________________________________
> 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 ijurado at econcept.es  Mon Jul 25 15:00:50 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Mon, 25 Jul 2011 13:00:50 +0200
Subject: [Freeswitch-users] Altering arbitrary channels
In-Reply-To: <CAFiqYumgaRcFofreA+4HB9qc+EaoN4RumNS-j2vGpKJs+7WpBg@mail.gmail.com>
References: <CABj_V-m=MGKeSSYZ+tNU1mvyoKDrf-18s7WEO2XK_tEmafjQnA@mail.gmail.com>
	<CAFiqYumgaRcFofreA+4HB9qc+EaoN4RumNS-j2vGpKJs+7WpBg@mail.gmail.com>
Message-ID: <CABj_V-kZbzmTBbvRa-bDK_Sib_H_yJJZz1x-dzyn+J1vgbg2Eg@mail.gmail.com>

On Mon, Jul 25, 2011 at 12:42 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_setvar

For some reason I was failing to find that wiki page.  I assumed that
the dialplan actions were limited to what is offered by mod_dptools.

Thank you very much :-)

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From avi at avimarcus.net  Mon Jul 25 15:10:33 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 25 Jul 2011 14:10:33 +0300
Subject: [Freeswitch-users] Altering arbitrary channels
In-Reply-To: <CABj_V-kZbzmTBbvRa-bDK_Sib_H_yJJZz1x-dzyn+J1vgbg2Eg@mail.gmail.com>
References: <CABj_V-m=MGKeSSYZ+tNU1mvyoKDrf-18s7WEO2XK_tEmafjQnA@mail.gmail.com>
	<CAFiqYumgaRcFofreA+4HB9qc+EaoN4RumNS-j2vGpKJs+7WpBg@mail.gmail.com>
	<CABj_V-kZbzmTBbvRa-bDK_Sib_H_yJJZz1x-dzyn+J1vgbg2Eg@mail.gmail.com>
Message-ID: <CALNZuwm0gum58e6VNXRtx-J4+dWH6w=hO6S4x_zbdAd3En4_uA@mail.gmail.com>

I think it's the opposite - the ESL tools are limited by mod_commands, but
in the dialplan you can do either. Dialplan tools act on the current channel
without having to specify a UUID, the mod_commands you need to specify a
uuid to use them.
-Avi



On Mon, Jul 25, 2011 at 2:00 PM, Isaac Jurado <ijurado at econcept.es> wrote:

> On Mon, Jul 25, 2011 at 12:42 PM, Steven Ayre <steveayre at gmail.com> wrote:
> >
> > http://wiki.freeswitch.org/wiki/Mod_commands#uuid_setvar
>
> For some reason I was failing to find that wiki page.  I assumed that
> the dialplan actions were limited to what is offered by mod_dptools.
>
> Thank you very much :-)
>
> --
> Isaac Jurado
> Internet Busines Solutions eConcept
>
> _______________________________________________
> 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 peter.olsson at visionutveckling.se  Mon Jul 25 16:06:24 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 25 Jul 2011 14:06:24 +0200
Subject: [Freeswitch-users] Altering arbitrary channels
In-Reply-To: <CALNZuwm0gum58e6VNXRtx-J4+dWH6w=hO6S4x_zbdAd3En4_uA@mail.gmail.com>
References: <CABj_V-m=MGKeSSYZ+tNU1mvyoKDrf-18s7WEO2XK_tEmafjQnA@mail.gmail.com>
	<CAFiqYumgaRcFofreA+4HB9qc+EaoN4RumNS-j2vGpKJs+7WpBg@mail.gmail.com>
	<CABj_V-kZbzmTBbvRa-bDK_Sib_H_yJJZz1x-dzyn+J1vgbg2Eg@mail.gmail.com>,
	<CALNZuwm0gum58e6VNXRtx-J4+dWH6w=hO6S4x_zbdAd3En4_uA@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F8C@cooper>

Actually, using ESL you can use mod_dptools as well. Using SendMsg to queue the execution of an app into a specific channel.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Avi Marcus [avi at avimarcus.net]
Skickat: den 25 juli 2011 13:10
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Altering arbitrary channels

I think it's the opposite - the ESL tools are limited by mod_commands, but in the dialplan you can do either. Dialplan tools act on the current channel without having to specify a UUID, the mod_commands you need to specify a uuid to use them.
-Avi



On Mon, Jul 25, 2011 at 2:00 PM, Isaac Jurado <ijurado at econcept.es<mailto:ijurado at econcept.es>> wrote:
On Mon, Jul 25, 2011 at 12:42 PM, Steven Ayre <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
>
> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_setvar

For some reason I was failing to find that wiki page.  I assumed that
the dialplan actions were limited to what is offered by mod_dptools.

Thank you very much :-)

--
Isaac Jurado
Internet Busines Solutions eConcept

_______________________________________________
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<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4e2d504232761845388632!


From steveayre at gmail.com  Mon Jul 25 16:30:18 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 25 Jul 2011 13:30:18 +0100
Subject: [Freeswitch-users] Altering arbitrary channels
In-Reply-To: <CABj_V-kZbzmTBbvRa-bDK_Sib_H_yJJZz1x-dzyn+J1vgbg2Eg@mail.gmail.com>
References: <CABj_V-m=MGKeSSYZ+tNU1mvyoKDrf-18s7WEO2XK_tEmafjQnA@mail.gmail.com>
	<CAFiqYumgaRcFofreA+4HB9qc+EaoN4RumNS-j2vGpKJs+7WpBg@mail.gmail.com>
	<CABj_V-kZbzmTBbvRa-bDK_Sib_H_yJJZz1x-dzyn+J1vgbg2Eg@mail.gmail.com>
Message-ID: <CAFiqYu=Z3yp2-kkze6jmyCxAq83-etzjXeFByGd3WZk0CujVaQ@mail.gmail.com>

There are 2 types of commands... dialplan apps (which execute on the current
channel) and API commands (which may or may not involve a channel, when they
do they always need its UUID).

APIs are generally executed from the console but can be used in the
dialplan. It's a slightly different syntax though:
<action application="set" data="result=${api arg1 arg2 arg3}"/>

It doesn't have to be used with set, it can be used anywhere:
<condition field="${api arg1 arg2 arg3}" expression="^true$">
    <action application="bridge" data="${api arg1 arg2 arg3}"/>
</condition>

Any module can register either type of command...
  mod_dptools contains the generic dialplan apps
  mod_commands contains the generic API commands
  module-specific stuff will be registered by that module, e.g. the sofia
API is registered by mod_sofia

-Steve


On 25 July 2011 12:00, Isaac Jurado <ijurado at econcept.es> wrote:

> On Mon, Jul 25, 2011 at 12:42 PM, Steven Ayre <steveayre at gmail.com> wrote:
> >
> > http://wiki.freeswitch.org/wiki/Mod_commands#uuid_setvar
>
> For some reason I was failing to find that wiki page.  I assumed that
> the dialplan actions were limited to what is offered by mod_dptools.
>
> Thank you very much :-)
>
> --
> Isaac Jurado
> Internet Busines Solutions eConcept
>
> _______________________________________________
> 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 darcy at thevoiphighway.com  Fri Jul 22 08:43:41 2011
From: darcy at thevoiphighway.com (VoIPHwy)
Date: Fri, 22 Jul 2011 00:43:41 -0400
Subject: [Freeswitch-users] bridge_answer_timeout problem
Message-ID: <E72CAEBEF02948C58BD41885721B88BA@DWP>

Hello, I am having a problem with bridge_answer_timeout

I included the dial plan below.   This does timeout and it does transfer but there is no audio when it goes to either voice mail or an ivr.  
If it goes to another extension or pstn number, there is audio.    The trace shows messages being played.   
I did a wireshark trace and it did not show any rtp going out after the bridge_answer_timeout.
In the dial plan, 8205 goes to a dial plan that is the voice mail for extension 205 and if I dial 8205 direct it works.    I have installed the latest git.

<include>
    <extension name="205">
         <condition field="destination_number" expression="^(205)$">
            <action application="set" data="proxy_media=true"/>
            <action application="set" data="continue_on_fail=true"/>
            <action application="set" data="hangup_after_bridge=false"/>
            <action application="bridge" data="{bridge_answer_timeout=5}sofia/${use_profile}/16133429652 at toronto2.voip.ms"/>
            <action application="transfer" data="8205 XML office.voiphwy.com"/>    
        </condition>
    </extension>
</include>

Any help would be appreciated.

Darcy Primrose
The VoIP Highway
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From slash at nat-tele.com  Fri Jul 22 18:24:19 2011
From: slash at nat-tele.com (slash at nat-tele.com)
Date: Fri, 22 Jul 2011 09:24:19 -0500
Subject: [Freeswitch-users]  rxfax on b-leg
Message-ID: <22a18816be6d633b717a87152f104de5@nat-tele.com>



 Hello guys!

 I wonder if there is a way to make rxfax recieve fax on outgoing call? 
 The idea is that registered FS user dials out, than tone_detect is used 
 to detect fax tone and start rxfax. The difference is that rxfax should 
 be run on B-leg, instead of A-leg in incoming fax call scenario.

 I've tried to make a test config with outgoing call, rxfax & 
 tone_detect, but when I dial out and remote user starts sending fax to 
 me, FS detects a tone correctly, starts rxfax, but on my leg (A-leg, i 
 hear fax tones) and B-leg (with real fax machine) is hanged up.

 Are there are solutions to explain somehow to rxfax that it should be 
 run on B-leg, not on A-leg?

 Thanks in advance!

 With best regards,
 Kirill.


From rmartinez at redvoiss.net  Fri Jul 22 19:56:01 2011
From: rmartinez at redvoiss.net (Ricardo Martinez)
Date: Fri, 22 Jul 2011 11:56:01 -0400
Subject: [Freeswitch-users] Problem with Freeswitch/Sangoma D100.
Message-ID: <aa54cbcc6870ad57562eae38969f10a9@mail.gmail.com>

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.-
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From darcy at primrose.ws  Fri Jul 22 22:12:04 2011
From: darcy at primrose.ws (Darcy)
Date: Fri, 22 Jul 2011 14:12:04 -0400
Subject: [Freeswitch-users] bridge_answer_timeout with no audio
Message-ID: <2206E71B464245AA9D01FD826E61F942@DWP>

Hello, I am having a problem with bridge_answer_timeout, also having email problems, hope this isn?t duplicated

I included the dial plan below.   This does timeout and it does transfer but there is no audio when it goes to either voice mail or an ivr.  
If it goes to another extension or pstn number, there is audio.    The trace shows messages being played.   
I did a wireshark trace and it did not show any rtp going out after the bridge_answer_timeout.
In the dial plan, 8205 goes to a dial plan that is the voice mail for extension 205 and if I dial 8205 direct it works.    I have installed the latest git.

<include>
    <extension name="205">
         <condition field="destination_number" expression="^(205)$">
            <action application="set" data="proxy_media=true"/>
            <action application="set" data="continue_on_fail=true"/>
            <action application="set" data="hangup_after_bridge=false"/>
            <action application="bridge" data="{bridge_answer_timeout=5}sofia/${use_profile}/16133429652 at toronto2.voip.ms"/>
            <action application="transfer" data="8205 XML office.voiphwy.com"/>    
        </condition>
    </extension>
</include>

I get the following from the timeout:

2011-07-22 12:34:13.013195 [DEBUG] switch_ivr_bridge.c:409 Answer timeout hit on sofia/internal/16133429652 at toronto2.voip.ms.
2011-07-22 12:34:13.013195 [DEBUG] switch_channel.c:2739 (sofia/internal/16133429652 at toronto2.voip.ms) Callstate Change EARLY -> HANGUP
2011-07-22 12:34:13.013195 [NOTICE] switch_ivr_bridge.c:410 Hangup sofia/internal/16133429652 at toronto2.voip.ms [CS_EXCHANGE_MEDIA] [ALLOTTED_TIMEOUT]

I see this which indicates the message is playing

2011-07-22 12:34:13.025050 [NOTICE] mod_dptools.c:930 Channel [sofia/internal/200 at office.voiphwy.com] has been answered
EXECUTE sofia/internal/200 at office.voiphwy.com sleep(1000)
2011-07-22 12:34:13.036856 [DEBUG] switch_core_session.c:855 Send signal sofia/internal/200 at office.voiphwy.com [BREAK]
2011-07-22 12:34:13.036856 [DEBUG] sofia.c:5086 Channel sofia/internal/200 at office.voiphwy.com entering state [completed][200]
2011-07-22 12:34:13.106193 [DEBUG] switch_core_session.c:855 Send signal sofia/internal/200 at office.voiphwy.com [BREAK]
2011-07-22 12:34:13.106193 [DEBUG] switch_core_session.c:855 Send signal sofia/internal/200 at office.voiphwy.com [BREAK]
2011-07-22 12:34:13.106193 [DEBUG] switch_core_session.c:855 Send signal sofia/internal/200 at office.voiphwy.com [BREAK]
2011-07-22 12:34:13.106193 [DEBUG] sofia.c:5086 Channel sofia/internal/200 at office.voiphwy.com entering state [ready][200]
EXECUTE sofia/internal/200 at office.voiphwy.com voicemail(default office.voiphwy.com 205)
2011-07-22 12:34:14.166175 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en]
2011-07-22 12:34:14.180219 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-person.wav] (en:en)
2011-07-22 12:34:14.180219 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
/lo2011-07-22 12:34:15.530174 [DEBUG] switch_ivr_play_say.c:1649 done playing file
g2011-07-22 12:34:15.650171 [DEBUG] switch_ivr_play_say.c:244 Handle say:[205] (en:en)
2011-07-22 12:34:15.650171 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms

Any help would be appreciated.

Darcy Primrose
The VoIP Highway
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From darcy at thevoiphighway.com  Fri Jul 22 20:44:19 2011
From: darcy at thevoiphighway.com (VoIPHwy)
Date: Fri, 22 Jul 2011 12:44:19 -0400
Subject: [Freeswitch-users] bridge_answer_timeout problem
Message-ID: <19CCCDC85C3146F8B670C0BDADC8E6C0@DWP>

Hello, I am having a problem with bridge_answer_timeout

I included the dial plan below.   This does timeout and it does transfer but there is no audio when it goes to either voice mail or an ivr.  
If it goes to another extension or pstn number, there is audio.    The trace shows messages being played.   
I did a wireshark trace and it did not show any rtp going out after the bridge_answer_timeout.
In the dial plan, 8205 goes to a dial plan that is the voice mail for extension 205 and if I dial 8205 direct it works.    I have installed the latest git.

<include>
    <extension name="205">
         <condition field="destination_number" expression="^(205)$">
            <action application="set" data="proxy_media=true"/>
            <action application="set" data="continue_on_fail=true"/>
            <action application="set" data="hangup_after_bridge=false"/>
            <action application="bridge" data="{bridge_answer_timeout=5}sofia/${use_profile}/16133429652 at toronto2.voip.ms"/>
            <action application="transfer" data="8205 XML office.voiphwy.com"/>    
        </condition>
    </extension>
</include>

I get the following from the timeout:

2011-07-22 12:34:13.013195 [DEBUG] switch_ivr_bridge.c:409 Answer timeout hit on sofia/internal/16133429652 at toronto2.voip.ms.
2011-07-22 12:34:13.013195 [DEBUG] switch_channel.c:2739 (sofia/internal/16133429652 at toronto2.voip.ms) Callstate Change EARLY -> HANGUP
2011-07-22 12:34:13.013195 [NOTICE] switch_ivr_bridge.c:410 Hangup sofia/internal/16133429652 at toronto2.voip.ms [CS_EXCHANGE_MEDIA] [ALLOTTED_TIMEOUT]

I see this which indicates the message is playing

2011-07-22 12:34:13.025050 [NOTICE] mod_dptools.c:930 Channel [sofia/internal/200 at office.voiphwy.com] has been answered
EXECUTE sofia/internal/200 at office.voiphwy.com sleep(1000)
2011-07-22 12:34:13.036856 [DEBUG] switch_core_session.c:855 Send signal sofia/internal/200 at office.voiphwy.com [BREAK]
2011-07-22 12:34:13.036856 [DEBUG] sofia.c:5086 Channel sofia/internal/200 at office.voiphwy.com entering state [completed][200]
2011-07-22 12:34:13.106193 [DEBUG] switch_core_session.c:855 Send signal sofia/internal/200 at office.voiphwy.com [BREAK]
2011-07-22 12:34:13.106193 [DEBUG] switch_core_session.c:855 Send signal sofia/internal/200 at office.voiphwy.com [BREAK]
2011-07-22 12:34:13.106193 [DEBUG] switch_core_session.c:855 Send signal sofia/internal/200 at office.voiphwy.com [BREAK]
2011-07-22 12:34:13.106193 [DEBUG] sofia.c:5086 Channel sofia/internal/200 at office.voiphwy.com entering state [ready][200]
EXECUTE sofia/internal/200 at office.voiphwy.com voicemail(default office.voiphwy.com 205)
2011-07-22 12:34:14.166175 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en]
2011-07-22 12:34:14.180219 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-person.wav] (en:en)
2011-07-22 12:34:14.180219 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
/lo2011-07-22 12:34:15.530174 [DEBUG] switch_ivr_play_say.c:1649 done playing file
g2011-07-22 12:34:15.650171 [DEBUG] switch_ivr_play_say.c:244 Handle say:[205] (en:en)
2011-07-22 12:34:15.650171 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms

Any help would be appreciated.

Darcy Primrose
The VoIP Highway
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From db3l.net at gmail.com  Sat Jul 23 03:29:59 2011
From: db3l.net at gmail.com (David Bolen)
Date: Fri, 22 Jul 2011 19:29:59 -0400
Subject: [Freeswitch-users] SIP re-invite issue with session timers
Message-ID: <m2tyaduc5k.fsf@valheru.db3l.homeip.net>

While testing some outbound T.38 fax calls, they were getting quickly
disconnected, and I found that I was running into problems related to
the session timer during the re-invite process.  I was wondering if
anyone else has run into anything similar?  I'm not sure if it's just
T.38 related, or would have occurred on any re-invite scenario.

In the trace below, I think my gateway (Flowroute) reacts a little
strangely since it appears to refuse a session timer that it itself
chose during the initial invite process (instead seeming to want to
flip who handles the refresh), but while odd, that's not necessarily
invalid, is it?  I also noticed that FreeSWITCH promoted the timer to
a requirement during the re-invite.  But the eventual call disconnect
seems due to FreeSWITCH never completing renegotiation - the last
thing it transmits is a 100 response.

Disabling session timers (enable-timer=false) in the profile gets
around the problem, but it seems like keeping the timers, if possible,
is preferable in terms of an additional safety net.

Any thoughts or suggestions would be welcome.  Does this appear
strictly a gateway side issue, or is something also not working
properly on the FreeSWITCH side?

-- David


Note: git-b1d1beb is the same as f1de377 from 7/16 plus one local
      change to mod_conference.  The dialed number has been replaced
      by 11111111111, my public switch address with 111.111.111.111,
      and I've excluded the Proxy-Authorization header throughout.

The initial call setup seems ok:

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

send 1390 bytes to udp/[216.115.69.144]:5060 at 21:29:19.155730:
   ------------------------------------------------------------------------
   INVITE sip:11111111111 at sip.flowroute.com SIP/2.0
   Via: SIP/2.0/UDP 111.111.111.111:5080;rport;branch=z9hG4bKK8mr3pcre4a2j
   Max-Forwards: 70
   From: "" <sip:XXXX at sip.flowroute.com;transport=udp>;tag=KcKN11gj6arQS
   To: <sip:11111111111 at sip.flowroute.com>
   Call-ID: 7fe743c9-2f4c-122f-9790-f23c91931894
   CSeq: 15349016 INVITE
   Contact: <sip:gw+flowroute at 111.111.111.111:5080;transport=udp;gw=flowroute>
   Expires: 600
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b1d1beb 2011-07-18 23-08-16 -0400
   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: 213
   X-FS-Support: update_display
   Remote-Party-ID: <sip:0000000000 at sip.flowroute.com>;party=calling;screen=yes;privacy=off
   
   v=0
   o=FreeSWITCH 1311342219 1311342220 IN IP4 111.111.111.111
   s=FreeSWITCH
   c=IN IP4 111.111.111.111
   t=0 0
   m=audio 27940 RTP/AVP 9 0 8 3 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
recv 305 bytes from udp/[216.115.69.144]:5060 at 21:29:19.194573:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 111.111.111.111:5080;rport=5080;branch=z9hG4bKK8mr3pcre4a2j
   From: "" <sip:XXXX at sip.flowroute.com;transport=udp>;tag=KcKN11gj6arQS
   To: <sip:11111111111 at sip.flowroute.com>
   Call-ID: 7fe743c9-2f4c-122f-9790-f23c91931894
   CSeq: 15349016 INVITE
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 687 bytes from udp/[216.115.69.144]:5060 at 21:29:19.839258:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   From: "" <sip:XXXX at sip.flowroute.com;transport=udp>;tag=KcKN11gj6arQS
   To: <sip:11111111111 at sip.flowroute.com>;tag=SDvtmh399-gK0a9c4821
   Via: SIP/2.0/UDP 111.111.111.111:5080;rport=5080;branch=z9hG4bKK8mr3pcre4a2j
   Call-ID: 7fe743c9-2f4c-122f-9790-f23c91931894
   CSeq: 15349016 INVITE
   Record-Route: <sip:216.115.69.133;lr>
   Record-Route: <sip:216.115.69.144;lr>
   Contact: <sip:+11111111111 at 67.16.125.60:5060;transport=udp>
   Content-Length: 184
   Content-Type: application/sdp
   
   v=0
   o=- 3829 19246 IN IP4 67.16.125.60
   s=-
   c=IN IP4 67.16.125.60
   t=0 0
   m=audio 50330 RTP/AVP 0 101
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:20
   a=maxptime:20
   ------------------------------------------------------------------------
recv 850 bytes from udp/[216.115.69.144]:5060 at 21:29:21.587130:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   From: "" <sip:XXXX at sip.flowroute.com;transport=udp>;tag=KcKN11gj6arQS
   To: <sip:11111111111 at sip.flowroute.com>;tag=SDvtmh399-gK0a9c4821
   Via: SIP/2.0/UDP 111.111.111.111:5080;rport=5080;branch=z9hG4bKK8mr3pcre4a2j
   Call-ID: 7fe743c9-2f4c-122f-9790-f23c91931894
   CSeq: 15349016 INVITE
   Record-Route: <sip:216.115.69.133;lr>
   Record-Route: <sip:216.115.69.144;lr>
   Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay,  multipart/mixed
   Contact: <sip:+11111111111 at 67.16.125.60:5060;transport=udp>
   Supported: timer,replaces
   Session-Expires: 10800;refresher=uas
   Content-Length: 184
   Content-Type: application/sdp
   
   v=0
   o=- 3829 19246 IN IP4 67.16.125.60
   s=-
   c=IN IP4 67.16.125.60
   t=0 0
   m=audio 50330 RTP/AVP 0 101
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:20
   a=maxptime:20
   ------------------------------------------------------------------------
send 826 bytes to udp/[216.115.69.144]:5060 at 21:29:21.587678:
   ------------------------------------------------------------------------
   ACK sip:+11111111111 at 67.16.125.60:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 111.111.111.111:5080;rport;branch=z9hG4bKmHeH5HXUBD1me
   Route: <sip:216.115.69.144;lr>
   Route: <sip:216.115.69.133;lr>
   Max-Forwards: 70
   From: "" <sip:XXXX at sip.flowroute.com;transport=udp>;tag=KcKN11gj6arQS
   To: <sip:11111111111 at sip.flowroute.com>;tag=SDvtmh399-gK0a9c4821
   Call-ID: 7fe743c9-2f4c-122f-9790-f23c91931894
   CSeq: 15349016 ACK
   Contact: <sip:gw+flowroute at 111.111.111.111:5080;transport=udp;gw=flowroute>
   Content-Length: 0
   
   ------------------------------------------------------------------------

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

The gateway appears to have specified a uas-refreshed session timer in
its 200 message.  Below, the re-invite happens for T.38.  Here I noticed
the new Require: timer option (with the session-expires from before),
which is rejected, then oddly the gateway re-invites but apparently
with a uac-refreshed timer.  FreeSWITCH then sends a 100, and nothing else.


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

send 1638 bytes to udp/[216.115.69.144]:5060 at 21:29:22.108341:
   ------------------------------------------------------------------------
   INVITE sip:+11111111111 at 67.16.125.60:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 111.111.111.111:5080;rport;branch=z9hG4bKNt796ceZ8NQ7S
   Route: <sip:216.115.69.144;lr>
   Route: <sip:216.115.69.133;lr>
   Max-Forwards: 70
   From: "" <sip:XXXX at sip.flowroute.com;transport=udp>;tag=KcKN11gj6arQS
   To: <sip:11111111111 at sip.flowroute.com>;tag=SDvtmh399-gK0a9c4821
   Call-ID: 7fe743c9-2f4c-122f-9790-f23c91931894
   CSeq: 15349017 INVITE
   Contact: <sip:gw+flowroute at 111.111.111.111:5080;transport=udp;gw=flowroute>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b1d1beb 2011-07-18 23-08-16 -0400
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
   Require: timer
   Supported: timer, precondition, path, replaces
   Session-Expires: 10800;refresher=uas
   Min-SE: 120
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 322
   X-FS-Support: update_display
   Remote-Party-ID: <sip:0000000000 at sip.flowroute.com>;party=calling;screen=yes;privacy=off
   
   v=0
   o=FreeSWITCH 1311342219 1311342221 IN IP4 111.111.111.111
   s=FreeSWITCH
   c=IN IP4 111.111.111.111
   t=0 0
   m=image 27940 udptl t38
   a=T38FaxVersion:0
   a=T38MaxBitRate:14400
   a=T38FaxFillBitRemoval
   a=T38FaxRateManagement:transferredTCF
   a=T38FaxMaxBuffer:2000
   a=T38FaxMaxDatagram:400
   a=T38FaxUdpEC:t38UDPRedundancy
   ------------------------------------------------------------------------
recv 330 bytes from udp/[216.115.69.144]:5060 at 21:29:22.147343:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 111.111.111.111:5080;rport=5080;branch=z9hG4bKNt796ceZ8NQ7S
   From: "" <sip:XXXX at sip.flowroute.com;transport=udp>;tag=KcKN11gj6arQS
   To: <sip:11111111111 at sip.flowroute.com>;tag=SDvtmh399-gK0a9c4821
   Call-ID: 7fe743c9-2f4c-122f-9790-f23c91931894
   CSeq: 15349017 INVITE
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 338 bytes from udp/[216.115.69.144]:5060 at 21:29:22.170802:
   ------------------------------------------------------------------------
   SIP/2.0 420 Bad Extension
   From: "" <sip:XXXX at sip.flowroute.com;transport=udp>;tag=KcKN11gj6arQS
   To: <sip:11111111111 at sip.flowroute.com>;tag=SDvtmh399-gK0a9c4821
   Via: SIP/2.0/UDP 111.111.111.111:5080;rport=5080;branch=z9hG4bKNt796ceZ8NQ7S
   Call-ID: 7fe743c9-2f4c-122f-9790-f23c91931894
   CSeq: 15349017 INVITE
   Unsupported: timer
   
   ------------------------------------------------------------------------
send 446 bytes to udp/[216.115.69.144]:5060 at 21:29:22.171034:
   ------------------------------------------------------------------------
   ACK sip:+11111111111 at 67.16.125.60:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 111.111.111.111:5080;rport;branch=z9hG4bKNt796ceZ8NQ7S
   Route: <sip:216.115.69.144;lr>
   Route: <sip:216.115.69.133;lr>
   Max-Forwards: 70
   From: "" <sip:XXXX at sip.flowroute.com;transport=udp>;tag=KcKN11gj6arQS
   To: <sip:11111111111 at sip.flowroute.com>;tag=SDvtmh399-gK0a9c4821
   Call-ID: 7fe743c9-2f4c-122f-9790-f23c91931894
   CSeq: 15349017 ACK
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 1160 bytes from udp/[216.115.69.144]:5060 at 21:29:26.533911:
   ------------------------------------------------------------------------
   INVITE sip:gw+flowroute at 111.111.111.111:5080;transport=udp;gw=flowroute SIP/2.0
   Record-Route: <sip:216.115.69.144;lr>
   Record-Route: <sip:216.115.69.133;lr>
   From: <sip:11111111111 at sip.flowroute.com>;tag=SDvtmh399-gK0a9c4821;tag=SDvtmh399-gK0a9c4821
   To: "" <sip:XXXX at sip.flowroute.com;transport=udp>;tag=KcKN11gj6arQS
   Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK1dbe.2add7494aaa097ffa4b75892aba82f61.0
   Via: SIP/2.0/UDP 216.115.69.133;branch=z9hG4bK1dbe.bbcd1cf6f99ecd7a297b5dc33dd676dd.0
   Via: SIP/2.0/UDP 67.16.125.60:5060;branch=z9hG4bKmthalp308g505g83r340sb0000hc3.1
   Call-ID: 7fe743c9-2f4c-122f-9790-f23c91931894
   CSeq: 4313 INVITE
   Max-Forwards: 67
   Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay,  multipart/mixed
   Contact: <sip:+11111111111 at 67.16.125.60:5060;transport=udp>
   Supported: timer,replaces
   Session-Expires: 10800;refresher=uac
   Min-SE: 90
   Content-Length: 172
   Content-Type: application/sdp
   
   v=0
   o=- 3829 19247 IN IP4 67.16.125.60
   s=-
   c=IN IP4 67.16.125.60
   t=0 0
   m=image 50330 udptl t38
   a=T38FaxRateManagement:transferredTCF
   a=T38FaxUdpEC:t38UDPRedundancy
   ------------------------------------------------------------------------
send 688 bytes to udp/[216.115.69.144]:5060 at 21:29:26.534587:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK1dbe.2add7494aaa097ffa4b75892aba82f61.0
   Via: SIP/2.0/UDP 216.115.69.133;branch=z9hG4bK1dbe.bbcd1cf6f99ecd7a297b5dc33dd676dd.0
   Via: SIP/2.0/UDP 67.16.125.60:5060;branch=z9hG4bKmthalp308g505g83r340sb0000hc3.1
   Record-Route: <sip:216.115.69.144;lr>
   Record-Route: <sip:216.115.69.133;lr>
   From: <sip:11111111111 at sip.flowroute.com>;tag=SDvtmh399-gK0a9c4821;tag=SDvtmh399-gK0a9c4821
   To: "" <sip:XXXX at sip.flowroute.com;transport=udp>;tag=KcKN11gj6arQS
   Call-ID: 7fe743c9-2f4c-122f-9790-f23c91931894
   CSeq: 4313 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-b1d1beb 2011-07-18 23-08-16 -0400
   Content-Length: 0
   
   ------------------------------------------------------------------------

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

At this point the fax actually proceeds for a few seconds, but I get a BYE
after about 10 seconds:

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

recv 708 bytes from udp/[216.115.69.144]:5060 at 21:29:36.285179:
   ------------------------------------------------------------------------
   BYE sip:gw+flowroute at 111.111.111.111:5080;transport=udp;gw=flowroute SIP/2.0
   Record-Route: <sip:216.115.69.144;lr>
   Record-Route: <sip:216.115.69.133;lr>
   From: <sip:11111111111 at sip.flowroute.com>;tag=SDvtmh399-gK0a9c4821;tag=SDvtmh399-gK0a9c4821
   To: "" <sip:XXXX at sip.flowroute.com;transport=udp>;tag=KcKN11gj6arQS
   Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bKecbe.d96adce16997b042396d38106987a638.0
   Via: SIP/2.0/UDP 216.115.69.133;branch=z9hG4bKecbe.75fcc1a3bc1f3a68e4c0241e8b6b0298.0
   Via: SIP/2.0/UDP 67.16.125.60:5060;branch=z9hG4bKmthalp308g505g83r340sd00001d3.1
   Call-ID: 7fe743c9-2f4c-122f-9790-f23c91931894
   CSeq: 4314 BYE
   Max-Forwards: 67
   Content-Length: 0
   
   ------------------------------------------------------------------------
             - - - - - - - - - - - - - - - - - - - - - - - - -



From sanja_angelova at hotmail.com  Mon Jul 25 11:48:25 2011
From: sanja_angelova at hotmail.com (Sanja Angelova)
Date: Mon, 25 Jul 2011 09:48:25 +0200
Subject: [Freeswitch-users] (no subject)
Message-ID: <DUB109-W4708BC652AAED846BBF33AFE330@phx.gbl>


Dear all!

I am new to FreeSwitch and I am trying to map a number for example 654321 to sip adress like this 1005 at 10.1.1.1 with configuring a local DNS server. When I write enum 654321 in fs_cli it gives me:  

Offered Routes:
Order    Pref    Service       Route
==============================================================================
10    100    E2U+sip       sofia/${use_profile}/1005 at 10.1.1.1

and seems to be fine.
But when I make a call from my soft phone, the FreeSwitch ends up my call and it gives me error 503 service unavailable. 
I have no idea what the problem is, can you please help me?

Thank you in advance
 		 	   		  
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From slash at nat-tele.com  Mon Jul 25 11:58:18 2011
From: slash at nat-tele.com (slash at nat-tele.com)
Date: Mon, 25 Jul 2011 11:58:18 +0400
Subject: [Freeswitch-users] rxfax on b-leg
In-Reply-To: <4E2ACE39.2080007@coppice.org>
References: <6c1cf419782ba6781928a095d3bc19bc@nat-tele.com>
	<4E2ACE39.2080007@coppice.org>
Message-ID: <4851d0b4a640bf82e068c5d6cc6079ed@nat-tele.com>

 On Sat, 23 Jul 2011 21:35:53 +0800, Steve Underwood 
 <steveu at coppice.org> wrote:
> On 07/22/2011 12:56 AM, slash at nat-tele.com wrote:
>>   Hello!
>>
>>
>>   I wonder if there is a way to make rxfax recieve fax on outgoing 
>> call?
>>   The idea is that registered FS user dials out, than tone_detect is 
>> used
>>   to detect fax tone and start rxfax. The difference is that rxfax 
>> should
>>   be run on B-leg, instead of A-leg in incoming fax call scenario.
>>
>>   I've tried to make a test config with outgoing call, rxfax&
>>   tone_detect, but when I dial out and remote user starts sending 
>> fax to
>>   me, FS detects a tone correctly, starts rxfax, but on my leg 
>> (A-leg, i
>>   hear fax tones) and B-leg (with real fax machine) is hanged up.
>>
>>   Are there are solutions to explain somehow to rxfax that it should 
>> be
>>   run on B-leg, not on A-leg?
>>
>>   Thanks in advance!
>>
>>   With best regards,
>>   Kirill.
> If you use rxfax at the outgoing end of a call, mod_spandsp expects 
> to
> get that FAX by the polling procedure (i.e. the one you use when you
> dial into a FAX server to collect an information FAX from it). If you
> want to freely act as a simple FAX transmitter or receiver in the 
> middle
> of a call, the current module code may not properly allow for that. 
> As
> someone else said, there are tricks related to call transfers which 
> can
> be used to coax the fax instance into caller or answerer mode. 
> Spandsp
> does pretty much everything the FAX spec allows. If there is demand 
> we
> could provide access to more of that functionality.
>
> Steve
>

 The idea with manual call transfer is good, but it's hard to remember
 that every time you should ask remote end to press *1 before starting 
 fax. It's much more
 comfortable for everybody to be able use rxfax as a simple FAX 
 transmitter or
 receiver in the middle of a call... There is a demand for that 
 functionality : )

 With best regards,
 Kirill.



From ajaysharma842000 at yahoo.co.in  Mon Jul 25 12:49:38 2011
From: ajaysharma842000 at yahoo.co.in (Jayakrishnanu)
Date: Mon, 25 Jul 2011 01:49:38 -0700 (PDT)
Subject: [Freeswitch-users]  sipx conference API
Message-ID: <32129940.post@talk.nabble.com>


Hi Folks
Is there any way to add a person in to existing conference?

> I configured sipx and I successfully made calls through call controller
> API
> but when I tried to implement conference web service API I could get the
> response from the argument xml_list,volume_in,kick,lock but when I tried
> to dial a number in to the conference using "curl --digest -k -X PUT
> https://userid:Password at hostname:8085/conference/conference-
> name/dial/XXXXX/dial-number".I couldnt get what is the
> <endpoint_module_name> for the same.

Can anybody give me the correct url for dial service in conference and how
can I figure what is the endpoint-name in my system?
-- 
View this message in context: http://old.nabble.com/sipx-conference-API-tp32129940p32129940.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From sanja_angelova at hotmail.com  Mon Jul 25 15:27:08 2011
From: sanja_angelova at hotmail.com (Sanja Angelova)
Date: Mon, 25 Jul 2011 13:27:08 +0200
Subject: [Freeswitch-users] Help
Message-ID: <DUB109-W127E1362329DCC21A432D3BFE330@phx.gbl>






Dear all!

I am new to FreeSwitch and I am trying to map a number 
for example 654321 to sip adress like this 1005 at 10.1.1.1 with 
configuring a local DNS server. When I write enum 654321 in fs_cli it 
gives me:  

Offered Routes:
Order    Pref    Service       Route
==============================================================================
10    100    E2U+sip       sofia/${use_profile}/1005 at 10.1.1.1

and seems to be fine.
But when I make a call from my soft phone, the FreeSwitch ends up my call and it gives me error 503 service unavailable. 
I have no idea what the problem is, can you please help me?

Thank you in advance
 		 	   		  
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From rmartinez at redvoiss.net  Mon Jul 25 17:37:41 2011
From: rmartinez at redvoiss.net (Ricardo Martinez)
Date: Mon, 25 Jul 2011 09:37:41 -0400
Subject: [Freeswitch-users] Problem with Freeswitch/Sangoma D100.
In-Reply-To: aa54cbcc6870ad57562eae38969f10a9@mail.gmail.com
References: aa54cbcc6870ad57562eae38969f10a9@mail.gmail.com
Message-ID: <bfb8637cd962b2c7a91d38c65ae6d28e@mail.gmail.com>

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 SDP of the initial
INVITE message has the parameters :

?a=rtpmap:18 G729/8000?

?a=fmtp:18 annexb=no?



Then the INVITE transmitted to the gateway B it also has the same
parameters.

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.

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.-
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From msc at freeswitch.org  Mon Jul 25 19:07:39 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 25 Jul 2011 08:07:39 -0700
Subject: [Freeswitch-users] Asterisk dial status
In-Reply-To: <1311533112.57294.YahooMailClassic@web161004.mail.bf1.yahoo.com>
References: <6A6B4C284AD15042B429EB9D904544AD021FD89DC2@NY1-EXMB-01.ip-soft.net>
	<1311533112.57294.YahooMailClassic@web161004.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxVMsRo8UmqPosT0HSQGvZWg=VjyPKwqGVkpUgRR_cLk=A@mail.gmail.com>

What you are trying to do is not possible in this fashion. The bridge occurs
in the same thread as the perl script. The thread can only do one thing at a
time.

If you need to know the dial status while the thread is still running you'll
want a separate event socket app running and listening for channel status
events. If you just need to know the dial status once the bridge is complete
(i.e. once the call ends) then just use a api_hangup_hook chan var to launch
a script to do whatever it is that you need done.

-MC

On Sun, Jul 24, 2011 at 11:45 AM, Sam <lakersman2006 at yahoo.com> wrote:

> For the life of me I am still unable to get the bridged call status, when I
> subscribe to an "event" I am not here is my perl code snippet below, what am
> I doing wrong?
>
>
> my $fs = new ESL::ESLconnection($host, $port, $password);
>
> $session->answer();
>
> if ($session->ready ())
> {
>     #set bridge settings
>     $session->execute("set", "ringback=$ringback_tone");
>     $session->execute("set", "instant_ringback=true");
>     $session->execute("set", "ignore_early_media=false");
>     $session->execute("set", "call_timeout=60");
>
>     $session->execute("set", "progress_timeout=15");
>     $session->execute("set", "origination_caller_id_number=$caller_id");
>     $session->execute("set", "continue_on_fail=false");
>     $session->execute("bridge",
> "sofia/gateway/provider1/$destination_number");
>
>     $fs->sendRecv("event plain ALL");
>     $fs->send("linger");
>
>     while($fs->connected())
>     {
>         $data = $fs->recvEvent();
>
>         if ( $data =~ m/Answered-State: (\w+)/m )
>         {
>            my $state = $1;
>
>            #print "Channel state is $state\n";
>
>            if ( $state eq 'answered' ) {
>
>              freeswitch::consoleLog("INFO", "CALL ANSWERED\n");
>
>            }
>            else
>            {
>                 freeswitch::consoleLog("INFO", "NO CALLS ANSWERED\n");
>            }
>          }
>     }
> }
>
> $session->hangup();
>
> return 1;
>
>
> --- On *Fri, 7/22/11, Hector Geraldino <Hector.Geraldino at ip-soft.net>*wrote:
>
>
> From: Hector Geraldino <Hector.Geraldino at ip-soft.net>
> Subject: Re: [Freeswitch-users] Asterisk dial status
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Friday, July 22, 2011, 2:28 PM
>
>
> It does.
>
>
>
> You can check the events by connecting to FS on inbound mode, or using FS
> outbound socket connection to an application designed by you.
>
>
>
> As you already have a perl script to interact with FS, you can modify it to
> connect to FS (inbound mode) and listen for events. You can subscribe to the
> events of a particular call, or filter an specific set of events. I haven?t
> used Asterisk before so I?m not sure how the AGI fits on the FS
> architecture.
>
>
>
> You can get more info at:
>
> http://wiki.freeswitch.org/wiki/Mod_event_socket
>
>
>
> And a working demo at:
>
> http://wiki.freeswitch.org/wiki/Perl_freeswitch_client_example
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sam
> *Sent:* Friday, July 22, 2011 4:23 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Asterisk dial status
>
>
>
> Sam,
>
> Does this require event sockets? Or how would I check these events?
>
> --- On *Fri, 7/22/11, Hector Geraldino <Hector.Geraldino at ip-soft.net>*wrote:
>
>
> From: Hector Geraldino <Hector.Geraldino at ip-soft.net>
> Subject: Re: [Freeswitch-users] Asterisk dial status
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Friday, July 22, 2011, 11:57 AM
>
> Hi Sam,
>
>
>
> My guess is that you can subscribe to the events of the channel and
> evaluate the channel events. Look at:
>
>
>
> http://wiki.freeswitch.org/wiki/Event_List
>
>
>
> specifically the channel events list.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sam
> *Sent:* Friday, July 22, 2011 2:32 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Asterisk dial status
>
>
>
> Hi,
>
> I am trying to port over an Asterisk AGI script that does a call bridge to
> Freeswitch using perl, I wanted to know what is the equivalent of the
> asterisk dial status like: ANSWER, NO_ANSWER, CANCEL, BUSY, CONGESTION,
> CHANUNAVAIL ? I wanted to know if what methods can be used to gets these
> statues?
>
>
>
>
> -----Inline Attachment Follows-----
>
> _______________________________________________
> 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
>
>
>
> -----Inline Attachment Follows-----
>
> _______________________________________________
> 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://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From ijurado at econcept.es  Mon Jul 25 19:47:23 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Mon, 25 Jul 2011 17:47:23 +0200
Subject: [Freeswitch-users] Altering arbitrary channels
In-Reply-To: <CAFiqYu=Z3yp2-kkze6jmyCxAq83-etzjXeFByGd3WZk0CujVaQ@mail.gmail.com>
References: <CABj_V-m=MGKeSSYZ+tNU1mvyoKDrf-18s7WEO2XK_tEmafjQnA@mail.gmail.com>
	<CAFiqYumgaRcFofreA+4HB9qc+EaoN4RumNS-j2vGpKJs+7WpBg@mail.gmail.com>
	<CABj_V-kZbzmTBbvRa-bDK_Sib_H_yJJZz1x-dzyn+J1vgbg2Eg@mail.gmail.com>
	<CAFiqYu=Z3yp2-kkze6jmyCxAq83-etzjXeFByGd3WZk0CujVaQ@mail.gmail.com>
Message-ID: <CABj_V-mWjSPZ4Wz0+9PE1=OGcAB6u4qLSfMYeXFBJZ=FD6i+DQ@mail.gmail.com>

On Mon, Jul 25, 2011 at 2:30 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
> There are 2 types of commands... dialplan apps (which execute on the
> current channel) and API commands (which may or may not involve a
> channel, when they do they always need its UUID).
>
> APIs are generally executed from the console but can be used in the
> dialplan. It's a slightly different syntax though:
> <action application="set" data="result=${api arg1 arg2 arg3}"/>

The mod_commands wiki page shows a slightly different syntax:

<action application="set" data="api_result=${sched_api(+5 none avmd
${uuid} start)}"/>

Are the brackets optional?

And also, a command like "uuid_setvar" probably does not generate any
value if it is evaluated like an expression.  Which value would be
assigned to the channel variable in such case?

Thanks again.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From kbdfck at gmail.com  Mon Jul 25 21:00:53 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Mon, 25 Jul 2011 21:00:53 +0400
Subject: [Freeswitch-users] Asterisk dial status
In-Reply-To: <CAKzWOxVMsRo8UmqPosT0HSQGvZWg=VjyPKwqGVkpUgRR_cLk=A@mail.gmail.com>
References: <6A6B4C284AD15042B429EB9D904544AD021FD89DC2@NY1-EXMB-01.ip-soft.net>
	<1311533112.57294.YahooMailClassic@web161004.mail.bf1.yahoo.com>
	<CAKzWOxVMsRo8UmqPosT0HSQGvZWg=VjyPKwqGVkpUgRR_cLk=A@mail.gmail.com>
Message-ID: <CA+GmxbZxLuWicC6Yz4vA1_yWr4O=Pw44sWJrJ+AAQtTBfc31Fg@mail.gmail.com>

Actually you have no other way except parsing channel events not only from
parent channel but from child channels too since there can be many child
channels in bridge attempt.
Actually, FS doesn't have a direct method like Asterisk DIALSTATUS. If you
answer incoming call and then doing bridge to some endpoint, you will see
ANSWER in disposition variables even in channel events.
I can be wrong, but I haven't found any other way as to parse hangup /
hangup_complete / answer events of parent and child channels to correctly
determine disposition, cause codes and so on in all cases.

2011/7/25 Michael Collins <msc at freeswitch.org>

> What you are trying to do is not possible in this fashion. The bridge
> occurs in the same thread as the perl script. The thread can only do one
> thing at a time.
>
> If you need to know the dial status while the thread is still running
> you'll want a separate event socket app running and listening for channel
> status events. If you just need to know the dial status once the bridge is
> complete (i.e. once the call ends) then just use a api_hangup_hook chan var
> to launch a script to do whatever it is that you need done.
>
> -MC
>
>
> On Sun, Jul 24, 2011 at 11:45 AM, Sam <lakersman2006 at yahoo.com> wrote:
>
>>   For the life of me I am still unable to get the bridged call status,
>> when I subscribe to an "event" I am not here is my perl code snippet below,
>> what am I doing wrong?
>>
>>
>> my $fs = new ESL::ESLconnection($host, $port, $password);
>>
>> $session->answer();
>>
>> if ($session->ready ())
>> {
>>     #set bridge settings
>>     $session->execute("set", "ringback=$ringback_tone");
>>     $session->execute("set", "instant_ringback=true");
>>     $session->execute("set", "ignore_early_media=false");
>>     $session->execute("set", "call_timeout=60");
>>
>>     $session->execute("set", "progress_timeout=15");
>>     $session->execute("set", "origination_caller_id_number=$caller_id");
>>     $session->execute("set", "continue_on_fail=false");
>>     $session->execute("bridge",
>> "sofia/gateway/provider1/$destination_number");
>>
>>     $fs->sendRecv("event plain ALL");
>>     $fs->send("linger");
>>
>>     while($fs->connected())
>>     {
>>         $data = $fs->recvEvent();
>>
>>         if ( $data =~ m/Answered-State: (\w+)/m )
>>         {
>>            my $state = $1;
>>
>>            #print "Channel state is $state\n";
>>
>>            if ( $state eq 'answered' ) {
>>
>>              freeswitch::consoleLog("INFO", "CALL ANSWERED\n");
>>
>>            }
>>            else
>>            {
>>                 freeswitch::consoleLog("INFO", "NO CALLS ANSWERED\n");
>>            }
>>          }
>>     }
>> }
>>
>> $session->hangup();
>>
>> return 1;
>>
>>
>> --- On *Fri, 7/22/11, Hector Geraldino <Hector.Geraldino at ip-soft.net>*wrote:
>>
>>
>> From: Hector Geraldino <Hector.Geraldino at ip-soft.net>
>> Subject: Re: [Freeswitch-users] Asterisk dial status
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Date: Friday, July 22, 2011, 2:28 PM
>>
>>
>>  It does.
>>
>>
>>
>> You can check the events by connecting to FS on inbound mode, or using FS
>> outbound socket connection to an application designed by you.
>>
>>
>>
>> As you already have a perl script to interact with FS, you can modify it
>> to connect to FS (inbound mode) and listen for events. You can subscribe to
>> the events of a particular call, or filter an specific set of events. I
>> haven?t used Asterisk before so I?m not sure how the AGI fits on the FS
>> architecture.
>>
>>
>>
>> You can get more info at:
>>
>> http://wiki.freeswitch.org/wiki/Mod_event_socket
>>
>>
>>
>> And a working demo at:
>>
>> http://wiki.freeswitch.org/wiki/Perl_freeswitch_client_example
>>
>>
>>
>>
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sam
>> *Sent:* Friday, July 22, 2011 4:23 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] Asterisk dial status
>>
>>
>>
>> Sam,
>>
>> Does this require event sockets? Or how would I check these events?
>>
>> --- On *Fri, 7/22/11, Hector Geraldino <Hector.Geraldino at ip-soft.net>*wrote:
>>
>>
>> From: Hector Geraldino <Hector.Geraldino at ip-soft.net>
>> Subject: Re: [Freeswitch-users] Asterisk dial status
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Date: Friday, July 22, 2011, 11:57 AM
>>
>> Hi Sam,
>>
>>
>>
>> My guess is that you can subscribe to the events of the channel and
>> evaluate the channel events. Look at:
>>
>>
>>
>> http://wiki.freeswitch.org/wiki/Event_List
>>
>>
>>
>> specifically the channel events list.
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sam
>> *Sent:* Friday, July 22, 2011 2:32 PM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* [Freeswitch-users] Asterisk dial status
>>
>>
>>
>> Hi,
>>
>> I am trying to port over an Asterisk AGI script that does a call bridge to
>> Freeswitch using perl, I wanted to know what is the equivalent of the
>> asterisk dial status like: ANSWER, NO_ANSWER, CANCEL, BUSY, CONGESTION,
>> CHANUNAVAIL ? I wanted to know if what methods can be used to gets these
>> statues?
>>
>>
>>
>>
>> -----Inline Attachment Follows-----
>>
>> _______________________________________________
>> 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
>>
>>
>>
>> -----Inline Attachment Follows-----
>>
>> _______________________________________________
>> 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://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From msc at freeswitch.org  Mon Jul 25 21:09:43 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 25 Jul 2011 10:09:43 -0700
Subject: [Freeswitch-users] Altering arbitrary channels
In-Reply-To: <CABj_V-mWjSPZ4Wz0+9PE1=OGcAB6u4qLSfMYeXFBJZ=FD6i+DQ@mail.gmail.com>
References: <CABj_V-m=MGKeSSYZ+tNU1mvyoKDrf-18s7WEO2XK_tEmafjQnA@mail.gmail.com>
	<CAFiqYumgaRcFofreA+4HB9qc+EaoN4RumNS-j2vGpKJs+7WpBg@mail.gmail.com>
	<CABj_V-kZbzmTBbvRa-bDK_Sib_H_yJJZz1x-dzyn+J1vgbg2Eg@mail.gmail.com>
	<CAFiqYu=Z3yp2-kkze6jmyCxAq83-etzjXeFByGd3WZk0CujVaQ@mail.gmail.com>
	<CABj_V-mWjSPZ4Wz0+9PE1=OGcAB6u4qLSfMYeXFBJZ=FD6i+DQ@mail.gmail.com>
Message-ID: <CAKzWOxWNZc=+gEWarXop6AxLjE1sE5VNi7VjOdv6PFqBVsQNDQ@mail.gmail.com>

On Mon, Jul 25, 2011 at 8:47 AM, Isaac Jurado <ijurado at econcept.es> wrote:

> On Mon, Jul 25, 2011 at 2:30 PM, Steven Ayre <steveayre at gmail.com> wrote:
> >
> > There are 2 types of commands... dialplan apps (which execute on the
> > current channel) and API commands (which may or may not involve a
> > channel, when they do they always need its UUID).
> >
> > APIs are generally executed from the console but can be used in the
> > dialplan. It's a slightly different syntax though:
> > <action application="set" data="result=${api arg1 arg2 arg3}"/>
>
> The mod_commands wiki page shows a slightly different syntax:
>
> <action application="set" data="api_result=${sched_api(+5 none avmd
> ${uuid} start)}"/>
>
> Are the brackets optional?
>
Curly braces are not optional when using this syntax. When using the set dp
app and having ${api_name} in the data argument it will evaluate that
argument as an API command.


>
> And also, a command like "uuid_setvar" probably does not generate any
> value if it is evaluated like an expression.  Which value would be
> assigned to the channel variable in such case?
>
Not sure what you mean. There's no technical reason why you couldn't do
this:

<action application="set" data="${uuid_setvar(${uuid} var_name
var_value)}"/>

However, obviously there's no need to use uuid_setvar on the current channel
since you can just use the set application.

-MC


>
> Thanks again.
>
> --
> Isaac Jurado
> Internet Busines Solutions eConcept
>
> _______________________________________________
> 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
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>
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From ijurado at econcept.es  Mon Jul 25 21:49:50 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Mon, 25 Jul 2011 19:49:50 +0200
Subject: [Freeswitch-users] Altering arbitrary channels
In-Reply-To: <CAKzWOxWNZc=+gEWarXop6AxLjE1sE5VNi7VjOdv6PFqBVsQNDQ@mail.gmail.com>
References: <CABj_V-m=MGKeSSYZ+tNU1mvyoKDrf-18s7WEO2XK_tEmafjQnA@mail.gmail.com>
	<CAFiqYumgaRcFofreA+4HB9qc+EaoN4RumNS-j2vGpKJs+7WpBg@mail.gmail.com>
	<CABj_V-kZbzmTBbvRa-bDK_Sib_H_yJJZz1x-dzyn+J1vgbg2Eg@mail.gmail.com>
	<CAFiqYu=Z3yp2-kkze6jmyCxAq83-etzjXeFByGd3WZk0CujVaQ@mail.gmail.com>
	<CABj_V-mWjSPZ4Wz0+9PE1=OGcAB6u4qLSfMYeXFBJZ=FD6i+DQ@mail.gmail.com>
	<CAKzWOxWNZc=+gEWarXop6AxLjE1sE5VNi7VjOdv6PFqBVsQNDQ@mail.gmail.com>
Message-ID: <CABj_V-koaaFYQ8otaB6VGAkni4Hv8zq6zyzY0-kLQaCj3Kb5Sg@mail.gmail.com>

On Mon, Jul 25, 2011 at 7:09 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>>> <action application="set" data="result=${api arg1 arg2 arg3}"/>
>>
>> The mod_commands wiki page shows a slightly different syntax:
>>
>> <action application="set" data="api_result=${sched_api(+5 none avmd
>> ${uuid} start)}"/>
>>
>> Are the brackets optional?
>
> Curly braces are not optional when using this syntax.

I was actually asking about the parenthesis surrounding the API command
arguments.  Seems like the "bracket" term is a british dialect.  Sorry
for the confusing.

Anyway, from your "uuid_setvar" example below I can grasp that I should
better use parenthesis.

>> And also, a command like "uuid_setvar" probably does not generate any
>> value if it is evaluated like an expression. ?Which value would be
>> assigned to the channel variable in such case?
>
> Not sure what you mean. There's no technical reason why you couldn't do
> this:
> <action application="set" data="${uuid_setvar(${uuid} var_name
> var_value)}"/>

In all the examples I've seen, the use of "set" is something like:

<action application="set" data="foo=bar"/>

So I understood that the "foo=" part was mandatory.  Considering that
uuid_setvar seems to evaluate to nothing/null/empty, I was wondering if,
by writing something like the following:

<action application="set" data="foo=${uuid_setvar(${uuid} var_name
var_value)}"/>

What would be actually assigned to the "foo" variable.  But never mind,
if I can skip the "foo=" part it is enough for me.

Thank you very much.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From lakersman2006 at yahoo.com  Mon Jul 25 21:56:45 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Mon, 25 Jul 2011 10:56:45 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk dial status
In-Reply-To: <CAKzWOxVMsRo8UmqPosT0HSQGvZWg=VjyPKwqGVkpUgRR_cLk=A@mail.gmail.com>
Message-ID: <1311616605.8729.YahooMailClassic@web161003.mail.bf1.yahoo.com>

What I want to do is play different sound files back to the caller when the call is NO_ANSWER, BUSY, etc. So I don't think your suggestion would work.

--- On Mon, 7/25/11, Michael Collins <msc at freeswitch.org> wrote:

From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Asterisk dial status
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Monday, July 25, 2011, 8:07 AM

What you are trying to do is not possible in this fashion. The bridge occurs in the same thread as the perl script. The thread can only do one thing at a time.?
If you need to know the dial status while the thread is still running you'll want a separate event socket app running and listening for channel status events. If you just need to know the dial status once the bridge is complete (i.e. once the call ends) then just use a api_hangup_hook chan var to launch a script to do whatever it is that you need done.

-MC

On Sun, Jul 24, 2011 at 11:45 AM, Sam <lakersman2006 at yahoo.com> wrote:

For the life of me I am still unable to get the bridged call status, when I subscribe to an "event" I am not here is my perl code snippet below, what am I doing wrong?



my $fs = new ESL::ESLconnection($host, $port, $password);

$session->answer();

if ($session->ready ()) 
{
??? #set bridge settings
??? $session->execute("set", "ringback=$ringback_tone");?? 

??? $session->execute("set", "instant_ringback=true");??? ??? 
??? $session->execute("set", "ignore_early_media=false"); 
??? $session->execute("set", "call_timeout=60");??? ??? ??? ??? ??? ??? ??? 

???
 $session->execute("set", "progress_timeout=15");
??? $session->execute("set", "origination_caller_id_number=$caller_id");
??? $session->execute("set", "continue_on_fail=false");

??? $session->execute("bridge", "sofia/gateway/provider1/$destination_number");
???? 
??? $fs->sendRecv("event plain ALL");
??? $fs->send("linger");
??? 
??? while($fs->connected())

??? {
??? ??? $data = $fs->recvEvent();
??? ??? 
??? ??? if ( $data =~ m/Answered-State: (\w+)/m ) 
??? ??? {
??? ??? ?? my $state = $1;
??? ??? 
??? ??? ?? #print
 "Channel state is $state\n";
??? ??? 
??? ??? ?? if ( $state eq 'answered' ) {
??? ??? 
??? ??? ??? ?freeswitch::consoleLog("INFO", "CALL ANSWERED\n");
??? ??? 
??? ??? ?? }

??? ??? ?? else
??? ??? ?? {
??? ??? ?? ??? ?freeswitch::consoleLog("INFO", "NO CALLS ANSWERED\n");
??? ??? ?? }
??? ??? ?}??? 
??? }
}
?? 
$session->hangup();
?
return 1;


--- On Fri, 7/22/11, Hector Geraldino
 <Hector.Geraldino at ip-soft.net> wrote:


From: Hector Geraldino <Hector.Geraldino at ip-soft.net>
Subject: Re: [Freeswitch-users] Asterisk dial status
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

Date: Friday, July 22, 2011, 2:28 PM

It does. ?
You can check the events by connecting to FS on inbound mode, or using FS outbound socket connection to an application designed by you.  ?
As you already have a perl script to interact with FS, you can modify it to connect to FS (inbound mode) and listen for events. You can subscribe to the
 events of a particular call, or filter an specific set of events. I haven?t used Asterisk before so I?m not sure how the AGI fits on the FS architecture. ?
You can get more info at:http://wiki.freeswitch.org/wiki/Mod_event_socket
 ?And a
 working demo at: http://wiki.freeswitch.org/wiki/Perl_freeswitch_client_example
 ? ? ?
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sam

Sent: Friday, July 22, 2011 4:23 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Asterisk dial status ?
Sam,

Does this require event sockets? Or how would I check these events?

--- On Fri, 7/22/11, Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:

From: Hector Geraldino <Hector.Geraldino at ip-soft.net>
Subject: Re: [Freeswitch-users] Asterisk dial status

To: "FreeSWITCH Users Help"
 <freeswitch-users at lists.freeswitch.org>
Date: Friday, July 22, 2011, 11:57 AMHi Sam,
?My guess is that you can subscribe to the events of the channel and evaluate the channel events. Look at:
?http://wiki.freeswitch.org/wiki/Event_List
?specifically the channel events list.?
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sam

Sent: Friday, July 22, 2011 2:32
 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Asterisk dial status?
Hi,

I am trying to port over an Asterisk AGI script that does a call bridge to Freeswitch using perl, I wanted to know what is the equivalent of the asterisk dial status like: ANSWER, NO_ANSWER, CANCEL, BUSY, CONGESTION, CHANUNAVAIL ? I wanted to know if what methods can be used to gets these statues?
?
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From mitch.johnson7 at gmail.com  Mon Jul 25 22:25:13 2011
From: mitch.johnson7 at gmail.com (Mitch Johnson)
Date: Mon, 25 Jul 2011 14:25:13 -0400
Subject: [Freeswitch-users] One way calling issue over an internal trunk
In-Reply-To: <mailman.13657.1311590598.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.13657.1311590598.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <EA173C56-B66A-484C-BB97-8422BEA38944@gmail.com>

Concerning the NAT.  I will always be on the internal network, all external phones will come in on a VPN and then connect onto FreeSWITCH using the VPN.

I did the ./freeswitch -nonat.

I also changed the sip-ip and rtp-ip to the internal ip of the Freeswitch server on both external and internal xml files:

Internal:
<param name="rtp-ip" value="172.16.200.60"/>
 <param name="sip-ip" value="172.16.200.60"/>

external:
<param name="rtp-ip" value="172.16.200.60"/>
    <param name="sip-ip" value="172.16.200.60"/>
    <param name="ext-rtp-ip" value="local-network-acl"/>
    <param name="ext-sip-ip" value="local-network-acl"/>

I'm still plodding around FreeSWITCH so if I did this wrong please let me know.  

Thanks,

Mitch


On Jul 25, 2011, at 6:43 AM, freeswitch-users-request at lists.freeswitch.org wrote:

> From: Steven Ayre <steveayre at gmail.com>
> Subject: Re: [Freeswitch-users] One way calling issue over an internal trunk
> Date: July 25, 2011 3:05:39 AM EDT
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> I'll always be on the internal network so should never need the nat.
> 
> Try starting FS with the -nonat option. It'll start faster and skip all the stuff where it tries to find your external IP and open routes through the NAT firewall.
> 
> On your sip profiles set sip-ip and rtp-ip to your internal IP.
> 
> Once you've done that you should have 2-way audio to anywhere on your LAN.
> 
> -Steve
> 
> 
> 
> 
> On 25 July 2011 05:08, Mitch Johnson <mitch.johnson7 at gmail.com> wrote:
> Sorry for the delayed response, got tied up in other things.  A quick recap. I can dial from my call manager to Freeswitch but cannot dial from freeswitch to call manager.
> 
> I wasn't sure what you meant by setting the ext-sip-ip and ext-rtp-ip.  So I replaced the auto-nat with local-network-acl.  I'll always be on the internal network so should never need the nat.
> 
> I saw in the acl.conf.xml that it automatically populates all my local subnets.  The networks are correct under the domains of the acl.conf.
> 
> I'm not sure where else to look.  Any help would be greatly appreciated.
> 
> Thanks so much,
> 
> Mitch
> 
> On Jul 19, 2011, at 11:29 PM, freeswitch-users-request at lists.freeswitch.org wrote:
> 
>> From: Brian West <brian at freeswitch.org>
>> Subject: Re: [Freeswitch-users] One way calling issue over an internal trunk
>> Date: July 19, 2011 8:55:02 PM EDT
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> 
>> 
>> you need to set the ext-sip-ip and ext-rtp-ip and deal with setting local-network-acl to the proper network/mask because you're answering with RFC1918 media IP's to the cisco? its retarded and will just start sending to those IP's and they'll usually never make it back to you.
>> 
>> /b
>> 
>> On Jul 19, 2011, at 7:16 PM, Mitch Johnson wrote:
>> 
>>> I have a connection between my FreeSWITCH and a Cisco CallManager:
>>> 
>>> This is from the default.xml
>>> 
>>> <extension name="Dial to CM8">
>>>      <condition field="destination_number" expression="^(800\d)$">
>>>        <action application="bridge" data="sofia/internal/$1 at 172.16.200.100"/>
>>>      </condition>
>>>     </extension>
>> 
> 

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From msc at freeswitch.org  Mon Jul 25 22:36:27 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 25 Jul 2011 11:36:27 -0700
Subject: [Freeswitch-users] Asterisk dial status
In-Reply-To: <1311616605.8729.YahooMailClassic@web161003.mail.bf1.yahoo.com>
References: <CAKzWOxVMsRo8UmqPosT0HSQGvZWg=VjyPKwqGVkpUgRR_cLk=A@mail.gmail.com>
	<1311616605.8729.YahooMailClassic@web161003.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxVgAXisUH_gJGwYZXBwYZpPrA=h8Rq6eXU7kfFZ-y9FYQ@mail.gmail.com>

On Mon, Jul 25, 2011 at 10:56 AM, Sam <lakersman2006 at yahoo.com> wrote:

> What I want to do is play different sound files back to the caller when the
> call is NO_ANSWER, BUSY, etc. So I don't think your suggestion would work.
>
I think maybe you have a case of "Asteriskitis" in all this. :)

FreeSWITCH makes it easy to do this without actually checking the dial
status. The only thing you need to do is come up with a dialplan extension
that handles the various failure notifications to the caller. Consider an
extremely simple case:

<extension name="bridge_with_failover">
  <condition field="destination_number" expression="1?(\d{10})">
    <action application="set" data="hangup_after_bridge=false"/>
    <action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"/>
    <action application="bridge" data="sofia/gateway/mygw/+1$1">
    <!-- the stuff after the bridge only gets executed on the failure
conditions listed above -->
    <action application="playback" data="call_failed.wav"/>
    <action application="hangup"/>
  </condition>
</extension>

In the above case, if the call works then there's no problem. If the call
fails, dialplan processing continues on and plays to the caller an error. If
you want to get fancy and have something specific happen then you can either
create some dialplan extensions that play various sound files, or you can
launch a dialplan script (like your perl script) and check
${bridge_hangup_cause} to see what "really" happened on the bridge attempt.

Enjoy!
-MC
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From jeff at jefflenk.com  Mon Jul 25 22:52:31 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 25 Jul 2011 11:52:31 -0700 (PDT)
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <DUB109-W4708BC652AAED846BBF33AFE330@phx.gbl>
References: <DUB109-W4708BC652AAED846BBF33AFE330@phx.gbl>
Message-ID: <1311619951536-6619518.post@n2.nabble.com>

go ahead and attach a small debug log here and perhaps someone will have a
look

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/no-subject-tp6618626p6619518.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gavin.henry at gmail.com  Mon Jul 25 22:59:40 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 25 Jul 2011 19:59:40 +0100
Subject: [Freeswitch-users] Asterisk dial status
In-Reply-To: <CAKzWOxVgAXisUH_gJGwYZXBwYZpPrA=h8Rq6eXU7kfFZ-y9FYQ@mail.gmail.com>
References: <CAKzWOxVMsRo8UmqPosT0HSQGvZWg=VjyPKwqGVkpUgRR_cLk=A@mail.gmail.com>
	<1311616605.8729.YahooMailClassic@web161003.mail.bf1.yahoo.com>
	<CAKzWOxVgAXisUH_gJGwYZXBwYZpPrA=h8Rq6eXU7kfFZ-y9FYQ@mail.gmail.com>
Message-ID: <CAA8_NKD7-iSw3zoFWKij0MSm_yopJs=MtSEo5b4aT0oJPxwK8w@mail.gmail.com>

Why not bgapi and listen for the CHANNEL_ANSWER event?

On 25 July 2011 19:36, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Jul 25, 2011 at 10:56 AM, Sam <lakersman2006 at yahoo.com> wrote:
>
>> What I want to do is play different sound files back to the caller when
>> the call is NO_ANSWER, BUSY, etc. So I don't think your suggestion would
>> work.
>>
> I think maybe you have a case of "Asteriskitis" in all this. :)
>
> FreeSWITCH makes it easy to do this without actually checking the dial
> status. The only thing you need to do is come up with a dialplan extension
> that handles the various failure notifications to the caller. Consider an
> extremely simple case:
>
> <extension name="bridge_with_failover">
>   <condition field="destination_number" expression="1?(\d{10})">
>     <action application="set" data="hangup_after_bridge=false"/>
>     <action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"/>
>     <action application="bridge" data="sofia/gateway/mygw/+1$1">
>     <!-- the stuff after the bridge only gets executed on the failure
> conditions listed above -->
>     <action application="playback" data="call_failed.wav"/>
>     <action application="hangup"/>
>   </condition>
> </extension>
>
> In the above case, if the call works then there's no problem. If the call
> fails, dialplan processing continues on and plays to the caller an error. If
> you want to get fancy and have something specific happen then you can either
> create some dialplan extensions that play various sound files, or you can
> launch a dialplan script (like your perl script) and check
> ${bridge_hangup_cause} to see what "really" happened on the bridge attempt.
>
> Enjoy!
> -MC
>
> _______________________________________________
> 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
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>
>


-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk
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From steveayre at gmail.com  Mon Jul 25 23:18:11 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 25 Jul 2011 20:18:11 +0100
Subject: [Freeswitch-users] Help
In-Reply-To: <DUB109-W127E1362329DCC21A432D3BFE330@phx.gbl>
References: <DUB109-W127E1362329DCC21A432D3BFE330@phx.gbl>
Message-ID: <CAFiqYu=_uZZVg48VXQcwaiuqJc0Tx6W-BRNXsY8Op9NXqXiepw@mail.gmail.com>

Can you start off by collecting a debug-level log of the call that's
failing? It should give some indiication of the problem.

-Steve



2011/7/25 Sanja Angelova <sanja_angelova at hotmail.com>

>  Dear all!
>
> I am new to FreeSwitch and I am trying to map a number for example 654321
> to sip adress like this 1005 at 10.1.1.1 with configuring a local DNS server.
> When I write enum 654321 in fs_cli it gives me:
>
> Offered Routes:
> Order    Pref    Service       Route
>
> ==============================================================================
> 10    100    E2U+sip       sofia/${use_profile}/1005 at 10.1.1.1
>
> and seems to be fine.
> But when I make a call from my soft phone, the FreeSwitch ends up my call
> and it gives me error 503 service unavailable.
> I have no idea what the problem is, can you please help me?
>
> Thank you in advance
>
> _______________________________________________
> 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
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>
>
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From msc at freeswitch.org  Mon Jul 25 23:25:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 25 Jul 2011 12:25:50 -0700
Subject: [Freeswitch-users] Asterisk dial status
In-Reply-To: <CAA8_NKD7-iSw3zoFWKij0MSm_yopJs=MtSEo5b4aT0oJPxwK8w@mail.gmail.com>
References: <CAKzWOxVMsRo8UmqPosT0HSQGvZWg=VjyPKwqGVkpUgRR_cLk=A@mail.gmail.com>
	<1311616605.8729.YahooMailClassic@web161003.mail.bf1.yahoo.com>
	<CAKzWOxVgAXisUH_gJGwYZXBwYZpPrA=h8Rq6eXU7kfFZ-y9FYQ@mail.gmail.com>
	<CAA8_NKD7-iSw3zoFWKij0MSm_yopJs=MtSEo5b4aT0oJPxwK8w@mail.gmail.com>
Message-ID: <CAKzWOxU2UwXUPM7Dd8FQRcppQ2y0CTMNiDf2ZgQ0ofup5pwA_g@mail.gmail.com>

On Mon, Jul 25, 2011 at 11:59 AM, Gavin Henry <gavin.henry at gmail.com> wrote:

> Why not bgapi and listen for the CHANNEL_ANSWER event?
>
Dealer's choice. You can totally do that if you so desire. It all depends on
what the ultimate goal is. If you just need to have a few calls hit the dp,
attempt bridge, then play messages based on failure types then the dialplan
itself can handle it with just continue_on_fail and transfer apps. You can
throw in a dp script if you have some specific logic you wish to apply.
Keeps it clean and simple. However, if you are wanting to do some async
stuff like real-time status updates, etc. then event socket w/ bgapi
originates is definitely the way to go.

-MC
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From Joshua.Foshee at LogixCom.com  Mon Jul 25 23:54:30 2011
From: Joshua.Foshee at LogixCom.com (Joshua Foshee)
Date: Mon, 25 Jul 2011 14:54:30 -0500
Subject: [Freeswitch-users] Can't get variable set in dial plan
Message-ID: <06502C073AD9394AADB3CA7FD94931BC0779052F@okc1x1.Logixcom.com>

I am trying to set a variable in the dial plan and use it on a
condition. I have pushed the variable setting as close to the condition
for testing and this is what I get. Why does it not see the variable
set?

 

<extension name="public_did">

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

        <action inline="true" application="set" data="hours=1"/>

    </condition>

    <condition field="hours" expression="1">

        <action application="answer"/>

        <action application="ivr" data="GAC-BH"/>

        <action application="hangup"/>

        <anti-action application="answer"/>

        <anti-action application="ivr" data="GAC-BH"/>

        <anti-action application="hangup"/>

    </condition>

  </extension>

 

Here is the console output

 

Dialplan: FreeTDM/1:7/405790xxx8 Regex (PASS) [public_did]
destination_number(405790xxx8) =~ /^(.*)$/ break=on-false

Dialplan: FreeTDM/1:7/405790xxx8 Action set(hours=1) INLINE

EXECUTE FreeTDM/1:7/405790xxx8 set(hours=1)

2011-07-25 12:05:25.191357 [DEBUG] mod_dptools.c:1063
FreeTDM/1:7/405790xxx8 SET [hours]=[1]

Dialplan: FreeTDM/1:7/405790xxx8 Regex (FAIL) [public_did] hours() =~
/1/ break=on-false

Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action answer()

Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action ivr(GAC-BH)

Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action hangup()

 

 

 

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From Joshua.Foshee at LogixCom.com  Tue Jul 26 00:11:39 2011
From: Joshua.Foshee at LogixCom.com (Joshua Foshee)
Date: Mon, 25 Jul 2011 15:11:39 -0500
Subject: [Freeswitch-users] Callcenter MOD question
Message-ID: <06502C073AD9394AADB3CA7FD94931BC07790532@okc1x1.Logixcom.com>

I am setting up a queue and I have ring all set with everything else
sent to default pretty much. What I am trying to accomplish is that
agents have phone that show 4 lines for their extension. If they are on
the phone with an outbound call I want to skip trying to send them a
call. How is this accomplished where the callcenter mod knows to not
send a call to them. All agents are static assigned in the callcenter
config file.

 

Copy of config file below.

 

<configuration name="callcenter.conf" description="CallCenter">

  <settings>

  </settings>

<queues>

    <queue name="sales at default">

      <param name="strategy" value="ring-all"/>

      <param name="moh-sound" value="$${hold_music}"/>

      <param name="time-base-score" value="queue"/>

      <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="14400"/>

      <param name="max-wait-time" value="0"/>

      <param name="max-wait-time-with-no-agent" value="120"/>

    </queue>

    <queue name="collection at default">

      <param name="strategy" value="ring-all"/>

      <param name="moh-sound" value="$${hold_music}"/>

      <param name="time-base-score" value="system"/>

      <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"/>

      <param name="max-wait-time" value="0"/>

      <param name="max-wait-time-with-no-agent" value="120"/>

    </queue>

  </queues>

 

  <agents>

    <agent name="513 at default" type="callback"
contact="[call_timeout=10]user/513 at 192.168.1.3" status="Available"
max-no-answer="3" wrap-up time="10" reject-delay-time="10"
busy-delay-time="60" />

 </agents>

 

<tiers>

    <tier agent="513 at default" queue="sales at default" level="1"
position="1"/>

    <tier agent="513 at default" queue="collection at default" level="1"
position="1"/>

</tiers>

 

</configuration>

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From steveayre at gmail.com  Tue Jul 26 00:39:34 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 25 Jul 2011 21:39:34 +0100
Subject: [Freeswitch-users] Can't get variable set in dial plan
In-Reply-To: <06502C073AD9394AADB3CA7FD94931BC0779052F@okc1x1.Logixcom.com>
References: <06502C073AD9394AADB3CA7FD94931BC0779052F@okc1x1.Logixcom.com>
Message-ID: <CAFiqYum3GpP8u9aHs85pmmPBC+53RazfNV7oupBLKMU7bFZB-Q@mail.gmail.com>

Variables are not fields. Fields are a specific set of attributes of the
call... you can however check variables in a condition, it just has a
slightly different syntax. Try this.

<extension name="public_did">

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

        <action inline="true" application="set" data="hours=1"/>

    </condition>

    <condition field="${hours}" expression="^1$">

        <action application="answer"/>

        <action application="ivr" data="GAC-BH"/>

        <action application="hangup"/>

        <anti-action application="answer"/>

        <anti-action application="ivr" data="GAC-BH"/>

        <anti-action application="hangup"/>

    </condition>

  </extension>

Also note I changed your regex of "1" to "^1$". The first matches any string
containing a digit 1 including 10, 21, 100, 200351. The latter matches the
start (^) and end ($) of the string so will only match the exact string of
1.

-Steve

On 25 July 2011 20:54, Joshua Foshee <Joshua.Foshee at logixcom.com> wrote:

> I am trying to set a variable in the dial plan and use it on a condition. I
> have pushed the variable setting as close to the condition for testing and
> this is what I get. Why does it not see the variable set?****
>
> ** **
>
> <extension name="public_did">****
>
>     <condition field="destination_number" expression="^(.*)$">****
>
>         <action inline="true" application="set" data="hours=1"/>****
>
>     </condition>****
>
>     <condition field="hours" expression="1">****
>
>         <action application="answer"/>****
>
>         <action application="ivr" data="GAC-BH"/>****
>
>         <action application="hangup"/>****
>
>         <anti-action application="answer"/>****
>
>         <anti-action application="ivr" data="GAC-BH"/>****
>
>         <anti-action application="hangup"/>****
>
>     </condition>****
>
>   </extension>****
>
> ** **
>
> Here is the console output****
>
> ** **
>
> Dialplan: FreeTDM/1:7/405790xxx8 Regex (PASS) [public_did]
> destination_number(405790xxx8) =~ /^(.*)$/ break=on-false****
>
> Dialplan: FreeTDM/1:7/405790xxx8 Action set(hours=1) INLINE****
>
> EXECUTE FreeTDM/1:7/405790xxx8 set(hours=1)****
>
> 2011-07-25 12:05:25.191357 [DEBUG] mod_dptools.c:1063
> FreeTDM/1:7/405790xxx8 SET [hours]=[1]****
>
> Dialplan: FreeTDM/1:7/405790xxx8 Regex (FAIL) [public_did] hours() =~ /1/
> break=on-false****
>
> Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action answer()****
>
> Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action ivr(GAC-BH)****
>
> Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action hangup()****
>
> ** **
>
> ** **
>
> ** **
>
> _______________________________________________
> 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
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From gavin.henry at gmail.com  Tue Jul 26 01:01:35 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 25 Jul 2011 22:01:35 +0100
Subject: [Freeswitch-users] Asterisk dial status
In-Reply-To: <CAKzWOxU2UwXUPM7Dd8FQRcppQ2y0CTMNiDf2ZgQ0ofup5pwA_g@mail.gmail.com>
References: <CAKzWOxVMsRo8UmqPosT0HSQGvZWg=VjyPKwqGVkpUgRR_cLk=A@mail.gmail.com>
	<1311616605.8729.YahooMailClassic@web161003.mail.bf1.yahoo.com>
	<CAKzWOxVgAXisUH_gJGwYZXBwYZpPrA=h8Rq6eXU7kfFZ-y9FYQ@mail.gmail.com>
	<CAA8_NKD7-iSw3zoFWKij0MSm_yopJs=MtSEo5b4aT0oJPxwK8w@mail.gmail.com>
	<CAKzWOxU2UwXUPM7Dd8FQRcppQ2y0CTMNiDf2ZgQ0ofup5pwA_g@mail.gmail.com>
Message-ID: <CAA8_NKDa5sOf-=fQ_do=zf+SFGQDaNjXb5uddy9gQzK7f1Bk_A@mail.gmail.com>

Yeah, that's what I'm doing and just finished tonight. Still a bit to go but
gist is:

1. Follow @SureVoIPLabs
2. Bot auto follows you back if not already following
3. Sends you a DM with a thank you and instructions.
4. You then send a DM like so:

d @SureVoIPLabs my_mobile_number/number_I_wish_to_call

5. Do some tariff checks and fire off a bgapi with 2 or 10min limit
6. Send you a DM back

All free but limited to UK for now. Will be live next week or so.  Full
version with international and App auth coming with credit topups via DM too
:)

Gav.

http://www.surevoip.co.uk
> On Mon, Jul 25, 2011 at 11:59 AM, Gavin Henry <gavin.henry at gmail.com>
wrote:
>
>> Why not bgapi and listen for the CHANNEL_ANSWER event?
>>
> Dealer's choice. You can totally do that if you so desire. It all depends
on
> what the ultimate goal is. If you just need to have a few calls hit the
dp,
> attempt bridge, then play messages based on failure types then the
dialplan
> itself can handle it with just continue_on_fail and transfer apps. You can
> throw in a dp script if you have some specific logic you wish to apply.
> Keeps it clean and simple. However, if you are wanting to do some async
> stuff like real-time status updates, etc. then event socket w/ bgapi
> originates is definitely the way to go.
>
> -MC
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From bryansmart at bryansmart.com  Tue Jul 26 01:42:17 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Mon, 25 Jul 2011 17:42:17 -0400
Subject: [Freeswitch-users] Getting Started with Speech Recognition
Message-ID: <6FE58ED3-806D-440C-BF33-ABE32D159A72@bryansmart.com>

I've been attempting to experiment with this part of Freeswitch, but it seems that this is an area that hasn't yet received a lot of attention with regard to docs and samples. Perhaps a some of you that are experienced in this area wouldn't mind clearing up some of my confusion?

First, I have FS working fine (outside of ASR). Pocketsphinx is built and loaded. As for me, I have some experience with VXML, and generally know how a grammar works, but the last phone ASR projects that I was involved with were several years back on Tellme Studio.

First, as to grammars, I'm not sure, but it seems as though FS has used different formats at different times. The mod_pocketsphinx page says that JSGF is the current format. Can the JSGF grammar files be loaded without being compiled to another format? I ask that, as some of the examples, such as the LUA Directory example, talk about compiling grammars, and the format shown there doesn't appear exactly the same as what I see from the JSGF examples. Is the info about compiling grammars obsolete?

Next, I'm not clear about the use of the detect_speech command. The wiki page lists some example forms of the command, but doesn't explain the purpose of the different forms, nor their arguments.

Here are some examples:

detect_speech <mod_name> <gram_name> <gram_path> [<addr>]

>From the examples, I see that "pocket sphinx" can be used for the module. I think that gram_path is the path to the grammar file, without the ".gram" suffix, and uses the grammar folder as a base. However, what is addr? An address? To, or for, what?

detect_speech grammaron <gram_name>

I think that

detect_speech grammaron <gram_name>

Might be used when a large .gram file has been loaded, and the code needs to enable a grammar for a specific context. However, I don't see a command for just loading a .gram file without also specifying a grammar name. If I use this grammaron command, will it search all files in the grammar folder for grammars that match the name, or is this just for enabling a grammar that I've already loaded through other means?

Maybe grammaron is used to activate a grammar that was disabled with

detect_speech grammaroff <gram_name>

But, if so, then what is the purpose of

detect_speech nogrammar <gram_name>

Does that also disable a grammar? Maybe it deletes the grammar from memory, also?

For:

detect_speech param <name> <value>

What are the possible parameter names and acceptable values?

detect_speech start_input_timers

What are the purposes of the start_input_timers? Does this cause ASR to time out if no input is recognized? Does it cause ASR to pause for a time before beginning to recognize speech?

I've tried to write up a simple grammar with only a few terms, and have a LUA IVR script attempt to detect speech using this grammar. My input callback is never called with any sort of speech event when I speak, though, and I don't receive any FS console errors. I must be making some incorrect assumptions about how the commands are used, or how the grammar is formatted.

Perhaps my tests aren't working because there is a problem with Pocketsphinx. I can't try the Pizza demo to verify that Pocketsphinx is working correctly, as I'm unable to download its grammar while bkw.org is offline, and no other sources for the grammar come up on a Google search.

Finally, I'd like to know if there are any compelling reasons to use one scripting language over another for speech-driven IVRs. I have experience with Lua, Javascript, and PHP, so could use any of them. The Freeswitch book uses Lua a lot, so that's what I've been trying so far. Perhaps I'd be better served with Javascript?

Thanks for any pointers!

Bryan



From Joshua.Foshee at LogixCom.com  Tue Jul 26 01:55:22 2011
From: Joshua.Foshee at LogixCom.com (Joshua Foshee)
Date: Mon, 25 Jul 2011 16:55:22 -0500
Subject: [Freeswitch-users] Can't get variable set in dial plan
Message-ID: <06502C073AD9394AADB3CA7FD94931BC07790545@okc1x1.Logixcom.com>

That fixed it. The fix was to change hours to ${hours} in the condition
field.

 

Thanks,

Josh

 

From: Steven Ayre [mailto:steveayre at gmail.com] 
Sent: Monday, July 25, 2011 3:40 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Can't get variable set in dial plan

 

Variables are not fields. Fields are a specific set of attributes of the
call... you can however check variables in a condition, it just has a
slightly different syntax. Try this.

<extension name="public_did">

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

        <action inline="true" application="set" data="hours=1"/>

    </condition>

    <condition field="${hours}" expression="^1$">

        <action application="answer"/>

        <action application="ivr" data="GAC-BH"/>

        <action application="hangup"/>

        <anti-action application="answer"/>

        <anti-action application="ivr" data="GAC-BH"/>

        <anti-action application="hangup"/>

    </condition>

  </extension>

 
Also note I changed your regex of "1" to "^1$". The first matches any
string containing a digit 1 including 10, 21, 100, 200351. The latter
matches the start (^) and end ($) of the string so will only match the
exact string of 1.

-Steve

On 25 July 2011 20:54, Joshua Foshee <Joshua.Foshee at logixcom.com> wrote:

I am trying to set a variable in the dial plan and use it on a
condition. I have pushed the variable setting as close to the condition
for testing and this is what I get. Why does it not see the variable
set?

 

<extension name="public_did">

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

        <action inline="true" application="set" data="hours=1"/>

    </condition>

    <condition field="hours" expression="1">

        <action application="answer"/>

        <action application="ivr" data="GAC-BH"/>

        <action application="hangup"/>

        <anti-action application="answer"/>

        <anti-action application="ivr" data="GAC-BH"/>

        <anti-action application="hangup"/>

    </condition>

  </extension>

 

Here is the console output

 

Dialplan: FreeTDM/1:7/405790xxx8 Regex (PASS) [public_did]
destination_number(405790xxx8) =~ /^(.*)$/ break=on-false

Dialplan: FreeTDM/1:7/405790xxx8 Action set(hours=1) INLINE

EXECUTE FreeTDM/1:7/405790xxx8 set(hours=1)

2011-07-25 12:05:25.191357 [DEBUG] mod_dptools.c:1063
FreeTDM/1:7/405790xxx8 SET [hours]=[1]

Dialplan: FreeTDM/1:7/405790xxx8 Regex (FAIL) [public_did] hours() =~
/1/ break=on-false

Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action answer()

Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action ivr(GAC-BH)

Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action hangup()

 

 

 


_______________________________________________
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 rmartinez at redvoiss.net  Tue Jul 26 02:38:57 2011
From: rmartinez at redvoiss.net (Ricardo Martinez)
Date: Mon, 25 Jul 2011 18:38:57 -0400
Subject: [Freeswitch-users] SDP Append
Message-ID: <50a0ed0896c1011707f782e2c2cc4c80@mail.gmail.com>

Hello.

I?m trying to append two distinct SDP fields in the outgoing call leg from
Freeswitch.  Right now I?m only capable to attach only one :



<extension name="add-annexb=no" continue="true">

        <condition field="${switch_r_sdp}" expression="/(.*)(a=fmtp:18
annexb=no)(.*)/s">

                <action application="export"
data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/>

        </condition>

</extension>



Because when I?m try to attach a second one, it rewrites the first one.



Thanks.

Ricardo Martinez.-
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From nsirugudi at gmail.com  Tue Jul 26 04:21:15 2011
From: nsirugudi at gmail.com (Narendra Sirugudi)
Date: Tue, 26 Jul 2011 05:51:15 +0530
Subject: [Freeswitch-users] NAT and Freeswitch
Message-ID: <CAP18h9osfM6+NUgJkJ7tNfz=mLQzcvcDR1t5k5VsoFXGwH66+w@mail.gmail.com>

__________________________
|                                            |
|________
____|_____                                            ___________
||              |                     |
|  74.1.1.1               74.1.1.2 |                   |
|| SIP UA  |---------------------| NAT
Server|------------------------------------------| Freeswitch  |
||________|10.10.10.10    |__________|         Public Network
|___________|
|                                             |                    74.1.1.0
|    Private Network 10.10.10.0   |
|__________________________|

Hi,

In the above topology, the typical issue one faces is that the SDP(rtp)
IP&port in SIP invite sent from the SIP UA to Freeswitch will contain the
private Ip address of the SIP UA.

Assuming source NAT, the typical solution is to use a STUN server  in the
public network, which will help the SIP UA to discover the public IP&port
(for its private IP&port). The UA then sends the Invite with the SDP(RTP)
IP&port set to the discovered public IP&port. (I believe Freeswitch does not
have a STUN server within itself.)

But suppose that we do not want to use the STUN solution. Does Freeswitch
itself offer any solutions in this case ?

thanks,
--naren
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From gcd at i.ph  Tue Jul 26 04:28:04 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 26 Jul 2011 08:28:04 +0800
Subject: [Freeswitch-users] rxfax on b-leg
In-Reply-To: <4851d0b4a640bf82e068c5d6cc6079ed@nat-tele.com>
References: <6c1cf419782ba6781928a095d3bc19bc@nat-tele.com>
	<4E2ACE39.2080007@coppice.org>
	<4851d0b4a640bf82e068c5d6cc6079ed@nat-tele.com>
Message-ID: <CAHKqMb5QrmWiTMSVu=KFs__bLqNO_5P-boFpqBRh8_yx7Gw0VQ@mail.gmail.com>

re pressing *1. it's the caller (A-party) who performs this. i agree tone
detection would be a better feature. however, having *1 as backup is also
nice.

On Mon, Jul 25, 2011 at 3:58 PM, <slash at nat-tele.com> wrote:

>  On Sat, 23 Jul 2011 21:35:53 +0800, Steve Underwood
>  <steveu at coppice.org> wrote:
> > On 07/22/2011 12:56 AM, slash at nat-tele.com wrote:
> >>   Hello!
> >>
> >>
> >>   I wonder if there is a way to make rxfax recieve fax on outgoing
> >> call?
> >>   The idea is that registered FS user dials out, than tone_detect is
> >> used
> >>   to detect fax tone and start rxfax. The difference is that rxfax
> >> should
> >>   be run on B-leg, instead of A-leg in incoming fax call scenario.
> >>
> >>   I've tried to make a test config with outgoing call, rxfax&
> >>   tone_detect, but when I dial out and remote user starts sending
> >> fax to
> >>   me, FS detects a tone correctly, starts rxfax, but on my leg
> >> (A-leg, i
> >>   hear fax tones) and B-leg (with real fax machine) is hanged up.
> >>
> >>   Are there are solutions to explain somehow to rxfax that it should
> >> be
> >>   run on B-leg, not on A-leg?
> >>
> >>   Thanks in advance!
> >>
> >>   With best regards,
> >>   Kirill.
> > If you use rxfax at the outgoing end of a call, mod_spandsp expects
> > to
> > get that FAX by the polling procedure (i.e. the one you use when you
> > dial into a FAX server to collect an information FAX from it). If you
> > want to freely act as a simple FAX transmitter or receiver in the
> > middle
> > of a call, the current module code may not properly allow for that.
> > As
> > someone else said, there are tricks related to call transfers which
> > can
> > be used to coax the fax instance into caller or answerer mode.
> > Spandsp
> > does pretty much everything the FAX spec allows. If there is demand
> > we
> > could provide access to more of that functionality.
> >
> > Steve
> >
>
>  The idea with manual call transfer is good, but it's hard to remember
>  that every time you should ask remote end to press *1 before starting
>  fax. It's much more
>  comfortable for everybody to be able use rxfax as a simple FAX
>  transmitter or
>  receiver in the middle of a call... There is a demand for that
>  functionality : )
>
>  With best regards,
>  Kirill.
>
>
> _______________________________________________
> 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
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>
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From msc at freeswitch.org  Tue Jul 26 04:32:18 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 25 Jul 2011 17:32:18 -0700
Subject: [Freeswitch-users] sipx conference API
In-Reply-To: <32129940.post@talk.nabble.com>
References: <32129940.post@talk.nabble.com>
Message-ID: <CAKzWOxUTciAHpvMPjatStUF8s5u3jEm9ueqJXJmJoKKEBnGU5Q@mail.gmail.com>

Is this a FreeSWITCH question or a SIPx question?
-MC

On Mon, Jul 25, 2011 at 1:49 AM, Jayakrishnanu <ajaysharma842000 at yahoo.co.in
> wrote:

>
> Hi Folks
> Is there any way to add a person in to existing conference?
>
> > I configured sipx and I successfully made calls through call controller
> > API
> > but when I tried to implement conference web service API I could get the
> > response from the argument xml_list,volume_in,kick,lock but when I tried
> > to dial a number in to the conference using "curl --digest -k -X PUT
> > https://userid:Password at hostname:8085/conference/conference-
> > name/dial/XXXXX/dial-number".I couldnt get what is the
> > <endpoint_module_name> for the same.
>
> Can anybody give me the correct url for dial service in conference and how
> can I figure what is the endpoint-name in my system?
> --
> View this message in context:
> http://old.nabble.com/sipx-conference-API-tp32129940p32129940.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> 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
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>
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From nsirugudi at gmail.com  Tue Jul 26 04:55:41 2011
From: nsirugudi at gmail.com (Narendra Sirugudi)
Date: Tue, 26 Jul 2011 06:25:41 +0530
Subject: [Freeswitch-users] NAT and Freeswitch
In-Reply-To: <CAP18h9osfM6+NUgJkJ7tNfz=mLQzcvcDR1t5k5VsoFXGwH66+w@mail.gmail.com>
References: <CAP18h9osfM6+NUgJkJ7tNfz=mLQzcvcDR1t5k5VsoFXGwH66+w@mail.gmail.com>
Message-ID: <CAP18h9qrTsTo2mUKMuXY0n8RawGbN_5CZWayCOfV=6F2BbuQyg@mail.gmail.com>

Just in case my diagram got bungled:
The use case is:  Freeswitch in the public network, with a SIP UA in a
private network talking to Freeswitch across a NAT Server.
--naren
On Tue, Jul 26, 2011 at 5:51 AM, Narendra Sirugudi <nsirugudi at gmail.com>wrote:

> __________________________
> |                                            |
> |________
> ____|_____                                            ___________
> ||              |                     |
> |  74.1.1.1               74.1.1.2 |                   |
> || SIP UA  |---------------------| NAT
> Server|------------------------------------------| Freeswitch  |
> ||________|10.10.10.10    |__________|         Public Network
> |___________|
> |                                             |                    74.1.1.0
> |    Private Network 10.10.10.0   |
> |__________________________|
>
> Hi,
>
> In the above topology, the typical issue one faces is that the SDP(rtp)
> IP&port in SIP invite sent from the SIP UA to Freeswitch will contain the
> private Ip address of the SIP UA.
>
> Assuming source NAT, the typical solution is to use a STUN server  in the
> public network, which will help the SIP UA to discover the public IP&port
> (for its private IP&port). The UA then sends the Invite with the SDP(RTP)
> IP&port set to the discovered public IP&port. (I believe Freeswitch does not
> have a STUN server within itself.)
>
> But suppose that we do not want to use the STUN solution. Does Freeswitch
> itself offer any solutions in this case ?
>
> thanks,
> --naren
>
>
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From curriegrad2004 at gmail.com  Tue Jul 26 05:39:56 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 25 Jul 2011 18:39:56 -0700
Subject: [Freeswitch-users] NAT and Freeswitch
In-Reply-To: <CAP18h9qrTsTo2mUKMuXY0n8RawGbN_5CZWayCOfV=6F2BbuQyg@mail.gmail.com>
References: <CAP18h9osfM6+NUgJkJ7tNfz=mLQzcvcDR1t5k5VsoFXGwH66+w@mail.gmail.com>
	<CAP18h9qrTsTo2mUKMuXY0n8RawGbN_5CZWayCOfV=6F2BbuQyg@mail.gmail.com>
Message-ID: <CAMWiMy-oXiqct_HBak8k7qQnHcdwcdM1kJKxb5UjxMZErftpvQ@mail.gmail.com>

The SIP UA only needs to be configured to use NAT in this case. The FS
Server does not need any configuration in this case.

On Mon, Jul 25, 2011 at 5:55 PM, Narendra Sirugudi <nsirugudi at gmail.com> wrote:
> Just in case my diagram got bungled:
> The use case is:? Freeswitch in the public network, with a SIP UA in a
> private network talking to Freeswitch across a NAT Server.
> --naren
> On Tue, Jul 26, 2011 at 5:51 AM, Narendra Sirugudi <nsirugudi at gmail.com>
> wrote:
>>
>> __________________________
>> |??????????????????????????????????????????? |
>> |________
>> ____|_____??????????????????????????????????????????? ___________
>> ||????????????? |???????????????????? |
>> |??74.1.1.1????????????? ?74.1.1.2 |???????????????????|
>> || SIP UA? |---------------------| NAT
>> Server|------------------------------------------| Freeswitch? |
>> ||________|10.10.10.10????|__________|???????? Public Network
>> |___________|
>> |?????????????????????????????????????????????|
>> 74.1.1.0
>> |??? Private Network 10.10.10.0?? |
>> |__________________________|
>>
>> Hi,
>>
>> In the above topology, the typical issue one faces is that the SDP(rtp)
>> IP&port in SIP invite sent from the SIP UA to Freeswitch will contain the
>> private Ip address of the SIP UA.
>>
>> Assuming source NAT, the typical solution is to use a STUN server? in the
>> public network, which will help the SIP UA to discover the public IP&port
>> (for its private IP&port). The UA then sends the Invite with the SDP(RTP)
>> IP&port set to the discovered public IP&port. (I believe Freeswitch does not
>> have a STUN server within itself.)
>>
>> But suppose that we do not want to use the STUN solution. Does Freeswitch
>> itself offer any solutions in this case ?
>>
>> thanks,
>> --naren
>>
>
> _______________________________________________
> 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 curriegrad2004 at gmail.com  Tue Jul 26 05:43:06 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 25 Jul 2011 18:43:06 -0700
Subject: [Freeswitch-users] NAT and Freeswitch
In-Reply-To: <CAMWiMy-oXiqct_HBak8k7qQnHcdwcdM1kJKxb5UjxMZErftpvQ@mail.gmail.com>
References: <CAP18h9osfM6+NUgJkJ7tNfz=mLQzcvcDR1t5k5VsoFXGwH66+w@mail.gmail.com>
	<CAP18h9qrTsTo2mUKMuXY0n8RawGbN_5CZWayCOfV=6F2BbuQyg@mail.gmail.com>
	<CAMWiMy-oXiqct_HBak8k7qQnHcdwcdM1kJKxb5UjxMZErftpvQ@mail.gmail.com>
Message-ID: <CAMWiMy9VaipVy1mMQJJ7dy5rU_ivodV+ySvrt1k=E9c_OBrXCg@mail.gmail.com>

I also forgot to mention that that the nf_nat_sip module in Linux
replaces the STUN's server role pretty well

On Mon, Jul 25, 2011 at 6:39 PM, curriegrad2004
<curriegrad2004 at gmail.com> wrote:
> The SIP UA only needs to be configured to use NAT in this case. The FS
> Server does not need any configuration in this case.
>
> On Mon, Jul 25, 2011 at 5:55 PM, Narendra Sirugudi <nsirugudi at gmail.com> wrote:
>> Just in case my diagram got bungled:
>> The use case is:? Freeswitch in the public network, with a SIP UA in a
>> private network talking to Freeswitch across a NAT Server.
>> --naren
>> On Tue, Jul 26, 2011 at 5:51 AM, Narendra Sirugudi <nsirugudi at gmail.com>
>> wrote:
>>>
>>> __________________________
>>> |??????????????????????????????????????????? |
>>> |________
>>> ____|_____??????????????????????????????????????????? ___________
>>> ||????????????? |???????????????????? |
>>> |??74.1.1.1????????????? ?74.1.1.2 |???????????????????|
>>> || SIP UA? |---------------------| NAT
>>> Server|------------------------------------------| Freeswitch? |
>>> ||________|10.10.10.10????|__________|???????? Public Network
>>> |___________|
>>> |?????????????????????????????????????????????|
>>> 74.1.1.0
>>> |??? Private Network 10.10.10.0?? |
>>> |__________________________|
>>>
>>> Hi,
>>>
>>> In the above topology, the typical issue one faces is that the SDP(rtp)
>>> IP&port in SIP invite sent from the SIP UA to Freeswitch will contain the
>>> private Ip address of the SIP UA.
>>>
>>> Assuming source NAT, the typical solution is to use a STUN server? in the
>>> public network, which will help the SIP UA to discover the public IP&port
>>> (for its private IP&port). The UA then sends the Invite with the SDP(RTP)
>>> IP&port set to the discovered public IP&port. (I believe Freeswitch does not
>>> have a STUN server within itself.)
>>>
>>> But suppose that we do not want to use the STUN solution. Does Freeswitch
>>> itself offer any solutions in this case ?
>>>
>>> thanks,
>>> --naren
>>>
>>
>> _______________________________________________
>> 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 nsirugudi at gmail.com  Tue Jul 26 07:48:41 2011
From: nsirugudi at gmail.com (Narendra Sirugudi)
Date: Tue, 26 Jul 2011 09:18:41 +0530
Subject: [Freeswitch-users] NAT and Freeswitch
In-Reply-To: <CAMWiMy-oXiqct_HBak8k7qQnHcdwcdM1kJKxb5UjxMZErftpvQ@mail.gmail.com>
References: <CAP18h9osfM6+NUgJkJ7tNfz=mLQzcvcDR1t5k5VsoFXGwH66+w@mail.gmail.com>
	<CAP18h9qrTsTo2mUKMuXY0n8RawGbN_5CZWayCOfV=6F2BbuQyg@mail.gmail.com>
	<CAMWiMy-oXiqct_HBak8k7qQnHcdwcdM1kJKxb5UjxMZErftpvQ@mail.gmail.com>
Message-ID: <CAP18h9pxFEmsRZTrO7VpcLVvo8OuwDjra67cn551y22iMWdJiw@mail.gmail.com>

Hi,
Thanks for your reply.

Maybe my problem definition is not fully clear:
For my use case, having addtional configuration on the SIP UA (softphone)
side is not feasible. I am looking for solution on the Freeswitch
side(sitting on the public network). Not all SIP UA clients (softphones)
will have the solution of working around NAT, let alone STUN client support.

I observed the Freeswitch has some features like AutoNAT etc. Aren't these
of any help for the use case i have ? I just started reading on the AutoNAT
feature. The documentation of AutoNAT seems to suggest that freeswitch
itself sits behind a NAT. This is not the same as my use case.

To summarize: Does freeswitch have any method of detecting that the remote
SIP UA is behind a NAT and taking some action based on that ?
Any suggestions are welcome.

thanks,
--naren
On Tue, Jul 26, 2011 at 7:09 AM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> The SIP UA only needs to be configured to use NAT in this case. The FS
> Server does not need any configuration in this case.
>
>
>
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From curriegrad2004 at gmail.com  Tue Jul 26 09:23:19 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 25 Jul 2011 22:23:19 -0700
Subject: [Freeswitch-users] NAT and Freeswitch
In-Reply-To: <CAP18h9pxFEmsRZTrO7VpcLVvo8OuwDjra67cn551y22iMWdJiw@mail.gmail.com>
References: <CAP18h9osfM6+NUgJkJ7tNfz=mLQzcvcDR1t5k5VsoFXGwH66+w@mail.gmail.com>
	<CAP18h9qrTsTo2mUKMuXY0n8RawGbN_5CZWayCOfV=6F2BbuQyg@mail.gmail.com>
	<CAMWiMy-oXiqct_HBak8k7qQnHcdwcdM1kJKxb5UjxMZErftpvQ@mail.gmail.com>
	<CAP18h9pxFEmsRZTrO7VpcLVvo8OuwDjra67cn551y22iMWdJiw@mail.gmail.com>
Message-ID: <CAMWiMy_5OTymJZ5Zf-BeefQ_vOU9Ot_ZbTSdd16yx9OJWSk3Dw@mail.gmail.com>

In this case there's nothing I can think of that FreeSWITCH can do in
this case if the remote endpoint doesn't support NAT at all.

You may want to try the nf_nat_sip module at the remote end, but it
may not be the solution you're looking for

On Mon, Jul 25, 2011 at 8:48 PM, Narendra Sirugudi <nsirugudi at gmail.com> wrote:
>
> Hi,
> Thanks for your reply.
>
> Maybe my problem definition is not fully clear:
> For my use case, having addtional configuration on the SIP UA (softphone)
> side is not feasible. I am looking for solution on the Freeswitch
> side(sitting on the public network). Not all SIP UA clients?(softphones)
> will have the solution of working around NAT, let alone STUN client support.
>
> I observed the Freeswitch has some features like AutoNAT etc. Aren't these
> of any help for the use case i have ? I just started reading on the AutoNAT
> feature. The documentation of AutoNAT seems to suggest that freeswitch
> itself sits behind a NAT. This is not the same as my use case.
>
> To summarize: Does freeswitch have any method of detecting that the remote
> SIP UA is behind a NAT and taking some action based on that ?
> Any suggestions are welcome.
>
> thanks,
> --naren
> On Tue, Jul 26, 2011 at 7:09 AM, curriegrad2004 <curriegrad2004 at gmail.com>
> wrote:
>>
>> The SIP UA only needs to be configured to use NAT in this case. The FS
>> Server does not need any configuration in this case.
>>
>>
>
> _______________________________________________
> 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 sanja_angelova at hotmail.com  Tue Jul 26 13:13:08 2011
From: sanja_angelova at hotmail.com (Sanja Angelova)
Date: Tue, 26 Jul 2011 11:13:08 +0200
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <1311619951536-6619518.post@n2.nabble.com>
References: <DUB109-W4708BC652AAED846BBF33AFE330@phx.gbl>,
	<1311619951536-6619518.post@n2.nabble.com>
Message-ID: <DUB109-W39D61CBC6141AE40CDC19FFE320@phx.gbl>


This is the result of debug:

2011-07-26 10:59:53.419631 [NOTICE] switch_channel.c:816 New Channel sofia/internal/1002 at 192.168.107.40 [0dfc0f77-b8f8-4146-82ad-42258dbc9526]
2011-07-26 10:59:53.419631 [INFO] mod_dialplan_xml.c:331 Processing John <1002>->654321 in context default
2011-07-26 10:59:53.439632 [NOTICE] switch_ivr.c:1613 Transfer sofia/internal/1002 at 192.168.107.40 to enum[654321 at default]
2011-07-26 10:59:53.439632 [NOTICE] switch_channel.c:816 New Channel sofia/internal/1005 at 192.168.107.17 [9cc08e20-0711-44c5-a026-d19c39e35e52]
2011-07-26 10:59:53.439632 [NOTICE] sofia.c:5416 Hangup sofia/internal/1005 at 192.168.107.17 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2011-07-26 10:59:53.439632 [INFO] mod_dptools.c:2647 Originate Failed.  Cause: NORMAL_TEMPORARY_FAILURE
2011-07-26 10:59:53.439632 [NOTICE] mod_dptools.c:2761 Hangup sofia/internal/1002 at 192.168.107.40 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
2011-07-26 10:59:53.459804 [WARNING] sofia_reg.c:1241 SIP auth challenge (INVITE) on sofia profile 'internal' for [654321 at 192.168.107.40] from ip 192.168.107.40
2011-07-26 10:59:53.459804 [NOTICE] switch_core_session.c:1304 Session 11 (sofia/internal/1002 at 192.168.107.40) Ended
2011-07-26 10:59:53.459804 [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/1002 at 192.168.107.40 [CS_DESTROY]
2011-07-26 10:59:53.459804 [NOTICE] switch_channel.c:816 New Channel sofia/internal/1002 at 192.168.107.40 [fc7c637b-442e-484f-b7e2-50968cd3c5b3]
2011-07-26 10:59:53.459804 [NOTICE] switch_core_session.c:1304 Session 12 (sofia/internal/1005 at 192.168.107.17) Ended
2011-07-26 10:59:53.459804 [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/1005 at 192.168.107.17 [CS_DESTROY]
2011-07-26 10:59:53.459804 [INFO] mod_dialplan_xml.c:331 Processing John <1002>->654321 in context default
2011-07-26 10:59:53.480469 [NOTICE] switch_ivr.c:1613 Transfer sofia/internal/1002 at 192.168.107.40 to enum[654321 at default]
2011-07-26 10:59:53.499658 [NOTICE] switch_channel.c:816 New Channel sofia/internal/1005 at 192.168.107.17 [5987db35-d31e-44d7-b59a-fbc5b77e0e75]
2011-07-26 10:59:54.500195 [NOTICE] sofia.c:5416 Hangup sofia/internal/1005 at 192.168.107.17 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2011-07-26 10:59:54.500195 [NOTICE] switch_core_session.c:1304 Session 14 (sofia/internal/1005 at 192.168.107.17) Ended
2011-07-26 10:59:54.500195 [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/1005 at 192.168.107.17 [CS_DESTROY]
2011-07-26 10:59:54.500195 [INFO] mod_dptools.c:2647 Originate Failed.  Cause: NORMAL_TEMPORARY_FAILURE
2011-07-26 10:59:54.500195 [NOTICE] mod_dptools.c:2761 Hangup sofia/internal/1002 at 192.168.107.40 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
2011-07-26 10:59:54.548796 [NOTICE] switch_core_session.c:1304 Session 13 (sofia/internal/1002 at 192.168.107.40) Ended
2011-07-26 10:59:54.548796 [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/1002 at 192.168.107.40 [CS_DESTROY]



> Date: Mon, 25 Jul 2011 11:52:31 -0700
> From: jeff at jefflenk.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] (no subject)
> 
> go ahead and attach a small debug log here and perhaps someone will have a
> look
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/no-subject-tp6618626p6619518.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> 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 acrow at integrafin.co.uk  Tue Jul 26 16:09:40 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Tue, 26 Jul 2011 13:09:40 +0100
Subject: [Freeswitch-users] Intercept taking too long
Message-ID: <4E2EAE84.6080407@integrafin.co.uk>

Hi,

I have set up a system with 6 snom 370s and FS and I'm seeing a long 
delay (up to 3 seconds) when trying to do a pickup using the BLF.

FS receives the **<dest> as soon as the button is pressed but it seems 
the delay is in tearing down the call to the original destination. This 
is much longer than it took on an Elastix install with 6 Grandstream 
GXP2000s.

Is there anything I can do to speed this up? It's causing some confusion 
for our users as they are pressing the pickup button and then start 
greeting the caller straight away, not realising it's going to take 
another couple of seconds for them to arrive.

Cheers

Alex

-- 
This message is intended only for the addressee and may contain
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the message along with any attachments and notify us immediately.

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From frank at rosengart.de  Tue Jul 26 18:07:08 2011
From: frank at rosengart.de (Frank Rosengart)
Date: Tue, 26 Jul 2011 16:07:08 +0200
Subject: [Freeswitch-users] Intercept taking too long
In-Reply-To: <4E2EAE84.6080407@integrafin.co.uk>
References: <4E2EAE84.6080407@integrafin.co.uk>
Message-ID: <4E2ECA0C.9080104@rosengart.de>

On 07/26/2011 02:09 PM, Alex Crow wrote:

> I have set up a system with 6 snom 370s and FS and I'm seeing a long 
> delay (up to 3 seconds) when trying to do a pickup using the BLF.

We are using Snom 360 with 7.3.30, FS from git around mid of june. The
manual pickup with **<extension> is prompt, no delay.

Do you have the delay an manual pickup, too?

How does your uuid-to-pickup database lookup look like?



Frank



PS: I'm still stuck with the BLF thing: busy (LED on) works, but not
blinking (monitored ext is ringing) - there is no notify sent to the phones.


From adavidm at gmail.com  Tue Jul 26 18:27:38 2011
From: adavidm at gmail.com (David Martin)
Date: Tue, 26 Jul 2011 15:27:38 +0100
Subject: [Freeswitch-users] Cisco CUCM 6.1 and Freeswitch SIP trunk
In-Reply-To: <DC79FD7E-539B-43D7-89B1-33B6304607EB@gmail.com>
References: <CA+ViVc99bqe2e8AAy-ROxLA_LPumQv7q50ygPNQjf_KMzypKAg@mail.gmail.com>
	<CAKDfjgd0R0Sb_BYpmyHLjmv-oPajQ_uw_5ev2ioPDrXXM_3qpw@mail.gmail.com>
	<80E395DA-B3F2-42CE-AB9F-098096B32B45@gmail.com>
	<CAD-qRkjh2-St09KcqbMEA=muQw5tzHckV=TVVMRdFHLYqLr6OQ@mail.gmail.com>
	<CA+ViVc9ydZfjWwTVVkDxRugAC+s4iV9TPRLEU3v=h7emfnf98Q@mail.gmail.com>
	<CAKDfjgfh2162ysaDc3-91-qNR4wV9R5K-Ds-OtSYOXdgiDZD9w@mail.gmail.com>
	<CA+ViVc_U+iEifVcwb=n=keR4EO8rTV-SGFEu7XO0q=NSzykGqw@mail.gmail.com>
	<CA+ViVc-oY3ZV6AYiDkHeSVJucjm=pLnha0=gW4P2A91Ux=TUSQ@mail.gmail.com>
	<CAKDfjgczmND40J-ygWUCsJ64sSb+USenco87MFmm9mMTRGmRNg@mail.gmail.com>
	<FAAAC9F6-ACCE-4A17-BCEE-EED2FE2DAE1A@gmail.com>
	<CAKDfjgcGU7ea4HmC726gHQu2R+jTW6G=u8_Shqyv30L=NuriOA@mail.gmail.com>
	<DC79FD7E-539B-43D7-89B1-33B6304607EB@gmail.com>
Message-ID: <CA+ViVc8osJSbF2C=egNpBjiVWkewhowGRMSggZpwGTykWKBaTg@mail.gmail.com>

FIXED!

the software MTP i was using was broken somehow. Have changed to another
(identical) MTP resource and all is well.

Thanks for all your help Kristian, much appreciated!

Just got to learn how to use Freeswitch now!

Regards

David



On 22 July 2011 23:23, David Martin <adavidm at gmail.com> wrote:

> Kristian,
>
> I'm starting to see some of that pain already! It's not wasted effort
> though as I have removed the ubuntu packages and built freeswitch from git
> so I can more easily try out different modules and versions of software.
> Getting mod_h323 or mod_opal to compile is proving more tricky but I can
> tinker with that at the same time as pursuing the SIP route, which I would
> definitely prefer.
>
> I will take your advice and post the question to the lists. I'm also going
> to raise a call with our cisco partner, we pay them enough money for support
> so they can earn it for a change. I will let the list know what I find out
> and if I get it fixed I will write something up.
>
> Thanks yet again
>
> David
>
> Sent from my iPad
>
> On 22 Jul 2011, at 14:42, Kristian Kielhofner <kris at kriskinc.com> wrote:
>
> > David,
> >
> >  I'd hate to see you have to go that route (it carries its own
> > headaches).  You should try sticking with SIP.
> >
> >  This is really more of a Cisco question.  Have you tried asking on
> >
> > http://puck.nether.net/mailman/listinfo/cisco-nsp
> >
> > -or-
> >
> > http://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >  Even Voiceops may have some clues:
> >
> > http://puck.nether.net/mailman/listinfo/voiceops
> >
> >  Of course if you find the answer here it would be nice to have you
> > come back and let us know what you needed to do to CUCM to get it to
> > act sane...
> >
> > On Thu, Jul 21, 2011 at 6:28 PM, David Martin <adavidm at gmail.com> wrote:
> >> Thanks for your assistance. I might try mod_opal and use h323 instead
> then.
> >>
> >> Thanks again.
> >>
> >> David
> >>
> >> Sent from my iPad
> >>
> >> On 21 Jul 2011, at 20:44, Kristian Kielhofner <kris at kriskinc.com>
> wrote:
> >>
> >>> The issue now is CUCM never sends an SDP.  Not in the INVITE, not in
> >>> the ACK.  If the INVITE contained an SDP you'd be fine with the
> >>> default configuration.  If it didn't but the ACK did you'd be fine
> >>> with 3pcc enabled.
> >>>
> >>> Right now CUCM isn't providing either.  That's your problem, it's
> >>> somewhere on the Cisco side, and I've never seen it before.
> >>> Unfortunately I don't have any specific recommendations for you.
> >>>
> >>> On Thu, Jul 21, 2011 at 8:46 AM, David Martin <adavidm at gmail.com>
> wrote:
> >>>> For completeness, here is the INVITE with "Disable Early Media"
> unchecked:
> >>>>
>  ------------------------------------------------------------------------
> >>>>    INVITE sip:65131111 at 10.201.27.20:5060 SIP/2.0
> >>>>    Date: Thu, 21 Jul 2011 12:42:02 GMT
> >>>>    Call-Info:
> >>>> <sip:10.200.170.3:5060
> >;method="NOTIFY;Event=telephone-event;Duration=500"
> >>>>    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE,
> REFER,
> >>>> SUBSCRIBE, NOTIFY, PUBLISH
> >>>>    From: "Ryan Girdlestone"
> >>>> <sip:64105554 at 10.200.170.3
> >;tag=90509204-400f-4cb6-9be7-cb5410a83a11-54376643
> >>>>    Allow-Events: presence, kpml
> >>>>    Supported: timer,replaces
> >>>>    Min-SE:  1800
> >>>>    Remote-Party-ID: "Ryan Girdlestone"
> >>>> <sip:64105554 at 10.200.170.3>;party=calling;screen=yes;privacy=off
> >>>>    Content-Length: 0
> >>>>    User-Agent: Cisco-CUCM6.1
> >>>>    To: <sip:65131111 at 10.201.27.20>
> >>>>    Contact: <sip:64105554 at 10.200.170.3:5060>
> >>>>    Expires: 180
> >>>>    Call-ID: d3b6ce80-e2811e9a-33-3aac80a at 10.200.170.3
> >>>>    Via: SIP/2.0/UDP 10.200.170.3:5060;branch=z9hG4bK4d55790bf1
> >>>>    CSeq: 101 INVITE
> >>>>    Session-Expires:  1800
> >>>>    Max-Forwards: 70
> >>>>
> >>>>
> >>>>
> >>>> On 21 July 2011 13:10, David Martin <adavidm at gmail.com> wrote:
> >>>>>
> >>>>> Kristian,
> >>>>> Here are some screenshots from the trunk config.
> >>>>> http://img200.imageshack.us/img200/9405/siptrunk.png
> >>>>> and for the sip profile
> >>>>> http://img191.imageshack.us/img191/3863/sipprofile.png
> >>>>> I have tried the early media option both ways with the same result.
> Is the
> >>>>> MTP payload type correct?
> >>>>> Appreciate your help on this.
> >>>>> David
> >>>>>
> >>>>> On 21 July 2011 11:13, Kristian Kielhofner <kris at kriskinc.com>
> wrote:
> >>>>>>
> >>>>>> David,
> >>>>>>
> >>>>>>  The problem here is that even with 3pcc enabled the ACK from
> >>>>>> CallManager doesn't have an SDP.  CallManager never provides an SDP
> >>>>>> during this entire trace.  Not in the INVITE (preferred) and not in
> >>>>>> the ACK (as I already mentioned).
> >>>>>>
> >>>>>>  This is odd, even for CallManager.  Are you sure you have MTP
> enabled:
> >>>>>>
> >>>>>> http://blog.krisk.org/2010/05/another-sip-gotcha-cisco.html
> >>>>>>
> >>>>>> On Thu, Jul 21, 2011 at 3:45 AM, David Martin <adavidm at gmail.com>
> wrote:
> >>>>>>> Gotcha, not sure how I missed the INVITE the first time, but here
> is a
> >>>>>>> new
> >>>>>>> pastebin that should have everything:
> >>>>>>> http://pastebin.com/T3eqms8N
> >>>>>>> thanks in advance once again.
> >>>>>>> David
> >>>>>>>
> >>>>>>>
> >>>>>>> On 21 July 2011 02:19, Christopher Rienzo <cmrienzo at gmail.com>
> wrote:
> >>>>>>>>
> >>>>>>>> Line 7 is the 200 OK response to the INVITE.
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> On Wed, Jul 20, 2011 at 6:54 PM, David Martin <adavidm at gmail.com>
> >>>>>>>> wrote:
> >>>>>>>>>
> >>>>>>>>> Forgive my confusion, I'm new to freeswitch. Does the invite not
> >>>>>>>>> start at
> >>>>>>>>> line 7 on the pastebin?
> >>>>>>>>>
> >>>>>>>>> David
> >>>>>>>>>
> >>>>>>>>> Sent from my iPhone
> >>>>>>>>>
> >>>>>>>>> On 20 Jul 2011, at 07:22 PM, Kristian Kielhofner <
> kris at kriskinc.com>
> >>>>>>>>> wrote:
> >>>>>>>>>
> >>>>>>>>>> That trace is missing the INVITE that starts the transaction.
> >>>>>>>>>>
> >>>>>>>>>> On Wed, Jul 20, 2011 at 11:48 AM, David Martin <
> adavidm at gmail.com>
> >>>>>>>>>> wrote:
> >>>>>>>>>>> All,
> >>>>>>>>>>> Firstly apologies if this is not the correct place to ask, but
> I
> >>>>>>>>>>> am
> >>>>>>>>>>> having
> >>>>>>>>>>> problems getting CUCM 6.1 to talk with freeswitch over a SIP
> trunk
> >>>>>>>>>>> (I
> >>>>>>>>>>> only
> >>>>>>>>>>> need inbound from Cisco -> Freeswitch). I have enabled 3pcc,
> but
> >>>>>>>>>>> am
> >>>>>>>>>>> getting
> >>>>>>>>>>> the following sofia trace:
> >>>>>>>>>>> http://pastebin.com/g9VmvxFe
> >>>>>>>>>>> Can anyone point me in the right direction? I have also tried
> >>>>>>>>>>> enabling
> >>>>>>>>>>> MTP
> >>>>>>>>>>> on the Cisco side, but this does not make any difference.
> >>>>>>>>>>> Thanks in advance.
> >>>>>>>>>>> David
> >>>>>>>>>>> _______________________________________________
> >>>>>>>>>>> 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
> >>>>>>>>>>>
> >>>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>> --
> >>>>>>>>>> Kristian Kielhofner
> >>>>>>>>>>
> >>>>>>>>>> _______________________________________________
> >>>>>>>>>> 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
> >>>>>>>>>
> >>>>>>>>> _______________________________________________
> >>>>>>>>> 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
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> _______________________________________________
> >>>>>>>> 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
> >>>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>> _______________________________________________
> >>>>>>> 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
> >>>>>>>
> >>>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> --
> >>>>>> Kristian Kielhofner
> >>>>>>
> >>>>>> _______________________________________________
> >>>>>> 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
> >>>>>
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> 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
> >>>>
> >>>>
> >>>
> >>>
> >>>
> >>> --
> >>> Kristian Kielhofner
> >>>
> >>> _______________________________________________
> >>> 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
> >>
> >> _______________________________________________
> >> 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
> >>
> >
> >
> >
> > --
> > Kristian Kielhofner
> >
> > _______________________________________________
> > 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 kris at kriskinc.com  Tue Jul 26 18:53:04 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 26 Jul 2011 10:53:04 -0400
Subject: [Freeswitch-users] Cisco CUCM 6.1 and Freeswitch SIP trunk
In-Reply-To: <CA+ViVc8osJSbF2C=egNpBjiVWkewhowGRMSggZpwGTykWKBaTg@mail.gmail.com>
References: <CA+ViVc99bqe2e8AAy-ROxLA_LPumQv7q50ygPNQjf_KMzypKAg@mail.gmail.com>
	<CAKDfjgd0R0Sb_BYpmyHLjmv-oPajQ_uw_5ev2ioPDrXXM_3qpw@mail.gmail.com>
	<80E395DA-B3F2-42CE-AB9F-098096B32B45@gmail.com>
	<CAD-qRkjh2-St09KcqbMEA=muQw5tzHckV=TVVMRdFHLYqLr6OQ@mail.gmail.com>
	<CA+ViVc9ydZfjWwTVVkDxRugAC+s4iV9TPRLEU3v=h7emfnf98Q@mail.gmail.com>
	<CAKDfjgfh2162ysaDc3-91-qNR4wV9R5K-Ds-OtSYOXdgiDZD9w@mail.gmail.com>
	<CA+ViVc_U+iEifVcwb=n=keR4EO8rTV-SGFEu7XO0q=NSzykGqw@mail.gmail.com>
	<CA+ViVc-oY3ZV6AYiDkHeSVJucjm=pLnha0=gW4P2A91Ux=TUSQ@mail.gmail.com>
	<CAKDfjgczmND40J-ygWUCsJ64sSb+USenco87MFmm9mMTRGmRNg@mail.gmail.com>
	<FAAAC9F6-ACCE-4A17-BCEE-EED2FE2DAE1A@gmail.com>
	<CAKDfjgcGU7ea4HmC726gHQu2R+jTW6G=u8_Shqyv30L=NuriOA@mail.gmail.com>
	<DC79FD7E-539B-43D7-89B1-33B6304607EB@gmail.com>
	<CA+ViVc8osJSbF2C=egNpBjiVWkewhowGRMSggZpwGTykWKBaTg@mail.gmail.com>
Message-ID: <CAKDfjgdRxsf0WJCzEsOknCmWMnNQcOV8mrGRGTOXWF1EbRxQKg@mail.gmail.com>

Good to hear.

On Tue, Jul 26, 2011 at 10:27 AM, David Martin <adavidm at gmail.com> wrote:
> FIXED!
> the software MTP i was using was broken somehow. Have changed to another
> (identical) MTP resource and all is well.
> Thanks for all your help Kristian, much?appreciated!
> Just got to learn how to use Freeswitch now!
> Regards
> David

-- 
Kristian Kielhofner


From david.ponzone at ipeva.fr  Tue Jul 26 20:37:25 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 26 Jul 2011 18:37:25 +0200
Subject: [Freeswitch-users] NAT and Freeswitch
In-Reply-To: <CAMWiMy_5OTymJZ5Zf-BeefQ_vOU9Ot_ZbTSdd16yx9OJWSk3Dw@mail.gmail.com>
References: <CAP18h9osfM6+NUgJkJ7tNfz=mLQzcvcDR1t5k5VsoFXGwH66+w@mail.gmail.com>
	<CAP18h9qrTsTo2mUKMuXY0n8RawGbN_5CZWayCOfV=6F2BbuQyg@mail.gmail.com>
	<CAMWiMy-oXiqct_HBak8k7qQnHcdwcdM1kJKxb5UjxMZErftpvQ@mail.gmail.com>
	<CAP18h9pxFEmsRZTrO7VpcLVvo8OuwDjra67cn551y22iMWdJiw@mail.gmail.com>
	<CAMWiMy_5OTymJZ5Zf-BeefQ_vOU9Ot_ZbTSdd16yx9OJWSk3Dw@mail.gmail.com>
Message-ID: <B519BBC8-D4E7-4583-86D2-A82248E449E5@ipeva.fr>

More precisely, FreeSWITCH does not have any issues to handle a remote endpoint behind NAT if the endpoint correctly sets the rport parameter in the Via field (RFC3581).
In case the endpoint does not, FreeSWITCH can force rport, meaning it will answer to the endpoint as if rport was set.
This mechanism does not have anything to do with STUN/ICE or other external mechanism and it works perfectly.

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

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

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




Le 26/07/2011 ? 07:23, curriegrad2004 a ?crit :

> In this case there's nothing I can think of that FreeSWITCH can do in
> this case if the remote endpoint doesn't support NAT at all.
> 
> You may want to try the nf_nat_sip module at the remote end, but it
> may not be the solution you're looking for
> 
> On Mon, Jul 25, 2011 at 8:48 PM, Narendra Sirugudi <nsirugudi at gmail.com> wrote:
>> 
>> Hi,
>> Thanks for your reply.
>> 
>> Maybe my problem definition is not fully clear:
>> For my use case, having addtional configuration on the SIP UA (softphone)
>> side is not feasible. I am looking for solution on the Freeswitch
>> side(sitting on the public network). Not all SIP UA clients (softphones)
>> will have the solution of working around NAT, let alone STUN client support.
>> 
>> I observed the Freeswitch has some features like AutoNAT etc. Aren't these
>> of any help for the use case i have ? I just started reading on the AutoNAT
>> feature. The documentation of AutoNAT seems to suggest that freeswitch
>> itself sits behind a NAT. This is not the same as my use case.
>> 
>> To summarize: Does freeswitch have any method of detecting that the remote
>> SIP UA is behind a NAT and taking some action based on that ?
>> Any suggestions are welcome.
>> 
>> thanks,
>> --naren
>> On Tue, Jul 26, 2011 at 7:09 AM, curriegrad2004 <curriegrad2004 at gmail.com>
>> wrote:
>>> 
>>> The SIP UA only needs to be configured to use NAT in this case. The FS
>>> Server does not need any configuration in this case.
>>> 
>>> 
>> 
>> _______________________________________________
>> 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
>> 
>> 
> 
> _______________________________________________
> 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 alex at digitalmail.com  Tue Jul 26 15:50:20 2011
From: alex at digitalmail.com (Alex Lake)
Date: Tue, 26 Jul 2011 12:50:20 +0100
Subject: [Freeswitch-users] Combining AND and ELSE in freeswitch conditions
Message-ID: <4E2EA9FC.9060706@digitalmail.com>

Hello all, I'm a Freeswitch newbie struggling to get my head around the 
best way to do some TOD/BH routing.
I've got a LUA script that will work out whether or not we're in a bank 
holiday, but I wanted to do the combinatorial logic within Freeswitch 
dial plan.

Something like:

if (Weekday AND (NOT BankHoliday) AND InWorkingHours) {
     Transfer to mainIVR
} else {
     Transfer to voicemail
}

What is the most elegant way to achieve this?
The Wiki pages don't appear to explain how one can do ANDs with ELSE, as 
the anti-action will only get executed if it's only the last condition 
that fails.

Thanks!





From michael.knop at hcu-hamburg.de  Tue Jul 26 18:52:27 2011
From: michael.knop at hcu-hamburg.de (michael knop)
Date: Tue, 26 Jul 2011 16:52:27 +0200
Subject: [Freeswitch-users] FS to a Sonus SIP trunk
Message-ID: <4E2ED4AB.1030305@hcu-hamburg.de>

Hi all!

I?m trying to connect my FS to a Sonus SIP trunk. I followed the 
instruction at

   http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus

but it did not work. At the beginning of a call voice quality is good. 
After a while it changes to choppy.

I don?t know if it?s the same problem: When I call the Tetris extension 
via Sonus SIP trunk the sound is too fast and I?m getting log entries 
like the following one:

[...]
2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME 
not supported, changing our end from 20 to 10
2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from 
PCMA at 20ms@8000hz to PCMA at 10ms@8000hz
2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global 
timer resolution to 10ms to handle interval 10
2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer 
[soft] 80 bytes per 10ms
2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec 
sofia/external/+4940... at 193...:5060 PCMA/8000 10 ms 80 samples 64000 bits
2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write 
Buffer at 160 bytes to accommodate 320->160
[...]

This problem is fixed by adding the following line to 
conf/sip_profiles/external.xml:

<param name="rtp-autofix-timing" value="false"/>

Any hints?

/micha


From msc at freeswitch.org  Wed Jul 27 00:37:59 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 26 Jul 2011 13:37:59 -0700
Subject: [Freeswitch-users] Combining AND and ELSE in freeswitch
	conditions
In-Reply-To: <4E2EA9FC.9060706@digitalmail.com>
References: <4E2EA9FC.9060706@digitalmail.com>
Message-ID: <CAKzWOxUzhjf_x_Rg8FomNKav3V7-eS-yONQSDp9UFsJc29=bdA@mail.gmail.com>

Hi Alex,

Welcome to FreeSWITCH! We're glad you are here. As to your question, this is
common for new ones. Don't worry - once you get all the pieces together you
will appreciate the power of the XML dialplan.

The key to doing AND logic is to "stack" conditions. However, your situation
is a bit more complex because you are checking 3 different values in
addition to the destination phone number. In this case I would create a
special dialplan extension whose job it is to determine if the bank is open
or closed, and set a channel variable. Then, use that channel variable in a
second extension:

<!-- This extension sets the chan var ${bank_open} to "false" by default -->
<extension name="Determine if bank is open" continue="true">
  <condition>
    <!-- this is an "absolute" condition - it will always be true, so this
action will always take place -->
    <action application="set" data="bank_open=false" inline="true"/>
  </condition>
  <condition field="wday" expression="2-5"/>
  <condition field="${bank_holiday}" expression="false"/>
  <condition field="hour" expression"x-y">
    <!-- this action only gets set if all 3 conditions above evaluate to
true -->
    <action application="set" data="bank_open=true" inline="true"/>
  </condition>
</extension>

Okay, so now you have ${bank_open} that will be true or false. It's just a
matter of routing now:
<extension name="Determine if bank is open" continue="true">
  <condition field="destination_number" expression="^1?\d{10}$"
break="never"/>
  <condition field="${bank_open}" expression="true">
    <action application="transfer" data="ivr_destination"/>
    <anti-action application="transfer" data="vm_destination"/>
  </condition>
</extension>

There is one catch here: At some point you have to define the variable
${bank_holiday} for this to work. :)

I hope this makes sense. Also, this topic is covered in detail in chapters 5
and 8 of the FreeSWITCH book that we wrote, so you may wish to acquire that.
Let us know if you run into any trouble putting this into action.

-MC

On Tue, Jul 26, 2011 at 4:50 AM, Alex Lake <alex at digitalmail.com> wrote:

> Hello all, I'm a Freeswitch newbie struggling to get my head around the
> best way to do some TOD/BH routing.
> I've got a LUA script that will work out whether or not we're in a bank
> holiday, but I wanted to do the combinatorial logic within Freeswitch
> dial plan.
>
> Something like:
>
> if (Weekday AND (NOT BankHoliday) AND InWorkingHours) {
>     Transfer to mainIVR
> } else {
>     Transfer to voicemail
> }
>
> What is the most elegant way to achieve this?
> The Wiki pages don't appear to explain how one can do ANDs with ELSE, as
> the anti-action will only get executed if it's only the last condition
> that fails.
>
> Thanks!
>
>
>
>
> _______________________________________________
> 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 bracken_dave at yahoo.com  Wed Jul 27 03:33:38 2011
From: bracken_dave at yahoo.com (Dave Bracken)
Date: Tue, 26 Jul 2011 16:33:38 -0700 (PDT)
Subject: [Freeswitch-users] (no subject)
Message-ID: <1311723218.97772.yint-ygo-j2me@web114508.mail.gq1.yahoo.com>

Don?t waste your time!.. http://bamboo-nails.com/important.php?diID=13fu0


From simpot at simpot.com  Wed Jul 27 03:48:46 2011
From: simpot at simpot.com (Dmitry Saratsky)
Date: Wed, 27 Jul 2011 02:48:46 +0300
Subject: [Freeswitch-users] Dailplan question?
Message-ID: <52114D605A462A4E9A50E588EE7D068601311C981D2F@mail.forest.simpot.com>

Is there any way to bridge/transfer call to dialplan logic?

Asterisk example: Local/XXXXXXXXXXX at outbound-allroutes

For example, if I have 3 external trunks to sip providers and DISA service, I want to bridge the call automatically to the regex suitable gateway in dialplan
I mean, how can I pass the call to known number (without specifying trunk/gateway, so the relevant gateway will be chosen by dialplan logic)

Thanks,
Dmitry.
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From freeswitch at simpot.com  Wed Jul 27 03:49:33 2011
From: freeswitch at simpot.com (Dmitry Saratsky)
Date: Wed, 27 Jul 2011 02:49:33 +0300
Subject: [Freeswitch-users] Dailplan question?
Message-ID: <000001cc4bee$ac8af7f0$05a0e7d0$@com>

Is there any way to bridge/transfer call to dialplan logic?

 

Asterisk example: Local/XXXXXXXXXXX at outbound-allroutes

 

For example, if I have 3 external trunks to sip providers and DISA service,
I want to bridge the call automatically to the regex suitable gateway in
dialplan

I mean, how can I pass the call to known number (without specifying
trunk/gateway, so the relevant gateway will be chosen by dialplan logic)

 

Thanks,

Dmitry.

 

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From nsirugudi at gmail.com  Wed Jul 27 03:58:33 2011
From: nsirugudi at gmail.com (Narendra Sirugudi)
Date: Wed, 27 Jul 2011 05:28:33 +0530
Subject: [Freeswitch-users] NAT and Freeswitch
In-Reply-To: <B519BBC8-D4E7-4583-86D2-A82248E449E5@ipeva.fr>
References: <CAP18h9osfM6+NUgJkJ7tNfz=mLQzcvcDR1t5k5VsoFXGwH66+w@mail.gmail.com>
	<CAP18h9qrTsTo2mUKMuXY0n8RawGbN_5CZWayCOfV=6F2BbuQyg@mail.gmail.com>
	<CAMWiMy-oXiqct_HBak8k7qQnHcdwcdM1kJKxb5UjxMZErftpvQ@mail.gmail.com>
	<CAP18h9pxFEmsRZTrO7VpcLVvo8OuwDjra67cn551y22iMWdJiw@mail.gmail.com>
	<CAMWiMy_5OTymJZ5Zf-BeefQ_vOU9Ot_ZbTSdd16yx9OJWSk3Dw@mail.gmail.com>
	<B519BBC8-D4E7-4583-86D2-A82248E449E5@ipeva.fr>
Message-ID: <CAP18h9p9be55HxpWXk72Tm5-sWgWw4dwXCdCY2Cz2Am1aY4_=w@mail.gmail.com>

David,
Thanks for your reply.

rfc 3581 is indeed helpful.

If my understanding is correct, this helps the SIP UA client to learn the
SIP session port as seen by the SIP server.

But i think this does not solve the issue of RTP IP and port in the SDP,
unless i am missing something.

The basic issue with SIP UA client sitting behind a NAT server is that the
SDP in the INVITE from the SIP UA, will contain the private IP & port of the
SIP UA. At best the SIP server can assume that the RTP IP will be the source
IP address seen on the IP header of the INVITE. But how to get the SIP UA
client's NATTED RTP port?

A public STUN server does solve this issue, wherein the client first queries
the STUN server to learn the natted RTP IP and port seen by the STUN server.
Many SIP UA clients (softphones) seem to have the support for STUN, but STUN
does not work with symmetric NAT. TURN, ICE are the suggested solutions. But
i am yet to see any SIP UA clients having TURN, ICE support. One
can definitely say that most SIP UA clients currently do not have TURN/ICE
support. Moreover TURN/ICE might have latency issues.

Hence one tends to still look at SIP server based solutions.

Please do get back with your thoughts. Its possible that i am missing
something.

thanks,
--naren




On Tue, Jul 26, 2011 at 10:07 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> More precisely, FreeSWITCH does not have any issues to handle a remote
> endpoint behind NAT if the endpoint correctly sets the rport parameter in
> the Via field (RFC3581).
> In case the endpoint does not, FreeSWITCH can force rport, meaning it will
> answer to the endpoint as if rport was set.
> This mechanism does not have anything to do with STUN/ICE or other external
> mechanism and it works perfectly.
>
>   David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 26/07/2011 ? 07:23, curriegrad2004 a ?crit :
>
>  In this case there's nothing I can think of that FreeSWITCH can do in
> this case if the remote endpoint doesn't support NAT at all.
>
> You may want to try the nf_nat_sip module at the remote end, but it
> may not be the solution you're looking for
>
> On Mon, Jul 25, 2011 at 8:48 PM, Narendra Sirugudi <nsirugudi at gmail.com>
> wrote:
>
>
> Hi,
>
> Thanks for your reply.
>
>
> Maybe my problem definition is not fully clear:
>
> For my use case, having addtional configuration on the SIP UA (softphone)
>
> side is not feasible. I am looking for solution on the Freeswitch
>
> side(sitting on the public network). Not all SIP UA clients (softphones)
>
> will have the solution of working around NAT, let alone STUN client
> support.
>
>
> I observed the Freeswitch has some features like AutoNAT etc. Aren't these
>
> of any help for the use case i have ? I just started reading on the AutoNAT
>
> feature. The documentation of AutoNAT seems to suggest that freeswitch
>
> itself sits behind a NAT. This is not the same as my use case.
>
>
> To summarize: Does freeswitch have any method of detecting that the remote
>
> SIP UA is behind a NAT and taking some action based on that ?
>
> Any suggestions are welcome.
>
>
> thanks,
>
> --naren
>
> On Tue, Jul 26, 2011 at 7:09 AM, curriegrad2004 <curriegrad2004 at gmail.com>
>
> wrote:
>
>
>  The SIP UA only needs to be configured to use NAT in this case. The FS
>
>  Server does not need any configuration in this case.
>
>
>
>
> _______________________________________________
>
> 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
>
>
>
>
> _______________________________________________
> 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
>
>
>
> _______________________________________________
> 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 msc at freeswitch.org  Wed Jul 27 04:27:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 26 Jul 2011 17:27:53 -0700
Subject: [Freeswitch-users] Dailplan question?
In-Reply-To: <000001cc4bee$ac8af7f0$05a0e7d0$@com>
References: <000001cc4bee$ac8af7f0$05a0e7d0$@com>
Message-ID: <CAKzWOxU7yvgPjffsQGaF2sV+P5FG3h7x9DsUVOOjTYoyimTt8A@mail.gmail.com>

Check out the loopback channel:

http://wiki.freeswitch.org/wiki/Loopback

-MC

On Tue, Jul 26, 2011 at 4:49 PM, Dmitry Saratsky <freeswitch at simpot.com>wrote:

> Is there any way to bridge/transfer call to dialplan logic?****
>
> ** **
>
> Asterisk example: Local/XXXXXXXXXXX at outbound-allroutes****
>
> ** **
>
> For example, if I have 3 external trunks to sip providers and DISA service,
> I want to bridge the call automatically to the regex suitable gateway in
> dialplan****
>
> I mean, how can I pass the call to known number (without specifying
> trunk/gateway, so the relevant gateway will be chosen by dialplan logic)**
> **
>
> ** **
>
> Thanks,****
>
> Dmitry.****
>
> ** **
>
> _______________________________________________
> 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 freeswitch at simpot.com  Wed Jul 27 05:06:44 2011
From: freeswitch at simpot.com (Dmitry Saratsky)
Date: Wed, 27 Jul 2011 04:06:44 +0300
Subject: [Freeswitch-users] Dailplan question?
In-Reply-To: <CAKzWOxU7yvgPjffsQGaF2sV+P5FG3h7x9DsUVOOjTYoyimTt8A@mail.gmail.com>
References: <000001cc4bee$ac8af7f0$05a0e7d0$@com>
	<CAKzWOxU7yvgPjffsQGaF2sV+P5FG3h7x9DsUVOOjTYoyimTt8A@mail.gmail.com>
Message-ID: <000501cc4bf9$7371cec0$5a556c40$@com>

Thanks a lot!!!

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: 27 Jul 2011 03:28
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dailplan question?

 

Check out the loopback channel:

http://wiki.freeswitch.org/wiki/Loopback

-MC

On Tue, Jul 26, 2011 at 4:49 PM, Dmitry Saratsky <freeswitch at simpot.com>
wrote:

Is there any way to bridge/transfer call to dialplan logic?

 

Asterisk example: Local/XXXXXXXXXXX at outbound-allroutes

 

For example, if I have 3 external trunks to sip providers and DISA service,
I want to bridge the call automatically to the regex suitable gateway in
dialplan

I mean, how can I pass the call to known number (without specifying
trunk/gateway, so the relevant gateway will be chosen by dialplan logic)

 

Thanks,

Dmitry.

 


_______________________________________________
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 gavin.henry at gmail.com  Wed Jul 27 10:37:48 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Wed, 27 Jul 2011 07:37:48 +0100
Subject: [Freeswitch-users] DTMF echo/crossed?
Message-ID: <CAA8_NKAPZcSXjBzRuxxAiZrChOZF17zZ_iG6L3jg7tjmMjbTzw@mail.gmail.com>

Hi all,

We generally don't suffer any type of echo deploying customer
multi-tenant systems, but an interesting one was reported by a
customer who said they could hear DTMF on the line that they were not
generating?

This is part of a business centre multi-tenant setup with just one
user on that domain who heard the DTMF. They are lots of domains on
that system. Any suggestions?

Thanks,

Gavin.

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


From sanja_angelova at hotmail.com  Wed Jul 27 11:18:22 2011
From: sanja_angelova at hotmail.com (Sanja Angelova)
Date: Wed, 27 Jul 2011 09:18:22 +0200
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <1311723218.97772.yint-ygo-j2me@web114508.mail.gq1.yahoo.com>
References: <1311723218.97772.yint-ygo-j2me@web114508.mail.gq1.yahoo.com>
Message-ID: <DUB109-W1251500D91928E5B6610C38FE350@phx.gbl>


That is very nice of you! Thank you!
 > Date: Tue, 26 Jul 2011 16:33:38 -0700
> From: bracken_dave at yahoo.com
> To: edavis51 at sbcglobal.net; elaine.kaiser at acs-inc.com; employment at itagroup.com; zacfedak at yahoo.com; freeswitch-users at lists.freeswitch.org; gemerson at theshurngroup.com; grishamservices at sbcglobal.net; jgierer at justourthought.com
> Subject: Re: [Freeswitch-users] (no subject)
> 
> Don?t waste your time!.. http://bamboo-nails.com/important.php?diID=13fu0
> 
> _______________________________________________
> 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 gavin.henry at gmail.com  Wed Jul 27 11:37:19 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Wed, 27 Jul 2011 08:37:19 +0100
Subject: [Freeswitch-users] DTMF echo/crossed?
In-Reply-To: <CAA8_NKAPZcSXjBzRuxxAiZrChOZF17zZ_iG6L3jg7tjmMjbTzw@mail.gmail.com>
References: <CAA8_NKAPZcSXjBzRuxxAiZrChOZF17zZ_iG6L3jg7tjmMjbTzw@mail.gmail.com>
Message-ID: <CAA8_NKDot+b+UGFwHWEpP9UkzcNV-7EswwNkMZB9j9Ys-8btNQ@mail.gmail.com>

Forgot to say this is pure SIP (before the carriers)

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


From peter.olsson at visionutveckling.se  Wed Jul 27 11:48:55 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 27 Jul 2011 09:48:55 +0200
Subject: [Freeswitch-users] DTMF echo/crossed?
In-Reply-To: <CAA8_NKAPZcSXjBzRuxxAiZrChOZF17zZ_iG6L3jg7tjmMjbTzw@mail.gmail.com>
References: <CAA8_NKAPZcSXjBzRuxxAiZrChOZF17zZ_iG6L3jg7tjmMjbTzw@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F92@cooper>

My guess would be that the other party was generating DTMF?

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Gavin Henry [gavin.henry at gmail.com]
Skickat: den 27 juli 2011 08:37
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] DTMF echo/crossed?

Hi all,

We generally don't suffer any type of echo deploying customer
multi-tenant systems, but an interesting one was reported by a
customer who said they could hear DTMF on the line that they were not
generating?

This is part of a business centre multi-tenant setup with just one
user on that domain who heard the DTMF. They are lots of domains on
that system. Any suggestions?

Thanks,

Gavin.

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

_______________________________________________
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

!DSPAM:4e2fb31132766269912559!



From david.ponzone at ipeva.fr  Wed Jul 27 13:42:31 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 27 Jul 2011 11:42:31 +0200
Subject: [Freeswitch-users] NAT and Freeswitch
In-Reply-To: <CAP18h9p9be55HxpWXk72Tm5-sWgWw4dwXCdCY2Cz2Am1aY4_=w@mail.gmail.com>
References: <CAP18h9osfM6+NUgJkJ7tNfz=mLQzcvcDR1t5k5VsoFXGwH66+w@mail.gmail.com>
	<CAP18h9qrTsTo2mUKMuXY0n8RawGbN_5CZWayCOfV=6F2BbuQyg@mail.gmail.com>
	<CAMWiMy-oXiqct_HBak8k7qQnHcdwcdM1kJKxb5UjxMZErftpvQ@mail.gmail.com>
	<CAP18h9pxFEmsRZTrO7VpcLVvo8OuwDjra67cn551y22iMWdJiw@mail.gmail.com>
	<CAMWiMy_5OTymJZ5Zf-BeefQ_vOU9Ot_ZbTSdd16yx9OJWSk3Dw@mail.gmail.com>
	<B519BBC8-D4E7-4583-86D2-A82248E449E5@ipeva.fr>
	<CAP18h9p9be55HxpWXk72Tm5-sWgWw4dwXCdCY2Cz2Am1aY4_=w@mail.gmail.com>
Message-ID: <DE052219-F288-402F-920C-E111157F8F25@ipeva.fr>

That's called rtp auto adjust.
The SIP server will not trust what it's in the SDP, but the real ip/source of the RTP flow coming to its own RTP port.


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

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

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




Le 27/07/2011 ? 01:58, Narendra Sirugudi a ?crit :

> David,
> Thanks for your reply.
>  
> rfc 3581 is indeed helpful.
>  
> If my understanding is correct, this helps the SIP UA client to learn the SIP session port as seen by the SIP server.
>  
> But i think this does not solve the issue of RTP IP and port in the SDP, unless i am missing something.
>  
> The basic issue with SIP UA client sitting behind a NAT server is that the SDP in the INVITE from the SIP UA, will contain the private IP & port of the SIP UA. At best the SIP server can assume that the RTP IP will be the source IP address seen on the IP header of the INVITE. But how to get the SIP UA client's NATTED RTP port?
>  
> A public STUN server does solve this issue, wherein the client first queries the STUN server to learn the natted RTP IP and port seen by the STUN server. Many SIP UA clients (softphones) seem to have the support for STUN, but STUN does not work with symmetric NAT. TURN, ICE are the suggested solutions. But i am yet to see any SIP UA clients having TURN, ICE support. One can definitely say that most SIP UA clients currently do not have TURN/ICE support. Moreover TURN/ICE might have latency issues.
>  
> Hence one tends to still look at SIP server based solutions.
>  
> Please do get back with your thoughts. Its possible that i am missing something.
>  
> thanks,
> --naren
>  
> 
> 
>  
> On Tue, Jul 26, 2011 at 10:07 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> More precisely, FreeSWITCH does not have any issues to handle a remote endpoint behind NAT if the endpoint correctly sets the rport parameter in the Via field (RFC3581).
> In case the endpoint does not, FreeSWITCH can force rport, meaning it will answer to the endpoint as if rport was set.
> This mechanism does not have anything to do with STUN/ICE or other external mechanism and it works perfectly.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 26/07/2011 ? 07:23, curriegrad2004 a ?crit :
> 
>> In this case there's nothing I can think of that FreeSWITCH can do in
>> this case if the remote endpoint doesn't support NAT at all.
>> 
>> You may want to try the nf_nat_sip module at the remote end, but it
>> may not be the solution you're looking for
>> 
>> On Mon, Jul 25, 2011 at 8:48 PM, Narendra Sirugudi <nsirugudi at gmail.com> wrote:
>>> 
>>> Hi,
>>> Thanks for your reply.
>>> 
>>> Maybe my problem definition is not fully clear:
>>> For my use case, having addtional configuration on the SIP UA (softphone)
>>> side is not feasible. I am looking for solution on the Freeswitch
>>> side(sitting on the public network). Not all SIP UA clients (softphones)
>>> will have the solution of working around NAT, let alone STUN client support.
>>> 
>>> I observed the Freeswitch has some features like AutoNAT etc. Aren't these
>>> of any help for the use case i have ? I just started reading on the AutoNAT
>>> feature. The documentation of AutoNAT seems to suggest that freeswitch
>>> itself sits behind a NAT. This is not the same as my use case.
>>> 
>>> To summarize: Does freeswitch have any method of detecting that the remote
>>> SIP UA is behind a NAT and taking some action based on that ?
>>> Any suggestions are welcome.
>>> 
>>> thanks,
>>> --naren
>>> On Tue, Jul 26, 2011 at 7:09 AM, curriegrad2004 <curriegrad2004 at gmail.com>
>>> wrote:
>>>> 
>>>> The SIP UA only needs to be configured to use NAT in this case. The FS
>>>> Server does not need any configuration in this case.
>>>> 
>>>> 
>>> 
>>> _______________________________________________
>>> 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
>>> 
>>> 
>> 
>> _______________________________________________
>> 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
> 
> 
> _______________________________________________
> 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
> 
> 
> _______________________________________________
> 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 david.ponzone at ipeva.fr  Wed Jul 27 13:51:38 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 27 Jul 2011 11:51:38 +0200
Subject: [Freeswitch-users] FS to a Sonus SIP trunk
In-Reply-To: <4E2ED4AB.1030305@hcu-hamburg.de>
References: <4E2ED4AB.1030305@hcu-hamburg.de>
Message-ID: <D2E38177-1466-4EA5-BC3A-419A7E6351B0@ipeva.fr>

Sonus sucks.

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

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

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




Le 26/07/2011 ? 16:52, michael knop a ?crit :

> Hi all!
> 
> I?m trying to connect my FS to a Sonus SIP trunk. I followed the 
> instruction at
> 
>   http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus
> 
> but it did not work. At the beginning of a call voice quality is good. 
> After a while it changes to choppy.
> 
> I don?t know if it?s the same problem: When I call the Tetris extension 
> via Sonus SIP trunk the sound is too fast and I?m getting log entries 
> like the following one:
> 
> [...]
> 2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME 
> not supported, changing our end from 20 to 10
> 2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from 
> PCMA at 20ms@8000hz to PCMA at 10ms@8000hz
> 2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global 
> timer resolution to 10ms to handle interval 10
> 2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer 
> [soft] 80 bytes per 10ms
> 2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec 
> sofia/external/+4940... at 193...:5060 PCMA/8000 10 ms 80 samples 64000 bits
> 2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write 
> Buffer at 160 bytes to accommodate 320->160
> [...]
> 
> This problem is fixed by adding the following line to 
> conf/sip_profiles/external.xml:
> 
> <param name="rtp-autofix-timing" value="false"/>
> 
> Any hints?
> 
> /micha
> 
> _______________________________________________
> 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 michael.knop at hcu-hamburg.de  Wed Jul 27 14:48:03 2011
From: michael.knop at hcu-hamburg.de (michael knop)
Date: Wed, 27 Jul 2011 12:48:03 +0200
Subject: [Freeswitch-users] FS to a Sonus SIP trunk
In-Reply-To: <D2E38177-1466-4EA5-BC3A-419A7E6351B0@ipeva.fr>
References: <4E2ED4AB.1030305@hcu-hamburg.de>
	<D2E38177-1466-4EA5-BC3A-419A7E6351B0@ipeva.fr>
Message-ID: <4E2FECE3.7050901@hcu-hamburg.de>

full ack


Am 27.07.2011 11:51, schrieb David Ponzone:
> Sonus sucks.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel: 01 74 03 18 97
> gsm: 06 66 98 76 34
>
> Service ClientIPeva
> tel: 0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>- www.ipeva-studio.com
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. /*/IPeva/*/d?cline toute responsabilit? au
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous
> n'?tes pas destinataire de ce message, merci de le d?truire
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 26/07/2011 ? 16:52, michael knop a ?crit :
>
>> Hi all!
>>
>> I?m trying to connect my FS to a Sonus SIP trunk. I followed the
>> instruction at
>>
>> http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus
>>
>> but it did not work. At the beginning of a call voice quality is good.
>> After a while it changes to choppy.
>>
>> I don?t know if it?s the same problem: When I call the Tetris extension
>> via Sonus SIP trunk the sound is too fast and I?m getting log entries
>> like the following one:
>>
>> [...]
>> 2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME
>> not supported, changing our end from 20 to 10
>> 2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from
>> PCMA at 20ms@8000hz to PCMA at 10ms@8000hz
>> 2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global
>> timer resolution to 10ms to handle interval 10
>> 2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer
>> [soft] 80 bytes per 10ms
>> 2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec
>> sofia/external/+4940... at 193...:5060 PCMA/8000 10 ms 80 samples 64000 bits
>> 2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write
>> Buffer at 160 bytes to accommodate 320->160
>> [...]
>>
>> This problem is fixed by adding the following line to
>> conf/sip_profiles/external.xml:
>>
>> <param name="rtp-autofix-timing" value="false"/>
>>
>> Any hints?
>>
>> /micha
>>
>> _______________________________________________
>> 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
>> <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
>
>
>
> _______________________________________________
> 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 acrow at integrafin.co.uk  Wed Jul 27 14:51:46 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Wed, 27 Jul 2011 11:51:46 +0100
Subject: [Freeswitch-users] Intercept taking too long
In-Reply-To: <4E2ECA0C.9080104@rosengart.de>
References: <4E2EAE84.6080407@integrafin.co.uk> <4E2ECA0C.9080104@rosengart.de>
Message-ID: <4E2FEDC2.9030101@integrafin.co.uk>



On 26/07/11 15:07, Frank Rosengart wrote:
> On 07/26/2011 02:09 PM, Alex Crow wrote:
>
>> I have set up a system with 6 snom 370s and FS and I'm seeing a long
>> delay (up to 3 seconds) when trying to do a pickup using the BLF.
> We are using Snom 360 with 7.3.30, FS from git around mid of june. The
> manual pickup with **<extension>  is prompt, no delay.
>
> Do you have the delay an manual pickup, too?
>
> How does your uuid-to-pickup database lookup look like?
>
>
>
> Frank
>
>
>
> PS: I'm still stuck with the BLF thing: busy (LED on) works, but not
> blinking (monitored ext is ringing) - there is no notify sent to the phones.
>
Frank,

I know if "Peer to Peer Call Completion" in Advanced menu on the Snom is 
enabled, this stops BLF working properly. Must be disabled.

Also if "Publish Presence" in Advance is disabled I had issues with the 
BLF stopping working after a rereg, so I left that and "Publish Presence 
on Bootup" in the identity page enabled.

Cheers

Alex

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

"Transact" is operated by Integrated Financial Arrangements plc
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Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)



From dome at tel.co.th  Wed Jul 27 17:37:29 2011
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 27 Jul 2011 20:37:29 +0700
Subject: [Freeswitch-users] FS as registra server behind SBC
Message-ID: <CAMFECSCCsjBpJ1MzPbP=8uo_WAkAjnnt6bHG5q=r2Ny0O6skvQ@mail.gmail.com>

Dear All,
           I found problem when setup FS behind  Huwei SBC.
my network diagram is.
                            NAT                     Public IP
          NAT 10.x.x.x
[SIP Client 1] ----- [ADSL Router] ----------------------------  SBC
  ----------------------------  FS

/
[SIP Client 2] ----- [ADSL Router] --------------------------/

It's work fine when call incomming and play sound. But i got 403 from SBC
when i try to call Local extension or originate call to registerd user.

I thing nat problem can someone show me sip profile config work in this
case.


BG

Dome C.
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From adam.kelloway at newpace.ca  Wed Jul 27 17:53:21 2011
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Wed, 27 Jul 2011 10:53:21 -0300
Subject: [Freeswitch-users] Can record_min_sec be used in record dial plan
	tool?
Message-ID: <4E301851.2020507@newpace.ca>

Hi there,

In regards to http://wiki.freeswitch.org/wiki/Variable_record_min_sec :

I can't seem to get record_min_sec to work when using the record dial 
plan tool. It doesn't detect that a recording is too short and therefore 
doesn't delete the file.

A look at the code seems to suggest that it is only checked when you are 
using the media_bug_answer_req feature and/or the record_session dial 
plan tool. Is this true? Is there any way I can enable this for the 
record dial plan tool?

My extension does the following:

...
<action application="set" data="record_min_sec=2"/>
<action application="record" data="/tmp/${uuid}.wav 30 200"/>
...

I also tried the following, with no difference in results:

...
<action application="set" data="RECORD_MIN_SEC=2"/>
<action application="record" data="/tmp/${uuid}.wav 30 200"/>
...

Thanks,

Adam


From xing2kin at yahoo.com  Wed Jul 27 07:54:19 2011
From: xing2kin at yahoo.com (king2kin)
Date: Tue, 26 Jul 2011 20:54:19 -0700 (PDT)
Subject: [Freeswitch-users] Busy Status of a pool of IVR Channels;
	any HelloWorld app module for reference?
In-Reply-To: <CAKzWOxU7yvgPjffsQGaF2sV+P5FG3h7x9DsUVOOjTYoyimTt8A@mail.gmail.com>
Message-ID: <1311738859.25202.YahooMailClassic@web39705.mail.mud.yahoo.com>

Hi Folks,
?
I am a newbie to FreeSwitch, read some docs and ebook on?FreeSwitch, and successfully compiled and tested it on Windows 2003 during the past?two weeks.
?
Now I'd like to develop a FreeSwitch-SIP-VoIP based IVR and integrate it together with?our in-house speech app server, and run them on a server machine on public network. The inbound and outboud calls between the FreeSwitch IVR and external cell phones?go through?the 3rd-party?SIP-VoIP-PSTN gateway provider.
?
Here is my use-case:
?
reserve?50 and 100?sip voip channels for inbound and outbound ivr calls, respectively. All the?50?inbound sip channels?are bound to a single DID phone number so that any cell phone users may call in to interact with an IVR dialplan script by the DID phone number; also, our?scheduler application working as a client of FreeSwitch?to make use of all the 100 outbound sip channels to periodically dial our users' cell phone numbers and execute ivr dialplan script, save the results.
?
Then, I get two questions here:
?
1. When?a cell phone user calls in the IVR?by DID,?will?FreeSwitch automatically pick up an idle channel (or extension) from the pool of 50 inbound sip channels or we have to develop a dialpan application to manually pick up an idle one from the pool according to some status variable (idle or busy) of sip channel? Similarily, when making outbound calls from the IVR to cell phones, we also need check whether a channel in the pool of outbound sip channels is idle or not by a status variable.?
?
2. Is there any HelloWorld app module in C or C++ for FreeSwitch for reference when one tries to develop a custom application module from the scratch? It'd better have two functions (sync one, async one).
?
?To handle events and messages between these inbound?or outbound calls and our speech app server, I am going to write a custom diaplan application module?in C or C++. After?roughly?reading the source codes of mod_flite and mod_portaudio (I ever compiled and run the two projects 3 years ago),?I am still not very clear where to start my app,?and how to expose?its functions to FreeSwitch.
?
Thanks!
?
?
?
?
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From chrisbware at interfree.it  Wed Jul 27 19:35:20 2011
From: chrisbware at interfree.it (chrisbware at interfree.it)
Date: 27 Jul 2011 15:35:20 -0000
Subject: [Freeswitch-users] Gateway real-time
Message-ID: <20110727153520.30965.qmail@community22.interfree.it>

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From lakersman2006 at yahoo.com  Wed Jul 27 19:59:41 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Wed, 27 Jul 2011 08:59:41 -0700 (PDT)
Subject: [Freeswitch-users] Gateway real-time
In-Reply-To: <20110727153520.30965.qmail@community22.interfree.it>
Message-ID: <1311782381.45296.YahooMailClassic@web161003.mail.bf1.yahoo.com>

You do know about Asterisk Realtime?

--- On Wed, 7/27/11, chrisbware at interfree.it <chrisbware at interfree.it> wrote:

From: chrisbware at interfree.it <chrisbware at interfree.it>
Subject: [Freeswitch-users] Gateway real-time
To: freeswitch-users at lists.freeswitch.org
Date: Wednesday, July 27, 2011, 8:35 AM


Hi guys,
?
I've successfully configured FS to work with mod_xml_curl and a cgi, in order to configure Sofia channel.
In this way, all Sofia configuration (including gateway definition) is loaded from my DB.
?
Next step I'd like to do is add a gateway on the fly, so that when gateway parameters are on DB table, it's
automatically loaded in FS. I hate asterisk "sip reload" method and I hope there's another solution with FS, not based on
config reload.
?
Can you help?
?
Thanks.



-------------------------------------------------------------------------------
Valore legale alle tue mail
InterfreePEC - la tua Posta Elettronica Certificata
http://pec.interfree.it
-------------------------------------------------------------------------------



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

_______________________________________________
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
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From msc at freeswitch.org  Wed Jul 27 20:05:25 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Jul 2011 09:05:25 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxX1ObRXNcALC2gEk6BWhXxKCVeoDLXNrt+h1E3etN-F2w@mail.gmail.com>

Hey all,

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

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

Thanks!
-MC
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From chrisbware at interfree.it  Wed Jul 27 20:10:57 2011
From: chrisbware at interfree.it (chrisbware at interfree.it)
Date: 27 Jul 2011 16:10:57 -0000
Subject: [Freeswitch-users] Gateway real-time
Message-ID: <20110727161057.18435.qmail@community17.interfree.it>

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

Hello!

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

Thanks!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From steveayre at gmail.com  Wed Jul 27 20:40:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 27 Jul 2011 17:40:01 +0100
Subject: [Freeswitch-users] Gateway real-time
In-Reply-To: <20110727153520.30965.qmail@community22.interfree.it>
References: <20110727153520.30965.qmail@community22.interfree.it>
Message-ID: <CAFiqYumwtVqWwLQcCK3N6xyCSzJZED5S_3R1nx+rkHWzeoHWHA@mail.gmail.com>

That's not possible. Sofia needs you to do a 'sofia profile <name> rescan'
to pick up new gateways, and if a gateway has changed its parameters you
need to do 'sofia profile <name> killgw <gwname>' then rescan.

There's a reason for it too - FS has to register with the gateway then keep
pinging it with OPTIONS requests. That means the gateway needs an object
stored in memory. It wouldn't be ideal for FS to constantly poll a database
looking for changes for something that might never change, so you're left
with rescan.

You could however have a DB trigger that runs a script that connects via ESL
and does that for you though...

-Steve



On 27 July 2011 16:35, <chrisbware at interfree.it> wrote:

> Hi guys,
>
> I've successfully configured FS to work with mod_xml_curl and a cgi, in
> order to configure Sofia channel.
> In this way, all Sofia configuration (including gateway definition) is
> loaded from my DB.
>
> Next step I'd like to do is add a gateway on the fly, so that when gateway
> parameters are on DB table, it's
> automatically loaded in FS. I hate asterisk "sip reload" method and I hope
> there's another solution with FS, not based on
> config reload.
>
> Can you help?
>
> Thanks.
>
>
> -------------------------------------------------------------------------------
> Valore legale alle tue mail
> InterfreePEC - la tua Posta Elettronica Certificata
> http://pec.interfree.it
>
> -------------------------------------------------------------------------------
>
> _______________________________________________
> 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 anu at familytv.com  Wed Jul 27 20:45:41 2011
From: anu at familytv.com (Anirudha Shimpi)
Date: Wed, 27 Jul 2011 10:45:41 -0600
Subject: [Freeswitch-users] Well, that didn't work very well did it? ...
Message-ID: <00b301cc4c7c$9ee11ab0$dca35010$@familytv.com>

What does that mean? We are having issues where Freeswitch will stop
answering calls and transferring them to conferences, does this message have
something to do with it? If so, what is the cause?

 

Anirudha Shimpi

 

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

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

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

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

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From yehavi.bourvine at gmail.com  Wed Jul 27 22:54:35 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 27 Jul 2011 21:54:35 +0300
Subject: [Freeswitch-users] FS to a Sonus SIP trunk
In-Reply-To: <4E2ED4AB.1030305@hcu-hamburg.de>
References: <4E2ED4AB.1030305@hcu-hamburg.de>
Message-ID: <CAAtWKeGecHp4683w+-zi-mTNGqjV0M=cMj+jL6GafWFNNi8B-g@mail.gmail.com>

Hello Micha,

  How much time does it take until the sound is choppy?
We have been connected to Sonus provider up to a week ago. Incoming calls
started being choppy after about 15 minutes (outgoing calls were ok). We
also had inconsistent problems with DTMF. We ended it by changing a
supplier...

                     __Yehavi:

2011/7/26 michael knop <michael.knop at hcu-hamburg.de>

> Hi all!
>
> I?m trying to connect my FS to a Sonus SIP trunk. I followed the
> instruction at
>
>   http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus
>
> but it did not work. At the beginning of a call voice quality is good.
> After a while it changes to choppy.
>
> I don?t know if it?s the same problem: When I call the Tetris extension
> via Sonus SIP trunk the sound is too fast and I?m getting log entries
> like the following one:
>
> [...]
> 2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME
> not supported, changing our end from 20 to 10
> 2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from
> PCMA at 20ms@8000hz to PCMA at 10ms@8000hz
> 2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global
> timer resolution to 10ms to handle interval 10
> 2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer
> [soft] 80 bytes per 10ms
> 2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec
> sofia/external/+4940... at 193...:5060 PCMA/8000 10 ms 80 samples 64000 bits
> 2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write
> Buffer at 160 bytes to accommodate 320->160
> [...]
>
> This problem is fixed by adding the following line to
> conf/sip_profiles/external.xml:
>
> <param name="rtp-autofix-timing" value="false"/>
>
> Any hints?
>
> /micha
>
> _______________________________________________
> 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 juraj.fabo at gmail.com  Wed Jul 27 21:21:06 2011
From: juraj.fabo at gmail.com (Juraj Fabo)
Date: Wed, 27 Jul 2011 19:21:06 +0200
Subject: [Freeswitch-users] freetdm api - ftdm_channel_read()
Message-ID: <CAAqVCyTi9m0+JPhjt_sLLkEa97ijFLmETjPoFRUcEQJRmQsN8w@mail.gmail.com>

Hi

I'm prototyping an application which uses libfreetdm above the sangoma
isdn stack and sangoma a104d card.
I would like to ask for help about proper reading of voice from the
particular channel.

According to return value of sangoma_get_rx_queue_sz() the default rx
queue size is 10.
I did tests when application skipped several reads on given channel,
then called ftdm_channel_read() w/ zero timeout  in a loop while the
return value of read was successfull - it was really 10 frames which
were then possible to read.

How can an application do this reading of a channel by the means of
the freetdm api in more nice way?
I did not find a better way yet, than to read it in the loop.
Any way to explicitly read the number of available frames?

Actually, why the ftdm_channel_read() does not read them ALL at once
and sets the number of read bytes via *datalen output parameter?
(assuming provided *data buffer is large enough)
In my tests, always 160 bytes were returned in one particular read.

thank you

juraj


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

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

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

Thanks.

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From joaocarlosleme at gmail.com  Thu Jul 28 00:18:37 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Wed, 27 Jul 2011 13:18:37 -0700
Subject: [Freeswitch-users] Bad sound quality and Abnormally large timer gap
	100953 detected!
Message-ID: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>

I just set up Freeswitch Precompiled for Windows x64 (
http://files-sync.freeswitch.org/windows/installer/) on a VPS Windows Server
2008.
Sound is really bad and I'm getting the following:

2011-07-27 12:33:15.012695 [WARNING] switch_time.c:244 Abnormally large
timer gap 100953 detected!
Do you have your kernel timer frequency set to lower than 1,000Hz? You may
experience audio problems.

also

2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was
executed late by 6 seconds 1 heartbeat (core)
2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was
executed late by 4 seconds 2 check_ip (core)
2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was
executed late by 8 seconds 1 heartbeat (core)
2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was
executed late by 8 seconds 2 check_ip (core)

What can I do?

Also, yesterday, when I first set it up, it was all good.

Thanks,
John
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From gmaruzz at gmail.com  Thu Jul 28 00:25:57 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 27 Jul 2011 22:25:57 +0200
Subject: [Freeswitch-users] Bad sound quality and Abnormally large timer
 gap 100953 detected!
In-Reply-To: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
Message-ID: <CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>

You can't count on the time keeping of a vps.

Use windows directly on real hardware, not a virtual machine.

-giovanni

On 7/27/11, Joao Leme <joaocarlosleme at gmail.com> wrote:
> I just set up Freeswitch Precompiled for Windows x64 (
> http://files-sync.freeswitch.org/windows/installer/) on a VPS Windows Server
> 2008.
> Sound is really bad and I'm getting the following:
>
> 2011-07-27 12:33:15.012695 [WARNING] switch_time.c:244 Abnormally large
> timer gap 100953 detected!
> Do you have your kernel timer frequency set to lower than 1,000Hz? You may
> experience audio problems.
>
> also
>
> 2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was
> executed late by 6 seconds 1 heartbeat (core)
> 2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was
> executed late by 4 seconds 2 check_ip (core)
> 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was
> executed late by 8 seconds 1 heartbeat (core)
> 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was
> executed late by 8 seconds 2 check_ip (core)
>
> What can I do?
>
> Also, yesterday, when I first set it up, it was all good.
>
> Thanks,
> John
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From joaocarlosleme at gmail.com  Thu Jul 28 00:51:20 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Wed, 27 Jul 2011 13:51:20 -0700
Subject: [Freeswitch-users] Bad sound quality and Abnormally large timer
 gap 100953 detected!
In-Reply-To: <CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
Message-ID: <CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>

Thanks! So there is no other way around?

On Wed, Jul 27, 2011 at 1:25 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> You can't count on the time keeping of a vps.
>
> Use windows directly on real hardware, not a virtual machine.
>
> -giovanni
>
> On 7/27/11, Joao Leme <joaocarlosleme at gmail.com> wrote:
> > I just set up Freeswitch Precompiled for Windows x64 (
> > http://files-sync.freeswitch.org/windows/installer/) on a VPS Windows
> Server
> > 2008.
> > Sound is really bad and I'm getting the following:
> >
> > 2011-07-27 12:33:15.012695 [WARNING] switch_time.c:244 Abnormally large
> > timer gap 100953 detected!
> > Do you have your kernel timer frequency set to lower than 1,000Hz? You
> may
> > experience audio problems.
> >
> > also
> >
> > 2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was
> > executed late by 6 seconds 1 heartbeat (core)
> > 2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was
> > executed late by 4 seconds 2 check_ip (core)
> > 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was
> > executed late by 8 seconds 1 heartbeat (core)
> > 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was
> > executed late by 8 seconds 2 check_ip (core)
> >
> > What can I do?
> >
> > Also, yesterday, when I first set it up, it was all good.
> >
> > Thanks,
> > John
> >
>
> --
> Sent from my mobile device
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peter.olsson at visionutveckling.se  Thu Jul 28 01:06:42 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 27 Jul 2011 23:06:42 +0200
Subject: [Freeswitch-users] Bad sound quality and Abnormally large timer
 gap 100953 detected!
In-Reply-To: <CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>,
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>

No, it's never "safe" (not for production at least) to run VoIP on virtual machines. The reason for better results yesterday was probably because the hardware had less total load by that time, so the problem is probably both the virtual machine, plus the hardware has too much load on it.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Joao Leme [joaocarlosleme at gmail.com]
Skickat: den 27 juli 2011 22:51
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!

Thanks! So there is no other way around?

On Wed, Jul 27, 2011 at 1:25 PM, Giovanni Maruzzelli <gmaruzz at gmail.com<mailto:gmaruzz at gmail.com>> wrote:
You can't count on the time keeping of a vps.

Use windows directly on real hardware, not a virtual machine.

-giovanni

On 7/27/11, Joao Leme <joaocarlosleme at gmail.com<mailto:joaocarlosleme at gmail.com>> wrote:
> I just set up Freeswitch Precompiled for Windows x64 (
> http://files-sync.freeswitch.org/windows/installer/) on a VPS Windows Server
> 2008.
> Sound is really bad and I'm getting the following:
>
> 2011-07-27 12:33:15.012695 [WARNING] switch_time.c:244 Abnormally large
> timer gap 100953 detected!
> Do you have your kernel timer frequency set to lower than 1,000Hz? You may
> experience audio problems.
>
> also
>
> 2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was
> executed late by 6 seconds 1 heartbeat (core)
> 2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was
> executed late by 4 seconds 2 check_ip (core)
> 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was
> executed late by 8 seconds 1 heartbeat (core)
> 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was
> executed late by 8 seconds 2 check_ip (core)
>
> What can I do?
>
> Also, yesterday, when I first set it up, it was all good.
>
> Thanks,
> John
>

--
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618<tel:%2B39-347-2665618>

_______________________________________________
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<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4e307b2032763660377611!


From msc at freeswitch.org  Thu Jul 28 01:21:14 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Jul 2011 14:21:14 -0700
Subject: [Freeswitch-users] Gateway real-time
In-Reply-To: <CAFiqYumwtVqWwLQcCK3N6xyCSzJZED5S_3R1nx+rkHWzeoHWHA@mail.gmail.com>
References: <20110727153520.30965.qmail@community22.interfree.it>
	<CAFiqYumwtVqWwLQcCK3N6xyCSzJZED5S_3R1nx+rkHWzeoHWHA@mail.gmail.com>
Message-ID: <CAKzWOxWuhXG5eCVNbBNJ3n7nMUKjfQFgdrr8Z5DXkkFX3KPo+g@mail.gmail.com>

On Wed, Jul 27, 2011 at 9:40 AM, Steven Ayre <steveayre at gmail.com> wrote:

> That's not possible. Sofia needs you to do a 'sofia profile <name> rescan'
> to pick up new gateways, and if a gateway has changed its parameters you
> need to do 'sofia profile <name> killgw <gwname>' then rescan.
>
> There's a reason for it too - FS has to register with the gateway then keep
> pinging it with OPTIONS requests. That means the gateway needs an object
> stored in memory. It wouldn't be ideal for FS to constantly poll a database
> looking for changes for something that might never change, so you're left
> with rescan.
>
> You could however have a DB trigger that runs a script that connects via
> ESL and does that for you though...
>
> -Steve
>
> What about Tony's "Enterprise vs. shuttlecraft" trick?
http://wiki.freeswitch.org/wiki/Clarification:gateways#Clarification

You can trigger a gateway to be "active" only when a particular user
registers, i.e. an inbound registration triggers an outbound registration. I
don't know if it will work in this scenario but it's worth looking into...

-MC
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From lakersman2006 at yahoo.com  Thu Jul 28 01:24:55 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Wed, 27 Jul 2011 14:24:55 -0700 (PDT)
Subject: [Freeswitch-users] Bad sound quality and Abnormally large timer
	gap 100953 detected!
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>,
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
Message-ID: <1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>

I run Asterisk on a virtual machine and it runs quite well even with approximately 50 to 60 simultaneous callers and tha'ts even with media passing thru it.



________________________________
From: Peter Olsson <peter.olsson at visionutveckling.se>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, July 27, 2011 2:06 PM
Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!

No, it's never "safe" (not for production at least) to run VoIP on virtual machines. The reason for better results yesterday was probably because the hardware had less total load by that time, so the problem is probably both the virtual machine, plus the hardware has too much load on it.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Joao Leme [joaocarlosleme at gmail.com]
Skickat: den 27 juli 2011 22:51
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!

Thanks! So there is no other way around?

On Wed, Jul 27, 2011 at 1:25 PM, Giovanni Maruzzelli <gmaruzz at gmail.com<mailto:gmaruzz at gmail.com>> wrote:
You can't count on the time keeping of a vps.

Use windows directly on real hardware, not a virtual machine.

-giovanni

On 7/27/11, Joao Leme <joaocarlosleme at gmail.com<mailto:joaocarlosleme at gmail.com>> wrote:
> I just set up Freeswitch Precompiled for Windows x64 (
> http://files-sync.freeswitch.org/windows/installer/) on a VPS Windows Server
> 2008.
> Sound is really bad and I'm getting the following:
>
> 2011-07-27 12:33:15.012695 [WARNING] switch_time.c:244 Abnormally large
> timer gap 100953 detected!
> Do you have your kernel timer frequency set to lower than 1,000Hz? You may
> experience audio problems.
>
> also
>
> 2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was
> executed late by 6 seconds 1 heartbeat (core)
> 2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was
> executed late by 4 seconds 2 check_ip (core)
> 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was
> executed late by 8 seconds 1 heartbeat (core)
> 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was
> executed late by 8 seconds 2 check_ip (core)
>
> What can I do?
>
> Also, yesterday, when I first set it up, it was all good.
>
> Thanks,
> John
>

--
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618<tel:%2B39-347-2665618>

_______________________________________________
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<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4e307b2032763660377611!

_______________________________________________
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 marketing at cluecon.com  Thu Jul 28 01:55:16 2011
From: marketing at cluecon.com (Michael Collins)
Date: Wed, 27 Jul 2011 14:55:16 -0700
Subject: [Freeswitch-users] FINAL UPDATE: ClueCon Hotel
Message-ID: <CAKzWOxWR+8HYOmU2qPQ=8ChWaEcUzd6xo2RAazOC9iJ8Sd=QPA@mail.gmail.com>

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

http://cluecon.com/hotel

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

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From msc at freeswitch.org  Thu Jul 28 01:57:22 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Jul 2011 14:57:22 -0700
Subject: [Freeswitch-users] Bad sound quality and Abnormally large timer
 gap 100953 detected!
In-Reply-To: <1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>

On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006 at yahoo.com> wrote:

> I run Asterisk on a virtual machine and it runs quite well even with
> approximately 50 to 60 simultaneous callers and tha'ts even with media
> passing thru it.
>
That's actually pretty impressive. What version of Asterisk do you use? And
do you have to continually make sacrifices to the digital gods to keep them
appeased? :P

-MC
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From frankie.k.yiu at gmail.com  Thu Jul 28 02:09:18 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Wed, 27 Jul 2011 15:09:18 -0700
Subject: [Freeswitch-users] Maximum session question
Message-ID: <CAGV=Xw0+ri1uqjNqO7j+jWQfcWGjnKvhJZU5DZOscVA7VU48uQ@mail.gmail.com>

Hi there,

I was running a test and I have reached the maximum session of 1000.

1000 session(s) since startup
1000 session(s) 0/100
1000 session(s) max
min idle cpu 0.00/99.80

Is that mean the maximum that I can dial is limit to 1000 only (or whatever
the limit that I set) and I have to restart freeswitch?  Or did I forget to
"close" a session?

Thanks,
Frankie
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From msc at freeswitch.org  Thu Jul 28 02:15:01 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Jul 2011 15:15:01 -0700
Subject: [Freeswitch-users] Maximum session question
In-Reply-To: <CAGV=Xw0+ri1uqjNqO7j+jWQfcWGjnKvhJZU5DZOscVA7VU48uQ@mail.gmail.com>
References: <CAGV=Xw0+ri1uqjNqO7j+jWQfcWGjnKvhJZU5DZOscVA7VU48uQ@mail.gmail.com>
Message-ID: <CAKzWOxVOr8f3j__L4+8b1e9G7bahZLdpiPfvXLhiUBP=urL1RQ@mail.gmail.com>

You may have an older version of FS. When did you last update?
-MC

On Wed, Jul 27, 2011 at 3:09 PM, Frankie Yiu <frankie.k.yiu at gmail.com>wrote:

> Hi there,
>
> I was running a test and I have reached the maximum session of 1000.
>
> 1000 session(s) since startup
> 1000 session(s) 0/100
> 1000 session(s) max
> min idle cpu 0.00/99.80
>
> Is that mean the maximum that I can dial is limit to 1000 only (or whatever
> the limit that I set) and I have to restart freeswitch?  Or did I forget to
> "close" a session?
>
> Thanks,
> Frankie
>
> _______________________________________________
> 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 avi at avimarcus.net  Thu Jul 28 02:20:01 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 28 Jul 2011 01:20:01 +0300
Subject: [Freeswitch-users] Bad sound quality and Abnormally large timer
 gap 100953 detected!
In-Reply-To: <CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
Message-ID: <CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>

I'm running FS under Xen provided by Linode. I only run max 10 concurrent
channels in production, but an MOH test sounded good through 200 concurrent
channels.
Linode says the CPU is the most underused resource on their systems...

-Avi


On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>> I run Asterisk on a virtual machine and it runs quite well even with
>> approximately 50 to 60 simultaneous callers and tha'ts even with media
>> passing thru it.
>>
> That's actually pretty impressive. What version of Asterisk do you use? And
> do you have to continually make sacrifices to the digital gods to keep them
> appeased? :P
>
> -MC
>
> _______________________________________________
> 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 lakersman2006 at yahoo.com  Thu Jul 28 02:21:16 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Wed, 27 Jul 2011 15:21:16 -0700 (PDT)
Subject: [Freeswitch-users] Bad sound quality and Abnormally large timer
	gap 100953 detected!
In-Reply-To: <CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
Message-ID: <1311805276.65250.YahooMailNeo@web161015.mail.bf1.yahoo.com>

I am running asterisk1.6.2.15 and the only real issues we encounter are occasional inband DTMF problems, but that could be related to Sonus issues.



________________________________
From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, July 27, 2011 2:57 PM
Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!





On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006 at yahoo.com> wrote:

I run Asterisk on a virtual machine and it runs quite well even with approximately 50 to 60 simultaneous callers and tha'ts even with media passing thru it.
>
That's actually pretty impressive. What version of Asterisk do you use? And do you have to continually make sacrifices to the digital gods to keep them appeased? :P

-MC
_______________________________________________
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
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From lakersman2006 at yahoo.com  Thu Jul 28 02:31:01 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Wed, 27 Jul 2011 15:31:01 -0700 (PDT)
Subject: [Freeswitch-users] Bad sound quality and Abnormally large timer
	gap 100953 detected!
In-Reply-To: <CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
	<CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>
Message-ID: <1311805861.95768.YahooMailNeo@web161013.mail.bf1.yahoo.com>

i forgot to mention that the 50 to 60 calls also are doing majority of g729 transcoding as well which does put th cpu to 85 to 90% full utilization. So don't knock virtual machines.



________________________________
From: Avi Marcus <avi at avimarcus.net>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, July 27, 2011 3:20 PM
Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!


I'm running FS under Xen provided by Linode. I only run max 10 concurrent channels in production, but an MOH test sounded good through 200 concurrent channels.
Linode says the CPU is the most underused resource on their systems...


-Avi


On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc at freeswitch.org> wrote:


>
>
>On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>I run Asterisk on a virtual machine and it runs quite well even with approximately 50 to 60 simultaneous callers and tha'ts even with media passing thru it.
>>
>That's actually pretty impressive. What version of Asterisk do you use? And do you have to continually make sacrifices to the digital gods to keep them appeased? :P
>
>-MC
>_______________________________________________
>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
>
>

_______________________________________________
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
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From frankie.k.yiu at gmail.com  Thu Jul 28 02:45:16 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Wed, 27 Jul 2011 15:45:16 -0700
Subject: [Freeswitch-users] Maximum session question
Message-ID: <CAGV=Xw3z0EY_8mdXQx1YgHiwH3pkoZYhpzYfo=CizR--T-jgwA@mail.gmail.com>

MC,

My freeswitch copy is about 3 months old.  Are you implying that the latest
version should not have this problem (that I can have unlimited session)?

Thanks,
Frankie


You may have an older version of FS. When did you last update?
-MC

On Wed, Jul 27, 2011 at 3:09 PM, Frankie Yiu <frankie.k.yiu at
gmail.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>wrote:

>* Hi there,*>**>* I was running a test and I have reached the maximum session of 1000.*>**>* 1000 session(s) since startup*>* 1000 session(s) 0/100*>* 1000 session(s) max*>* min idle cpu 0.00/99.80*>**>* Is that mean the maximum that I can dial is limit to 1000 only (or whatever*>* the limit that I set) and I have to restart freeswitch?  Or did I forget to*>* "close" a session?*>**>* Thanks,*>* Frankie*
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From bryansmart at bryansmart.com  Thu Jul 28 02:46:41 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Wed, 27 Jul 2011 18:46:41 -0400
Subject: [Freeswitch-users] Discovered problem with PocketSphinx build?
Message-ID: <4097DFE6-DA8E-47BC-A8DC-90AF44E0C575@bryansmart.com>

Tried a build with a latest GIT, with PocketSphinx enabled, and get:

Creating mod_pocketsphinx.la...
/bin/sed: can't read /usr/src/freeswitch/libs/pocketsphinx-0.7/../sphinxbase/src
/libsphinxbase/libsphinxbase.la: No such file or directory
quiet_libtool: link: `/usr/src/freeswitch/libs/pocketsphinx-0.7/../sphinxbase/sr
c/libsphinxbase/libsphinxbase.la' is not a valid libtool archive
cat: .libs/mod_pocketsphinx.log: No such
make[5]: *** [mod_pocketsphinx.la] Error 1

I don't pretend to fully understand the build system, but the makefile seems to be trying to go to libs/sphinxbase for files, when that directory doesn't exist. The currently downloaded version of sphinxbase has the version appended to the directory name, as in sphinxbase-0.7. The Sphinx readme warns that the root should be named sphinxbase.

I did:

ln -s sphinxbase-0.7 sphinxbase

and the build finished without problems.

PocketSphinx works fine on a hardware phone using ULAW, but, when I use a softphone with 16Khz Speex, it doesn't. Instead, I get this

2011-07-27 21:43:14.651803 [WARNING] mod_pocketsphinx.c:147 Can't open speech model /usr/local/freeswitch/grammar/model/wsj1.

The only item in the model directory is communicator. I thought wsj1 was the old model. What in Freeswitch would try to use wsj1 over communicator based on the codec/rate? The mod_pocketsphinx wiki page says that 16Khz is a supported sampling rate, so I'm guessing that there is a special reason why Speex doesn't work, or else that this is a bug.

Bryan

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From avi at avimarcus.net  Thu Jul 28 02:51:52 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 28 Jul 2011 01:51:52 +0300
Subject: [Freeswitch-users] Maximum session question
In-Reply-To: <CAGV=Xw3z0EY_8mdXQx1YgHiwH3pkoZYhpzYfo=CizR--T-jgwA@mail.gmail.com>
References: <CAGV=Xw3z0EY_8mdXQx1YgHiwH3pkoZYhpzYfo=CizR--T-jgwA@mail.gmail.com>
Message-ID: <CALNZuw=fhknKqcSB7jRArHEB3e9E6q3UA3ALt=-4X0oU18wrDA@mail.gmail.com>

The sessions max is just supposed to be how many CONCURRENT sessions are
possible, not how many before you have to restart..

UP 0 years, 12 days, 13 hours, 24 minutes, 25 seconds, 999 milliseconds, 970
microseconds
4999 session(s) since startup
4 session(s) 0/30 <--current channel count
1000 session(s) max
min idle cpu 0.00/100.00

Does it not let me make new calls? Did you hit F10 and do "fsctl pause" by
mistake? "fsctl resume" will change that. The latest GIT checks the
pause/resume status in the F2 "status" call.

-Avi


On Thu, Jul 28, 2011 at 1:45 AM, Frankie Yiu <frankie.k.yiu at gmail.com>wrote:

> MC,
>
> My freeswitch copy is about 3 months old.  Are you implying that the latest
> version should not have this problem (that I can have unlimited session)?
>
> Thanks,
> Frankie
>
>
> You may have an older version of FS. When did you last update?
> -MC
>
> On Wed, Jul 27, 2011 at 3:09 PM, Frankie Yiu <frankie.k.yiu at gmail.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>wrote:
>
> >* Hi there,*>**>* I was running a test and I have reached the maximum session of 1000.*>**>* 1000 session(s) since startup*>* 1000 session(s) 0/100*>* 1000 session(s) max*>* min idle cpu 0.00/99.80*>**>* Is that mean the maximum that I can dial is limit to 1000 only (or whatever*>* the limit that I set) and I have to restart freeswitch?  Or did I forget to*>* "close" a session?*>**>* Thanks,*>* Frankie*
>
>
>
> _______________________________________________
> 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 Joshua.Foshee at LogixCom.com  Thu Jul 28 02:56:20 2011
From: Joshua.Foshee at LogixCom.com (JFoshee)
Date: Wed, 27 Jul 2011 15:56:20 -0700 (PDT)
Subject: [Freeswitch-users] Operator panel or HUD
Message-ID: <1311807380280-6628067.post@n2.nabble.com>

Does anyone know of any software paid or free that is similar to the
following but for Freeswitch?

http://www.getisymphony.com/compare-features/

http://help.fonality.com/HUD/HUD2_vs._HUD3#Features

http://www.astassistant.com/

If not what do you think a budget might be for something like this?

Josh


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Operator-panel-or-HUD-tp6628067p6628067.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From michel.daggelinckx at gmail.com  Thu Jul 28 03:08:33 2011
From: michel.daggelinckx at gmail.com (Michel Daggelinckx)
Date: Thu, 28 Jul 2011 01:08:33 +0200
Subject: [Freeswitch-users] Operator panel or HUD
In-Reply-To: <1311807380280-6628067.post@n2.nabble.com>
References: <1311807380280-6628067.post@n2.nabble.com>
Message-ID: <4E309A71.9080800@gmail.com>

if have used HUD Lite before and its a very handy tool.
for me it beats all existing operator panels.
cos freeswitch uses sockets it must be possible to make a HUD like tool 
that interacts directly with freeswitch, or a module can be created for 
that purpose.


On donderdag 28 juli 2011 0:56:20, JFoshee wrote:
> Does anyone know of any software paid or free that is similar to the
> following but for Freeswitch?
> 
> http://www.getisymphony.com/compare-features/
> 
> http://help.fonality.com/HUD/HUD2_vs._HUD3#Features
> 
> http://www.astassistant.com/
> 
> If not what do you think a budget might be for something like this?
> 
> Josh
> 
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Operator-panel-or-HUD-tp6628067p6628067.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> Free
SWITCH-users mailing list
> FreeSWITCH-users at 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 Jul 28 03:25:15 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Jul 2011 16:25:15 -0700
Subject: [Freeswitch-users] Operator panel or HUD
In-Reply-To: <1311807380280-6628067.post@n2.nabble.com>
References: <1311807380280-6628067.post@n2.nabble.com>
Message-ID: <CAKzWOxURf3Z0T3Vo8tAnX2GApdMYqr21UMuCGk0APH9gv8aqOA@mail.gmail.com>

There is a legitimate need for this kind of functionality in FreeSWITCH. I
would love to see someone take the bull by the horns and start work on
something like this.

-MC

On Wed, Jul 27, 2011 at 3:56 PM, JFoshee <Joshua.Foshee at logixcom.com> wrote:

> Does anyone know of any software paid or free that is similar to the
> following but for Freeswitch?
>
> http://www.getisymphony.com/compare-features/
>
> http://help.fonality.com/HUD/HUD2_vs._HUD3#Features
>
> http://www.astassistant.com/
>
> If not what do you think a budget might be for something like this?
>
> Josh
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Operator-panel-or-HUD-tp6628067p6628067.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> 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 jmesquita at freeswitch.org  Thu Jul 28 03:30:13 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 27 Jul 2011 20:30:13 -0300
Subject: [Freeswitch-users] Operator panel or HUD
In-Reply-To: <CAKzWOxURf3Z0T3Vo8tAnX2GApdMYqr21UMuCGk0APH9gv8aqOA@mail.gmail.com>
References: <1311807380280-6628067.post@n2.nabble.com>
	<CAKzWOxURf3Z0T3Vo8tAnX2GApdMYqr21UMuCGk0APH9gv8aqOA@mail.gmail.com>
Message-ID: <CABH8QX2xoj960GV=Gg57baOzhho9eYne7K7UkVad4JVSoAGn8A@mail.gmail.com>

I have done something based on Qt and if there is enough interest on keeping
the tool alive, I can see if the sponsor is willing to make it open source.
You guys really think there is a big need for that?

Regards,
Jo?o Mesquita



On Wed, Jul 27, 2011 at 8:25 PM, Michael Collins <msc at freeswitch.org> wrote:

> There is a legitimate need for this kind of functionality in FreeSWITCH. I
> would love to see someone take the bull by the horns and start work on
> something like this.
>
> -MC
>
>
> On Wed, Jul 27, 2011 at 3:56 PM, JFoshee <Joshua.Foshee at logixcom.com>wrote:
>
>> Does anyone know of any software paid or free that is similar to the
>> following but for Freeswitch?
>>
>> http://www.getisymphony.com/compare-features/
>>
>> http://help.fonality.com/HUD/HUD2_vs._HUD3#Features
>>
>> http://www.astassistant.com/
>>
>> If not what do you think a budget might be for something like this?
>>
>> Josh
>>
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Operator-panel-or-HUD-tp6628067p6628067.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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 msc at freeswitch.org  Thu Jul 28 03:32:41 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Jul 2011 16:32:41 -0700
Subject: [Freeswitch-users] Operator panel or HUD
In-Reply-To: <CABH8QX2xoj960GV=Gg57baOzhho9eYne7K7UkVad4JVSoAGn8A@mail.gmail.com>
References: <1311807380280-6628067.post@n2.nabble.com>
	<CAKzWOxURf3Z0T3Vo8tAnX2GApdMYqr21UMuCGk0APH9gv8aqOA@mail.gmail.com>
	<CABH8QX2xoj960GV=Gg57baOzhho9eYne7K7UkVad4JVSoAGn8A@mail.gmail.com>
Message-ID: <CAKzWOxXm3Gj+OXBenfRQdHYCdZDCUpS=vkDwt_VEd-wd3zhRvA@mail.gmail.com>

I think the need real, I just don't know how "big" it is. The problem is
that these are all closed source.
-MC

2011/7/27 Jo?o Mesquita <jmesquita at freeswitch.org>

> I have done something based on Qt and if there is enough interest on
> keeping the tool alive, I can see if the sponsor is willing to make it open
> source. You guys really think there is a big need for that?
>
> Regards,
> Jo?o Mesquita
>
>
>
>
> On Wed, Jul 27, 2011 at 8:25 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> There is a legitimate need for this kind of functionality in FreeSWITCH. I
>> would love to see someone take the bull by the horns and start work on
>> something like this.
>>
>> -MC
>>
>>
>> On Wed, Jul 27, 2011 at 3:56 PM, JFoshee <Joshua.Foshee at logixcom.com>wrote:
>>
>>> Does anyone know of any software paid or free that is similar to the
>>> following but for Freeswitch?
>>>
>>> http://www.getisymphony.com/compare-features/
>>>
>>> http://help.fonality.com/HUD/HUD2_vs._HUD3#Features
>>>
>>> http://www.astassistant.com/
>>>
>>> If not what do you think a budget might be for something like this?
>>>
>>> Josh
>>>
>>>
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/Operator-panel-or-HUD-tp6628067p6628067.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> 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
>>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 cmcureau at gmail.com  Thu Jul 28 03:50:57 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Wed, 27 Jul 2011 18:50:57 -0500
Subject: [Freeswitch-users] Operator panel or HUD
In-Reply-To: <CAKzWOxXm3Gj+OXBenfRQdHYCdZDCUpS=vkDwt_VEd-wd3zhRvA@mail.gmail.com>
References: <1311807380280-6628067.post@n2.nabble.com>
	<CAKzWOxURf3Z0T3Vo8tAnX2GApdMYqr21UMuCGk0APH9gv8aqOA@mail.gmail.com>
	<CABH8QX2xoj960GV=Gg57baOzhho9eYne7K7UkVad4JVSoAGn8A@mail.gmail.com>
	<CAKzWOxXm3Gj+OXBenfRQdHYCdZDCUpS=vkDwt_VEd-wd3zhRvA@mail.gmail.com>
Message-ID: <3564462343509373363@unknownmsgid>

I have considered playing around with this before when I was experimenting
with mod_event_socket. I was working with java though. I'd be interested in
picking up some of this...


On Jul 27, 2011, at 6:33 PM, Michael Collins <msc at freeswitch.org> wrote:

I think the need real, I just don't know how "big" it is. The problem is
that these are all closed source.
-MC

2011/7/27 Jo?o Mesquita <jmesquita at freeswitch.org>

> I have done something based on Qt and if there is enough interest on
> keeping the tool alive, I can see if the sponsor is willing to make it open
> source. You guys really think there is a big need for that?
>
> Regards,
> Jo?o Mesquita
>
>
>
>
> On Wed, Jul 27, 2011 at 8:25 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> There is a legitimate need for this kind of functionality in FreeSWITCH. I
>> would love to see someone take the bull by the horns and start work on
>> something like this.
>>
>> -MC
>>
>>
>> On Wed, Jul 27, 2011 at 3:56 PM, JFoshee <Joshua.Foshee at logixcom.com>wrote:
>>
>>> Does anyone know of any software paid or free that is similar to the
>>> following but for Freeswitch?
>>>
>>> http://www.getisymphony.com/compare-features/
>>>
>>> http://help.fonality.com/HUD/HUD2_vs._HUD3#Features
>>>
>>> http://www.astassistant.com/
>>>
>>> If not what do you think a budget might be for something like this?
>>>
>>> Josh
>>>
>>>
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/Operator-panel-or-HUD-tp6628067p6628067.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> 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
>>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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
>
>
_______________________________________________
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
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From michel.daggelinckx at gmail.com  Thu Jul 28 04:27:47 2011
From: michel.daggelinckx at gmail.com (Michel Daggelinckx)
Date: Thu, 28 Jul 2011 02:27:47 +0200
Subject: [Freeswitch-users] Operator panel or HUD
In-Reply-To: <CABH8QX2xoj960GV=Gg57baOzhho9eYne7K7UkVad4JVSoAGn8A@mail.gmail.com>
References: <1311807380280-6628067.post@n2.nabble.com>
	<CAKzWOxURf3Z0T3Vo8tAnX2GApdMYqr21UMuCGk0APH9gv8aqOA@mail.gmail.com>
	<CABH8QX2xoj960GV=Gg57baOzhho9eYne7K7UkVad4JVSoAGn8A@mail.gmail.com>
Message-ID: <4E30AD03.7050901@gmail.com>

It would really benefit the end user who prefers a visual display of 
whats important to him/her.
Also point and click or drag and drop control of some system features 
will ease the use and acceptance of freeswitch.
The most important thing to operators/agents is great functionality 
(already done) and ease of use where i think a good HUD/panel system can 
provide a valuable interface to freeswitch for non tech users.
The techs will probably like the possibility of an API to connect other 
programs to the HUD/panel to exchange data from the phone system for a 
CRM for example

Michel


On 28/07/2011 1:30, Jo?o Mesquita wrote:
> I have done something based on Qt and if there is enough interest on 
> keeping the tool alive, I can see if the sponsor is willing to make it 
> open source. You guys really think there is a big need for that?
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Wed, Jul 27, 2011 at 8:25 PM, Michael Collins <msc at freeswitch.org 
> <mailto:msc at freeswitch.org>> wrote:
>
>     There is a legitimate need for this kind of functionality in
>     FreeSWITCH. I would love to see someone take the bull by the horns
>     and start work on something like this.
>
>     -MC
>
>
>     On Wed, Jul 27, 2011 at 3:56 PM, JFoshee
>     <Joshua.Foshee at logixcom.com <mailto:Joshua.Foshee at logixcom.com>>
>     wrote:
>
>         Does anyone know of any software paid or free that is similar
>         to the
>         following but for Freeswitch?
>
>         http://www.getisymphony.com/compare-features/
>
>         http://help.fonality.com/HUD/HUD2_vs._HUD3#Features
>
>         http://www.astassistant.com/
>
>         If not what do you think a budget might be for something like
>         this?
>
>         Josh
>
>
>         --
>         View this message in context:
>         http://freeswitch-users.2379917.n2.nabble.com/Operator-panel-or-HUD-tp6628067p6628067.html
>         Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>         _______________________________________________
>         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
>         <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
>
>
>
>     _______________________________________________
>     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
>     <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
>
>
>
>
> _______________________________________________
> 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

*Michel Daggelinckx*

The fact that you talk in your head

Doesn't mean your thinking.

Contact me: Google Talk michel.daggelinckx at gmail.com MSN 
micheldaggelinckx at hotmail.com
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From justlikeef at gmail.com  Thu Jul 28 05:51:10 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Wed, 27 Jul 2011 21:51:10 -0400
Subject: [Freeswitch-users] Operator panel or HUD
In-Reply-To: <1311807380280-6628067.post@n2.nabble.com>
References: <1311807380280-6628067.post@n2.nabble.com>
Message-ID: <201107272151.10782.justlikeef@gmail.com>

Blue.box has some of this functionality:

Switchboard gives you a snapshot status of each extension and is near real time.  It also does call control, as does the callcontrol module.

There is a callcenter supervisor module that will show queue status and refreshes every 30 seconds.  It will be near real time soon.

There is a visual voice mail module, but it seems to have some features that need to be addressed.  I understand that significant improvements to the subscriber interface are soon to come.

The permission system is not as granular as these seem to be.



I don't know of anything that has Outlook integration at this point.

On Wednesday 27 July 2011 18:56:20 JFoshee wrote:
> Does anyone know of any software paid or free that is similar to the
> following but for Freeswitch?
> 
> http://www.getisymphony.com/compare-features/
> 
> http://help.fonality.com/HUD/HUD2_vs._HUD3#Features
> 
> http://www.astassistant.com/
> 
> If not what do you think a budget might be for something like this?
> 
> Josh
> 
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Operator-panel-or-HUD-tp6628067p6628067.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> 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 Joshua.Foshee at LogixCom.com  Thu Jul 28 06:08:28 2011
From: Joshua.Foshee at LogixCom.com (JFoshee)
Date: Wed, 27 Jul 2011 19:08:28 -0700 (PDT)
Subject: [Freeswitch-users] Operator panel or HUD
In-Reply-To: <201107272151.10782.justlikeef@gmail.com>
References: <1311807380280-6628067.post@n2.nabble.com>
	<201107272151.10782.justlikeef@gmail.com>
Message-ID: <1311818908755-6628428.post@n2.nabble.com>

Yes we use blue.box today. I like it as it is a great interface. Where are
some of these modules that you are talking about? They are not on the most
current git that I have? Also what the client is really looking for is
something for end user like small client app that can use. I don't mind
going open source or private. The budget would be around 8-9K.

Josh

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Operator-panel-or-HUD-tp6628067p6628428.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From justlikeef at gmail.com  Thu Jul 28 07:05:25 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Wed, 27 Jul 2011 23:05:25 -0400
Subject: [Freeswitch-users] Operator panel or HUD
In-Reply-To: <1311818908755-6628428.post@n2.nabble.com>
References: <1311807380280-6628067.post@n2.nabble.com>
	<201107272151.10782.justlikeef@gmail.com>
	<1311818908755-6628428.post@n2.nabble.com>
Message-ID: <201107272305.25444.justlikeef@gmail.com>

The first round of switchboard made it into the git repo before Whistle pushed everything to the back burner, so it should be there.

My github repo has a whole bunch of patches, additional functionality that I am currently using in production at customer sites that has not been pulled into the official tree, but I have been told for the last 6 months will be done any day.  Unfortunately, testing on my Callcenter modules has been limited, so there may be bugs.  And the UI currently refreshes every 30 seconds, so it is not real time.  That will change as soon as I know where 2600 is going with it so I know what to do.

pyite or janderson on #2600-dev can answer the question on the new client view.  Someone guessed two weeks about three or four weeks ago based on the imminent release of the Whistle version, which has not been released yet.

On Wednesday 27 July 2011 22:08:28 JFoshee wrote:
> Yes we use blue.box today. I like it as it is a great interface. Where are
> some of these modules that you are talking about? They are not on the most
> current git that I have? Also what the client is really looking for is
> something for end user like small client app that can use. I don't mind
> going open source or private. The budget would be around 8-9K.
> 
> Josh
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Operator-panel-or-HUD-tp6628067p6628428.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> 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 justlikeef at gmail.com  Thu Jul 28 07:07:13 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Wed, 27 Jul 2011 23:07:13 -0400
Subject: [Freeswitch-users] Operator panel or HUD
In-Reply-To: <1311818908755-6628428.post@n2.nabble.com>
References: <1311807380280-6628067.post@n2.nabble.com>
	<201107272151.10782.justlikeef@gmail.com>
	<1311818908755-6628428.post@n2.nabble.com>
Message-ID: <201107272307.14058.justlikeef@gmail.com>

One more thing:  I would be happy to intertain a development project, as would 2600 I am sure.  Both of us will do either open source (preferable) or private development, and that budget would be able to get you a fair distance down the road...

On Wednesday 27 July 2011 22:08:28 JFoshee wrote:
> Yes we use blue.box today. I like it as it is a great interface. Where are
> some of these modules that you are talking about? They are not on the most
> current git that I have? Also what the client is really looking for is
> something for end user like small client app that can use. I don't mind
> going open source or private. The budget would be around 8-9K.
> 
> Josh
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Operator-panel-or-HUD-tp6628067p6628428.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> 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 covici at ccs.covici.com  Thu Jul 28 07:29:16 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 27 Jul 2011 23:29:16 -0400
Subject: [Freeswitch-users] Operator panel or HUD
In-Reply-To: <CAKzWOxURf3Z0T3Vo8tAnX2GApdMYqr21UMuCGk0APH9gv8aqOA@mail.gmail.com>
References: <1311807380280-6628067.post@n2.nabble.com>
	<CAKzWOxURf3Z0T3Vo8tAnX2GApdMYqr21UMuCGk0APH9gv8aqOA@mail.gmail.com>
Message-ID: <6598.1311823756@ccs.covici.com>

Doesn't  fusionpbx do all or most of those funactionality.

Michael Collins <msc at freeswitch.org> wrote:

> There is a legitimate need for this kind of functionality in FreeSWITCH. I
> would love to see someone take the bull by the horns and start work on
> something like this.
> 
> -MC
> 
> On Wed, Jul 27, 2011 at 3:56 PM, JFoshee <Joshua.Foshee at logixcom.com> wrote:
> 
> > Does anyone know of any software paid or free that is similar to the
> > following but for Freeswitch?
> >
> > http://www.getisymphony.com/compare-features/
> >
> > http://help.fonality.com/HUD/HUD2_vs._HUD3#Features
> >
> > http://www.astassistant.com/
> >
> > If not what do you think a budget might be for something like this?
> >
> > Josh
> >
> >
> > --
> > View this message in context:
> > http://freeswitch-users.2379917.n2.nabble.com/Operator-panel-or-HUD-tp6628067p6628067.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _______________________________________________
> > 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
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> 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

-- 
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 gmaruzz at gmail.com  Thu Jul 28 11:37:04 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 28 Jul 2011 09:37:04 +0200
Subject: [Freeswitch-users] Bad sound quality and Abnormally large timer
 gap 100953 detected!
In-Reply-To: <1311805861.95768.YahooMailNeo@web161013.mail.bf1.yahoo.com>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
	<CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>
	<1311805861.95768.YahooMailNeo@web161013.mail.bf1.yahoo.com>
Message-ID: <CALXCt0oLi81a_K+zWv3GiOmEv+JYyE7o85XUy4bRNaAA-wnWiw@mail.gmail.com>

Hey guys 'n gal,

the original post is asking for Windows!!!

-giovanni


On 7/28/11, Sam <lakersman2006 at yahoo.com> wrote:
> i forgot to mention that the 50 to 60 calls also are doing majority of g729
> transcoding as well which does put th cpu to 85 to 90% full utilization. So
> don't knock virtual machines.
>
>
>
> ________________________________
> From: Avi Marcus <avi at avimarcus.net>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, July 27, 2011 3:20 PM
> Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer
> gap 100953 detected!
>
>
> I'm running FS under Xen provided by Linode. I only run max 10 concurrent
> channels in production, but an MOH test sounded good through 200 concurrent
> channels.
> Linode says the CPU is the most underused resource on their systems...
>
>
> -Avi
>
>
> On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc at freeswitch.org>
> wrote:
>
>
>>
>>
>>On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>>I run Asterisk on a virtual machine and it runs quite well even with
>> approximately 50 to 60 simultaneous callers and tha'ts even with media
>> passing thru it.
>>>
>>That's actually pretty impressive. What version of Asterisk do you use? And
>> do you have to continually make sacrifices to the digital gods to keep
>> them appeased? :P
>>
>>-MC
>>_______________________________________________
>>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
>>
>>
>
> _______________________________________________
> 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

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From michael.knop at hcu-hamburg.de  Thu Jul 28 11:46:15 2011
From: michael.knop at hcu-hamburg.de (michael knop)
Date: Thu, 28 Jul 2011 09:46:15 +0200
Subject: [Freeswitch-users] FS to a Sonus SIP trunk
In-Reply-To: <CAAtWKeGecHp4683w+-zi-mTNGqjV0M=cMj+jL6GafWFNNi8B-g@mail.gmail.com>
References: <4E2ED4AB.1030305@hcu-hamburg.de>
	<CAAtWKeGecHp4683w+-zi-mTNGqjV0M=cMj+jL6GafWFNNi8B-g@mail.gmail.com>
Message-ID: <4E3113C7.4040107@hcu-hamburg.de>

Yehavi,

unfortunately changing the provider is not an option. If I can?t get 
FreeSWITCH to work with Sonus I can?t switch to FreeSWITCH. Then if have 
to stay using Asterisk.

To me it looks like that its nondeterministic how much time it takes 
until the sound is choppy. Sometimes sound quality is brilliant for 
several minutes. Sometimes the call starts with choppy sound.

/micha


Am 27.07.2011 20:54, schrieb Yehavi Bourvine:
> Hello Micha,
>    How much time does it take until the sound is choppy?
> We have been connected to Sonus provider up to a week ago. Incoming
> calls started being choppy after about 15 minutes (outgoing calls were
> ok). We also had inconsistent problems with DTMF. We ended it by
> changing a supplier...
>                       __Yehavi:
>
> 2011/7/26 michael knop <michael.knop at hcu-hamburg.de
> <mailto:michael.knop at hcu-hamburg.de>>
>
>     Hi all!
>
>     I?m trying to connect my FS to a Sonus SIP trunk. I followed the
>     instruction at
>
>     http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus
>
>     but it did not work. At the beginning of a call voice quality is good.
>     After a while it changes to choppy.
>
>     I don?t know if it?s the same problem: When I call the Tetris extension
>     via Sonus SIP trunk the sound is too fast and I?m getting log entries
>     like the following one:
>
>     [...]
>     2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME
>     not supported, changing our end from 20 to 10
>     2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from
>     PCMA at 20ms@8000hz to PCMA at 10ms@8000hz
>     2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global
>     timer resolution to 10ms to handle interval 10
>     2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer
>     [soft] 80 bytes per 10ms
>     2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec
>     sofia/external/+4940... at 193...:5060 PCMA/8000 10 ms 80 samples 64000
>     bits
>     2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write
>     Buffer at 160 bytes to accommodate 320->160
>     [...]
>
>     This problem is fixed by adding the following line to
>     conf/sip_profiles/external.xml:
>
>     <param name="rtp-autofix-timing" value="false"/>
>
>     Any hints?
>
>     /micha
>
>     _______________________________________________
>     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
>     <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
>
>
>
> _______________________________________________
> 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 tomasz at kopacki.eu  Thu Jul 28 11:52:13 2011
From: tomasz at kopacki.eu (Tomasz Kopacki)
Date: Thu, 28 Jul 2011 07:52:13 +0000
Subject: [Freeswitch-users] partial xml reloading
Message-ID: <443EC226AAEABB48B58CAF9D56D80AB4AB0721@hektor.dom.local>

Hi,
i have big dialplan(100k+ extensions) divided into several files. Is there a way to reload only one of those files ? reloading whole dp takes lots of time.

BR,
Tomek

From gcd at i.ph  Thu Jul 28 12:04:54 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Thu, 28 Jul 2011 16:04:54 +0800
Subject: [Freeswitch-users] partial xml reloading
In-Reply-To: <443EC226AAEABB48B58CAF9D56D80AB4AB0721@hektor.dom.local>
References: <443EC226AAEABB48B58CAF9D56D80AB4AB0721@hektor.dom.local>
Message-ID: <CAHKqMb5LO+XRKQCZHP2mc4yG2Vgs26cixxVmG9Cw20JELrtCjg@mail.gmail.com>

hello tomasz,

it's better to use mod_lcr than sifting 100k entries. let an SQL server do
the lookup.

-nandy
On Thu, Jul 28, 2011 at 3:52 PM, Tomasz Kopacki <tomasz at kopacki.eu> wrote:

> Hi,
> i have big dialplan(100k+ extensions) divided into several files. Is there
> a way to reload only one of those files ? reloading whole dp takes lots of
> time.
>
> BR,
> Tomek
> _______________________________________________
> 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
>
-------------- next part --------------
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From steveayre at gmail.com  Thu Jul 28 12:48:51 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 28 Jul 2011 09:48:51 +0100
Subject: [Freeswitch-users] partial xml reloading
In-Reply-To: <CAHKqMb5LO+XRKQCZHP2mc4yG2Vgs26cixxVmG9Cw20JELrtCjg@mail.gmail.com>
References: <443EC226AAEABB48B58CAF9D56D80AB4AB0721@hektor.dom.local>
	<CAHKqMb5LO+XRKQCZHP2mc4yG2Vgs26cixxVmG9Cw20JELrtCjg@mail.gmail.com>
Message-ID: <CAFiqYum9srrO8fSuzxYZ4zAe4fVUa9gmdkyi9+jEQoo901Ktjg@mail.gmail.com>

Or mod_xml_curl

On 28 July 2011 09:04, Nandy Dagondon <gcd at i.ph> wrote:

> hello tomasz,
>
> it's better to use mod_lcr than sifting 100k entries. let an SQL server do
> the lookup.
>
> -nandy
>
> On Thu, Jul 28, 2011 at 3:52 PM, Tomasz Kopacki <tomasz at kopacki.eu> wrote:
>
>> Hi,
>> i have big dialplan(100k+ extensions) divided into several files. Is there
>> a way to reload only one of those files ? reloading whole dp takes lots of
>> time.
>>
>> BR,
>> Tomek
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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
>
>
-------------- next part --------------
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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110728/985ae01f/attachment.html 

From peter.olsson at visionutveckling.se  Thu Jul 28 13:18:23 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 28 Jul 2011 11:18:23 +0200
Subject: [Freeswitch-users] Bad sound quality and Abnormally large timer
 gap 100953 detected!
In-Reply-To: <CALXCt0oLi81a_K+zWv3GiOmEv+JYyE7o85XUy4bRNaAA-wnWiw@mail.gmail.com>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
	<CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>
	<1311805861.95768.YahooMailNeo@web161013.mail.bf1.yahoo.com>,
	<CALXCt0oLi81a_K+zWv3GiOmEv+JYyE7o85XUy4bRNaAA-wnWiw@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F95@cooper>

I feel I need to explain my last statement a little further :)

First of all, you can run FS on virtual machines (even though I never recommend it), but you need to be in control of the virtual environment, or totally trust the ones controlling it.

There are two basic problems when running on virtual hardware - timing and network I/O. For timing to work you need to make sure that the one controlling the virtual environment NEVER give out more CPU cores then there are available on the hardware. If they give out more, and your machine need to share resources with another machine, you will be in trouble. As soon as the other virtual machine starts to load the CPU, your machine will get bad timing. Also - usually a one core VM will get better timing then a two+ core VM.

Also, to be safe on the network side, you need your own NIC, that is not shared with other machines.

If you feel you can control this, you might be successful with the setup, if you can't control this, you're either lucky (and things work anyway), or you'll get bad audio.

Also, just passing RTP for normal call isn't the most critical issue (you don't even have to use a timer there), the most critical timing issues will occur in conferences or when using a local IVR on the machine.

I've done lots of testing for this in my lab, since many of my customers wants to run our system (which uses FS for IVR/conferencing) on ESXi virtual hosts, and I've found the above to be quite critical to succeed with your virtual implementation.

/Peter

____________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Giovanni Maruzzelli [gmaruzz at gmail.com]
Skickat: den 28 juli 2011 09:37
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!

Hey guys 'n gal,

the original post is asking for Windows!!!

-giovanni


On 7/28/11, Sam <lakersman2006 at yahoo.com> wrote:
> i forgot to mention that the 50 to 60 calls also are doing majority of g729
> transcoding as well which does put th cpu to 85 to 90% full utilization. So
> don't knock virtual machines.
>
>
>
> ________________________________
> From: Avi Marcus <avi at avimarcus.net>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, July 27, 2011 3:20 PM
> Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer
> gap 100953 detected!
>
>
> I'm running FS under Xen provided by Linode. I only run max 10 concurrent
> channels in production, but an MOH test sounded good through 200 concurrent
> channels.
> Linode says the CPU is the most underused resource on their systems...
>
>
> -Avi
>
>
> On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc at freeswitch.org>
> wrote:
>
>
>>
>>
>>On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>>I run Asterisk on a virtual machine and it runs quite well even with
>> approximately 50 to 60 simultaneous callers and tha'ts even with media
>> passing thru it.
>>>
>>That's actually pretty impressive. What version of Asterisk do you use? And
>> do you have to continually make sacrifices to the digital gods to keep
>> them appeased? :P
>>
>>-MC
>>_______________________________________________
>>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
>>
>>
>
> _______________________________________________
> 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

--
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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:4e31126d32765770742302!



From gmaruzz at celliax.org  Thu Jul 28 13:33:28 2011
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 28 Jul 2011 11:33:28 +0200
Subject: [Freeswitch-users] Bad sound quality and Abnormally large timer
 gap 100953 detected!
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F95@cooper>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
	<CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>
	<1311805861.95768.YahooMailNeo@web161013.mail.bf1.yahoo.com>
	<CALXCt0oLi81a_K+zWv3GiOmEv+JYyE7o85XUy4bRNaAA-wnWiw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F95@cooper>
Message-ID: <CALXCt0powR2Zp_D=3ywf6h6o13GjjbD_ndAEaNRB1C3yRJMvMA@mail.gmail.com>

Peter,

that's a pretty clear explanation, and answers many frequently asked questions.

Can you please add it to the wiki (eg: "FS on virtual machines", and
add links to it from "amazon" and "aws" related pages) ?

I'm sure lot of people will find it interesting, and hopefully some
other people (and/or you) will add more info and experiences to it.

-giovanni

On 7/28/11, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> I feel I need to explain my last statement a little further :)
>
> First of all, you can run FS on virtual machines (even though I never
> recommend it), but you need to be in control of the virtual environment, or
> totally trust the ones controlling it.
>
> There are two basic problems when running on virtual hardware - timing and
> network I/O. For timing to work you need to make sure that the one
> controlling the virtual environment NEVER give out more CPU cores then there
> are available on the hardware. If they give out more, and your machine need
> to share resources with another machine, you will be in trouble. As soon as
> the other virtual machine starts to load the CPU, your machine will get bad
> timing. Also - usually a one core VM will get better timing then a two+ core
> VM.
>
> Also, to be safe on the network side, you need your own NIC, that is not
> shared with other machines.
>
> If you feel you can control this, you might be successful with the setup, if
> you can't control this, you're either lucky (and things work anyway), or
> you'll get bad audio.
>
> Also, just passing RTP for normal call isn't the most critical issue (you
> don't even have to use a timer there), the most critical timing issues will
> occur in conferences or when using a local IVR on the machine.
>
> I've done lots of testing for this in my lab, since many of my customers
> wants to run our system (which uses FS for IVR/conferencing) on ESXi virtual
> hosts, and I've found the above to be quite critical to succeed with your
> virtual implementation.
>
> /Peter
>
> ____________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Giovanni Maruzzelli
> [gmaruzz at gmail.com]
> Skickat: den 28 juli 2011 09:37
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer
> gap 100953 detected!
>
> Hey guys 'n gal,
>
> the original post is asking for Windows!!!
>
> -giovanni
>
>
> On 7/28/11, Sam <lakersman2006 at yahoo.com> wrote:
>> i forgot to mention that the 50 to 60 calls also are doing majority of
>> g729
>> transcoding as well which does put th cpu to 85 to 90% full utilization.
>> So
>> don't knock virtual machines.
>>
>>
>>
>> ________________________________
>> From: Avi Marcus <avi at avimarcus.net>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Wednesday, July 27, 2011 3:20 PM
>> Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large
>> timer
>> gap 100953 detected!
>>
>>
>> I'm running FS under Xen provided by Linode. I only run max 10 concurrent
>> channels in production, but an MOH test sounded good through 200
>> concurrent
>> channels.
>> Linode says the CPU is the most underused resource on their systems...
>>
>>
>> -Avi
>>
>>
>> On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>>
>>
>>>
>>>
>>>On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>
>>>I run Asterisk on a virtual machine and it runs quite well even with
>>> approximately 50 to 60 simultaneous callers and tha'ts even with media
>>> passing thru it.
>>>>
>>>That's actually pretty impressive. What version of Asterisk do you use?
>>> And
>>> do you have to continually make sacrifices to the digital gods to keep
>>> them appeased? :P
>>>
>>>-MC
>>>_______________________________________________
>>>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
>>>
>>>
>>
>> _______________________________________________
>> 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
>
> --
> Sent from my mobile device
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:4e31126d32765770742302!
>
>
> _______________________________________________
> 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
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From michael.knop at hcu-hamburg.de  Thu Jul 28 14:14:50 2011
From: michael.knop at hcu-hamburg.de (michael knop)
Date: Thu, 28 Jul 2011 12:14:50 +0200
Subject: [Freeswitch-users] FS to a Sonus SIP trunk
In-Reply-To: <4E2ED4AB.1030305@hcu-hamburg.de>
References: <4E2ED4AB.1030305@hcu-hamburg.de>
Message-ID: <4E31369A.70605@hcu-hamburg.de>

Update:

Call starts with good sound quality. After the following log message 
sound is choppy:

2011-07-28 12:07:22.639151 [DEBUG] sofia.c:5094 Duplicate SDP
v=0
o=Sonus_UAC 8739 8900 IN IP4 XXX.XXX.XXX.XXX
s=SIP Media Capabilities
c=IN IP4 YYY.YYY.YYY.YYY
t=0 0
m=audio 20320 RTP/AVP 8 0 18 100
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=maxptime:10

/micha

Am 26.07.2011 16:52, schrieb michael knop:
> Hi all!
>
> I?m trying to connect my FS to a Sonus SIP trunk. I followed the
> instruction at
>
>     http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus
>
> but it did not work. At the beginning of a call voice quality is good.
> After a while it changes to choppy.
>
> I don?t know if it?s the same problem: When I call the Tetris extension
> via Sonus SIP trunk the sound is too fast and I?m getting log entries
> like the following one:
>
> [...]
> 2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME
> not supported, changing our end from 20 to 10
> 2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from
> PCMA at 20ms@8000hz to PCMA at 10ms@8000hz
> 2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global
> timer resolution to 10ms to handle interval 10
> 2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer
> [soft] 80 bytes per 10ms
> 2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec
> sofia/external/+4940... at 193...:5060 PCMA/8000 10 ms 80 samples 64000 bits
> 2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write
> Buffer at 160 bytes to accommodate 320->160
> [...]
>
> This problem is fixed by adding the following line to
> conf/sip_profiles/external.xml:
>
> <param name="rtp-autofix-timing" value="false"/>
>
> Any hints?
>
> /micha


From avi at avimarcus.net  Thu Jul 28 15:24:01 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 28 Jul 2011 14:24:01 +0300
Subject: [Freeswitch-users] partial xml reloading
In-Reply-To: <CAFiqYum9srrO8fSuzxYZ4zAe4fVUa9gmdkyi9+jEQoo901Ktjg@mail.gmail.com>
References: <443EC226AAEABB48B58CAF9D56D80AB4AB0721@hektor.dom.local>
	<CAHKqMb5LO+XRKQCZHP2mc4yG2Vgs26cixxVmG9Cw20JELrtCjg@mail.gmail.com>
	<CAFiqYum9srrO8fSuzxYZ4zAe4fVUa9gmdkyi9+jEQoo901Ktjg@mail.gmail.com>
Message-ID: <CALNZuw=GhirkoSQA5LGG0z3_9qsBDvrH+Uncs63U0fARPaUegw@mail.gmail.com>

If your situation is "simple" e.g. an incoming DID -> a particular endpoint,
then using mod_odbc_query or a simple lua script sounds good.
If it's "just" this, then I would think mod_xml_curl is very much overkill,
in terms of complexity of setup and that ALL dialplan routing needs to hit
the XML first.

-Avi


On Thu, Jul 28, 2011 at 11:48 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Or mod_xml_curl
>
>
> On 28 July 2011 09:04, Nandy Dagondon <gcd at i.ph> wrote:
>
>> hello tomasz,
>>
>> it's better to use mod_lcr than sifting 100k entries. let an SQL server do
>> the lookup.
>>
>> -nandy
>>
>>  On Thu, Jul 28, 2011 at 3:52 PM, Tomasz Kopacki <tomasz at kopacki.eu>wrote:
>>
>>> Hi,
>>> i have big dialplan(100k+ extensions) divided into several files. Is
>>> there a way to reload only one of those files ? reloading whole dp takes
>>> lots of time.
>>>
>>> BR,
>>> Tomek
>>> _______________________________________________
>>> 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
>>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 bryansmart at bryansmart.com  Thu Jul 28 15:52:33 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Thu, 28 Jul 2011 07:52:33 -0400
Subject: [Freeswitch-users] Bad sound quality and Abnormally large timer
 gap 100953 detected!
In-Reply-To: <CALXCt0powR2Zp_D=3ywf6h6o13GjjbD_ndAEaNRB1C3yRJMvMA@mail.gmail.com>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
	<CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>
	<1311805861.95768.YahooMailNeo@web161013.mail.bf1.yahoo.com>
	<CALXCt0oLi81a_K+zWv3GiOmEv+JYyE7o85XUy4bRNaAA-wnWiw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F95@cooper>
	<CALXCt0powR2Zp_D=3ywf6h6o13GjjbD_ndAEaNRB1C3yRJMvMA@mail.gmail.com>
Message-ID: <04020B71-9CAA-458A-98D0-189A82E8F5F1@bryansmart.com>

> Peter, thanks for those details.

What sort of trade-offs, if any, are useful to improve audio for conferences and IVRs on VMs? Will increasing the interval setting for a conference help? What about using larger frames for codecs? I know that latency, beyond a point, is a serious quality issue of its own. Still, do you know what compromises provide the best improvement?

Bryan

> On 7/28/11, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
>> I feel I need to explain my last statement a little further :)
>> 
>> First of all, you can run FS on virtual machines (even though I never
>> recommend it), but you need to be in control of the virtual environment, or
>> totally trust the ones controlling it.
>> 
>> There are two basic problems when running on virtual hardware - timing and
>> network I/O. For timing to work you need to make sure that the one
>> controlling the virtual environment NEVER give out more CPU cores then there
>> are available on the hardware. If they give out more, and your machine need
>> to share resources with another machine, you will be in trouble. As soon as
>> the other virtual machine starts to load the CPU, your machine will get bad
>> timing. Also - usually a one core VM will get better timing then a two+ core
>> VM.
>> 
>> Also, to be safe on the network side, you need your own NIC, that is not
>> shared with other machines.
>> 
>> If you feel you can control this, you might be successful with the setup, if
>> you can't control this, you're either lucky (and things work anyway), or
>> you'll get bad audio.
>> 
>> Also, just passing RTP for normal call isn't the most critical issue (you
>> don't even have to use a timer there), the most critical timing issues will
>> occur in conferences or when using a local IVR on the machine.
>> 
>> I've done lots of testing for this in my lab, since many of my customers
>> wants to run our system (which uses FS for IVR/conferencing) on ESXi virtual
>> hosts, and I've found the above to be quite critical to succeed with your
>> virtual implementation.
>> 
>> /Peter
>> 
>> ____________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Giovanni Maruzzelli
>> [gmaruzz at gmail.com]
>> Skickat: den 28 juli 2011 09:37
>> Till: FreeSWITCH Users Help
>> ?mne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer
>> gap 100953 detected!
>> 
>> Hey guys 'n gal,
>> 
>> the original post is asking for Windows!!!
>> 
>> -giovanni
>> 
>> 
>> On 7/28/11, Sam <lakersman2006 at yahoo.com> wrote:
>>> i forgot to mention that the 50 to 60 calls also are doing majority of
>>> g729
>>> transcoding as well which does put th cpu to 85 to 90% full utilization.
>>> So
>>> don't knock virtual machines.
>>> 
>>> 
>>> 
>>> ________________________________
>>> From: Avi Marcus <avi at avimarcus.net>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Sent: Wednesday, July 27, 2011 3:20 PM
>>> Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large
>>> timer
>>> gap 100953 detected!
>>> 
>>> 
>>> I'm running FS under Xen provided by Linode. I only run max 10 concurrent
>>> channels in production, but an MOH test sounded good through 200
>>> concurrent
>>> channels.
>>> Linode says the CPU is the most underused resource on their systems...
>>> 
>>> 
>>> -Avi
>>> 
>>> 
>>> On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc at freeswitch.org>
>>> wrote:
>>> 
>>> 
>>>> 
>>>> 
>>>> On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>> 
>>>> I run Asterisk on a virtual machine and it runs quite well even with
>>>> approximately 50 to 60 simultaneous callers and tha'ts even with media
>>>> passing thru it.
>>>>> 
>>>> That's actually pretty impressive. What version of Asterisk do you use?
>>>> And
>>>> do you have to continually make sacrifices to the digital gods to keep
>>>> them appeased? :P
>>>> 
>>>> -MC
>>>> _______________________________________________
>>>> 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
>>>> 
>>>> 
>>> 
>>> _______________________________________________
>>> 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
>> 
>> --
>> Sent from my mobile device
>> 
>> Sincerely,
>> 
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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:4e31126d32765770742302!
>> 
>> 
>> _______________________________________________
>> 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
>> 
> 
> -- 
> Sent from my mobile device
> 
> Sincerely,
> 
> Giovanni Maruzzelli
> Cell : +39-347-2665618
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 bryansmart at bryansmart.com  Thu Jul 28 16:06:16 2011
From: bryansmart at bryansmart.com (Bryan Smart)
Date: Thu, 28 Jul 2011 08:06:16 -0400
Subject: [Freeswitch-users] Discovered problem with PocketSphinx build?
In-Reply-To: <4097DFE6-DA8E-47BC-A8DC-90AF44E0C575@bryansmart.com>
References: <4097DFE6-DA8E-47BC-A8DC-90AF44E0C575@bryansmart.com>
Message-ID: <CB9CFB6B-F4C5-409D-AD83-D01D390CB6EB@bryansmart.com>

I partially figured out my problem.

Communicator only works for 8Khz. For 16Khz, you need WSJ1. I wonder why it isn't included with Freeswitch?

WSJ1 is available from this page:
http://www.speech.cs.cmu.edu/sphinx/models/

Direct link for the latest release:
http://www.speech.cs.cmu.edu/sphinx/models/wsj_jan2008/wsj_all_sc_5000_20080401.tar.gz

Rename the extracted folder to wsj1, and place in grammar/model.

For FS to use it, in pocketsphinx.conf.xml, set wideband-model:

<param name="wideband-model" value="wsj1"/>

ASR works fine for me now at 16Khz.

Should I add this to the wiki? Maybe FS could have automatically installed wsj1, but I was clueless about the required build option.

Bryan

On Jul 27, 2011, at 6:46 PM, Bryan Smart wrote:

Tried a build with a latest GIT, with PocketSphinx enabled, and get:

Creating mod_pocketsphinx.la...
/bin/sed: can't read /usr/src/freeswitch/libs/pocketsphinx-0.7/../sphinxbase/src
/libsphinxbase/libsphinxbase.la: No such file or directory
quiet_libtool: link: `/usr/src/freeswitch/libs/pocketsphinx-0.7/../sphinxbase/sr
c/libsphinxbase/libsphinxbase.la' is not a valid libtool archive
cat: .libs/mod_pocketsphinx.log: No such
make[5]: *** [mod_pocketsphinx.la] Error 1

I don't pretend to fully understand the build system, but the makefile seems to be trying to go to libs/sphinxbase for files, when that directory doesn't exist. The currently downloaded version of sphinxbase has the version appended to the directory name, as in sphinxbase-0.7. The Sphinx readme warns that the root should be named sphinxbase.

I did:

ln -s sphinxbase-0.7 sphinxbase

and the build finished without problems.

PocketSphinx works fine on a hardware phone using ULAW, but, when I use a softphone with 16Khz Speex, it doesn't. Instead, I get this

2011-07-27 21:43:14.651803 [WARNING] mod_pocketsphinx.c:147 Can't open speech model /usr/local/freeswitch/grammar/model/wsj1.

The only item in the model directory is communicator. I thought wsj1 was the old model. What in Freeswitch would try to use wsj1 over communicator based on the codec/rate? The mod_pocketsphinx wiki page says that 16Khz is a supported sampling rate, so I'm guessing that there is a special reason why Speex doesn't work, or else that this is a bug.

Bryan


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From Nabble at slickdeals.endjunk.com  Thu Jul 28 18:01:16 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 28 Jul 2011 07:01:16 -0700 (PDT)
Subject: [Freeswitch-users] Maximum session question
In-Reply-To: <CALNZuw=fhknKqcSB7jRArHEB3e9E6q3UA3ALt=-4X0oU18wrDA@mail.gmail.com>
References: <CAGV=Xw3z0EY_8mdXQx1YgHiwH3pkoZYhpzYfo=CizR--T-jgwA@mail.gmail.com>
	<CALNZuw=fhknKqcSB7jRArHEB3e9E6q3UA3ALt=-4X0oU18wrDA@mail.gmail.com>
Message-ID: <1311861676781-6629963.post@n2.nabble.com>


Avi Marcus-2 wrote:
> 
> The sessions max is just supposed to be how many CONCURRENT sessions are
> possible, not how many before you have to restart..
Does that mean my Seagate DockStar can handle 1,000 CONCURRENT sessions as
shown below?

root at DockStar:/# fs_cli 
            _____ ____     ____ _     ___            
           |  ___/ ___|   / ___| |   |_ _|           
           | |_  \___ \  | |   | |    | |            
           |  _|  ___) | | |___| |___ | |            
           |_|   |____/   \____|_____|___|           

*******************************************************
* Anthony Minessale II, Ken Rice, Michael Jerris      *
* FreeSWITCH (http://www.freeswitch.org)              *
* Paypal Donations Appreciated: paypal at freeswitch.org *
* Brought to you by ClueCon http://www.cluecon.com/   *
*******************************************************

Type /help <enter> to see a list of commands


+OK log level  [7]
freeswitch at internal> status

UP 0 years, 0 days, 0 hours, 0 minutes, 13 seconds, 812 milliseconds, 275
microseconds
0 session(s) since startup
0 session(s) 0/30
1000 session(s) max
min idle cpu 0.00/0.00

freeswitch at internal> version
FreeSWITCH Version 1.0.head (git-9795dd2 2011-03-26 11-07-34 -0500)

freeswitch at internal>


-----
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/Maximum-session-question-tp6628038p6629963.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From avi at avimarcus.net  Thu Jul 28 18:07:33 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 28 Jul 2011 17:07:33 +0300
Subject: [Freeswitch-users] Maximum session question
In-Reply-To: <1311861676781-6629963.post@n2.nabble.com>
References: <CAGV=Xw3z0EY_8mdXQx1YgHiwH3pkoZYhpzYfo=CizR--T-jgwA@mail.gmail.com>
	<CALNZuw=fhknKqcSB7jRArHEB3e9E6q3UA3ALt=-4X0oU18wrDA@mail.gmail.com>
	<1311861676781-6629963.post@n2.nabble.com>
Message-ID: <CALNZuwnoSEmcCD=EvVkgwhuk4FfV3cWXfVj=EaOyz2n7dn6+Gw@mail.gmail.com>

hehe. Once the CPU usage starts building up, I think FS will scale that down
automatically.
-Avi


On Thu, Jul 28, 2011 at 5:01 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> Avi Marcus-2 wrote:
> >
> > The sessions max is just supposed to be how many CONCURRENT sessions are
> > possible, not how many before you have to restart..
> Does that mean my Seagate DockStar can handle 1,000 CONCURRENT sessions as
> shown below?
>
> root at DockStar:/# fs_cli
>            _____ ____     ____ _     ___
>           |  ___/ ___|   / ___| |   |_ _|
>           | |_  \___ \  | |   | |    | |
>           |  _|  ___) | | |___| |___ | |
>           |_|   |____/   \____|_____|___|
>
> *******************************************************
> * Anthony Minessale II, Ken Rice, Michael Jerris      *
> * FreeSWITCH (http://www.freeswitch.org)              *
> * Paypal Donations Appreciated: paypal at freeswitch.org *
> * Brought to you by ClueCon http://www.cluecon.com/   *
> *******************************************************
>
> Type /help <enter> to see a list of commands
>
>
> +OK log level  [7]
> freeswitch at internal> status
>
> UP 0 years, 0 days, 0 hours, 0 minutes, 13 seconds, 812 milliseconds, 275
> microseconds
> 0 session(s) since startup
> 0 session(s) 0/30
> 1000 session(s) max
> min idle cpu 0.00/0.00
>
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (git-9795dd2 2011-03-26 11-07-34 -0500)
>
> freeswitch at internal>
>
>
> -----
> 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/Maximum-session-question-tp6628038p6629963.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> 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
>
-------------- next part --------------
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From Nabble at slickdeals.endjunk.com  Thu Jul 28 18:23:46 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 28 Jul 2011 07:23:46 -0700 (PDT)
Subject: [Freeswitch-users] Maximum session question
In-Reply-To: <CALNZuwnoSEmcCD=EvVkgwhuk4FfV3cWXfVj=EaOyz2n7dn6+Gw@mail.gmail.com>
References: <CAGV=Xw3z0EY_8mdXQx1YgHiwH3pkoZYhpzYfo=CizR--T-jgwA@mail.gmail.com>
	<CALNZuw=fhknKqcSB7jRArHEB3e9E6q3UA3ALt=-4X0oU18wrDA@mail.gmail.com>
	<1311861676781-6629963.post@n2.nabble.com>
	<CALNZuwnoSEmcCD=EvVkgwhuk4FfV3cWXfVj=EaOyz2n7dn6+Gw@mail.gmail.com>
Message-ID: <1311863026945-6630048.post@n2.nabble.com>


Avi Marcus-2 wrote:
> 
> hehe. Once the CPU usage starts building up, I think FS will scale that
> down
> automatically.
> -Avi
This makes more sense. 

-----
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/Maximum-session-question-tp6628038p6630048.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From peter.olsson at visionutveckling.se  Thu Jul 28 19:02:28 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 28 Jul 2011 17:02:28 +0200
Subject: [Freeswitch-users] Bad sound quality and Abnormally large timer
 gap 100953 detected!
In-Reply-To: <CALXCt0powR2Zp_D=3ywf6h6o13GjjbD_ndAEaNRB1C3yRJMvMA@mail.gmail.com>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
	<CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>
	<1311805861.95768.YahooMailNeo@web161013.mail.bf1.yahoo.com>
	<CALXCt0oLi81a_K+zWv3GiOmEv+JYyE7o85XUy4bRNaAA-wnWiw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F95@cooper>,
	<CALXCt0powR2Zp_D=3ywf6h6o13GjjbD_ndAEaNRB1C3yRJMvMA@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F96@cooper>

Thanks,

I will make sure to add this to the wiki, I also have some more detailed information after all my testing, so I will make sure to get it all upp on the wiki.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Giovanni Maruzzelli [gmaruzz at celliax.org]
Skickat: den 28 juli 2011 11:33
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!

Peter,

that's a pretty clear explanation, and answers many frequently asked questions.

Can you please add it to the wiki (eg: "FS on virtual machines", and
add links to it from "amazon" and "aws" related pages) ?

I'm sure lot of people will find it interesting, and hopefully some
other people (and/or you) will add more info and experiences to it.

-giovanni

On 7/28/11, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> I feel I need to explain my last statement a little further :)
>
> First of all, you can run FS on virtual machines (even though I never
> recommend it), but you need to be in control of the virtual environment, or
> totally trust the ones controlling it.
>
> There are two basic problems when running on virtual hardware - timing and
> network I/O. For timing to work you need to make sure that the one
> controlling the virtual environment NEVER give out more CPU cores then there
> are available on the hardware. If they give out more, and your machine need
> to share resources with another machine, you will be in trouble. As soon as
> the other virtual machine starts to load the CPU, your machine will get bad
> timing. Also - usually a one core VM will get better timing then a two+ core
> VM.
>
> Also, to be safe on the network side, you need your own NIC, that is not
> shared with other machines.
>
> If you feel you can control this, you might be successful with the setup, if
> you can't control this, you're either lucky (and things work anyway), or
> you'll get bad audio.
>
> Also, just passing RTP for normal call isn't the most critical issue (you
> don't even have to use a timer there), the most critical timing issues will
> occur in conferences or when using a local IVR on the machine.
>
> I've done lots of testing for this in my lab, since many of my customers
> wants to run our system (which uses FS for IVR/conferencing) on ESXi virtual
> hosts, and I've found the above to be quite critical to succeed with your
> virtual implementation.
>
> /Peter
>
> ____________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Giovanni Maruzzelli
> [gmaruzz at gmail.com]
> Skickat: den 28 juli 2011 09:37
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer
> gap 100953 detected!
>
> Hey guys 'n gal,
>
> the original post is asking for Windows!!!
>
> -giovanni
>
>
> On 7/28/11, Sam <lakersman2006 at yahoo.com> wrote:
>> i forgot to mention that the 50 to 60 calls also are doing majority of
>> g729
>> transcoding as well which does put th cpu to 85 to 90% full utilization.
>> So
>> don't knock virtual machines.
>>
>>
>>
>> ________________________________
>> From: Avi Marcus <avi at avimarcus.net>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Wednesday, July 27, 2011 3:20 PM
>> Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large
>> timer
>> gap 100953 detected!
>>
>>
>> I'm running FS under Xen provided by Linode. I only run max 10 concurrent
>> channels in production, but an MOH test sounded good through 200
>> concurrent
>> channels.
>> Linode says the CPU is the most underused resource on their systems...
>>
>>
>> -Avi
>>
>>
>> On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>>
>>
>>>
>>>
>>>On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>
>>>I run Asterisk on a virtual machine and it runs quite well even with
>>> approximately 50 to 60 simultaneous callers and tha'ts even with media
>>> passing thru it.
>>>>
>>>That's actually pretty impressive. What version of Asterisk do you use?
>>> And
>>> do you have to continually make sacrifices to the digital gods to keep
>>> them appeased? :P
>>>
>>>-MC
>>>_______________________________________________
>>>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
>>>
>>>
>>
>> _______________________________________________
>> 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
>
> --
> Sent from my mobile device
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> 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
>

--
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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:4e312d4a32762001915859!



From peter.olsson at visionutveckling.se  Thu Jul 28 19:16:05 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 28 Jul 2011 17:16:05 +0200
Subject: [Freeswitch-users] Bad sound quality and Abnormally large timer
 gap 100953 detected!
In-Reply-To: <04020B71-9CAA-458A-98D0-189A82E8F5F1@bryansmart.com>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
	<CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>
	<1311805861.95768.YahooMailNeo@web161013.mail.bf1.yahoo.com>
	<CALXCt0oLi81a_K+zWv3GiOmEv+JYyE7o85XUy4bRNaAA-wnWiw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F95@cooper>
	<CALXCt0powR2Zp_D=3ywf6h6o13GjjbD_ndAEaNRB1C3yRJMvMA@mail.gmail.com>,
	<04020B71-9CAA-458A-98D0-189A82E8F5F1@bryansmart.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F97@cooper>

It all depends really. On some solutions (depending on hypervisor, hardware and guest OS) the main problem can be that 20ms is a too short period to sleep, in that case a larger frame (ie 40ms) might help out. But mostly the problem is that the clock is not accurate enough, and fluxtuates from time to time (sometimes 20ms is 10, next round 30 etc).

After my latest testing I've found out that latest ESXi (4.1) works quite good, as long as you make sure not to share out more CPU's to the guests then you have actual cores on the hardware, and makes sure to use a separate NIC for that guest OS. I will summarize the results from my tests on the wiki soon, but I'm away for now, and won't get access to all the information I need until I get back - after the ClueCon event(!).

If using any kind of cloud environment (like the VPS the original questions was about) that's not under your own control, I highly recommend to solve it another way. Those solutions are usually to much loaded to get an accurate time. However, I do know that using for instance EC2 (xen) has worked for some, but that would also require that host to be on Linux (the original question was about Windows).

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Bryan Smart [bryansmart at bryansmart.com]
Skickat: den 28 juli 2011 13:52
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer gap 100953 detected!

> Peter, thanks for those details.

What sort of trade-offs, if any, are useful to improve audio for conferences and IVRs on VMs? Will increasing the interval setting for a conference help? What about using larger frames for codecs? I know that latency, beyond a point, is a serious quality issue of its own. Still, do you know what compromises provide the best improvement?

Bryan

> On 7/28/11, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
>> I feel I need to explain my last statement a little further :)
>>
>> First of all, you can run FS on virtual machines (even though I never
>> recommend it), but you need to be in control of the virtual environment, or
>> totally trust the ones controlling it.
>>
>> There are two basic problems when running on virtual hardware - timing and
>> network I/O. For timing to work you need to make sure that the one
>> controlling the virtual environment NEVER give out more CPU cores then there
>> are available on the hardware. If they give out more, and your machine need
>> to share resources with another machine, you will be in trouble. As soon as
>> the other virtual machine starts to load the CPU, your machine will get bad
>> timing. Also - usually a one core VM will get better timing then a two+ core
>> VM.
>>
>> Also, to be safe on the network side, you need your own NIC, that is not
>> shared with other machines.
>>
>> If you feel you can control this, you might be successful with the setup, if
>> you can't control this, you're either lucky (and things work anyway), or
>> you'll get bad audio.
>>
>> Also, just passing RTP for normal call isn't the most critical issue (you
>> don't even have to use a timer there), the most critical timing issues will
>> occur in conferences or when using a local IVR on the machine.
>>
>> I've done lots of testing for this in my lab, since many of my customers
>> wants to run our system (which uses FS for IVR/conferencing) on ESXi virtual
>> hosts, and I've found the above to be quite critical to succeed with your
>> virtual implementation.
>>
>> /Peter
>>
>> ____________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Giovanni Maruzzelli
>> [gmaruzz at gmail.com]
>> Skickat: den 28 juli 2011 09:37
>> Till: FreeSWITCH Users Help
>> ?mne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer
>> gap 100953 detected!
>>
>> Hey guys 'n gal,
>>
>> the original post is asking for Windows!!!
>>
>> -giovanni
>>
>>
>> On 7/28/11, Sam <lakersman2006 at yahoo.com> wrote:
>>> i forgot to mention that the 50 to 60 calls also are doing majority of
>>> g729
>>> transcoding as well which does put th cpu to 85 to 90% full utilization.
>>> So
>>> don't knock virtual machines.
>>>
>>>
>>>
>>> ________________________________
>>> From: Avi Marcus <avi at avimarcus.net>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Sent: Wednesday, July 27, 2011 3:20 PM
>>> Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large
>>> timer
>>> gap 100953 detected!
>>>
>>>
>>> I'm running FS under Xen provided by Linode. I only run max 10 concurrent
>>> channels in production, but an MOH test sounded good through 200
>>> concurrent
>>> channels.
>>> Linode says the CPU is the most underused resource on their systems...
>>>
>>>
>>> -Avi
>>>
>>>
>>> On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc at freeswitch.org>
>>> wrote:
>>>
>>>
>>>>
>>>>
>>>> On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>>
>>>> I run Asterisk on a virtual machine and it runs quite well even with
>>>> approximately 50 to 60 simultaneous callers and tha'ts even with media
>>>> passing thru it.
>>>>>
>>>> That's actually pretty impressive. What version of Asterisk do you use?
>>>> And
>>>> do you have to continually make sacrifices to the digital gods to keep
>>>> them appeased? :P
>>>>
>>>> -MC
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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
>>
>> --
>> Sent from my mobile device
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> 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
>>
>
> --
> Sent from my mobile device
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


_______________________________________________
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

!DSPAM:4e314e1632764912765535!



From skrishnamurthy at attinteractive.com  Thu Jul 28 21:53:23 2011
From: skrishnamurthy at attinteractive.com (Srinivasan Krishnamurthy)
Date: Thu, 28 Jul 2011 10:53:23 -0700
Subject: [Freeswitch-users] Trunk without authentication configuration
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F97@cooper>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
	<CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>
	<1311805861.95768.YahooMailNeo@web161013.mail.bf1.yahoo.com>
	<CALXCt0oLi81a_K+zWv3GiOmEv+JYyE7o85XUy4bRNaAA-wnWiw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F95@cooper>
	<CALXCt0powR2Zp_D=3ywf6h6o13GjjbD_ndAEaNRB1C3yRJMvMA@mail.gmail.com>,
	<04020B71-9CAA-458A-98D0-189A82E8F5F1@bryansmart.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F97@cooper>
Message-ID: <237050FF255D844CBDF840D42334EB9B6783F80F88@SFOEXM02.YELLOWPAGES.LOCAL>

I have configured a new gateway for outbound calls from FS as

   <list name="custom_carrier" default="deny">
     <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> 
   </list>
In acl.conf.xml

Created dialplan\default\01_custom_carrier.xml file as
<include>
<!-- Dial any 10 digit number 1(2223334444) here -->
	<extension name="Dial Out Custom Gateway">
		<condition field="destination_number" expression="^9(1\d{10})$">
			<action application="bridge" data="sofia/gateway/custom_carrier/$1"/>
		</condition>
	</extension>
</include>

Created sip_profiles\external\custom_carrier.xml
<include>
	<gateway name="custom_carrier">
		<param name="username" value="anything"/>
		<param name="password" value="anything"/>
		<param name="auth-calls" value="false"/>
		<param name="apply-outbound-acl" value="custom_carrier"/>
		<param name="apply-inbound-acl" value="custom_carrier"/>		
		<param name="register" value="false"/>
	</gateway>
</include>

When I make an outbound call from registered extension (1001) xlite I get the following error. (I also tried using SJ phone)

2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel sofia/external/14152484142 entering state [terminated][503]
2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup sofia/external/14152484142 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal sofia/external/14152484142 [KILL]
2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/14152484142 [BREAK]
2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314 (sofia/external/14152484142) Running State Change CS_HANGUP
2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498 (sofia/external/14152484142) State HANGUP
2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435 sofia/external/14152484142 Overriding SIP cause 503 with 503 from the other leg

I have attached the complete log too.

Iam a new FC users. I have referred to the following thread
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.html

and also read the section "Connecting to the world with gateways" in FreeSwitch book several times.

Any help would be appreciated.

Note: Iam able to make outbond calls from softphone through the trunk without FS.

Srini
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From aksrini at hotmail.com  Thu Jul 28 22:10:52 2011
From: aksrini at hotmail.com (Srinivasan Krishnamurthy)
Date: Thu, 28 Jul 2011 11:10:52 -0700
Subject: [Freeswitch-users] Trunk without authentication configuration
Message-ID: <COL107-W31E17797051BB054AF2BBFAA340@phx.gbl>







I have configured a new gateway for outbound calls from
FS as

 

   <list
name="custom_carrier" default="deny">

     <node
type="allow" cidr=xxx.xxx.xxx.xxx/32"/> 

   </list>

In acl.conf.xml

 

Created dialplan\default\01_custom_carrier.xml file as
<include>

<!-- Dial any 10 digit number 1(2223334444) here
-->

      <extension
name="Dial Out Custom Gateway">

            <condition
field="destination_number" expression="^9(1\d{10})$">

                  <action
application="bridge"
data="sofia/gateway/custom_carrier/$1"/>

            </condition>

      </extension>

</include>

 

Created sip_profiles\external\custom_carrier.xml

<include>

      <gateway
name="custom_carrier">

            <param
name="username" value="anything"/>

            <param
name="password" value="anything"/>

            <param
name="auth-calls" value="false"/>

            <param
name="apply-outbound-acl" value="custom_carrier"/>

            <param
name="apply-inbound-acl" value="custom_carrier"/>          

            <param
name="register" value="false"/>

      </gateway>

</include>

 

When I make an outbound call from registered extension
(1001) xlite I get the following error. (I also tried using SJ phone)

 

2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel
sofia/external/14152484142 entering state [terminated][503]

2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup
sofia/external/14152484142 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]

2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145
Send signal sofia/external/14152484142 [KILL]

2011-07-28 10:10:24.388677 [DEBUG]
switch_core_session.c:1022 Send signal sofia/external/14152484142 [BREAK]

2011-07-28 10:10:24.388677 [DEBUG]
switch_core_state_machine.c:314 (sofia/external/14152484142) Running State
Change CS_HANGUP

2011-07-28 10:10:24.388677 [DEBUG]
switch_core_state_machine.c:498 (sofia/external/14152484142) State HANGUP

2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435
sofia/external/14152484142 Overriding SIP cause 503 with 503 from the other leg

 

I have attached the complete log too.

 

Iam a new FC users. I have referred to the following
thread http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.html

 

and also read the section "Connecting to the world
with gateways" in FreeSwitch book several times.

 

Any help would be appreciated.

 

Note: Iam able to make outbond calls from softphone
through the trunk without FS.

 

Srini

 		 	   		  
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From msc at freeswitch.org  Thu Jul 28 23:31:12 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Jul 2011 12:31:12 -0700
Subject: [Freeswitch-users] Trunk without authentication configuration
In-Reply-To: <COL107-W31E17797051BB054AF2BBFAA340@phx.gbl>
References: <COL107-W31E17797051BB054AF2BBFAA340@phx.gbl>
Message-ID: <CAKzWOxUUyDBnFe0suDOh1SgiC-HfpkFomnGiQVgDjHP=fDJhuA@mail.gmail.com>

I recommend two things:
#1 turn on siptrace at fs_cli: sofia global siptrace on
#2 put the log into pastebin.freeswitch.org and use "FreeSWITCH Log" as the
syntax highlighting
Put the pastebin link in this email thread.

It looks to me like you may just not be supplying the correct credentials to
the carrier, but you won't know unless you look at the sip trace to see
what, exactly, is happening.

-MC

On Thu, Jul 28, 2011 at 11:10 AM, Srinivasan Krishnamurthy <
aksrini at hotmail.com> wrote:

>  I have configured a new gateway for outbound calls from FS as
>
> **** **
>
>    <list name="custom_carrier" default="deny">
>
>      <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/>
>
>    </list>
>
> In acl.conf.xml
>
> ** **
>
> Created dialplan\default\01_custom_carrier.xml file as <include>
>
> <!-- Dial any 10 digit number 1(2223334444) here -->
>
>       <extension name="Dial Out Custom Gateway">
>
>             <condition field="destination_number"
> expression="^9(1\d{10})$">
>
>                   <action application="bridge"
> data="sofia/gateway/custom_carrier/$1"/>
>
>             </condition>
>
>       </extension>
>
> </include>
>
> ** **
>
> Created sip_profiles\external\custom_carrier.xml
>
> <include>
>
>       <gateway name="custom_carrier">
>
>             <param name="username" value="anything"/>
>
>             <param name="password" value="anything"/>
>
>             <param name="auth-calls" value="false"/>
>
>             <param name="apply-outbound-acl" value="custom_carrier"/>
>
>             <param name="apply-inbound-acl" value="custom_carrier"/>
>
>
>             <param name="register" value="false"/>
>
>       </gateway>
>
> </include>
>
> ** **
>
> When I make an outbound call from registered extension (1001) xlite I get
> the following error. (I also tried using SJ phone)
>
> ** **
>
> 2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel
> sofia/external/14152484142 entering state [terminated][503]
>
> 2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup
> sofia/external/14152484142 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
>
> 2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal
> sofia/external/14152484142 [KILL]
>
> 2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal
> sofia/external/14152484142 [BREAK]
>
> 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/14152484142) Running State Change CS_HANGUP
>
> 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498
> (sofia/external/14152484142) State HANGUP
>
> 2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435
> sofia/external/14152484142 Overriding SIP cause 503 with 503 from the other
> leg
>
> ** **
>
> I have attached the complete log too.
>
> ** **
>
> Iam a new FC users. I have referred to the following thread
> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.html
>
> ** **
>
> and also read the section "Connecting to the world with gateways" in
> FreeSwitch book several times.
>
> ** **
>
> Any help would be appreciated.
>
> ** **
>
> Note: Iam able to make outbond calls from softphone through the trunk
> without FS.
>
> ** **
>
> Srini
>
> _______________________________________________
> 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 msc at freeswitch.org  Thu Jul 28 23:33:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Jul 2011 12:33:38 -0700
Subject: [Freeswitch-users] Discovered problem with PocketSphinx build?
In-Reply-To: <CB9CFB6B-F4C5-409D-AD83-D01D390CB6EB@bryansmart.com>
References: <4097DFE6-DA8E-47BC-A8DC-90AF44E0C575@bryansmart.com>
	<CB9CFB6B-F4C5-409D-AD83-D01D390CB6EB@bryansmart.com>
Message-ID: <CAKzWOxWQ7TFvA9qRM6+yaUG14A8fTW6MoKhbciY5XvHE6Y5PRg@mail.gmail.com>

If you have confirmed the 8k vs. 16kHz then yes, please add to the wiki.
-MC

On Thu, Jul 28, 2011 at 5:06 AM, Bryan Smart <bryansmart at bryansmart.com>wrote:

> I partially figured out my problem.
>
> Communicator only works for 8Khz. For 16Khz, you need WSJ1. I wonder why it
> isn't included with Freeswitch?
>
> WSJ1 is available from this page:
> http://www.speech.cs.cmu.edu/sphinx/models/
>
> Direct link for the latest release:
>
> http://www.speech.cs.cmu.edu/sphinx/models/wsj_jan2008/wsj_all_sc_5000_20080401.tar.gz
>
> Rename the extracted folder to wsj1, and place in grammar/model.
>
> For FS to use it, in pocketsphinx.conf.xml, set wideband-model:
>
> <param name="wideband-model" value="wsj1"/>
>
> ASR works fine for me now at 16Khz.
>
> Should I add this to the wiki? Maybe FS could have automatically installed
> wsj1, but I was clueless about the required build option.
>
> Bryan
>
> On Jul 27, 2011, at 6:46 PM, Bryan Smart wrote:
>
> Tried a build with a latest GIT, with PocketSphinx enabled, and get:
>
> Creating mod_pocketsphinx.la...
>
> /bin/sed: can't read
> /usr/src/freeswitch/libs/pocketsphinx-0.7/../sphinxbase/src
> /libsphinxbase/libsphinxbase.la: No such file or directory
>
> quiet_libtool: link:
> `/usr/src/freeswitch/libs/pocketsphinx-0.7/../sphinxbase/sr
> c/libsphinxbase/libsphinxbase.la' is not a valid libtool archive
>
> cat: .libs/mod_pocketsphinx.log: No such
> make[5]: *** [mod_pocketsphinx.la] Error 1
>
>
> I don't pretend to fully understand the build system, but the makefile
> seems to be trying to go to libs/sphinxbase for files, when that directory
> doesn't exist. The currently downloaded version of sphinxbase has the
> version appended to the directory name, as in sphinxbase-0.7. The Sphinx
> readme warns that the root should be named sphinxbase.
>
> I did:
>
> ln -s sphinxbase-0.7 sphinxbase
>
> and the build finished without problems.
>
> PocketSphinx works fine on a hardware phone using ULAW, but, when I use a
> softphone with 16Khz Speex, it doesn't. Instead, I get this
>
> 2011-07-27 21:43:14.651803 [WARNING] mod_pocketsphinx.c:147 Can't open
> speech model /usr/local/freeswitch/grammar/model/wsj1.
>
> The only item in the model directory is communicator. I thought wsj1 was
> the old model. What in Freeswitch would try to use wsj1 over communicator
> based on the codec/rate? The mod_pocketsphinx wiki page says that 16Khz is a
> supported sampling rate, so I'm guessing that there is a special reason why
> Speex doesn't work, or else that this is a bug.
>
> Bryan
>
>
>
> _______________________________________________
> 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 gavin.henry at gmail.com  Thu Jul 28 23:48:24 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Thu, 28 Jul 2011 20:48:24 +0100
Subject: [Freeswitch-users] DTMF echo/crossed?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F92@cooper>
References: <CAA8_NKAPZcSXjBzRuxxAiZrChOZF17zZ_iG6L3jg7tjmMjbTzw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F92@cooper>
Message-ID: <CAA8_NKBEvbA4KoKvRDz2OnxwhbO8PayZaMNB50-ZLdsJuC8aHQ@mail.gmail.com>

Nah, don't think so.

On 27 July 2011 08:48, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> My guess would be that the other party was generating DTMF?
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Gavin Henry [gavin.henry at gmail.com]
> Skickat: den 27 juli 2011 08:37
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] DTMF echo/crossed?
>
> Hi all,
>
> We generally don't suffer any type of echo deploying customer
> multi-tenant systems, but an interesting one was reported by a
> customer who said they could hear DTMF on the line that they were not
> generating?
>
> This is part of a business centre multi-tenant setup with just one
> user on that domain who heard the DTMF. They are lots of domains on
> that system. Any suggestions?
>
> Thanks,
>
> Gavin.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.surevoip.co.uk
>
> _______________________________________________
> 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
>
> !DSPAM:4e2fb31132766269912559!
>
>
> _______________________________________________
> 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
>



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


From freeswitch at simpot.com  Fri Jul 29 01:22:49 2011
From: freeswitch at simpot.com (Dmitry Saratsky)
Date: Fri, 29 Jul 2011 00:22:49 +0300
Subject: [Freeswitch-users] Trunk without authentication configuration
In-Reply-To: <237050FF255D844CBDF840D42334EB9B6783F80F88@SFOEXM02.YELLOWPAGES.LOCAL>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>	<CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>	<1311805861.95768.YahooMailNeo@web161013.mail.bf1.yahoo.com>	<CALXCt0oLi81a_K+zWv3GiOmEv+JYyE7o85XUy4bRNaAA-wnWiw@mail.gmail.com>	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F95@cooper>	<CALXCt0powR2Zp_D=3ywf6h6o13GjjbD_ndAEaNRB1C3yRJMvMA@mail.gmail.com>,
	<04020B71-9CAA-458A-98D0-189A82E8F5F1@bryansmart.com>	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F97@cooper>
	<237050FF255D844CBDF840D42334EB9B6783F80F88@SFOEXM02.YELLOWPAGES.LOCAL>
Message-ID: <000001cc4d6c$808ddfd0$81a99f70$@com>

I think you should try to change "custom_carrier" under dialplan and
sip_profile to "xxx.xxx.xxx.xxx":


   <list name="custom_carrier" default="deny">
     <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> 
   </list>
In acl.conf.xml

Created dialplan\default\01_custom_carrier.xml file as <include>
<!-- Dial any 10 digit number 1(2223334444) here -->
	<extension name="Dial Out Custom Gateway">
		<condition field="destination_number"
expression="^9(1\d{10})$">
			<action application="bridge"
data="sofia/gateway/xxx.xxx.xxx.xxx/$1"/>
		</condition>
	</extension>
</include>

Created sip_profiles\external\custom_carrier.xml
<include>
	<gateway name="xxx.xxx.xxx.xxx">
		<param name="username" value="anything"/>
		<param name="password" value="anything"/>
		<param name="auth-calls" value="false"/>
		<param name="apply-outbound-acl" value="custom_carrier"/>
		<param name="apply-inbound-acl" value="custom_carrier"/>

		<param name="register" value="false"/>
	</gateway>
</include>


Regards,
Dmitry.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Srinivasan Krishnamurthy
Sent: 28 Jul 2011 20:53
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Trunk without authentication configuration

I have configured a new gateway for outbound calls from FS as

   <list name="custom_carrier" default="deny">
     <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> 
   </list>
In acl.conf.xml

Created dialplan\default\01_custom_carrier.xml file as <include>
<!-- Dial any 10 digit number 1(2223334444) here -->
	<extension name="Dial Out Custom Gateway">
		<condition field="destination_number"
expression="^9(1\d{10})$">
			<action application="bridge"
data="sofia/gateway/custom_carrier/$1"/>
		</condition>
	</extension>
</include>

Created sip_profiles\external\custom_carrier.xml
<include>
	<gateway name="custom_carrier">
		<param name="username" value="anything"/>
		<param name="password" value="anything"/>
		<param name="auth-calls" value="false"/>
		<param name="apply-outbound-acl" value="custom_carrier"/>
		<param name="apply-inbound-acl" value="custom_carrier"/>

		<param name="register" value="false"/>
	</gateway>
</include>

When I make an outbound call from registered extension (1001) xlite I get
the following error. (I also tried using SJ phone)

2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel
sofia/external/14152484142 entering state [terminated][503]
2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup
sofia/external/14152484142 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal
sofia/external/14152484142 [KILL]
2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal
sofia/external/14152484142 [BREAK]
2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314
(sofia/external/14152484142) Running State Change CS_HANGUP
2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498
(sofia/external/14152484142) State HANGUP
2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435
sofia/external/14152484142 Overriding SIP cause 503 with 503 from the other
leg

I have attached the complete log too.

Iam a new FC users. I have referred to the following thread
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.htm
l

and also read the section "Connecting to the world with gateways" in
FreeSwitch book several times.

Any help would be appreciated.

Note: Iam able to make outbond calls from softphone through the trunk
without FS.

Srini



From skrishnamurthy at attinteractive.com  Fri Jul 29 01:41:26 2011
From: skrishnamurthy at attinteractive.com (Srinivasan Krishnamurthy)
Date: Thu, 28 Jul 2011 14:41:26 -0700
Subject: [Freeswitch-users] Trunk without authentication configuration
In-Reply-To: <CAKzWOxUUyDBnFe0suDOh1SgiC-HfpkFomnGiQVgDjHP=fDJhuA@mail.gmail.com>
References: <COL107-W31E17797051BB054AF2BBFAA340@phx.gbl>
	<CAKzWOxUUyDBnFe0suDOh1SgiC-HfpkFomnGiQVgDjHP=fDJhuA@mail.gmail.com>
Message-ID: <237050FF255D844CBDF840D42334EB9B6783F80F89@SFOEXM02.YELLOWPAGES.LOCAL>

Logs are attached http://pastebin.freeswitch.org/16910
I don't see FS sending INVITE out to gateway.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Thursday, July 28, 2011 12:31 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Trunk without authentication configuration

I recommend two things:
#1 turn on siptrace at fs_cli: sofia global siptrace on
#2 put the log into pastebin.freeswitch.org<http://pastebin.freeswitch.org> and use "FreeSWITCH Log" as the syntax highlighting
Put the pastebin link in this email thread.

It looks to me like you may just not be supplying the correct credentials to the carrier, but you won't know unless you look at the sip trace to see what, exactly, is happening.

-MC

On Thu, Jul 28, 2011 at 11:10 AM, Srinivasan Krishnamurthy <aksrini at hotmail.com<mailto:aksrini at hotmail.com>> wrote:

I have configured a new gateway for outbound calls from FS as



   <list name="custom_carrier" default="deny">

     <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/>

   </list>

In acl.conf.xml



Created dialplan\default\01_custom_carrier.xml file as <include>

<!-- Dial any 10 digit number 1(2223334444) here -->

      <extension name="Dial Out Custom Gateway">

            <condition field="destination_number" expression="^9(1\d{10})$">

                  <action application="bridge" data="sofia/gateway/custom_carrier/$1"/>

            </condition>

      </extension>

</include>



Created sip_profiles\external\custom_carrier.xml

<include>

      <gateway name="custom_carrier">

            <param name="username" value="anything"/>

            <param name="password" value="anything"/>

            <param name="auth-calls" value="false"/>

            <param name="apply-outbound-acl" value="custom_carrier"/>

            <param name="apply-inbound-acl" value="custom_carrier"/>

            <param name="register" value="false"/>

      </gateway>

</include>



When I make an outbound call from registered extension (1001) xlite I get the following error. (I also tried using SJ phone)



2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel sofia/external/14152484142 entering state [terminated][503]

2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup sofia/external/14152484142 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]

2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal sofia/external/14152484142 [KILL]

2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/14152484142 [BREAK]

2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314 (sofia/external/14152484142) Running State Change CS_HANGUP

2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498 (sofia/external/14152484142) State HANGUP

2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435 sofia/external/14152484142 Overriding SIP cause 503 with 503 from the other leg



I have attached the complete log too.



Iam a new FC users. I have referred to the following thread http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.html



and also read the section "Connecting to the world with gateways" in FreeSwitch book several times.



Any help would be appreciated.



Note: Iam able to make outbond calls from softphone through the trunk without FS.



Srini

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

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From joaocarlosleme at gmail.com  Fri Jul 29 01:52:36 2011
From: joaocarlosleme at gmail.com (Joao Leme)
Date: Thu, 28 Jul 2011 14:52:36 -0700
Subject: [Freeswitch-users] Bad sound quality and Abnormally large timer
 gap 100953 detected!
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F97@cooper>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
	<CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>
	<1311805861.95768.YahooMailNeo@web161013.mail.bf1.yahoo.com>
	<CALXCt0oLi81a_K+zWv3GiOmEv+JYyE7o85XUy4bRNaAA-wnWiw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F95@cooper>
	<CALXCt0powR2Zp_D=3ywf6h6o13GjjbD_ndAEaNRB1C3yRJMvMA@mail.gmail.com>
	<04020B71-9CAA-458A-98D0-189A82E8F5F1@bryansmart.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F97@cooper>
Message-ID: <CANGieu3PhUEZNYhY82Wjb3jLskQjbP2COrtFAfrWPWRMxtTmFg@mail.gmail.com>

Thanks you all! I decided to keep hosting locally instead of upgrading from
VPS to a Dedicated Server and got DSL service with static ip to be able to
log in remotely.
Thanks again,
John

On Thu, Jul 28, 2011 at 8:16 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> It all depends really. On some solutions (depending on hypervisor, hardware
> and guest OS) the main problem can be that 20ms is a too short period to
> sleep, in that case a larger frame (ie 40ms) might help out. But mostly the
> problem is that the clock is not accurate enough, and fluxtuates from time
> to time (sometimes 20ms is 10, next round 30 etc).
>
> After my latest testing I've found out that latest ESXi (4.1) works quite
> good, as long as you make sure not to share out more CPU's to the guests
> then you have actual cores on the hardware, and makes sure to use a separate
> NIC for that guest OS. I will summarize the results from my tests on the
> wiki soon, but I'm away for now, and won't get access to all the information
> I need until I get back - after the ClueCon event(!).
>
> If using any kind of cloud environment (like the VPS the original questions
> was about) that's not under your own control, I highly recommend to solve it
> another way. Those solutions are usually to much loaded to get an accurate
> time. However, I do know that using for instance EC2 (xen) has worked for
> some, but that would also require that host to be on Linux (the original
> question was about Windows).
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Bryan Smart [
> bryansmart at bryansmart.com]
> Skickat: den 28 juli 2011 13:52
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Bad sound quality and Abnormally large timer
> gap 100953 detected!
>
> > Peter, thanks for those details.
>
> What sort of trade-offs, if any, are useful to improve audio for
> conferences and IVRs on VMs? Will increasing the interval setting for a
> conference help? What about using larger frames for codecs? I know that
> latency, beyond a point, is a serious quality issue of its own. Still, do
> you know what compromises provide the best improvement?
>
> Bryan
>
> > On 7/28/11, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> >> I feel I need to explain my last statement a little further :)
> >>
> >> First of all, you can run FS on virtual machines (even though I never
> >> recommend it), but you need to be in control of the virtual environment,
> or
> >> totally trust the ones controlling it.
> >>
> >> There are two basic problems when running on virtual hardware - timing
> and
> >> network I/O. For timing to work you need to make sure that the one
> >> controlling the virtual environment NEVER give out more CPU cores then
> there
> >> are available on the hardware. If they give out more, and your machine
> need
> >> to share resources with another machine, you will be in trouble. As soon
> as
> >> the other virtual machine starts to load the CPU, your machine will get
> bad
> >> timing. Also - usually a one core VM will get better timing then a two+
> core
> >> VM.
> >>
> >> Also, to be safe on the network side, you need your own NIC, that is not
> >> shared with other machines.
> >>
> >> If you feel you can control this, you might be successful with the
> setup, if
> >> you can't control this, you're either lucky (and things work anyway), or
> >> you'll get bad audio.
> >>
> >> Also, just passing RTP for normal call isn't the most critical issue
> (you
> >> don't even have to use a timer there), the most critical timing issues
> will
> >> occur in conferences or when using a local IVR on the machine.
> >>
> >> I've done lots of testing for this in my lab, since many of my customers
> >> wants to run our system (which uses FS for IVR/conferencing) on ESXi
> virtual
> >> hosts, and I've found the above to be quite critical to succeed with
> your
> >> virtual implementation.
> >>
> >> /Peter
> >>
> >> ____________________________
> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> >> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Giovanni
> Maruzzelli
> >> [gmaruzz at gmail.com]
> >> Skickat: den 28 juli 2011 09:37
> >> Till: FreeSWITCH Users Help
> >> ?mne: Re: [Freeswitch-users] Bad sound quality and Abnormally large
> timer
> >> gap 100953 detected!
> >>
> >> Hey guys 'n gal,
> >>
> >> the original post is asking for Windows!!!
> >>
> >> -giovanni
> >>
> >>
> >> On 7/28/11, Sam <lakersman2006 at yahoo.com> wrote:
> >>> i forgot to mention that the 50 to 60 calls also are doing majority of
> >>> g729
> >>> transcoding as well which does put th cpu to 85 to 90% full
> utilization.
> >>> So
> >>> don't knock virtual machines.
> >>>
> >>>
> >>>
> >>> ________________________________
> >>> From: Avi Marcus <avi at avimarcus.net>
> >>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> >>> Sent: Wednesday, July 27, 2011 3:20 PM
> >>> Subject: Re: [Freeswitch-users] Bad sound quality and Abnormally large
> >>> timer
> >>> gap 100953 detected!
> >>>
> >>>
> >>> I'm running FS under Xen provided by Linode. I only run max 10
> concurrent
> >>> channels in production, but an MOH test sounded good through 200
> >>> concurrent
> >>> channels.
> >>> Linode says the CPU is the most underused resource on their systems...
> >>>
> >>>
> >>> -Avi
> >>>
> >>>
> >>> On Thu, Jul 28, 2011 at 12:57 AM, Michael Collins <msc at freeswitch.org>
> >>> wrote:
> >>>
> >>>
> >>>>
> >>>>
> >>>> On Wed, Jul 27, 2011 at 2:24 PM, Sam <lakersman2006 at yahoo.com> wrote:
> >>>>
> >>>> I run Asterisk on a virtual machine and it runs quite well even with
> >>>> approximately 50 to 60 simultaneous callers and tha'ts even with media
> >>>> passing thru it.
> >>>>>
> >>>> That's actually pretty impressive. What version of Asterisk do you
> use?
> >>>> And
> >>>> do you have to continually make sacrifices to the digital gods to keep
> >>>> them appeased? :P
> >>>>
> >>>> -MC
> >>>> _______________________________________________
> >>>> 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
> >>>>
> >>>>
> >>>
> >>> _______________________________________________
> >>> 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
> >>
> >> --
> >> Sent from my mobile device
> >>
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >> _______________________________________________
> >> 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
> >>
> >
> > --
> > Sent from my mobile device
> >
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> 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
>
> !DSPAM:4e314e1632764912765535!
>
>
> _______________________________________________
> 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 skrishnamurthy at attinteractive.com  Fri Jul 29 01:54:55 2011
From: skrishnamurthy at attinteractive.com (Srinivasan Krishnamurthy)
Date: Thu, 28 Jul 2011 14:54:55 -0700
Subject: [Freeswitch-users] Trunk without authentication configuration
In-Reply-To: <000001cc4d6c$808ddfd0$81a99f70$@com>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
	<CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>
	<1311805861.95768.YahooMailNeo@web161013
	<04020B71-9CAA-458A-98D0-189A82E8F5F1@bryansmart.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F97@cooper>
	<237050FF255D844CBDF840D42334EB9B6783F80F88@SFOEXM02.YELLOWPAGES.LOCAL>
	<000001cc4d6c$808ddfd0$81a99f70$@com>
Message-ID: <237050FF255D844CBDF840D42334EB9B6783F80F8A@SFOEXM02.YELLOWPAGES.LOCAL>

Thanks for your suggestion. I tried but got Invalid gateway err!!!

-----------------------------------------------------------------------------
EXECUTE sofia/internal/1001 at 10.10.2.3:5070 bridge(sofia/gateway/xxx.xxx.xxx.xxx/14152484142)
2011-07-28 14:48:39.163987 [ERR] mod_sofia.c:3492 Invalid Gateway
2011-07-28 14:48:39.163987 [NOTICE] mod_sofia.c:3779 Close Channel N/A [CS_NEW]
2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:428 () Running State Change CS_DESTROY
2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY
2011-07-28 14:48:39.163987 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY
2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY going to sleep
2011-07-28 14:48:39.163987 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
2011-07-28 14:48:39.163987 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
2011-07-28 14:48:39.163987 [INFO] mod_dptools.c:2356 Originate Failed.  Cause: INVALID_NUMBER_FORMAT
-------------------------------------------------------------------------------
Regards
Srini


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Dmitry Saratsky
Sent: Thursday, July 28, 2011 2:23 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Trunk without authentication configuration

I think you should try to change "custom_carrier" under dialplan and
sip_profile to "xxx.xxx.xxx.xxx":


   <list name="custom_carrier" default="deny">
     <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> 
   </list>
In acl.conf.xml

Created dialplan\default\01_custom_carrier.xml file as <include>
<!-- Dial any 10 digit number 1(2223334444) here -->
	<extension name="Dial Out Custom Gateway">
		<condition field="destination_number"
expression="^9(1\d{10})$">
			<action application="bridge"
data="sofia/gateway/xxx.xxx.xxx.xxx/$1"/>
		</condition>
	</extension>
</include>

Created sip_profiles\external\custom_carrier.xml
<include>
	<gateway name="xxx.xxx.xxx.xxx">
		<param name="username" value="anything"/>
		<param name="password" value="anything"/>
		<param name="auth-calls" value="false"/>
		<param name="apply-outbound-acl" value="custom_carrier"/>
		<param name="apply-inbound-acl" value="custom_carrier"/>

		<param name="register" value="false"/>
	</gateway>
</include>


Regards,
Dmitry.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Srinivasan Krishnamurthy
Sent: 28 Jul 2011 20:53
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Trunk without authentication configuration

I have configured a new gateway for outbound calls from FS as

   <list name="custom_carrier" default="deny">
     <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> 
   </list>
In acl.conf.xml

Created dialplan\default\01_custom_carrier.xml file as <include>
<!-- Dial any 10 digit number 1(2223334444) here -->
	<extension name="Dial Out Custom Gateway">
		<condition field="destination_number"
expression="^9(1\d{10})$">
			<action application="bridge"
data="sofia/gateway/custom_carrier/$1"/>
		</condition>
	</extension>
</include>

Created sip_profiles\external\custom_carrier.xml
<include>
	<gateway name="custom_carrier">
		<param name="username" value="anything"/>
		<param name="password" value="anything"/>
		<param name="auth-calls" value="false"/>
		<param name="apply-outbound-acl" value="custom_carrier"/>
		<param name="apply-inbound-acl" value="custom_carrier"/>

		<param name="register" value="false"/>
	</gateway>
</include>

When I make an outbound call from registered extension (1001) xlite I get
the following error. (I also tried using SJ phone)

2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel
sofia/external/14152484142 entering state [terminated][503]
2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup
sofia/external/14152484142 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal
sofia/external/14152484142 [KILL]
2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal
sofia/external/14152484142 [BREAK]
2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314
(sofia/external/14152484142) Running State Change CS_HANGUP
2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498
(sofia/external/14152484142) State HANGUP
2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435
sofia/external/14152484142 Overriding SIP cause 503 with 503 from the other
leg

I have attached the complete log too.

Iam a new FC users. I have referred to the following thread
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.htm
l

and also read the section "Connecting to the world with gateways" in
FreeSwitch book several times.

Any help would be appreciated.

Note: Iam able to make outbond calls from softphone through the trunk
without FS.

Srini


_______________________________________________
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 msc at freeswitch.org  Fri Jul 29 04:01:59 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Jul 2011 17:01:59 -0700
Subject: [Freeswitch-users] Trunk without authentication configuration
In-Reply-To: <237050FF255D844CBDF840D42334EB9B6783F80F89@SFOEXM02.YELLOWPAGES.LOCAL>
References: <COL107-W31E17797051BB054AF2BBFAA340@phx.gbl>
	<CAKzWOxUUyDBnFe0suDOh1SgiC-HfpkFomnGiQVgDjHP=fDJhuA@mail.gmail.com>
	<237050FF255D844CBDF840D42334EB9B6783F80F89@SFOEXM02.YELLOWPAGES.LOCAL>
Message-ID: <CAKzWOxUKueTBHL--POpf4hVdJE-6onbkWYCOCZk71-R_Xjz2Dw@mail.gmail.com>

On Thu, Jul 28, 2011 at 2:41 PM, Srinivasan Krishnamurthy <
skrishnamurthy at attinteractive.com> wrote:

> Logs are attached http://pastebin.freeswitch.org/16910****
>
> I don?t see FS sending INVITE out to gateway.****
>
> **
>
I just noticed that you have what appear to be profile parameters in your
gateway config. Try a minimal gateway config. Make a copy of
sip-profiles/external/example.xml and fill in only the items you absolutely
need. Make sure you put these in the sip-profile (external.xml) file:

            <param name="auth-calls" value="false"/>

            <param name="apply-outbound-acl" value="custom_carrier"/>

            <param name="apply-inbound-acl" value="custom_carrier"/>

Be sure to reloadxml and then reload mod_sofia. Try again...

-MC
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From freeswitch at simpot.com  Fri Jul 29 04:58:07 2011
From: freeswitch at simpot.com (Dmitry Saratsky)
Date: Fri, 29 Jul 2011 03:58:07 +0300
Subject: [Freeswitch-users] Trunk without authentication configuration
In-Reply-To: <237050FF255D844CBDF840D42334EB9B6783F80F8A@SFOEXM02.YELLOWPAGES.LOCAL>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>	<CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>	<1311805861.95768.YahooMailNeo@web161013	<04020B71-9CAA-458A-98D0-189A82E8F5F1@bryansmart.com>	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F97@cooper>	<237050FF255D844CBDF840D42334EB9B6783F80F88@SFOEXM02.YELLOWPAGES.LOCAL>	<000001cc4d6c$808ddfd0$81a99f70$@com>
	<237050FF255D844CBDF840D42334EB9B6783F80F8A@SFOEXM02.YELLOWPAGES.LOCAL>
Message-ID: <000101cc4d8a$947a2440$bd6e6cc0$@com>

Did you restarted FS after that change? Or you only have reloaded xmls???

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Srinivasan Krishnamurthy
Sent: 29 Jul 2011 00:55
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Trunk without authentication configuration

Thanks for your suggestion. I tried but got Invalid gateway err!!!

----------------------------------------------------------------------------
-
EXECUTE sofia/internal/1001 at 10.10.2.3:5070
bridge(sofia/gateway/xxx.xxx.xxx.xxx/14152484142)
2011-07-28 14:48:39.163987 [ERR] mod_sofia.c:3492 Invalid Gateway
2011-07-28 14:48:39.163987 [NOTICE] mod_sofia.c:3779 Close Channel N/A
[CS_NEW]
2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:428 ()
Running State Change CS_DESTROY
2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A)
State DESTROY
2011-07-28 14:48:39.163987 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY
2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A)
State DESTROY going to sleep
2011-07-28 14:48:39.163987 [ERR] switch_ivr_originate.c:2493 Cannot create
outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
2011-07-28 14:48:39.163987 [DEBUG] switch_ivr_originate.c:3308 Originate
Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
2011-07-28 14:48:39.163987 [INFO] mod_dptools.c:2356 Originate Failed.
Cause: INVALID_NUMBER_FORMAT
----------------------------------------------------------------------------
---
Regards
Srini


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Dmitry
Saratsky
Sent: Thursday, July 28, 2011 2:23 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Trunk without authentication configuration

I think you should try to change "custom_carrier" under dialplan and
sip_profile to "xxx.xxx.xxx.xxx":


   <list name="custom_carrier" default="deny">
     <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> 
   </list>
In acl.conf.xml

Created dialplan\default\01_custom_carrier.xml file as <include>
<!-- Dial any 10 digit number 1(2223334444) here -->
	<extension name="Dial Out Custom Gateway">
		<condition field="destination_number"
expression="^9(1\d{10})$">
			<action application="bridge"
data="sofia/gateway/xxx.xxx.xxx.xxx/$1"/>
		</condition>
	</extension>
</include>

Created sip_profiles\external\custom_carrier.xml
<include>
	<gateway name="xxx.xxx.xxx.xxx">
		<param name="username" value="anything"/>
		<param name="password" value="anything"/>
		<param name="auth-calls" value="false"/>
		<param name="apply-outbound-acl" value="custom_carrier"/>
		<param name="apply-inbound-acl" value="custom_carrier"/>

		<param name="register" value="false"/>
	</gateway>
</include>


Regards,
Dmitry.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Srinivasan Krishnamurthy
Sent: 28 Jul 2011 20:53
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Trunk without authentication configuration

I have configured a new gateway for outbound calls from FS as

   <list name="custom_carrier" default="deny">
     <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/> 
   </list>
In acl.conf.xml

Created dialplan\default\01_custom_carrier.xml file as <include>
<!-- Dial any 10 digit number 1(2223334444) here -->
	<extension name="Dial Out Custom Gateway">
		<condition field="destination_number"
expression="^9(1\d{10})$">
			<action application="bridge"
data="sofia/gateway/custom_carrier/$1"/>
		</condition>
	</extension>
</include>

Created sip_profiles\external\custom_carrier.xml
<include>
	<gateway name="custom_carrier">
		<param name="username" value="anything"/>
		<param name="password" value="anything"/>
		<param name="auth-calls" value="false"/>
		<param name="apply-outbound-acl" value="custom_carrier"/>
		<param name="apply-inbound-acl" value="custom_carrier"/>

		<param name="register" value="false"/>
	</gateway>
</include>

When I make an outbound call from registered extension (1001) xlite I get
the following error. (I also tried using SJ phone)

2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel
sofia/external/14152484142 entering state [terminated][503]
2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup
sofia/external/14152484142 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal
sofia/external/14152484142 [KILL]
2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal
sofia/external/14152484142 [BREAK]
2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314
(sofia/external/14152484142) Running State Change CS_HANGUP
2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498
(sofia/external/14152484142) State HANGUP
2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435
sofia/external/14152484142 Overriding SIP cause 503 with 503 from the other
leg

I have attached the complete log too.

Iam a new FC users. I have referred to the following thread
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.htm
l

and also read the section "Connecting to the world with gateways" in
FreeSwitch book several times.

Any help would be appreciated.

Note: Iam able to make outbond calls from softphone through the trunk
without FS.

Srini


_______________________________________________
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


_______________________________________________
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 erin.omeara at salmonbaytechnology.com  Fri Jul 29 05:43:10 2011
From: erin.omeara at salmonbaytechnology.com (Erin O'Meara)
Date: Thu, 28 Jul 2011 18:43:10 -0700
Subject: [Freeswitch-users] No audio, only on outgoing calls
Message-ID: <CAB4Oy0UgRWg6Dg-BXJKyQMcGCfO+8nbDT1Kn0PhPXe=A=w4-1A@mail.gmail.com>

I have a blue.box server setup with a cloudhost, setup IPKall & DIDforSale.
Both incoming calls come thru to my phone connected to the blue.box, I setup
outgoing calls with CallwithUs and a Toll Free termination provider. All
outgoing calls from the phone thru the blue.box connect, but no audio in
either direction.

The phone is behind a PFSense Firewall/Router and I have a second phone
connected to an Elastix PBX that I use for my day to day business and
everything works as expected. I have disable the Block All firewall rule and
replaced it with Pass All and am pretty sure at this point its not a
Firewall issue.

When I dial the extension's number, I can see voicemail wav files playing
but again hear nothing. So from what I can determine this is a freeswitch to
phone issue.

I have chatted on IRC with people at #2600 and #Freeswitch and googled
the symptoms and come up no solution.

Hoping some has seen this before and has a solution so I don't have to smash
anything in frustration.

Regards,

206.905.9520
http://salmonbaytechnology.com
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From brad at tech21.com  Fri Jul 29 06:32:51 2011
From: brad at tech21.com (Brad Mina)
Date: Thu, 28 Jul 2011 19:32:51 -0700
Subject: [Freeswitch-users] Trunk without authentication configuration
In-Reply-To: <000101cc4d8a$947a2440$bd6e6cc0$@com>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
	<CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>
	<04020B71-9CAA-458A-98D0-189A82E8F5F1@bryansmart.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F97@cooper>
	<237050FF255D844CBDF840D42334EB9B6783F80F88@SFOEXM02.YELLOWPAGES.LOCAL>
	<000001cc4d6c$808ddfd0$81a99f70$@com>
	<237050FF255D844CBDF840D42334EB9B6783F80F8A@SFOEXM02.YELLOWPAGES.LOCAL>
	<000101cc4d8a$947a2440$bd6e6cc0$@com>
Message-ID: <CAGRr058eCwzSrM=ovjGk3osJp56M8f=Youc0ge5TBSgsWfjHsg@mail.gmail.com>

Might want to doublecheck the format your carrier accepts for outgoing
calls.

Also, attatch a full siptrace with your debug log, that helps a lot.

freeswitch> sofia profile external siptrace on

On Thu, Jul 28, 2011 at 5:58 PM, Dmitry Saratsky <freeswitch at simpot.com>wrote:

> Did you restarted FS after that change? Or you only have reloaded xmls???
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Srinivasan Krishnamurthy
> Sent: 29 Jul 2011 00:55
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Trunk without authentication configuration
>
> Thanks for your suggestion. I tried but got Invalid gateway err!!!
>
>
> ----------------------------------------------------------------------------
> -
> EXECUTE sofia/internal/1001 at 10.10.2.3:5070
> bridge(sofia/gateway/xxx.xxx.xxx.xxx/14152484142)
> 2011-07-28 14:48:39.163987 [ERR] mod_sofia.c:3492 Invalid Gateway
> 2011-07-28 14:48:39.163987 [NOTICE] mod_sofia.c:3779 Close Channel N/A
> [CS_NEW]
> 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:428 ()
> Running State Change CS_DESTROY
> 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A)
> State DESTROY
> 2011-07-28 14:48:39.163987 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY
> 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A)
> State DESTROY going to sleep
> 2011-07-28 14:48:39.163987 [ERR] switch_ivr_originate.c:2493 Cannot create
> outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
> 2011-07-28 14:48:39.163987 [DEBUG] switch_ivr_originate.c:3308 Originate
> Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
> 2011-07-28 14:48:39.163987 [INFO] mod_dptools.c:2356 Originate Failed.
> Cause: INVALID_NUMBER_FORMAT
>
> ----------------------------------------------------------------------------
> ---
> Regards
> Srini
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Dmitry
> Saratsky
> Sent: Thursday, July 28, 2011 2:23 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Trunk without authentication configuration
>
> I think you should try to change "custom_carrier" under dialplan and
> sip_profile to "xxx.xxx.xxx.xxx":
>
>
>   <list name="custom_carrier" default="deny">
>     <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/>
>   </list>
> In acl.conf.xml
>
> Created dialplan\default\01_custom_carrier.xml file as <include>
> <!-- Dial any 10 digit number 1(2223334444) here -->
>        <extension name="Dial Out Custom Gateway">
>                <condition field="destination_number"
> expression="^9(1\d{10})$">
>                        <action application="bridge"
> data="sofia/gateway/xxx.xxx.xxx.xxx/$1"/>
>                </condition>
>        </extension>
> </include>
>
> Created sip_profiles\external\custom_carrier.xml
> <include>
>        <gateway name="xxx.xxx.xxx.xxx">
>                <param name="username" value="anything"/>
>                <param name="password" value="anything"/>
>                <param name="auth-calls" value="false"/>
>                <param name="apply-outbound-acl" value="custom_carrier"/>
>                <param name="apply-inbound-acl" value="custom_carrier"/>
>
>                <param name="register" value="false"/>
>        </gateway>
> </include>
>
>
> Regards,
> Dmitry.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Srinivasan Krishnamurthy
> Sent: 28 Jul 2011 20:53
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Trunk without authentication configuration
>
> I have configured a new gateway for outbound calls from FS as
>
>   <list name="custom_carrier" default="deny">
>     <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/>
>   </list>
> In acl.conf.xml
>
> Created dialplan\default\01_custom_carrier.xml file as <include>
> <!-- Dial any 10 digit number 1(2223334444) here -->
>        <extension name="Dial Out Custom Gateway">
>                <condition field="destination_number"
> expression="^9(1\d{10})$">
>                        <action application="bridge"
> data="sofia/gateway/custom_carrier/$1"/>
>                </condition>
>        </extension>
> </include>
>
> Created sip_profiles\external\custom_carrier.xml
> <include>
>        <gateway name="custom_carrier">
>                <param name="username" value="anything"/>
>                <param name="password" value="anything"/>
>                <param name="auth-calls" value="false"/>
>                <param name="apply-outbound-acl" value="custom_carrier"/>
>                <param name="apply-inbound-acl" value="custom_carrier"/>
>
>                <param name="register" value="false"/>
>        </gateway>
> </include>
>
> When I make an outbound call from registered extension (1001) xlite I get
> the following error. (I also tried using SJ phone)
>
> 2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel
> sofia/external/14152484142 entering state [terminated][503]
> 2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup
> sofia/external/14152484142 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
> 2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal
> sofia/external/14152484142 [KILL]
> 2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal
> sofia/external/14152484142 [BREAK]
> 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/14152484142) Running State Change CS_HANGUP
> 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498
> (sofia/external/14152484142) State HANGUP
> 2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435
> sofia/external/14152484142 Overriding SIP cause 503 with 503 from the
> other
> leg
>
> I have attached the complete log too.
>
> Iam a new FC users. I have referred to the following thread
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.htm
> l
>
> and also read the section "Connecting to the world with gateways" in
> FreeSwitch book several times.
>
> Any help would be appreciated.
>
> Note: Iam able to make outbond calls from softphone through the trunk
> without FS.
>
> Srini
>
>
> _______________________________________________
> 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
>
>
> _______________________________________________
> 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
>
>
> _______________________________________________
> 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 dome at tel.co.th  Fri Jul 29 08:55:22 2011
From: dome at tel.co.th (Dome Charoenyost)
Date: Fri, 29 Jul 2011 11:55:22 +0700
Subject: [Freeswitch-users] FS as registra server behind SBC
In-Reply-To: <CAMFECSCCsjBpJ1MzPbP=8uo_WAkAjnnt6bHG5q=r2Ny0O6skvQ@mail.gmail.com>
References: <CAMFECSCCsjBpJ1MzPbP=8uo_WAkAjnnt6bHG5q=r2Ny0O6skvQ@mail.gmail.com>
Message-ID: <CAMFECSBG-6r62CG8xCzWOdO1Qgh6uChJq3O8+Uhf8sVZzKML5g@mail.gmail.com>

Work fine after reconfig Huiwei SBC .

Dome C.

2011/7/27 Dome Charoenyost <dome at tel.co.th>:
> Dear All,
> ? ? ? ? ? ?I found problem when setup FS behind ?Huwei SBC.
> my network diagram is.
> ? ? ? ? ? ? ? ? ? ? ? ? ? ? NAT ? ? ? ? ? ? ? ? ? ? Public IP
> ? ? ? ? ? NAT 10.x.x.x
> [SIP Client 1] ----- [ADSL Router] ---------------------------- ?SBC
> ??---------------------------- ?FS
>
> /
> [SIP Client 2] ----- [ADSL Router] --------------------------/
> It's work fine when call incomming and play sound. But i got 403 from SBC
> when i try to call Local extension or originate call to registerd user.
> I thing nat problem can someone show me sip profile config work in this
> case.
>
> BG
> Dome C.


From cmcureau at gmail.com  Fri Jul 29 09:28:43 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Fri, 29 Jul 2011 00:28:43 -0500
Subject: [Freeswitch-users] Help with choppy audio after attended transfer
Message-ID: <CAL0fNgWOuhAeMgLS3UsD4tSqdDd2FGSqXw4T-H9AnfVLBGOr5Q@mail.gmail.com>

I'm having some issues with extremely choppy audio after a call has been
sent to another extension via an automated transfer.  The audio is great
when the call is answered.  Shortly after, the transfer button is pressed
and the incoming call hears music on hold.  The music on hold is sent to the
caller sounds fine as does the conversation between extensions.  When the
transfer is completed, the caller hears what sounds like someone speaking
through a fan (though slower) but incoming audio sounds fine.

Since it's such a large log, I posted it to the FreeSWITCH pastebin:
http://pastebin.freeswitch.org/16911

I'm thinking that it has something to do with the transition from MOH to the
internal extension, but I can't figure out what is happening.

Any ideas?
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From brad at tech21.com  Fri Jul 29 12:19:03 2011
From: brad at tech21.com (Brad Mina)
Date: Fri, 29 Jul 2011 01:19:03 -0700
Subject: [Freeswitch-users] FS as registra server behind SBC
In-Reply-To: <CAMFECSBG-6r62CG8xCzWOdO1Qgh6uChJq3O8+Uhf8sVZzKML5g@mail.gmail.com>
References: <CAMFECSCCsjBpJ1MzPbP=8uo_WAkAjnnt6bHG5q=r2Ny0O6skvQ@mail.gmail.com>
	<CAMFECSBG-6r62CG8xCzWOdO1Qgh6uChJq3O8+Uhf8sVZzKML5g@mail.gmail.com>
Message-ID: <CAGRr058MY3s7uYNRAC+xLqM5N1F9J0Smm+Y_bxaYOVsT+KappQ@mail.gmail.com>

Can you possibly tell us the configuration you're using? It might be useful
to document this.

On Thu, Jul 28, 2011 at 9:55 PM, Dome Charoenyost <dome at tel.co.th> wrote:

> Work fine after reconfig Huiwei SBC .
>
> Dome C.
>
> 2011/7/27 Dome Charoenyost <dome at tel.co.th>:
> > Dear All,
> >            I found problem when setup FS behind  Huwei SBC.
> > my network diagram is.
> >                             NAT                     Public IP
> >           NAT 10.x.x.x
> > [SIP Client 1] ----- [ADSL Router] ----------------------------  SBC
> >   ----------------------------  FS
> >
> > /
> > [SIP Client 2] ----- [ADSL Router] --------------------------/
> > It's work fine when call incomming and play sound. But i got 403 from SBC
> > when i try to call Local extension or originate call to registerd user.
> > I thing nat problem can someone show me sip profile config work in this
> > case.
> >
> > BG
> > Dome C.
>
> _______________________________________________
> 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 michael.knop at hcu-hamburg.de  Fri Jul 29 13:58:11 2011
From: michael.knop at hcu-hamburg.de (michael knop)
Date: Fri, 29 Jul 2011 11:58:11 +0200
Subject: [Freeswitch-users] FS to a Sonus SIP trunk
In-Reply-To: <4E31369A.70605@hcu-hamburg.de>
References: <4E2ED4AB.1030305@hcu-hamburg.de> <4E31369A.70605@hcu-hamburg.de>
Message-ID: <4E328433.8070701@hcu-hamburg.de>

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.

/micha


From frank at telonium.com  Fri Jul 29 18:39:19 2011
From: frank at telonium.com (Frank Park)
Date: Fri, 29 Jul 2011 10:39:19 -0400
Subject: [Freeswitch-users] DTMF echo/crossed?
In-Reply-To: <CAA8_NKBEvbA4KoKvRDz2OnxwhbO8PayZaMNB50-ZLdsJuC8aHQ@mail.gmail.com>
References: <CAA8_NKAPZcSXjBzRuxxAiZrChOZF17zZ_iG6L3jg7tjmMjbTzw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F92@cooper>
	<CAA8_NKBEvbA4KoKvRDz2OnxwhbO8PayZaMNB50-ZLdsJuC8aHQ@mail.gmail.com>
Message-ID: <CAAp+uUEgx0edOT7wN=fgPejC53U7JG4=_O+VXnQM+Brc0CVX6g@mail.gmail.com>

I do experience this every so often as well..

On Thu, Jul 28, 2011 at 3:48 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> Nah, don't think so.
>
> On 27 July 2011 08:48, Peter Olsson <peter.olsson at visionutveckling.se>
> wrote:
> > My guess would be that the other party was generating DTMF?
> >
> > /Peter
> > ________________________________________
> > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Gavin Henry [
> gavin.henry at gmail.com]
> > Skickat: den 27 juli 2011 08:37
> > Till: FreeSWITCH Users Help
> > ?mne: [Freeswitch-users] DTMF echo/crossed?
> >
> > Hi all,
> >
> > We generally don't suffer any type of echo deploying customer
> > multi-tenant systems, but an interesting one was reported by a
> > customer who said they could hear DTMF on the line that they were not
> > generating?
> >
> > This is part of a business centre multi-tenant setup with just one
> > user on that domain who heard the DTMF. They are lots of domains on
> > that system. Any suggestions?
> >
> > Thanks,
> >
> > Gavin.
> >
> > --
> > http://www.suretecsystems.com/services/openldap/
> > http://www.surevoip.co.uk
> >
> > _______________________________________________
> > 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
> >
> > !DSPAM:4e2fb31132766269912559!
> >
> >
> > _______________________________________________
> > 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
> >
>
>
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.surevoip.co.uk
>
> _______________________________________________
> 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
>



-- 

----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----
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From anthony.minessale at gmail.com  Fri Jul 29 19:44:47 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 29 Jul 2011 10:44:47 -0500
Subject: [Freeswitch-users] Help with choppy audio after attended
	transfer
In-Reply-To: <CAL0fNgWOuhAeMgLS3UsD4tSqdDd2FGSqXw4T-H9AnfVLBGOr5Q@mail.gmail.com>
References: <CAL0fNgWOuhAeMgLS3UsD4tSqdDd2FGSqXw4T-H9AnfVLBGOr5Q@mail.gmail.com>
Message-ID: <CAKbxfG-UOWoisXbbZaEH0Ruzm7=e2oobyvwqM1vJ-pQFtDuOow@mail.gmail.com>

probably ptime related thing.
you should have included the sip trace "sofia global siptrace on"


On Fri, Jul 29, 2011 at 12:28 AM, Chris Cureau <cmcureau at gmail.com> wrote:
> I'm having some issues with extremely choppy audio after a call has been
> sent to another extension via an automated transfer.? The audio is great
> when the call is answered.? Shortly after, the transfer button is pressed
> and the incoming call hears music on hold.? The music on hold is sent to the
> caller sounds fine as does the conversation between extensions.? When the
> transfer is completed, the caller hears what sounds like someone speaking
> through a fan (though slower) but incoming audio sounds fine.
>
> Since it's such a large log, I posted it to the FreeSWITCH pastebin:
> http://pastebin.freeswitch.org/16911
>
> I'm thinking that it has something to do with the transition from MOH to the
> internal extension, but I can't figure out what is happening.
>
> Any ideas?
>
> _______________________________________________
> 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
>
>



-- 
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 norm at voicenetwork.ca  Fri Jul 29 19:56:40 2011
From: norm at voicenetwork.ca (Norman Tomlins)
Date: Fri, 29 Jul 2011 11:56:40 -0400
Subject: [Freeswitch-users] DTMF echo/crossed?
In-Reply-To: <CAA8_NKAPZcSXjBzRuxxAiZrChOZF17zZ_iG6L3jg7tjmMjbTzw@mail.gmail.com>
References: <CAA8_NKAPZcSXjBzRuxxAiZrChOZF17zZ_iG6L3jg7tjmMjbTzw@mail.gmail.com>
Message-ID: <CALdmMT6LZPaJu92Hb1CBEtRkEmb4tBzjJdh3bFLyP+ECrpGVUg@mail.gmail.com>

Gavin,

If you are using the LinkSys SPA2102's you will get DTMF digits when
people are talking.   You will need to increase the
DTMF_Tx_Strict_Hold_Off_Time to about 80.

This will stop random DTMF while people are talking


Norman Tomlins
Voice Network Inc.



On Wed, Jul 27, 2011 at 2:37 AM, Gavin Henry <gavin.henry at gmail.com> wrote:
> Hi all,
>
> We generally don't suffer any type of echo deploying customer
> multi-tenant systems, but an interesting one was reported by a
> customer who said they could hear DTMF on the line that they were not
> generating?
>
> This is part of a business centre multi-tenant setup with just one
> user on that domain who heard the DTMF. They are lots of domains on
> that system. Any suggestions?
>
> Thanks,
>
> Gavin.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.surevoip.co.uk
>
> _______________________________________________
> 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 dhairyavora at gmail.com  Fri Jul 29 12:07:00 2011
From: dhairyavora at gmail.com (Dhairya Vora)
Date: Fri, 29 Jul 2011 13:37:00 +0530
Subject: [Freeswitch-users] unable to connect to freeswitch server using
	x-lite
Message-ID: <CAHq18AxZ1Ky_5RhaOpANTWkAhZCOuKhmoe3dpd49VWCk67V5Vw@mail.gmail.com>

I have installed freeswitch on centos
-----------------------------------------------------
[root at localhost ~]# uname -r
2.6.18-238.19.1.el5
------------------------------------------------------

I have installed x-lite on a different machine in same lan.

when i try to connect x-lite to freeswitch with valid id, password, request
times out.

any idea to solve this?

-- 
Dhairya Vora(newbie)
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From msc at freeswitch.org  Fri Jul 29 20:35:16 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jul 2011 09:35:16 -0700
Subject: [Freeswitch-users] unable to connect to freeswitch server using
	x-lite
In-Reply-To: <CAHq18AxZ1Ky_5RhaOpANTWkAhZCOuKhmoe3dpd49VWCk67V5Vw@mail.gmail.com>
References: <CAHq18AxZ1Ky_5RhaOpANTWkAhZCOuKhmoe3dpd49VWCk67V5Vw@mail.gmail.com>
Message-ID: <CAKzWOxU=KFh8wcGHrypZKQ5oJ7X0Ua9fmDjuDabNAF==t2h-3w@mail.gmail.com>

make sure that the centos firewall isn't on and blocking everything. You can
run "lokkit" from the linux command line and turn off firewall and SELinux
and try again.
-MC

On Fri, Jul 29, 2011 at 1:07 AM, Dhairya Vora <dhairyavora at gmail.com> wrote:

> I have installed freeswitch on centos
> -----------------------------------------------------
> [root at localhost ~]# uname -r
> 2.6.18-238.19.1.el5
> ------------------------------------------------------
>
> I have installed x-lite on a different machine in same lan.
>
> when i try to connect x-lite to freeswitch with valid id, password, request
> times out.
>
> any idea to solve this?
>
> --
> Dhairya Vora(newbie)
>
>
> _______________________________________________
> 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 skrishnamurthy at attinteractive.com  Fri Jul 29 21:19:20 2011
From: skrishnamurthy at attinteractive.com (Srinivasan Krishnamurthy)
Date: Fri, 29 Jul 2011 10:19:20 -0700
Subject: [Freeswitch-users] Trunk without authentication configuration
In-Reply-To: <CAGRr058eCwzSrM=ovjGk3osJp56M8f=Youc0ge5TBSgsWfjHsg@mail.gmail.com>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
	<CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>
	<04020B71-9CAA-458A-98D0-189A82E8F5F1@bryansmart.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F97@cooper>
	<237050FF255D844CBDF840D42334EB9B6783F80F88@SFOEXM02.YELLOWPAGES.LOCAL>
	<000001cc4d6c$808ddfd0$81a99f70$@com>
	<237050FF255D844CBDF840D42334EB9B6783F80F8A@SFOEXM02.YELLOWPAGES.LOCAL>
	<000101cc4d8a$947a2440$bd6e6cc0$@com>
	<CAGRr058eCwzSrM=ovjGk3osJp56M8f=Youc0ge5TBSgsWfjHsg@mail.gmail.com>
Message-ID: <237050FF255D844CBDF840D42334EB9B6783F80F8E@SFOEXM02.YELLOWPAGES.LOCAL>

Now my sip_profiles\external\custom_carrier.xml looks like

<include>
                <gateway name="custom_carrier">
                                <param name="auth-calls" value="false"/>
                                <param name="apply-outbound-acl" value="custom_carrier"/>
                                <param name="apply-inbound-acl" value="custom_carrier"/>
                </gateway>
</include>

I have restarted FS.  Enabled sofia log in debug ?sofia tracelevel 6? and sip trace is on too.
Refer to complete log @ http://pastebin.freeswitch.org/16918

2011-07-29 10:00:31.372738 [ERR] mod_sofia.c:3492 Invalid Gateway
2011-07-29 10:00:31.372738 [NOTICE] mod_sofia.c:3779 Close Channel N/A [CS_NEW]
2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:428 () Running State Change CS_DESTROY
2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY
2011-07-29 10:00:31.372738 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY
2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY going to sleep
2011-07-29 10:00:31.372738 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]

Regards
Srini

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brad Mina
Sent: Thursday, July 28, 2011 7:33 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Trunk without authentication configuration

Might want to doublecheck the format your carrier accepts for outgoing calls.

Also, attatch a full siptrace with your debug log, that helps a lot.

freeswitch> sofia profile external siptrace on
On Thu, Jul 28, 2011 at 5:58 PM, Dmitry Saratsky <freeswitch at simpot.com<mailto:freeswitch at simpot.com>> wrote:
Did you restarted FS after that change? Or you only have reloaded xmls???

-----Original Message-----
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:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of
Srinivasan Krishnamurthy
Sent: 29 Jul 2011 00:55
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Trunk without authentication configuration

Thanks for your suggestion. I tried but got Invalid gateway err!!!

----------------------------------------------------------------------------
-
EXECUTE sofia/internal/1001 at 10.10.2.3:5070<http://1001 at 10.10.2.3:5070>
bridge(sofia/gateway/xxx.xxx.xxx.xxx/14152484142<tel:14152484142>)
2011-07-28 14:48:39.163987 [ERR] mod_sofia.c:3492 Invalid Gateway
2011-07-28 14:48:39.163987 [NOTICE] mod_sofia.c:3779 Close Channel N/A
[CS_NEW]
2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:428 ()
Running State Change CS_DESTROY
2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A)
State DESTROY
2011-07-28 14:48:39.163987 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY
2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A)
State DESTROY going to sleep
2011-07-28 14:48:39.163987 [ERR] switch_ivr_originate.c:2493 Cannot create
outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
2011-07-28 14:48:39.163987 [DEBUG] switch_ivr_originate.c:3308 Originate
Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
2011-07-28 14:48:39.163987 [INFO] mod_dptools.c:2356 Originate Failed.
Cause: INVALID_NUMBER_FORMAT
----------------------------------------------------------------------------
---
Regards
Srini


-----Original Message-----
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:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Dmitry
Saratsky
Sent: Thursday, July 28, 2011 2:23 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Trunk without authentication configuration

I think you should try to change "custom_carrier" under dialplan and
sip_profile to "xxx.xxx.xxx.xxx":


  <list name="custom_carrier" default="deny">
    <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/>
  </list>
In acl.conf.xml

Created dialplan\default\01_custom_carrier.xml file as <include>
<!-- Dial any 10 digit number 1(2223334444) here -->
       <extension name="Dial Out Custom Gateway">
               <condition field="destination_number"
expression="^9(1\d{10})$">
                       <action application="bridge"
data="sofia/gateway/xxx.xxx.xxx.xxx/$1"/>
               </condition>
       </extension>
</include>

Created sip_profiles\external\custom_carrier.xml
<include>
       <gateway name="xxx.xxx.xxx.xxx">
               <param name="username" value="anything"/>
               <param name="password" value="anything"/>
               <param name="auth-calls" value="false"/>
               <param name="apply-outbound-acl" value="custom_carrier"/>
               <param name="apply-inbound-acl" value="custom_carrier"/>

               <param name="register" value="false"/>
       </gateway>
</include>


Regards,
Dmitry.

-----Original Message-----
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:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of
Srinivasan Krishnamurthy
Sent: 28 Jul 2011 20:53
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Trunk without authentication configuration

I have configured a new gateway for outbound calls from FS as

  <list name="custom_carrier" default="deny">
    <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/>
  </list>
In acl.conf.xml

Created dialplan\default\01_custom_carrier.xml file as <include>
<!-- Dial any 10 digit number 1(2223334444) here -->
       <extension name="Dial Out Custom Gateway">
               <condition field="destination_number"
expression="^9(1\d{10})$">
                       <action application="bridge"
data="sofia/gateway/custom_carrier/$1"/>
               </condition>
       </extension>
</include>

Created sip_profiles\external\custom_carrier.xml
<include>
       <gateway name="custom_carrier">
               <param name="username" value="anything"/>
               <param name="password" value="anything"/>
               <param name="auth-calls" value="false"/>
               <param name="apply-outbound-acl" value="custom_carrier"/>
               <param name="apply-inbound-acl" value="custom_carrier"/>

               <param name="register" value="false"/>
       </gateway>
</include>

When I make an outbound call from registered extension (1001) xlite I get
the following error. (I also tried using SJ phone)

2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel
sofia/external/14152484142<tel:14152484142> entering state [terminated][503]
2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup
sofia/external/14152484142<tel:14152484142> [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal
sofia/external/14152484142<tel:14152484142> [KILL]
2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal
sofia/external/14152484142<tel:14152484142> [BREAK]
2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314
(sofia/external/14152484142<tel:14152484142>) Running State Change CS_HANGUP
2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498
(sofia/external/14152484142<tel:14152484142>) State HANGUP
2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435
sofia/external/14152484142<tel:14152484142> Overriding SIP cause 503 with 503 from the other
leg

I have attached the complete log too.

Iam a new FC users. I have referred to the following thread
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.htm
l

and also read the section "Connecting to the world with gateways" in
FreeSwitch book several times.

Any help would be appreciated.

Note: Iam able to make outbond calls from softphone through the trunk
without FS.

Srini


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

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_______________________________________________
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-------------- next part --------------
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From msc at freeswitch.org  Fri Jul 29 21:33:41 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jul 2011 10:33:41 -0700
Subject: [Freeswitch-users] Trunk without authentication configuration
In-Reply-To: <237050FF255D844CBDF840D42334EB9B6783F80F8E@SFOEXM02.YELLOWPAGES.LOCAL>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
	<CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>
	<04020B71-9CAA-458A-98D0-189A82E8F5F1@bryansmart.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F97@cooper>
	<237050FF255D844CBDF840D42334EB9B6783F80F88@SFOEXM02.YELLOWPAGES.LOCAL>
	<000001cc4d6c$808ddfd0$81a99f70$@com>
	<237050FF255D844CBDF840D42334EB9B6783F80F8A@SFOEXM02.YELLOWPAGES.LOCAL>
	<000101cc4d8a$947a2440$bd6e6cc0$@com>
	<CAGRr058eCwzSrM=ovjGk3osJp56M8f=Youc0ge5TBSgsWfjHsg@mail.gmail.com>
	<237050FF255D844CBDF840D42334EB9B6783F80F8E@SFOEXM02.YELLOWPAGES.LOCAL>
Message-ID: <CAKzWOxUC8tA4OYcXiuZ2P=yaCprUrvOzZ-N3DRgP_k7EAMk1Cg@mail.gmail.com>

Srini,

I think you misunderstood what my suggestion. What you've got in your
gateway definition are parameters that belong in the SIP profile. Again,
look in conf/sip_profiles/external/example.xml. Don't add any parameters
that are not already in that file. A gateway definition should be no more
complicated than this:

<include>

  <gateway name="custom_carrier">

    <param name="username" value="my_username"/>

    <param name="realm" value="target_realm"/>

    <param name="password" value="my_password"/>

    <param name="register" value="false"/>
  </gateway>
</include>

Also, when doing a sip trace, simply do this:
sofia global siptrace on

If you do that sofialoglevel stuff you'll end up with way more information
than you need.

-MC

On Fri, Jul 29, 2011 at 10:19 AM, Srinivasan Krishnamurthy <
skrishnamurthy at attinteractive.com> wrote:

> Now my sip_profiles\external\custom_carrier.xml looks like****
>
> ** **
>
> <include>****
>
>                 <gateway name="custom_carrier">****
>
>                                 <param name="auth-calls" value="false"/>**
> **
>
>                                 <param name="apply-outbound-acl"
> value="custom_carrier"/>****
>
>                                 <param name="apply-inbound-acl"
> value="custom_carrier"/>                  ****
>
>                 </gateway>****
>
> </include>****
>
> ** **
>
> I have restarted FS.  Enabled sofia log in debug ?sofia tracelevel 6? and
> sip trace is on too.****
>
> Refer to complete log @ http://pastebin.freeswitch.org/16918****
>
> ** **
>
> 2011-07-29 10:00:31.372738 [ERR] mod_sofia.c:3492 Invalid Gateway****
>
> 2011-07-29 10:00:31.372738 [NOTICE] mod_sofia.c:3779 Close Channel N/A
> [CS_NEW]****
>
> 2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:428 ()
> Running State Change CS_DESTROY****
>
> 2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:438 (N/A)
> State DESTROY****
>
> 2011-07-29 10:00:31.372738 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY****
>
> 2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:438 (N/A)
> State DESTROY going to sleep****
>
> 2011-07-29 10:00:31.372738 [ERR] switch_ivr_originate.c:2493 Cannot create
> outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]****
>
> ** **
>
> Regards****
>
> Srini****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brad Mina
> *Sent:* Thursday, July 28, 2011 7:33 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Trunk without authentication
> configuration****
>
> ** **
>
> Might want to doublecheck the format your carrier accepts for outgoing
> calls.****
>
> ** **
>
> Also, attatch a full siptrace with your debug log, that helps a lot.
>
> freeswitch> sofia profile external siptrace on****
>
> On Thu, Jul 28, 2011 at 5:58 PM, Dmitry Saratsky <freeswitch at simpot.com>
> wrote:****
>
> Did you restarted FS after that change? Or you only have reloaded xmls???*
> ***
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Srinivasan Krishnamurthy****
>
> Sent: 29 Jul 2011 00:55****
>
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Trunk without authentication configuration
>
> Thanks for your suggestion. I tried but got Invalid gateway err!!!
>
>
> ----------------------------------------------------------------------------
> -
> EXECUTE sofia/internal/1001 at 10.10.2.3:5070
> bridge(sofia/gateway/xxx.xxx.xxx.xxx/14152484142)
> 2011-07-28 14:48:39.163987 [ERR] mod_sofia.c:3492 Invalid Gateway
> 2011-07-28 14:48:39.163987 [NOTICE] mod_sofia.c:3779 Close Channel N/A
> [CS_NEW]
> 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:428 ()
> Running State Change CS_DESTROY
> 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A)
> State DESTROY
> 2011-07-28 14:48:39.163987 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY
> 2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A)
> State DESTROY going to sleep
> 2011-07-28 14:48:39.163987 [ERR] switch_ivr_originate.c:2493 Cannot create
> outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
> 2011-07-28 14:48:39.163987 [DEBUG] switch_ivr_originate.c:3308 Originate
> Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
> 2011-07-28 14:48:39.163987 [INFO] mod_dptools.c:2356 Originate Failed.
> Cause: INVALID_NUMBER_FORMAT
>
> ----------------------------------------------------------------------------
> ---
> Regards
> Srini
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Dmitry
> Saratsky
> Sent: Thursday, July 28, 2011 2:23 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Trunk without authentication configuration
>
> I think you should try to change "custom_carrier" under dialplan and
> sip_profile to "xxx.xxx.xxx.xxx":
>
>
>   <list name="custom_carrier" default="deny">
>     <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/>
>   </list>
> In acl.conf.xml
>
> Created dialplan\default\01_custom_carrier.xml file as <include>
> <!-- Dial any 10 digit number 1(2223334444) here -->
>        <extension name="Dial Out Custom Gateway">
>                <condition field="destination_number"
> expression="^9(1\d{10})$">
>                        <action application="bridge"
> data="sofia/gateway/xxx.xxx.xxx.xxx/$1"/>
>                </condition>
>        </extension>
> </include>
>
> Created sip_profiles\external\custom_carrier.xml
> <include>
>        <gateway name="xxx.xxx.xxx.xxx">
>                <param name="username" value="anything"/>
>                <param name="password" value="anything"/>
>                <param name="auth-calls" value="false"/>
>                <param name="apply-outbound-acl" value="custom_carrier"/>
>                <param name="apply-inbound-acl" value="custom_carrier"/>
>
>                <param name="register" value="false"/>
>        </gateway>
> </include>
>
>
> Regards,
> Dmitry.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Srinivasan Krishnamurthy
> Sent: 28 Jul 2011 20:53
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Trunk without authentication configuration
>
> I have configured a new gateway for outbound calls from FS as
>
>   <list name="custom_carrier" default="deny">
>     <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/>
>   </list>
> In acl.conf.xml
>
> Created dialplan\default\01_custom_carrier.xml file as <include>
> <!-- Dial any 10 digit number 1(2223334444) here -->
>        <extension name="Dial Out Custom Gateway">
>                <condition field="destination_number"
> expression="^9(1\d{10})$">
>                        <action application="bridge"
> data="sofia/gateway/custom_carrier/$1"/>
>                </condition>
>        </extension>
> </include>
>
> Created sip_profiles\external\custom_carrier.xml
> <include>
>        <gateway name="custom_carrier">
>                <param name="username" value="anything"/>
>                <param name="password" value="anything"/>
>                <param name="auth-calls" value="false"/>
>                <param name="apply-outbound-acl" value="custom_carrier"/>
>                <param name="apply-inbound-acl" value="custom_carrier"/>
>
>                <param name="register" value="false"/>
>        </gateway>
> </include>
>
> When I make an outbound call from registered extension (1001) xlite I get
> the following error. (I also tried using SJ phone)
>
> 2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel
> sofia/external/14152484142 entering state [terminated][503]
> 2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup
> sofia/external/14152484142 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
> 2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal
> sofia/external/14152484142 [KILL]
> 2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal
> sofia/external/14152484142 [BREAK]
> 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/14152484142) Running State Change CS_HANGUP
> 2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498
> (sofia/external/14152484142) State HANGUP
> 2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435
> sofia/external/14152484142 Overriding SIP cause 503 with 503 from the
> other
> leg
>
> I have attached the complete log too.
>
> Iam a new FC users. I have referred to the following thread
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.htm
> l
>
> and also read the section "Connecting to the world with gateways" in
> FreeSwitch book several times.
>
> Any help would be appreciated.
>
> Note: Iam able to make outbond calls from softphone through the trunk
> without FS.
>
> Srini
>
>
> _______________________________________________
> 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
>
>
> _______________________________________________
> 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
>
>
> _______________________________________________
> 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****
>
> ** **
>
> _______________________________________________
> 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 skrishnamurthy at attinteractive.com  Fri Jul 29 22:26:10 2011
From: skrishnamurthy at attinteractive.com (Srinivasan Krishnamurthy)
Date: Fri, 29 Jul 2011 11:26:10 -0700
Subject: [Freeswitch-users] Trunk without authentication configuration
In-Reply-To: <CAKzWOxUC8tA4OYcXiuZ2P=yaCprUrvOzZ-N3DRgP_k7EAMk1Cg@mail.gmail.com>
References: <CANGieu1Rqywgog01xy6icvZkFk2Z-O7e01MsQaz2QXs7J4Yriw@mail.gmail.com>
	<CALXCt0pq5Uq0A7T2pQHEq1oG_HUAwh2u=PMFVYidDb8Z_MZddw@mail.gmail.com>
	<CANGieu39n4M-MO56T0fbZ6jOWc_imQcoHQUhGS9Fk=Wr-pEPVw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F93@cooper>
	<1311801895.24333.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxXqEbWNknZB9Aub5XSoj32wBWs7se1CdpMAq_NANRrUeg@mail.gmail.com>
	<CALNZuwkXqEAVbybsZAy0k0p8ifROqixRdsEtss_bC94atcu3GA@mail.gmail.com>
	<04020B71-9CAA-458A-98D0-189A82E8F5F1@bryansmart.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59E51D2F97@cooper>
	<237050FF255D844CBDF840D42334EB9B6783F80F88@SFOEXM02.YELLOWPAGES.LOCAL>
	<000001cc4d6c$808ddfd0$81a99f70$@com>
	<237050FF255D844CBDF840D42334EB9B6783F80F8A@SFOEXM02.YELLOWPAGES.LOCAL>
	<000101cc4d8a$947a2440$bd6e6cc0$@com>
	<CAGRr058eCwzSrM=ovjGk3osJp56M8f=Youc0ge5TBSgsWfjHsg@mail.gmail.com>
	<237050FF255D844CBDF840D42334EB9B6783F80F8E@SFOEXM02.YELLOWPAGES.LOCAL>
	<CAKzWOxUC8tA4OYcXiuZ2P=yaCprUrvOzZ-N3DRgP_k7EAMk1Cg@mail.gmail.com>
Message-ID: <237050FF255D844CBDF840D42334EB9B6783F80F8F@SFOEXM02.YELLOWPAGES.LOCAL>

Thanks Michael now it works.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, July 29, 2011 10:34 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Trunk without authentication configuration

Srini,

I think you misunderstood what my suggestion. What you've got in your gateway definition are parameters that belong in the SIP profile. Again, look in conf/sip_profiles/external/example.xml. Don't add any parameters that are not already in that file. A gateway definition should be no more complicated than this:

<include>
  <gateway name="custom_carrier">
    <param name="username" value="my_username"/>
    <param name="realm" value="target_realm"/>
    <param name="password" value="my_password"/>
    <param name="register" value="false"/>
  </gateway>
</include>

Also, when doing a sip trace, simply do this:
sofia global siptrace on

If you do that sofialoglevel stuff you'll end up with way more information than you need.

-MC

On Fri, Jul 29, 2011 at 10:19 AM, Srinivasan Krishnamurthy <skrishnamurthy at attinteractive.com<mailto:skrishnamurthy at attinteractive.com>> wrote:
Now my sip_profiles\external\custom_carrier.xml looks like

<include>
                <gateway name="custom_carrier">
                                <param name="auth-calls" value="false"/>
                                <param name="apply-outbound-acl" value="custom_carrier"/>
                                <param name="apply-inbound-acl" value="custom_carrier"/>
                </gateway>
</include>

I have restarted FS.  Enabled sofia log in debug "sofia tracelevel 6' and sip trace is on too.
Refer to complete log @ http://pastebin.freeswitch.org/16918

2011-07-29 10:00:31.372738 [ERR] mod_sofia.c:3492 Invalid Gateway
2011-07-29 10:00:31.372738 [NOTICE] mod_sofia.c:3779 Close Channel N/A [CS_NEW]
2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:428 () Running State Change CS_DESTROY
2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY
2011-07-29 10:00:31.372738 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY
2011-07-29 10:00:31.372738 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY going to sleep
2011-07-29 10:00:31.372738 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]

Regards
Srini

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:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Brad Mina
Sent: Thursday, July 28, 2011 7:33 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Trunk without authentication configuration

Might want to doublecheck the format your carrier accepts for outgoing calls.

Also, attatch a full siptrace with your debug log, that helps a lot.

freeswitch> sofia profile external siptrace on
On Thu, Jul 28, 2011 at 5:58 PM, Dmitry Saratsky <freeswitch at simpot.com<mailto:freeswitch at simpot.com>> wrote:
Did you restarted FS after that change? Or you only have reloaded xmls???

-----Original Message-----
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:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of
Srinivasan Krishnamurthy
Sent: 29 Jul 2011 00:55
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Trunk without authentication configuration

Thanks for your suggestion. I tried but got Invalid gateway err!!!

----------------------------------------------------------------------------
-
EXECUTE sofia/internal/1001 at 10.10.2.3:5070<http://1001 at 10.10.2.3:5070>
bridge(sofia/gateway/xxx.xxx.xxx.xxx/14152484142<tel:14152484142>)
2011-07-28 14:48:39.163987 [ERR] mod_sofia.c:3492 Invalid Gateway
2011-07-28 14:48:39.163987 [NOTICE] mod_sofia.c:3779 Close Channel N/A
[CS_NEW]
2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:428 ()
Running State Change CS_DESTROY
2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A)
State DESTROY
2011-07-28 14:48:39.163987 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY
2011-07-28 14:48:39.163987 [DEBUG] switch_core_state_machine.c:438 (N/A)
State DESTROY going to sleep
2011-07-28 14:48:39.163987 [ERR] switch_ivr_originate.c:2493 Cannot create
outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
2011-07-28 14:48:39.163987 [DEBUG] switch_ivr_originate.c:3308 Originate
Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
2011-07-28 14:48:39.163987 [INFO] mod_dptools.c:2356 Originate Failed.
Cause: INVALID_NUMBER_FORMAT
----------------------------------------------------------------------------
---
Regards
Srini


-----Original Message-----
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:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Dmitry
Saratsky
Sent: Thursday, July 28, 2011 2:23 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Trunk without authentication configuration

I think you should try to change "custom_carrier" under dialplan and
sip_profile to "xxx.xxx.xxx.xxx":


  <list name="custom_carrier" default="deny">
    <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/>
  </list>
In acl.conf.xml

Created dialplan\default\01_custom_carrier.xml file as <include>
<!-- Dial any 10 digit number 1(2223334444) here -->
       <extension name="Dial Out Custom Gateway">
               <condition field="destination_number"
expression="^9(1\d{10})$">
                       <action application="bridge"
data="sofia/gateway/xxx.xxx.xxx.xxx/$1"/>
               </condition>
       </extension>
</include>

Created sip_profiles\external\custom_carrier.xml
<include>
       <gateway name="xxx.xxx.xxx.xxx">
               <param name="username" value="anything"/>
               <param name="password" value="anything"/>
               <param name="auth-calls" value="false"/>
               <param name="apply-outbound-acl" value="custom_carrier"/>
               <param name="apply-inbound-acl" value="custom_carrier"/>

               <param name="register" value="false"/>
       </gateway>
</include>


Regards,
Dmitry.

-----Original Message-----
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:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of
Srinivasan Krishnamurthy
Sent: 28 Jul 2011 20:53
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Trunk without authentication configuration

I have configured a new gateway for outbound calls from FS as

  <list name="custom_carrier" default="deny">
    <node type="allow" cidr=xxx.xxx.xxx.xxx/32"/>
  </list>
In acl.conf.xml

Created dialplan\default\01_custom_carrier.xml file as <include>
<!-- Dial any 10 digit number 1(2223334444) here -->
       <extension name="Dial Out Custom Gateway">
               <condition field="destination_number"
expression="^9(1\d{10})$">
                       <action application="bridge"
data="sofia/gateway/custom_carrier/$1"/>
               </condition>
       </extension>
</include>

Created sip_profiles\external\custom_carrier.xml
<include>
       <gateway name="custom_carrier">
               <param name="username" value="anything"/>
               <param name="password" value="anything"/>
               <param name="auth-calls" value="false"/>
               <param name="apply-outbound-acl" value="custom_carrier"/>
               <param name="apply-inbound-acl" value="custom_carrier"/>

               <param name="register" value="false"/>
       </gateway>
</include>

When I make an outbound call from registered extension (1001) xlite I get
the following error. (I also tried using SJ phone)

2011-07-28 10:10:24.388677 [DEBUG] sofia.c:4200 Channel
sofia/external/14152484142<tel:14152484142> entering state [terminated][503]
2011-07-28 10:10:24.388677 [NOTICE] sofia.c:4836 Hangup
sofia/external/14152484142<tel:14152484142> [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2011-07-28 10:10:24.388677 [DEBUG] switch_channel.c:2145 Send signal
sofia/external/14152484142<tel:14152484142> [KILL]
2011-07-28 10:10:24.388677 [DEBUG] switch_core_session.c:1022 Send signal
sofia/external/14152484142<tel:14152484142> [BREAK]
2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:314
(sofia/external/14152484142<tel:14152484142>) Running State Change CS_HANGUP
2011-07-28 10:10:24.388677 [DEBUG] switch_core_state_machine.c:498
(sofia/external/14152484142<tel:14152484142>) State HANGUP
2011-07-28 10:10:24.388677 [DEBUG] mod_sofia.c:435
sofia/external/14152484142<tel:14152484142> Overriding SIP cause 503 with 503 from the other
leg

I have attached the complete log too.

Iam a new FC users. I have referred to the following thread
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070294.htm
l

and also read the section "Connecting to the world with gateways" in
FreeSwitch book several times.

Any help would be appreciated.

Note: Iam able to make outbond calls from softphone through the trunk
without FS.

Srini


_______________________________________________
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<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


_______________________________________________
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<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


_______________________________________________
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<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


_______________________________________________
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<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 lloydie.t at gmail.com  Fri Jul 29 22:38:25 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Fri, 29 Jul 2011 19:38:25 +0100
Subject: [Freeswitch-users] Help setting up SIP reg
Message-ID: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>

*Hi I need a little help setting up a SIP registration for a provider that
does not use auth.*

*All I have is info below.*
**

*
*


SBC/Proxy IP:        80.40.150.150:5060

Authentication:      Trusted IP ? 88.221.85.33

Assigned DDI:        01869******, 01869******

DTMF Method:         RFC2833

Status:              Live

No. of trunks:       2x

Session Timer:       1800

Profile*:*             Generic (35060)


Apparently the following is used for *

[vibe]

type = friend

host = 80.40.150.150
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From steveayre at gmail.com  Fri Jul 29 22:55:53 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 29 Jul 2011 19:55:53 +0100
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
Message-ID: <CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>

Look at the cidr attribute in the user directory to authenticate by IP:
http://wiki.freeswitch.org/wiki/Acl#Users

-Steve

On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com> wrote:

> *Hi I need a little help setting up a SIP registration for a provider that
> does not use auth.*
>
> *All I have is info below.*
> **
>
> *
> *
>
>
> SBC/Proxy IP:        80.40.150.150:5060
>
> Authentication:      Trusted IP ? 88.221.85.33
>
> Assigned DDI:        01869******, 01869******
>
> DTMF Method:         RFC2833
>
> Status:              Live
>
> No. of trunks:       2x
>
> Session Timer:       1800
>
> Profile*:*             Generic (35060)
>
>
> Apparently the following is used for *
>
> [vibe]
>
> type = friend
>
> host = 80.40.150.150
>
> _______________________________________________
> 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 msc at freeswitch.org  Fri Jul 29 23:10:48 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jul 2011 12:10:48 -0700
Subject: [Freeswitch-users] invoking php scripts from dialplan
In-Reply-To: <1311205533.33756.YahooMailClassic@web161011.mail.bf1.yahoo.com>
References: <1311205533.33756.YahooMailClassic@web161011.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxV0iLDBfc37po=svoH5xL-y6+pTQF7OcuxLtL_xca3eFQ@mail.gmail.com>

Sorry for the delayed response...

I went back to reproduce your symptoms and I noticed something with the port
number you chose (8021). What other REALLY important service is running on
that port? Hint: event socket.

I tried this exact config using port 8040 instead of 8021 and it worked like
a champ.

-MC

On Wed, Jul 20, 2011 at 4:45 PM, Sam <lakersman2006 at yahoo.com> wrote:

> Hi,
>
> I wanted to know is there another method of invoking a php script from the
> dialplan besides using the following because I still cannot get the script
> to run for wahtever reason. Yes, I have opened up the ports on my firewall.
> When I run the php script directly from console it outputs fine, but i
> cannot get it to output via the dialplan.
>
> <!-- dialplan.xml -->
>
> <extension name="outbound-socket">
> <condition field="destination_number" expression="^55(522)$">
> <action application="set" data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/>
> <action application="socket" data="127.0.0.1:8021 async full"/>
> </condition>
> </extension>
>
> //I then ran fs_ivrd with this command:
> /usr/local/freeswitch/bin/fs_ivrd -h 127.0.0.1 -p 8021
>
>
> //my php script: ivrd-demo.php
> #!/usr/bin/php -q
>
> <?php
>
> // set a couple of things so we dont kill the system
> ob_implicit_flush(true);
> set_time_limit(30);
>
> // Open stdin so we can read the data in
> $in = fopen("php://stdin", "r");
>
> // Connect
> echo "connect\n\n";
>
> // Answer
> echo "sendmsg\n";
> echo "call-command: execute\n";
> echo "execute-app-name: answer\n\n";
>
> // Play a prompt
> echo "sendmsg\n";
> echo "call-command: execute\n";
> echo "execute-app-name: playback\n";
> echo "execute-app-arg: /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n";
>
> // Wait
> sleep(5);
>
> // Hangup
> echo "sendmsg\n";
> echo "call-command: hangup\n\n";
>
> fclose($in);
>
> ?>
>
>
>
> _______________________________________________
> 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 erin.omeara at salmonbaytechnology.com  Fri Jul 29 23:38:29 2011
From: erin.omeara at salmonbaytechnology.com (Erin O'Meara)
Date: Fri, 29 Jul 2011 12:38:29 -0700
Subject: [Freeswitch-users] No audio, only on outgoing calls
In-Reply-To: <CAB4Oy0UgRWg6Dg-BXJKyQMcGCfO+8nbDT1Kn0PhPXe=A=w4-1A@mail.gmail.com>
References: <CAB4Oy0UgRWg6Dg-BXJKyQMcGCfO+8nbDT1Kn0PhPXe=A=w4-1A@mail.gmail.com>
Message-ID: <CAB4Oy0Vdxkqe6PLA5xjb439AFaJDYXabYcCvtgtC18=vOamg2A@mail.gmail.com>

All Working.

Switched out phones, the model does not seem to work with Freeswitch.

Strange that incoming calls worked but not outgoing.

Phone was a GrandStream BT201

Regards,

206.905.9520
http://salmonbaytechnology.com



On Thu, Jul 28, 2011 at 6:43 PM, Erin O'Meara <
erin.omeara at salmonbaytechnology.com> wrote:

> I have a blue.box server setup with a cloudhost, setup IPKall & DIDforSale.
> Both incoming calls come thru to my phone connected to the blue.box, I setup
> outgoing calls with CallwithUs and a Toll Free termination provider. All
> outgoing calls from the phone thru the blue.box connect, but no audio in
> either direction.
>
> The phone is behind a PFSense Firewall/Router and I have a second phone
> connected to an Elastix PBX that I use for my day to day business and
> everything works as expected. I have disable the Block All firewall rule and
> replaced it with Pass All and am pretty sure at this point its not a
> Firewall issue.
>
> When I dial the extension's number, I can see voicemail wav files playing
> but again hear nothing. So from what I can determine this is a freeswitch to
> phone issue.
>
> I have chatted on IRC with people at #2600 and #Freeswitch and googled
> the symptoms and come up no solution.
>
> Hoping some has seen this before and has a solution so I don't have to
> smash anything in frustration.
>
>  Regards,
>
> 206.905.9520
> http://salmonbaytechnology.com
>
>
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From gavin.henry at gmail.com  Fri Jul 29 23:43:41 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 29 Jul 2011 20:43:41 +0100
Subject: [Freeswitch-users] DTMF echo/crossed?
In-Reply-To: <CALdmMT6LZPaJu92Hb1CBEtRkEmb4tBzjJdh3bFLyP+ECrpGVUg@mail.gmail.com>
References: <CAA8_NKAPZcSXjBzRuxxAiZrChOZF17zZ_iG6L3jg7tjmMjbTzw@mail.gmail.com>
	<CALdmMT6LZPaJu92Hb1CBEtRkEmb4tBzjJdh3bFLyP+ECrpGVUg@mail.gmail.com>
Message-ID: <CAA8_NKDcyE3foSFeahBS-Wq4heGunG0hVoMn4_Tvk1-P+yudPg@mail.gmail.com>

On 29 July 2011 16:56, Norman Tomlins <norm at voicenetwork.ca> wrote:
> Gavin,
>
> If you are using the LinkSys SPA2102's you will get DTMF digits when
> people are talking. ? You will need to increase the
> DTMF_Tx_Strict_Hold_Off_Time to about 80.
>
> This will stop random DTMF while people are talking

OK, thanks. This was on Yealink T28p and a Panasonic KX-TGP550 SIP phone system


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


From rupa at rupa.com  Sat Jul 30 00:33:35 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 29 Jul 2011 15:33:35 -0500
Subject: [Freeswitch-users] curl.run and curl parameter
In-Reply-To: <BANLkTimgBJHhSw=2xmQWwdqyFoUWXeAc5Q@mail.gmail.com>
References: <BANLkTimgBJHhSw=2xmQWwdqyFoUWXeAc5Q@mail.gmail.com>
Message-ID: <CAJw9H-MjPP9K+swJ_F5tZfa6OAD+UBs0oAOvkHXmHap07_0SoQ@mail.gmail.com>

On Fri, Jul 1, 2011 at 2:02 AM, Daniel Knoll <daniel at danielknoll.de> wrote:
> Hey Freeswitch and Javascript Guys,
>
> i would like to use curl.run to post a file to googles speech api. How
> can i add curl parameters like "-F filename" to pass the filename.
>
> curl.run("POST", "https://www.google.com/speech-api/v1/",
> "recognize?xjerr=1&client=chromium&lang=de-DE&lm=builtin:dictation",
> my_callback, "my callback_arg\n",, Content-Type: audio/x-flac;
> rate=16000);
>
>
> Has anyone an idea ?

no.  That would be a new feature.  If you want it, jira is the right
place to ask for it.

-- 
-Rupa


From lloydie.t at gmail.com  Sat Jul 30 00:40:11 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Fri, 29 Jul 2011 21:40:11 +0100
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
Message-ID: <CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>

Sorry, example is not clear to me.
I don't understand why a user config is relevant to sip registration for a
provider.
An example will help me more. Maybe CIDR attribute in a sip_profile gateway
could help.


On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com> wrote:

> Look at the cidr attribute in the user directory to authenticate by IP:
> http://wiki.freeswitch.org/wiki/Acl#Users
>
> -Steve
>
> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com> wrote:
>
>> *Hi I need a little help setting up a SIP registration for a provider
>> that does not use auth.*
>>
>> *All I have is info below.*
>> **
>>
>> *
>> *
>>
>>
>> SBC/Proxy IP:        80.40.150.150:5060
>>
>> Authentication:      Trusted IP ? 88.221.85.33
>>
>> Assigned DDI:        01869******, 01869******
>>
>> DTMF Method:         RFC2833
>>
>> Status:              Live
>>
>> No. of trunks:       2x
>>
>> Session Timer:       1800
>>
>> Profile*:*             Generic (35060)
>>
>>
>> Apparently the following is used for *
>>
>> [vibe]
>>
>> type = friend
>>
>> host = 80.40.150.150
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 leo at megapin.com  Sat Jul 30 00:14:57 2011
From: leo at megapin.com (Leo)
Date: Fri, 29 Jul 2011 13:14:57 -0700
Subject: [Freeswitch-users] Please Help! CANCEL 481 Problem Call dons't hunup
Message-ID: <5B4F4E33BA91412DB98475E6967DE386@LEODE9798812EB>

Hi:

I have an issue with FS connected to a SIP gateway. After the Freeswitch update(FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05 -0500).

When I receive a call from the SIP Gateway and try to hangup with "CANCEL" message the call before he call is answered,

Our freeswitch replay  SIP 481 Call Transaction does not exist. 

Please help!

Leo Liu
---------------------------------------------------------------
INVITE sip:9501351350010 at 38.106.1.72 SIP/2.0
Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
From: "calling from" <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
To: "9501351350010" <sip:9501351350010 at 38.106.1.72>
Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
Contact: <sip:8675582928388 at 204.110.11.29:5060>
CSeq: 237 INVITE
Max-Forwards: 70
User-Agent: ABC-Phone
Supported: replaces
Content-Type: application/sdp
Content-Length: 141

v=0
o=- 34791986 84556339 IN IP4 204.110.11.29
s=SIP CALL
c=IN IP4 204.110.11.29
t=0 0
m=audio 35974 RTP/AVP 18
a=rtpmap:18 G729/8000

-----------------------------------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
From: "calling from" <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
To: "9501351350010" <sip:9501351350010 at 38.106.1.72>
Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
CSeq: 237 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05 -0500
Content-Length: 0

-----------------------------------------------------------------------------------------------------
INVITE sip:88359501351350010 at 38.106.1.83 SIP/2.0
Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKBvFBUj7NNcQ1B
Max-Forwards: 69
From: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
To: <sip:88359501351350010 at 38.106.1.83>
Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
CSeq: 15529052 INVITE
Contact: <sip:mod_sofia at 38.106.1.72:5080>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05 -0500
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: 141
X-FS-Support: update_display
Remote-Party-ID: "calling from" <sip:8675582928388 at 38.106.1.72>;party=calling;screen=yes;privacy=off

v=0
o=- 34791986 84556339 IN IP4 204.110.11.29
s=SIP CALL
c=IN IP4 204.110.11.29
t=0 0
m=audio 35974 RTP/AVP 18
a=rtpmap:18 G729/8000

-----------------------------------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKBvFBUj7NNcQ1B
From: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
To: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
Date: Fri, 11 Aug 2000 05:54:04 GMT
Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 15529052 INVITE
Allow-Events: telephone-event
Content-Length: 0

-----------------------------------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKBvFBUj7NNcQ1B
From: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
To: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
Date: Fri, 11 Aug 2000 05:54:04 GMT
Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 15529052 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:88359501351350010 at 38.106.1.83:5060>
Content-Type: application/sdp
Content-Length: 200

v=0
o=CiscoSystemsSIP-GW-UserAgent 8984 5356 IN IP4 38.106.1.83
s=SIP Call
c=IN IP4 38.106.1.83
t=0 0
m=audio 17050 RTP/AVP 18
c=IN IP4 38.106.1.83
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes



ACK sip:88359501351350010 at 38.106.1.83:5060 SIP/2.0
Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKc583vDrSjNDmQ
Max-Forwards: 70
From: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
To: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
CSeq: 15529052 ACK
Contact: <sip:mod_sofia at 38.106.1.72:5080>
Content-Length: 0


SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
From: "calling from" <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
CSeq: 237 INVITE
Contact: <sip:9501351350010 at 38.106.1.72:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05 -0500
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-Type: application/sdp
Content-Disposition: session
Content-Length: 200
Remote-Party-ID: "Outbound Call" <sip:88359501351350010 at 38.106.1.72>;party=calling;privacy=off;screen=no

v=0
o=CiscoSystemsSIP-GW-UserAgent 8984 5356 IN IP4 38.106.1.83
s=SIP Call
c=IN IP4 38.106.1.83
t=0 0
m=audio 17050 RTP/AVP 18
c=IN IP4 38.106.1.83
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes

-----------------------------------------------------------------------------------------------------
ACK sip:9501351350010 at 38.106.1.72 SIP/2.0
Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
From: "calling from" <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
Contact: <sip:8675582928388 at 204.110.11.29:5060>
Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
CSeq: 237 ACK
Max-Forwards: 70
User-Agent: ABC-Phone
Content-Length: 0

-----------------------------------------------------------------------------------------------------
CANCEL sip:9501351350010 at 38.106.1.72 SIP/2.0
Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
From: "calling from" <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
CSeq: 237 CANCEL
Max-Forwards: 70
Content-Length: 0

-----------------------------------------------------------------------------------------------------
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
From: "calling from" <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
CSeq: 237 CANCEL
Content-Length: 0


-----------------------------------------------------------------------------------------------------
ACK sip:9501351350010 at 38.106.1.72 SIP/2.0
Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
From: "calling from" <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
Contact: <sip:8675582928388 at 204.110.11.29:5060>
Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
CSeq: 237 ACK
Max-Forwards: 70
User-Agent: ABC-Phone
Content-Length: 0

-----------------------------------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 38.106.1.83:5060;branch=z9hG4bK2CE48C77
From: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
To: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
CSeq: 101 BYE
Timestamp: 965973271 0.000381
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05 -0500
Content-Length: 0

-----------------------------------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 38.106.1.83:5060;branch=z9hG4bK2CE48C77
From: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
To: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
CSeq: 101 BYE
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05 -0500
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Content-Length: 0

-----------------------------------------------------------------------------------------------------
BYE sip:8675582928388 at 204.110.11.29:5060 SIP/2.0
Via: SIP/2.0/UDP 38.106.1.72;rport;branch=z9hG4bKSj5XgvcHUpm6c
Max-Forwards: 70
From: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
To: "calling from" <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
CSeq: 15529071 BYE
Contact: <sip:9501351350010 at 38.106.1.72:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05 -0500
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=102;text="RECOVERY_ON_TIMER_EXPIRE"
Content-Length: 0


-----------------------------------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 38.106.1.72;rport;branch=z9hG4bKSj5XgvcHUpm6c
From: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
To: "calling from" <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
CSeq: 15529071 BYE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0




__________ Information from ESET NOD32 Antivirus, version of virus signature database 6335 (20110729) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From anthony.minessale at gmail.com  Sat Jul 30 00:57:03 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 29 Jul 2011 15:57:03 -0500
Subject: [Freeswitch-users] Please Help! CANCEL 481 Problem Call dons't
	hunup
In-Reply-To: <5B4F4E33BA91412DB98475E6967DE386@LEODE9798812EB>
References: <5B4F4E33BA91412DB98475E6967DE386@LEODE9798812EB>
Message-ID: <CAKbxfG_RM--mL2-OjgYssBHrO+-baA-4xYjco-CuFHZToCvE2w@mail.gmail.com>

1) You did not try the latest trunk before reporting an issue.
2) You are reporting a bug on the mailing list please see
http://jira.freeswitch.org
3) You need to get a trace of the FS console
    sofia global siptrace on first
4) At quick glance, it looks like you are sending a cancel to a
channel that has already been answered so you are having some kind of
nat problem where the 200ok from FS is not reaching the caller.
5) in case you forgot point 1, try latest git


2011/7/29 Leo <leo at megapin.com>:
> Hi:
>
> I have an issue with FS connected to a SIP gateway. After the Freeswitch
> update(FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05 -0500).
>
> When I receive a call from the SIP Gateway and try to hangup?with
> "CANCEL"?message the call before he call is answered,
>
> Our freeswitch?replay ?SIP 481 Call Transaction does not exist.
>
> Please help!
>
> Leo Liu
> ---------------------------------------------------------------
> INVITE sip:9501351350010 at 38.106.1.72 SIP/2.0
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> To: "9501351350010" <sip:9501351350010 at 38.106.1.72>
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> Contact: <sip:8675582928388 at 204.110.11.29:5060>
> CSeq: 237 INVITE
> Max-Forwards: 70
> User-Agent: ABC-Phone
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 141
>
> v=0
> o=- 34791986 84556339 IN IP4 204.110.11.29
> s=SIP CALL
> c=IN IP4 204.110.11.29
> t=0 0
> m=audio 35974 RTP/AVP 18
> a=rtpmap:18 G729/8000
>
> -----------------------------------------------------------------------------------------------------
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> To: "9501351350010" <sip:9501351350010 at 38.106.1.72>
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 INVITE
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500
> Content-Length: 0
>
> -----------------------------------------------------------------------------------------------------
> INVITE sip:88359501351350010 at 38.106.1.83 SIP/2.0
> Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKBvFBUj7NNcQ1B
> Max-Forwards: 69
> From: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> To: <sip:88359501351350010 at 38.106.1.83>
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> CSeq: 15529052 INVITE
> Contact: <sip:mod_sofia at 38.106.1.72:5080>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500
> 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: 141
> X-FS-Support: update_display
> Remote-Party-ID: "calling from"
> <sip:8675582928388 at 38.106.1.72>;party=calling;screen=yes;privacy=off
>
> v=0
> o=- 34791986 84556339 IN IP4 204.110.11.29
> s=SIP CALL
> c=IN IP4 204.110.11.29
> t=0 0
> m=audio 35974 RTP/AVP 18
> a=rtpmap:18 G729/8000
>
> -----------------------------------------------------------------------------------------------------
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKBvFBUj7NNcQ1B
> From: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> To: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
> Date: Fri, 11 Aug 2000 05:54:04 GMT
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> Server: Cisco-SIPGateway/IOS-12.x
> CSeq: 15529052 INVITE
> Allow-Events: telephone-event
> Content-Length: 0
>
> -----------------------------------------------------------------------------------------------------
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKBvFBUj7NNcQ1B
> From: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> To: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
> Date: Fri, 11 Aug 2000 05:54:04 GMT
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> Server: Cisco-SIPGateway/IOS-12.x
> CSeq: 15529052 INVITE
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
> NOTIFY, INFO, UPDATE, REGISTER
> Allow-Events: telephone-event
> Contact: <sip:88359501351350010 at 38.106.1.83:5060>
> Content-Type: application/sdp
> Content-Length: 200
>
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 8984 5356 IN IP4 38.106.1.83
> s=SIP Call
> c=IN IP4 38.106.1.83
> t=0 0
> m=audio 17050 RTP/AVP 18
> c=IN IP4 38.106.1.83
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=yes
>
>
>
> ACK sip:88359501351350010 at 38.106.1.83:5060 SIP/2.0
> Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKc583vDrSjNDmQ
> Max-Forwards: 70
> From: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> To: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> CSeq: 15529052 ACK
> Contact: <sip:mod_sofia at 38.106.1.72:5080>
> Content-Length: 0
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 INVITE
> Contact: <sip:9501351350010 at 38.106.1.72:5060;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500
> 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-Type: application/sdp
> Content-Disposition: session
> Content-Length: 200
> Remote-Party-ID: "Outbound Call"
> <sip:88359501351350010 at 38.106.1.72>;party=calling;privacy=off;screen=no
>
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 8984 5356 IN IP4 38.106.1.83
> s=SIP Call
> c=IN IP4 38.106.1.83
> t=0 0
> m=audio 17050 RTP/AVP 18
> c=IN IP4 38.106.1.83
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=yes
>
> -----------------------------------------------------------------------------------------------------
> ACK sip:9501351350010 at 38.106.1.72 SIP/2.0
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> Contact: <sip:8675582928388 at 204.110.11.29:5060>
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 ACK
> Max-Forwards: 70
> User-Agent: ABC-Phone
> Content-Length: 0
>
> -----------------------------------------------------------------------------------------------------
> CANCEL sip:9501351350010 at 38.106.1.72 SIP/2.0
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 CANCEL
> Max-Forwards: 70
> Content-Length: 0
>
> -----------------------------------------------------------------------------------------------------
> SIP/2.0 481 Call/Transaction Does Not Exist
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 CANCEL
> Content-Length: 0
>
> -----------------------------------------------------------------------------------------------------
> ACK sip:9501351350010 at 38.106.1.72 SIP/2.0
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> Contact: <sip:8675582928388 at 204.110.11.29:5060>
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 ACK
> Max-Forwards: 70
> User-Agent: ABC-Phone
> Content-Length: 0
>
> -----------------------------------------------------------------------------------------------------
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 38.106.1.83:5060;branch=z9hG4bK2CE48C77
> From: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
> To: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> CSeq: 101 BYE
> Timestamp: 965973271 0.000381
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500
> Content-Length: 0
>
> -----------------------------------------------------------------------------------------------------
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 38.106.1.83:5060;branch=z9hG4bK2CE48C77
> From: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
> To: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> CSeq: 101 BYE
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Content-Length: 0
>
> -----------------------------------------------------------------------------------------------------
> BYE sip:8675582928388 at 204.110.11.29:5060 SIP/2.0
> Via: SIP/2.0/UDP 38.106.1.72;rport;branch=z9hG4bKSj5XgvcHUpm6c
> Max-Forwards: 70
> From: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> To: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 15529071 BYE
> Contact: <sip:9501351350010 at 38.106.1.72:5060;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Reason: Q.850;cause=102;text="RECOVERY_ON_TIMER_EXPIRE"
> Content-Length: 0
>
> -----------------------------------------------------------------------------------------------------
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 38.106.1.72;rport;branch=z9hG4bKSj5XgvcHUpm6c
> From: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> To: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 15529071 BYE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Length: 0
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature
> database 6335 (20110729) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
> _______________________________________________
> 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
>
>



-- 
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 sos at sokhapkin.dyndns.org  Sat Jul 30 00:57:40 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 29 Jul 2011 16:57:40 -0400
Subject: [Freeswitch-users] Please Help! CANCEL 481 Problem Call dons't
	hunup
In-Reply-To: <5B4F4E33BA91412DB98475E6967DE386@LEODE9798812EB>
References: <5B4F4E33BA91412DB98475E6967DE386@LEODE9798812EB>
Message-ID: <201107291657.40246.sos@sokhapkin.dyndns.org>

CANCEL is applicable to unanswered call only. You're trying to 'cancel' 
answered call. Only BYE is applicable this case.

On Friday 29 July 2011, Leo wrote:
> Hi:
> 
> I have an issue with FS connected to a SIP gateway. After the Freeswitch
> update(FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500).
> 
> When I receive a call from the SIP Gateway and try to hangup with "CANCEL"
> message the call before he call is answered,
> 
> Our freeswitch replay  SIP 481 Call Transaction does not exist.
> 
> Please help!
> 
> Leo Liu
> ---------------------------------------------------------------
> INVITE sip:9501351350010 at 38.106.1.72 SIP/2.0
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236
> T6 To: "9501351350010" <sip:9501351350010 at 38.106.1.72>
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> Contact: <sip:8675582928388 at 204.110.11.29:5060>
> CSeq: 237 INVITE
> Max-Forwards: 70
> User-Agent: ABC-Phone
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 141
> 
> v=0
> o=- 34791986 84556339 IN IP4 204.110.11.29
> s=SIP CALL
> c=IN IP4 204.110.11.29
> t=0 0
> m=audio 35974 RTP/AVP 18
> a=rtpmap:18 G729/8000
> 
> ---------------------------------------------------------------------------
> -------------------------- SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236
> T6 To: "9501351350010" <sip:9501351350010 at 38.106.1.72>
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 INVITE
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500 Content-Length: 0
> 
> ---------------------------------------------------------------------------
> -------------------------- INVITE sip:88359501351350010 at 38.106.1.83 SIP/2.0
> Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKBvFBUj7NNcQ1B
> Max-Forwards: 69
> From: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> To: <sip:88359501351350010 at 38.106.1.83>
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> CSeq: 15529052 INVITE
> Contact: <sip:mod_sofia at 38.106.1.72:5080>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500 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: 141
> X-FS-Support: update_display
> Remote-Party-ID: "calling from"
> <sip:8675582928388 at 38.106.1.72>;party=calling;screen=yes;privacy=off
> 
> v=0
> o=- 34791986 84556339 IN IP4 204.110.11.29
> s=SIP CALL
> c=IN IP4 204.110.11.29
> t=0 0
> m=audio 35974 RTP/AVP 18
> a=rtpmap:18 G729/8000
> 
> ---------------------------------------------------------------------------
> -------------------------- SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKBvFBUj7NNcQ1B
> From: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> To: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
> Date: Fri, 11 Aug 2000 05:54:04 GMT
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> Server: Cisco-SIPGateway/IOS-12.x
> CSeq: 15529052 INVITE
> Allow-Events: telephone-event
> Content-Length: 0
> 
> ---------------------------------------------------------------------------
> -------------------------- SIP/2.0 200 OK
> Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKBvFBUj7NNcQ1B
> From: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> To: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
> Date: Fri, 11 Aug 2000 05:54:04 GMT
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> Server: Cisco-SIPGateway/IOS-12.x
> CSeq: 15529052 INVITE
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
> NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event
> Contact: <sip:88359501351350010 at 38.106.1.83:5060>
> Content-Type: application/sdp
> Content-Length: 200
> 
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 8984 5356 IN IP4 38.106.1.83
> s=SIP Call
> c=IN IP4 38.106.1.83
> t=0 0
> m=audio 17050 RTP/AVP 18
> c=IN IP4 38.106.1.83
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=yes
> 
> 
> 
> ACK sip:88359501351350010 at 38.106.1.83:5060 SIP/2.0
> Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKc583vDrSjNDmQ
> Max-Forwards: 70
> From: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> To: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> CSeq: 15529052 ACK
> Contact: <sip:mod_sofia at 38.106.1.72:5080>
> Content-Length: 0
> 
> 
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236
> T6 To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 INVITE
> Contact: <sip:9501351350010 at 38.106.1.72:5060;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500 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-Type: application/sdp
> Content-Disposition: session
> Content-Length: 200
> Remote-Party-ID: "Outbound Call"
> <sip:88359501351350010 at 38.106.1.72>;party=calling;privacy=off;screen=no
> 
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 8984 5356 IN IP4 38.106.1.83
> s=SIP Call
> c=IN IP4 38.106.1.83
> t=0 0
> m=audio 17050 RTP/AVP 18
> c=IN IP4 38.106.1.83
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=yes
> 
> ---------------------------------------------------------------------------
> -------------------------- ACK sip:9501351350010 at 38.106.1.72 SIP/2.0
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236
> T6 To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> Contact: <sip:8675582928388 at 204.110.11.29:5060>
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 ACK
> Max-Forwards: 70
> User-Agent: ABC-Phone
> Content-Length: 0
> 
> ---------------------------------------------------------------------------
> -------------------------- CANCEL sip:9501351350010 at 38.106.1.72 SIP/2.0
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236
> T6 To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 CANCEL
> Max-Forwards: 70
> Content-Length: 0
> 
> ---------------------------------------------------------------------------
> -------------------------- SIP/2.0 481 Call/Transaction Does Not Exist
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236
> T6 To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 CANCEL
> Content-Length: 0
> 
> 
> ---------------------------------------------------------------------------
> -------------------------- ACK sip:9501351350010 at 38.106.1.72 SIP/2.0
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236
> T6 To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> Contact: <sip:8675582928388 at 204.110.11.29:5060>
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 ACK
> Max-Forwards: 70
> User-Agent: ABC-Phone
> Content-Length: 0
> 
> ---------------------------------------------------------------------------
> -------------------------- SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 38.106.1.83:5060;branch=z9hG4bK2CE48C77
> From: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
> To: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> CSeq: 101 BYE
> Timestamp: 965973271 0.000381
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500 Content-Length: 0
> 
> ---------------------------------------------------------------------------
> -------------------------- SIP/2.0 200 OK
> Via: SIP/2.0/UDP 38.106.1.83:5060;branch=z9hG4bK2CE48C77
> From: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
> To: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> CSeq: 101 BYE
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces
> Content-Length: 0
> 
> ---------------------------------------------------------------------------
> -------------------------- BYE sip:8675582928388 at 204.110.11.29:5060 SIP/2.0
> Via: SIP/2.0/UDP 38.106.1.72;rport;branch=z9hG4bKSj5XgvcHUpm6c
> Max-Forwards: 70
> From: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> To: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236
> T6 Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 15529071 BYE
> Contact: <sip:9501351350010 at 38.106.1.72:5060;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer,
> precondition, path, replaces
> Reason: Q.850;cause=102;text="RECOVERY_ON_TIMER_EXPIRE"
> Content-Length: 0
> 
> 
> ---------------------------------------------------------------------------
> -------------------------- SIP/2.0 200 OK
> Via: SIP/2.0/UDP 38.106.1.72;rport;branch=z9hG4bKSj5XgvcHUpm6c
> From: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> To: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236
> T6 Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 15529071 BYE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Length: 0
> 
> 
> 
> 
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 6335 (20110729) __________
> 
> The message was checked by ESET NOD32 Antivirus.
> 
> http://www.eset.com


From msc at freeswitch.org  Sat Jul 30 00:57:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jul 2011 13:57:58 -0700
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
	<CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
Message-ID: <CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>

Are you registering with the provider or are they registering with you? If
they register with you then a user example is appropriate. If you are
registering with them then all you need is a gateway configured.
-MC

On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <lloydie.t at gmail.com> wrote:

> Sorry, example is not clear to me.
> I don't understand why a user config is relevant to sip registration for a
> provider.
> An example will help me more. Maybe CIDR attribute in a sip_profile gateway
> could help.
>
>
> On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Look at the cidr attribute in the user directory to authenticate by IP:
>> http://wiki.freeswitch.org/wiki/Acl#Users
>>
>> -Steve
>>
>> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com> wrote:
>>
>>> *Hi I need a little help setting up a SIP registration for a provider
>>> that does not use auth.*
>>>
>>> *All I have is info below.*
>>> **
>>>
>>> *
>>> *
>>>
>>>
>>> SBC/Proxy IP:        80.40.150.150:5060
>>>
>>> Authentication:      Trusted IP ? 88.221.85.33
>>>
>>> Assigned DDI:        01869******, 01869******
>>>
>>> DTMF Method:         RFC2833
>>>
>>> Status:              Live
>>>
>>> No. of trunks:       2x
>>>
>>> Session Timer:       1800
>>>
>>> Profile*:*             Generic (35060)
>>>
>>>
>>> Apparently the following is used for *
>>>
>>> [vibe]
>>>
>>> type = friend
>>>
>>> host = 80.40.150.150
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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
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From krice at freeswitch.org  Sat Jul 30 01:02:17 2011
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 29 Jul 2011 16:02:17 -0500
Subject: [Freeswitch-users] Please Help! CANCEL 481 Problem Call dons't
 hunup
In-Reply-To: <5B4F4E33BA91412DB98475E6967DE386@LEODE9798812EB>
Message-ID: <CA588A09.19FBA%krice@freeswitch.org>

Wrong answer... That call is clearly answered before for CANCEL, that?s what
those 200 OKs and ACKs mean

So your end point trying to hang up is getting a 481 as the call is already
in progress and it has ACK?d that fact...

It should instead of sending a CANCEL be sending a bye at that point

K




On 7/29/11 3:14 PM, "Leo" <leo at megapin.com> wrote:

> Hi:
>  
> I have an issue with FS connected to a SIP gateway. After the Freeswitch
> update(FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05 -0500).
>  
> When I receive a call from the SIP Gateway and try to hangup with "CANCEL"
> message the call before he call is answered,
>  
> Our freeswitch replay  SIP 481 Call Transaction does not exist.
> 
> Please help!
>  
> Leo Liu
> ---------------------------------------------------------------
> INVITE sip:9501351350010 at 38.106.1.72 SIP/2.0
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> To: "9501351350010" <sip:9501351350010 at 38.106.1.72>
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> Contact: <sip:8675582928388 at 204.110.11.29:5060>
> CSeq: 237 INVITE
> Max-Forwards: 70
> User-Agent: ABC-Phone
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 141
>  
> v=0
> o=- 34791986 84556339 IN IP4 204.110.11.29
> s=SIP CALL
> c=IN IP4 204.110.11.29
> t=0 0
> m=audio 35974 RTP/AVP 18
> a=rtpmap:18 G729/8000
>  
> ------------------------------------------------------------------------------
> -----------------------
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> To: "9501351350010" <sip:9501351350010 at 38.106.1.72>
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 INVITE
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500
> Content-Length: 0
>  
> ------------------------------------------------------------------------------
> -----------------------
> INVITE sip:88359501351350010 at 38.106.1.83 SIP/2.0
> Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKBvFBUj7NNcQ1B
> Max-Forwards: 69
> From: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> To: <sip:88359501351350010 at 38.106.1.83>
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> CSeq: 15529052 INVITE
> Contact: <sip:mod_sofia at 38.106.1.72:5080>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500
> 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: 141
> X-FS-Support: update_display
> Remote-Party-ID: "calling from"
> <sip:8675582928388 at 38.106.1.72>;party=calling;screen=yes;privacy=off
>  
> v=0
> o=- 34791986 84556339 IN IP4 204.110.11.29
> s=SIP CALL
> c=IN IP4 204.110.11.29
> t=0 0
> m=audio 35974 RTP/AVP 18
> a=rtpmap:18 G729/8000
>  
> ------------------------------------------------------------------------------
> -----------------------
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKBvFBUj7NNcQ1B
> From: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> To: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
> Date: Fri, 11 Aug 2000 05:54:04 GMT
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> Server: Cisco-SIPGateway/IOS-12.x
> CSeq: 15529052 INVITE
> Allow-Events: telephone-event
> Content-Length: 0
>  
> ------------------------------------------------------------------------------
> -----------------------
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKBvFBUj7NNcQ1B
> From: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> To: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
> Date: Fri, 11 Aug 2000 05:54:04 GMT
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> Server: Cisco-SIPGateway/IOS-12.x
> CSeq: 15529052 INVITE
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
> NOTIFY, INFO, UPDATE, REGISTER
> Allow-Events: telephone-event
> Contact: <sip:88359501351350010 at 38.106.1.83:5060>
> Content-Type: application/sdp
> Content-Length: 200
>  
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 8984 5356 IN IP4 38.106.1.83
> s=SIP Call
> c=IN IP4 38.106.1.83
> t=0 0
> m=audio 17050 RTP/AVP 18
> c=IN IP4 38.106.1.83
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=yes
>  
>  
>  
> ACK sip:88359501351350010 at 38.106.1.83:5060 SIP/2.0
> Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKc583vDrSjNDmQ
> Max-Forwards: 70
> From: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> To: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> CSeq: 15529052 ACK
> Contact: <sip:mod_sofia at 38.106.1.72:5080>
> Content-Length: 0
>  
> 
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 INVITE
> Contact: <sip:9501351350010 at 38.106.1.72:5060;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500
> 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-Type: application/sdp
> Content-Disposition: session
> Content-Length: 200
> Remote-Party-ID: "Outbound Call"
> <sip:88359501351350010 at 38.106.1.72>;party=calling;privacy=off;screen=no
>  
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 8984 5356 IN IP4 38.106.1.83
> s=SIP Call
> c=IN IP4 38.106.1.83
> t=0 0
> m=audio 17050 RTP/AVP 18
> c=IN IP4 38.106.1.83
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=yes
>  
> ------------------------------------------------------------------------------
> -----------------------
> ACK sip:9501351350010 at 38.106.1.72 SIP/2.0
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> Contact: <sip:8675582928388 at 204.110.11.29:5060>
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 ACK
> Max-Forwards: 70
> User-Agent: ABC-Phone
> Content-Length: 0
>  
> ------------------------------------------------------------------------------
> -----------------------
> CANCEL sip:9501351350010 at 38.106.1.72 SIP/2.0
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 CANCEL
> Max-Forwards: 70
> Content-Length: 0
>  
> ------------------------------------------------------------------------------
> -----------------------
> SIP/2.0 481 Call/Transaction Does Not Exist
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 CANCEL
> Content-Length: 0
>  
> 
> ------------------------------------------------------------------------------
> -----------------------
> ACK sip:9501351350010 at 38.106.1.72 SIP/2.0
> Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236
> From: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> To: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> Contact: <sip:8675582928388 at 204.110.11.29:5060>
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 237 ACK
> Max-Forwards: 70
> User-Agent: ABC-Phone
> Content-Length: 0
>  
> ------------------------------------------------------------------------------
> -----------------------
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 38.106.1.83:5060;branch=z9hG4bK2CE48C77
> From: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
> To: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> CSeq: 101 BYE
> Timestamp: 965973271 0.000381
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500
> Content-Length: 0
>  
> ------------------------------------------------------------------------------
> -----------------------
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 38.106.1.83:5060;branch=z9hG4bK2CE48C77
> From: <sip:88359501351350010 at 38.106.1.83>;tag=7DAE5EAC-1DD0
> To: "calling from" <sip:8675582928388 at 38.106.1.72>;tag=9r2ScyaN12DpB
> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55
> CSeq: 101 BYE
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Content-Length: 0
>  
> ------------------------------------------------------------------------------
> -----------------------
> BYE sip:8675582928388 at 204.110.11.29:5060 SIP/2.0
> Via: SIP/2.0/UDP 38.106.1.72;rport;branch=z9hG4bKSj5XgvcHUpm6c
> Max-Forwards: 70
> From: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> To: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 15529071 BYE
> Contact: <sip:9501351350010 at 38.106.1.72:5060;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05
> -0500
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Reason: Q.850;cause=102;text="RECOVERY_ON_TIMER_EXPIRE"
> Content-Length: 0
>  
> 
> ------------------------------------------------------------------------------
> -----------------------
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 38.106.1.72;rport;branch=z9hG4bKSj5XgvcHUpm6c
> From: "9501351350010" <sip:9501351350010 at 38.106.1.72>;tag=ZD2XZveKXjHFc
> To: "calling from"
> <sip:8675582928388 at 204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6
> Call-ID: ID236ThB-18461-236-2NeaL at 204.110.11.29
> CSeq: 15529071 BYE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Length: 0
>  
>  
> 
> 
> __________ Information from ESET NOD32 Antivirus, version of virus signature
> database 6335 (20110729) __________
> 
> The message was checked by ESET NOD32 Antivirus.
> 
> http://www.eset.com
> 
> 
> _______________________________________________
> 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 rupa at rupa.com  Sat Jul 30 01:52:05 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 29 Jul 2011 16:52:05 -0500
Subject: [Freeswitch-users] Load Balance Trunks
In-Reply-To: <COL109-W4456F57D6EF8C3739998F5F45F0@phx.gbl>
References: <col109-w22D1131E50A2495C1521C9F4590@phx.gbl>
	<C71A4485-E533-48AE-AC3F-C59D7A1E79CA@ipeva.fr>
	<COL109-W612603C4C7BCC734CBD974F45B0@phx.gbl>
	<BANLkTinT9O5+hzT1YczsmQV5ouL9EQ_2TQ@mail.gmail.com>
	<233898962-1309536985-cardhu_decombobulator_blackberry.rim.net-2106826141-@b13.c4.bise7.blackberry>
	<BANLkTikjYw=0UDyGdipBzadj0_kSoOVS2Q@mail.gmail.com>
	<1637192224-1309538673-cardhu_decombobulator_blackberry.rim.net-1258156530-@b13.c4.bise7.blackberry>
	<BANLkTineHeUHCMpKNJS7msb6xCZzBqmkQg@mail.gmail.com>
	<COL109-W4456F57D6EF8C3739998F5F45F0@phx.gbl>
Message-ID: <CAJw9H-NU3gOO0j6QvqL2dZ8QpfJXi24TM0ix7D0_mpPVHx2+rQ@mail.gmail.com>

I realize this is late, but...

There are so many ways to solve this.  My approach?

Use mod_lcr to define your trunks, mod_lcr will load balance (well
choose random order at each "cost" level).

use mod_nibblebill to keep track of your "account" for each trunk.
Have a cron job at midnight that resets your "balance".  When the
trunk hits 0 it'll disconnect any active calls and any new calls will
not complete on trunks that have a 0 balance.

On Tue, Jul 5, 2011 at 10:21 AM, Abid Saleem <abid_freeswitch at live.com> wrote:
> Hi Micheal, Avi and All,
> Sorry for a little late response as I was away. I have seen quite a few
> questions from you guys, so I am answering them in one email as below.
> 1- How does the provider notify you that each trunk has used its allotted
> time?
> Abid -> They have some counter in their IMS network to count on mins per
> trunk per day and they inform us by email.
> 2- Are these trunks inbound only?
> Abid -> No. All these are Outbound Trunks. We just use them to send outgoing
> calls to our provider.
> 3- What happens when a call extends more than 120 mins on a trunk, would the
> call be disconnected?
> Abid -> The call is not disconnected right away but they send us a
> notification the next day. There is no real-time disconnection.
> 4- And then no more calls that day on that trunk?
> Abid -> Calls do not stop connecting immediately but they keep going.
> Currently their notification process is manual not automatic blocking.
> Please help me if you can. Thanks.
> Regards
> -----------------
> Abid Saleem
> Technical Manager NGN
> Terminus Technologies
> ________________________________
> Date: Fri, 1 Jul 2011 09:49:02 -0700
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Load Balance Trunks
>
> On Fri, Jul 1, 2011 at 9:46 AM, <hmkias at gmail.com> wrote:
>
> Lcr entry could be activated on the the trunks. A cron job could monitor the
> usage and add or remove the trunk for the day.
>
> Are these trunks inbound only?
> -MC
>
> _______________________________________________ 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
> _______________________________________________
> 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
>
>



-- 
-Rupa


From msc at freeswitch.org  Sat Jul 30 01:59:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Jul 2011 14:59:44 -0700
Subject: [Freeswitch-users] Load Balance Trunks
In-Reply-To: <CAJw9H-NU3gOO0j6QvqL2dZ8QpfJXi24TM0ix7D0_mpPVHx2+rQ@mail.gmail.com>
References: <col109-w22D1131E50A2495C1521C9F4590@phx.gbl>
	<C71A4485-E533-48AE-AC3F-C59D7A1E79CA@ipeva.fr>
	<COL109-W612603C4C7BCC734CBD974F45B0@phx.gbl>
	<BANLkTinT9O5+hzT1YczsmQV5ouL9EQ_2TQ@mail.gmail.com>
	<233898962-1309536985-cardhu_decombobulator_blackberry.rim.net-2106826141-@b13.c4.bise7.blackberry>
	<BANLkTikjYw=0UDyGdipBzadj0_kSoOVS2Q@mail.gmail.com>
	<1637192224-1309538673-cardhu_decombobulator_blackberry.rim.net-1258156530-@b13.c4.bise7.blackberry>
	<BANLkTineHeUHCMpKNJS7msb6xCZzBqmkQg@mail.gmail.com>
	<COL109-W4456F57D6EF8C3739998F5F45F0@phx.gbl>
	<CAJw9H-NU3gOO0j6QvqL2dZ8QpfJXi24TM0ix7D0_mpPVHx2+rQ@mail.gmail.com>
Message-ID: <CAKzWOxUCcja1UB5qzqRZG-RL3JWcHk7uKf-WWgZ7PyW2b0x3Bg@mail.gmail.com>

Rupa!!! Glad you chimed in. :)
-MC

On Fri, Jul 29, 2011 at 2:52 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> I realize this is late, but...
>
> There are so many ways to solve this.  My approach?
>
> Use mod_lcr to define your trunks, mod_lcr will load balance (well
> choose random order at each "cost" level).
>
> use mod_nibblebill to keep track of your "account" for each trunk.
> Have a cron job at midnight that resets your "balance".  When the
> trunk hits 0 it'll disconnect any active calls and any new calls will
> not complete on trunks that have a 0 balance.
>
> On Tue, Jul 5, 2011 at 10:21 AM, Abid Saleem <abid_freeswitch at live.com>
> wrote:
> > Hi Micheal, Avi and All,
> > Sorry for a little late response as I was away. I have seen quite a few
> > questions from you guys, so I am answering them in one email as below.
> > 1- How does the provider notify you that each trunk has used its allotted
> > time?
> > Abid -> They have some counter in their IMS network to count on mins per
> > trunk per day and they inform us by email.
> > 2- Are these trunks inbound only?
> > Abid -> No. All these are Outbound Trunks. We just use them to send
> outgoing
> > calls to our provider.
> > 3- What happens when a call extends more than 120 mins on a trunk, would
> the
> > call be disconnected?
> > Abid -> The call is not disconnected right away but they send us a
> > notification the next day. There is no real-time disconnection.
> > 4- And then no more calls that day on that trunk?
> > Abid -> Calls do not stop connecting immediately but they keep going.
> > Currently their notification process is manual not automatic blocking.
> > Please help me if you can. Thanks.
> > Regards
> > -----------------
> > Abid Saleem
> > Technical Manager NGN
> > Terminus Technologies
> > ________________________________
> > Date: Fri, 1 Jul 2011 09:49:02 -0700
> > From: msc at freeswitch.org
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Load Balance Trunks
> >
> > On Fri, Jul 1, 2011 at 9:46 AM, <hmkias at gmail.com> wrote:
> >
> > Lcr entry could be activated on the the trunks. A cron job could monitor
> the
> > usage and add or remove the trunk for the day.
> >
> > Are these trunks inbound only?
> > -MC
> >
> > _______________________________________________ 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
> > _______________________________________________
> > 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
> >
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> 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 rupa at rupa.com  Sat Jul 30 02:10:33 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 29 Jul 2011 17:10:33 -0500
Subject: [Freeswitch-users] Load Balance Trunks
In-Reply-To: <CAKzWOxUCcja1UB5qzqRZG-RL3JWcHk7uKf-WWgZ7PyW2b0x3Bg@mail.gmail.com>
References: <col109-w22D1131E50A2495C1521C9F4590@phx.gbl>
	<C71A4485-E533-48AE-AC3F-C59D7A1E79CA@ipeva.fr>
	<COL109-W612603C4C7BCC734CBD974F45B0@phx.gbl>
	<BANLkTinT9O5+hzT1YczsmQV5ouL9EQ_2TQ@mail.gmail.com>
	<233898962-1309536985-cardhu_decombobulator_blackberry.rim.net-2106826141-@b13.c4.bise7.blackberry>
	<BANLkTikjYw=0UDyGdipBzadj0_kSoOVS2Q@mail.gmail.com>
	<1637192224-1309538673-cardhu_decombobulator_blackberry.rim.net-1258156530-@b13.c4.bise7.blackberry>
	<BANLkTineHeUHCMpKNJS7msb6xCZzBqmkQg@mail.gmail.com>
	<COL109-W4456F57D6EF8C3739998F5F45F0@phx.gbl>
	<CAJw9H-NU3gOO0j6QvqL2dZ8QpfJXi24TM0ix7D0_mpPVHx2+rQ@mail.gmail.com>
	<CAKzWOxUCcja1UB5qzqRZG-RL3JWcHk7uKf-WWgZ7PyW2b0x3Bg@mail.gmail.com>
Message-ID: <CAJw9H-MV=cphGJJMy2Vj8-cgeFkUb5os22-fBzvxZhmwOL1hMg@mail.gmail.com>

On Fri, Jul 29, 2011 at 4:59 PM, Michael Collins <msc at freeswitch.org> wrote:
> Rupa!!! Glad you chimed in. :)
> -MC
>

MC!!!  Just catching up on the list.  Been busy as heck but will drop
by irc eventually. Hope ya'll have fun at ClueCon

-- 
-Rupa


From lloydie.t at gmail.com  Sat Jul 30 07:34:10 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Sat, 30 Jul 2011 04:34:10 +0100
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
	<CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
	<CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>
Message-ID: <CAE=3qOacZh50iepxJKJD5UvtesxZAsN7RnPKNe1zdLO9VmFsuA@mail.gmail.com>

I am registering with a them. I could not find suitable example in
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples which

On 29 July 2011 21:57, Michael Collins <msc at freeswitch.org> wrote:

> Are you registering with the provider or are they registering with you? If
> they register with you then a user example is appropriate. If you are
> registering with them then all you need is a gateway configured.
> -MC
>
>
> On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <lloydie.t at gmail.com> wrote:
>
>> Sorry, example is not clear to me.
>> I don't understand why a user config is relevant to sip registration for a
>> provider.
>> An example will help me more. Maybe CIDR attribute in a sip_profile
>> gateway could help.
>>
>>
>> On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> Look at the cidr attribute in the user directory to authenticate by IP:
>>> http://wiki.freeswitch.org/wiki/Acl#Users
>>>
>>> -Steve
>>>
>>> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>
>>>> *Hi I need a little help setting up a SIP registration for a provider
>>>> that does not use auth.*
>>>>
>>>> *All I have is info below.*
>>>> **
>>>>
>>>> *
>>>> *
>>>>
>>>>
>>>> SBC/Proxy IP:        80.40.150.150:5060
>>>>
>>>> Authentication:      Trusted IP ? 88.221.85.33
>>>>
>>>> Assigned DDI:        01869******, 01869******
>>>>
>>>> DTMF Method:         RFC2833
>>>>
>>>> Status:              Live
>>>>
>>>> No. of trunks:       2x
>>>>
>>>> Session Timer:       1800
>>>>
>>>> Profile*:*             Generic (35060)
>>>>
>>>>
>>>> Apparently the following is used for *
>>>>
>>>> [vibe]
>>>>
>>>> type = friend
>>>>
>>>> host = 80.40.150.150
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 lloydie.t at gmail.com  Sat Jul 30 07:59:09 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Sat, 30 Jul 2011 04:59:09 +0100
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAE=3qOacZh50iepxJKJD5UvtesxZAsN7RnPKNe1zdLO9VmFsuA@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
	<CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
	<CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>
	<CAE=3qOacZh50iepxJKJD5UvtesxZAsN7RnPKNe1zdLO9VmFsuA@mail.gmail.com>
Message-ID: <CAE=3qOZsNuiimFBZJTf4SbPRA8bRq_hJfPUunEnT8rxZj3qTnA@mail.gmail.com>

Hi, dialling in produces the following error.

2011-07-30 04:56:07.818936 [DEBUG] sofia.c:6517 IP 80.40.150.150 Rejected by
acl "domains". Falling back to Digest auth.
2011-07-30 04:56:07.826367 [WARNING] sofia_reg.c:1246 SIP auth challenge
(INVITE) on sofia profile 'internal' for [01869******@172.16.XXX.XXX] from
ip 80.40.150.150


On 30 July 2011 04:34, lloyd thomas <lloydie.t at gmail.com> wrote:

> I am registering with a them. I could not find suitable example in
> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples which
>
>
> On 29 July 2011 21:57, Michael Collins <msc at freeswitch.org> wrote:
>
>> Are you registering with the provider or are they registering with you? If
>> they register with you then a user example is appropriate. If you are
>> registering with them then all you need is a gateway configured.
>> -MC
>>
>>
>> On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>
>>> Sorry, example is not clear to me.
>>> I don't understand why a user config is relevant to sip registration for
>>> a provider.
>>> An example will help me more. Maybe CIDR attribute in a sip_profile
>>> gateway could help.
>>>
>>>
>>> On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>>> Look at the cidr attribute in the user directory to authenticate by IP:
>>>> http://wiki.freeswitch.org/wiki/Acl#Users
>>>>
>>>> -Steve
>>>>
>>>> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>
>>>>> *Hi I need a little help setting up a SIP registration for a provider
>>>>> that does not use auth.*
>>>>>
>>>>> *All I have is info below.*
>>>>> **
>>>>>
>>>>> *
>>>>> *
>>>>>
>>>>>
>>>>> SBC/Proxy IP:        80.40.150.150:5060
>>>>>
>>>>> Authentication:      Trusted IP ? 88.221.85.33
>>>>>
>>>>> Assigned DDI:        01869******, 01869******
>>>>>
>>>>> DTMF Method:         RFC2833
>>>>>
>>>>> Status:              Live
>>>>>
>>>>> No. of trunks:       2x
>>>>>
>>>>> Session Timer:       1800
>>>>>
>>>>> Profile*:*             Generic (35060)
>>>>>
>>>>>
>>>>> Apparently the following is used for *
>>>>>
>>>>> [vibe]
>>>>>
>>>>> type = friend
>>>>>
>>>>> host = 80.40.150.150
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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 lloydie.t at gmail.com  Sat Jul 30 08:31:09 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Sat, 30 Jul 2011 05:31:09 +0100
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAE=3qOZsNuiimFBZJTf4SbPRA8bRq_hJfPUunEnT8rxZj3qTnA@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
	<CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
	<CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>
	<CAE=3qOacZh50iepxJKJD5UvtesxZAsN7RnPKNe1zdLO9VmFsuA@mail.gmail.com>
	<CAE=3qOZsNuiimFBZJTf4SbPRA8bRq_hJfPUunEnT8rxZj3qTnA@mail.gmail.com>
Message-ID: <CAE=3qObVuLOhp2GD29Da-GfzB_qkbkyDP4kPmRxjdbJi7XeirQ@mail.gmail.com>

OK Inbound working with:

  <!--2circles -->
    <list name="domains" default="allow">
      <node type="allow" cidr="80.40.150.150/32"/>
    </list>

Just need to sort outbound.

On 30 July 2011 04:59, lloyd thomas <lloydie.t at gmail.com> wrote:

> Hi, dialling in produces the following error.
>
> 2011-07-30 04:56:07.818936 [DEBUG] sofia.c:6517 IP 80.40.150.150 Rejected
> by acl "domains". Falling back to Digest auth.
> 2011-07-30 04:56:07.826367 [WARNING] sofia_reg.c:1246 SIP auth challenge
> (INVITE) on sofia profile 'internal' for [01869******@172.16.XXX.XXX] from
> ip 80.40.150.150
>
>
>
> On 30 July 2011 04:34, lloyd thomas <lloydie.t at gmail.com> wrote:
>
>> I am registering with a them. I could not find suitable example in
>> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples which
>>
>>
>> On 29 July 2011 21:57, Michael Collins <msc at freeswitch.org> wrote:
>>
>>> Are you registering with the provider or are they registering with you?
>>> If they register with you then a user example is appropriate. If you are
>>> registering with them then all you need is a gateway configured.
>>> -MC
>>>
>>>
>>> On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>
>>>> Sorry, example is not clear to me.
>>>> I don't understand why a user config is relevant to sip registration for
>>>> a provider.
>>>> An example will help me more. Maybe CIDR attribute in a sip_profile
>>>> gateway could help.
>>>>
>>>>
>>>> On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com> wrote:
>>>>
>>>>> Look at the cidr attribute in the user directory to authenticate by IP:
>>>>> http://wiki.freeswitch.org/wiki/Acl#Users
>>>>>
>>>>> -Steve
>>>>>
>>>>> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>
>>>>>> *Hi I need a little help setting up a SIP registration for a provider
>>>>>> that does not use auth.*
>>>>>>
>>>>>> *All I have is info below.*
>>>>>> **
>>>>>>
>>>>>> *
>>>>>> *
>>>>>>
>>>>>>
>>>>>> SBC/Proxy IP:        80.40.150.150:5060
>>>>>>
>>>>>> Authentication:      Trusted IP ? 88.221.85.33
>>>>>>
>>>>>> Assigned DDI:        01869******, 01869******
>>>>>>
>>>>>> DTMF Method:         RFC2833
>>>>>>
>>>>>> Status:              Live
>>>>>>
>>>>>> No. of trunks:       2x
>>>>>>
>>>>>> Session Timer:       1800
>>>>>>
>>>>>> Profile*:*             Generic (35060)
>>>>>>
>>>>>>
>>>>>> Apparently the following is used for *
>>>>>>
>>>>>> [vibe]
>>>>>>
>>>>>> type = friend
>>>>>>
>>>>>> host = 80.40.150.150
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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 member at linkedin.com  Sat Jul 30 23:14:45 2011
From: member at linkedin.com (srinivasulareddy kondreddy via LinkedIn)
Date: Sat, 30 Jul 2011 19:14:45 +0000 (UTC)
Subject: [Freeswitch-users] Invitation to connect on LinkedIn
Message-ID: <942927269.1135145.1312053285281.JavaMail.app@ela4-bed80.prod>

LinkedIn
------------




    srinivasulareddy kondreddy requested to add you as a connection on LinkedIn:
  
------------------------------------------

Zohair,

I'd like to add you to my professional network on LinkedIn.

- srinivasulareddy

Accept invitation from srinivasulareddy kondreddy
http://www.linkedin.com/e/kwhdv8-gqqyne72-j/GrULditvq6UNXRBIaIGnrCzzv0aYgRNIau2nD4Rpg0D0u0BXjFSyiLl/blk/I1565027291_3/1BpC5vrmRLoRZcjkkZt5YCpnlOt3RApnhMpmdzgmhxrSNBszYPnP4VczsOc3kSdj59bQ4Nd59piSoObPgTej4OcjkMdPcLrCBxbOYWrSlI/EML_comm_afe/

View invitation from srinivasulareddy kondreddy
http://www.linkedin.com/e/kwhdv8-gqqyne72-j/GrULditvq6UNXRBIaIGnrCzzv0aYgRNIau2nD4Rpg0D0u0BXjFSyiLl/blk/I1565027291_3/3dvcjAOdP8MdjoRckALqnpPbOYWrSlI/svi/ 
------------------------------------------

DID YOU KNOW you can showcase your professional knowledge on LinkedIn to receive job/consulting offers and enhance your professional reputation? Posting replies to questions on LinkedIn Answers puts you in front of the world's professional community.
http://www.linkedin.com/e/kwhdv8-gqqyne72-j/abq/inv-24/
 
-- 
(c) 2011, LinkedIn Corporation
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From moises.silva at gmail.com  Sun Jul 31 01:43:14 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 30 Jul 2011 17:43:14 -0400
Subject: [Freeswitch-users] freetdm api - ftdm_channel_read()
In-Reply-To: <CAAqVCyTi9m0+JPhjt_sLLkEa97ijFLmETjPoFRUcEQJRmQsN8w@mail.gmail.com>
References: <CAAqVCyTi9m0+JPhjt_sLLkEa97ijFLmETjPoFRUcEQJRmQsN8w@mail.gmail.com>
Message-ID: <CAA4nhyDc9i=raCOk1mzbJ3vVJYSAAKHGwT0b7wyMbaZrA+Wz=Q@mail.gmail.com>

Hi Juraj,

see your questions answered below ...

On Wed, Jul 27, 2011 at 1:21 PM, Juraj Fabo <juraj.fabo at gmail.com> wrote:
> According to return value of sangoma_get_rx_queue_sz() the default rx
> queue size is 10.

That is correct. And you can change that in the FreeTDM wanpipe.conf
configuration to set a different default or by using
FTDM_COMMAND_SET_RX/TX_QUEUE_SIZE

> How can an application do this reading of a channel by the means of
> the freetdm api in more nice way?
> I did not find a better way yet, than to read it in the loop.
> Any way to explicitly read the number of available frames?

I just committed new code to allow access to the internal iostats
structure (which contains the size of the queue and number of elements
on it). See src/include/freetdm.h ftdm_channel_iostats_t, bear in mind
you must enable the feature using FTDM_COMMAND_SWITCH_IOSTATS, then
you can retrieve the stats with FTDM_COMMAND_GET_IOSTATS. I added an
example of usage in mod_freetdm/mod_freetdm.c

Do "git pull" to get it.

> Actually, why the ftdm_channel_read() does not read them ALL at once
> and sets the number of read bytes via *datalen output parameter?
> (assuming provided *data buffer is large enough)
> In my tests, always 160 bytes were returned in one particular read.

Did you verify this by providing bigger buffer and datalen?

The reason is that you typically want to handle voice as soon as is
available, otherwise you add delay in your audio path, therefore you
never leave your voice to accumulate in the driver's buffer's.
FreeSWITCH and all other users of the FreeTDM API work this way.

In the future you might want consider asking this questions in the
freeswitch-dev mailing list to have better chances of a prompt
response (as long as the question involves C development with
FreeSWITCH/FreeTDM internal APIs).

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 fieldpeak at gmail.com  Sun Jul 31 12:46:18 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sun, 31 Jul 2011 16:46:18 +0800
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
	FreeRadius server
Message-ID: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>

Hello Gurus,

i met a issue when using
mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to works
with freeradius server+mysql for AAA, the details is below, Could
anyone give any hints, Thanks in advance.

i setup a dial plan "unitest_rad-ANI-auth" as wiki above, however,
when i dialed 601 to trigger the dial plan, the console show errors,
it looks "h323-conf-id" is not in the directory, then i tried to add
this attribute to the dictionary, however, it does not help, in the
wiki, it mentioned the rad_auth.conf.xml contains <param
name="dictionary"
value="/usr/local/etc/radiusclient/dictionary.all"/>, however i did
not find the file "dictionary.all" at that directory, so i use
dictionary. BTW, the freeradius server + mysql works well.

console errors:

EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in , in
38516060333, in 003282, out AUTH_RESULT)
2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate initial
structure.
2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed configuration.
2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set authserver
:= 127.0.0.1:1812:gateway.
2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set dictionary
:= /usr/local/etc/radiusclient/dictionary.
2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile :=
/var/run/radius.seq.
2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile :=
/usr/local/etc/radiusclient/port-id-map.
2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set default_realm := .
2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_timeout := 3.
2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_retries := 2.
2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_deadtime := 0.
2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set bindaddr := *.
2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius:
User-Name: 38516060333
2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius:
User-Password: 003282
2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius:
Called-station-Id is empty, ignoring...
2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
attribute: h323-conf-id
2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown attribute:
key:h323-conf-id, not found in dictionary
2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort sending
radius packet.
2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error occured
during RADIUS Authentication(RC=-1)
2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error occured
during radius authorization.
EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  AUTH_RESULT=
EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  billing_model=)
2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  billing_model=
EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  credit_amount=)
2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  credit_amount=
EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  currency=)
2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  preffered_lang=)
2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  preffered_lang=

added below in the dictionary(/usr/local/etc/radiusclient/dictionary):

ATTRIBUTE	h323-conf-id		1008	string


dial plan:
<extension name="unitest_rad-ANI-auth">
    <condition field="destination_number" expression="^601$">
      <action application="log" data="INFO  Before Auth "/>

      <action inline="true" application="set"
data="CALLID=h323-conf-id=${uuid}"/>
      <action inline="true" application="set"
data="SERVICENUM=h323-prompt-id=${destination_number}"/>
      <action inline="true" application="set"
data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
 <!--      <action inline="true" application="set"
data="CALLINGNUMBER=${caller_id_number}"/> -->
      <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/>
      <action inline="true" application="set" data="USERNAME=38516060333"/>
 <!--      <action inline="true" application="set" data="USERNAME=209354"/> -->
      <action inline="true" application="set" data="PASSWD=003282"/>
 <!--      <action inline="true" application="set"
data="DIALED_NUMBER=16094191500"/>  -->

      <action application="sleep" data="2000"/>
      <action application="auth_function" data="in ${DIALED_NUMBER},
in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>


      <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
      <action application="log" data="INFO  billing_model=${billing_model}"/>
      <action application="log" data="INFO  credit_amount=${credit_amount}"/>
      <action application="log" data="INFO  currency=${currency}"/>
      <action application="log" data="INFO  preffered_lang=${preffered_lang}"/>
      <action application="log" data="INFO  credit_time=${credit_time}"/>
      <action application="log" data="INFO
h323_ivr_duration=${h323_ivr_duration}"/>
      <action application="log" data="INFO  return_code=${return_code}"/>
      <!-- <action application="execute_extension" data="AUTH XML default"/> -->
    </condition>
  </extension>

 radius_cdr.conf.xml:
 <configuration name="radius_cdr.conf" description="RADIUS CDR Configuration">

        <settings>

                <!-- location of the radius dictionary files -->

                <param name="dictionary"
value="/usr/local/freeswitch/conf/radius/dictionary"/>

                <!-- number of retries for each server -->

                <param name="radius_retries" value="3"/>

                <!-- number of seconds to wait between retries -->

                <param name="radius_timeout" value="5"/>

                <!-- accounting servers, up to 8 allowed -->

                <!-- value is "host:port:secret", port is optional -->

                <!-- use IP ADDRESSES, not hostnames -->

                <param name="acct_server" value="127.0.0.1:1813:testing123"/>


        </settings>

</configuration>

  the FS version:
  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400)

  Regards,
  Charles


From fieldpeak at gmail.com  Sun Jul 31 19:33:53 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sun, 31 Jul 2011 23:33:53 +0800
Subject: [Freeswitch-users] How to realize FS auth user (for REGISTERATION)
	using standard Radius interface
Message-ID: <CAOTmbMwoYZjHtAJnT8igORaBfpy8qnvyLn3n2i4m3e4mqAceCg@mail.gmail.com>

Dear frindes,

i would to like to have all of users data in mysql db and autenticate
user(for registeration) thru standard Radius interface e.g. FreeRadius,
meaning when SIP user send REGISTRATION to FS, FS will send Radius request
to FreeRadius, then FS auth user OK or NOK according to the response from
FreeRadius.

i read the wiki, there are some modules related, e.g. Mod_rad_auth,
Mod_xml_radius, Mod_radius_cdr, however, no one provides a definite example
for my case, could anyone help advise an example or any hint how to make it
done,
BTW, i can thought about using mod_xml_curl, however i don't want to use it
if there is a better one.

Thanks!

Regards,
Charles
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